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2313 Commits

Author SHA1 Message Date
Chris Robinson fd83e006ec Release 1.15 2012-12-04 13:49:16 -08:00
Chris Robinson e51574c764 Trace the list of supported backends 2012-12-04 13:46:51 -08:00
Chris Robinson 893a1bedbc Use +/-90 degrees for stereo sources with non-HRTF stereo output. 2012-12-04 05:30:06 -08:00
Chris Robinson 4621a7b426 Add support for float32 output to CoreAudio 2012-12-03 11:50:02 -08:00
Chris Robinson 4808313e70 Use the correct multiplier for int32 output with CoreAudio 2012-12-03 10:44:02 -08:00
Chris Robinson 1fb9311d82 Lock the device before calling aluHandleDisconnect
PulseAudio causes an assert if being relocked inside a callback on the worker
thread, where aluHandleDisconnect is called. We can assume it's already locked
there, so just make sure the device is locked before being calling it.
2012-12-02 11:30:23 -08:00
Chris Robinson dd34daed42 Trace the opened device name in alcCaptureOpenDevice 2012-12-02 06:37:34 -08:00
Chris Robinson 830428fb06 Use WARN when implicitly deleting resources with the context 2012-12-02 01:17:26 -08:00
Chris Robinson 0ca7ab2cf8 Explicitly check for malloc.h 2012-11-25 19:20:35 -08:00
Chris Robinson a0c39989f5 malloc.h isn't standard, and is only needed for _aligned_malloc 2012-11-25 02:33:20 -08:00
Chris Robinson 209d09ae13 Avoid locking PulseAudio's mainloop when starting and stopping capture
The mainloop is already locked by the caller, and double-locking interferes
with the subsequent wait.
2012-11-19 16:50:03 -08:00
Chris Robinson 0c030f7d9c Fix a comment describing the sinc function 2012-11-18 19:02:50 -08:00
Chris Robinson 2d800e99e2 Fix off-by-one error in the makehrtf dither and add a resample option
Also fixes some size_t type warnings
2012-11-18 09:29:58 -08:00
Chris Robinson 0518ecca14 Avoid storing the SSE state if it's not supported 2012-11-13 15:20:36 -08:00
Chris Robinson 43b9027611 Return the integer from fastf2i with MSVC 2012-11-13 15:12:46 -08:00
Chris Robinson 40ea4f4b46 Use int instead of long for the IMA4 tables 2012-11-10 03:40:36 -08:00
Chris Robinson 4f624cb734 Load Int and UInt samples as float with alBufferData
This shouldn't really ever happen, as there's no 32-bit (u)int formats
alBufferData can take.
2012-11-10 03:36:37 -08:00
Chris Robinson b03013d6a7 Specify that the config file being installed is a sample 2012-11-06 03:52:41 -08:00
Chris Robinson 9e79d144de Install the config sample by default 2012-11-05 04:20:15 -08:00
Chris Robinson 40b3f60cde Install the config example into share/openal instead of /etc/openal
/etc/openal/alsoft.conf is still used for configuration, only the sample's
install placement is changed.
2012-11-05 04:14:05 -08:00
Chris Robinson 2b27d243ca Use stdout to print alstream information 2012-11-04 22:37:10 -08:00
Chris Robinson 2f07c46e20 Include standard headers before conditional ones 2012-11-04 22:36:34 -08:00
Chris Robinson 3da1b540fd Use the stored buffer metrics for the mmdevapi buffer size 2012-11-04 22:35:35 -08:00
Chris Robinson 92dde81296 Add rudimentary latency tracking for mmdevapi
This won't be as granular as it could be, since it only updates when the wakeup
event trips (which may or may not happen more often than OpenAL's mix updates).
A more correct method would be to query GetCurrentPadding directly, but that
would require sending a message to the processing thread and waiting for a
reply, since we can't guarantee COM on the calling thread.
2012-11-04 08:35:39 -08:00
Chris Robinson e2368eb960 Use an array to specify the offset for each channel of the device buffer
This effectively inverts the DevChannels array
2012-11-04 04:41:11 -08:00
Chris Robinson 3e71a7cb94 Rename some struct fields for DirectSound 2012-11-04 01:18:39 -08:00
Chris Robinson b92ad830ef Avoid recreating the DSound primary buffer 2012-11-04 00:37:31 -07:00
Chris Robinson fc7adccd6f Fix retrieving the source's write offset 2012-11-01 23:41:18 -07:00
Chris Robinson 97bf718d0e Pass the offset latency properties to the set handler 2012-11-01 18:35:20 -07:00
Chris Robinson eb6360e1f8 Set the proper error when trying to set AL_SAMPLE_OFFSET_LATENCY_SOFT or AL_SEC_OFFSET_LATENCY_SOFT 2012-11-01 00:16:44 -07:00
Chris Robinson 41b1ba146a Add an example to get and show the source latency 2012-10-31 23:53:10 -07:00
Chris Robinson 6450d0cbca Avoid prefixing function pointers in the streaming example 2012-10-31 22:38:27 -07:00
Chris Robinson 4e281350a0 Print the name of the device when opening it in the example apps 2012-10-31 17:16:56 -07:00
Chris Robinson e868dad61c Make a decimal value a float type 2012-10-31 06:58:16 -07:00
Chris Robinson 4a3d36a176 Finalize AL_SOFT_source_latency 2012-10-31 05:09:42 -07:00
Chris Robinson f3dddb6e63 Add an example program showing how to apply reverb to a source 2012-10-31 04:04:49 -07:00
Chris Robinson 3c3e3b123b Add a couple comments to the code 2012-10-26 20:11:59 -07:00
Chris Robinson f02a9935c6 Avoid branching when clamping and converting float samples to integer 2012-10-26 15:31:14 -07:00
Chris Robinson 3711735ed9 Add the AL subdir to the list of include flags in the pkg-config file
This allows apps to include "al.h", "alc.h", etc, like with Creative's SDK and
CMake's FindOpenAL module, or as "AL/al.h" and "AL/alc.h" like before.
2012-10-26 00:02:39 -07:00
Chris Robinson 2c802cc10a Reorder and comment some function in alhelpers.c 2012-10-25 23:53:46 -07:00
Chris Robinson e2e502970c Move some math functions to where they're used 2012-10-25 17:24:22 -07:00
Chris Robinson 404d601666 Add a couple missing RESTRICTs 2012-10-25 17:02:21 -07:00
Chris Robinson 252fc40f78 Fix some types 2012-10-25 15:37:52 -07:00
Chris Robinson fb21a73b4e Add the RESTRICT keyword to a function pointer definition's parameters 2012-10-25 14:52:09 -07:00
Chris Robinson 138a4c6d92 Add explicit casts for MSVC 2012-10-25 14:46:27 -07:00
Chris Robinson c802bfccef Remove the float math wrapper functions 2012-10-25 14:16:35 -07:00
Chris Robinson fae4959db3 Add retrieval of missing source properties 2012-10-21 11:36:27 -07:00
Chris Robinson 8a67e5cfa2 Use an asterisk to denote a requested format option in the trace 2012-10-16 07:46:04 -07:00
Chris Robinson 617def3ef0 Require an alignment declaration for SSE support 2012-10-15 02:23:08 -07:00
Chris Robinson 4f5aafe59f Put the HRTF DirectParams into an anonymous struct 2012-10-15 01:37:37 -07:00
Chris Robinson a14383d6cc Constify the direct and send parameters given to the mixer 2012-10-15 01:31:58 -07:00
Chris Robinson fe1903fb56 Use a separate step to deinterlace reverb samples 2012-10-14 11:55:03 -07:00
Chris Robinson f0d27e5f91 Remove the unused Device parameter 2012-10-14 11:29:28 -07:00
Chris Robinson a852669dce Store the output buffers in the DirectParams struct 2012-10-14 11:21:52 -07:00
Chris Robinson 3240894cd5 Remove the now-unused Source parameter from the DryMix methods 2012-10-14 11:04:36 -07:00
Chris Robinson 0b679167c9 Store some more HRTF info in the DirectParams struct 2012-10-14 10:57:11 -07:00
Chris Robinson 5ac0a93fba Mark some parameters as const and/or RESTRICT 2012-10-14 10:27:15 -07:00
Chris Robinson f7655d44a2 Move alhelpers and alffmpeg code to a common sub-directory 2012-10-14 01:55:39 -07:00
Chris Robinson 38e6bfb702 Use a helper method to convert i64 values to the proper types 2012-10-14 01:41:49 -07:00
Chris Robinson 20ede6de59 Check i64 property ranges before passing them to the int handlers
Note that some properties (e.g. AL_BUFFER, AL_DIRECT_FILTER) actually take
unsigned int values, and so are checked against the unsigned range even though
they eventually get casted to an int. The int handler casts them back as
needed.
2012-10-14 00:55:46 -07:00
Chris Robinson 5c70e6b9ab Don't add the dry gain to the mixing matrix 2012-10-13 11:10:58 -07:00
Chris Robinson 9bd3b88a58 Silence an implicit int-to-float conversion warning on MSVC 2012-10-13 11:00:43 -07:00
Chris Robinson e60281c927 Don't use a loop to determine the next power of 2 2012-10-13 08:58:48 -07:00
Chris Robinson 01136ecd51 List the in-progress AL_SOFT_source_latency extension 2012-10-13 00:59:46 -07:00
Chris Robinson 1212523470 Implement the double and int64 source setters
Note that currently the int64 setters do not range check before being passed to
the int setters, erroneously chopping off the upper bits.
2012-10-13 00:56:39 -07:00
Chris Robinson 73105b2d4b Fix copying the listener Up vector 2012-10-12 23:35:00 -07:00
Chris Robinson 06ab9cf70e Use the enum to set the default distance model 2012-10-12 23:28:51 -07:00
Chris Robinson a3d4868ef4 Initialize the listener velocity parameter 2012-10-12 08:01:02 -07:00
Chris Robinson 815b03855b Apply the listener matrix to the listener velocity once 2012-10-12 07:56:37 -07:00
Chris Robinson 6750bbb1d7 Avoid needlessly copying the listener matrix 2012-10-12 07:45:20 -07:00
Chris Robinson 43c8c4fa29 Apply the listener position to the transformation matrix 2012-10-12 07:38:29 -07:00
Chris Robinson 25b8a95987 Check the distance against epsilon to determine if it matches 2012-10-12 07:10:51 -07:00
Chris Robinson 99fcd6f816 Properly scale wide-stereo sources 2012-10-10 04:08:35 -07:00
Chris Robinson 513c6db0d5 Avoid mixing to a send slot if the gain is too low 2012-10-10 03:52:45 -07:00
Chris Robinson 1b7f01b7cc The max step is no longer dependent on the channel count 2012-10-10 03:40:01 -07:00
Chris Robinson a4f21aaa64 Set Hrtf.Moving after setting the source params instead of after mixing 2012-10-09 06:54:21 -07:00
Chris Robinson 39bc2ba65c Build the listener matrix separately 2012-10-09 06:19:36 -07:00
Chris Robinson d598f82722 Don't include alListener.h from alMain.h 2012-10-09 04:48:12 -07:00
Chris Robinson c22408cb2a Store a pointer to the listener in the context 2012-10-09 04:44:27 -07:00
Chris Robinson 8c3419ed48 Define RESTRICT in config.h instead of alMain.h 2012-10-09 01:46:02 -07:00
Chris Robinson a98361ebbe Don't assume x86 or x64 to use IsProcessorFeaturePresent 2012-10-08 00:43:51 -07:00
Chris Robinson 999f0783c5 Fix a type conversion warning 2012-10-08 00:32:35 -07:00
Chris Robinson 2a6dd2f243 Fix a type mismatch 2012-10-07 08:19:09 -07:00
Chris Robinson f3349221c6 Add missing include 2012-10-07 08:18:53 -07:00
Chris Robinson 65c7c468cc Use IsProcessorFeaturePresent to detect SSE on Windows 2012-10-07 08:18:22 -07:00
Chris Robinson ebdf636759 Revert "Use __cpuid from intrin.h in Windows when available"
This reverts commit 6b870714a9.
2012-10-07 08:08:11 -07:00
Chris Robinson 6b870714a9 Use __cpuid from intrin.h in Windows when available 2012-10-07 05:36:25 -07:00
Chris Robinson 708f3ccc3d Add trace, warn, and error markers to logged output 2012-10-07 04:31:31 -07:00
Chris Robinson 84f4afa5f2 Lock the device using the pulseaudio main loop 2012-10-07 02:25:59 -07:00
Chris Robinson 991e21a712 Avoid unnecessarily locking the list lock 2012-10-07 00:13:23 -07:00
Chris Robinson 1a9cf7ab80 Shorten format trace messages 2012-10-06 23:32:36 -07:00
Chris Robinson 9501f1cb9e Define WIN32_LEAN_AND_MEAN when including windows.h in alMain.h 2012-10-06 21:36:15 -07:00
Chris Robinson d52cb3708b Include windows.h before cpuid.h to avoid __cpuid clashes 2012-10-06 21:30:03 -07:00
Chris Robinson fa8d374613 Rename a couple functions
Since they don't use the stack anymore.
2012-10-05 06:44:55 -07:00
Chris Robinson 0d26b0dcc2 Put temporary storage in the device instead of on the stack 2012-10-05 06:42:26 -07:00
Chris Robinson a6287fd407 Add a special resampler for matching sample rates 2012-10-05 06:03:19 -07:00
Chris Robinson 16bdf79d4c Round up the device's default slot alignment 2012-10-05 00:32:41 -07:00
Chris Robinson 0b47844606 Fix potential buffer overflow with temp sample space 2012-10-05 00:27:29 -07:00
Chris Robinson 468759e6ab Scale and round NumUpdates when PulseAudio changes the sample rate 2012-10-04 23:36:59 -07:00
Chris Robinson f138774d78 Avoid void* arithmetic 2012-10-04 06:16:16 -07:00
Chris Robinson 068da3fe83 Make sure the output buffer pointer is updated in case multiple iterations are needed 2012-10-03 02:35:43 -07:00
Chris Robinson 29b11db13f Update the example to use avcodec_decode_audio4 2012-10-03 02:01:52 -07:00
Chris Robinson dca6254b35 Update some ffmpeg functions 2012-10-03 00:29:08 -07:00
Chris Robinson cccec40e87 Fix some signed/unsigned comparisons 2012-10-02 18:55:39 -07:00
Chris Robinson b5d9d8e005 Reset maxlength if PulseAudio updates the playback rate 2012-10-02 17:01:35 -07:00
Chris Robinson 825c5b5282 Use lrintf to fast convert floats to ints when possible 2012-09-28 04:20:55 -07:00
Chris Robinson 25ca179389 Decrease the default BUFFERSIZE to 2048 2012-09-28 03:30:57 -07:00
Chris Robinson 8bd0592e3f Combine early reflections and late reverb in VerbPass 2012-09-27 17:30:35 -07:00
Chris Robinson 06a0d1b86e Use an unsigned type for the sample position fraction 2012-09-27 13:52:45 -07:00
Chris Robinson 41b2c2f3bb Remove an unneeded parameter from the resampler 2012-09-27 02:46:15 -07:00
Chris Robinson 68faef5b84 Ensure the device's default effect slot is properly aligned 2012-09-26 18:21:03 -07:00
Chris Robinson abf1309fb6 Update a comment 2012-09-26 18:04:22 -07:00
Chris Robinson d3891af36e Ensure effect slots are aligned 2012-09-26 18:00:38 -07:00
Chris Robinson ed82194260 Deinterlace audio when loading it into the stack.
It may be better to deinterlace it on load into the buffer, though I don't
imagine this will affect much. It could be nice for consistency's sake, though
it would also complicate things if we ever want to support direct access to
buffer data.
2012-09-26 17:16:25 -07:00
Chris Robinson 012e61b105 Round when calculating NumUpdates from PulseAudio's minreq 2012-09-25 11:32:49 -07:00
Chris Robinson d19d7321d9 Remove an unnecessary include 2012-09-24 20:39:12 -07:00
Chris Robinson d4e82d441e Combine multiple loops into one 2012-09-24 19:07:07 -07:00
Chris Robinson 2ceea21f76 Avoid mixing silence to output
"Silence" being less than -100dB.
2012-09-24 15:33:44 -07:00
Chris Robinson 12327da4fe Remove SSE resamplers. They aren't gaining us much this way. 2012-09-24 14:47:06 -07:00
Chris Robinson 52d403b0d1 Make sure PulseAudio sets an update size that's a multiple of 4 samples with SSE 2012-09-20 15:01:49 -07:00
Chris Robinson ccd35bffe7 Help ensure the update size is a multiple of 4 with SSE 2012-09-20 14:54:28 -07:00
Chris Robinson d1c93525ed Fix matrix multiply used by the SSE cubic resampler
Also remove the 4-sample loop. It's not terribly effective.
2012-09-18 18:09:43 -07:00
Chris Robinson f0732426f2 Precision control bits don't exist with SSE 2012-09-18 15:08:36 -07:00
Chris Robinson eedbd1e81c Improve a couple error traces with the HRTF loader 2012-09-18 12:15:43 -07:00
Chris Robinson 6005e7f27f fpu_control.h is no longer used 2012-09-18 11:28:47 -07:00
Chris Robinson e779e64c95 Win64 doesn't allow _controlfp or __control87_2 to set the precision control bits 2012-09-18 11:20:48 -07:00
Chris Robinson 3baaf99a6e Fix a typo in a comment 2012-09-17 07:48:57 -07:00
Chris Robinson 4a86983a6f Avoid correcting for more non-existing clicks 2012-09-16 09:04:37 -07:00
Chris Robinson c19d9d3452 Don't try to correct for non-existing clicks. 2012-09-16 08:42:36 -07:00
Chris Robinson 79b6d89b3f Explicitly give the wet buffer 1 channel 2012-09-16 08:27:50 -07:00
Chris Robinson 535e8fd0f7 Check the proper value for setting WetPendingClicks 2012-09-16 08:15:03 -07:00
Chris Robinson a240abb6c8 Avoid building redundant mixers 2012-09-16 08:14:26 -07:00
Chris Robinson dd9d30e248 Implement an SSE MixSend method 2012-09-16 06:19:39 -07:00
Chris Robinson 95535ce99e Properly restore the SSE control word with __control87_2 2012-09-16 06:01:48 -07:00
Chris Robinson df2e82da78 Properly handle the SSE control word 2012-09-16 05:35:38 -07:00
Chris Robinson a1daec66e2 Use __control87_2 when available 2012-09-16 01:44:45 -07:00
Chris Robinson 657ee85136 Use a struct to store the FPU mode 2012-09-16 01:35:16 -07:00
Chris Robinson 965306b296 Add missing source property retrievals 2012-09-14 09:02:36 -07:00
Chris Robinson 28086f6cb7 Implement an SSE cubic resampler 2012-09-14 07:01:58 -07:00
Chris Robinson 45bb010b28 Shorten a warning 2012-09-14 05:05:20 -07:00
Chris Robinson 90e0c74721 Implement an SSE linear resampler 2012-09-14 04:48:53 -07:00
Chris Robinson a20341ac2a Ensure the ResampledBuffer is aligned 2012-09-14 04:16:49 -07:00
Chris Robinson 7635afcb52 Use a source param for the resampler and move them to the mixer source 2012-09-14 04:13:18 -07:00
Chris Robinson 74aee374a6 Remove an unused header 2012-09-14 03:10:12 -07:00
Chris Robinson 4f70766b95 Move a couple macros to more appropriate headers 2012-09-14 02:52:37 -07:00
Chris Robinson 9f58edd7a8 Fix up some more header includes 2012-09-14 02:42:36 -07:00
Chris Robinson 0f3a575a09 Don't include alu.h in alMain.h 2012-09-14 02:14:29 -07:00
Chris Robinson ec74fb9ba2 Enable flush-to-zero mode when possible 2012-09-14 01:10:19 -07:00
Chris Robinson ff809f2271 Move a couple methods to where they're used 2012-09-12 07:57:50 -07:00
Chris Robinson 41f52fdd6a Restore support for the previous mhr format 2012-09-12 07:25:05 -07:00
Chris Robinson aa827790bd Add a note about the hrtf_tables filename markers 2012-09-12 05:34:42 -07:00
Chris Robinson 38e7bd5359 Properly scale the HRTF delays when writing them 2012-09-12 05:11:53 -07:00
Chris Robinson 613ce3f976 Update fnamelist after handling % markers 2012-09-12 04:55:03 -07:00
Chris Robinson ce8a0a9fd5 Use PATH_MAX for the fname length 2012-09-12 03:45:26 -07:00
Chris Robinson f59e4ba7c2 Improve parsing hrtf_tables filenames 2012-09-12 03:41:41 -07:00
Chris Robinson df469bfdea Fix HRTF data format description 2012-09-11 19:55:45 -07:00
Chris Robinson f4642953d4 Warn about SSE performance with non-multiple-of-4 update sizes 2012-09-11 11:17:41 -07:00
Chris Robinson 88b0c30152 Use the correct size for defaultCoeffs 2012-09-11 10:43:53 -07:00
Chris Robinson 01183ad93b Minor cleanups for variable declarations 2012-09-11 09:40:08 -07:00
Chris Robinson 3f97af4c8c Don't use count-specific functions to write to the output buffer 2012-09-11 06:41:24 -07:00
Chris Robinson 4c5cb2189b Use a non-interleaved DryBuffer 2012-09-11 06:32:42 -07:00
Chris Robinson 98ff6f990a Do the filtering separately from the mixing 2012-09-11 05:56:19 -07:00
Chris Robinson aaa81e00b7 Use HUGE_VAL instead of dividing by 0 for infinity, where possible 2012-09-11 04:45:43 -07:00
Chris Robinson 1c44a96768 Silence a couple MSVC warnings 2012-09-11 03:29:00 -07:00
Chris Robinson 34fef456c8 Add example HRTF definition files for makehrtf 2012-09-11 03:19:03 -07:00
Chris Robinson 7e81918f7b Update HRTF code
This update allows for much more flexibility in the HRTF data. It also allows
for HRTF table file names to include "%r" to represent the device's playback
rate (e.g. if you set hrtf-%r.mhr, then it will try to use hrtf-44100.mhr or
hrtf-48000.mhr depending if the device's output rate is 44100 or 48000,
respectively).

The makehrtf utility has also been updated to support more options and input
file formats, as well as the new mhr format.
2012-09-11 02:11:51 -07:00
Chris Robinson 1b840a3db8 Again try to get _WIN32_IE to behave right with MinGW 2012-09-11 01:39:05 -07:00
Chris Robinson 4e440fff62 Remove a hack for MinGW 2012-09-10 00:19:13 -07:00
Chris Robinson 81208acd6e Add missing returns 2012-09-09 21:27:54 -07:00
Chris Robinson af3b5c23c8 Try to ensure mixing samples in multiples of 4 2012-09-09 04:27:47 -07:00
Chris Robinson 9f208417df Use the right method to clear a __m128 to 0 2012-09-09 04:09:31 -07:00
Chris Robinson f1ce139997 Use SSE to do 4 samples at once (non-HRTF direct mix), instead of to apply a matrix row
MaxCHannels no longer needs to be a multiple of 4 now, either.
2012-09-09 04:02:54 -07:00
Chris Robinson f56dddfa73 Implement MixDirect_SSE separately from the C and Neon versions 2012-09-09 00:53:54 -07:00
Chris Robinson 2bf1979d4a Move the target effect slot to the SendParams struct 2012-09-08 22:32:30 -07:00
Chris Robinson 89cab3cd39 Remove STACK_DATA_SIZE in favor of BUFFERSIZE 2012-09-08 22:09:34 -07:00
Chris Robinson 6723d27e06 Remove some unused code 2012-09-08 21:53:04 -07:00
Chris Robinson 1c03025090 Separate the resampling and mixing steps 2012-09-08 21:34:36 -07:00
Chris Robinson 7a37a63b8e Workaround MinGW __declspec(align(x)) bug 2012-09-08 19:24:07 -07:00
Chris Robinson 2b5fdad2c8 Free the global ALSA config after opening capture devices too 2012-09-02 14:26:46 -07:00
Lauri Kasanen a6f9ede849 alsa: Free the global config cache to remove a lot of reported leaks
Signed-off-by: Lauri Kasanen <curaga@operamail.com>
2012-09-02 14:16:23 -07:00
Chris Robinson 78dac1bf4a Set the error if setting a new integer source offset fails 2012-09-01 19:31:43 -07:00
Chris Robinson 795909c39e Minor cleanups for mixer_inc 2012-08-29 02:43:19 -07:00
Chris Robinson 3c042a054d Add an SSE-enhanced path for applying the mixer matrix 2012-08-29 01:56:04 -07:00
Chris Robinson 332d66c183 Make sure device buffers are 16-byte aligned 2012-08-29 01:40:42 -07:00
Chris Robinson de006947ac Increase MaxChannels to be a multiple of 4 2012-08-29 01:11:55 -07:00
Chris Robinson 8a128e5d25 Set an error closer to where it occurs for setting source properties 2012-08-29 00:25:01 -07:00
Chris Robinson 34cbacf424 Add helpers to set source properties 2012-08-28 22:16:55 -07:00
Chris Robinson f2fa5e7c88 Print the error when reporting snd_pcm_delay failure 2012-08-27 22:52:37 -07:00
Chris Robinson 3b735cc34a Use _aligned_free for pointers returned by _aligned_malloc 2012-08-24 17:45:16 -07:00
Chris Robinson 238caa1492 Constify some parameters 2012-08-21 16:01:11 -07:00
Chris Robinson e5ebe345ad Add the option to retrieve the source offset and latency in seconds 2012-08-20 15:57:27 -07:00
Chris Robinson 965608356f Fix getting the RW offsets with alGetSourcei64vSOFT 2012-08-20 15:33:28 -07:00
Chris Robinson bc1ce7b3ac Add methods to get source properties as doubles 2012-08-20 15:26:35 -07:00
Chris Robinson 6a3619c40f Fix a parameter name 2012-08-20 14:50:43 -07:00
Chris Robinson 50fa9a44be Properly export alGetSource3i64SOFT 2012-08-20 14:48:08 -07:00
Chris Robinson 4937a48bd9 Use helper functions to read source properties
And make sure the 64-bit int getters to get 32-bit int values
2012-08-20 14:16:58 -07:00
Chris Robinson 987a81c4de Use the correct 64-bit int type for the extension functions 2012-08-20 12:22:00 -07:00
Chris Robinson f5e0500df4 Add a macro to help make a 64-bit value 2012-08-19 22:31:55 -07:00
Chris Robinson 37924887e8 Use a default method to provide a dummy latency 2012-08-19 12:31:59 -07:00
Chris Robinson 0db7c4131b Hold the device lock with PulseAudio between mixing and writing 2012-08-18 17:20:20 -07:00
Chris Robinson a56a974149 Hold the device lock with ALSA between mixing and writing 2012-08-18 17:18:53 -07:00
Chris Robinson af7f3768fa Use non-blocking access for ALSA playback 2012-08-18 17:01:56 -07:00
Chris Robinson 0865db564f Move the device lock into the backend function table
For backend-specific implementations: this should hold the audio mixer loop for
playback devices, and provide recursive mutex behavior.
2012-08-18 15:58:04 -07:00
Chris Robinson 2b020040b4 Keep the backend device open until the last reference is released
This should generally happen with the next ALCdevice_DecRef call unless the
device is in the middle of being used in another thread, in which case it's
probably best to not remove the resources.
2012-08-18 14:49:42 -07:00
Chris Robinson 3ae5fcbd7e Add the start of AL_SOFT_source_latency
This extension will provide a way for apps to get accurate latency and playback
position information
2012-08-18 11:06:39 -07:00
Chris Robinson ac4fc4026c Add a device method to retrieve the active latency
This is effectively the time until the next update will be heard by the user,
or the closest approximation thereof, in nanoseconds.
2012-08-17 13:38:52 -07:00
Chris Robinson 9737f68315 Place the alignment attribute before the variable declaration 2012-08-16 10:42:48 -07:00
Chris Robinson 2a8c26e390 Remove extra parenthesis used in the aligned attribute declaration 2012-08-16 10:33:37 -07:00
Chris Robinson 3c395a82db Fix aligned allocator fallback 2012-08-16 10:30:13 -07:00
Chris Robinson 15c6b9b165 MSVC wants __declspec(align), not __declspec(aligned) 2012-08-16 10:23:25 -07:00
Chris Robinson 147fd9de4e Avoid dynamic handling of libsndio
This makes it safer and easier to handle as OpenBSD updates it
2012-08-16 10:12:50 -07:00
Chris Robinson 0bca771a88 Apply HRTF mixer coefficients with stepping using SSE 2012-08-15 08:19:04 -07:00
Chris Robinson f4ff63e271 Check the max CPUID functions before calling them 2012-08-15 06:28:45 -07:00
Chris Robinson bbcf4e8c1d Make sure sources are 16-byte aligned
They contain fields that require 16-byte alignment for SSE (and Neon?)
acceleration.
2012-08-15 05:54:13 -07:00
Chris Robinson 2859357939 Add wrapper methods to ensure aligned allocations 2012-08-15 05:50:40 -07:00
Chris Robinson 2cbb565d09 Ensure some HRTF-related parameters are aligned 2012-08-15 04:10:41 -07:00
Chris Robinson 4d157d2849 Make sure the working coefficients are aligned for the SSE mixer 2012-08-15 03:43:36 -07:00
Chris Robinson 64cd763e19 Remove unused filter functions 2012-08-15 01:48:47 -07:00
Chris Robinson c3f387da70 Check for and use the -msse switch when dealing with SSE intrinsics 2012-08-15 01:46:42 -07:00
Chris Robinson 6ebeb3b4f7 Check the correct macro for ARM Neon support 2012-08-15 01:32:17 -07:00
Chris Robinson 2e18baa3c8 Don't bother handling MMX since we don't use it 2012-08-15 01:29:19 -07:00
Chris Robinson c9e87a19a9 Add cmake options to disable or require support for CPU extensions 2012-08-15 01:24:50 -07:00
Chris Robinson 3b87e29e63 Move mixers into separate source files 2012-08-15 01:01:55 -07:00
Chris Robinson e9a20cb985 Add an SSE-enhanced HRTF mixer using intrinsics
Not as good as it could be, but it's something
2012-08-14 03:53:07 -07:00
Chris Robinson cfc5205569 Fix a typo 2012-08-13 22:31:58 -07:00
Chris Robinson 17dfaa3aae Add a config option to disable use of CPU extensions 2012-08-13 10:37:49 -07:00
Chris Robinson 0a3eba08d8 Allow building multiple types of mixers, selecting based on CPU caps 2012-08-13 09:18:46 -07:00
Chris Robinson add53e28c2 Check for some CPU extensions 2012-08-13 09:07:01 -07:00
Chris Robinson f42f655ea9 Don't unlock the pulseaudio mainloop when mixing
Lock contention could cause a lengthy delay between mixing and writing the
audio. There shouldn't be any risk of dead-locking as the device lock should
never be held when the mainloop gets locked.
2012-08-12 03:52:38 -07:00
Chris Robinson 8304fc1bef Handle the scalar values a bit better 2012-08-11 06:20:24 -07:00
Chris Robinson ea84fee952 Use a more efficient method to blend HRTF delays and coefficients 2012-08-11 02:16:34 -07:00
Chris Robinson d6fd52a0e5 Add an option for wide stereo sources
Replaces the old stereodup option
2012-08-09 05:38:07 -07:00
Chris Robinson 0dde7dd88a Use a more efficient method to offset the speaker angles when computing gains 2012-08-09 02:31:19 -07:00
Chris Robinson 1323e11701 Interpolate between HRTF delay samples using the delay fraction 2012-07-20 15:08:11 -07:00
Chris Robinson 7ac9944bb3 Increase the directional change sensitivity for the HRTF delta 2012-07-20 14:38:49 -07:00
Chris Robinson e32e2fd6b8 Fix an inadvertent change in the HRTF coefficient stepping value array size 2012-07-19 11:49:43 -07:00
Chris Robinson 6bd535bed0 Use wrappers for float-typed math functions 2012-06-29 02:12:36 -07:00
Chris Robinson 524c88c402 Test the squared length of a vector before normalizing 2012-06-28 20:54:10 -07:00
Chris Robinson 1c096b101d Minor typo fixes 2012-06-28 19:57:23 -07:00
Chris Robinson 583dc8dbca Don't use all caps for enum value names 2012-06-28 18:49:49 -07:00
Chris Robinson befa4e7528 Don't try to set FE_TOWARDZERO when it's not available 2012-06-28 18:27:45 -07:00
Chris Robinson bc6751237c Clamp the Y position before trying to get the elevation 2012-06-28 17:26:00 -07:00
Chris Robinson 6f3be31b60 Add some includes 2012-06-20 03:01:19 -07:00
Chris Robinson 8001d86751 Add missing include 2012-06-19 22:15:19 -07:00
Chris Robinson acca3e5a21 Use a proper method to wait for a PulseAudio operation 2012-06-19 10:37:09 -07:00
Chris Robinson d17f835e74 Remove an unused include 2012-06-19 04:02:55 -07:00
Chris Robinson ef1f8aa9a0 Don't needlessly verify a device 2012-06-19 00:25:45 -07:00
Chris Robinson a5ac6eb524 Remove an outdated comment about _REENTRANT 2012-06-18 02:04:10 -07:00
Chris Robinson ac84d95c6e Silence some GCC warnings 2012-06-17 22:17:03 -07:00
Chris Robinson 785f52aa29 Only try float32 with DSound when specifically requested
Some hardware drivers seem to have a bug where a buffer descriptor with a
32-bit float sample type ends up creating a non-float32 buffer.
2012-06-17 04:13:01 -07:00
Chris Robinson 051195a991 Avoid failing if ALSA can't set a buffer or period time near to our request
These functions should not fail, but there seem to be bugs that cause them to
sometimes. We're flexible enough to handle almost anything it throws back at us
though, so anything is better than nothing here.
2012-06-16 23:02:59 -07:00
Chris Robinson 90ca6675c2 Allow PulseAudio to spawn a server by default
Since only one backend is used at a time now, the issues with device ownership
aren't really prevelant anymore. An application that tries to open all
enumerated devices won't run into the problem of spawning a server and end up
preventing other backend devices from working.
2012-06-16 22:37:45 -07:00
Chris Robinson 0db9730a50 Add an option to allow PulseAudio to move streams
This could be made the default, if there's a safe way to update the AL device
specifier.
2012-06-16 22:32:19 -07:00
Chris Robinson ad2643d8af Clarify some comments, fix some definition ordering, and add some includes 2012-06-15 23:46:09 -07:00
Chris Robinson 46840ee787 Add DEVPKEY_Device_FriendlyName declaration for mingw-w64 compatibility 2012-06-14 18:10:10 -07:00
Chris Robinson 1e0282a238 Add some more entries to .gitignore 2012-05-09 16:30:22 -07:00
Chris Robinson 259e4f5b38 Rename AllDevice -> AllDevices 2012-05-09 16:28:16 -07:00
Chris Robinson e44229c74e Clamp the ALSA period time between min and max if setting near fails 2012-05-08 01:23:53 -07:00
Chris Robinson 92827cc467 Clamp the ALSA buffer time between min and max if setting near fails
It seems there's some bugs with snd_pcm_hw_params_set_buffer_time_near that
cause it to return "invalid argument" despite a valid nearby buffer time being
available. Clamping the buffer time seems to mitigate the issue a bit.
2012-05-07 11:42:59 -07:00
Chris Robinson dd3d5bc2d4 Look for libsndio.so.2 instead of libsndio.so 2012-05-05 16:19:51 -07:00
Chris Robinson b4c7b3309a Only use _FPU_GETCW and _FPU_SETCW for x86 targets 2012-05-03 12:15:51 -07:00
Chris Robinson 010a66487d Use 2 chained one-pole filters for the wet path 2012-04-30 09:33:00 -07:00
Chris Robinson aed35fd811 Use ComputeAngleGains for multi-channel sources and remove the unused lookup table stuff 2012-04-29 05:04:46 -07:00
Chris Robinson 8d9838012d Remove the stereodup option
There's better ways to do it now with "wide" sound sources.
2012-04-29 04:44:53 -07:00
Chris Robinson 57d36ec582 Fix a couple comment typos, and a minor cleanup 2012-04-29 03:39:08 -07:00
Chris Robinson febc2a50cb Use ComputeAngleGains for 3D sounds 2012-04-28 13:06:16 -07:00
Chris Robinson 74e128bd0f Fix ComputeAngleGains
The old code would improperly handle speakers just inside or outside the
coverage area if it had to be inverted. It also didn't properly handle when
only one speaker was outside of the covered area.
2012-04-28 12:12:02 -07:00
Chris Robinson c34d78f417 Use ComputeAngleGains for the echo and dedicated effects 2012-04-28 08:24:09 -07:00
Chris Robinson 611bd0b2d3 Add a method to calculate gains given a sound point and its half-width, and use it for reverb
The half-width ranges from 0 to pi, and essentially specifies the coverage area
around the listener. At 0, it's an infinitely small point sound and behaves
like a usual panning sound. At pi/2 it covers half the area, and at pi it
covers the whole area.
2012-04-28 08:21:53 -07:00
Chris Robinson 267d38cf20 Flip the parameters to aluCart2LUTpos, so it behaves a bit more like atan2 2012-04-28 03:58:58 -07:00
Chris Robinson 26456f13e1 Pass the direct/send params to the mixer instead of accessing through the source 2012-04-28 03:31:13 -07:00
Chris Robinson 2484c44db0 Apply the listener gain with the filter 2012-04-28 03:13:37 -07:00
Chris Robinson a8a13fcbd8 Move the HrtfParams into the DirectParams 2012-04-28 02:23:53 -07:00
Chris Robinson 0a193c5645 Move the direct and send params into separate different types 2012-04-28 01:32:52 -07:00
Chris Robinson 2450ee1723 Use separate methods for the dry and wet mixing loops 2012-04-27 23:46:51 -07:00
Chris Robinson 67f1cdfcbe Store the speaker angles in the device 2012-04-27 07:38:17 -07:00
Chris Robinson 549d542a31 Reorganize some ALsource fields 2012-04-27 01:36:13 -07:00
Chris Robinson 2cdb58fb85 The mixer always uses floats 2012-04-26 07:50:17 -07:00
Chris Robinson 68ab232364 Use a separate loop to actually update DataPosFrac and DataPosInt 2012-04-26 06:44:37 -07:00
Chris Robinson 871d61f3f9 Add and fixup some ALsource comments 2012-04-26 03:26:22 -07:00
Chris Robinson b08bd3746e Add AL_EXT_SOURCE_RADIUS to alext.h 2012-04-26 02:05:49 -07:00
Chris Robinson 39ab0b0aa3 Add AL_EXT_STEREO_ANGLES to alext.h 2012-04-26 01:19:25 -07:00
Chris Robinson 5c70cacc13 Fixup some comments in ALu.c 2012-04-26 00:59:17 -07:00
Chris Robinson 8c2cd057b2 Fix a minor typo 2012-04-25 19:37:12 -07:00
Chris Robinson fdc4133ac9 Partially convert alEffect.c and alFilter.c to the new error handling 2012-04-25 00:24:02 -07:00
Chris Robinson 3833c021c0 Move a comment 2012-04-25 00:03:01 -07:00
Chris Robinson cafbd9461e Fix up alEffect.c and alFilter.c a bit 2012-04-24 23:06:10 -07:00
Chris Robinson 6995ac1ed2 Change alAuxEffectSlot.c to the new error handling 2012-04-24 22:57:23 -07:00
Chris Robinson 58e3dc3d52 Add a SIGTRAP for when alGetError is called without a context 2012-04-24 21:41:01 -07:00
Chris Robinson 3e49e79140 Couple small error handling fixups 2012-04-24 03:52:21 -07:00
Chris Robinson dec8e81211 Change alState.c to the new error handling 2012-04-24 03:09:28 -07:00
Chris Robinson 4188a5f3f3 Fix compiling al.h with C++ 2012-04-24 02:51:23 -07:00
Chris Robinson 31d124582b Switch alBuffer.c to the new error handling scheme, and get rid of more hungarian notation 2012-04-24 00:17:05 -07:00
Chris Robinson 248e079301 Switch alListener.c to the new error handling scheme, and get rid of more hungarian notation 2012-04-23 21:35:05 -07:00
Chris Robinson 5ce850570f Add try/catch-like macros to handle errors, and convert alSource.c to use them 2012-04-23 19:46:05 -07:00
Chris Robinson 648464a2da Check the right playback device file with OSS 2012-04-22 05:06:44 -07:00
Chris Robinson 29f77003c0 Some alSource.c cleanups 2012-04-21 05:56:03 -07:00
Chris Robinson dfa38ed7a5 Reformat alc/h and al.h, and redo the comments 2012-04-21 04:34:08 -07:00
Chris Robinson 66c918c84b Remove export pragmas from headers
Available information suggests it's only useful for Mac OS9 and earlier (not
OSX).
2012-04-20 22:53:47 -07:00
Chris Robinson 77f55da6c7 Use a macro to declare the function and enum list members 2012-04-20 22:38:03 -07:00
Chris Robinson d6af7cdcb2 Rename another variable to avoid hungarian notation 2012-04-20 22:37:36 -07:00
Chris Robinson 0ad8047820 Combine ALCenum and ALenum lists 2012-04-20 01:24:58 -07:00
Chris Robinson 2115f578e9 Reorganize ALC functions and fixup some comments 2012-04-20 01:18:38 -07:00
Chris Robinson 511b4790d0 Remove hungarian notation from dsound and mmdevapi 2012-04-19 23:42:02 -07:00
Chris Robinson d6491e21f9 Remove some more hungarian notation 2012-04-19 23:28:34 -07:00
Chris Robinson cff0f938e5 Remove hungarian notation from the winmm backend 2012-04-19 23:22:11 -07:00
Chris Robinson 5418519637 Remove some more hungarian notation 2012-04-19 23:00:58 -07:00
Chris Robinson b023dbe7b8 Remove hungarian notation from the device and context structs 2012-04-19 22:50:11 -07:00
Chris Robinson 8092cd9205 Remove hungarian notation from some function parameters 2012-04-19 22:40:40 -07:00
Chris Robinson 60da5bd338 Remove an unused field 2012-04-19 22:32:04 -07:00
Chris Robinson c9e64596a4 Use a consistent name for the self-id field 2012-04-19 22:28:01 -07:00
Chris Robinson 08bd5e8b37 Fix comment and rename the buffer's self-id 2012-04-19 22:14:02 -07:00
Chris Robinson 125b743e5c Remove hungarian notation from sources 2012-04-19 21:46:29 -07:00
Chris Robinson 9a3eb397f4 Print the device name in printALCInfo 2012-04-16 23:27:13 -07:00
Chris Robinson 84c1f2bfd0 Avoid shadowing a couple variables 2012-04-16 23:09:36 -07:00
Chris Robinson d7ba49ea62 Clear the position fraction when setting a new source offset 2012-04-16 22:13:39 -07:00
Chris Robinson e47557630e Use a double to store the source offset and don't use milliseconds for seconds 2012-04-16 22:11:03 -07:00
Chris Robinson f6a1e53f19 Read ALSOFT_TRAP_*ERROR when reading the config 2012-04-16 20:13:50 -07:00
Chris Robinson a0a015f19f Set a 1MB stack size for created threads 2012-04-16 19:15:48 -07:00
Chris Robinson e3bea63f94 Check for a recent libavutil for the example, too 2012-04-07 12:35:59 -07:00
Chris Robinson 6b154025ab Release 1.14 2012-03-28 08:55:17 -07:00
Chris Robinson b9b2bdb37a Make sure we have a recent-enough ffmpeg 2012-03-27 08:52:21 -07:00
Chris Robinson 7b35c77062 Add a missing round() for MSVC 2012-03-27 08:44:08 -07:00
Chris Robinson c23cfdcfa1 Make sure M_PI is defined for makehrtf 2012-03-27 08:20:04 -07:00
Chris Robinson 929e9bfac1 Avoid DOS-style line-endings and stdint types for the makehrtf utility 2012-03-25 19:57:40 -07:00
Chris Robinson b51afb74da Add a utility to generate OpenAL Soft's HRTF data from the MIT KEMAR data 2012-03-22 22:45:03 -07:00
Chris Robinson 5a43cd3b64 Use avformat_free_context to free a format context that failed to open 2012-03-21 13:54:13 -07:00
Chris Robinson 5d08563975 Handle very small speed of sound values 2012-03-18 08:20:08 -07:00
Chris Robinson adb83ad3c0 Avoid converting positions to meters 2012-03-18 08:09:59 -07:00
Chris Robinson 2d0e028993 Base the initial reverb decay on the source's dry path attenuation 2012-03-18 08:06:21 -07:00
Chris Robinson 703d72f20e Use the correct loop count for reading the elevation offsets from external HRTF files 2012-03-17 03:42:07 -07:00
Chris Robinson b7d67d8bcc Don't error when querying unknown enums for the channel config and sample format 2012-03-15 15:22:58 -07:00
Chris Robinson 7e5b13aabd Finalize ALC_SOFT_loopback 2012-03-15 15:21:15 -07:00
Chris Robinson 7fbcb4b9e6 Use the actual distance from the ref distance for air absorption and initial reverb decay 2012-03-14 22:45:52 -07:00
Chris Robinson 767a5b3c73 Mark the GlobalContext pointer as volatile 2012-03-14 00:13:16 -07:00
Chris Robinson a08080bd82 Avoid modifying the ALCdevice in portaudio's open method 2012-03-13 22:18:51 -07:00
Chris Robinson 415bf58a06 Store the portaudio stream parameters with the device 2012-03-13 21:52:47 -07:00
Chris Robinson 3b517bfa59 Initialize the default effect after opening the device 2012-03-13 19:02:26 -07:00
Chris Robinson e356527a88 Rename GetReverbEffect to LoadReverbPreset 2012-03-13 15:48:51 -07:00
Chris Robinson 07e34cc3cf Use enums to specify the default channel configuration and sample type 2012-03-13 15:32:44 -07:00
Chris Robinson f7f1dd1346 Initialize the default effect when opening the device and update it with the other effect slots 2012-03-13 15:07:38 -07:00
Chris Robinson 92ccacdc77 Pass the device to InitializeEffect and return the error enum from it 2012-03-13 14:58:34 -07:00
Chris Robinson b5ed2a5351 Pass a device to the effect update functions 2012-03-13 14:49:58 -07:00
Chris Robinson 55c5e4fe92 Rename ForcedEffect to DefaultEffect 2012-03-13 14:38:09 -07:00
Chris Robinson 3b165809cc Minor coreaudio fix, and make sure to set the device name 2012-03-13 13:48:42 -07:00
Chris Robinson 5c59e6e704 Try to ensure the full mmdevapi buffer is used 2012-03-13 01:39:57 -07:00
Chris Robinson ef2f237276 Try to ensure at least 2 mmdevapi updates 2012-03-13 01:35:31 -07:00
Chris Robinson 0af14c7a11 Try to find a multiple of mmdevapi's period size nearest to the wanted update size 2012-03-13 01:32:15 -07:00
Chris Robinson 8ef777ebbe Restore the FPU mode in an error path 2012-03-12 23:58:12 -07:00
Chris Robinson 305d790a19 Revert the default stereo layout back to -90 and +90 degrees
For the time being, until better stereo spatialization can be implemented and
headphones can be better detected.
2012-03-12 22:49:10 -07:00
Chris Robinson 1f3c1091d3 Finalize AL_SOFT_direct_channels 2012-03-12 22:46:13 -07:00
Chris Robinson 663b43f4f1 Attempted fix for the coreaudio backend 2012-03-12 21:58:40 -07:00
Chris Robinson 3e0cf0f239 Simply (hopefuly) the explanation about the HRIR set ordering 2012-03-12 18:42:21 -07:00
Chris Robinson 560f34e3df Channel angle is already in radians, not degrees 2012-03-12 18:29:46 -07:00
Chris Robinson 335232207d With DirectChannels enabled, don't try to mix channels that have no matching output 2012-03-12 18:27:25 -07:00
Chris Robinson 93e71f0554 Some HRTF clarifications 2012-03-12 17:54:16 -07:00
Chris Robinson 0efbf7bc9e Fix typo 2012-03-12 17:15:44 -07:00
Chris Robinson 3daaa078fb Add a note about the HRTF data being minimum-phase reconstructed 2012-03-12 17:14:42 -07:00
Chris Robinson 647448a32b Reset HAVE_MMDEVAPI before checking the backends 2012-03-12 15:33:56 -07:00
Chris Robinson f7a8e3db35 Don't assume the calculated HRTF delta for the gain difference is less than 1 2012-03-12 13:31:19 -07:00
Chris Robinson 7747c320a8 Remove an unused property key definition 2012-03-12 12:56:52 -07:00
Chris Robinson ebc69bc870 Minor corrections to the hrtf.txt 2012-03-12 00:58:14 -07:00
Chris Robinson 10e2b7ecbf Document the hrtf_tables config option 2012-03-11 19:29:19 -07:00
Chris Robinson 842a698f2b Add a text file explaining OpenAL Soft's HRTF support and data file format 2012-03-11 19:27:40 -07:00
Chris Robinson f5f519aa80 Properly make the source velocity relative to the listener velocity 2012-03-09 23:53:45 -08:00
Chris Robinson 7db29a2332 Clamp the upper and lower bound of the doppler velocity scale more like the original 2012-03-09 22:38:26 -08:00
Chris Robinson 6e70ae9958 The listener velocity is specified in world coordinates, too 2012-03-08 18:27:05 -08:00
Chris Robinson e190fc9416 Refactor the doppler shift calculations 2012-03-08 18:19:15 -08:00
Chris Robinson 1a4a0abd1a Pin the DLL for Windows
The mmdevapi backend does not react well to being unloaded dynamically. It has
a message-handling thread running in the background which can't quit before
DllMain is called with DLL_PROCESS_DETACH, at which point it's too late to
safely message and wait for it shutddown, thus it can continue running after
the DLL is unloaded from memory.
2012-03-08 17:42:16 -08:00
Chris Robinson dddacc25a5 Store a duplicate of the mmdevapi device ID 2012-03-06 13:15:55 -08:00
Chris Robinson 2ba2da467c Add a comment explaining the pulseaudio prop_filter 2012-03-06 12:01:15 -08:00
Chris Robinson f2b64fd111 Update the ALCdevice in winmm's reset method instead of open 2012-03-06 10:14:05 -08:00
Chris Robinson 6623f69476 Mark the device as running for capture, too 2012-03-05 15:10:02 -08:00
Chris Robinson db36574667 Hold the list lock while calling the backend
Pretty ugly, but we mustn't hold the device lock
2012-03-05 14:53:25 -08:00
Chris Robinson 3131104595 Filter out a couple pulseaudio stream properties by setting them to 0-length data
It seems Qt/Phonon will globally set media.role and phonon.streamid properties,
causing them to apply even to our streams if they happen to be loaded in the
same process. This is particularly a problem because media.role gets set to
"event", preventing the streams from showing in pavucontrol.
2012-03-05 13:00:51 -08:00
Chris Robinson f64bdc4bd4 Returned pa_operations can be NULL 2012-03-05 08:37:50 -08:00
Chris Robinson fe6e532c87 Use a separate backend callback to start playback of the device
This allows us to properly update the ALCdevice and its resources with the new
parameters before starting playback, instead of expecting the mixer to block
and wait after it has begun.

This also lets us avoid holding the device lock while resetting and starting
the device, which helps prevent lock inversion on some backends (ie, one thread
locking A then B, and another thread locking B then A), ultimately allowing
certain backends to asynchronously update the ALCdevice without risk of lockup.
Capture still has issues here, however.
2012-03-05 07:11:09 -08:00
Chris Robinson 5cdeeb47f3 Avoid some uninitialized warnings 2012-03-05 00:28:57 -08:00
Chris Robinson bc1bb7f90f Print and handle errors from pa_stream_readable_size 2012-03-05 00:11:28 -08:00
Chris Robinson 0f0141b9dd Set prebuf to 0 since we're handling the stream start manually 2012-03-04 09:24:07 -08:00
Chris Robinson 302c08a14b Enforce range limits on periods and period_size. 2012-03-04 05:44:37 -08:00
Chris Robinson b643f13a6e Make sure the pulse stream is properly started and better handle buffer size changes 2012-03-04 05:30:42 -08:00
Chris Robinson 487d05a73e Invert a nested loop 2012-03-03 10:31:27 -08:00
Chris Robinson 91ed077eba Handle BS2B cross-feed right after click removal 2012-03-03 09:41:16 -08:00
Chris Robinson 49433c2074 Slight correction for handling the wave writer backend's timer wrapping 2012-03-03 08:50:15 -08:00
Chris Robinson a93ff6ef3c Slight correction for handling the null backend's timer wrapping 2012-03-03 08:47:23 -08:00
Chris Robinson 44e10c4f69 Remove some unneeded pulseaudio calls 2012-03-02 09:46:36 -08:00
Chris Robinson 7a832c4522 Don't unload libpulse
This sometimes spawns esoteric errors in dlclose
2012-03-02 09:27:30 -08:00
Chris Robinson 7d0faba424 Remove some unneeded markers 2012-03-02 09:08:17 -08:00
Chris Robinson 01d942628b Set prebuf to the expected buffer size
This is to make sure the stream starts even if PulseAudio increases tlength.
2012-03-02 05:38:11 -08:00
Chris Robinson 2a9012c94d Don't rely on PulseAudio to wake us up using early requests 2012-03-02 04:51:29 -08:00
Chris Robinson b571926a6d Avoid a leak if PortAudio fails to give a usable channel count 2012-03-02 02:14:04 -08:00
Chris Robinson f784aa704d Make sure the device gets stopped when closing even if there's no contexts 2012-03-02 00:10:24 -08:00
Chris Robinson 153affac12 Recalculate the new update size using the device's UpdateSize 2012-03-02 00:08:56 -08:00
Chris Robinson e5d365a3d9 Avoid an unnecessary doCapture check 2012-03-01 20:47:49 -08:00
Chris Robinson 6ffaabeb75 Update the layout config option names 2012-03-01 10:06:55 -08:00
Chris Robinson 06debb153d Fix a memory leak 2012-03-01 09:56:25 -08:00
Chris Robinson 51e6786f31 Set the func list after successfully connecting to PulseAudio 2012-03-01 09:22:37 -08:00
Chris Robinson 50ef19720d Make alcLoopbackOpenDeviceSOFT take a standard "device" name for a parameter 2012-03-01 08:30:21 -08:00
Chris Robinson db4d62d17c Print out a more descriptive name for the opened device in openal-info 2012-03-01 06:59:51 -08:00
Chris Robinson 8d24bd720f Don't use GUIDs to ID mmdevapi devices, and don't enumerate if not needed 2012-03-01 06:07:00 -08:00
Chris Robinson 55556f0e6c Avoid enumerating when opening the default ALSA device 2012-03-01 04:58:28 -08:00
Chris Robinson 7d9cabebff Use NULL to open the default PulseAudio device, and get the actual device name from the stream 2012-03-01 04:37:40 -08:00
Chris Robinson 9602c46809 Move some function definitions down 2012-03-01 04:12:47 -08:00
Chris Robinson 85daab0d5e Don't define lib handles when dynamic loading is disabled 2012-03-01 03:46:45 -08:00
Chris Robinson 105faa1935 Only support PA_API_VERSION 12 2012-03-01 03:29:28 -08:00
Chris Robinson b299b90964 Update the stored buffer attributes if PulseAudio changes it 2012-03-01 03:09:15 -08:00
Chris Robinson 346f1ac293 Don't allow the PulseAudio stream to move
Since the device specifier isn't updated to reflect the change, it's best to
not allow it to move. This should be fixed eventually.
2012-03-01 02:52:32 -08:00
Chris Robinson 6e113eea89 Don't force a sample spec when looking for the default device 2012-03-01 01:34:49 -08:00
Chris Robinson 54f906b347 Don't check if the pulseaudio sink is suspended when opening 2012-03-01 01:19:07 -08:00
Chris Robinson 4b252bc494 Constify a couple string pointers 2012-03-01 01:03:37 -08:00
Chris Robinson b192165fb4 Look for and enumerate the default PulseAudio source first 2012-02-29 23:52:10 -08:00
Chris Robinson f887f4efdd Use a separate function to create and connect a PulseAudio record stream 2012-02-29 23:46:30 -08:00
Chris Robinson b37e54eead Look for and enumerate the default PulseAudio sink first 2012-02-29 23:35:01 -08:00
Chris Robinson ea4f458c1d Avoid enumerating multiple devices with the same name 2012-02-29 23:21:38 -08:00
Chris Robinson 394c76d64f Avoid passing the ALCdevice to connect_playback_stream 2012-02-29 23:13:09 -08:00
Chris Robinson d34fc21a50 Print the enumerated devices from PulseAudio 2012-02-29 22:31:51 -08:00
Chris Robinson f5435aa2de Minor cleanups for OSS and Solaris 2012-02-29 21:45:44 -08:00
Chris Robinson e88f954cbf Allow reading audio from ALSA directly if the requested length is supported 2012-02-29 10:01:45 -08:00
Chris Robinson fe897f291b Don't store the frame size with pulse_data 2012-02-29 06:51:21 -08:00
Chris Robinson fbb6239299 Read capture data from pulseaudio as needed, avoiding a ring buffer 2012-02-29 05:57:54 -08:00
Chris Robinson ae1633497c Get and release the mmdevapi render client iface on the message thread 2012-02-29 05:54:58 -08:00
Chris Robinson 00b0e64bd2 Use local variables to determine how much to write to mmdevapi 2012-02-29 00:32:20 -08:00
Chris Robinson 2747c87893 Print out unsupported audio format info from ffmpeg 2012-02-28 04:04:52 -08:00
Chris Robinson 2285c79ded Give "OpenAL Soft" for the standard device name 2012-02-28 03:36:16 -08:00
Chris Robinson 230e784b9e Trace the name of the created device 2012-02-27 21:57:33 -08:00
Chris Robinson 4b88168edb Use a switch to handle the ALSA PCM state 2012-02-27 04:31:03 -08:00
Chris Robinson f29fbdf088 Print enumerated ALSA devices 2012-02-26 23:02:29 -08:00
Chris Robinson 233acff3c4 Print enumerated dsound devices 2012-02-26 22:55:50 -08:00
Chris Robinson 5e625c9a68 Watch for unsupported sample types for dsound capture 2012-02-26 22:03:27 -08:00
Chris Robinson da4408835d Print the mmdevapi device and GUID found 2012-02-26 21:59:27 -08:00
Chris Robinson b78b8f0c82 Use separate loops for getting and checking device attributes 2012-02-25 18:03:33 -08:00
Chris Robinson 484e8f07a0 Watch for CLSIDFromString errors 2012-02-23 21:41:24 -08:00
Chris Robinson 504cdadd60 Use an enum for the device type 2012-02-23 15:25:30 -08:00
Chris Robinson 45c10229d9 Print a message when loading the "none" reverb preset 2012-02-23 13:49:46 -08:00
Chris Robinson 973b92b8fd Minor rearrangements for the pulseaudio mixing loop 2012-02-22 18:33:28 -08:00
Chris Robinson 6ba0fd25a2 Remove an unneeded wrapper 2012-02-22 18:08:15 -08:00
Chris Robinson 1dde8ad60b Use the stored minreq size from PulseAudio for rounding off the length to write 2012-02-21 18:07:20 -08:00
Chris Robinson 45b9cbe2b3 Only probe for devices when a backend is loaded 2012-02-20 22:12:48 -08:00
Chris Robinson 14912229b1 Avoid default names for winmm and dsound 2012-02-20 22:03:08 -08:00
Chris Robinson 2b64df05e6 Avoid using a default name with the MMDevApi backend 2012-02-20 21:46:01 -08:00
Chris Robinson 01d8d956da Avoid using a default name when opening a PulseAudio device 2012-02-20 21:03:54 -08:00
Chris Robinson 4f0d862cbe Set the default ALSA device name when probing 2012-02-20 20:59:55 -08:00
Chris Robinson e2ccc6f98e Always use "OpenAL Soft" for the short device enumeration list 2012-02-20 20:45:22 -08:00
Chris Robinson d24ada7ab7 Don't accept the pulseaudio default name for capture 2012-02-20 20:29:20 -08:00
Chris Robinson 54e2316b65 Use a string to report unsupported capture sample types 2012-02-20 20:25:58 -08:00
Chris Robinson 58316a97d5 Override the resampler as needed right after calculating the step value 2012-02-20 20:08:45 -08:00
Chris Robinson 771ef3e8d5 Rename p -> hp for ALSA's hw params 2012-02-20 12:46:09 -08:00
Chris Robinson 65ed18960e Add a text file describing the available environment variables 2012-02-19 23:03:04 -08:00
Chris Robinson 799f8da333 Recognize "none" to disable the default reverb 2012-02-19 17:41:42 -08:00
Chris Robinson 40bb8bd9dc Remove the trailing double-underscore from a couple env vars 2012-02-19 17:24:01 -08:00
Chris Robinson 426f0273cb Remove mention of the environment variables from alsoftrc.sample
There will be a separate file to document all of them.
2012-02-19 17:20:02 -08:00
Chris Robinson 03ec2d6fc7 Check for the ALSOFT_DRIVERS env var for a list of backend drivers
This overrides the 'drivers' config option
2012-02-19 14:27:00 -08:00
Chris Robinson e4f14fdc61 Check the pulse spawn-server config option before testing a connection to pulseaudio 2012-02-19 14:07:44 -08:00
Chris Robinson 418c766493 Use memset to clear the bs2b filter's last sample and remove the unused is_clear function 2012-02-19 13:09:29 -08:00
Chris Robinson 4a65747a4b Add a COUNTOF macro to get the number of entries in a static array 2012-02-19 12:07:40 -08:00
Chris Robinson 6dc194ed1d Fix a misnamed variable 2012-02-19 11:56:01 -08:00
Chris Robinson 1aa69e77e9 Print a warning if using the 'format' config option 2012-02-19 11:54:24 -08:00
Chris Robinson 6ac16867b4 Only set a default slot for non-NULL effect types 2012-02-18 17:09:08 -08:00
Chris Robinson bbc679a5db Minor MMDevApi cleanups 2012-02-18 13:08:54 -08:00
Chris Robinson 592fb2b913 Make sure an ALSA capture device was found before accessing the map 2012-02-18 12:39:17 -08:00
Chris Robinson f68f3f5f86 Simplify ALSA parameter error checking 2012-02-18 12:34:58 -08:00
Chris Robinson 5f23860880 Rename a couple variables and fix their types 2012-02-18 11:07:54 -08:00
Chris Robinson faf59a7cba Fix some comments 2012-02-18 11:03:42 -08:00
Chris Robinson cc2cb1142b Remove unused variables 2012-02-17 12:32:43 -08:00
Chris Robinson eb8be88866 Use the correct type to store the count from IMMDeviceCollection_GetCount 2012-02-17 12:31:53 -08:00
Chris Robinson e43e1bd7f5 Silence another MSVC warning 2012-02-17 01:32:35 -08:00
Chris Robinson eb586b0ed7 Initialize a variable before use 2012-02-17 01:30:41 -08:00
Chris Robinson 75aa96a4a6 Silence an MSVC warning 2012-02-17 01:29:23 -08:00
Chris Robinson 10257f485a Support device enumeration with mmdevapi 2012-02-16 17:04:08 -08:00
Chris Robinson deee6a73f0 Fix a device count leak and make sure COM is cleaned up if an mmdevice open fails 2012-02-16 10:08:50 -08:00
Chris Robinson 84ba0a94c0 Make sure the source write position is at least 15ms ahead of the read 2012-02-16 00:11:50 -08:00
Chris Robinson 5d17cef55f Calculate the buffer time before modifying the frequency 2012-02-15 22:54:17 -08:00
Chris Robinson 70f1a7e3ca Get the mmdevapi device period after initialization 2012-02-15 22:40:01 -08:00
Chris Robinson 60785eab8e Move the endian test macro to alMain.h 2012-02-15 21:47:35 -08:00
Chris Robinson 7e9af6f406 Use a switch to check the device format with winmm 2012-02-15 21:13:13 -08:00
Chris Robinson 6b272ff585 Add a device parameter to alcLoopbackOpenDeviceSOFT 2012-02-15 20:52:49 -08:00
Chris Robinson fecd679f33 Add a device request flag for the sample type 2012-02-15 16:26:32 -08:00
Chris Robinson 2ab0e3521c Remove parameter names from typedefs 2012-02-15 15:23:23 -08:00
Chris Robinson 9afd6b2858 Separate the format config option into channels and sample-type 2012-02-15 15:22:44 -08:00
Chris Robinson 670c88a64c Define a minimum output sample rate and warn if the config option goes below it 2012-02-15 12:26:19 -08:00
Chris Robinson 425ff7cb00 Combine a couple if checks 2012-02-15 11:28:19 -08:00
Chris Robinson 1140b3ae83 Support signed and unsigned 32-bit int output 2012-02-14 11:44:57 -08:00
Chris Robinson 5e1d1a52ba Check for each ALSA prefix config option only as needed 2012-02-13 10:33:53 -08:00
Chris Robinson e394d14cda Use more proper enum names for the resampler 2012-02-12 08:45:19 -08:00
Chris Robinson 9f073b6f1b Use a proper enum value for setting the default distance model 2012-02-12 08:18:20 -08:00
Chris Robinson b375b130ba Add config options for card- and device-specific prefixes in ALSA device names 2012-02-11 09:36:40 -08:00
Chris Robinson 3fb6351212 Format the ALSA device name when enumerating them 2012-02-11 06:55:24 -08:00
Chris Robinson 61be01efc1 Rename SpeakerGain to ChannelGain 2012-02-10 00:29:47 -08:00
Chris Robinson 23bda38be0 Match the 7.1 channel angles with the speakers 2012-02-09 23:59:24 -08:00
Chris Robinson e6a0dd10a9 Pre-scale the channel angles 2012-02-09 23:52:20 -08:00
Chris Robinson ef68aa39f8 Rename non_virtual_channels to direct_channels 2012-02-09 23:35:17 -08:00
Chris Robinson b4b492c417 Improve checks for supported output formats from ALSA 2012-02-09 22:26:16 -08:00
Chris Robinson a98e381c8e Ensure ALbyte3 and ALubyte3 are the proper size 2012-02-07 08:05:01 -08:00
Chris Robinson 18f87b97d5 Always use the default speaker layout for loopback devices 2012-02-07 07:29:34 -08:00
Chris Robinson c211f1d3fe Don't enable BS2B for loopback devices 2012-02-07 07:06:38 -08:00
Chris Robinson deb84cfdac Improve a couple error messages with the full ALSA device control name 2012-02-07 07:04:33 -08:00
Chris Robinson 7ad1b45d11 Change the default stereo speaker positions to match the stereo channel positions
A separate "headphones" configuration may be nice to add for -90,+90, however
not many audio APIs can detect this. Ideally HRTF would be used with
headphones too, which largely ignores the speaker positions, however there
could be situations where this is unfeasible or unwanted.
2012-02-04 10:29:19 -08:00
Chris Robinson 47161150d7 Check for a 32-bit x86 processor before using MSVC's asm intrinsics 2012-02-03 19:34:06 -08:00
Chris Robinson 8e6832407a Only use fast float-to-int GCC assembly for x86 and x86_64 targets 2012-02-03 17:40:21 -08:00
Chris Robinson 9166432ccf Fix typo in the "CarpetedHallway" reverb preset 2012-02-01 16:31:37 -08:00
Chris Robinson 0107e74986 Use an enum for static const values 2012-01-31 03:43:52 -08:00
Chris Robinson c605e06e07 Make the EFX preset values explicit floats 2012-01-31 02:57:22 -08:00
Chris Robinson 15a8e58225 Implement capture for the DirectSound backend
Based on a patch by Jeffrey Knockel <jeff250.at.gmail.com>
2012-01-31 01:47:59 -08:00
Chris Robinson 4393b6b05f Only initialize the default reverb effect when a slot is available 2012-01-29 22:49:17 -08:00
Chris Robinson ed037698fd Reorder a couple effect presets 2012-01-29 22:16:39 -08:00
Chris Robinson e870c22f87 av_find_stream_info is also deprecated, in favor of avformat_find_stream_info 2012-01-29 20:06:30 -08:00
Chris Robinson fd6f03cf74 Use AVMEDIA_TYPE_AUDIO instead of the deprecated CODEC_TYPE_AUDIO 2012-01-29 16:21:57 -08:00
Chris Robinson 706f407ed5 Add some more reverb presets 2012-01-27 23:15:16 -08:00
Chris Robinson dd3abd0f16 Use sizeof(ALfloat) when finding the max stepping value
The mixer always reads from floats, so the size of the storage type is
unimportant.
2012-01-26 22:33:36 -08:00
Chris Robinson 3fda804217 Pass the uncompressed sample count to LoadData and ConvertData for IMA4 2012-01-26 16:53:21 -08:00
Chris Robinson 1d1440ac64 Make sure context attributes are specified for loopback devices 2012-01-26 16:31:46 -08:00
Chris Robinson 119768001c Make sure InitializeEffect on the default slot doesn't set a context error 2012-01-26 14:46:52 -08:00
Chris Robinson 3ee92a2b33 Return the error code from UpdateDeviceParams 2012-01-25 20:00:34 -08:00
Chris Robinson f87b607a53 Make sure the attributes list specifies a format for loopback devices 2012-01-25 19:54:55 -08:00
Chris Robinson b4ba83ec34 Avoid a bit of code duplication 2012-01-25 19:32:10 -08:00
Chris Robinson d807c4bf87 Don't process a default device slot if it doesn't exist 2012-01-25 18:48:26 -08:00
Chris Robinson 423f89c908 Fix a typo 2012-01-24 15:11:23 -08:00
Chris Robinson 8d07bc41c9 Some cleanups for the reverb list 2012-01-24 13:29:33 -08:00
Chris Robinson 86ef115e21 Set the source's slot send parameter to NULL if it's the AL_EFFECT_NULL type 2012-01-23 06:29:03 -08:00
Chris Robinson 1dd8aa9b6f Link with AudioToolbox.framework for the CoreAudio backend if it's found 2012-01-23 03:45:04 -08:00
Chris Robinson f520257463 Avoid duplicating some initialization code 2012-01-20 16:23:15 -08:00
Chris Robinson 00dc3088c8 Honor the disabled effects for the default reverb 2012-01-20 15:56:32 -08:00
Chris Robinson e52e4b1a8c Not all devices will have a default effect slot 2012-01-20 12:53:41 -08:00
Chris Robinson 0ef3a30b84 Fix WaveIn shutdown for real
The processing thread must be messaged when shutting down, and depending on the
state of the device, waveInReset may not do it. Additionally, all buffers must
be removed from the device before they can be unprepared and the device closed.
2012-01-20 12:36:23 -08:00
Chris Robinson b375897374 Remove an unnecessary else statement 2012-01-20 12:20:24 -08:00
Chris Robinson 2b2ce8bf3b Always message the WinMM processing thread when a buffer is done
So the processing thread can keep track of completed buffers when shutting
down, too.
2012-01-20 12:06:31 -08:00
Chris Robinson 47c8c15878 Fix a possible race condition when shutting down WinMM devices
If the processing threads are interrupted between the shutdown check and re-
adding the buffer back to the device, the device can be left with a buffer that
will get unprepared and freed while in use.
2012-01-20 03:02:21 -08:00
Chris Robinson fa36435290 Add a default-reverb config option, and rename __ALSOFT_FORCE_REVERB to match 2012-01-20 01:51:27 -08:00
Chris Robinson 4b83123cb8 Install efx-presets.h 2012-01-19 22:30:12 -08:00
Chris Robinson 37b17c8c78 Add the ability to specify a preset for forced reverb 2012-01-19 20:02:57 -08:00
Chris Robinson 51e81f4867 Add a global option to apply a reverb effect on source send 0
A special slot on the device is created and processed, so it can be shared
across all contexts on the device. Sources that don't have a slot set on send 0
will use this special slot instead.
2012-01-19 19:30:03 -08:00
Chris Robinson ae7b61b040 Avoid printing messages about device parameter changes in the backends 2012-01-17 16:38:58 -08:00
Chris Robinson 5a2a534240 Report parameter request changes in UpdateDeviceParams 2012-01-17 15:55:37 -08:00
Chris Robinson 09258ca5c1 Change the default output format to 32-bit float 2012-01-17 15:38:36 -08:00
Chris Robinson e0966d99a1 Retry opening the PortAudio device with 16-bit short if 32-bit float output fails 2012-01-17 15:26:22 -08:00
Chris Robinson 66a700729b Retry opening the DSound device with 16-bit short if 32-bit float output fails 2012-01-17 12:38:47 -08:00
Chris Robinson a7266442c2 Retry opening the WinMM device with 16-bit short if 32-bit float output fails 2012-01-17 12:21:52 -08:00
Chris Robinson cf34b9c04e Support 32-bit float in the WinMM backend 2012-01-17 12:17:41 -08:00
Chris Robinson 530597347c Safely insert the new context into the list 2012-01-12 06:13:24 -08:00
Chris Robinson c543611897 Add some casts 2012-01-11 00:55:02 -08:00
Chris Robinson d88596e13c There's no need to pass a buffer to avio_alloc_context 2012-01-11 00:43:24 -08:00
Chris Robinson c59bbe7c22 Handle a couple of special AVSEEK_ seek modes 2012-01-11 00:32:16 -08:00
Chris Robinson d4670faafd Fix compiling the example program, and support AL_SOFT_buffer_samples with it 2012-01-10 02:12:54 -08:00
Chris Robinson e772fb7d0e Finalize AL_SOFT_buffer_samples 2012-01-10 01:14:02 -08:00
Chris Robinson 29ba8f9f1a Remove the unnecessary AL_MULAW_SOFT, AL_ALAW_SOFT, and AL_IMA4_SOFT defines 2012-01-10 00:59:10 -08:00
Chris Robinson 61af4d715c Append _SOFT to the AL_SOFT_buffer_samples enums 2012-01-10 00:41:05 -08:00
Chris Robinson 535091e36c Make LockDevice, LockContext, etc, inline 2012-01-09 23:37:03 -08:00
Chris Robinson 5b7f8c1816 Check earlier for ffmpeg 2012-01-02 11:52:20 -08:00
Chris Robinson 18216ab1a9 Remove the warning when a context being released is globally current
Just about every implementation allows this, and there's no direct harm caused
by it. However, leave the warnings for when it's still current on specific
threads, as this can cause issues.
2012-01-01 15:36:31 -08:00
Chris Robinson 5500ed9209 Use the device lock when removing the context from the device's list 2012-01-01 15:22:58 -08:00
Chris Robinson 886f874ff3 Use a proper typedef for handling atomic pointer swaps 2012-01-01 15:00:03 -08:00
Chris Robinson 466cac328f Add an example program that streams audio using ffmpeg 2012-01-01 12:36:41 -08:00
Chris Robinson 06caac2619 Update some config option descriptions 2011-12-20 01:45:29 -08:00
Chris Robinson 2691c33b55 Use a string to specify the resampler config option 2011-12-20 01:36:01 -08:00
Chris Robinson 7f6844e48d Use a struct to associate channel enums and their angles 2011-12-20 01:17:11 -08:00
Chris Robinson a5a929ff72 Restore the old FPU mode if the effects fail to update with new device parameters 2011-12-20 00:35:32 -08:00
Chris Robinson 84b5311257 Use an explicit 0 for the "no error" enums 2011-12-03 08:27:51 -08:00
Chris Robinson 6a97f6a327 Reset the ALSA device if the number of available samples grows too large 2011-11-28 10:24:24 -08:00
Chris Robinson dfd02817fa Make ALSA XRUN when the number of available samples exceeds the buffer size 2011-11-28 09:12:30 -08:00
Chris Robinson 222d2363cd Don't fail to insert a map entry when the key exists and the limit is reached 2011-11-26 03:52:07 -08:00
Chris Robinson 14c117ffd3 Initialize listener matrix to identity 2011-11-26 03:37:37 -08:00
Chris Robinson d5df2ad3ce Define TlsDestructor for Windows static builds, too 2011-11-12 16:00:37 -08:00
Chris Robinson c56b2734fd Use loops to write reverb output 2011-11-02 08:15:10 -07:00
Chris Robinson 5eebe73f79 Use a loop to write the modulator effect output 2011-11-01 17:12:56 -07:00
Chris Robinson 21d400e3c6 Use panning for echo instead of specifying discrete channels 2011-11-01 17:03:11 -07:00
Chris Robinson d270188578 Append _SOFT to the ALC loopback format enums 2011-11-01 16:00:47 -07:00
Chris Robinson fcf9034c2b Calculate the listener matrix when a new orientation is specified
This is so the matrix isn't derived each time a source is updated, and it will
make supporting user-defined matrices easier.
2011-10-30 08:27:24 -07:00
Chris Robinson 956e6f95ec Don't translate the listener velocity 2011-10-30 05:49:17 -07:00
Chris Robinson 50913948a0 Don't transform/translate vectors with an invalid listener orientation 2011-10-30 05:47:14 -07:00
Chris Robinson a5c02556e6 Add a work-around for cross-compiling with MinGW 2011-10-27 21:22:00 -07:00
Chris Robinson 319f40462f Initialize/Uninitialize COM as needed 2011-10-24 22:11:16 -07:00
Chris Robinson e5374f60c9 Don't close devices that are left open on lib shutdown 2011-10-24 08:23:41 -07:00
Chris Robinson 75679c959a Don't update the device update parameters when PulseAudio does an async buffer update
It risks a deadlock, and though the UpdateSize won't change, we'll still fill
as much of the buffer as we can up to a multiple of it
2011-10-19 16:07:56 -07:00
Chris Robinson 628356a294 Set AL_STOPPED after setting AL_PLAYING when a source has nothing to play 2011-10-15 10:33:30 -07:00
Chris Robinson 0fba3b0e1e Use the local Device variable 2011-10-11 22:30:58 -07:00
Chris Robinson 92a650c827 Use the local variable instead of re-reading the source property 2011-10-11 22:29:06 -07:00
Chris Robinson 3bbde05e0e Use the square root to build the panning table instead of sin/cos 2011-10-10 09:39:02 -07:00
Chris Robinson 3324bab81e Mark a couple more arrays with restrict 2011-10-08 03:08:13 -07:00
Chris Robinson 81133769de Return the key's value from the map when it's removed 2011-10-06 06:39:13 -07:00
Chris Robinson a7d3779dfa Check for the HRTF object instead of a flag 2011-10-06 01:16:07 -07:00
Chris Robinson 75fee8c11f Add min/max/clamp methods for 64-bit int 2011-10-04 22:39:35 -07:00
Chris Robinson 9e11b0a8c6 Fix maximum BufferSize calculation 2011-10-04 22:35:01 -07:00
Chris Robinson 5861920453 Rename a variable to avoid confusion 2011-10-04 22:16:54 -07:00
Chris Robinson eb277cc578 Remove some unneeded templates and parameters 2011-10-04 09:55:36 -07:00
Chris Robinson 20fd648729 Convert samples to float when copying to the stack 2011-10-04 09:47:08 -07:00
Chris Robinson 11caba9807 Use sample frames when handling the buffer length 2011-10-04 02:08:45 -07:00
Chris Robinson 35b4b31d57 Track the temp stack buffer using sample frames instead of bytes 2011-10-03 10:06:09 -07:00
Chris Robinson a5c74e324c Make sure GCC properly errors when checking visiblity attributes 2011-10-02 22:12:31 -07:00
Chris Robinson e1cbf06653 Don't use a double when calculating the HRTF coefficients 2011-10-02 22:05:03 -07:00
Chris Robinson 5a1959f0cf Make a floating-point constant a float type 2011-10-02 22:01:51 -07:00
Chris Robinson 7f5074949e Don't use inline asm with MSVC on 64-bit 2011-10-02 21:44:08 -07:00
Chris Robinson cf10a8d321 Get rid of some unnecessary casting 2011-10-02 16:54:45 -07:00
Chris Robinson a05006b06a Rename some parameters 2011-10-02 07:31:26 -07:00
Chris Robinson eb0b96752a Remove the OriginalAlign buffer field, and calculate it as needed 2011-10-01 19:52:07 -07:00
Chris Robinson 2f453eba35 Fix potential divide-by-0 2011-10-01 08:37:42 -07:00
Chris Robinson 8b2e1fdd9a Add buffer properties to get the internal format, and the length in bytes, samples, and seconds
The provided buffer lengths correspond to the source offsets, in that the byte
length specifies the end of the byte offset (ie, when the buffer is used for a
static source, the offset will range between 0 (inclusive) and the byte length
(exclusive)). Although an application could use the AL_SIZE, AL_CHANNELS,
AL_BITS, and AL_FREQUENCY properties to find the length in samples and seconds,
the byte length cannot be reliably calculated this way.
2011-10-01 06:19:55 -07:00
Chris Robinson c99b32a8ec Use tokens matching the string 2011-10-01 04:56:45 -07:00
Chris Robinson 92f95e4d94 Add support for AL_EXT_ALAW 2011-10-01 04:55:03 -07:00
Chris Robinson d1011f8d37 Add AL_EXT_MULAW and AL_EXT_ALAW tokens to alext.h 2011-10-01 04:05:39 -07:00
Chris Robinson a486045f7c Improve index calculation for reverb modulation updates 2011-10-01 02:38:28 -07:00
Chris Robinson 0e7dc93681 Add some formatting spaces to the reverb state struct 2011-10-01 02:09:17 -07:00
Chris Robinson a7bbf7a741 Rename LOWPASSFREQCUTOFF to LOWPASSFREQREF 2011-09-30 23:07:15 -07:00
Chris Robinson bfa782f329 Define the values for F_PI and F_PI_2 directly instead of using M_PI 2011-09-30 22:53:47 -07:00
Chris Robinson bed37a148c Use raise(SIGTRAP) instead of kill(getpid(),SIGTRAP) 2011-09-30 20:46:18 -07:00
Chris Robinson bf19186223 Apply the initial wet send reverb decay before clamping the gains 2011-09-30 17:51:21 -07:00
Chris Robinson 087e75d47f Multiply by the reciprocal value instead of dividing 2011-09-29 22:29:10 -07:00
Chris Robinson 9e2600cac4 Use fast float-to-int conversions for some mixer-related code 2011-09-29 06:55:00 -07:00
Chris Robinson 3c27c24dec Set the mixer FPU mode when setting a new effect on an effect slot 2011-09-29 05:25:01 -07:00
Chris Robinson 1b81642db5 Set the mixer FPU mode for device parameter updates 2011-09-29 05:15:20 -07:00
Chris Robinson d3a6de3f86 Set the FPU into single-precision mode for mixer updates 2011-09-29 05:10:15 -07:00
Chris Robinson f4925a0e6a Use inline functions to set/restore the FPU mode for mixer updates 2011-09-29 04:03:18 -07:00
Chris Robinson b6b3ca6e6f Use inline assembly for fast float-to-int conversions 2011-09-29 03:51:46 -07:00
Chris Robinson 53572da7de Set toward-zero rounding when updating in alDeferUpdatesSOFT 2011-09-28 22:02:36 -07:00
Chris Robinson 8a0e0265bc Clamp the low-pass reference gain to -60 dB instead of -40 dB 2011-09-25 11:32:51 -07:00
Chris Robinson 0ada7b2da5 Copy only the gains needed for the channel being mixed 2011-09-24 22:24:26 -07:00
Chris Robinson 1dee1245ff A static source only has one buffer in the queue 2011-09-24 21:21:59 -07:00
Chris Robinson e9da6950ee Centralize the Lookup and Remove macros 2011-09-24 18:34:45 -07:00
Chris Robinson 12abd75aed Only warn when a symbol fails to load 2011-09-24 14:52:54 -07:00
Chris Robinson 044774ed72 Rename the MMDevAPI backend's MessageProc function 2011-09-24 13:55:44 -07:00
Chris Robinson cf56b0733b Look for and use atan2f, log10f, and floorf 2011-09-24 12:17:39 -07:00
Chris Robinson 2b0a63003f Cast a 64-bit variable to a 32-bit int 2011-09-24 00:52:58 -07:00
Chris Robinson deef98658b Don't require libm for the float-type math functions 2011-09-24 00:42:14 -07:00
Chris Robinson d9544d006b Use aluAtan instead of atan 2011-09-24 00:04:34 -07:00
Chris Robinson 6a2b2a0aed Silence MSVC warnings in the HRTF code 2011-09-23 23:16:42 -07:00
Chris Robinson 9060d0cc36 Add a WAVEFORM_FRACONE macro and minor cleanups for the ring modulator 2011-09-23 23:11:35 -07:00
Chris Robinson b4f9f89480 Use float types for the resamplers instead of double 2011-09-23 23:03:59 -07:00
Chris Robinson a9b6b284c0 Change a double type to a float 2011-09-23 22:50:33 -07:00
Chris Robinson da62f50528 Check for asinf and use it 2011-09-23 22:44:34 -07:00
Chris Robinson e84e38cc19 Make AIRABSORBGAINHF a float type 2011-09-23 22:37:20 -07:00
Chris Robinson 9266a37276 Move ConeScale and ZScale to ALu.c and alu.h, and make them floats 2011-09-23 22:33:37 -07:00
Chris Robinson ff8ee68852 Add atomic functions for GCC inline asm 2011-09-23 12:42:13 -07:00
Chris Robinson 470b506952 Use macros for float-typed PI values, to avoid manual casts everywhere 2011-09-22 11:17:01 -07:00
Chris Robinson de8a447ea5 More MSVC precision casts 2011-09-22 01:20:49 -07:00
Chris Robinson b2f4520ba1 Add a few more precision casts 2011-09-22 01:09:02 -07:00
Chris Robinson a4b1239f45 Use cosf and sinf when available
Also clear away a few more MSVC precision warnings
2011-09-22 01:00:44 -07:00
Chris Robinson 43350f9066 Silence some MSVC precision warnings 2011-09-22 00:35:08 -07:00
Chris Robinson 0988023d30 Constify some parameters 2011-09-22 00:31:42 -07:00
Chris Robinson e81e20446f Don't use the -fvisibility switch if the visibility attributes didn't work 2011-09-21 23:41:52 -07:00
Chris Robinson 64080250ff Actually check for hidden visibility 2011-09-20 18:35:27 -07:00
Chris Robinson 4e19f82a2b Try to use "default" and "hidden" visibility when "protected" and "internal" aren't available 2011-09-20 16:53:05 -07:00
Chris Robinson 97743e32c2 Remove a couple type size checks 2011-09-20 15:47:45 -07:00
Chris Robinson b8d53fb7e9 Check for SLES/OpenSLES.h along with SLES/OpenSLES_Android.h 2011-09-20 15:43:00 -07:00
Chris Robinson 390d94ec97 Define _WIN32_WINNT on the command line with _WIN32 2011-09-20 14:59:01 -07:00
Chris Robinson e01092a0da Move Sleep implementation into helper.c and emulate sched_yield for Windows 2011-09-20 14:43:53 -07:00
Chris Robinson 312e9a436d Fix indentation 2011-09-20 13:58:45 -07:00
Chris Robinson e43470da7a Make a separate log level to trace reference counts, to avoid log spam for traces 2011-09-20 12:24:23 -07:00
Chris Robinson d3e7a26db6 Fix a warning message 2011-09-20 11:58:26 -07:00
Chris Robinson 8c2a2a5c9d Trace created device and context pointers 2011-09-20 11:55:40 -07:00
Chris Robinson bf2ac6625c Avoid size_t and ssize_t in some places 2011-09-20 11:40:56 -07:00
Chris Robinson edb386eb1e Add a ConfigValueStr function, to return the string through a parameter 2011-09-19 11:29:18 -07:00
Chris Robinson cc3dd648ac Check the number of available samples before calling the capture method 2011-09-18 20:27:34 -07:00
Chris Robinson 98b2fa6cb5 Use the ALC types for the device format enum values 2011-09-18 19:43:59 -07:00
Chris Robinson b615c7d0a2 Use arrays instead of large switches to decompose formats 2011-09-18 19:06:19 -07:00
Chris Robinson 7e06a10f73 Return int and float config values through a parameter
This allows the getter functions to return whether or not the option exists
without a separate call and check.
2011-09-18 16:16:55 -07:00
Chris Robinson d46db2f648 Avoid a buffer underrun when getting the end of a config option value 2011-09-18 11:46:39 -07:00
Chris Robinson 6d77aabb23 Update a config file comment 2011-09-18 11:25:44 -07:00
Chris Robinson a7b9ee6bcb Allow loading multiple HRTF tables through a comma-separated list 2011-09-18 11:10:32 -07:00
Chris Robinson 0e5c221a3c Dynamically allocate space for HRTF tables 2011-09-18 10:09:32 -07:00
Chris Robinson 25f7c27043 Store the HRTF tables used in the device 2011-09-18 09:52:40 -07:00
Chris Robinson ec3b78e52d Fix a mutex leak in a capture open error path 2011-09-16 02:17:51 -07:00
Chris Robinson 4e903c84cf Fix comment on external HRTF file format 2011-09-15 02:27:02 -07:00
Chris Robinson ae3ec4e767 Trace compatible HRTF formats when loading instead of on error 2011-09-15 02:05:48 -07:00
Chris Robinson fd22226242 Make alcSetError static 2011-09-14 11:58:51 -07:00
Chris Robinson fe1e046ea6 Remove a couple ineffective alcSetError calls 2011-09-14 11:55:57 -07:00
Chris Robinson 2171aaa35c Use IsDebuggerPresent instead of the VC-only __try/__except 2011-09-14 02:10:42 -07:00
Chris Robinson 97024151e4 Return an error from the CaptureSamples method instead of setting it 2011-09-14 02:01:35 -07:00
Chris Robinson 62dfea43c6 Limit the minimum ALSA capture size to 100ms, and the maximum period size to 50ms 2011-09-13 23:51:22 -07:00
Chris Robinson a4f8408150 Use maxu to clamp PulseAudio's capture buffer size 2011-09-13 21:53:34 -07:00
Chris Robinson 80f2b787ef Pre-apply the deferred source offset only when it's playing or paused 2011-09-12 06:04:40 -07:00
Chris Robinson 404cfde33e Rename the ALEffect_ macros to ALeffectState_ to reflect what they work on 2011-09-12 05:59:23 -07:00
Chris Robinson 145bc6da6b Be a bit more robust while handling the global device list 2011-09-12 03:57:53 -07:00
Chris Robinson 69b10e2d7c Remove the context from being current before removing it from the device 2011-09-12 03:35:21 -07:00
Chris Robinson 25e86e8e91 Get rid of IsContext 2011-09-12 03:18:27 -07:00
Chris Robinson 780417ba39 Use VerifyDevice to validate a device handle instead of IsDevice 2011-09-12 02:17:31 -07:00
Chris Robinson 9f159bfb33 Look for __ALSOFT_TRAP_ERROR to trap both AL and ALC errors 2011-09-12 01:50:57 -07:00
Chris Robinson c2e9cda2d1 Shorten some warning messages 2011-09-12 01:21:54 -07:00
Chris Robinson 1d6ecee64f Use the active source list when stopping sources on disconnect 2011-09-12 01:11:46 -07:00
Chris Robinson 3ee2826565 Minor cleanup for applying click-removal to wet buffers 2011-09-12 01:09:01 -07:00
Chris Robinson 0ceea27b44 Remove the effect slot parameter from the effect process method 2011-09-12 00:44:52 -07:00
Chris Robinson 0dc5837b82 Fix a possible race condition when setting an effect on a slot 2011-09-11 20:51:30 -07:00
Chris Robinson b4cbb3380a Fix non-GCC CompExchangePtr implementations 2011-09-11 09:39:29 -07:00
Chris Robinson d9cb2baf4a Don't increment the context reference count when locked 2011-09-11 09:29:24 -07:00
Chris Robinson 8f456f68ff Remove GetLockedContext 2011-09-11 09:28:30 -07:00
Chris Robinson 54a421ec58 Get rid of some more GetLockedContext calls 2011-09-11 09:13:01 -07:00
Chris Robinson 6ce420afe3 Use function pointers to set type-specific effect properties 2011-09-11 08:59:47 -07:00
Chris Robinson 8028770901 Avoid accessing EAX-only reverb properties for non-EAX updates 2011-09-11 08:50:48 -07:00
Chris Robinson d51b93f869 Remove the union from the effect parameter types 2011-09-11 07:42:23 -07:00
Chris Robinson ab2b62c980 Use function pointers to set type-specific filter properties 2011-09-11 07:34:03 -07:00
Chris Robinson 97fa7cc9ac More lock avoidance 2011-09-11 06:58:49 -07:00
Chris Robinson 2648a9b011 Just reference the context when checking for extensions 2011-09-11 04:32:34 -07:00
Chris Robinson 1fdc25df75 Use a RWLock to protect access to a buffer instead of the device lock 2011-09-11 03:57:40 -07:00
Chris Robinson 4db24af59a Avoid holding the context lock for the buffer functions 2011-09-11 02:37:52 -07:00
Chris Robinson d85c12c385 Avoid holding the context lock when setting auxiliary slot properties 2011-09-11 02:01:31 -07:00
Chris Robinson ccd8cbc2a9 Make some listener properties volatile 2011-09-11 01:26:09 -07:00
Chris Robinson 1075cce7b3 Mark some source properties volatile 2011-09-11 01:18:57 -07:00
Chris Robinson 89f438df71 Small fixes to setting the source's direct filter and auxiliary send 2011-09-11 00:58:33 -07:00
Chris Robinson 97027a7ad9 Avoid holding the context lock while setting source parameters
Note that some parameters still need the lock when getting or setting them.
2011-09-11 00:47:31 -07:00
Chris Robinson 3a8ef2f0d6 Initialize source parameters before adding the source to the map 2011-09-10 21:21:07 -07:00
Chris Robinson 4c8e6dead1 Don't defer stopped and initial source state changes
Typically when these are set, an app will be releasing the buffer from the
source, possibly to immediately reuse it. There would be an extra burden placed
on the app if it was required to process updates before it could fully release
the buffer from the source.
2011-09-10 21:02:50 -07:00
Chris Robinson c19a7408d9 Add a comment about when FreeContext and FreeDevice are called 2011-09-10 20:52:19 -07:00
Chris Robinson c7b9d27b85 Trace the device pointer when warning about leftover objects 2011-09-10 20:49:16 -07:00
Chris Robinson 30510dc478 Set the device-connected flag to false first when handling a disconnect 2011-09-10 19:22:46 -07:00
Chris Robinson b788ae52b0 Condense some code 2011-09-10 19:19:32 -07:00
Chris Robinson ccbd67eab6 Clean up some code comments 2011-09-10 19:14:14 -07:00
Chris Robinson f21b1eefda Lock the lists while probing devices 2011-09-10 09:12:02 -07:00
Chris Robinson 6cd55a0a10 Clean up some comments 2011-09-10 09:00:01 -07:00
Chris Robinson 0c5c63f4ba Slightly modify a warning 2011-09-10 07:49:35 -07:00
Chris Robinson 3dcd1e793e Properly destroy contexts that are left on the device when it's closed 2011-09-10 07:35:48 -07:00
Chris Robinson 419294cddd Avoid locking the device when adding and removing contexts
Also, don't keep track of the number of contexts per device
2011-09-10 07:18:29 -07:00
Chris Robinson b9448a476e Avoid holding the list lock when not needed 2011-09-10 05:19:08 -07:00
Chris Robinson 238a747414 Warn about contexts that are errantly left current
These won't show by default in release mode, unless the log level is changed to
show warnings.
2011-09-10 03:18:31 -07:00
Chris Robinson c35433ec7a Hold an extra reference on the device for each of its contexts 2011-09-10 03:01:24 -07:00
Chris Robinson e8c9da0124 Add some reference counting to the device 2011-09-10 02:43:07 -07:00
Chris Robinson 18cf6e3210 Fix a parameter type 2011-09-10 02:13:52 -07:00
Chris Robinson c9205a58ae Use alGetEnumValue to check for supported filters and effects 2011-09-10 02:09:18 -07:00
Chris Robinson 23d693cd06 Add an option to trap device errors in a debugger 2011-09-10 01:23:59 -07:00
Chris Robinson 724ad0d893 Add an option to trap context errors in a debugger 2011-09-10 01:12:34 -07:00
Chris Robinson 7a0df8553f Allow PulseAudio's tlength to grow to make sure there's at least two updates 2011-09-09 22:30:24 -07:00
Chris Robinson 477215883f Use an array to get the device format from the user config string 2011-09-09 21:33:18 -07:00
Chris Robinson 05114815ba Avoid an unnecessary lock 2011-09-02 15:57:50 -07:00
Chris Robinson 9080d5fda0 Do an atomic compare-exchange on the global context when destroying a context 2011-09-02 02:57:21 -07:00
Chris Robinson 108b43458f Fix multichannel wet gainhf calculation 2011-09-02 02:54:00 -07:00
Chris Robinson 3dfd67680c Use an atomic exchange to get the old global context when setting a new one 2011-09-02 02:32:35 -07:00
Chris Robinson 3a2fa18744 Log the function name instead of the source file and line number 2011-09-02 02:00:37 -07:00
Chris Robinson 370c9f553d Trace the reverb buffer size 2011-09-02 01:13:22 -07:00
Chris Robinson e80aa008b2 Combine the reverb effects
Updating and processing still differs depending on whether standard or EAX
reverb is used or not. The only functional difference should be that the
allocated buffer (and subsequent offsets) take into account the modulation
and echo times.
2011-09-01 20:06:51 -07:00
Chris Robinson 812d91cbf8 Don't recreate the effect when switching between the dedicated effect types 2011-09-01 18:51:24 -07:00
Chris Robinson 942c8bf8a7 Combine the dedicated (dialog/LFE) effects 2011-09-01 18:01:14 -07:00
Chris Robinson b6b3e2f1cc Fix a comment 2011-08-31 14:01:15 -07:00
Chris Robinson ee5ff943f3 Fix a potential infinite loop when removing an effect slot 2011-08-31 13:36:47 -07:00
Chris Robinson b28f48c1bd Store the filter gains directly in the source instead of duplicate filter objects 2011-08-31 02:18:16 -07:00
Chris Robinson 49e2fa428f Avoid the context lock when generating and deleting effects and filters 2011-08-31 02:15:03 -07:00
Chris Robinson 2fcf97e207 Avoid the context lock when generating and deleting buffers 2011-08-31 01:13:02 -07:00
Chris Robinson f3ac3cd1e2 Don't set a thread priority if real-time isn't requested 2011-08-31 01:05:01 -07:00
Chris Robinson dd74757319 Add a function to initialize a RWLock 2011-08-31 01:00:19 -07:00
Chris Robinson ab0b5d9890 Hold the context lock only as needed when handling the listener 2011-08-31 00:33:33 -07:00
Chris Robinson 23f576360c Avoid locking the context while getting and setting global state 2011-08-31 00:19:27 -07:00
Chris Robinson d7bd9c7559 Use the active effect slot list for pending updates in alDeferUpdatesSOFT 2011-08-30 23:33:01 -07:00
Chris Robinson 9fb91f70aa Rename GetReffedContext to GetContextRef 2011-08-30 23:28:38 -07:00
Chris Robinson a32a3bcda8 Use a more appropriate error for trying to delete an effect slot that's in use 2011-08-30 20:57:14 -07:00
Chris Robinson 16dc73ff48 Hold the lock only as needed when generating and deleting effect slots 2011-08-30 20:49:49 -07:00
Chris Robinson 0a1321aaae Add a limit to the UIntMap size and use it for sources and effect slots 2011-08-30 20:33:47 -07:00
Chris Robinson 189add1d5a Use a separate array for the auxiliary slots in the mixer 2011-08-30 20:13:42 -07:00
Chris Robinson 755062fb76 Fix compilation with MSVC 2011-08-30 17:55:52 -07:00
Chris Robinson 7d577832cd Add functions to retrieve the source from the source map while removing it 2011-08-30 17:32:49 -07:00
Chris Robinson c4866afbe0 Use a RWLock and atomic exchanges for thunk handling 2011-08-30 17:24:58 -07:00
Chris Robinson ee60248d19 Lock the context while deleting sources only as needed 2011-08-30 00:33:01 -07:00
Chris Robinson d546813c05 Don't lock the context while checking the validity of objects 2011-08-30 00:04:02 -07:00
Chris Robinson f02d9e22d7 Do a pointer exchange when replacing the source buffer queue 2011-08-29 23:55:24 -07:00
Chris Robinson cc67f45892 Use the reference counting functions for the buffer and auxiliary slot refs 2011-08-29 23:10:02 -07:00
Chris Robinson c6a110ec59 Hold the context lock as needed while generating and deleting sources 2011-08-29 22:51:18 -07:00
Chris Robinson 7408396fd4 Use a generic int type to handle enum swaps 2011-08-29 21:30:12 -07:00
Chris Robinson da081b81c4 Do a compare-exchange to set the context error
This allows for unlocked context access when getting the error
2011-08-29 20:52:13 -07:00
Chris Robinson 500ad776ea Exchange the AL error when retrieving it 2011-08-29 20:34:47 -07:00
Chris Robinson e4a2b69b37 Make specific functions to lock/unlock UIntMap access 2011-08-29 20:21:28 -07:00
Chris Robinson 8eaa9bb469 Manually lock the map while iterating through its contents 2011-08-29 20:05:50 -07:00
Chris Robinson b283dd3682 Use a read-write lock to protect access to the UInt maps 2011-08-29 19:15:22 -07:00
Chris Robinson 4a5dd73a3c Make the source's NeedsUpdate volatile 2011-08-29 13:43:59 -07:00
Chris Robinson 950570b8c3 Do a proper exchange when setting a new state 2011-08-29 13:29:37 -07:00
Chris Robinson 72beb577b6 Lock the context as needed for the defer and process calls 2011-08-29 13:22:07 -07:00
Chris Robinson de65ee08c9 Make ALCcontext's DeferUpdates field volatile 2011-08-29 11:55:01 -07:00
Chris Robinson c06f4eb7f7 Hold a reference on the global context 2011-08-29 11:10:17 -07:00
Chris Robinson 0abe13e3ad GCC's atomic built-ins were introduced in 4.1.0, so check for that version 2011-08-29 10:51:58 -07:00
Chris Robinson 2ac808ffa2 Fix typo 2011-08-29 01:01:20 -07:00
Chris Robinson 58078e2c1e Use atomic exchanges when checking for updates to objects' internal parameters 2011-08-29 00:50:55 -07:00
Chris Robinson 01503f8a7b Use an enum instead of an boolean for some flags
They still only hold AL_TRUE or AL_FALSE, but some systems can't properly
handle atomic swaps of 1-byte types.
2011-08-29 00:03:09 -07:00
Chris Robinson aa99e1220b Increment the context reference count when locking 2011-08-28 23:49:50 -07:00
Chris Robinson 783375af56 Use a list of contexts in the device instead of an array 2011-08-28 19:28:41 -07:00
Chris Robinson 1d9f21f6dc Avoid a global list of contexts 2011-08-28 18:53:23 -07:00
Chris Robinson 0900c281a3 Check if the context is valid only if it's replacing the current thread context 2011-08-28 18:17:06 -07:00
Chris Robinson f8c4b16c30 Hold a reference on the context when being set as current on the thread
This should vastly improve efficiency for retrieving the current thread
context, as we can be assured the stored context pointer will always be valid.
2011-08-28 18:08:48 -07:00
Chris Robinson c567339718 Rename ALCcontext_Deref to ALCcontext_DecRef 2011-08-28 17:38:55 -07:00
Chris Robinson 5eceb593e9 Emulate pthread TLS functions in Windows 2011-08-28 17:21:01 -07:00
Chris Robinson 82244f298c Add reference counting to the ALC contexts 2011-08-28 15:44:03 -07:00
Chris Robinson 3ab20d4ccb Print an error for invalid speaker layout keys 2011-08-28 10:25:40 -07:00
Chris Robinson ccc2644e6b Free the WaveIn buffers' data after unpreparing them all 2011-08-25 11:33:13 -07:00
Chris Robinson 7e2155d937 Return an ALC error enum from the OpenCapture backend method 2011-08-24 14:44:15 -07:00
Chris Robinson c696d4dbb2 Return an ALC error enum from the OpenPlayback backend method 2011-08-24 14:24:48 -07:00
Chris Robinson f5195ee4f2 Remove the Buffer field from the source 2011-08-23 09:55:22 -07:00
Chris Robinson 9f25ad42c1 Explicitly set BufferListStart's prev field 2011-08-23 09:27:35 -07:00
Chris Robinson 1d4dc7d7a9 Add AL_SOFTX_deferred_updates to the context's extension list 2011-08-22 18:57:28 -07:00
Chris Robinson 65d42083e1 Prevent source and effect slot updates from occuring while updates are deferred 2011-08-22 17:13:03 -07:00
Chris Robinson 6992a67dd9 Remove some g_ prefixes 2011-08-22 07:50:08 -07:00
Chris Robinson 922f5fa081 Rename a couple variables 2011-08-22 07:44:22 -07:00
Chris Robinson 199df253ca Always use the thunk functions to get object IDs 2011-08-22 07:40:14 -07:00
Chris Robinson c1f435cc18 Don't use the thunk array for object lookups 2011-08-22 07:22:02 -07:00
Chris Robinson 858592832f Defer source offset changes requested by the app 2011-08-21 00:49:04 -07:00
Chris Robinson a97211b572 Use -1 to mark an invalid source offset instead of 0 2011-08-21 00:35:37 -07:00
Chris Robinson f196a9fc67 Defer source state changes from alSourcePlay/Pause/Stop/Rewind calls 2011-08-20 23:59:24 -07:00
Chris Robinson 7ea71d8bfa Use a separate function to set the source state 2011-08-20 06:31:10 -07:00
Chris Robinson 9b615a0add Remove some spaces 2011-08-20 04:58:50 -07:00
Chris Robinson 4540034039 Move the GUID initialization to the helper 2011-08-20 04:55:11 -07:00
Chris Robinson 29d58f3d84 Set the loopback backend functions first 2011-08-20 04:43:47 -07:00
Chris Robinson 5290c820b4 Print the ALC_DEFAULT_ALL_DEVICES_SPECIFIER when available 2011-08-20 04:16:22 -07:00
Chris Robinson 22ad15e1f1 Move some functions into a separate helper source file 2011-08-20 04:01:07 -07:00
Chris Robinson 80b8abb0ec Don't enumerate "PulseAudio Default" in the ALL_DEVICES list 2011-08-20 03:00:38 -07:00
Chris Robinson 529ff3197a Don't expose "DirectSound Default" in the ALL_DEVICES list
This is under the assumption that one of the devices in ALL_DEVICES corresponds
to the default. This is not true for all backends (eg, ALSA).
2011-08-20 02:49:15 -07:00
Chris Robinson b036bc9e04 More closely emulate pthread_once in Windows 2011-08-20 01:07:27 -07:00
Chris Robinson 9989f33fc2 Move backend sources to a separate sub-directory 2011-08-20 00:41:47 -07:00
Chris Robinson ba4456d24a Use the proper variable type 2011-08-19 23:05:54 -07:00
Chris Robinson 00718ca03f Use the string ID instead of numeral index to reference the ALSA card
The index has a habit of changing, where the ID should be more consistent.
2011-08-19 20:47:08 -07:00
Chris Robinson 6b8475b435 Add options to specify alternate ALSA device prefixes 2011-08-19 04:02:36 -07:00
Chris Robinson e51295c79e Don't scale the wet send by the number of channels
Each channel should be treated as a distinct audio source, so multiple channels
should act as multiple sources.
2011-08-19 03:59:57 -07:00
Chris Robinson b40aa8636b Remove a couple unneeded variables 2011-08-19 03:01:57 -07:00
Chris Robinson b3645df56f Delete the mutex if the device fails to open 2011-08-19 02:58:21 -07:00
Chris Robinson 711f0a29ec Make sure there's a valid backend before trying to open a device 2011-08-19 02:55:36 -07:00
Chris Robinson e8e1286572 Remove the "via *" portion of the enumerated devices 2011-08-19 01:59:03 -07:00
Chris Robinson 3d722b9ba0 Don't remove backends that fail to initialize 2011-08-19 01:54:55 -07:00
Chris Robinson 4ac0a507e0 Use only the first available backend for playback or capture devices
This should help cut down on the number of duplicate devices when enumerating
on a system that supports multiple backends. It will also avoid the problem of
backends that don't play well with others (eg, PulseAudio).
2011-08-19 01:41:16 -07:00
Chris Robinson d54908d946 Attempt to connect a context when pulseaudio initializes 2011-08-19 01:40:16 -07:00
Chris Robinson 6ba5128f72 Use separate lists for playback and capture backends 2011-08-19 00:30:05 -07:00
Chris Robinson c96a6c2b1b Remove a typedef 2011-08-18 23:58:00 -07:00
Chris Robinson 0b0588cc37 Load backend libs on init 2011-08-18 23:56:11 -07:00
Chris Robinson 742bb12498 Make DSoundLoad return a boolean 2011-08-18 01:17:42 -07:00
Chris Robinson f00017c8c1 Avoid some LP types 2011-08-18 00:31:44 -07:00
Chris Robinson d976d02936 Only attempt to load MMDevApi and DSound once, on init 2011-08-18 00:13:43 -07:00
Chris Robinson 45f0886eba Set the proper union member to 0 2011-08-17 20:44:05 -07:00
Chris Robinson 057fef88ae Remove backends that fail to initialize 2011-08-17 06:15:43 -07:00
Chris Robinson 8148626ba3 Allow the backend init functions to return failure 2011-08-17 06:05:26 -07:00
Chris Robinson 14ffb8bd9b Coalesce some mixer sample converters 2011-08-17 02:33:25 -07:00
Chris Robinson 2f8510db31 Avoid reading some more source parameters multiple times when updating 2011-08-17 01:54:09 -07:00
Chris Robinson 38a4d9a6c5 Get rid of the remaining min/max usage 2011-08-16 18:53:04 -07:00
Chris Robinson 9f5bf5f930 Rename minF/maxF/clampF to minf/maxf/clampf for consistency 2011-08-16 18:40:21 -07:00
Chris Robinson 04dad28228 Use mini/maxi/clampi and minu/maxu/clampu to replace min/max calls 2011-08-16 18:33:10 -07:00
Chris Robinson a96f817738 Avoid a float->int->float conversion 2011-08-16 06:13:55 -07:00
Chris Robinson 8a51a7ea2d Use inline minF/maxF/clampF functions instead of the __min/__max macros 2011-08-16 04:21:58 -07:00
Chris Robinson b3cb511c06 Fix wet path conehf calculation 2011-08-13 06:58:05 -07:00
Chris Robinson 7e3d02d991 Make sure there's a buffer to render loopback samples to 2011-08-12 15:44:21 -07:00
Chris Robinson 412cec73de Avoid a temporary throw-away buffer for mixing null output 2011-08-12 15:42:36 -07:00
Chris Robinson a72c0f7470 Use an existing function to clear and free the context 2011-08-12 05:40:09 -07:00
Chris Robinson 45a245d7de Round the HRTF sample step count 2011-08-11 21:43:21 -07:00
Chris Robinson 1e60481040 Use ARM Neon vector ops to apply HRTFs when available 2011-08-11 19:02:45 -07:00
Chris Robinson cda4c738e8 Don't declare i in a for() statement 2011-08-08 13:34:42 -07:00
Chris Robinson f7c176e8ca Use a message thread to handle MMDevAPI backend methods
Because MMDevAPI uses COM, CoInitialize must be called on the thread(s) where
the device calls are made. OpenAL has no such requirement, thus no guarantee
that COM will be (or remain) initialized on the calling thread(s).
2011-08-03 10:42:29 -07:00
Chris Robinson 3e2bc847d8 Avoid closing devices and deinitializing backends on process exit
They are still closed and deinitialized when the DLL is simply being detached
(via FreeLibrary), however.
2011-08-03 09:46:18 -07:00
Chris Robinson b5092a6212 Delay reading the config file on Windows, too 2011-08-03 09:37:58 -07:00
Chris Robinson 81d401eab5 Use specialized click removal loops for mono and stereo output 2011-07-30 05:43:47 -07:00
Chris Robinson f986f7dfad Wait until all headers are unprepared before freeing the buffer 2011-07-28 08:27:52 -07:00
Chris Robinson bcbc1dea25 Make the thread event auto-resetting 2011-07-28 08:23:19 -07:00
Chris Robinson c8326ab328 Remove the unnecessary wave header event
It's only ever set before sending the quit message, which will set the thread
event. It's also only waited on just before the thread event is used to signal
the end of the thread.
2011-07-28 08:18:43 -07:00
Chris Robinson 91c6bc58dc Use unnamed events for winmm 2011-07-28 08:16:09 -07:00
Chris Robinson c722579c18 Make the DSound and MMDevAPI events initially non-signaled 2011-07-24 05:29:43 -07:00
Chris Robinson 2f3bd5b5d9 Move an include to where it makes more sense to be 2011-07-23 05:53:53 -07:00
Chris Robinson fad80a4f92 Don't cast for the float versions of math functions 2011-07-23 05:52:10 -07:00
Chris Robinson 1b0a55f44c Use real-time priority by default on Windows
It's actually "time-critical", but it seems normal priority doesn't cope as
well when the system is under moderate load.
2011-07-23 05:44:55 -07:00
Chris Robinson 14b359beb0 Fix cmake check 2011-07-23 04:36:03 -07:00
Chris Robinson f32876d4bd Check for the finite() function along with _finite() 2011-07-23 04:23:54 -07:00
Chris Robinson 13f9dbe582 Look for ieeefp.h and include it when appropriate 2011-07-23 04:06:29 -07:00
Chris Robinson 8dfa2560b1 Use a better method for determining if shared functions exist 2011-07-23 00:48:19 -07:00
Chris Robinson 0c8c3b16d2 Round UpdateSize to the nearest multiple of MMDevAPI's period size 2011-07-21 18:53:15 -07:00
Chris Robinson 3d2812bd35 Reset the MMDevAPI and DSound events before starting playback 2011-07-21 17:04:33 -07:00
Chris Robinson a0e5cbc47f Fix DSound UpdateSize calculation when NumUpdates is clamped 2011-07-21 16:53:25 -07:00
Chris Robinson d5ddc6acee Make sure some state values are finite 2011-07-21 16:18:21 -07:00
Chris Robinson 30750d8763 Create and close the DSound event handle with the device 2011-07-21 03:38:43 -07:00
Chris Robinson 3e4659076a Use an event to signal the mmdevapi backend to mix
Also, use a multiple of the default period size for the update size, since the
event apparently triggers on period boundaries.
2011-07-21 03:32:20 -07:00
Chris Robinson 7cd6327969 Use DSound notifications to wake up the mixer thread 2011-07-21 02:48:09 -07:00
Chris Robinson 77128d298e Add an option to emulate EAX reverb using standard reverb 2011-07-20 22:53:46 -07:00
Chris Robinson 1e8a379a03 Add a config option to boost reverb output 2011-07-20 22:26:16 -07:00
Chris Robinson b889823b21 Set real-time priority for the WaveOut backend 2011-07-20 19:59:47 -07:00
Chris Robinson 9723c553b6 Scale reverb output based on the number of channels
A scale of sqrt(2/numchannels) is used so the perceived volume matches as if it
was stereo output, which seems to match other implementations.
2011-07-20 19:57:24 -07:00
Chris Robinson 6bfb0371eb Make sure some effect property values are finite 2011-07-20 03:33:06 -07:00
Chris Robinson 1348bf2476 Make sure some listener property values are finite 2011-07-20 03:27:05 -07:00
Chris Robinson 36446f3a58 Make sure source vectors are finite values 2011-07-20 03:18:46 -07:00
Chris Robinson 5a484f7493 Look for an isfinite function 2011-07-20 03:18:11 -07:00
Chris Robinson 244d60fb96 Slightly improve the gain change delta calculation 2011-07-19 00:43:31 -07:00
Chris Robinson 0ae9638aa5 Add a lost comment 2011-07-18 17:44:37 -07:00
Chris Robinson 55b893c784 Improve buffer metrics when using the MMDevAPI backend 2011-07-17 19:42:03 -07:00
Chris Robinson c9b6bc848f Simplify some reverb update calculations 2011-07-17 17:51:13 -07:00
Chris Robinson f97310c757 Apply the click removal offset first, before decreasing it 2011-07-17 07:23:19 -07:00
Chris Robinson 87b3520d02 Fix GetMovingHrtfCoeffs return type 2011-07-17 07:16:10 -07:00
Chris Robinson 887de0523f Scale the delay down once for the post-stepping mix loop 2011-07-17 02:34:35 -07:00
Chris Robinson a7a1f802a8 Invert the device channel map 2011-07-17 01:55:25 -07:00
Chris Robinson 163cc62a00 Add a couple new functions to handle deferred updates
Currently no-ops, they will be used in place of alcSuspendContext and
alcProcessContext for batching updates since the mentioned functions have no
coherent functionality between system implementations.
2011-07-16 16:59:20 -07:00
Chris Robinson 5f566ebf05 Fade between HRTF coefficients, to reduce noise from sudden changes 2011-07-16 16:24:01 -07:00
Chris Robinson 1622986467 Fix some format types 2011-07-16 04:21:40 -07:00
Chris Robinson ce6db53e5f Apply the slot gain during the effect update method 2011-07-16 03:35:51 -07:00
Chris Robinson 292ea0607d Request a slot update when the gain changes 2011-07-16 03:15:28 -07:00
Chris Robinson 3bd7256dc8 Pass the effect slot to the effect update method 2011-07-16 03:02:16 -07:00
Chris Robinson da2429a1d0 Allow effect slots to be updated asynchronously
Updates when the slot changes effect type is still sychronous, however, to
ensure a proper state for the Process method call. Fixing this would
essentially require all effects to work from the same state.
2011-07-16 02:41:02 -07:00
Chris Robinson 54b5f35125 Warn about HRTF being disabled in the compatibility check function 2011-07-16 00:39:13 -07:00
Chris Robinson 23bfb03d2a Improve the external HRTF data file format
Although more parameters are read from the file, some must currently conform to
specific values. In the future, these restrictions may be lifted.
2011-07-16 00:22:01 -07:00
Chris Robinson 7381b1a696 Make the sample rate part of the HRTF struct 2011-07-15 23:45:03 -07:00
Chris Robinson 99bcb7a0f6 Use both root paths when looking for Wine headers, and disable them by default 2011-07-13 01:49:43 -07:00
Chris Robinson 17773c5644 Rename the ERROR macro to ERR 2011-07-13 01:43:00 -07:00
Chris Robinson e1e47c6e43 Fix CMAKE_FIND_ROOT_PATH in XCompile.txt 2011-07-13 01:31:17 -07:00
Chris Robinson 2bb3d31d5c Avoid holding the global list lock while calling some capture methods 2011-07-13 01:21:30 -07:00
Chris Robinson 2a7bf1234b Use the correct send slot when mixing 2011-07-11 22:32:27 -07:00
Chris Robinson e56e462317 Fix default room air absorption 2011-07-11 22:07:37 -07:00
Chris Robinson 3245e992c3 Always pass the slot's effect to the update method 2011-07-11 01:51:24 -07:00
Chris Robinson 1299fa2801 Avoid some alListenerfv calls while under the context lock 2011-07-11 01:21:29 -07:00
Chris Robinson 52c78194b3 Update the context when an auxiliary slot is modified 2011-07-11 01:13:58 -07:00
Chris Robinson 58466a304b Use a flag to signifiy that all sources need updating 2011-07-11 01:05:42 -07:00
Chris Robinson 51c09e94b8 Avoid some alGet* duplication 2011-07-11 00:51:18 -07:00
Chris Robinson 22ccaf7f4a Trace device update size and count 2011-07-11 00:05:53 -07:00
Chris Robinson b237ebba44 Trace when the format is being requested 2011-07-10 23:57:45 -07:00
Chris Robinson 6038bbd6ec Avoid redundantly tracing HRTF status 2011-07-10 23:50:55 -07:00
Chris Robinson 84a67ef766 Trace device parameters when updating 2011-07-10 22:54:39 -07:00
Chris Robinson e03df97919 Accept "openal soft" and "openal-soft" to open a device 2011-07-10 22:36:10 -07:00
Chris Robinson 7bedd9ef7a Replace some AL_PRINT calls with proper logging 2011-07-10 22:32:44 -07:00
Chris Robinson 72cd670636 Use logging macros in the opensl backend 2011-07-10 22:31:24 -07:00
Chris Robinson 9b028e76d2 Use logging macros in the wave backend 2011-07-10 22:29:25 -07:00
Chris Robinson 479b4c9531 Use logging macros in the solaris backend 2011-07-10 22:28:34 -07:00
Chris Robinson 28f7d2151e Use logging macros in the winmm backend 2011-07-10 22:27:46 -07:00
Chris Robinson cd1560e2b0 Use logging macros in the coreaudio backend 2011-07-10 22:26:02 -07:00
Chris Robinson 87b4c93af9 Use logging macros for the portaudio backend 2011-07-10 22:22:38 -07:00
Chris Robinson 0393d4ba71 Use logging macros in the sndio backend 2011-07-10 22:20:34 -07:00
Chris Robinson 717a0592ba Use logging macros in the oss backend 2011-07-10 22:17:12 -07:00
Chris Robinson a67aff220f Use logging macros in the null backend 2011-07-10 22:15:16 -07:00
Chris Robinson e47e9c5ac3 Use logging macros in the mmdevapi backend 2011-07-10 22:13:57 -07:00
Chris Robinson 4cd9367520 Use logging macros in the dsound backend 2011-07-10 21:57:32 -07:00
Chris Robinson c4ac014a0c Use the logging macros for the alsa backend 2011-07-10 21:53:32 -07:00
Chris Robinson 21c99ca1e8 Use the logging macros in the pulseaudio backend 2011-07-10 21:51:41 -07:00
Chris Robinson 2cd3e9bcdb Use the logging macro for thunk errors 2011-07-10 21:49:01 -07:00
Chris Robinson f56fbbb777 Use logging macros for config handling 2011-07-10 21:44:42 -07:00
Chris Robinson b64156aa2b Add some macros for logging at various levels, and use them 2011-07-10 21:30:25 -07:00
Chris Robinson 5cb4129bc7 Use a per-device lock instead of a global one 2011-07-06 05:07:36 -07:00
Chris Robinson bed264ad77 Use a Lock/UnlockDevice function instead of locking a NULL context 2011-07-06 02:58:43 -07:00
Chris Robinson f8be0448b0 Don't modify the source's WetGain[HF]Auto because of a slot's state 2011-07-06 00:07:37 -07:00
Chris Robinson 75fa891e0a Store and use the send slot in the source params 2011-07-05 14:14:20 -07:00
Chris Robinson 21574f7e16 Scale the WetGain parameter directly instead of on each application 2011-07-05 14:00:11 -07:00
Chris Robinson 0903a7f1e0 Apply wet path calculations with their dry path equivalents
The auxiliary slot (and related effect) properties are retrieved at the same
time as the source properties. This will help ensure each property is accessed
from the source and slot only once.
2011-07-05 11:00:52 -07:00
Chris Robinson c8d385c998 Use a helper to tell if an effect type is reverb 2011-07-05 04:36:41 -07:00
Chris Robinson 4c35bfbc10 Read the source's RoomRolloffFactor once 2011-07-05 03:55:16 -07:00
Chris Robinson 83e322c5f3 Force front-centered for mono sounds that comes from the listener under HRTF 2011-07-05 03:36:14 -07:00
Chris Robinson 6020b964c7 Fix some formatting 2011-07-04 22:36:51 -07:00
Chris Robinson b2d5367830 Fix rolloff calculations for auxiliary sends 2011-07-04 07:32:23 -07:00
Chris Robinson d307ee8c2f Implement a lerped lookup of the HRTF coefficients/delays
Code supplied by Christopher Fitzgerald
2011-07-04 07:20:38 -07:00
Chris Robinson 202f57552a Use a proper enum for the distance model 2011-07-03 19:39:19 -07:00
Chris Robinson 5ba864b6da Calculate the wet gain decay time attenuation with linear values 2011-07-03 19:22:14 -07:00
Chris Robinson 30e97d5223 Read the RoomRolloffFactor property with the other properties 2011-07-03 03:34:40 -07:00
Chris Robinson 69be02c0fd Set the default auxiliary sends to MAX_SENDS
Since the property is ALC_MAX_AUXILIARY_SENDS, it makes more sense to default
it to the maximum and let the app lower it on request.
2011-07-03 03:18:18 -07:00
Chris Robinson eefda551de Properly handle input samples for changes with the FIR coefficients
This makes sure previously processed samples are not changed due to changes in
the FIR filter. It also applies the coefficients in ascending order with the
values, to improve caching.
2011-07-03 02:14:33 -07:00
Chris Robinson ac7ca07221 Get rid of a couple typedefs 2011-07-02 21:33:53 -07:00
Chris Robinson b043390ec2 Update a trace 2011-07-02 19:02:45 -07:00
Chris Robinson 4c94ba6d8a Properly save the original FPU state when using _controlfp 2011-07-02 02:56:06 -07:00
Chris Robinson 7b87519c15 Move functions to where they're used 2011-07-02 02:54:05 -07:00
Chris Robinson 14bfe0293a Use separate functions to find valid ALC loopback device formats 2011-07-02 02:51:33 -07:00
Chris Robinson 931d422443 Use separate ALC enums for loopback device formats 2011-07-02 02:25:34 -07:00
Chris Robinson 895150eebe Update HRTF coefficients 2011-07-01 05:08:57 -07:00
Chris Robinson a5ccae5779 Do air absorption with linear gain values
This copies the way the calculation is done with the wet path
2011-07-01 02:30:15 -07:00
Chris Robinson 1635132fd5 Use a template function to handle processing ring modulator types 2011-07-01 01:48:04 -07:00
Chris Robinson 048aee66cf Avoid accessing source properties more than once when updating parameters 2011-07-01 01:46:56 -07:00
Chris Robinson aea2868ce1 Avoid calling some al[Get]Effect functions while the context is locked 2011-06-30 20:59:35 -07:00
Chris Robinson 723755788d Rename Suspend/ProcessContext since they are locking a mutex 2011-06-30 18:10:04 -07:00
Chris Robinson 032d0836a7 Declare some variables closer to where they're used 2011-06-29 23:32:48 -07:00
Chris Robinson 5736bbc3e4 Add a source property to skip channel virtualization for multi-channel buffers
With virtualization disabled, channels are mapped directly from input to output
and any input channel that is missing an output is dropped. Any virtualization
effects (such as HRTF filters) for positional cues are also skipped.

The idea is to allow applications a way to play pre-filtered audio that channel
vitualization effects could interfere with.
2011-06-29 23:18:49 -07:00
Chris Robinson 1b773a8585 Select the correct resampler 2011-06-28 00:30:00 -07:00
Chris Robinson 3f0214ed6b Implement capture support for the CoreAudio backend 2011-06-27 23:49:17 -07:00
Chris Robinson ea83608ee4 Avoid a junk sample at the start of the ring buffer, and allocate the buffer
memory at the end of the ring buffer itself.
2011-06-27 14:30:45 -07:00
Chris Robinson 8cae1aee80 Clamp PortAudio output to stereo 2011-06-27 01:00:34 -07:00
Chris Robinson 3db061e670 Combine the two pa_load definitions into one 2011-06-26 15:40:15 -07:00
Chris Robinson efaaf9b5f9 Don't bother with a per-context lock for disconnecting devices 2011-06-26 01:26:50 -07:00
Chris Robinson c451bf370d Remove an incorrect ProcessContext call 2011-06-26 01:22:29 -07:00
Chris Robinson 212b187447 Add a function to check if a device is compatible with the Hrtf filter 2011-06-25 21:39:36 -07:00
Chris Robinson ac9a6e891f Fix the cross-compile toolchain script
CMAKE_INSTALL_PREFIX is now properly set, and the pkgconfig env vars are set so
it won't find .pc files for the default compiler.
2011-06-25 13:49:40 -07:00
Chris Robinson 1fc44d5788 Select the mixer during a source update 2011-06-25 00:13:56 -07:00
Chris Robinson 913c70557d Fix formatter for size_t 2011-06-24 15:03:29 -07:00
Chris Robinson e8b785677d Make sure to clear HAVE_SNDIO before checking backends 2011-06-23 18:39:16 -07:00
Chris Robinson 5c17eb69cb Fix comment typo 2011-06-23 17:02:57 -07:00
Chris Robinson f6d1e0f299 Avoid setting a thread priority for OpenBSD
It apparently causes a negative performance impact regardless of which values
are set.
2011-06-23 15:42:35 -07:00
Chris Robinson e42bdc2022 Fix device Update parameters for sndio 2011-06-23 15:42:16 -07:00
Chris Robinson cf44f499b8 Be silent about failing to load shared libs 2011-06-22 20:48:45 -07:00
Chris Robinson 39c984fda8 Actually load libsndio.so 2011-06-22 20:45:45 -07:00
Chris Robinson 9c23bc53b7 Update the default drivers config option comment 2011-06-22 19:31:27 -07:00
Chris Robinson 96b6cd46e6 Add a FreeBSD sndio backend 2011-06-22 19:29:13 -07:00
Chris Robinson bc6e3879d4 Properly handle the PortAudio lib in OpenBSD 2011-06-22 16:43:37 -07:00
Chris Robinson 3dcfd0dc98 Use 'Distance' for the unclamped distance value, since the clamped value is
only used for distance attenuation
2011-06-21 12:55:21 -07:00
Chris Robinson 40b8934487 Calculate wet gain when needed 2011-06-18 16:45:26 -07:00
Chris Robinson 0d39ac6528 Allow alThunkAddEntry to return a proper error 2011-06-17 23:59:25 -07:00
Chris Robinson 157e1394d6 Open the log file with config initialization 2011-06-17 18:54:05 -07:00
Chris Robinson edf78b077e Use separate lines for HRTF history and values 2011-06-17 16:22:39 -07:00
Chris Robinson 753e64c76c Do stereo duplication in two passes
First using the rear channels, then using the normal stereo channels
2011-06-17 16:20:18 -07:00
Chris Robinson 3a072a294d Avoid resetting a flag that isn't on 2011-06-17 14:59:10 -07:00
Chris Robinson 23ae3c69b5 Handle the device's disconnected status earlier when playing a source 2011-06-17 13:21:02 -07:00
Chris Robinson bef9c5e2d5 Avoid some al[Get]Effect* calls from under the context lock 2011-06-17 13:07:47 -07:00
Chris Robinson 58add24830 Call al[Get]Filterf directly from the *v equivalents
There are no filter parameters which take multiple values, and besides some
duplicated checks, they call straight through anyway.
2011-06-16 14:40:37 -07:00
Chris Robinson fdffbfd2c4 Use the predefined filter macros instead of hardcoded values 2011-06-16 10:53:01 -07:00
Chris Robinson 357f841e88 Fix namespacing of EFX filter property values 2011-06-16 10:52:17 -07:00
Chris Robinson 7d73ac4405 Avoid calling some AL functions from under the context lock
There are still some more occurances to clear out (deletion in gen error,
effects, some filters), which shall be coming up. There is a possibility for a
deadlock between the listlock and the global/context lock, if another attempt
to get the listlock is made while under the context lock.
2011-06-16 09:14:41 -07:00
Chris Robinson c429a5be25 The DirectSound DLL is always loaded by the Win32 API 2011-06-16 06:17:57 -07:00
Chris Robinson fba58fb1f0 Make sure user configuration is always initialized for Windows 2011-06-16 05:36:16 -07:00
Chris Robinson dad7f1cf0f Prefer DllMain when in Windows 2011-06-16 05:34:13 -07:00
Chris Robinson 8bee7a2b7f Don't suspend individual contexts when updating 2011-06-16 05:29:38 -07:00
Chris Robinson 632b4e6332 Avoid some channel order duplication 2011-06-16 00:00:40 -07:00
Chris Robinson f26798bd19 Remove some unnecessary casts 2011-06-15 23:52:21 -07:00
Chris Robinson 7f4dcefc80 Remove the databuffer functions and structs
It's been disabled for a while now, and not likely to be re-enabled.
2011-06-15 23:46:35 -07:00
Chris Robinson d702cb9b49 Avoid an unnecessary lock when retrieving the current context 2011-06-15 23:22:34 -07:00
Chris Robinson 2081228055 Declare some variables where they're used 2011-06-15 06:45:51 -07:00
Chris Robinson f85283c036 Don't prematurely set the HRTF flag 2011-06-15 02:00:41 -07:00
Chris Robinson 14d6ab384d Use a flag to specify when a device is running 2011-06-15 01:59:07 -07:00
Chris Robinson 62036745f3 Separate config init from library init
OSX has a bug where the file descriptor resource limit can't be increased once
certain functions (such as fopen) are used. Since lib initialization happens in
a global constructor and includes opening and parsing config files, it prevents
apps from being able to increase the rlimit once the lib is loaded, which is
usually before main().

By delaying config init, it allows apps the opportunity to change the rlimit
before calling certain OpenAL functions.
2011-06-14 09:43:33 -07:00
Chris Robinson 264274685a Throw an error when no global initialization is available 2011-06-14 08:47:53 -07:00
Chris Robinson 1babf28491 Unset the local context if it's being destroyed 2011-06-14 08:27:09 -07:00
Chris Robinson 5a15dc4ddf Avoid multiple list searches when destroying devices and contexts 2011-06-14 07:04:08 -07:00
Chris Robinson 0f782b385b Create TLS for contexts earlier 2011-06-14 06:57:51 -07:00
Chris Robinson 1534fcfafd Remove some unneeded includes 2011-06-14 05:06:39 -07:00
Chris Robinson 1c7ed6e693 Avoid using function pointers directly in the ALSA backend
Also make it use the lib wrappers
2011-06-14 05:00:59 -07:00
Chris Robinson 0c5c56cf88 Avoid using function pointers directly in the PulseAudio backend
Also, switch to the lib wrapper functions
2011-06-14 04:45:52 -07:00
Chris Robinson 53c5275aa4 Use a proper enum for the probe type 2011-06-14 04:02:58 -07:00
Chris Robinson c93d7a1721 Open the logfile in append/text mode 2011-06-14 03:04:00 -07:00
Chris Robinson 9c7dad1bdf Combine multiple probe functions into one 2011-06-14 02:28:15 -07:00
Chris Robinson bfe09b933e Trace errors in the opensl callback 2011-06-14 01:51:44 -07:00
Chris Robinson 823f5779e1 Set format_pcm according to the device format 2011-06-14 01:48:13 -07:00
Chris Robinson 96a18ba1d5 Make fewer assumptions on the OpenSL buffer count 2011-06-12 19:36:28 -07:00
Chris Robinson b8bb41107c Don't be silent when an OpenSL error occurs 2011-06-12 18:10:01 -07:00
Chris Robinson ccffb7e380 Add missing semicolon 2011-06-12 16:30:51 -07:00
Chris Robinson 39088b2035 Uninline the timeGetTime wrapper 2011-06-12 05:04:36 -07:00
Chris Robinson 8db05b8c17 Avoid defining a macro twice 2011-06-12 04:45:03 -07:00
Chris Robinson ba069ded40 Add an OpenSL backend
Currently for Android's OpenSL ES implementation
2011-06-12 04:41:42 -07:00
Chris Robinson 6b8209fdc9 Cleanup dynamic loading in the PortAudio backend 2011-06-12 04:37:32 -07:00
Chris Robinson ccf62fc525 Add wrappers to handle dyanmically loading libs 2011-06-12 04:36:24 -07:00
Chris Robinson 6bfab6427a Avoid calling some function pointers directly 2011-06-12 03:55:02 -07:00
Chris Robinson aab4f85383 Change HRTF table user input format 2011-06-03 03:15:13 -07:00
Chris Robinson 99b9208896 Improve default HRTF filter delays
They're now rounded from the source's float delays
2011-06-03 03:09:13 -07:00
Chris Robinson 7402e3743d Readd the default HRTF filter (based on the diffuse set) 2011-06-03 01:47:33 -07:00
Chris Robinson 7ddfacb58f Use a minimum phase HRTF data set
This reduces the coefficient size from 128 down to 32, with a set of delays
2011-06-03 01:06:00 -07:00
Chris Robinson c7a80418d9 Separate device-type parameter checks for alcGetIntegerv 2011-05-29 23:20:33 -07:00
Chris Robinson e98104dac8 Avoid holding locks once the context is removed from the lists 2011-05-29 23:03:59 -07:00
Chris Robinson 884f668855 Uninline a couple more functions 2011-05-29 03:28:34 -07:00
Chris Robinson 2bbe9d1b1f Uninline a couple mutex wrapper functions 2011-05-29 03:08:26 -07:00
Chris Robinson 91c3cffe41 Protect the mixer while the device's context array is being updated 2011-05-29 03:04:32 -07:00
Chris Robinson 60a55aa897 Hold the global lock while the device is being updated 2011-05-29 02:56:00 -07:00
Chris Robinson 9325a78043 Use a separate lock for device and context list accesses 2011-05-29 02:26:46 -07:00
Chris Robinson cd0e80126c Require a lock for IsDevice calls 2011-05-28 22:21:11 -07:00
Chris Robinson 1778c0d7d8 Get the device format directly from the string 2011-05-28 21:37:33 -07:00
Chris Robinson b07901f1a0 Set the subformat type when an unknown one is encountered 2011-05-28 21:21:37 -07:00
Chris Robinson a15a1abb21 Trace the format that is set when reporting HRTF disabled 2011-05-28 20:40:46 -07:00
Chris Robinson 7da2eea8f0 Add a 5.1 device format that uses side channels instead of back 2011-05-28 19:35:32 -07:00
Chris Robinson 596a1af7d9 Switch to the diffuse HRTF data set 2011-05-26 15:43:26 -07:00
Chris Robinson 6b6a7757ce Clear HRTF coefficients for the LFE channel 2011-05-25 19:04:19 -07:00
Chris Robinson 8a6266239c Rename Quad to Quadraphonic 2011-05-24 14:34:53 -07:00
Chris Robinson d83af679bd Fix handling of GUID definitions 2011-05-22 13:29:32 -07:00
Chris Robinson af005668da Avoid using an unnecessary temp variable 2011-05-21 23:14:55 -07:00
Chris Robinson d07e94056d Use libuuid when it has MMDevApi's UIDs 2011-05-21 21:57:44 -07:00
Chris Robinson 4d7aab1df4 Make some declarations static or static const 2011-05-21 21:28:27 -07:00
Chris Robinson c3cbfecbae Remove duplicate macros 2011-05-21 21:16:30 -07:00
Chris Robinson a9d9553fff Use a multi-dimensional array for the panning LUT 2011-05-21 17:45:54 -07:00
Chris Robinson 5cdf774ea7 Use the stored sample and channel count 2011-05-21 17:36:54 -07:00
Chris Robinson b75d4beb1a Dedicated dialog should go in front 2011-05-20 09:50:31 -07:00
Chris Robinson 35a0430c81 Store the separate effect parameters in a union 2011-05-20 09:36:36 -07:00
Chris Robinson 1b5caa4112 Check against the correct limits 2011-05-20 09:19:53 -07:00
Chris Robinson cbbb04eb40 Always fully normalize the position when using HRTF 2011-05-20 07:58:05 -07:00
Chris Robinson 7396aaf2c2 Fix HRTF elevation calculation 2011-05-19 17:57:14 -07:00
Chris Robinson 555ccce4ee Update the device update fields after initializing the MMDevApi device 2011-05-19 15:55:27 -07:00
Chris Robinson b838d123f8 Multiply the history offset in the filtering funcitons 2011-05-18 18:52:47 -07:00
Chris Robinson 1d3926e187 Remove some more duplication 2011-05-18 18:49:41 -07:00
Chris Robinson 194f8fc791 Uninline lpCoeffCalc 2011-05-18 18:42:25 -07:00
Chris Robinson c316766144 Use lerp() in a couple more places 2011-05-18 18:09:04 -07:00
Chris Robinson c00b2f9254 Avoid looking in the buffer queue to find the frame size 2011-05-18 17:02:18 -07:00
Chris Robinson 35c38e0b5e Update alsoftrc.sample 2011-05-18 15:13:11 -07:00
Chris Robinson 87ef4eeb36 Add cmake options to require specific backends
Note that this alone doesn't gurantee run-time availability for backends libs
that are loaded at run-time.
2011-05-18 05:34:21 -07:00
Chris Robinson 06f5f0b7c8 Make mixers channel-count agnostic
As a result, this combines/removes a bunch of mixers
2011-05-17 13:25:15 -07:00
Chris Robinson 9f91ff4734 Move the channel loop to the outside of the sample mixing loop 2011-05-17 12:49:26 -07:00
Chris Robinson f7936b4709 Avoid multiple getenv calls to the same var 2011-05-17 10:15:53 -07:00
Chris Robinson 60965e3a6c Determine the function export attribute using CMake 2011-05-17 09:13:55 -07:00
Chris Robinson b5cb2d633c Avoid including config.h in the utilities 2011-05-17 09:03:54 -07:00
Chris Robinson 810734718b Always look for libwinmm 2011-05-17 08:36:26 -07:00
Chris Robinson ca48c2e51b Modify the cross-compile toolchain file to only need a HOST option 2011-05-17 06:53:36 -07:00
Chris Robinson 86ed2ee72b Add mmdevapi to alsoftrc.sample 2011-05-17 05:04:08 -07:00
Chris Robinson 2abeed069b Improve handling of format support for MMDevApi 2011-05-17 04:54:09 -07:00
Chris Robinson 569106acd3 Fix COM initialization with MMDevApi 2011-05-16 10:31:38 -07:00
Chris Robinson 507d2f8afb Properly set the channel order for WinMM playback 2011-05-16 10:28:17 -07:00
Chris Robinson 54769a823e Implement playback using MMDevApi 2011-05-16 04:13:37 -07:00
Chris Robinson cf3287f8ec Fix closing of waveOut devices 2011-05-16 03:16:47 -07:00
Chris Robinson c5fbc9f05c Define GUIDs in ALc.c instead of the individual backend source files 2011-05-15 21:47:50 -07:00
Chris Robinson 2940b0f390 Add a skeleton backend for MMDevApi 2011-05-15 20:26:25 -07:00
Chris Robinson 442e41f825 Add an option to use Wine's Windows headers when building under MinGW 2011-05-15 20:25:09 -07:00
Chris Robinson e95b718cf4 Improve handling of device channels with DSound 2011-05-15 05:24:27 -07:00
Chris Robinson 143c08bfcb Use channel config strings in log output 2011-05-15 04:49:18 -07:00
Chris Robinson b3902c8950 Add functions to get strings from the device format 2011-05-15 04:03:15 -07:00
Chris Robinson 7210796cb8 Combine multiple loops 2011-05-15 02:12:42 -07:00
Chris Robinson 949cfe2ceb Re-enable duplication for stereo sounds 2011-05-15 00:18:28 -07:00
Chris Robinson c39f27d5f8 Set the duplicate-stereo flag properly after the device format is reset 2011-05-13 21:27:46 -07:00
Chris Robinson c9ac4f06ff Remove an unused macro and rename another 2011-05-07 04:35:49 -07:00
Chris Robinson 674b14faa3 Make a couple more pointers restricted 2011-05-07 04:32:53 -07:00
Chris Robinson a8fdcd30a5 Some minor cleanups 2011-05-07 04:31:43 -07:00
Chris Robinson db3cb23c1d Add an option to reverse Z panning of mono sources
Applications that are not built around OpenAL's orientation system need to
convert their given vectors to it. Depending on how this is done, it can lead
to proper stereo (left-right) panning with improper surround sound (front-back)
panning, which thusly sounds correct with stereo output and incorrect with 4+
channel output.

This option is intended to help fix playback of such applications on surround
sound systems, without having to resort to forcing stereo output.
2011-05-07 03:54:46 -07:00
Chris Robinson 7972f7a2da Combine duplicate arrays 2011-05-06 23:25:15 -07:00
Chris Robinson 8c3d50be28 Skip LFE with HRTF output 2011-05-06 04:46:21 -07:00
Chris Robinson 4198f3062c Remove the device channel matrix 2011-05-06 04:42:54 -07:00
Chris Robinson c2a6efc77a Use the panning algorithms to build the multi-channel source matrix
This prevents the stereo-duplication option from working. It should hopefully
be restored soon.
2011-05-06 04:37:10 -07:00
Chris Robinson 6928487aa1 Remove some redundancy 2011-05-06 02:53:22 -07:00
Chris Robinson e608317e70 Fix device flag value 2011-05-06 01:59:32 -07:00
Chris Robinson 64b5d7a519 Check the correct flag in the dsound backend 2011-05-06 01:53:13 -07:00
Chris Robinson eef41e76d7 Don't clear the request flag if a new context doesn't specify a frequency 2011-05-06 01:50:19 -07:00
Chris Robinson d583f7f307 Update sources automatically when device parameters change 2011-05-06 01:06:01 -07:00
Chris Robinson f97eca34df Minor source update fixes 2011-05-06 00:51:12 -07:00
Chris Robinson 1363e044ac Remove the head_dampen option
Now that HRTF is properly supported, it's not needed
2011-05-06 00:38:10 -07:00
Chris Robinson 9967d4eadd Mark a couple more pointers as restricted 2011-05-06 00:33:58 -07:00
Chris Robinson 2972de1047 Use function pointers to call the source mixer 2011-05-06 00:20:40 -07:00
Chris Robinson 38db4115fc Fix a comment 2011-05-05 22:36:26 -07:00
Chris Robinson 0e824914b2 Use signed samples for 8-bit storage 2011-05-05 18:54:10 -07:00
Chris Robinson 7832a6d1d8 Add a warning when HRTF is disabled 2011-05-03 20:23:35 -07:00
Chris Robinson 0cce93a1f8 Fix WinMM backend compilation 2011-05-03 17:12:05 -07:00
Chris Robinson 679f2480c9 Add a device flag for channel config requests 2011-05-03 17:11:07 -07:00
Chris Robinson db7ca2c7ca Combine the mono and multi-channel mixer templates 2011-05-03 16:34:28 -07:00
Chris Robinson e2850df07e Use the same filters for mono sources and multi-channel 2011-05-03 16:18:46 -07:00
Chris Robinson 48def2fecb Properly handle elevations that don't have angles up to 180 2011-05-03 03:15:28 -07:00
Chris Robinson ba236ad108 Add a frequency-request device flag 2011-05-03 02:29:26 -07:00
Chris Robinson 9a28402b90 Fix non-spatialized mono sources 2011-05-02 18:33:32 -07:00
Chris Robinson 8c705cc4ea Mark some mixer pointers as restricted 2011-05-02 17:14:57 -07:00
Chris Robinson 24aaa24b5c Check for the C99 restrict keyword
GCC does not default to C99 mode yet, so does not know restrict, however it
still allows using __restrict in its place
2011-05-02 17:06:32 -07:00
Chris Robinson fd795a7eeb Add casts to silence some warnings 2011-05-02 15:04:35 -07:00
Chris Robinson abdc254e55 Use a pointer-to-arrays for the local HRTF coefficients 2011-05-02 13:35:13 -07:00
Chris Robinson f843b7e2e3 Implement HRTF mixers for multi-channel sources 2011-05-02 02:22:30 -07:00
Chris Robinson eea86ab891 Use flags instead of separate bools 2011-05-01 18:18:37 -07:00
Chris Robinson d7a452588c Document the hrtf config option 2011-05-01 14:16:05 -07:00
Chris Robinson 56d9418119 Add an HRTF filter for mono sources
The data is based on the KEMAR HRTF data provided by MIT, which can be found at
<http://sound.media.mit.edu/resources/KEMAR.html>. The compact measurements
were used. See hrtf_tables.inc for more information.

The filter is only available for stereo output, using a 44100hz playback rate.
Note also that it currently only applies to mono sounds, and the cf_level and
head_dampen config options are ignored while it is active.
2011-05-01 13:59:44 -07:00
Chris Robinson 0109104027 Add some source fields for HRTF data 2011-05-01 13:21:56 -07:00
Chris Robinson c9c965ef85 Add a device flag for enabling HRTF 2011-05-01 13:19:23 -07:00
Chris Robinson af3adf5244 Be silent if the PulseAudio context fails to connect while probing 2011-04-30 17:58:37 -07:00
Chris Robinson 43f97b4da1 Use the new format names when possible 2011-04-29 11:55:59 -07:00
Chris Robinson 63f07db303 Add the SOFT moniker for the loopback extension functions and enums 2011-04-28 11:37:23 -07:00
Chris Robinson 2e9ca89dd3 Reorder some cases 2011-04-27 21:05:46 -07:00
Chris Robinson 8586aeabe8 Minor fix for 24-bit conversions to float and double 2011-04-27 19:26:27 -07:00
Chris Robinson c031b4d5d0 Allow MSVC to build a static lib
Based on a patch by Michał Cichoń <michcic@gmail.com>
2011-04-27 16:33:44 -07:00
Chris Robinson d91b3464a4 Fix 24-bit sample loading for big-endian 2011-04-27 15:46:30 -07:00
Chris Robinson 650d94e1a6 Change some sample type values 2011-04-27 12:20:50 -07:00
Chris Robinson 543134a6d2 Add loopback device properties to the ALCenum list 2011-04-27 12:18:24 -07:00
Chris Robinson ab2626f9df Add a couple in-progress extensions to the extension lists 2011-04-27 12:15:16 -07:00
Chris Robinson d50f9249d6 Add support for packed 24-bit samples 2011-04-26 18:17:50 -07:00
Chris Robinson 10a9753183 Add a compatibility option to treat cone angles as half angles
All previous versions of the library treated the source cone angles as half
angles, which is contrary to the spec. Setting the __ALSOFT_HALF_ANGLE_CONES
environment variable to "true" or "1" restores the buggy behavior for
compatibility with applications that expect it.

This is not a config file option because new apps should not be made to depend
on the old behavior.
2011-04-22 23:17:27 -07:00
Chris Robinson 1e8718fe87 Convert full-width cone angle source properties to half-width
The spec intends the property values to be the full angle encompassed by the
cones, but the calculation interprets them as the angle from the center
point.
2011-04-22 22:59:58 -07:00
Chris Robinson 678d0b87d0 Apply the device matrix to the multi-channel source matrix
Mono sources and effects already output according to the available output
device channels. Multiplying the device matrix with the source matrix results
in a matrix that has the same effect as applying the source matrix followed by
the device matrix, so all the channel remixing can be done in one place.
2011-04-14 21:03:37 -07:00
Chris Robinson 319dfce300 Fix LFE channel output 2011-04-14 15:20:25 -07:00
Chris Robinson c3e86933d0 Only set relevant device matrix entries 2011-04-12 17:08:02 -07:00
Chris Robinson 936acf8c10 Allow the buffer_samples API to be retrieved 2011-04-10 09:48:37 -07:00
Chris Robinson 358d402cd0 Move Convert_IMA4 into the template declarations 2011-03-21 09:42:03 -07:00
Chris Robinson d11593eb5c Use nested loops when converting data 2011-03-19 12:40:14 -07:00
Chris Robinson 4bdedc7fde Use more appropriate enum values 2011-03-18 07:28:06 -07:00
Chris Robinson 9be625d6e8 Add some new format names 2011-03-17 08:42:25 -07:00
Chris Robinson 898b245bb2 Make LoadData return an error if the dst format is not valid or compatible 2011-03-16 19:39:31 -07:00
Chris Robinson be37f10fed Add alIsBufferFormatSupportedSOFT 2011-03-16 13:57:00 -07:00
Chris Robinson a87f651b4d Combine ConvertInput* and ConvertOutput* helpers
The UserFmt* enum types are a complete set of all recognized channel configs and
sample types, so casting Fmt* enum types to them is valid.
2011-03-16 13:32:57 -07:00
Chris Robinson 575195a283 Add alGetBufferSamplesSOFT 2011-03-16 12:56:39 -07:00
Chris Robinson 5621ee4239 Buffers made with alBufferSamplesSOFT always reference the internal format
This is not necessarily the same as the format of the originating data
2011-03-16 12:24:53 -07:00
Chris Robinson f5c4e67aef Add alBufferSubSamplesSOFT 2011-03-16 12:13:17 -07:00
Chris Robinson 2f7de9d696 Add alBufferSamplesSOFT, as an initial start to AL_SOFT_buffer_samples 2011-03-16 11:29:22 -07:00
Chris Robinson 9cee723e92 Pass the number of compressed frames to LoadData 2011-03-16 10:00:24 -07:00
Chris Robinson 4a1c0fedca Use the defines for the UserFmt* and Fmt* types 2011-03-16 05:47:07 -07:00
Chris Robinson b510958f9e NoErr should be noErr 2011-03-15 11:52:18 -07:00
Chris Robinson d6c8bb35b4 Add a CoreAudio backend
Code courtesy of Garin Hiebert <garinh@cheesetoast.net>
2011-03-15 04:58:56 -07:00
Chris Robinson 4899674b6c Move ALC_ENUMERATE_ALL_EXT enums to alc.h, to match other systems' headers 2011-03-14 09:04:41 -07:00
Chris Robinson 4eeed5376f Invert the device matrix row/column
It is accessed now as mat[target][source]
2011-03-13 08:46:54 -07:00
Chris Robinson 77ad538e68 Remove more unneeded casts 2011-03-13 07:11:09 -07:00
Chris Robinson b699aca65b Remove unnecessary casts 2011-03-13 06:34:41 -07:00
Chris Robinson ffd19fe961 Rename ConvertData* to ConvertInput* 2011-03-13 00:05:19 -08:00
Chris Robinson 208912ceac Fix DSound floating-point sample assumptions 2011-03-12 22:52:33 -08:00
Chris Robinson 89c253ce41 Recognize ALC_EXT_DEDICATED with openal-info 2011-03-12 21:28:40 -08:00
Chris Robinson b1d4508d90 Use a loop to make sure the requested effect is not disabled 2011-03-12 20:54:49 -08:00
Chris Robinson 8249dc8559 Combine multiple lists into one 2011-03-12 20:37:22 -08:00
Chris Robinson 59fe4f8a36 Don't return enums for disabled effects 2011-03-12 20:27:46 -08:00
Chris Robinson acb493665d Add dedicated to the list of excludefx 2011-03-12 20:15:21 -08:00
Chris Robinson 73123bc313 Add support for ALC_EXT_DEDICATED 2011-03-12 20:11:25 -08:00
Chris Robinson 947aa1342c Add ALC_EXT_DEDICATED to alext.h 2011-03-12 07:19:42 -08:00
Chris Robinson 20c8f9fe31 Add AL_EXT_FOLDBACK to alext.h 2011-03-12 06:35:45 -08:00
Chris Robinson 66ee3bc217 Better protect the device for multi-threading access 2011-03-12 06:23:17 -08:00
Chris Robinson 031a2a1b1e Implement a basic non-real-time loopback device
Currently it behaves just like a normal device except contexts are only
processed during calls to alcRenderSamples. Additionally, the ALC_SYNC and
ALC_REFRESH context attributes are not valid for these devices, and there are
two new context attributes to specify the rendering format: ALC_FORMAT_TYPE
and ALC_FORMAT_CHANNELS. These each take one of the type and channel enums
added.
This stuff is subject to change.
2011-03-11 00:13:44 -08:00
Chris Robinson 9c87b73ad5 Fix a compiler warning about checking differing enum types
One of the enum types is actually a subset of the other, and share the same
integer values for compatible enumation values
2011-03-10 01:27:42 -08:00
Chris Robinson 97139a530f Filter NaN sample values when converting floats and doubles 2011-03-04 16:18:03 -08:00
Chris Robinson b483b1e380 Release 1.13 2011-02-15 12:10:44 -08:00
Chris Robinson d9126eeb2c Finalize the loop points extension as AL_SOFT_loop_points 2011-02-09 22:54:12 -08:00
Chris Robinson d8048e136f BufferFreq doesn't need to be a float 2011-02-09 11:19:33 -08:00
Chris Robinson fd64fea279 Use the correct winmm device ID type 2011-02-08 10:33:18 -08:00
Chris Robinson 667fd24c84 Add an option to force usage of static MSVC runtimes 2011-02-08 00:34:25 -08:00
Chris Robinson 9ffb38d6be Prevent unused variable warnings when monotonic clock isn't available 2011-02-08 00:22:22 -08:00
Chris Robinson 13397feccf Don't inline CalcLimitedHfRatio 2011-02-07 23:58:40 -08:00
Chris Robinson 57b05fb6d4 Add a missing enum value for PA_API_VERSION 11 2011-02-07 23:53:07 -08:00
Chris Robinson 41a2c25b5a Include cguid.h for GUID_NULL 2011-02-07 22:46:54 -08:00
Chris Robinson b214249290 Check for internal visibility, since that's what is used 2011-02-07 13:55:48 -08:00
Chris Robinson 8dad1c382b Add the standard "revision" number to the library name 2011-02-06 02:39:22 -08:00
Chris Robinson 6713a47d38 Fix looping on loop points 2011-02-06 01:07:37 -08:00
Chris Robinson 02e0223fe8 Re-enable channel-config-specific layout options 2011-01-23 23:57:39 -08:00
Chris Robinson 6c4add67e1 Don't duplicate the device list append function 2011-01-23 23:16:55 -08:00
Chris Robinson 2b7324c06e Avoid using a variable for array sizes and const variable initialization 2011-01-23 22:30:18 -08:00
Chris Robinson c95bd6cadf Fix loop handling when traversing the buffer queue backwards 2011-01-23 15:04:42 -08:00
Chris Robinson d73694d3e6 More improvements to the AudioIO backend 2010-12-09 23:53:32 -08:00
Chris Robinson 5d607f9484 Better check format type returned by AUDIO_SETINFO 2010-12-09 23:47:11 -08:00
Chris Robinson 816c38e9e4 Add signed 8-bit output to the AudioIO backend 2010-12-09 23:43:29 -08:00
Chris Robinson 09130e5dcd Fix some AudioIO backend warnings 2010-12-09 23:41:54 -08:00
Chris Robinson b87447787a Cleanup some ok() macros 2010-12-09 23:32:47 -08:00
Chris Robinson 5a548f8366 Another fix for float-to-int conversions 2010-12-09 22:47:06 -08:00
Chris Robinson 023658605e Constify some variables 2010-12-09 19:47:08 -08:00
Chris Robinson 031111014b Be consistent with variable name use 2010-12-09 18:09:47 -08:00
Chris Robinson b5b9303e66 Fix up some conversions 2010-12-09 17:37:57 -08:00
Chris Robinson 3deaa2fad3 Combine stereo and multi-channel templates 2010-12-09 17:14:12 -08:00
Chris Robinson c3cd4c80b2 Use a matrix when rendering to the dry buffer 2010-12-09 16:37:23 -08:00
Chris Robinson be3f3c4bf0 Make better use of the type range when converting from float 2010-12-09 05:06:29 -08:00
Chris Robinson d1ca8b4470 Use simple loops instead of several explicit lines for the mono mixer 2010-12-09 05:05:42 -08:00
Chris Robinson 72e3386c6a Be a bit more generic with the multi-channel mixer loops 2010-12-09 04:41:52 -08:00
Chris Robinson 4010a6b745 Remove some now-unused functions 2010-12-08 02:56:24 -08:00
Chris Robinson 0218957eb5 Use "internal" visibility by default
All functions callable by apps, directly or indirectly (eg, through function
pointers), are marked as "protected", and internal visibility can provide
benefits over hidden when we know functions won't be called by other modules.
2010-12-08 02:32:09 -08:00
Chris Robinson cd65ab7dfa Add an IMA4 encoder 2010-12-05 03:34:19 -08:00
Chris Robinson c4adf541c1 Set the playing flag to TRUE when the dsound buffer starts playing 2010-12-04 20:22:12 -08:00
Chris Robinson 88e3a22772 Separate device format into 'channel config' and 'sample type' components 2010-12-04 19:50:00 -08:00
Chris Robinson 191803ad53 Fix wave and null device calculations when the timer wraps 2010-12-03 23:48:59 -08:00
Chris Robinson 1b7be672fa No need to clamp the app-specified pitch 2010-12-03 23:08:26 -08:00
Chris Robinson 9581325b9c Rename SrcFmt -> UserFmt 2010-12-03 22:33:41 -08:00
Chris Robinson 59990acc2a Add functions to write (signed) byte and ushort samples to devices 2010-12-02 16:36:37 -08:00
Chris Robinson 0bf7716332 Remove incomplete EXT_buffer_sub_data, as it's (temporarily) superceded by SOFT_buffer_sub_data
The EXT version can come back when other implementations take an interest
2010-12-02 01:46:43 -08:00
Chris Robinson 3339e5dc6a Use memcpy to copy a string instead of sprintf 2010-12-02 01:12:49 -08:00
Chris Robinson 30b3f68ddf Make a function static 2010-12-02 01:05:29 -08:00
Chris Robinson 75552f2765 Remove unused macro 2010-12-01 23:21:18 -08:00
Chris Robinson 4a0bf9fd44 Template-ize the device buffer writing functions 2010-12-01 21:50:49 -08:00
Chris Robinson fa28098a83 Some general cleanups 2010-12-01 20:34:29 -08:00
Chris Robinson 646409cbdf Don't limit IMA4 decoding to 2 channels 2010-12-01 18:36:07 -08:00
Chris Robinson 9e8fb89a52 Rename OUTPUTCHANNELS to something more descriptive 2010-12-01 18:33:17 -08:00
Chris Robinson 14441e0e53 Rename some variables 2010-12-01 14:19:57 -08:00
Chris Robinson 7f5814c4d8 Use min/max to clamp the IMA4 sample and index values 2010-12-01 14:07:21 -08:00
Chris Robinson e3afc3587e Fix handling of -32768 in the muLaw encoder 2010-12-01 02:24:39 -08:00
Chris Robinson 8a52c44d15 Don'f fail if realloc returns NULL for 0 sizes 2010-12-01 02:00:41 -08:00
Chris Robinson e6e18732b5 Make sure the muLaw sample encoder is inlined 2010-11-30 18:06:54 -08:00
Chris Robinson 3f8ae8f1ce Fix IMA4 decoding 2010-11-30 16:54:30 -08:00
Chris Robinson a79129835c Fix some comments and line spacing 2010-11-30 13:04:48 -08:00
Chris Robinson 6fcaecdc9f Fix int-to-double conversion 2010-11-30 12:26:37 -08:00
Chris Robinson 8dc8090888 Make muLaw decoding work similar to encoding 2010-11-30 01:26:43 -08:00
Chris Robinson 4db2f1ec7d Implement a muLaw encoder 2010-11-30 01:20:40 -08:00
Chris Robinson 06fa030260 Rename the IMA4 block decoder function 2010-11-30 01:00:37 -08:00
Chris Robinson 018f9fc278 Handle all channel configurations for double 2010-11-30 00:32:13 -08:00
Chris Robinson c0f6d46cd8 Allow LoadData to handle IMA4 data 2010-11-30 00:28:02 -08:00
Chris Robinson 8e43b1fe04 Let the storable format enum values share the input format enum values where they overlap 2010-11-29 23:57:07 -08:00
Chris Robinson 04df919a1c Clean up muLaw format loading a bit 2010-11-29 23:46:49 -08:00
Chris Robinson f82e628f08 Allow converting IMA4 to types other than short 2010-11-29 22:55:46 -08:00
Chris Robinson 07a7c03adb Check for a couple missing input formats and remove an unneeded default case 2010-11-29 22:07:07 -08:00
Chris Robinson e800f8c0a7 Let DecomposeInputType verify the format passed to alBufferSubDataSOFT 2010-11-29 22:01:54 -08:00
Chris Robinson d02dbef9d2 Pass the input channel config and sample type to LoadData 2010-11-29 21:55:47 -08:00
Chris Robinson 55262bc6e9 Make the decompose functions return false when given an unhandled format 2010-11-29 20:34:33 -08:00
Chris Robinson 51ed22afb9 Combine two switch blocks 2010-11-29 20:17:15 -08:00
Chris Robinson 0cf32e2946 Reorder some parameters for consistency 2010-11-29 19:48:18 -08:00
Chris Robinson 693ab4d8bc Fix source offset calculations 2010-11-29 19:36:33 -08:00
Chris Robinson e4869b447c Store the decomposed source format in the buffer 2010-11-29 19:27:33 -08:00
Chris Robinson 547f356131 Keep a handle on the buffer with the format when queueing new buffers 2010-11-29 17:35:22 -08:00
Chris Robinson 017ab1b3fd Add IMA4 as a source data type 2010-11-29 17:14:22 -08:00
Chris Robinson 07ef566de1 Add support for (unsigned) int input formats 2010-11-29 17:00:22 -08:00
Chris Robinson 3c8a29a179 Reorder and rename parameters for the format decomposition functions 2010-11-29 15:19:39 -08:00
Chris Robinson f74032d244 Uninline a couple more functions 2010-11-29 15:01:27 -08:00
Chris Robinson eb489ac886 Store double formats as float
The mixer will cut it down to float anyway, so it's unnecessary and needlessly
creates more mixer functions
2010-11-28 23:07:57 -08:00
Chris Robinson 986628b8b2 Avoid temp storage 2010-11-28 22:50:27 -08:00
Chris Robinson da4f5c6f96 Uninline some functions
Also add -Winline to the compiler command line to watch for future inline problems
2010-11-28 19:52:17 -08:00
Chris Robinson f09ae196a8 Avoid making Resampler[Pre]Padding static 2010-11-28 17:37:14 -08:00
Chris Robinson 3e702ae2bc Improve static builds with MinGW 2010-11-28 16:19:11 -08:00
Chris Robinson 490d1c6a8e Prefer GCC's constructor, even in Windows 2010-11-28 14:27:07 -08:00
Chris Robinson 0e748d98d5 Rename some stuff for consistency 2010-11-28 13:08:51 -08:00
Chris Robinson c41e893361 Don't store the internal format in the buffer
The type and channel config are good enough
2010-11-28 12:53:35 -08:00
Chris Robinson 2fd8d6916b Support rear formats directly in the mixer 2010-11-27 22:55:20 -08:00
Chris Robinson b4eaf1ab5a Avoid a potential overflow when setting the source offset 2010-11-27 22:22:14 -08:00
Chris Robinson 3c8e9a7f24 Clean up the source byte offset code 2010-11-27 22:13:29 -08:00
Chris Robinson 67ec72284d Remove some unneeded muLaw special cases 2010-11-27 22:06:51 -08:00
Chris Robinson 754a122aa0 Support muLaw with LoadData 2010-11-27 22:00:24 -08:00
Chris Robinson 0486d633c6 Remove an unnecessary assert 2010-11-27 20:41:09 -08:00
Chris Robinson 09b063002b Be much more flexible with loading and converting buffer data 2010-11-27 20:33:37 -08:00
Chris Robinson 8828f4d0e5 Don't inline the decompose function 2010-11-27 15:37:51 -08:00
Chris Robinson 9495cf71fd Decompose input buffer formats and use it when converting uncompressed data 2010-11-27 15:33:33 -08:00
Chris Robinson 4d0f70439c Rename a function 2010-11-27 14:30:13 -08:00
Chris Robinson d3bb5d4fcb Add a basic WaveOut device
It can still use more work, but it seems to work
2010-11-27 14:13:40 -08:00
Chris Robinson 61315d4dfb Shorten a couple really long lines 2010-11-27 00:51:21 -08:00
Chris Robinson 2a123a4760 Support 64-bit double buffers directly 2010-11-27 00:32:53 -08:00
Chris Robinson 4689ffb72d muLaw converts to 16-bit, not float 2010-11-27 00:17:19 -08:00
Chris Robinson d2895c7985 Use the enums in a couple more places 2010-11-27 00:15:07 -08:00
Chris Robinson 6abb9d151e Use enums when selecting the mixer 2010-11-26 23:42:30 -08:00
Chris Robinson 08827efae3 Remove an unnecessary 'struct' keyword 2010-11-26 20:35:47 -08:00
Chris Robinson 16c05d2c05 Avoid some more code duplication 2010-11-26 20:22:14 -08:00
Chris Robinson 4d4d699780 Fix step calculations 2010-11-26 20:17:05 -08:00
Chris Robinson ca1ec4b221 Fix some comments 2010-11-26 18:01:29 -08:00
Chris Robinson facb922f3e Properly clamp high pitch values 2010-11-26 17:47:43 -08:00
Chris Robinson d7987677ac Subtract the fractional position separately 2010-11-26 15:51:13 -08:00
Chris Robinson 5fbbfa482f Rename some variables to be clearer 2010-11-26 12:59:45 -08:00
Chris Robinson 7287b5cd3c Make the buffer padding size dependent on the resampler 2010-11-26 02:53:15 -08:00
Chris Robinson 0692cf304d Don't allocate padding with the buffer itself 2010-11-26 02:31:17 -08:00
Chris Robinson 88ebe7e4c1 Decrease the buffer size more 2010-11-26 02:14:11 -08:00
Chris Robinson d647ed60e8 Convert a few divisions to multiplications 2010-11-26 01:35:40 -08:00
Chris Robinson 1cb29ece08 Make a FRACTIONONE macro, and use it 2010-11-26 01:21:46 -08:00
Chris Robinson de8b7fbc5f Share the interpolation functions and use them in the reverb effect 2010-11-26 01:07:54 -08:00
Chris Robinson 98d78f7ff9 Slightly improve channel gain calculations 2010-11-26 00:24:54 -08:00
Chris Robinson 3d1eb7dcad Add a missing break statement 2010-11-25 23:13:50 -08:00
Chris Robinson e1c0b5ea24 Call MixSource directly, instead of through a function pointer 2010-11-25 23:09:18 -08:00
Chris Robinson 30820c1bde Add a cubic resampler 2010-11-25 23:05:44 -08:00
Chris Robinson 46b9802902 Force the point resampler when the source plays at normal speed 2010-11-25 22:15:38 -08:00
Chris Robinson b6fb28de1a Pass a pointer to the samplers so they can read just the samples they need 2010-11-25 22:04:17 -08:00
Chris Robinson 17fc4ba5e3 Add "pre-padding" to the temp buffer
This will help to implement resamplers that want previous samples
2010-11-25 21:42:15 -08:00
Chris Robinson 596514f7aa Make the buffer queue a double-linked list 2010-11-25 18:30:57 -08:00
Chris Robinson 6e3a6f90e9 Be less wave-in-specific in the winmm backend 2010-11-25 17:37:34 -08:00
Chris Robinson fbd6439bd2 Make the first enumerated DirectSound device use the default, and fix the name 2010-11-25 14:35:32 -08:00
Chris Robinson f72c735f9e Make sure the target buffer size is frame aligned 2010-11-25 14:23:03 -08:00
Chris Robinson 9b23fc5db8 Minor rework of the template macros 2010-11-25 14:07:52 -08:00
Chris Robinson aaff188b3b Remove the cosine resampler
It's too slow when auxiliary sends are being used.
2010-11-25 13:49:51 -08:00
Chris Robinson 2595d85cca Double-up the interpolation functions (and make them use doubles) 2010-11-25 13:45:35 -08:00
Chris Robinson 2c96c29938 Rename some "template" functions 2010-11-25 13:22:54 -08:00
Chris Robinson 65919b7037 Always make sure there's a sample available to mix, for pending click removal 2010-11-25 13:17:51 -08:00
Chris Robinson f1bb7c33f1 Fix a bunch of comments 2010-11-25 12:07:12 -08:00
Chris Robinson 8e7a4046d0 Don't "convert" LOKI quad to MCFORMATS quad 2010-11-25 11:49:35 -08:00
Chris Robinson f4e0c6a348 Use a temporary stack buffer to store the source buffer data to be mixed
This makes it much easier to pack the playable buffer data so that it's a
continuous data stream. This could normally be broken up otherwise, such as
when using loop points or small queued buffers.
Currently it defaults to a 16KB stack buffer, but this can be overridden at
compile time.
2010-11-25 11:16:19 -08:00
Chris Robinson 885a5e9f90 Use 0-based access for the buffer data 2010-11-24 20:44:21 -08:00
Chris Robinson 22e3e9a18b Support 8-bit sample data in the mixer 2010-11-24 19:41:25 -08:00
Chris Robinson 445cde2dd4 Check the sample byte size after the resampler 2010-11-24 17:23:09 -08:00
Chris Robinson 3dd58cf765 Get rid of a useless union 2010-11-24 17:10:41 -08:00
Chris Robinson cd87061749 Change the mixer macros to declare functions 2010-11-24 17:07:50 -08:00
Chris Robinson 1311e051d9 Set mixer sample pos and frac when needed 2010-11-24 14:54:46 -08:00
Chris Robinson c9363832fe Fully progress the source to a stopped state when no buffers are playable 2010-11-24 10:16:53 -08:00
Chris Robinson 6ab22e7cbb Reorder some stored source params 2010-11-24 10:01:50 -08:00
Chris Robinson bb13f7f234 Make sure the buffer padding is always cleared before looking for the next buffer 2010-11-24 07:20:41 -08:00
Chris Robinson 13ba253aec Calculate ALSA buffer length with the other values 2010-11-23 07:24:27 -08:00
Chris Robinson c58469e234 Handle the timer wrapping in the wave and null output devices 2010-11-21 11:34:16 -08:00
Chris Robinson 57a03bc2b5 Use helpers to write 16- and 32-bit values 2010-11-21 06:54:41 -08:00
Chris Robinson a7b4e0b5c5 Get rid of the device timer stuff 2010-11-21 06:32:59 -08:00
Chris Robinson bbb45e326c Avoid temporary float variables 2010-11-21 03:58:37 -08:00
Chris Robinson dfe3e402a8 Pass the data pointer and proper sampler to the mixer macro 2010-11-21 03:33:33 -08:00
Chris Robinson e884da50ea Move aluMixData back to ALu.c 2010-11-21 02:51:18 -08:00
Chris Robinson 4c576631ad Hide AL_EXTX_sample_buffer_object
It hasn't been going anywhere, and it's current use is dubious. Leave the code
in for now, but it may be removed later.
2010-11-21 02:36:34 -08:00
Chris Robinson 5f6f046d25 Use a double when converting millisecond offsets to byte offsets 2010-11-07 21:41:04 -08:00
Chris Robinson 65e033c64b Increase the max number of source sends to 4, and decrease the default to 1 2010-11-06 15:08:54 -07:00
Chris Robinson 07346da740 Use the number of sends the device is configured for in more places 2010-11-06 14:27:33 -07:00
Chris Robinson c95f182a90 Remove unnecessary comments and reorder some lines 2010-11-06 14:07:30 -07:00
Chris Robinson 289c74788e Use a switch to select the appropriate channel-count mixer 2010-11-06 12:21:46 -07:00
Chris Robinson 65a8d9278d Declare a variable where it's used 2010-11-01 12:44:56 -07:00
Chris Robinson 887c830d56 Remove explicit use of -funroll-loops
I'm not sure this really helps much.
2010-10-15 15:21:23 -07:00
Chris Robinson 6db4e08efb Fix fpu state handling with _controlfp 2010-10-15 10:35:47 -07:00
Chris Robinson 4f81bce800 Improve a macro to reference its parameter 2010-10-11 12:24:28 -07:00
Chris Robinson 932fcfa8bf Fix 64-bit compilation 2010-10-11 12:24:07 -07:00
Chris Robinson 314832e35a Rename a couple more variables 2010-10-10 04:00:50 -07:00
Chris Robinson 6a9dd185e6 Improve naming and placement of a couple variables 2010-10-10 03:47:57 -07:00
Chris Robinson 2172827f43 Simplify a calculation 2010-10-09 15:24:24 -07:00
Chris Robinson 90e9cd970a Check resampler mode first when selecting the mixer 2010-09-26 13:53:02 -07:00
Chris Robinson 845ca95888 Get a variable where it's needed 2010-09-26 13:30:46 -07:00
Chris Robinson 92d8d2f8a4 Pass the device to the mix function and localize the dry mix and click buffers 2010-09-26 12:23:22 -07:00
Chris Robinson 7cd52b2857 Clarify a check 2010-09-26 01:29:28 -07:00
Chris Robinson 9fbd6c6c3f Make the SourceMix function a method of the ALsource struct 2010-09-26 01:15:27 -07:00
Chris Robinson 08fdc5fa98 Avoid updating the source in MixSource 2010-09-25 23:49:57 -07:00
Chris Robinson 7e362249ff Convert shorts so that 0 remains 0 2010-09-25 21:13:40 -07:00
Chris Robinson 9fa2db0227 Don't try to open the ALSA device a second time if the first fails 2010-09-25 20:20:10 -07:00
Chris Robinson 4c75a72a90 Avoid some code duplication 2010-09-25 20:18:07 -07:00
Chris Robinson a0b2778161 Localize more variables and remove unnecessary ones 2010-09-25 14:19:36 -07:00
Chris Robinson bc6d94cd25 Avoid sampling beyond the end of the loop 2010-09-24 23:32:50 -07:00
Chris Robinson e70141e57b Localize some variables to the mixer macros
The hope here is to help break the mixer loops into separate functions, so the
macro expansion doesn't cause the MixSource function to explode in size and
kill compilation speed.
2010-09-24 23:24:01 -07:00
Chris Robinson 2e70b82eaa Use separate mixing loops for the dry path and wet paths
This will increase CPU use a bit for sources that use auxiliary sends
(particularly with the cosine resampler), but it makes it more scalable when
adding more sends per source. Additionally, it will improve performance of
sources that don't use auxiliary sends.

The cosine resampler can probably be improved by using a lookup table instead
of calling cos() as there are 14 bits of fractional precision, so a 16384-
element float array would cover it all. This can also be reduced to a half or
quarter size if the precision loss is found to be acceptable.
2010-09-24 18:29:49 -07:00
Chris Robinson 117ed52091 Better protect against negative attenuation 2010-09-24 13:16:09 -07:00
Chris Robinson e9d5e53dbe Use ALfloat for the dry mix buffer 2010-09-23 17:44:02 -07:00
Chris Robinson e308694521 Remove an unnecessary variable 2010-09-23 16:51:30 -07:00
Chris Robinson 7e498012ff Rename some variables to be less confusing 2010-09-23 16:49:33 -07:00
Chris Robinson f98bf32be8 Fix line indentation 2010-09-23 15:08:30 -07:00
Chris Robinson 0f2ed8c84d Be consistent with a variable's use 2010-09-23 13:49:21 -07:00
Chris Robinson 8d67cdfaab Convert float to short similar to short to float 2010-09-23 08:43:39 -07:00
Chris Robinson 67ba924ed3 Store and mix signed 16-bit samples 2010-09-22 23:38:06 -07:00
Chris Robinson 764cfb6a0d Allow the mixing loop to access the different sample formats 2010-09-22 20:53:04 -07:00
Chris Robinson 4e3c945093 Compact some reused code 2010-09-22 18:49:46 -07:00
Chris Robinson 8558bc1d41 Allow accessing the buffer data as multiple data types in the mixer
Also reduces a bunch of indentation
2010-09-22 17:19:35 -07:00
Chris Robinson 03f56419d0 Duplicate stereo sources by default
The mixer is smart enough now to handle proper volume adjustments depending on
the number of output speakers
2010-09-22 11:09:34 -07:00
Chris Robinson 3ee9d24c3f Use separate macros for the different mixing loops 2010-09-22 09:38:24 -07:00
Chris Robinson 9b1ad6c873 Consolidate some code 2010-09-21 19:49:47 -07:00
Chris Robinson 97daaade8a Reduce some more indentation 2010-09-21 16:54:33 -07:00
Chris Robinson 8a1d5a21c3 Reduce some indentation 2010-09-21 15:12:08 -07:00
Chris Robinson 301a8ba3a4 Improve some range checks 2010-09-21 14:41:43 -07:00
Chris Robinson 2d4453968d Implement AL_SOFT_buffer_sub_data using the current AL_EXT_buffer_sub_data spec
This extension can be useful for some applications, but a full EXT version may
not be ready for some time. Using the SOFT moniker will help differentiate it
in case changes need to be done for the EXT version.
2010-09-21 09:43:35 -07:00
Chris Robinson a2580a85a8 Improve range checking for BufferSubData 2010-09-19 14:00:55 -07:00
Chris Robinson 8a0e15ff37 Store air absorption factor locally 2010-09-12 00:10:33 -07:00
Chris Robinson 8953a73fb1 Prevent effectiveDist from going negative 2010-09-11 23:57:15 -07:00
Chris Robinson 1a2883ae9a Pass dry-path click buffers as parameters to MixSource 2010-09-11 23:43:12 -07:00
Chris Robinson a2358ffc41 Source positions at and after the end of the buffer queue go back to 0 2010-09-09 18:27:14 -07:00
Chris Robinson 33eb2786f8 Avoid another non-descript incrementing variable 2010-09-09 16:47:54 -07:00
Chris Robinson c8ec4fd505 Avoid using a single-letter iterator value 2010-09-09 14:53:14 -07:00
Chris Robinson fe98ab5398 Combine multiple checks into one 2010-09-08 17:29:16 -07:00
Chris Robinson 76be7eb1e7 Better protect against sample overflow when converting float to short 2010-09-08 16:26:19 -07:00
Chris Robinson 6e32812bc1 Remove unnecessary check 2010-09-07 22:38:29 -07:00
Chris Robinson 50a7be6385 Fix comparison check 2010-09-07 22:23:37 -07:00
Chris Robinson 3763c95784 Add missing else 2010-09-07 16:36:42 -07:00
Chris Robinson 5fa5252648 Watch for a few more negative sizes 2010-09-07 16:33:17 -07:00
Chris Robinson 1e82561a22 Check for invalid negative sizes 2010-09-07 16:18:24 -07:00
Chris Robinson ba32a52bb6 Remove unused struct member 2010-09-07 16:07:35 -07:00
Chris Robinson 6027fda0ce Combine two arrays into one 2010-08-16 16:07:44 -07:00
Chris Robinson e5cae87502 Handle clicks by watching changes between updates 2010-08-16 14:53:21 -07:00
Chris Robinson ba0ec1b4ba Use the correct value to dereference an array 2010-08-16 11:42:18 -07:00
Chris Robinson 2acd6f52f1 Add a bit more information to the README 2010-08-15 16:55:11 -07:00
Chris Robinson b520228bc6 Consolidate some click removal calculations 2010-08-15 15:55:02 -07:00
Chris Robinson 343a8241ab Use a multiplier instead of if checks 2010-08-15 15:09:38 -07:00
Chris Robinson 00525274a0 Add click removal for wet sends 2010-08-15 14:52:12 -07:00
Chris Robinson 2915d9dbd9 Use click removal to handle harsh panning/volume changes 2010-08-15 00:19:38 -07:00
Chris Robinson 0c49e5ba56 Use click removal when starting a source 2010-08-14 22:59:55 -07:00
Chris Robinson b65bc45236 Add a corrective, click-remval sample value offset
This will hopefully be more effective and rebust at removing clicks and pops,
particularly from starts and stops in the middle of a sound. Based on an idea
by Ben Davis.
2010-08-14 21:30:14 -07:00
Chris Robinson 94b7e5a227 Rename Null Output to No Output 2010-08-13 22:06:38 -07:00
Chris Robinson 928572d384 Start the device when creating a context without specified attributes 2010-08-12 17:24:55 -07:00
Chris Robinson 2d1988bb6f Output on available channels only, for the echo and modulator effects 2010-08-09 06:12:00 -07:00
Chris Robinson e007dc3614 Fix early/late reverb panning, and only output reverb on available channels 2010-08-09 05:11:58 -07:00
Chris Robinson 2ed929eadc Update device parameters using a separate function 2010-08-09 00:28:48 -07:00
Chris Robinson f1924fa7ea Decrement context count when it's removed from the list 2010-08-08 02:48:28 -07:00
Chris Robinson df822513fa Don't re-init and re-allocate some stuff when the device is left running 2010-08-07 09:13:51 -07:00
Chris Robinson 762bdaeee1 Panning init cleanup 2010-08-07 08:49:29 -07:00
Chris Robinson 8003278a55 Combine non-attenuated source calculation functions 2010-08-07 06:57:31 -07:00
Chris Robinson 5f22d30fc9 Calculate the source stepping value with the param calculations 2010-08-07 05:43:16 -07:00
Chris Robinson d6dc855511 Set up increment once 2010-08-07 02:32:16 -07:00
Chris Robinson 8f49f2c25e Forward some more source properties instead of duplicating 2010-08-07 01:35:43 -07:00
Chris Robinson f4304ca062 Calculate the actual stepping value in the source update method 2010-08-07 00:38:02 -07:00
Chris Robinson 044662e4b7 Calculate gain steps once during a source mix 2010-08-06 07:55:00 -07:00
Chris Robinson d9d8319c22 Effectslot's Effectstate should never be NULL 2010-08-06 04:52:34 -07:00
Chris Robinson 56b094767b Remove an unnecessary check
The number of buffers played should never exceed the buffer count when playing,
and the buffer list is used later on regardless
2010-08-06 01:14:37 -07:00
Chris Robinson 98bc60cfb7 Combine stereo and stereo-duplicate mixers
Use a separate update method for stereo sources to the set appropriate channel
gains
2010-08-05 01:07:20 -07:00
Chris Robinson 2aacca76de Improve a warning message 2010-08-04 06:18:23 -07:00
Chris Robinson aea615edb3 Prevent re-probing while opening a Pulse device if no devices were found 2010-08-04 05:07:56 -07:00
Chris Robinson f6b2dfd09f Only except a NULL device name for Pulse if some devices were found 2010-08-04 04:56:07 -07:00
Chris Robinson 7f9bcfbc7d Do not enumerate PulseAudio devices if a context fails to connect 2010-08-04 04:21:39 -07:00
Chris Robinson 22cc5af1fe pa_context_errno can return a positive value 2010-08-04 04:11:38 -07:00
Chris Robinson 0dc3f1984e Move the core mixer functions to a separate source file 2010-08-03 23:19:36 -07:00
Chris Robinson e74976e645 Use a callback to specify the source update method 2010-08-03 23:10:00 -07:00
Chris Robinson 91278608c5 Add some spacing 2010-08-03 01:44:52 -07:00
Chris Robinson 8e2d765671 Move active source iteration to aluMixData 2010-08-03 01:09:02 -07:00
Chris Robinson 2af39e51f8 Separate speaker/panning initialization into another source file 2010-08-03 00:21:36 -07:00
Chris Robinson 1504cf8812 Use millisecond resolution for the Null and Wave Writer backends 2010-08-02 22:34:19 -07:00
Chris Robinson 3bece35bbd Prevent getting bad time values when pa_stream_get_time fails
Also makes sure that reported time won't go backwards
2010-08-02 22:27:32 -07:00
Chris Robinson ae41ad5d1e Use a better method to time the Null and Wave Writer backends
This better compensates for precision loss when converting milliseconds to the
sample rate
2010-08-02 20:04:52 -07:00
Chris Robinson ee61f7a55c Make sure to request a tlength that's at least minreq*2
And warn if Pulse returns a minreq that's too large
2010-08-02 17:24:40 -07:00
Chris Robinson 241ddf76d4 Enforce the device time to be a multiple of the reported resolution 2010-08-01 16:31:00 -07:00
Chris Robinson fe6e73ede9 Be a bit more verbose when a device fails to open 2010-08-01 16:20:28 -07:00
Chris Robinson 8dab4c418c Use Pulse's stream time for the device time 2010-07-31 21:33:46 -07:00
Chris Robinson 41e2c41ecc Combine duplicate lines into the mixing macro 2010-07-31 21:09:59 -07:00
Chris Robinson 30108c7855 Remove an unnecessary variable 2010-07-31 19:14:27 -07:00
Chris Robinson 40ba9db406 Only call Pulse functions from under a lock 2010-07-31 16:07:04 -07:00
Chris Robinson f1f2941511 Add experimental device attributes for apps to query
These will be part of the ALC_EXT_device_time extension. Behavior, values,
etc, are all subject to change.
2010-07-30 20:23:55 -07:00
Chris Robinson d886fb118b Signal the Pulse mainloop on underruns
Otherwise there are (rare) risks of not getting signaled after writing the
amount of free audio data the client knows about, leading to a mixer loop
lockup
2010-07-30 20:20:40 -07:00
Chris Robinson 46d46c49a8 Store the time precision with the device 2010-07-25 12:54:50 -07:00
Chris Robinson 40dd1fb853 Add a backend callback to get the device time
The method should monotonically count the number of nanoseconds the device has
active contexts
2010-07-24 19:14:32 -07:00
Chris Robinson ab77fc14d5 Make killNow volatile 2010-07-24 10:54:12 -07:00
Chris Robinson e0d0edc0fc Close the log file last during shutdown 2010-07-22 09:16:28 -07:00
Chris Robinson 7eb124d99c Keep count of the number of samples played 2010-07-22 08:11:44 -07:00
Chris Robinson 88f236ccd3 Make sure the device is stopped if there was an error setting up the first context 2010-07-21 14:50:39 -07:00
Chris Robinson 34a0c998cd Use CLOCK_MONOTONIC when available 2010-07-18 06:21:16 -07:00
Chris Robinson e658df493a Update PulseAudio using period-sized chunks
To match the behavior of other backends
2010-07-17 23:30:25 -07:00
Chris Robinson b2a257b6b0 Rename examples to utilities
There is only the one, and it more accurately reflects what it is
2010-07-17 01:02:51 -07:00
Chris Robinson cde577690a Set the backend HAVE_ macros off before checking their availability 2010-07-17 00:37:54 -07:00
Chris Robinson c85d53cfbe Remove the commit count from the version number
As it's only updated for releases, it's kinda useless
2010-07-16 00:18:21 -07:00
Chris Robinson 26127cd2f2 Use a long single buffer for WaveIn's capture buffers 2010-07-12 16:37:44 -07:00
Chris Robinson 42d9140afe Check the buffer's original format when queueing on a source 2010-07-12 15:53:11 -07:00
Chris Robinson 52ad28df92 Fix WaveIn device cleanup order 2010-07-08 17:37:35 -07:00
Chris Robinson 58ba5fd0dd Add missing static keyword to WinMM vars 2010-07-08 15:10:59 -07:00
Chris Robinson 71cc91cc4a Use the existing ring buffer methods for WaveIn 2010-07-07 02:47:29 -07:00
Chris Robinson 6db3463d17 Fix backslash alignment for the single-channel mixer macro 2010-07-06 16:32:58 -07:00
Chris Robinson 969818db6f Use a 100ms capture buffer minimum for PulseAudio, with 50ms fragments
This better matches what is given with the WaveIn backend
2010-07-04 16:27:42 -07:00
Chris Robinson d4b191d27d Avoid retrying the requested ALSA format after it failed 2010-07-04 15:48:47 -07:00
Chris Robinson 6b23f2dfa3 snd_pcm_recover should automatically re-prepare the stream as needed 2010-06-16 02:17:41 -07:00
Chris Robinson 17e6840f92 Prevent multiple WinMM devices from getting the same name, as well 2010-06-10 15:54:04 -07:00
Chris Robinson a1f987c016 Improve handling of lost DirectSound buffers 2010-06-09 23:32:35 -07:00
Chris Robinson cff805e0a3 Prevent multiple DirectSound devices from getting the same name, too 2010-06-09 23:11:34 -07:00
Chris Robinson 25a941666a Prevent multiple PulseAudio devices from getting the same name 2010-06-09 18:22:05 -07:00
Chris Robinson 8c59849ea4 Make sure the log file is set before initializing the rest of the lib 2010-06-08 02:02:48 -07:00
Chris Robinson f9e569099c Make the Wave Writer backend optional 2010-06-08 02:01:11 -07:00
Chris Robinson 6fd6e4eacf Remove unused variable 2010-06-07 03:49:21 -07:00
Chris Robinson ce07270973 Remove support for the old 'refresh' config option 2010-06-06 00:53:29 -07:00
Chris Robinson ac05863f72 Make sure the effectslot map is properly initialized 2010-06-06 00:21:13 -07:00
Chris Robinson 7f6df7695c Use an array of active sources when mixing
Prevents iterating over all allocated sources during mixing updates
2010-06-06 00:17:50 -07:00
Chris Robinson ed58571079 Use non-blocking mode for ALSA capture 2010-06-05 19:56:58 -07:00
Chris Robinson 80fbe8a788 Allow prepending backend names with - to remove them from the available devices 2010-06-05 19:33:06 -07:00
Chris Robinson 0fdec056fa Improve error recovery when capturing from ALSA 2010-06-05 17:41:33 -07:00
Chris Robinson 1915fe831f Read capture data from ALSA when querying the available sample count 2010-06-05 04:43:26 -07:00
Chris Robinson 98c01a40e6 Improve locking when calling the backend capture API 2010-06-05 04:42:05 -07:00
Chris Robinson 33234d605a Don't attempt to look for more samples from Pulse after a disconnect 2010-06-05 00:09:45 -07:00
Chris Robinson c6defe9a6d Read PulseAudio's capture data into the ring buffer when querying the sample count
This ensures we have all the samples available that the app knows about
2010-06-04 00:27:31 -07:00
Chris Robinson 244daba501 Don't overwrite data when the ring buffer overflows 2010-06-04 00:24:23 -07:00
Chris Robinson 43dadcd9e4 Use a UIntMap for the Databuffers 2010-06-03 19:35:08 -07:00
Chris Robinson 90db244b5f Skip mixing after checking looping validity 2010-06-03 03:58:04 -07:00
Chris Robinson e98f1205e5 Increase the temp buffer sizes to construct device names with 2010-05-31 19:04:41 -07:00
Chris Robinson 2205f91e5f Fix Pulse mainloop handling for device probing 2010-05-31 19:02:20 -07:00
Chris Robinson da1dee351d Update device naming to be more apparent about the device being used 2010-05-31 19:00:33 -07:00
Chris Robinson 694e9a5fec Always add the default PulseAudio device when probing 2010-05-30 23:04:04 -07:00
Chris Robinson 9ac5d8ad2d List all available PulseAudio sources 2010-05-30 21:05:05 -07:00
Chris Robinson 706b9f2357 Probe and expose all available PulseAudio sinks for the enumerate_all extension 2010-05-30 20:33:57 -07:00
Chris Robinson 55b3675dee Add a separate function to create/connect a pulseaudio context 2010-05-30 19:05:19 -07:00
Chris Robinson dc86563486 Pass the threaded main loop object to context/stream_state_callback 2010-05-30 18:19:01 -07:00
Chris Robinson ebccfa93c3 Don't unload libs when they're not needed
Some libs don't really like being unloaded and reloaded all the time, and the
benefits aren't that great
2010-05-28 23:00:59 -07:00
Chris Robinson e9f4576d4d Watch for changes to the PulseAudio sink device 2010-05-28 17:03:57 -07:00
Chris Robinson 880902fecc Add a Null Output device
This device will mix and fully process contexts as normal
2010-05-28 16:41:52 -07:00
Chris Robinson 96cf5b7282 Avoid constantly loading and unloading PortAudio
It seems this triggers a bug in the lib, causing a crash. It may not be a bad
idea to remove dynamic unloading/reloading all through-out.
2010-05-28 02:22:17 -07:00
Chris Robinson f89c9f7ea4 Connect a pulse stream on device open, to make sure the device isn't suspended
Also use the opportunity to get the stream's device's name, so later reset
calls will use the same device
2010-05-27 22:12:54 -07:00
Chris Robinson 27a4c871fd Update head dampening after a device reset 2010-05-26 23:36:55 -07:00
Chris Robinson 37190b2caf Add a function to get the frame size from a format 2010-05-24 21:51:30 -07:00
Chris Robinson c6b948323e Better watch for invalid values in alBufferSubDataEXT 2010-05-24 21:33:43 -07:00
Chris Robinson ad2957ed98 Reduce the mixing buffer size 2010-05-24 20:10:35 -07:00
Chris Robinson c469bfcb9e Make error checking consistent in alBuffer.c 2010-05-24 19:52:27 -07:00
Chris Robinson 3d16be1822 Fix alBufferiv error handling 2010-05-24 19:02:36 -07:00
Chris Robinson ad10d4ee74 Increment pointer when read 2010-05-24 17:40:16 -07:00
Chris Robinson 3ba3c1ca31 Pass the number of blocks to ConvertDataIMA4 2010-05-24 17:31:49 -07:00
Chris Robinson 4ad28964de Be careful of integer overflows when allocating buffer storage 2010-05-24 17:08:12 -07:00
Chris Robinson 15ce7fb8f7 Cleanup some variables and names 2010-05-24 03:28:00 -07:00
Chris Robinson 80a22b6484 Reorder some error checks and watch for negative buffer sizes 2010-05-24 03:20:14 -07:00
Chris Robinson 8cfac7c6b8 Don't bother storing the loop points in the source 2010-05-24 01:07:13 -07:00
Chris Robinson 9ac6b4c2c8 Be big endian safe with the IMA4 decoder 2010-05-22 15:37:32 -07:00
Chris Robinson 861a933b7b Use the ALSOFT_LOGFILE env var to specify the output for AL_PRINT 2010-05-21 18:16:27 -07:00
Chris Robinson 58e5404d61 Loop points should be specified in sample offsets 2010-05-21 15:52:35 -07:00
Chris Robinson 3676fdc4db Add mono and stereo channel masks for DSound output 2010-05-21 14:12:09 -07:00
Chris Robinson e7a555a953 Add an empty efx-creative.h header
This is for compatibility with projects that include it in Windows. Everything
from it is already defined in efx.h.
2010-05-21 11:21:35 -07:00
Chris Robinson 1e6ba29cd2 Consistency fix 2010-05-19 11:06:13 -07:00
Chris Robinson c5c83882b3 Use unsigned types when finding the source offsets 2010-05-19 10:38:28 -07:00
Chris Robinson ea3bc0518c A couple formatting fixes 2010-05-19 10:36:24 -07:00
Chris Robinson 2d46a1001e Use a map for the filter list 2010-05-18 17:54:45 -07:00
Chris Robinson 4edd7fe647 Use a map for the effects list 2010-05-18 17:41:06 -07:00
Chris Robinson ad5dfab428 Improve compatbility with latest AL_EXT_buffer_sub_data spec 2010-05-18 17:15:21 -07:00
Chris Robinson 85838c1d57 Define ARGSTACK and CALLSTACK only when needed for testing shared libs 2010-05-16 08:03:05 -07:00
Chris Robinson 4fbb3ccf66 Always look for winmm under Windows
It's needed for timeGetTime, even if the WinMM backend is disabled
2010-05-16 07:52:43 -07:00
Chris Robinson 7f533c6a25 Change inline -> __inline for MSVC 2010-05-16 06:02:27 -07:00
Chris Robinson 6dab9d54d1 Use the loop start when filling in padding data for looping sources 2010-05-13 06:52:44 -07:00
Chris Robinson 7ed5d12717 Pay attention to the source loop points 2010-05-13 04:53:21 -07:00
Chris Robinson 23b2c1f902 Only use a modulo to loop when the source is static 2010-05-13 02:12:50 -07:00
Chris Robinson 5d579dd1a8 Store loop points as uncompressed frame offsets in the source when a buffer is attached 2010-05-13 02:10:33 -07:00
Chris Robinson 0ce138341c Add initial structural support for AL_EXT_loop_points 2010-05-13 02:03:48 -07:00
Chris Robinson 61b682131e Reduce some indentation 2010-05-12 07:30:45 -07:00
Chris Robinson a53e3dda75 Don't needlessly expose a variable for the backends 2010-05-12 07:27:12 -07:00
Chris Robinson 2220c22c90 Move incomplete extension declarations to alMain.h 2010-05-12 06:59:47 -07:00
Chris Robinson e3a70e5021 Scale output of effects to compensate for device down-mixing 2010-05-12 04:56:03 -07:00
Chris Robinson af4faaf666 Use a UIntMap for the effect slot list 2010-05-12 02:20:14 -07:00
Chris Robinson f2548570b8 Store the buffer's original data size and block alignment 2010-05-12 01:36:09 -07:00
Chris Robinson d678f1d88b Store looping locally when mixing a source 2010-05-12 00:38:04 -07:00
Chris Robinson f0ce1bff3e Simplify and fix applying a source offset 2010-05-11 11:59:41 -07:00
Chris Robinson ab3c23478d Use a separate function to convert compressed bytes to frames 2010-05-11 11:06:48 -07:00
Chris Robinson 4e65e9bfa9 Only a streaming source can have a processed buffer 2010-05-11 08:53:38 -07:00
Chris Robinson 053aeee5d3 Rename a variable 2010-05-11 07:09:44 -07:00
Chris Robinson 1652dc02ab Look for the map insertion point using a binary search 2010-05-10 11:40:57 -07:00
Chris Robinson 099c2f3593 Use a simple loop to set multi-channel dry gains 2010-05-09 22:06:10 -07:00
Chris Robinson 0378422fcb Use a map to store sources and buffers
And do a lookup using a binary search instead of linear
2010-05-01 19:59:41 -07:00
Chris Robinson 0760415d08 Use doubles instead of floats for the source offset calculations 2010-04-28 17:58:20 -07:00
Chris Robinson 5730ccaa39 Combine AL and ALC function lists 2010-04-28 15:07:38 -07:00
Chris Robinson aaa7082aa8 Get the offset even if the current buffer is 0 2010-04-28 14:08:10 -07:00
Chris Robinson 5d7815beed Only apply ambient gain to available output channels 2010-04-27 11:39:54 -07:00
Chris Robinson a0ee2ccce7 Add a couple missing values to the *GetEnumValue tables 2010-04-25 08:40:37 -07:00
Chris Robinson 110be7f500 Add an option to scale the output mix 2010-04-23 07:54:43 -07:00
Chris Robinson 52663e8a44 Combine two loops into one 2010-04-23 07:23:38 -07:00
Chris Robinson 8646647b3e Update stereodup config option comment 2010-04-22 05:22:20 -07:00
Chris Robinson a04dac670e Make stereo duplication a property of the device
Further, don't use it with mono/stereo output
2010-04-20 03:57:40 -07:00
Chris Robinson 81dd71dd71 Stop and start the backend device only if new attributes are being requested 2010-04-17 00:50:28 -07:00
Chris Robinson 4d15391c88 Disable PulseAudio's write callback when disconnecting 2010-04-17 00:27:13 -07:00
Chris Robinson fc2473f826 Translate the source position separately
This is to handle the case where an app specifies the same values for the
source and listener, and expects centered panning. This fails due to floating-
point errors in the matrix, causing the result to be ever-so-slightly off of 0.

This error would normally be hidden by the position normalization, which will
not lengthen a distance shorter than the reference distance so the panning
would be nearly imperceptible. But that also fails if the reference distance
is set to 0, causing the position to expand to a full unit.

Keep the 4x4 matrix calculations, however. It will still be useful for the
requested listener matrix extension.
2010-04-16 02:09:53 -07:00
Chris Robinson 000d848f0b Return immediately when no layout is specified 2010-04-11 23:03:36 -07:00
Chris Robinson 3c5007352e More type fixes 2010-04-11 20:17:43 -07:00
Chris Robinson e0fa497948 Use a linear scaling when sending multi-channel sources to auxiliary slots 2010-04-11 15:20:46 -07:00
Chris Robinson f082c73539 Use proper types for some variables 2010-04-11 14:33:22 -07:00
Chris Robinson d4f8bd665f Only consider output speakers for the panning LUT 2010-04-11 14:23:10 -07:00
Chris Robinson 58d0db6eab Don't use LIB_INSTALL_DIR since that can be the full path 2010-04-11 12:42:47 -07:00
Chris Robinson 1544ddf0a5 Avoid reading buffer 0's size 2010-04-10 17:31:08 -07:00
Chris Robinson 488ec612bb Avoid hard-coding the channel count 2010-04-09 14:21:47 -07:00
Chris Robinson 484702a2e5 Get the speaker arrangement before setting up the channel matrix 2010-04-09 12:27:30 -07:00
Chris Robinson 371058fb9e Re-enable stereo duplication 2010-04-08 23:00:14 -07:00
Chris Robinson 97b00670f5 Use the proper type for the speaker-to-channel map 2010-04-08 18:05:24 -07:00
Chris Robinson 704b386cc5 Scale the ring modulator's mono input when expanding to multi-channel 2010-04-08 16:34:51 -07:00
Chris Robinson aace50ebf4 Store the panning LUT in the device 2010-04-08 15:58:11 -07:00
Chris Robinson 49201e892b Use a single layout for the virtual speakers
The downmix should probably be improved to take the angles into account
2010-04-08 15:41:32 -07:00
Chris Robinson 19b8deba66 Apply a scalar to the output downmix so channels don't get more than a 100% mix
All channels get the same scalar so channel balance (should be) maintained
2010-04-08 15:23:03 -07:00
Chris Robinson 068c0076af Set the device channel matrix when the device is reset 2010-04-08 15:02:03 -07:00
Chris Robinson 37c93a6ad8 Always mix internally at 8.1
The channels are remixed when writing to the output buffer. Stereo duplication
is currently broken, but this can be restored later
2010-04-08 14:47:12 -07:00
Chris Robinson 1b1c76da34 Invert device channel map array 2010-04-08 13:11:21 -07:00
Chris Robinson 5db1aec1d1 Make some functions static 2010-04-08 12:14:18 -07:00
Chris Robinson 05ca5c88ef Update alsoftrc.sample for the ring modulator 2010-04-08 10:44:35 -07:00
Chris Robinson 980167c99b Modulate with the requested waveform in the ring modulator 2010-04-08 10:37:09 -07:00
Chris Robinson 76f8c1e4e2 Implement a basic high-pass cutoff filter in the ring modulator 2010-04-08 09:36:20 -07:00
Chris Robinson b273d616ad Implement a skeleton Ring Modulator effect 2010-04-08 08:54:28 -07:00
Chris Robinson c16b895460 Fix non-dynamic PortAudio 2010-04-01 03:25:39 -07:00
Chris Robinson ec1a4884d9 Release 1.12.854 2010-03-29 22:05:21 -07:00
Chris Robinson bb4ed8b1db ALC_EXT_thread_local_context is now complete 2010-03-29 22:01:54 -07:00
Chris Robinson 4e54f93477 Make sure ALbyte and ALCbyte are signed 2010-03-28 23:24:41 -07:00
Chris Robinson 4ac1772149 No need to continue supporting _OPENAL32LIB 2010-03-28 02:12:32 -07:00
Chris Robinson 3f235c51f6 Remove a couple unnecessary lines 2010-03-26 21:58:19 -07:00
Chris Robinson 7bdf0d5156 Shorten some variable names 2010-03-26 00:41:27 -07:00
Chris Robinson d2a6fd58a3 Use unsigned types for mono and stereo source count hints 2010-03-25 18:39:16 -07:00
Chris Robinson 05098862c5 Fix comparison check 2010-03-24 23:48:38 -07:00
Chris Robinson c8dd0246da Fix alcGetEnumValue error handling 2010-03-24 22:57:35 -07:00
Chris Robinson b3c389c0a0 Constify and properly type some variables and struct members 2010-03-24 22:43:08 -07:00
Chris Robinson 15b9fc9099 Improve timing for the wave file writer 2010-03-24 22:20:55 -07:00
Chris Robinson 1c14c13c2f Fix 8-bit output with the Wave Writer 2010-03-24 16:58:58 -07:00
Chris Robinson b79b92967f Fix comment typo 2010-03-24 16:53:11 -07:00
Chris Robinson f77832fea2 Remove unnecessary NULL check 2010-03-24 03:43:37 -07:00
Chris Robinson 4ce8a17afd Reduce some indentation 2010-03-24 02:23:00 -07:00
Chris Robinson 702c62c9a2 Use a global handle to store the process-wide context 2010-03-23 21:05:29 -07:00
Chris Robinson 661059bf0f Rename alcMakeCurrent to alcSetThreadContext 2010-03-23 19:49:31 -07:00
Chris Robinson 7fb4c4ce4f Move extension function declarations to alext.h/efx.h 2010-03-23 18:21:55 -07:00
Chris Robinson 52834744fa Return if PortAudio fails to load in Windows 2010-03-23 17:45:37 -07:00
Chris Robinson 3e9934e810 Mark extension functions with AL_API/ALC_API 2010-03-23 17:44:01 -07:00
Chris Robinson 556e9b8fee Use protected visibility for exported API functions
Protected visibility is like default visibility, in that functions will be
"exported" from the library. However, it also guarantees that references to the
functions from within the library will be to the library's version, even if the
symbols are overriden by the application.
2010-03-22 23:10:29 -07:00
Chris Robinson ef92c63455 Define AL_BUILD_LIBRARY only when building the lib 2010-03-22 23:09:04 -07:00
Chris Robinson ae8858c516 Whitespace fixes 2010-03-22 15:12:20 -07:00
Chris Robinson 0691486813 Support PortAudio in Windows 2010-03-21 20:57:15 -07:00
Chris Robinson 49db3bf916 Cleanup and fixup Release* functions 2010-03-20 21:49:02 -07:00
Chris Robinson 99f28f25b0 Avoid calling alDelete* from alGen* 2010-03-20 21:38:05 -07:00
Chris Robinson 27358c8ce8 Don't expose enums for unsupported filter types 2010-03-20 20:30:43 -07:00
Chris Robinson 899fbdfe08 Add EAX Reverb and Echo parameters to the enum table 2010-03-20 20:24:50 -07:00
Chris Robinson eea23a012a Relax circular layout requirement 2010-03-20 00:54:44 -07:00
Chris Robinson 2235a53824 Reduce some indentation 2010-03-19 20:49:23 -07:00
Chris Robinson 52f82f0b94 Simplify removing the context handle from the device's context array 2010-03-19 14:53:32 -07:00
Chris Robinson a572b13743 Don't use deprecated macros 2010-03-19 14:34:18 -07:00
Chris Robinson 4340a6f8fe Remove another unnecessary assert 2010-03-19 01:35:57 -07:00
Chris Robinson 4be97dc735 Use proper storage type for the new format 2010-03-19 01:26:56 -07:00
Chris Robinson 799126df0e Use a better method to derive the offset from the data pointer 2010-03-18 19:32:03 -07:00
Alam Arias a65ec178df cast AL_SIZE value to ALint 2010-03-18 16:33:16 -07:00
Alam Arias cf4027df38 Mingw64's shlobj.h wants at least 0x501 2010-03-18 16:33:08 -07:00
Alam Arias 127b845960 DXSDK_DIR is MSVC only 2010-03-18 16:32:53 -07:00
Alam Arias e3637b8452 use ALintptrEXT in alcReverb.c 2010-03-18 16:32:49 -07:00
Alam Arias 745d36dc76 add _CRT_NONSTDC_NO_DEPRECATE define for MSVC 2010-03-18 16:32:44 -07:00
Chris Robinson 2355d6a4a0 Use ptrdiff types for databuffer offset/length handling
Should improve addressing for non-32-bit systems
2010-03-18 14:46:33 -07:00
Chris Robinson e1965fe7d9 Convert CR/LF -> LF
Spotted by Alam Arias
2010-03-18 14:23:31 -07:00
Chris Robinson 781444d12c Use size_t where appropriate
Spotted by Alam Arias
2010-03-18 14:21:06 -07:00
Chris Robinson b6571904ce Define _WIN32_IE to 0x400 only if it's a lesser value by default
Spotted by Alam Arias
2010-03-18 13:46:18 -07:00
Chris Robinson 97b25755e7 Support capture with PortAudio 2010-03-18 01:58:25 -07:00
Chris Robinson bdfafb454b Create WAVEFORMATEXTENSIBLE wave files for multi-channel and float support 2010-03-18 00:53:58 -07:00
Chris Robinson 0fee20a415 Simplifiy verification loops 2010-03-17 20:16:02 -07:00
Chris Robinson 186e0f3b04 Use a separate list for NULL-device extensions 2010-03-17 19:46:20 -07:00
Chris Robinson 2e6c5e590f Remove old comment 2010-03-17 19:28:01 -07:00
Chris Robinson 9d4e715ee7 Don't return unsupported effects from alGetEnumValue 2010-03-17 19:20:52 -07:00
Chris Robinson 932e5ec9b8 The enums are the same as with mcforamts, so advertise AL_EXT_MULAW too 2010-03-17 19:15:51 -07:00
Chris Robinson af5fb1d535 Remove unnecessary assert 2010-03-17 19:04:02 -07:00
Chris Robinson 2953daebf5 GetSourceOffset can't fail 2010-03-17 15:19:00 -07:00
Chris Robinson dd82b68175 Handle source offsets for muLaw formats 2010-03-17 15:15:12 -07:00
Chris Robinson 05a081a7b8 Allow selecting databuffer 0 2010-03-17 14:04:25 -07:00
Chris Robinson 9c417d33cc clock_gettime is available only when _POSIX_TIMERS is greater than 0 2010-03-17 12:04:33 -07:00
Chris Robinson 975f92fc84 Remove unnecessary stop call 2010-03-16 21:29:55 -07:00
Chris Robinson 0097beb84c Remove unused variable 2010-03-16 21:12:36 -07:00
Chris Robinson 65e16cc8bb Remove unneeded checks 2010-03-16 20:29:01 -07:00
Chris Robinson 2687155200 Get rid of more alIs* calls 2010-03-16 20:23:46 -07:00
Chris Robinson 49321961e8 Effect and filter ID 0 is valid 2010-03-16 19:14:05 -07:00
Chris Robinson 65f43637ce Avoid some alIs* calls while under the context lock
Instead, use functions that verify the ID by searching the object list and
returning its handle
2010-03-16 18:54:36 -07:00
Chris Robinson f37cfc486d Pass the context to alSetError 2010-03-16 17:35:51 -07:00
Chris Robinson a337a59fb8 Use alext.h for extension definitions 2010-03-16 16:25:30 -07:00
Chris Robinson ec917e8e2f Rename some struct members for consistency 2010-03-16 15:37:41 -07:00
Chris Robinson 89d84131a4 Avoid dereferencing a NULL buffer when unqueueing buffer 0 2010-03-15 16:10:21 -07:00
Chris Robinson 2033f60d24 Print an error when realloc fails 2010-03-15 14:28:38 -07:00
Chris Robinson d4159430f0 Set the buffer length for Pulse when opening the device
So it will properly scale with context frequency requests
2010-03-14 22:20:55 -07:00
Chris Robinson 14255a38fc Pulse's buffer size is not a good measurement for the update size
Lie and say 20ms instead, since there is no stable method to get such info
2010-03-14 21:45:23 -07:00
Chris Robinson 6c5f1b78d7 Reduce the mixing buffers to the next power of two 2010-03-14 03:30:42 -07:00
Chris Robinson 1f2d3a0132 Remove duplication of setting source angles 2010-03-14 03:23:18 -07:00
Chris Robinson c23e4ed412 Add support for AL_EXT_DOUBLE 2010-03-13 21:32:45 -08:00
Chris Robinson 8c0118e7ae Add support for AL_EXT_MULAW_MCFORMATS 2010-03-13 21:03:34 -08:00
Chris Robinson 9d4107a515 Add AL_EXT_MULAW_MCFORMATS tokens to alext.h 2010-03-13 18:39:12 -08:00
Chris Robinson d07b9c2c11 Rewrite openal-info
Output is mostly unchanged, however there are differences. Device-specific info
(including context-specific info) is separated from NULL-device queries, and
there is now a distinction between ALC versions of the NULL device and the
opened device. Additionally, an alternate playback device can be queried by
specifying its name as the first parameter.
2010-03-13 18:03:01 -08:00
Chris Robinson 3108a6a997 Limit PulseAudio's buffer size to the requested length 2010-03-11 09:47:26 -08:00
Chris Robinson b734cab6fc Fix the retrieval of the RW offsets using the correct functions 2010-03-10 07:33:30 -08:00
Chris Robinson 08b1bed38b Remove a couple incorrect comments 2010-03-10 07:16:53 -08:00
Chris Robinson df6320eb0a Remove duplicate copyright notice 2010-03-09 11:25:34 -08:00
Chris Robinson 80f1ec0d7e Protect some capture calls with a mutex 2010-03-09 10:14:52 -08:00
Chris Robinson 98f4f5a4f0 Add EFX types and enums to alext.h
The types and enums are in efx.h, which is included by alext.h. It's done this
way because EFX has a lot if definitions which would polute alext.h
2010-03-09 09:28:22 -08:00
Chris Robinson 8feb089f5c Build device lists only when needed 2010-03-09 06:09:29 -08:00
Chris Robinson 2ba3a88ace Probe physical devices separately from appending them to the device list 2010-03-09 05:44:18 -08:00
Chris Robinson 1f10195c47 Use powf when available 2010-03-07 22:12:33 -08:00
Chris Robinson dc40702b53 Remove unused source param count macro 2010-03-07 21:48:25 -08:00
Chris Robinson a9e0e57797 Keep UpdateSize updated when apps request a frequency 2010-02-24 18:51:57 -08:00
Chris Robinson 9b47048231 Silence some warnings 2010-02-16 05:44:39 -08:00
Chris Robinson c05a3d3549 Better handle run-time linking 2010-02-16 05:43:36 -08:00
Chris Robinson 2083884ae3 Properly check for functions in Win32 shared libs 2010-02-16 05:36:49 -08:00
Chris Robinson 2f644c7ed5 Constify some globals 2010-02-15 22:01:23 -08:00
Chris Robinson d17c4a098c Avoid probing for devices on lib initialization 2010-02-10 16:34:43 -08:00
Chris Robinson 96303893bc Set ALSA playback buffer and period size based on time
So rate adjustments won't modify the latency
2010-02-08 17:42:44 -08:00
Chris Robinson a329f9c84a Fix retrieval of the current global context when changing it 2010-02-08 16:58:26 -08:00
Chris Robinson fc96e965ae Always disable ALSA's resampler for playback
This means the frequency config option won't have an effect if the ALSA device
can't set an alternate sampling rate
2010-02-08 16:38:53 -08:00
Chris Robinson 365a62d63a Improve source write offset calculations 2010-02-07 20:11:24 -08:00
Chris Robinson 3a87c3c3ba Avoid using periods with PulseAudio, and add an option to specify its buffer length 2010-02-07 19:29:48 -08:00
Chris Robinson 54dc26b145 Enable the WinMM backend only if the lib check succeeds 2010-02-05 12:07:51 -08:00
Chris Robinson 78c1629aeb Disable ALSA's resampler when no specific output rate is requested 2010-02-05 11:45:17 -08:00
Chris Robinson bd518ed264 Make sure all requested data gets written to PulseAudio 2010-01-24 00:03:23 -08:00
Chris Robinson 56379bffac Use pa_channel_map_superset only when available with PulseAudio 0.9.15
pa_channel_map_equal can otherwise detect some compatible output modes
2010-01-22 21:20:08 -08:00
Chris Robinson 11d7a0e394 Specify the buffer size to ALSA, so the period count remains flexible
Under conditions where the period size is fixed (such as dmix), ALSA will still
attempt to provide the requested number of periods even if the requested period
size needs to be dramatically altered. In these cases, it would be better to
increase or decrease the period count, instead of growing or shrinking the
total buffer size and significantly changing the latency.
2010-01-20 01:19:32 -08:00
Chris Robinson 3ad86a1c1b Release 1.11.753 2010-01-17 17:46:27 -08:00
Chris Robinson 931f5875cd Fix speaker angle fixup 2010-01-12 09:11:46 -08:00
Chris Robinson f22b02a762 al_print doesn't need to be inline 2010-01-12 09:05:57 -08:00
Chris Robinson 505f61ff43 Properly trace the invalid speaker angle 2010-01-12 08:50:32 -08:00
Chris Robinson 0c2efa1253 Use GCC's format attribute for al_printf 2010-01-12 08:48:25 -08:00
Chris Robinson 5921e18be2 Fix some typos with the solaris backend 2010-01-12 08:28:07 -08:00
Chris Robinson 5b9e6fe440 Only allow 8- and 16-bit, and mono and stereo, wave file output 2010-01-12 08:26:25 -08:00
Chris Robinson 35ba7d70dc Consolidate some source offset handling code 2010-01-12 02:43:37 -08:00
Chris Robinson 026b4b8f4c Better handle large second offsets 2010-01-12 02:26:38 -08:00
Chris Robinson c924a50bed Don't use a flag to set an error when applying the source offset 2010-01-12 02:22:38 -08:00
Chris Robinson 574792d2e1 Fix Windows compilation 2010-01-11 23:47:03 -08:00
Chris Robinson ab2eb5a603 Check buffer size when copying padding instead of the data pointer 2010-01-11 10:58:33 -08:00
Chris Robinson 3d7f925d52 Add helpful values to the resampler enum 2010-01-11 07:02:29 -08:00
Chris Robinson 93b584ff84 Add a cosine resampler 2010-01-11 06:19:54 -08:00
Chris Robinson 705849ca73 Add an option for point resampling 2010-01-11 05:37:20 -08:00
Chris Robinson 1a57f095e8 Hold global lock during mixing only when iterating device contexts 2010-01-11 03:57:49 -08:00
Chris Robinson ac9c2e8c4a Cleanup OSS/Solaris loop a bit 2010-01-10 12:40:59 -08:00
Chris Robinson 45f8efec41 Return the default config string for empty entries 2010-01-09 06:48:43 -08:00
Chris Robinson ef7df2416d Hide OSS and Solaris devices if their files can't be stat'd 2010-01-09 06:40:50 -08:00
Chris Robinson cb113315f7 Promote AL_EXT_source_distance_model to complete 2010-01-09 03:27:16 -08:00
Chris Robinson d61f90a9c2 Make sure the device handle is valid when getting the device error 2010-01-09 03:25:13 -08:00
Chris Robinson 3d11fb219e Hold the default context extensions globally 2010-01-09 02:48:18 -08:00
Chris Robinson 4d9a368f96 Suppress connection refused errors
Since that just typically means the server isn't running
2010-01-08 01:43:27 -08:00
Chris Robinson 43220b604c Reduce some indentation 2010-01-08 01:21:57 -08:00
Chris Robinson 91f28fa87c Try the PulseAudio backend first 2010-01-08 01:08:27 -08:00
Chris Robinson 82547db2db Prepare the pcm handle after recovering 2010-01-06 23:02:28 -08:00
Chris Robinson 64ac3a4739 Use snd_pcm_recover in more places 2010-01-06 22:55:20 -08:00
Chris Robinson c90616b963 Use snd_pcm_recover to recover from certain device errors 2010-01-06 22:50:30 -08:00
Chris Robinson 0ceaa01c3d Adjust PulseAudio's capture latency 2009-12-30 15:33:16 -08:00
Chris Robinson 7d7e503603 Properly retrieve device-inspecific errors 2009-12-28 23:29:49 -08:00
Chris Robinson 7ee81eac0f Store ALC errors with the device when possible 2009-12-28 23:19:13 -08:00
Chris Robinson c9a08fc7b2 Remove some unneeded functions 2009-12-28 13:12:45 -08:00
Chris Robinson 69ab93a824 Add a function to check if a config option is set to a non-empty value 2009-12-28 13:08:15 -08:00
Chris Robinson 179b660eee Enforce a lower bound playback frequency of 8khz 2009-12-28 12:47:59 -08:00
Chris Robinson a0ef7be9a5 Improve layout parsing and allow using long names for speakers 2009-12-28 11:49:55 -08:00
Chris Robinson 2a735b14dc Add an option to spawn a PulseAudio server on request 2009-12-28 10:49:31 -08:00
Chris Robinson a294dd8d9b Be more lenient in signaling the mainloop 2009-12-28 10:15:59 -08:00
Chris Robinson dc1522282b Try to find a compatible format from Pulse's default sink 2009-12-27 11:05:59 -08:00
Chris Robinson 528a4c6a02 Update panning for existing AL contexts when creating a new one 2009-12-26 20:27:14 -08:00
Chris Robinson 1770a8b62b A NULL Pulse context won't have a valid error code 2009-12-26 12:18:16 -08:00
Chris Robinson 77a0a6de95 Wait for the buffer attrib update from pulse to finish 2009-12-26 11:30:41 -08:00
Chris Robinson c1fd977e79 Consolidate pulse capture failure path 2009-12-26 10:46:29 -08:00
Chris Robinson cbbda93292 Make sure a proper channel map is set for pulse capture 2009-12-26 09:17:03 -08:00
Chris Robinson c6340ce12d Clean up some loading checks 2009-12-26 08:49:11 -08:00
Chris Robinson 7bc739e965 Use the proper logging function 2009-12-26 08:16:25 -08:00
Chris Robinson 9f988808c7 Start and stop PortAudio's stream in the proper methods 2009-12-26 08:14:28 -08:00
Chris Robinson 6e9ce47a4b GetConfigValueBool should take an int, not a float 2009-12-26 07:42:57 -08:00
Chris Robinson 89f59c0e1c Shorten a long line 2009-12-26 07:42:16 -08:00
Chris Robinson 3793919892 Capture as much as possible directly from PulseAudio 2009-12-25 17:56:57 -08:00
Chris Robinson d1699d50f6 Add missing newline 2009-12-25 15:18:32 -08:00
Chris Robinson b94d950b1e Undefine no longer needed macro 2009-12-25 14:11:43 -08:00
Chris Robinson d3e77a0f63 Fix incorrect error check 2009-12-25 13:19:25 -08:00
Chris Robinson 965a71ee90 Only use pa_stream_set_buffer_attr_callback when available (0.9.15+) 2009-12-25 13:17:21 -08:00
Chris Robinson fd794b44b2 Use pa_stream_begin_write when available for "zero-copy" writing
Must be compiled against 0.9.16 or newer to be available. It will fall back to
the old method if the function is not available at run-time.
2009-12-25 13:11:17 -08:00
Chris Robinson c9dcba17a9 Check error returns instead of error string 2009-12-25 13:05:04 -08:00
Chris Robinson 88de4e15b3 Mark the source for updating when setting a buffer on it
Changing the buffer can change the playback format, which can require different
source parameter calculations. Queueing a buffer on a source that has no
buffers (or only buffers without a format, eg. buffer 0) has the same effect
2009-12-25 05:09:29 -08:00
Chris Robinson 7f0c6629b9 Use a 64-bit value to scale potentially-large numbers 2009-12-24 15:41:45 -08:00
Chris Robinson f6ca39403d Use the Pulse server-specified playback rate by default 2009-12-24 15:35:22 -08:00
Chris Robinson ea7cf35ea4 Use pa_xmalloc instead of pa_xmalloc0 2009-12-24 07:38:46 -08:00
Chris Robinson accea119d9 Update pulse capture ring buffer in 25ms chunks 2009-12-24 04:08:04 -08:00
Chris Robinson 6ea7b43c90 Don't force latency adjustment with PulseAudio 2009-12-23 13:59:58 -08:00
Chris Robinson ddfad996a9 Don't render mono as stereo 2009-12-21 02:59:30 -08:00
Chris Robinson d786f99094 Scale samples down when converting stereo to mono 2009-12-21 02:27:25 -08:00
Chris Robinson 008563450c Improve some indentations 2009-12-20 21:50:00 -08:00
Chris Robinson 69c238a2e6 Properly retrieve the device frequency 2009-12-20 21:19:57 -08:00
Chris Robinson b3fdf170b1 Fix header comment for AL_PITCH 2009-12-16 22:50:54 -08:00
Chris Robinson 56cc038603 Improve initial decay calculations for sends to reverb 2009-12-10 18:37:36 -08:00
Chris Robinson 4c83c689a8 Use the specified reverb air absorption value for wet sends 2009-12-10 18:22:45 -08:00
Chris Robinson 2b7953c472 Track whether a context is suspended or not 2009-12-09 12:14:53 -08:00
Chris Robinson 368ef1cea9 Store the source state locally after making sure it needs mixing 2009-12-09 11:56:53 -08:00
Chris Robinson bd696a6b37 Fix IMA ADPCM offset calculation 2009-12-09 09:50:09 -08:00
Chris Robinson a9fc272e39 Prettify CMake checks for compiler switches 2009-12-09 08:46:35 -08:00
Chris Robinson c5952d1ddc Move the pkg-config file to the base directory 2009-12-09 07:36:06 -08:00
Chris Robinson 656a406377 Use an inline function to calculate the low-pass filter coefficient 2009-12-09 07:21:59 -08:00
Chris Robinson 5fcd6cc510 Split non-attenuated source calculations into a separate function 2009-12-09 07:02:26 -08:00
Chris Robinson e09b6020a6 Fix typo for reverb creation 2009-12-09 06:46:06 -08:00
Chris Robinson 84d2d623b6 Add a head-dampening option
This simulates occlusion of the player's head for sounds coming from behind,
when outputing to mono or stereo
2009-12-08 14:18:07 -08:00
Chris Robinson 1694e5bd12 Store the original pre-clamped distance instead of re-computing it 2009-12-08 10:12:18 -08:00
Chris Robinson 5170e251f9 Use the correct wet filter history offset 2009-12-07 12:45:23 -08:00
Chris Robinson 4697e946d3 Internally store 32-bit float buffer data, and mix accordingly 2009-12-07 10:35:30 -08:00
Chris Robinson 9286e3984c Move default channel order setting out of the header 2009-12-07 04:19:33 -08:00
Chris Robinson 88f3d747b7 Add a simple README 2009-12-07 01:05:01 -08:00
Chris Robinson d4427c54ea Change internal order to match WFX 2009-12-07 00:51:27 -08:00
Chris Robinson 92ade80557 Fix 6.1 channel order 2009-12-07 00:49:56 -08:00
Chris Robinson 8138446b8c Restrict setting some wet path values to active sends only 2009-12-06 03:59:12 -08:00
Chris Robinson 58ecc7fad4 Check for potential NULL accesses with the IsBadWritePtr wrapper 2009-12-04 01:33:50 -08:00
Chris Robinson 311cc5c62f Don't wait to accept stream connections 2009-12-03 04:43:38 -08:00
Chris Robinson 4a928ea9fa Don't signal on synchronous changes and don't wait to accept context changes 2009-12-02 08:10:45 -08:00
Chris Robinson c1eb444a39 Remove some assumptions of 16-bit internal storage 2009-12-02 05:55:33 -08:00
Chris Robinson b5270e0bb3 Use a channel-map to specify the output device channel order 2009-12-02 04:03:51 -08:00
Chris Robinson 6cfc31777b Add an option for real-time priority mixing
Default to disable for now, as a safety precaution
2009-12-01 23:15:09 -08:00
Chris Robinson 0e1e8503e0 Properly accept PulseAudio's mainloop signal on connection failure 2009-12-01 22:42:43 -08:00
Chris Robinson 974d0b4e91 Fix retrieved update size from pulseaudio 2009-12-01 12:21:02 -08:00
Chris Robinson d66832a39c Accept a signal only when the pulseaudio stream/context is ready 2009-12-01 12:18:34 -08:00
Chris Robinson d5889a941f Clamp dry gain after cone attenuation is applied 2009-12-01 03:32:04 -08:00
Chris Robinson 6248e986e5 Don't complain if the length given by pulse isn't an exact multiple of the period size 2009-12-01 02:23:38 -08:00
Chris Robinson 786b98a013 Avoid setting a field redundantly 2009-11-29 23:07:04 -08:00
Chris Robinson f63d8dbf38 Enable real-time priority for ALSA, OSS, and DirectSound mixing loops 2009-11-29 23:02:21 -08:00
Chris Robinson 15aad09ce7 Check for the pthread_setschedparam function 2009-11-29 22:44:56 -08:00
Chris Robinson fdd314f7f8 Reset the kill flag after the thread has stopped 2009-11-29 10:22:12 -08:00
Chris Robinson 39b456d88c Add AL_EXT_double tokens to alext.h 2009-11-28 22:19:04 -08:00
Chris Robinson d930fb8d3d Call the Update effect method for null effects 2009-11-28 20:09:41 -08:00
Chris Robinson 3ec64eb0fd Call to the right databuffer function 2009-11-28 19:37:29 -08:00
Chris Robinson d81c9819b8 Remove unnecessary returns 2009-11-28 19:36:05 -08:00
Chris Robinson 98ce1d14c1 Update AL_EXTX_source_distance_model to require explicit enabling
The in-progress spec has been updated to reflect this
2009-11-27 20:05:21 -08:00
Chris Robinson 69f9ab88b9 Send multi-channel sources to auxiliary effect slots
They are downmixed to mono using a volume-preserving scalar, and passed
through a 1-pole low-pass filter (not chained)
2009-11-26 00:39:32 -08:00
Chris Robinson 1a0676f0eb Mark sources for updating when a new context is made from the device 2009-11-25 16:28:19 -08:00
Chris Robinson 658923175f Update source parameters only when they need changing 2009-11-25 16:21:47 -08:00
Chris Robinson 877f4340ba Base air absorption on distance attenuation, try 2 2009-11-24 21:19:11 -08:00
Chris Robinson 35a9ccb41f Revert "Base air absorption on the distance attenuation"
This reverts commit c7c2f9385f.

Fundamentally wrong. The -6dB rolloff is every doubling of the distance, and
the (scaled) -0.05dB rolloff is supposed to be every additional unit.
2009-11-24 01:39:34 -08:00
Chris Robinson c7c2f9385f Base air absorption on the distance attenuation 2009-11-23 16:25:37 -08:00
Chris Robinson d1f3dbb9a1 Make modulation index 0 map to offset 0 2009-11-23 04:14:12 -08:00
Chris Robinson cfb30fe975 Use a 4x4 matrix to transform coordinates 2009-11-23 04:13:51 -08:00
Chris Robinson 2d1b378ef2 Transform all relevant vectors for converting world-space to listener-space 2009-11-22 22:36:20 -08:00
Chris Robinson 06bbee2449 Add a function for 1-pole filtering 2009-11-22 21:20:27 -08:00
Chris Robinson 85bc300dfa ALCAPIENTRY is deprecated in favor of ALC_APIENTRY 2009-11-22 21:15:11 -08:00
Chris Robinson 0662adacb9 Fix alBufferDataStatic extension name in alext.h 2009-11-22 21:12:43 -08:00
Chris Robinson 65a153da6c Update the drivers config file comment 2009-11-22 15:43:05 -08:00
Chris Robinson c5ee010640 The echo should not take energy-attenuated input 2009-11-22 15:37:32 -08:00
Chris Robinson 1b18b4a0ee Allow unlisted backends to stay available when the drivers list ends with , 2009-11-22 01:06:05 -08:00
Chris Robinson 58765aab6b Print available devices before trying to open one 2009-11-19 19:02:16 -08:00
Chris Robinson 1483af1baa Watch for NULL and empty device lists 2009-11-19 18:49:24 -08:00
Chris Robinson 13979793f6 Display supported filters using a comma-separated listing 2009-11-19 18:46:23 -08:00
Chris Robinson 9c228e7625 Be clearer if device opening or context setup fails 2009-11-19 18:25:29 -08:00
Chris Robinson 6a667b36d1 Reorganize and improve the reverb effect
Code supplied by Christopher Fitzgerald. This update also implements the echo
and modulation parameters.
2009-11-19 14:05:04 -08:00
Chris Robinson fe37f1968d Better approximate the wet attenuation model for reverb 2009-11-19 11:06:41 -08:00
Chris Robinson 4565bdd7ce Make sure there's at least a one-sample delay for the echo's first tap 2009-11-19 10:42:41 -08:00
Chris Robinson 5f3c07ca3b Apply the reverb room rolloff factor for EAX reverb too 2009-11-19 10:29:10 -08:00
Chris Robinson 8d1da6a8ca Move NextPowerOf2 to alMain.h 2009-11-19 09:50:15 -08:00
Chris Robinson ff8e09495d Move aluCart2LUTpos to alu.h 2009-11-19 09:24:35 -08:00
Chris Robinson e868eef679 Make sure to set the new echo sample buffer and remove an unused macro 2009-11-18 17:34:08 -08:00
Chris Robinson ea90f781ff Release 1.10.622 2009-11-08 21:36:44 -08:00
Chris Robinson c1b161b44b Handle NULL effect objects as AL_EFFECT_NULL effects 2009-11-05 20:50:56 -08:00
Chris Robinson aa84b2e328 Handle setting AL_EFFECT_NULL effect types on a slot 2009-11-05 18:22:43 -08:00
Chris Robinson ebc1602816 Use the DXSDK_DIR variable when set 2009-11-02 14:57:48 -08:00
Chris Robinson dcd6a55529 Use a realtime clock for measuring time 2009-11-01 10:03:05 -08:00
Chris Robinson fb258a7416 Properly lock the device when setting it disconnected 2009-11-01 09:29:20 -08:00
Chris Robinson 5aed927887 Watch for stream and context failures from pulse, and report disconnects 2009-10-31 20:05:15 -07:00
Chris Robinson d964bb9f2f Fix length calculation for pulse 2009-10-31 19:30:12 -07:00
Chris Robinson 87dc7a5b97 Set a proper channel map for PulseAudio 2009-10-31 19:29:41 -07:00
Chris Robinson ae63ca481d Mix all updates PulseAudio requests 2009-10-31 17:59:30 -07:00
Chris Robinson 8e599103b6 Watch for buffer attribute changes from PulseAudio 2009-10-31 17:57:35 -07:00
Chris Robinson af0140e260 Use signals to watch for state changes
This partially reverts commit 617b6728d3
2009-10-31 17:52:26 -07:00
Chris Robinson 9d43907710 Properly free the device name 2009-10-31 13:25:26 -07:00
Chris Robinson a3fcd67a82 Remove some unneeded buffer fields, make padding a constant amount 2009-10-25 08:31:28 -07:00
Chris Robinson fe3a43e2d4 Remove the format and frequency from the source, get them manually 2009-10-25 06:26:19 -07:00
Chris Robinson e095047b02 Small fixups 2009-10-25 06:16:21 -07:00
Chris Robinson 31a106344f Calculate more store only once per loop 2009-10-25 05:34:47 -07:00
Chris Robinson b0a82aa420 Store the buffer format and frequency in the source when updated 2009-10-25 05:03:22 -07:00
Chris Robinson c003d6032d Retreive and store source params once per mix update 2009-10-25 04:39:36 -07:00
Chris Robinson 1665595138 Remove unneeded bufferlistitem properties 2009-10-25 02:59:31 -07:00
Chris Robinson bbccf76b45 Dereference left over buffers and slots when sources are released 2009-10-24 07:09:44 -07:00
Chris Robinson 1f47fa9243 Try 32-bit float before 16-bit, and stereo before mono 2009-10-24 03:20:47 -07:00
Chris Robinson 7b24a12a66 Check for the -pthread switch instead of using _REENTRANT 2009-10-23 10:36:32 -07:00
Chris Robinson 921e7ac3e9 Store the buffer handle directly in the source and buffer queue list 2009-10-22 09:31:26 -07:00
Chris Robinson 8d807add41 Store the bufferID in the buffer struct 2009-10-22 08:53:59 -07:00
Chris Robinson 8501fce38b Remove unused source fields 2009-10-21 15:40:09 -07:00
Chris Robinson 55d7984582 Set the calculated source parameters directly 2009-10-21 15:31:21 -07:00
Chris Robinson 8dbac9c0bd Store the calculated source parameters in the source struct 2009-10-21 14:25:54 -07:00
Chris Robinson 47b8510b0c Don't assume 16-bit samples 2009-10-21 13:50:53 -07:00
Chris Robinson 4e399b41cd Calculate source filter coefficients with the source parameters 2009-10-21 13:08:50 -07:00
Chris Robinson bdec1e037f Remove unnecessary variable 2009-10-21 12:48:04 -07:00
Chris Robinson 9924cce466 Make sure the ALSA device list is reset if no cards are found 2009-10-21 11:53:45 -07:00
Chris Robinson d3ecbd75af Clear the echo and delay lines even if the buffer doesn't change size 2009-10-21 02:03:33 -07:00
Chris Robinson 75b65ab2a2 Remove the Frequency field from the context 2009-10-20 13:47:55 -07:00
Chris Robinson eca8825ab4 Explicitly suspend the context being processed for mixing 2009-10-20 13:38:02 -07:00
Chris Robinson 9f9b7cf885 Make sure the context is properly locked while updating slots and sends 2009-10-20 13:02:28 -07:00
Chris Robinson cd695d7f15 Allow creating multiple contexts on a device 2009-10-20 12:58:19 -07:00
Chris Robinson 9c0ae597cd Make sure the effect slots and source sends are properly updated
Slots need to be updated due to possible frequency changes, and source sends
need to be updated in case some active sends were disabled
2009-10-20 12:53:07 -07:00
Chris Robinson 4461b4c56f Be more tolerant with resetting the fragment size 2009-10-20 12:28:51 -07:00
Chris Robinson 93361118b7 Store a device's contexts in a dynamic array 2009-10-20 11:54:04 -07:00
Chris Robinson 50f963f6c6 Don't try to clamp the listener-to-source velocity for local sources
Local sources will always have a 0 VLS value
2009-10-20 11:03:16 -07:00
Chris Robinson 67c3e47ed4 Avoid handling NaN when scaling float samples for output 2009-10-20 09:48:09 -07:00
Chris Robinson f14cf8289e Add a method to update device-dependant effect parameters.
The effect state's update method will be called afterwards
2009-10-20 08:31:44 -07:00
Chris Robinson 790aa68621 Reduce indentation for CalcSourceParams 2009-10-19 13:25:40 -07:00
Chris Robinson 6d1d61026d Be context-agnostic in the effect Create functions
This allows the effect Update functions to handle the playback frequency being
changed. By default the effects assume a maximum frequency of 192khz, however,
it can go higher at the cost of the sample buffers being cleared and the risk
of an abort() if reallocation fails
2009-10-19 07:46:53 -07:00
Chris Robinson a4e3ca933b EINTR is not a fatal write error 2009-10-19 00:33:00 -07:00
Chris Robinson 1fb58f9f8e Fix OSS and Solaris playback loop check 2009-10-17 22:17:59 -07:00
Chris Robinson 86e0b9386d Remove duplicate check in ALSA mmap playback 2009-10-16 10:57:58 -07:00
Chris Robinson c7c4cf221a Make the dry buffer part of the device 2009-10-09 01:06:33 -07:00
Chris Robinson d6e439244a Rename SetALCError for consistency 2009-10-07 04:32:14 -07:00
Chris Robinson 5ff225fa1b Add missing ) in pulseaudio backend 2009-10-02 07:34:49 -07:00
Chris Robinson 2a77e840cc Fix type casts for 7.1 output 2009-09-30 16:02:49 -07:00
Chris Robinson 00272883f6 Fallback to 16-bit or 8-bit, or mono, if ALSA doesn't accept the request 2009-09-27 08:35:10 -07:00
Chris Robinson bd6f3b1273 Report disconnect if the device fails to reset 2009-09-27 04:58:42 -07:00
Chris Robinson 4e9767bb22 Don't store the ALSA format 2009-09-27 04:03:11 -07:00
Chris Robinson 1d54b7a266 Fix capture, too 2009-09-27 03:57:19 -07:00
Chris Robinson c9506d939a Make sure ALSA properly loads, and fix the thread stop condition 2009-09-27 01:12:42 -07:00
Chris Robinson 21217824b7 Avoid duplicate error handling 2009-09-27 00:58:11 -07:00
Chris Robinson 8024df2991 Load backend libs on-demand 2009-09-27 00:21:40 -07:00
Chris Robinson f72b6fe717 Release 1.9.563 2009-09-26 01:42:21 -07:00
Chris Robinson 8d29d25d60 Condense some local variables 2009-09-24 14:51:40 -07:00
Chris Robinson af8e7ab984 Check the source start position in alSourcePlayv instead of the mixer 2009-09-24 12:24:10 -07:00
Chris Robinson 4289d5de7d Parse the attribute list and reset the device before creating the context 2009-09-24 11:31:01 -07:00
Chris Robinson 83734aff0e Slightly improve the multi-channel mixing loops 2009-09-24 08:52:51 -07:00
Chris Robinson 617b6728d3 Remove the state callbacks and watch for changes synchronously 2009-09-24 00:26:41 -07:00
Chris Robinson 7a8fd6cb20 Get rid of unnecessary messages from pulseaudio 2009-09-23 08:34:15 -07:00
Chris Robinson 55da339586 Properly clean up the pulse device 2009-09-23 08:19:00 -07:00
Chris Robinson 774eae0f2f Silence unused parameter warning 2009-09-23 07:10:23 -07:00
Chris Robinson 201b108a54 Attempt to reload backend libs when probing, if they didn't initially load 2009-09-23 01:33:58 -07:00
Chris Robinson bdbdbcea26 Update in properly-sized chunks for PulseAudio 2009-09-23 01:24:10 -07:00
Chris Robinson f6c1a21cf0 Properly flip the backend entries when sorting the device list 2009-09-22 04:42:46 -07:00
Chris Robinson c953072a15 Fix reset condition check 2009-09-22 00:16:21 -07:00
Chris Robinson 93f3e9357b (Re)allocate and clear the stereo-to-binaural filter as needed 2009-09-21 23:52:59 -07:00
Chris Robinson 8ea8b49464 Avoid locking a NULL context when a device is disconnected 2009-09-21 23:31:04 -07:00
Chris Robinson 62cef44dcb Reinitialize the stereo-to-binaural filter after resetting the device 2009-09-21 23:25:26 -07:00
Chris Robinson a2edc95207 Properly set the number of sends requested by the app 2009-09-21 23:18:39 -07:00
Chris Robinson 4cd41e3e08 Don't allow setting an invalid auxiliary send 2009-09-21 23:09:31 -07:00
Chris Robinson b641f993ac Hide the wave writer device if no output file is set 2009-09-21 05:03:40 -07:00
Chris Robinson e2905b6bc9 Properly handle forced 32-bit float output with DSound 2009-09-21 02:57:17 -07:00
Chris Robinson e11b59a80f Return imediately if trying to set an unknown format 2009-09-21 02:53:33 -07:00
Chris Robinson 0ae02e8ad4 Support floating point capture for backends that can handle it 2009-09-21 02:31:33 -07:00
Chris Robinson 2cbdffab86 Create the ALSA capture ring buffer with the proper size 2009-09-21 02:14:32 -07:00
Chris Robinson 86eefdb2a8 Mix all pending periods at once instead of one at a time 2009-09-20 00:44:39 -07:00
Chris Robinson 750399ccd7 Let verify_state restore from an ALSA underrun 2009-09-20 00:40:23 -07:00
Chris Robinson fb97ca6022 Remove left-over backend-specific period options 2009-09-16 23:29:32 -07:00
Chris Robinson cbcaa54173 Store the number of periods in the device instead of the buffer size
Also keep all the fields in sync
2009-09-16 22:58:54 -07:00
Chris Robinson 94e3fca702 Remove unnecessary parameters, and rename methods 2009-09-16 00:24:44 -07:00
Chris Robinson ffff104554 Restore the device frequency using the stream info 2009-09-16 00:08:11 -07:00
Chris Robinson 307a9dd1b2 Update the frequency with what the stream actually got set with 2009-09-15 23:14:14 -07:00
Chris Robinson 5dff8ea820 Properly terminate PortAudio when closing 2009-09-15 22:45:27 -07:00
Chris Robinson 49648fe22d Make a GetConfigValueBool function and use it 2009-09-15 22:18:13 -07:00
Chris Robinson 9f037e8980 Pass the device to aluMixData 2009-09-15 19:30:27 -07:00
Chris Robinson 12f81bcbb9 Move the stereo-to-binaural filter to the device 2009-09-15 19:06:47 -07:00
Chris Robinson 6636131d3b Pass the frame count to aluMixData 2009-09-15 18:19:00 -07:00
Chris Robinson eeea9631ce Split the source mixing loop into a separate function
Helps reduce the function size and indentation levels
2009-09-15 17:56:31 -07:00
Chris Robinson f9d228c30a Add missing context checks 2009-09-12 20:22:03 -07:00
Chris Robinson d516f8c4bd Update comment 2009-09-12 20:02:45 -07:00
Chris Robinson 5d1c3a0c63 Fix wave writer backend 2009-09-12 19:58:36 -07:00
Chris Robinson 396f2d6050 Mix the proper size for every update, and notify disconnects 2009-09-12 19:36:16 -07:00
Chris Robinson e25b01c3a2 Start Solaris playback when a context is ready 2009-09-12 18:19:52 -07:00
Chris Robinson b29a24afb0 Move the default out-of-tree build dir to build 2009-09-12 17:54:27 -07:00
Chris Robinson d6d1dba3d2 Initialize OpenAL with a constructor call instead of first-use 2009-09-12 17:49:08 -07:00
Chris Robinson 8b36a9d121 Get rid of deinit code duplication 2009-09-12 17:29:35 -07:00
Chris Robinson 59ed9338d8 Add support for the in-progress ALC_EXT_thread_local_context extension 2009-09-12 16:45:46 -07:00
Chris Robinson b444dea63b Fix compilation for Windows 2009-08-28 06:33:59 -07:00
Chris Robinson 1816c8bd37 Make a copy of the default device name for the app
The pointer to it could otherwise change if the device list is rebuilt, which
would be bad if the app has it
2009-08-27 20:13:35 -07:00
Chris Robinson c27948a7ee Don't free a NULL handle 2009-08-27 19:01:12 -07:00
Chris Robinson 832dc8a585 Remove unnecessary vars and rework device name checks 2009-08-27 18:55:18 -07:00
Chris Robinson bb121e68a6 Rebuild device lists when retrieving them 2009-08-27 06:09:33 -07:00
Chris Robinson 1ec26fd789 Dynamically allocate device lists 2009-08-27 03:55:36 -07:00
Chris Robinson 4ff4a7e3b9 Close some libs when deinitializing 2009-08-27 03:12:54 -07:00
Chris Robinson 6bb14e45ce Store a copy of the device name in the device 2009-08-27 02:53:09 -07:00
Chris Robinson 45dc804819 Store copies of the device names in the individual backends 2009-08-27 01:47:41 -07:00
Chris Robinson 81db01ebf1 Add a method to deinitialize backends 2009-08-26 23:45:00 -07:00
Chris Robinson 06c576a945 Support disconnect for OSS and Solaris 2009-08-26 21:49:38 -07:00
Chris Robinson 929b025fd6 Support disconnect for ALSA playback and capture
As a consequence for capture, mmap is no longer supported. Later on, supporting
mmap capture to write directly into the ring buffer is a possibility.
2009-08-26 21:27:10 -07:00
Chris Robinson 7976de05a5 Add base support for ALC_EXT_disconnect
Individual backends need to be updated to handle disconnection events
2009-08-26 19:15:17 -07:00
Chris Robinson 70c88879ab Add ALC_EXT_disconnect to alext.h 2009-08-26 18:29:25 -07:00
Chris Robinson 276ee63b99 Define PATH_MAX if it isn't already 2009-08-19 12:59:40 -07:00
Chris Robinson 487f0dde75 Print source and auxiliary slot debug warnings from alcDestroyContext 2009-08-16 16:11:22 -07:00
Chris Robinson c8f700930a More modifications and fixes for context checks 2009-08-16 16:02:13 -07:00
Chris Robinson decbe4df45 Modify some context checks 2009-08-16 15:09:36 -07:00
Chris Robinson 462f27c129 Use a function to retrieve the current context in an already-locked state
This should help prevent race-conditions with a context being destroyed between
breing retrieved and locked
2009-08-16 14:09:23 -07:00
Chris Robinson 2d1191697a Fix the Solaris backend 2009-08-16 13:16:41 -07:00
Chris Robinson 0b5a0d1e79 Reset the target if the databuffer being deleted is currently selected 2009-08-16 11:12:13 -07:00
Chris Robinson 0b7eb9f118 Add in-progress working extension AL_EXTX_sample_buffer_object 2009-08-16 00:54:08 -07:00
Chris Robinson dc26261065 Support 32-bit float output 2009-08-15 13:20:35 -07:00
Chris Robinson 9bea67fb1c Use a function to convert float to unsigned byte 2009-08-15 11:33:38 -07:00
Chris Robinson 8246bc3418 Use a custom function to get the requested format 2009-08-15 10:32:54 -07:00
Chris Robinson 01e32b58da Fix some size checks 2009-08-15 10:01:55 -07:00
Chris Robinson 9d05ff0468 Verify the device pointer is a valid device 2009-08-15 09:55:08 -07:00
Chris Robinson 510ccc7f17 Store the effect and filter lists in the device 2009-08-15 09:39:18 -07:00
Chris Robinson 43067ed2b8 Store the buffer list in the device 2009-08-15 09:14:08 -07:00
Chris Robinson 4e4c9aef15 Calculate the buffer size and frame size when needed 2009-08-14 13:08:25 -07:00
Chris Robinson 4031ee5ef0 Fix speaker override check 2009-08-14 07:27:19 -07:00
Chris Robinson e079291202 Remove unnecessary function parameters 2009-08-13 19:36:14 -07:00
Chris Robinson 826c641668 Hold the lock while finding a capture device to open 2009-08-13 19:19:16 -07:00
Chris Robinson 6ae845fd9e Don't let the device frequency change when PortAudio gets a new context 2009-08-13 17:13:37 -07:00
Chris Robinson a78fd58dca Create PulseAudio's playback stream when the AL context is created 2009-08-13 17:08:47 -07:00
Chris Robinson a8ae610489 Set context frequency only if it successfully starts 2009-08-13 15:35:10 -07:00
Chris Robinson b86a1e5797 Add a note about app and system defaults 2009-08-13 13:48:06 -07:00
Chris Robinson 45552a68ce Hold the lock while initializing context playback 2009-08-13 13:32:42 -07:00
Chris Robinson d46655fbb2 Support the ALC_FREQUENCY context attribute 2009-08-13 13:28:35 -07:00
Chris Robinson 87fee8c504 Start DSound playback at context creation 2009-08-13 13:11:05 -07:00
Chris Robinson 9ff0617807 Don't clamp the wave writer's update size 2009-08-13 12:35:42 -07:00
Chris Robinson 243939f94f Allow delaying playback start until context creation, and don't use UpdateSize to store the buffer size
This will make it possible to support the context attributes (frequency,
refresh, etc) for some backends
2009-08-13 12:28:46 -07:00
Chris Robinson 739385bd89 Add AL_EXT_BUFFER_DATA_STATIC to alext.h 2009-08-13 10:10:34 -07:00
Chris Robinson 32060f53cd Reset the buffer played count when the queue is removed via the AL_BUFFER property 2009-08-11 14:16:30 -07:00
Chris Robinson 4ee3972f15 Update queued buffer states when source looping is toggled 2009-08-11 07:57:18 -07:00
Chris Robinson 00a9cf4a85 Remove more unneeded variables 2009-07-07 21:11:13 -07:00
Chris Robinson 1dcd538ad9 Remove unneeded variables 2009-07-07 21:01:41 -07:00
Chris Robinson a8973514ba Don't load functions if libpulse failed to open 2009-07-07 12:51:18 -07:00
Chris Robinson 0ac9e57d28 Store the requested mono/stereo source count and sends in the device
Although the properties are set as context attributes, they are queried from
the device. If multiple contexts per device are ever supported, it would not
be straight forward about where to read the values from. This way, the
attributes can be treated as device-specific attributes that are updated when
a new context is created.
2009-07-06 03:09:01 -07:00
Chris Robinson 5460e85c40 Set the right access mode for non-mmap capture 2009-07-02 20:50:12 -07:00
Chris Robinson 432f0ba34c Specify the right format for ALSA recording 2009-07-02 20:32:19 -07:00
Chris Robinson 7e01e53445 Fix parsing of layout configuration string 2009-07-02 20:31:36 -07:00
Chris Robinson 043fe6edd2 Release 1.8.466 2009-06-07 22:30:08 -07:00
Chris Robinson ef68e5d0ff Protect ring buffer access with the lock 2009-06-07 21:34:05 -07:00
Chris Robinson c6f3a4e062 Move a couple inline functions into the header 2009-06-07 20:16:41 -07:00
Chris Robinson 87ed53c7f6 Don't read deprecated config files 2009-06-07 20:12:01 -07:00
Alam Arias 89e9437b84 add check for _WAVEFORMATEXTENSIBLE_ and define _WIN32_IE before inlcuding any header 2009-06-07 19:43:42 -07:00
Chris Robinson 1454c46b5f Use a thread-safe static inline function for printing 2009-06-07 15:42:15 -07:00
Chris Robinson 0fac1e9115 Store the max number of auxiliary slots in the device 2009-06-07 14:53:22 -07:00
Chris Robinson f3a3358e63 Fix for MSVC
MSVC won't accept zero-sized arrays at the end of structs, if that struct is
used in another struct that's not also at the end. This wastes a float for each
FILTER object
2009-06-06 23:33:53 -07:00
Chris Robinson 3c122b0bef Set an error if effect creation fails 2009-05-31 11:54:49 -07:00
Chris Robinson 7940003d06 Set an error if NaN is given for panning values 2009-05-31 11:34:07 -07:00
Chris Robinson 45b16ff456 Use the generic FILTER object for the master effect filter
Also fix a couple comments
2009-05-30 12:54:53 -07:00
Chris Robinson 9341f4f884 Fixup panning gain calculations
Clamp the panning vector magnitude to 1, and use an energy-reduction method as
the vector magnitude increases (to simulate reverb area occlusion)
2009-05-30 00:32:17 -07:00
Chris Robinson 2c20f26784 Apply slot gain on slot output, not input 2009-05-29 16:51:00 -07:00
Chris Robinson 2a21a449b4 Slot gain changes should have an immediate effect 2009-05-29 13:50:58 -07:00
Chris Robinson 74dc7090fd Don't expose effect-specific structures 2009-05-29 13:30:50 -07:00
Chris Robinson e12ac95d69 Use the MIN/MAX/DEFAULT macros for checking and setting the effect parameters 2009-05-29 12:40:08 -07:00
Chris Robinson 34ea7eba9e Add preliminary support for the EAX Reverb effect
Not all parameters are supported yet, though it is a little more fuctional than
standard reverb
2009-05-29 01:32:54 -07:00
Chris Robinson 6d3ba44f55 Don't update the devicevalues before returning success 2009-05-27 07:02:33 -07:00
Chris Robinson 5a8ad67660 Lock the context before getting the mmap region to update 2009-05-26 14:54:59 -07:00
Chris Robinson f1e442408b Set the correct data size 2009-05-26 14:51:57 -07:00
Chris Robinson a946af4463 Simplify ALSA PCM parameter retrieval 2009-05-26 10:08:27 -07:00
Chris Robinson e01cd92d56 Set avail_min so snd_pcm_wait will wait for the full update to be ready 2009-05-26 10:03:17 -07:00
Chris Robinson 923af8c3a6 Don't update the source's gains in the sample mixing loop
Update copies stored on the stack instead, then update the source after mixing
2009-05-26 09:23:49 -07:00
Chris Robinson ac04cf57fa Make sure _CRT_SECURE_NO_WARNINGS is always defined for MSVC 2009-05-26 06:02:40 -07:00
Chris Robinson 1f6c4e01db Properly fall back to non-mmap capture if the parameters fail 2009-05-25 16:06:03 -07:00
Chris Robinson 04135723a2 Get rid of the ugly ok() macro 2009-05-25 13:07:49 -07:00
Chris Robinson 93fbdb1e6c Redo OSS mixing loop 2009-05-25 12:12:37 -07:00
Chris Robinson c1de15f840 Update in descrete amounts using the update size 2009-05-25 12:01:14 -07:00
Chris Robinson 3be399c555 Do not open pulseaudio if it didn't load 2009-05-19 06:37:22 -07:00
Chris Robinson 1f5453075c Dynamically load PulseAudio 2009-05-19 06:35:12 -07:00
Chris Robinson 33a74f1594 Clear the filter history directly 2009-05-17 22:50:07 -07:00
Chris Robinson e4da1224cb Optionally install the sample configuration file 2009-05-17 22:36:45 -07:00
Chris Robinson 5c2a47620c Reformat the sample config file 2009-05-17 22:24:29 -07:00
Chris Robinson 1bfa7609d2 Make the filter functions global inline, and use it for echo 2009-05-16 23:34:26 -07:00
Chris Robinson 7e8501123a Set some common macros in alu.h 2009-05-16 23:26:39 -07:00
Chris Robinson c447eeb2c7 Use the source reference distance to specify full panning magnitude
Sources that are closer than the specified reference distance will not pan to
full magnitude, thus providing a smoother transition as it moves around near
the listener
2009-05-16 03:54:16 -07:00
Chris Robinson ad19cea6e0 Make sure DSSPEAKER_5POINT1 is defined 2009-05-14 05:24:18 -07:00
Chris Robinson 0ce29863e3 Prevent spawning PulseAudio if it's not already running 2009-05-06 00:19:35 -07:00
Chris Robinson 7a7a4844f4 Make the filter history buffer size flexible
This lets the filter history buffer be as big as needed for a given use, so
that it can have a size large enough for the more demanding cases, but not be
wasteful for lesser-demanding cases, while not incuring the overhead of an
added pointer indirection
2009-04-16 05:43:09 -07:00
Chris Robinson a2adbb1ab5 Add a PulseAudio backend 2009-04-16 05:17:42 -07:00
Chris Robinson a97fc792d8 Increase the default maximum number of sends to 2 2009-04-14 22:18:47 -07:00
Chris Robinson e9a6a1d6f5 Use a 2-pole filter for the wet path low-pass filter
This should help keep CPU use from increasing a lot when the number of sends
increases. Also changes the function names to reflect the difference
2009-04-14 22:04:18 -07:00
Chris Robinson f245f0ef87 Make the number of source sends variable
The highest value is clamped to MAX_SENDS
2009-04-13 20:33:41 -07:00
Chris Robinson a30f431b63 Reduce the number of send loops 2009-04-13 02:50:40 -07:00
Chris Robinson 58f48f20e2 Rename Standard Reverb to Reverb 2009-04-13 02:01:21 -07:00
Chris Robinson ee1b02b709 Fix comment regarding the 3rd echo tap 2009-04-12 18:54:26 -07:00
Chris Robinson 55c790c9ff Add the Echo effect 2009-04-12 16:01:10 -07:00
Chris Robinson 65b69f3308 Fix air absorption 2009-04-11 20:27:55 -07:00
Chris Robinson c67fbd72dd Pay attention to the MAX_SENDS value 2009-04-11 20:04:46 -07:00
Chris Robinson 30f57d0824 Make auxiliary effect slot count configurable 2009-04-11 18:16:05 -07:00
Chris Robinson 1c54018111 Move the WetBuffer into the effect slot object
This should make it easier to support multiple slots
2009-04-11 17:04:55 -07:00
Chris Robinson e80e1b6d9f Use LIB_SUFFIX when creating the default lib install dir
Thanks to Hanno Böck for pointing this out
2009-04-07 12:11:41 -07:00
Chris Robinson 437f00d6de Add missing comment markers 2009-04-07 11:41:41 -07:00
Chris Robinson fc6d6b1601 Use the square of the values when calculating the density gain 2009-03-21 10:42:42 -07:00
Chris Robinson 9306e36d52 Make sure M_PI is defined for reverb 2009-03-15 23:14:40 -07:00
Chris Robinson 61122a5093 Release 1.7.411 2009-03-14 22:24:59 -07:00
Chris Robinson a8a4ff8af1 Constify some parameters 2009-03-13 23:08:15 -07:00
Chris Robinson 8a857c35ee Calculate filter coefficients in aluMixData 2009-03-13 22:58:54 -07:00
Chris Robinson 89ddd7d8e5 Don't modify the device struct until playback succesfully starts 2009-03-13 22:16:01 -07:00
Chris Robinson 8d0c4ccb3b The room rolloff factor can go up to 10 2009-03-13 02:12:45 -07:00
Chris Robinson 8ee47d5573 Dynamically load dsound when possible 2009-03-10 02:46:42 -07:00
Chris Robinson 9e88011417 Add a cast for setting a dummy pointer value 2009-03-10 02:20:05 -07:00
Chris Robinson f8949ee7a1 Always add the default ALSA device even if no cards are present 2009-03-10 01:46:51 -07:00
Chris Robinson 62aa2d0ba7 Only expose the default ALSA device in standard enumeration 2009-03-10 01:28:01 -07:00
Chris Robinson ace047d625 Append the card and device indices to the ALSA device strings 2009-03-10 01:21:42 -07:00
Chris Robinson d6277db209 Fix alsoftrc configuration sample comments 2009-03-10 01:08:03 -07:00
Chris Robinson 7ebb28327f Remove the SDL backend 2009-03-10 01:03:39 -07:00
Chris Robinson 86931cbde4 Add a PortAudio backend 2009-03-10 00:55:29 -07:00
Chris Robinson 6d7be151dc Remove the unneeded path from the cross-compiler commands 2009-03-06 20:09:44 -08:00
Chris Robinson 07227b9806 Use a modified reverb model that obeys the reverb parameters better 2009-03-02 18:48:23 -08:00
Chris Robinson 8348d719cd Clamp gain of multichannel sources 2009-02-10 15:15:49 -08:00
Chris Robinson 0fcefd865b Install the pkgconfig file to the correct directory 2009-02-10 14:39:48 -08:00
Chris Robinson 57c2e9b5f8 Include assert.h for assert() 2009-02-02 11:18:33 -08:00
Chris Robinson 7d7fc39035 Use calculated distance from reference for air absorption 2009-01-30 10:56:25 -08:00
Chris Robinson 351105b3df Fix incorrect comment 2009-01-30 10:53:09 -08:00
Chris Robinson cbfc33215b Use M_PI since it is sure to be defined 2009-01-27 07:11:58 -08:00
Chris Robinson 5a93b56673 Fix ignored return value warning 2009-01-26 08:10:05 -08:00
Chris Robinson 1f4c69c17a Use a matrix for up- and down-mixing channels 2009-01-25 22:11:07 -08:00
Chris Robinson f5b19fad20 Duplicate stereo onto the side channels as well as the back 2009-01-25 19:54:50 -08:00
Chris Robinson bc60818e9a Don't read stereo layout when output is mono 2009-01-25 19:33:52 -08:00
Chris Robinson f82c88f016 Add options for user-configurable speaker arrangements 2009-01-25 19:20:47 -08:00
Chris Robinson 98e86decad 6.1 uses front- and back-center, not left- and right-back channels 2009-01-24 15:13:14 -08:00
Chris Robinson aaf2c0ebd4 Seperate mixing loops depending on source channel configuration 2009-01-24 13:57:01 -08:00
Chris Robinson 778b74cae1 Reimplement panning using lookup tables, based on a patch by Christian Borss
This allows speaker positions to be specified by discrete angles around the
listener, providing more flexibility and configurability in placement.
Additional patches to take advantage of this are forthcoming.
2009-01-24 10:38:04 -08:00
Chris Robinson 43ee1edd97 Properly capitalize the libname in Win32 2009-01-10 21:23:53 -08:00
Chris Robinson dd7e23740b Remove XCOMPILEWIN32 option in favor of CMake 2.6's cross-compiling caps 2008-12-12 11:19:38 -08:00
Chris Robinson 1acd6da745 Search for SDL.h in addition to SDL/SDL.h 2008-12-12 11:09:23 -08:00
Chris Robinson 3056f91ec5 Apply the dry filter to multi-channel sources
Unlike mono sources, they use 2 chained one-pole filters instead of 4
2008-12-10 11:54:13 -08:00
Chris Robinson ed03570e1a Fix some CMake checks 2008-12-07 02:20:17 -08:00
Chris Robinson 2ec0e48d06 Protect playback device opening with a mutex lock 2008-12-07 01:07:54 -08:00
Chris Robinson 8b54d59b8c Add an SDL backend 2008-12-07 01:05:39 -08:00
Chris Robinson f6a4dbabdd Add cast for setting a dummy pointer value 2008-12-06 12:17:29 -08:00
Chris Robinson a71c291bcb Let CMAKE_DEBUG_POSTFIX be exposed for graphical CMake front-ends 2008-12-02 02:37:51 -08:00
Chris Robinson 55b9ccc2de Implement AL_EXTX_source_distance_model
As with other EXTX extensions, this is subject to change and removal as the
spec gets worked on
2008-11-25 18:56:10 -08:00
Chris Robinson 67d67a3bf6 Release 1.6.372 2008-11-21 09:16:29 -08:00
Chris Robinson 591de1ecb4 Wait until one full fragment is empty before mixing 2008-11-21 07:52:13 -08:00
Chris Robinson 3dad17c263 Make the DSound emulated fragment count configurable 2008-11-21 07:48:03 -08:00
Chris Robinson eefc18170d Fix a comment 2008-11-20 10:09:37 -08:00
Chris Robinson 5f84c5339d Fix early reflection input 2008-11-19 09:14:40 -08:00
Chris Robinson 2d461379ef Handle ALSA capture errors a bit better 2008-11-19 09:01:03 -08:00
Chris Robinson c8123756ff Simplify in-sample low-pass filter coefficient calculation 2008-11-18 06:35:00 -08:00
Chris Robinson 76c7789ee7 Fix low-pass coefficient calculation 2008-11-18 04:31:24 -08:00
Chris Robinson 13a2e6ef1f Don't calculate reverb HF limit if air absorption is 1 2008-11-18 03:26:02 -08:00
Chris Robinson 0ecb34b850 Non-cross-compiled DLLs shouldn't have lib prefixed 2008-11-18 03:24:49 -08:00
Chris Robinson 1f86c48d95 Remove outdated comments and add copyright header 2008-11-17 09:32:25 -08:00
Chris Robinson 7e1295df9a DirectSound is not explicitly dependant on windows.h 2008-11-16 04:39:12 -08:00
Chris Robinson 482b160c8a Remove unneeded macro 2008-11-16 01:07:39 -08:00
Chris Robinson 181eb95b13 Use a better dB-to-linear gain convertion 2008-11-16 00:57:35 -08:00
Chris Robinson c0ccd31a3e Implement a new reverb effect
Code created and graciously provided by Christopher Fitzgerald
2008-11-16 00:29:49 -08:00
Chris Robinson d72b132c57 Add an option to disable specific EFX effect types 2008-11-14 07:13:59 -08:00
Chris Robinson 506912aed7 Add cross-compiling option 2008-11-13 23:13:15 -08:00
Chris Robinson 670d70d3c9 Allow specifying another config file with the ALSOFT_CONF env var 2008-11-13 07:58:39 -08:00
Chris Robinson 010f7d12f4 Don't ramp gains when starting a sound from the beginning 2008-11-13 05:48:38 -08:00
Chris Robinson da684564ea Avoid unnecessary floating-point math 2008-11-11 17:59:55 -08:00
Chris Robinson fc4c867f27 Add initial AL_EXTX_buffer_sub_data support
Note that this is an in-development extension, as noted by the EXTX moniker
instead of EXT. It's behavior is subject to change, and the extension string
will be removed (replaced with the official string once it's finalized).
Developers are discouraged from using this in production code, though feel
free to play around with it.
2008-11-11 05:57:32 -08:00
Chris Robinson 9ba30c4e20 Fix Win32 thread handle leak 2008-11-05 19:42:56 -08:00
Chris Robinson 15334e56cd Be more flexible with channel count when loading IMA4 data 2008-11-02 15:30:56 -08:00
Chris Robinson ffa42ff22c Seperate data converters into reusable functions 2008-11-02 01:24:57 -07:00
Chris Robinson a7c62dbabc More padding fixes 2008-11-01 16:37:32 -07:00
Chris Robinson eda1e41152 More buffer conversion refactoring 2008-11-01 16:35:16 -07:00
Chris Robinson af5a5b76ec Padding is not dependant on the frequency cutoff anymore 2008-10-31 23:32:28 -07:00
Chris Robinson 87f3a0dc16 Restructure buffer data conversion code a bit 2008-10-31 22:13:35 -07:00
Chris Robinson 2c80a80704 Fix typo preventing capture from opening 2008-10-27 23:37:56 -07:00
Chris Robinson 301a4c4a95 Append the driver and its version to the AL version string 2008-10-25 15:48:17 -07:00
Chris Robinson 8fc4a3b724 Make sure an appropriate error is set when opening a device fails 2008-10-24 19:58:49 -07:00
Chris Robinson cb6f040005 Use plughw for capture so ALSA can convert capture data 2008-10-14 09:50:37 -07:00
Chris Robinson b91c2e4a99 Include float.h if it exists, for _RC_CHOP and _MCW_RC 2008-10-14 09:47:32 -07:00
Chris Robinson 59a71b1454 Remove another unused source member 2008-10-10 01:31:31 -07:00
Chris Robinson 36f133a5ae Use a modulo to keep the buffer position in range for looping sources
A high pitch and low buffer size can cause a lot of unnecessary iterations
otherwise, that just decrement the position
2008-10-10 01:13:32 -07:00
Chris Robinson 74a58c0d09 Clamp source position to the buffer size when it stops 2008-10-09 23:54:31 -07:00
Chris Robinson bfa1107781 Remove unneeded source member variable 2008-10-09 23:44:48 -07:00
Chris Robinson 11397f7667 Commit missing changes 2008-10-09 20:58:12 -07:00
Chris Robinson 6e9e8239ef Only send one channel through the wet path 2008-10-09 04:02:34 -07:00
Chris Robinson af9932d28b Increase max pitch to 65536
This should be safe now
2008-10-09 02:50:00 -07:00
Chris Robinson 87ff8a65e9 Simplify the lerp function 2008-10-09 02:32:47 -07:00
Chris Robinson 7b6f207790 Don't apply the wet path for multi-channel buffers 2008-10-09 02:28:52 -07:00
Chris Robinson 8672008e43 Skip mixing if the read position is beyond the end of the buffer 2008-10-09 01:17:39 -07:00
Chris Robinson c8cd193346 The wet path should be silent if no effect is set on the slot 2008-10-09 01:07:02 -07:00
Chris Robinson be292e5f0b Don't hold the whole-number position in the fractional value
This will help prevent overflows when the max pitch is increased
2008-10-02 23:53:46 -07:00
Chris Robinson 3863dcc9cb Use a new low-pass filter, based on the I3DL2 spec
Many thanks to Christopher Fitzgerald, for helping with it
2008-10-02 22:20:42 -07:00
Chris Robinson a2568409fc Implement non-mmap ALSA capture 2008-09-29 17:24:50 -07:00
Chris Robinson 6567cdd7b5 Air absorption factor is applied to the dB value, not linear gain 2008-09-22 17:01:47 -07:00
Chris Robinson 5bbf55a401 Add a variable to override the default library type with 2008-09-16 07:43:38 -07:00
Chris Robinson 4a530e2146 Fixup some source parameter calculations 2008-09-16 07:36:48 -07:00
Chris Robinson 27ba8f7b60 Fix function pointer declarations 2008-09-16 06:19:27 -07:00
Chris Robinson 6bfdb57a5b Use a 12dB/oct rolloff instead of 24 for the lowpass filter 2008-09-13 02:46:14 -07:00
Chris Robinson 26e8ea60a5 Store pi as a static const 2008-09-13 00:44:48 -07:00
Chris Robinson f4ea188ffa Fix typo to get the proper minor ALC version 2008-09-12 18:07:12 -07:00
Chris Robinson 1266580420 Print EFX info when the extension is available 2008-09-12 18:05:23 -07:00
Chris Robinson 16d96eed7b Add a Solaris playback backend 2008-09-07 14:34:14 -07:00
Chris Robinson fa76168683 Clear the end of the buffer when at the end of the queue and not looping 2008-09-06 14:08:53 -07:00
Chris Robinson 5f3329b2c9 Don't export extension function symbols from the lib 2008-09-06 13:45:27 -07:00
Chris Robinson db541f3cfa Remove unneeded source struct member 2008-08-15 17:43:07 -07:00
Chris Robinson 3e19ba6ca8 Clear channel volumes when starting a source 2008-08-15 16:33:47 -07:00
Chris Robinson ac8c082b89 Overwrite the input wet sample with the output 2008-08-14 20:44:55 -07:00
Chris Robinson 084df2a229 Allow setting the EFX doppler factor source property 2008-08-14 16:14:16 -07:00
Chris Robinson 22557070ec Ramp channel gains to remove pops and clicks from abrupt changes
Thanks to Christopher Fitzgerald for helping me work on it
2008-08-14 05:43:52 -07:00
Chris Robinson f8ef66954c Include fenv.h if it exists for fesetround 2008-08-08 08:12:41 -07:00
Chris Robinson ef59901e7c Set FPU mode to round toward zero for mixing 2008-08-08 07:32:21 -07:00
Chris Robinson cfe620ccb5 Remove unnecessary casting 2008-08-08 00:21:25 -07:00
Chris Robinson 453b015225 Prevent a 0 or negative increment for the buffer position
Thanks to Christopher Fitzgerald for pointing these last two problems out
2008-08-05 20:51:30 -07:00
Chris Robinson c1cf9ae8f6 Pass a dummy variable to CreateThread to satisfy Win9x 2008-08-05 20:19:13 -07:00
Chris Robinson f1414c650e Release 1.5.304 2008-07-27 19:58:32 -07:00
Chris Robinson 869b041f2f Reduce the default buffer size to 4096
Should help with latency issues some people have and not put too much extra
burden on the mixer, hopefully
2008-07-26 21:07:08 -07:00
Chris Robinson 8621fbb015 Improve getting and setting EFX filter parameters 2008-07-26 18:32:45 -07:00
Chris Robinson 597e01153e Use arrays instead of pointer-to-arrays for the low-pass filter 2008-07-26 17:13:50 -07:00
Chris Robinson d3e5fcd13e Fix some calculations for the reverb buffer 2008-07-26 01:57:04 -07:00
Chris Robinson 3e0f9cc716 Make the filter processing function inline 2008-07-26 00:58:54 -07:00
Chris Robinson c7e49c9f57 Implement yet another low-pass filter
This one using the Butterworth IIR filter design
2008-07-25 19:31:12 -07:00
Chris Robinson e2ed8ff2bf Use a temp pointer when realloc()ing
So the original data isn't lost on out-of-memory conditions
2008-07-24 13:44:26 -07:00
Chris Robinson 559c786d0c Specify padding per buffer, and make sure it's large enough for the filter step 2008-07-24 00:41:25 -07:00
Chris Robinson c3a7480961 Don't advertise extra samples for mixing 2008-07-23 23:27:38 -07:00
Chris Robinson a75e75aef5 Implement an alternative low-pass filter
This method samples from the buffer so that it gets a time-correct 5khz stream,
which is subtracted from the original sample and has the high-frequency gain
applied, then added back.
A better method may be to average all the samples from the current one to the
one freq/5000 away, instead of bilinear filtering the two nearest freq/5000
apart. Processing cost will need to determine its viability
2008-07-23 22:29:53 -07:00
Chris Robinson d45b1207d0 Make sure the correct libname is used for pkg-config 2008-07-23 11:58:36 -07:00
Chris Robinson 5a9e24dea1 Add a warning when building on Windows with the DSound backend disabled 2008-07-23 00:51:55 -07:00
Chris Robinson dc8132e965 Add a pkg-config file to install on the system 2008-07-22 18:42:53 -07:00
Chris Robinson 20bcdc1ef6 Add AL_LOKI_IMA_ADPCM_format and AL_EXT_vorbis tokens to alext.h 2008-07-22 14:45:31 -07:00
Chris Robinson 93c0f61050 Clarify implicit destruction warnings 2008-07-22 14:29:27 -07:00
Chris Robinson 3a8c543f8b Move ALC_ENUMERATE_ALL_EXT tokens to alext.h 2008-07-22 13:50:08 -07:00
Chris Robinson 9e96156451 Add AL_LOKI_WAVE_format tokens to alext.h 2008-07-22 13:45:26 -07:00
Chris Robinson 19aabd6d02 Add ALC_LOKI_audio_channel tokens to alext.h 2008-07-22 13:16:58 -07:00
Chris Robinson 87b5d7bc7b Store extension list with a pointer, not a per-context array 2008-07-22 12:39:10 -07:00
Chris Robinson 646714b585 Set the new linking policy for CMake 2.6 to avoid warnings 2008-07-22 11:26:56 -07:00
Chris Robinson eba60c30c5 Don't force initialization when shutting down
Thanks to Michael Simms
2008-07-17 18:45:21 -07:00
Chris Robinson e66bb09156 Move (de)initialization into ALc.c and remove unneeded file 2008-07-17 18:38:07 -07:00
Chris Robinson 0042b1f80d Implement doppler factor source property 2008-07-15 02:33:05 -07:00
Chris Robinson 6d416ee734 Add the reverb room rolloff to the source room rolloff, not override 2008-07-15 02:23:53 -07:00
Chris Robinson 97d3a45aff Don't check the number of objects being deleted with the number currently allocated
Since apps can validly delete buffer 0, and delete the same source/buffer multiple times in a single call
2008-07-11 09:12:11 -07:00
Chris Robinson 9e9ff44b17 Use volatile for member variables that are changed in a different thread 2008-07-10 20:36:28 -07:00
Chris Robinson f369be148f Reduce the mix buffer sizes by half
Nearly 3MB is a bit much. Could reduce it further, but this is good enough for now.
2008-07-08 19:37:14 -07:00
Chris Robinson 3a09e446b3 Leave SourceToListener untransformed for use with untransformed velocities
Distance is also left untransformed so cone calculations with SoundToListener
are correct
2008-07-03 03:13:43 -07:00
Chris Robinson 2d0b12e49e Store thread return value in the struct to avoid void*-to-uint casting 2008-06-17 19:46:37 -07:00
Chris Robinson 55e2f294ac Allow for overriding the default lib destination with LIB_INSTALL_DIR
So 64-bit builds can be optionally redirected to lib64
2008-06-06 18:39:18 -07:00
Chris Robinson 2a4b467f08 Make the project explicitly C
So it doesn't search for an unneeded C++ compiler
2008-06-06 18:33:13 -07:00
Chris Robinson 2ae9da696a Release 1.4.272 2008-06-04 20:47:20 -07:00
Chris Robinson 376997c3ef Fix config value check 2008-06-04 20:44:51 -07:00
Chris Robinson e963782b8c Release 1.4.270 2008-06-04 19:46:31 -07:00
Chris Robinson 86c01aa62c Install openal-info if it is built 2008-06-04 19:40:52 -07:00
Chris Robinson 15783d25e7 Don't override the format config option in DSound 2008-06-04 18:33:02 -07:00
Chris Robinson 3243f69f21 Use %AppData%\alsoft.ini for the config file in Windows 2008-06-04 18:09:21 -07:00
Chris Robinson 10a87f510c Protect RingBufferSize calculation with the mutex 2008-06-04 17:01:44 -07:00
Chris Robinson ca6feeda29 Make sure the lib is initialized when shutting down
Pretty ugly, but the destructor sequence relies in the mutex being initialized
2008-05-18 20:17:31 -07:00
Chris Robinson cecf778de3 Add addiitonal copyright line 2008-05-18 18:44:17 -07:00
Chris Robinson fe79ab351a Add a simple example that prints out some OpenAL info 2008-05-18 18:40:53 -07:00
Chris Robinson dc0a3a6653 Remove -fno-strict-aliasing as the code should be safe, now 2008-05-18 17:46:45 -07:00
Chris Robinson fed346c285 Fix source calculations for AL_SOURCE_RELATIVE mode
Make sure the source position and direction are properly put into listener-
space before working with them, and don't calculate the listener velocity for
relative coordinates
2008-05-18 16:52:38 -07:00
Chris Robinson cad9b367a5 Use pthread_mutexattr_setkind_np as a fallback to set a recursive mutex type
Some systems (FreeBSD) don't like setting it through pthread_mutexattr_settype
2008-05-15 21:35:51 -07:00
Chris Robinson 6e86146a25 Prepare the ALSA PCM handle before starting capture
Thanks to Jason Daly for pointing it out
2008-05-06 16:05:36 -07:00
Chris Robinson 49d9695ad9 Check the right struct member for the filter type 2008-04-12 07:25:18 -07:00
Chris Robinson e15bc6b9ba Fill the correct capture device list 2008-04-05 20:33:19 -07:00
Chris Robinson 28093a6dcb constify the pointer that holds the filename 2008-03-22 19:05:00 -07:00
Chris Robinson 2af5498804 Define _WIN32_WINNT to 0x0500 when including windows.h
VC7 appears to require that value, or higher, set and fails otherwise
2008-03-01 01:39:42 -08:00
Chris Robinson 2b42d7fdb8 Don't start the DSound playback thread is startup failed 2008-03-01 00:57:37 -08:00
Chris Robinson 9b3c4faa34 Release 1.3.253 2008-02-21 01:28:18 -08:00
Chris Robinson 847b3e8b61 Preserve data and position when reallocating the reverb effect
Still not perfect, but better for when the size doesn't change
2008-02-18 21:58:58 -08:00
Chris Robinson b1bdd7c89f Add the first card after making sure there's a useable device on it 2008-02-15 21:48:00 -08:00
Chris Robinson ffe7a11866 Avoid a static variable for enumerating 2008-02-15 21:09:19 -08:00
Chris Robinson 8c3188bc7d Remove FrameSize struct member 2008-02-14 22:05:56 -08:00
Chris Robinson 880196e5a8 Remove unneeded headers 2008-02-13 19:42:47 -08:00
Chris Robinson d523b440e3 Wait for a PCM handle to be ready for more data instead of polling every 1ms or so 2008-02-13 19:34:47 -08:00
Chris Robinson f2875d5d13 Read in chunks of the fragment size, not expected buffer size 2008-02-12 21:19:09 -08:00
Chris Robinson b0720333cc UpdateSize is not used for capture devices 2008-02-12 19:50:49 -08:00
Chris Robinson c3446a10fa Rename UpdateFreq device field to UpdateSize 2008-02-12 19:38:27 -08:00
Chris Robinson fbc869eed7 Pretend DSound buffer fragment size is 1/4th the total buffer size 2008-02-12 19:24:08 -08:00
Chris Robinson e3e4dbbccf Make the backend list static 2008-02-11 17:24:04 -08:00
Chris Robinson 7fda645b16 Call InitAL at the start of some more ALC functions 2008-02-11 17:23:31 -08:00
Chris Robinson ec7f20644d Fast float-to-int function is no longer needed 2008-02-08 21:03:48 -08:00
Chris Robinson 3d5fa91703 Remove unnecessary casting 2008-02-08 21:01:05 -08:00
Chris Robinson e8acfb069c Remove explicit dependancy on ole32 and unused dxguid 2008-02-08 20:56:45 -08:00
Chris Robinson 4c38ed272d Enumerate DirectSound devices 2008-02-08 20:46:34 -08:00
Chris Robinson acbb4f8e6c Include alext.h instead of redefining some enums 2008-02-08 19:55:51 -08:00
Chris Robinson 13a3fe8a7a Remove unneeded device struct member 2008-02-08 19:41:47 -08:00
Chris Robinson 3dd3cd4ceb Prevent overflow of the device lists 2008-02-08 15:33:26 -08:00
Chris Robinson 80fcbc8adc Use all capture devices listed by ALSA, not just the first on each card 2008-02-08 15:22:17 -08:00
Chris Robinson 33680accea Don't remove the devices from the device list on unexpected shutdown
The close functions will remove it for us
2008-02-08 14:44:05 -08:00
Chris Robinson 081e593e2b Set the done flag immediately when entering the initialization
To prevent two threads from initializing at the same time (not that it's
likely to happen at this point).
2008-02-06 22:22:29 -08:00
Chris Robinson b4ffdfab81 Add an option for duplicating stereo sources on the back speakers 2008-02-06 22:18:50 -08:00
Chris Robinson 1ea4aac799 Add an alext.h header 2008-02-06 20:53:00 -08:00
Chris Robinson 4d6c292757 Fix race condition when starting winmm message poll thread 2008-02-03 00:28:00 -08:00
Chris Robinson 655a83c5c5 Use WAVEFORMATEXTENSIBLE for multichannel dsound output, and don't create a primary buffer 2008-02-01 08:17:57 -08:00
Chris Robinson e8b576bc25 Use the correct channel ordering for Windows 2008-01-27 07:04:13 -08:00
Chris Robinson 01404ed7af Fix output channel order for 6.1 and 7.1 2008-01-27 06:49:48 -08:00
Chris Robinson ad6237f6f7 Fix availibility amount calculation 2008-01-26 21:10:55 -08:00
Chris Robinson d06c64e1c4 aluBytesFromFormat returns bytes, not bits 2008-01-26 21:09:08 -08:00
Chris Robinson b4ad3186dd Update frame size after detecting the directsound output format 2008-01-25 17:05:14 -08:00
Chris Robinson 5f1c0450b3 Remove unnecessary Channels field 2008-01-25 16:38:37 -08:00
Chris Robinson 04b62f77e3 Set the output format according to the speaker setup reported by directsound 2008-01-25 05:02:00 -08:00
Chris Robinson 029dff3f40 Use both write pointers from the directsound buffer lock 2008-01-25 04:42:44 -08:00
Chris Robinson 342f639771 Release 1.2.218 2008-01-21 18:43:14 -08:00
Chris Robinson b5a2baafe5 Remove effect slot thunk entry when deallocated forcefully 2008-01-21 15:34:38 -08:00
Chris Robinson 781e4e5be4 Remove an unneceesary pointer check and decrease indentation 2008-01-21 14:54:15 -08:00
Chris Robinson f3dddb5e99 Remove unnecessary duplicate thunk lookups 2008-01-21 14:33:42 -08:00
Chris Robinson 9a5e892cad Small formatting updates 2008-01-20 22:16:28 -08:00
Chris Robinson 0317362662 Remove duplicate function 2008-01-20 19:22:39 -08:00
Chris Robinson 6b403a76e8 Don't access ALSource for every sample mix 2008-01-20 19:20:24 -08:00
Chris Robinson 6bf6c6fb7b More overflow protection 2008-01-20 13:27:54 -08:00
Chris Robinson 29618bee10 Prevent float samples from overflowing when converting to 16-bit 2008-01-20 00:43:02 -08:00
Chris Robinson db80f682c9 Clean a couple debug messages 2008-01-19 21:41:09 -08:00
Chris Robinson e1cdbac5a0 Close ALC first when exiting since devices might've been running when deleting stuff 2008-01-19 20:58:50 -08:00
Chris Robinson 543eb9d217 Don't use a multiple lists for extensions 2008-01-19 20:02:40 -08:00
Chris Robinson bc963463f3 Add an option for setting the max number of sources 2008-01-19 19:38:06 -08:00
Chris Robinson 86b7b3d54b Remove duplication of setting the max source count 2008-01-19 19:28:34 -08:00
Chris Robinson 799ba1f954 Use less ambiguous config file names 2008-01-19 18:18:14 -08:00
Chris Robinson 1a3e39e452 Remove some unnecessary duplicate math, which was making long lines 2008-01-19 00:49:05 -08:00
Chris Robinson 7dc73815ae Remove some branches 2008-01-18 21:39:09 -08:00
Chris Robinson 4caf2c7edd Implement AL_EFFECT_REVERB
Here is a quick description of how the reverb effect works:

 +--->---+*(4)
 |       V       new sample
 +-----+---+---+    |
 |extra|ltr|ref| <- +*(1)
 +-----+---+---+
   (3,5)*|   |*(2)
         +-->|
             V
         out sample

 1) Apply master reverb gain to incoming sample and place it at the head of the
    buffer. The master reverb gainhf was already applied when the source was
    initially mixed.
 2) Copy the delayed reflection sample to an output sample and apply the
    reflection gain.
 3) Apply the late reverb gain to the late reverb sample
 4) Copy the end of the buffer, applying a decay gain and the decay hf ratio,
    and add to the late reverb.
 5) Copy the late reverb sample, adding to the output sample.

 Then the head and sampling points are shifted forward, and done again for each
 new sample. The extra buffer length is determined by the Reverb Density
 property. A value of 0 gives a length of 0.1 seconds (long, with fairly
 distinct echos) , and 1 gives 0.075 seconds (short, indistinct echos).
 The decay gain is calculated such that after a number of loops to satisfy the
 Decay Time, a sample will be 1/32768th as powerful (virtually insignificant to
 the resulting output, and only getting further reduced). It is calculated as:

 DecayGain = pow(1.0f/32768.0f, 1.0/(DecayTime/ExtraLength));

 Things to note: Reverb Diffusion is not currently handled, nor is Decay HF
 Limit. Decay HF Ratios above 1 probably give incorrect results. Also, this
 method likely sucks, but it's the best I can come up with before release. :)
2008-01-18 21:25:40 -08:00
Chris Robinson 1b9d740244 Remove duplicated source freeing code 2008-01-18 00:56:01 -08:00
Chris Robinson 497ada9c77 Buffer size fixes. Partially reverts the ALSA buffer size "fix" 2008-01-18 00:28:25 -08:00
Chris Robinson 73e386dc6f Release effect slots when deleting sources 2008-01-17 21:17:08 -08:00
Chris Robinson 43cfc097de Don't dereference ALContext if there's no context yet
Patch by Evgeny A. Marchenko
2008-01-17 12:57:22 -08:00
Chris Robinson 6735fc7911 Add missing config.h includes 2008-01-16 14:09:04 -08:00
Chris Robinson be34dbe608 Don't include alAuxEffectSlot.h in alSource.h 2008-01-16 14:01:24 -08:00
Chris Robinson 70102e7cd3 Fix debug comments 2008-01-16 13:58:10 -08:00
Chris Robinson 8ad16145f6 Make sure sources are deleted with the context 2008-01-16 13:27:15 -08:00
Chris Robinson bb7b18d4e1 Keep track of references to effect slots, so they aren't deleted while in use 2008-01-16 13:20:09 -08:00
Chris Robinson 4742dedb45 Don't clamp wet gain if there's no send slot, and move slot gain calculation
To remove an extra if check
2008-01-16 13:00:35 -08:00
Chris Robinson 10a9bc62bf Store a reference to the effect slot in a source's send, not a copy 2008-01-16 12:43:25 -08:00
Chris Robinson 8cbbf17519 Allow getting and setting reverb parameters, as well as setting reverb effects
Reverb is still not implemented. The parameters just no-op for now.
2008-01-15 23:32:28 -08:00
Chris Robinson 24f433b938 Remove unneeded variables 2008-01-15 21:57:50 -08:00
Chris Robinson abc69dd3d0 Use acosf when available 2008-01-15 21:23:14 -08:00
Chris Robinson bf87aed459 Add reverb parameters 2008-01-15 20:32:20 -08:00
Chris Robinson 42306f93b3 Set default room rolloff factor 2008-01-15 19:59:19 -08:00
Chris Robinson 03ca50fa70 Use the previous low-pass filter again, as it seems to match the intended output better 2008-01-15 18:29:21 -08:00
Chris Robinson 7b1b8bd741 Set the default outer cone gainhf 2008-01-15 17:40:37 -08:00
Chris Robinson a11f25e47b Fix source and buffer out-of-memory conditions 2008-01-15 16:30:43 -08:00
Chris Robinson a6213ebfc7 Fix allocation of multiple effect slots, effects, and filters 2008-01-15 16:24:12 -08:00
Chris Robinson b95fcf5da1 Store effect slots in the context 2008-01-15 16:22:39 -08:00
Chris Robinson 707e596811 Don't append _struct to the effect slot struct name 2008-01-15 16:01:27 -08:00
Chris Robinson 5678041b92 Remove errant line 2008-01-15 15:53:58 -08:00
Chris Robinson 7f850d8fb0 Make sure a valid context is set for effect slots 2008-01-15 15:53:06 -08:00
Chris Robinson 7edc576e9c Fix some copy/paste errors 2008-01-15 15:37:54 -08:00
Chris Robinson 0041a09576 Move include so win32 gets it too, and remove unnecessary line 2008-01-15 12:45:24 -08:00
Chris Robinson 0f06254ecf Use a string for the detected backends, instead of a list 2008-01-15 12:43:43 -08:00
Chris Robinson ff5245ca8a Reword the backend options' text 2008-01-15 12:30:20 -08:00
Chris Robinson fe925ada59 Fix use of variables in checks 2008-01-15 12:25:51 -08:00
Chris Robinson b6596f38b9 Use CMAKE_BUILD_TYPE instead of custom options for compile modes 2008-01-15 10:00:56 -08:00
Chris Robinson 16fb3f6db5 Check for strncasecmp or _strnicmp 2008-01-14 16:30:18 -08:00
Chris Robinson a2e2522612 Let alIsExtensionPresent work with mixed-case extension names 2008-01-14 16:26:22 -08:00
Chris Robinson df07e8a65b Add support for AL_LOKI_quadriphonic 2008-01-14 16:11:15 -08:00
Chris Robinson 729f076c3b Reduce indentation 2008-01-14 15:38:15 -08:00
Chris Robinson dfc0118b8b Add an option for disabling ALSA mmap 2008-01-14 15:30:52 -08:00
Chris Robinson d9ef062caf Avoid busy waiting when waiting for suspend to clear 2008-01-14 13:23:49 -08:00
Chris Robinson 2b8ce3b4cf Let the mmap thread start the pcm stream when it's full
Instead of filling it with silence and starting it before the thread is active
2008-01-14 13:07:41 -08:00
Chris Robinson a27b855a39 Make sure the stream is playing when it's full 2008-01-14 12:59:44 -08:00
Chris Robinson 1634b69faf Don't restart the stream right after preparing it
It needs to be filled, first
2008-01-14 12:51:36 -08:00
Chris Robinson 59fa1f90d5 Don't attempt to recover the ALSA stream when filling silence
Since it was just prepared, any error would likely mean bigger problems
2008-01-14 12:49:21 -08:00
Chris Robinson e519338a2b Add missing stdio include 2008-01-14 10:55:13 -08:00
Chris Robinson 7a4870bd97 Close dangling devices when exiting 2008-01-14 10:54:33 -08:00
Chris Robinson a552e32a9a Destroy context if closing a device with one 2008-01-14 10:42:11 -08:00
Chris Robinson 29c6238b52 Keep track of open devices 2008-01-14 10:39:54 -08:00
Chris Robinson 38db8eb64b Reorder setting of some variables 2008-01-12 07:36:22 -08:00
Chris Robinson 3bbbf8a025 Merge branch 'master' into efx-experiment 2008-01-11 17:19:08 -08:00
Chris Robinson 978764cb6b Don't limit output wave filename size 2008-01-11 15:27:56 -08:00
Chris Robinson 893ecf1af2 Fix the Wave Writer's reliance on ftell
So output can work on FIFOs
2008-01-11 15:18:26 -08:00
Chris Robinson aa453b4e9e Use Sleep instead of usleep 2008-01-11 14:55:35 -08:00
Chris Robinson f8089d2026 Don't double-close a handle on error 2008-01-11 13:23:37 -08:00
Chris Robinson 40241b4e97 Don't attempt to open ALSA playback/capture if it didn't load 2008-01-11 13:00:30 -08:00
Chris Robinson f10408739e Add a wave file writing backend 2008-01-11 09:32:22 -08:00
Chris Robinson e1d0ad749b Use nanosleep instead of usleep
So not to rely on the non-standard unistd.h header
2008-01-11 08:15:44 -08:00
Chris Robinson 2a5a5b5c1b Add Sleep wrapper 2008-01-11 06:54:09 -08:00
Chris Robinson a97ecb8690 Add a timing wrapper, using gettimeofday 2008-01-11 06:01:51 -08:00
Chris Robinson bc56c00a9a Allow querying of ALC_MONO_SOURCES and ALC_STEREO_SOURCES
Based on a patch by Xavier Bouchoux
2008-01-10 08:24:23 -08:00
Chris Robinson 1e3ad2f9ce Use a more reliable thread loop for DSound, instead of a Win32 timer
Also use a simpler method for calculating the read/write location
2008-01-08 07:09:25 -08:00
Chris Robinson 9c97f07ec9 Remove unneeded silence field 2008-01-07 00:26:10 -08:00
Chris Robinson 312108a0d3 Try a different low-pass filter
Seems to be more correct, although it's not as powerful as the previous (which
may be a good thing)
2008-01-05 03:51:24 -08:00
Chris Robinson 5e48be27b8 Merge branch 'master' into efx-experiment 2008-01-04 14:40:38 -08:00
Chris Robinson 9ed574b399 Merge branch 'master' into efx-experiment 2008-01-01 06:29:11 -08:00
Chris Robinson 4e2f8e305e Fix wet volumes for multichannel path 2007-12-31 19:40:24 -08:00
Chris Robinson 3d78d93b40 Merge branch 'master' into efx-experiment 2007-12-31 19:34:52 -08:00
Chris Robinson 5a2f509104 Zero out wet send params when calculating source params
Instead of using a check in the mix loop
2007-12-31 19:13:18 -08:00
Chris Robinson d6288d63c6 Release the new stuff on exit for Windows, too 2007-12-22 13:18:05 -08:00
Chris Robinson 96bd020bd3 Use the right-channel sample... 2007-12-21 14:30:38 -08:00
Chris Robinson 8dc2aef058 Don't send stereo samples to the wet path without an auxiliary slot 2007-12-21 12:12:13 -08:00
Chris Robinson d43486f894 Fix extrenuous comment 2007-12-21 10:52:26 -08:00
Chris Robinson 229bc0d7e1 Merge branch 'master' into efx-experiment 2007-12-20 21:48:17 -08:00
Chris Robinson 7bf9ebf545 Allow auxiliary effect slot 0 when (un)setting the source auxiliary send 2007-12-18 23:51:21 -08:00
Chris Robinson 1e9302f564 Don't assume only one max send when setting source properties 2007-12-18 19:47:42 -08:00
Chris Robinson a433ae0ff4 Use the AuxSendAuto property of the effect slot 2007-12-18 19:43:54 -08:00
Chris Robinson e5609279da Implement AL_AUXILIARY_SEND_FILTER_GAIN_AUTO property 2007-12-18 19:13:41 -08:00
Chris Robinson afd3b71849 Implement AL_ROOM_ROLLOFF_FACTOR property 2007-12-18 19:03:40 -08:00
Chris Robinson 4cc64fa9a8 Don't allow properties on effect and filter name 0 2007-12-18 18:39:08 -08:00
Chris Robinson 7f55372727 Implement AL_EFFECT_TYPE property
Only the null effect type is accepted, right now
2007-12-18 18:15:52 -08:00
Chris Robinson 919b8c5bd1 Report one max auxiliary send per source 2007-12-18 18:13:49 -08:00
Chris Robinson a281027876 Implement AL_AUXILIARY_SEND_FILTER source property 2007-12-18 17:59:28 -08:00
Chris Robinson 72ce653bcf Remove unneeded struct member 2007-12-18 17:46:49 -08:00
Chris Robinson 17c8881444 Add AL_EFFECTSLOT_AUXILIARY_SEND_AUTO property (still unimplemented) 2007-12-18 17:41:44 -08:00
Chris Robinson 6aabc837a9 Fix initial WetMix setup 2007-12-18 17:21:13 -08:00
Chris Robinson 233e6f1765 Implement AL_AUXILIARY_SEND_FILTER_GAINHF_AUTO property 2007-12-18 16:54:22 -08:00
Chris Robinson 9adc20e4e7 Implement AL_EFFECTSLOT_GAIN property 2007-12-18 15:47:24 -08:00
Chris Robinson bdf16bf601 Add initial auxiliary send paths 2007-12-18 15:10:21 -08:00
Chris Robinson cf03bfa156 Implement AL_EFFECTSLOT_EFFECT property 2007-12-18 14:22:59 -08:00
Chris Robinson d4f556836a Only allow one auxiliary effect slot 2007-12-18 13:42:13 -08:00
Chris Robinson ebb30ec5b0 Implement AL_AIR_ABSORPTION_FACTOR source property 2007-12-18 13:37:07 -08:00
Chris Robinson 168d8e3bee Add AL_FILTER_LOWPASS support
Direct filters only, since auxiliary sends and slots aren't available yet
2007-12-17 22:42:38 -08:00
Chris Robinson 654788f2cf Implement AL_CONE_OUTER_GAINHF source property 2007-12-17 21:56:31 -08:00
Chris Robinson 06f2076f04 Implement AL_DIRECT_FILTER_GAINHF_AUTO source property 2007-12-17 21:33:05 -08:00
Chris Robinson fc43a13927 Implement basic lowpass filter path 2007-12-17 21:00:52 -08:00
Chris Robinson 28f80fa113 Add AL_METERS_PER_UNIT listener property 2007-12-17 19:40:43 -08:00
Chris Robinson 9ebf1c1431 Don't attempt to free null effects or filters 2007-12-17 19:12:49 -08:00
Chris Robinson 8795932420 0 is a valid filter and effect name 2007-12-17 19:10:22 -08:00
Chris Robinson be50a5f992 Export ALC_EXT_EFX extension
Preliminary testing can now begin. Almost nothing will work, but the bare
framework is laid out, and the functions can begin to be implemented.
2007-12-17 18:01:01 -08:00
Chris Robinson 7086f5f84c Add new listener property definition 2007-12-17 17:55:27 -08:00
Chris Robinson 670487b4dd Add effect type enums 2007-12-17 17:52:03 -08:00
Chris Robinson 0ac746df57 Add new functions to proc address table 2007-12-17 17:44:11 -08:00
Chris Robinson 7458273634 Add auxiliary effect slot function skeletons 2007-12-17 17:43:19 -08:00
Chris Robinson aa386041e7 Fix comment 2007-12-17 17:20:55 -08:00
Chris Robinson 44e0fa9325 Implement effect function skeletons 2007-12-17 17:20:11 -08:00
Chris Robinson 91d214d713 Release left over filters on close 2007-12-17 17:08:44 -08:00
Chris Robinson 635f673b95 Implement ALC properties 2007-12-17 16:55:07 -08:00
Chris Robinson 35a82970a3 Implement AL_DIRECT_FILTER source property
Only AL_FILTER_NULL filter types are currently supported, however
2007-12-17 16:28:16 -08:00
Chris Robinson ac9bef0b31 Add EFX source filter properties 2007-12-17 16:02:34 -08:00
Chris Robinson f6bb6cc891 Add AL_FILTER_TYPE enum 2007-12-17 15:49:26 -08:00
Chris Robinson 56ff747463 Implement filter function skeletons 2007-12-17 15:43:35 -08:00
94 changed files with 40554 additions and 9806 deletions
+5 -1
View File
@@ -1 +1,5 @@
CMakeConf
build
winbuild
include/sndio.h
include/sys
openal-soft.kdev4
+2917 -867
View File
File diff suppressed because it is too large Load Diff
+1112 -851
View File
File diff suppressed because it is too large Load Diff
+116 -38
View File
@@ -18,6 +18,16 @@
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#ifdef _WIN32
#ifdef __MINGW32__
#define _WIN32_IE 0x501
#else
#define _WIN32_IE 0x400
#endif
#endif
#include "config.h"
#include <stdlib.h>
#include <stdio.h>
#include <ctype.h>
@@ -25,6 +35,10 @@
#include "alMain.h"
#ifdef _WIN32_IE
#include <shlobj.h>
#endif
typedef struct ConfigEntry {
char *key;
char *value;
@@ -33,11 +47,11 @@ typedef struct ConfigEntry {
typedef struct ConfigBlock {
char *name;
ConfigEntry *entries;
size_t entryCount;
unsigned int entryCount;
} ConfigBlock;
static ConfigBlock *cfgBlocks;
static size_t cfgCount;
static unsigned int cfgCount;
static char buffer[1024];
@@ -48,7 +62,7 @@ static void LoadConfigFromFile(FILE *f)
while(fgets(buffer, sizeof(buffer), f))
{
size_t i = 0;
int i = 0;
while(isspace(buffer[i]))
i++;
@@ -60,6 +74,7 @@ static void LoadConfigFromFile(FILE *f)
if(buffer[0] == '[')
{
ConfigBlock *nextBlock;
unsigned int i;
i = 1;
while(buffer[i] && buffer[i] != ']')
@@ -67,7 +82,7 @@ static void LoadConfigFromFile(FILE *f)
if(!buffer[i])
{
AL_PRINT("config parse error: bad line \"%s\"\n", buffer);
ERR("config parse error: bad line \"%s\"\n", buffer);
continue;
}
buffer[i] = 0;
@@ -77,7 +92,7 @@ static void LoadConfigFromFile(FILE *f)
if(buffer[i] && !isspace(buffer[i]))
{
if(buffer[i] != '#')
AL_PRINT("config warning: extra data after block: \"%s\"\n", buffer+i);
WARN("config warning: extra data after block: \"%s\"\n", buffer+i);
break;
}
} while(buffer[i]);
@@ -88,7 +103,7 @@ static void LoadConfigFromFile(FILE *f)
if(strcasecmp(cfgBlocks[i].name, buffer+1) == 0)
{
nextBlock = cfgBlocks+i;
// AL_PRINT("found block '%s'\n", nextBlock->name);
TRACE("found block '%s'\n", nextBlock->name);
break;
}
}
@@ -98,7 +113,7 @@ static void LoadConfigFromFile(FILE *f)
nextBlock = realloc(cfgBlocks, (cfgCount+1)*sizeof(ConfigBlock));
if(!nextBlock)
{
AL_PRINT("config parse error: error reallocating config blocks\n");
ERR("config parse error: error reallocating config blocks\n");
continue;
}
cfgBlocks = nextBlock;
@@ -109,7 +124,7 @@ static void LoadConfigFromFile(FILE *f)
nextBlock->entries = NULL;
nextBlock->entryCount = 0;
// AL_PRINT("found new block '%s'\n", nextBlock->name);
TRACE("found new block '%s'\n", nextBlock->name);
}
curBlock = nextBlock;
continue;
@@ -123,7 +138,7 @@ static void LoadConfigFromFile(FILE *f)
if(!buffer[i] || buffer[i] == '#' || i == 0)
{
AL_PRINT("config parse error: malformed option line: \"%s\"\n", buffer);
ERR("config parse error: malformed option line: \"%s\"\n", buffer);
continue;
}
@@ -136,7 +151,7 @@ static void LoadConfigFromFile(FILE *f)
i++;
if(buffer[i] != '=')
{
AL_PRINT("config parse error: option without a value: \"%s\"\n", buffer);
ERR("config parse error: option without a value: \"%s\"\n", buffer);
continue;
}
}
@@ -147,20 +162,20 @@ static void LoadConfigFromFile(FILE *f)
/* Check if we already have this option set */
ent = curBlock->entries;
while((size_t)(ent-curBlock->entries) < curBlock->entryCount)
while((unsigned int)(ent-curBlock->entries) < curBlock->entryCount)
{
if(strcasecmp(ent->key, buffer) == 0)
break;
ent++;
}
if((size_t)(ent-curBlock->entries) >= curBlock->entryCount)
if((unsigned int)(ent-curBlock->entries) >= curBlock->entryCount)
{
/* Allocate a new option entry */
ent = realloc(curBlock->entries, (curBlock->entryCount+1)*sizeof(ConfigEntry));
if(!ent)
{
AL_PRINT("config parse error: error reallocating config entries\n");
ERR("config parse error: error reallocating config entries\n");
continue;
}
curBlock->entries = ent;
@@ -180,18 +195,19 @@ static void LoadConfigFromFile(FILE *f)
i++;
do {
i--;
} while(isspace(buffer[i]));
} while(i >= 0 && isspace(buffer[i]));
buffer[++i] = 0;
free(ent->value);
ent->value = strdup(buffer);
// AL_PRINT("found '%s' = '%s'\n", ent->key, ent->value);
TRACE("found '%s' = '%s'\n", ent->key, ent->value);
}
}
void ReadALConfig(void)
{
const char *str;
FILE *f;
cfgBlocks = calloc(1, sizeof(ConfigBlock));
@@ -199,16 +215,27 @@ void ReadALConfig(void)
cfgCount = 1;
#ifdef _WIN32
if(SHGetSpecialFolderPathA(NULL, buffer, CSIDL_APPDATA, FALSE) != FALSE)
{
size_t p = strlen(buffer);
snprintf(buffer+p, sizeof(buffer)-p, "\\alsoft.ini");
f = fopen(buffer, "rt");
if(f)
{
LoadConfigFromFile(f);
fclose(f);
}
}
#else
f = fopen("/etc/openal/config", "r");
f = fopen("/etc/openal/alsoft.conf", "r");
if(f)
{
LoadConfigFromFile(f);
fclose(f);
}
if(getenv("HOME") && *(getenv("HOME")))
if((str=getenv("HOME")) != NULL && *str)
{
snprintf(buffer, sizeof(buffer), "%s/.openalrc", getenv("HOME"));
snprintf(buffer, sizeof(buffer), "%s/.alsoftrc", str);
f = fopen(buffer, "r");
if(f)
{
@@ -217,15 +244,24 @@ void ReadALConfig(void)
}
}
#endif
if((str=getenv("ALSOFT_CONF")) != NULL && *str)
{
f = fopen(str, "r");
if(f)
{
LoadConfigFromFile(f);
fclose(f);
}
}
}
void FreeALConfig(void)
{
size_t i;
unsigned int i;
for(i = 0;i < cfgCount;i++)
{
size_t j;
unsigned int j;
for(j = 0;j < cfgBlocks[i].entryCount;j++)
{
free(cfgBlocks[i].entries[j].key);
@@ -241,45 +277,87 @@ void FreeALConfig(void)
const char *GetConfigValue(const char *blockName, const char *keyName, const char *def)
{
size_t i, j;
unsigned int i, j;
if(keyName)
if(!keyName)
return def;
if(!blockName)
blockName = "general";
for(i = 0;i < cfgCount;i++)
{
if(!blockName)
blockName = "general";
if(strcasecmp(cfgBlocks[i].name, blockName) != 0)
continue;
for(i = 0;i < cfgCount;i++)
for(j = 0;j < cfgBlocks[i].entryCount;j++)
{
if(strcasecmp(cfgBlocks[i].name, blockName) != 0)
continue;
for(j = 0;j < cfgBlocks[i].entryCount;j++)
if(strcasecmp(cfgBlocks[i].entries[j].key, keyName) == 0)
{
if(strcasecmp(cfgBlocks[i].entries[j].key, keyName) == 0)
TRACE("Found %s:%s = \"%s\"\n", blockName, keyName,
cfgBlocks[i].entries[j].value);
if(cfgBlocks[i].entries[j].value[0])
return cfgBlocks[i].entries[j].value;
return def;
}
}
}
TRACE("Key %s:%s not found\n", blockName, keyName);
return def;
}
int GetConfigValueInt(const char *blockName, const char *keyName, int def)
int ConfigValueExists(const char *blockName, const char *keyName)
{
const char *val = GetConfigValue(blockName, keyName, "");
if(!val[0]) return def;
return strtol(val, NULL, 0);
return !!val[0];
}
float GetConfigValueFloat(const char *blockName, const char *keyName, float def)
int ConfigValueStr(const char *blockName, const char *keyName, const char **ret)
{
const char *val = GetConfigValue(blockName, keyName, "");
if(!val[0]) return 0;
*ret = val;
return 1;
}
int ConfigValueInt(const char *blockName, const char *keyName, int *ret)
{
const char *val = GetConfigValue(blockName, keyName, "");
if(!val[0]) return 0;
*ret = strtol(val, NULL, 0);
return 1;
}
int ConfigValueUInt(const char *blockName, const char *keyName, unsigned int *ret)
{
const char *val = GetConfigValue(blockName, keyName, "");
if(!val[0]) return 0;
*ret = strtoul(val, NULL, 0);
return 1;
}
int ConfigValueFloat(const char *blockName, const char *keyName, float *ret)
{
const char *val = GetConfigValue(blockName, keyName, "");
if(!val[0]) return 0;
if(!val[0]) return def;
#ifdef HAVE_STRTOF
return strtof(val, NULL);
*ret = strtof(val, NULL);
#else
return (float)strtod(val, NULL);
*ret = (float)strtod(val, NULL);
#endif
return 1;
}
int GetConfigValueBool(const char *blockName, const char *keyName, int def)
{
const char *val = GetConfigValue(blockName, keyName, "");
if(!val[0]) return !!def;
return (strcasecmp(val, "true") == 0 || strcasecmp(val, "yes") == 0 ||
strcasecmp(val, "on") == 0 || atoi(val) != 0);
}
+100
View File
@@ -0,0 +1,100 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 2011 by Chris Robinson.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdlib.h>
#include "alMain.h"
#include "alFilter.h"
#include "alAuxEffectSlot.h"
#include "alError.h"
#include "alu.h"
typedef struct ALdedicatedState {
// Must be first in all effects!
ALeffectState state;
ALfloat gains[MaxChannels];
} ALdedicatedState;
static ALvoid DedicatedDestroy(ALeffectState *effect)
{
ALdedicatedState *state = (ALdedicatedState*)effect;
free(state);
}
static ALboolean DedicatedDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
{
(void)effect;
(void)Device;
return AL_TRUE;
}
static ALvoid DedicatedUpdate(ALeffectState *effect, ALCdevice *device, const ALeffectslot *Slot)
{
ALdedicatedState *state = (ALdedicatedState*)effect;
ALfloat Gain;
ALsizei s;
Gain = Slot->Gain * Slot->effect.Dedicated.Gain;
for(s = 0;s < MaxChannels;s++)
state->gains[s] = 0.0f;
if(Slot->effect.type == AL_EFFECT_DEDICATED_DIALOGUE)
ComputeAngleGains(device, atan2f(0.0f, 1.0f), 0.0f, Gain, state->gains);
else if(Slot->effect.type == AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT)
state->gains[LFE] = Gain;
}
static ALvoid DedicatedProcess(ALeffectState *effect, ALuint SamplesToDo, const ALfloat *RESTRICT SamplesIn, ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE])
{
ALdedicatedState *state = (ALdedicatedState*)effect;
const ALfloat *gains = state->gains;
ALuint i, c;
for(c = 0;c < MaxChannels;c++)
{
for(i = 0;i < SamplesToDo;i++)
SamplesOut[c][i] = SamplesIn[i] * gains[c];
}
}
ALeffectState *DedicatedCreate(void)
{
ALdedicatedState *state;
ALsizei s;
state = malloc(sizeof(*state));
if(!state)
return NULL;
state->state.Destroy = DedicatedDestroy;
state->state.DeviceUpdate = DedicatedDeviceUpdate;
state->state.Update = DedicatedUpdate;
state->state.Process = DedicatedProcess;
for(s = 0;s < MaxChannels;s++)
state->gains[s] = 0.0f;
return &state->state;
}
+184
View File
@@ -0,0 +1,184 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 2009 by Chris Robinson.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <math.h>
#include <stdlib.h>
#include "alMain.h"
#include "alFilter.h"
#include "alAuxEffectSlot.h"
#include "alError.h"
#include "alu.h"
typedef struct ALechoState {
// Must be first in all effects!
ALeffectState state;
ALfloat *SampleBuffer;
ALuint BufferLength;
// The echo is two tap. The delay is the number of samples from before the
// current offset
struct {
ALuint delay;
} Tap[2];
ALuint Offset;
/* The panning gains for the two taps */
ALfloat Gain[2][MaxChannels];
ALfloat FeedGain;
FILTER iirFilter;
ALfloat history[2];
} ALechoState;
static ALvoid EchoDestroy(ALeffectState *effect)
{
ALechoState *state = (ALechoState*)effect;
if(state)
{
free(state->SampleBuffer);
state->SampleBuffer = NULL;
free(state);
}
}
static ALboolean EchoDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
{
ALechoState *state = (ALechoState*)effect;
ALuint maxlen, i;
// Use the next power of 2 for the buffer length, so the tap offsets can be
// wrapped using a mask instead of a modulo
maxlen = fastf2u(AL_ECHO_MAX_DELAY * Device->Frequency) + 1;
maxlen += fastf2u(AL_ECHO_MAX_LRDELAY * Device->Frequency) + 1;
maxlen = NextPowerOf2(maxlen);
if(maxlen != state->BufferLength)
{
void *temp;
temp = realloc(state->SampleBuffer, maxlen * sizeof(ALfloat));
if(!temp)
return AL_FALSE;
state->SampleBuffer = temp;
state->BufferLength = maxlen;
}
for(i = 0;i < state->BufferLength;i++)
state->SampleBuffer[i] = 0.0f;
return AL_TRUE;
}
static ALvoid EchoUpdate(ALeffectState *effect, ALCdevice *Device, const ALeffectslot *Slot)
{
ALechoState *state = (ALechoState*)effect;
ALuint frequency = Device->Frequency;
ALfloat lrpan, cw, g, gain;
ALfloat dirGain;
ALuint i;
state->Tap[0].delay = fastf2u(Slot->effect.Echo.Delay * frequency) + 1;
state->Tap[1].delay = fastf2u(Slot->effect.Echo.LRDelay * frequency);
state->Tap[1].delay += state->Tap[0].delay;
lrpan = Slot->effect.Echo.Spread;
state->FeedGain = Slot->effect.Echo.Feedback;
cw = cosf(F_PI*2.0f * LOWPASSFREQREF / frequency);
g = 1.0f - Slot->effect.Echo.Damping;
state->iirFilter.coeff = lpCoeffCalc(g, cw);
gain = Slot->Gain;
for(i = 0;i < MaxChannels;i++)
{
state->Gain[0][i] = 0.0f;
state->Gain[1][i] = 0.0f;
}
dirGain = fabsf(lrpan);
/* First tap panning */
ComputeAngleGains(Device, atan2f(-lrpan, 0.0f), (1.0f-dirGain)*F_PI, gain, state->Gain[0]);
/* Second tap panning */
ComputeAngleGains(Device, atan2f(+lrpan, 0.0f), (1.0f-dirGain)*F_PI, gain, state->Gain[1]);
}
static ALvoid EchoProcess(ALeffectState *effect, ALuint SamplesToDo, const ALfloat *RESTRICT SamplesIn, ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE])
{
ALechoState *state = (ALechoState*)effect;
const ALuint mask = state->BufferLength-1;
const ALuint tap1 = state->Tap[0].delay;
const ALuint tap2 = state->Tap[1].delay;
ALuint offset = state->Offset;
ALfloat smp;
ALuint i, k;
for(i = 0;i < SamplesToDo;i++,offset++)
{
/* First tap */
smp = state->SampleBuffer[(offset-tap1) & mask];
for(k = 0;k < MaxChannels;k++)
SamplesOut[k][i] += smp * state->Gain[0][k];
/* Second tap */
smp = state->SampleBuffer[(offset-tap2) & mask];
for(k = 0;k < MaxChannels;k++)
SamplesOut[k][i] += smp * state->Gain[1][k];
// Apply damping and feedback gain to the second tap, and mix in the
// new sample
smp = lpFilter2P(&state->iirFilter, 0, smp+SamplesIn[i]);
state->SampleBuffer[offset&mask] = smp * state->FeedGain;
}
state->Offset = offset;
}
ALeffectState *EchoCreate(void)
{
ALechoState *state;
state = malloc(sizeof(*state));
if(!state)
return NULL;
state->state.Destroy = EchoDestroy;
state->state.DeviceUpdate = EchoDeviceUpdate;
state->state.Update = EchoUpdate;
state->state.Process = EchoProcess;
state->BufferLength = 0;
state->SampleBuffer = NULL;
state->Tap[0].delay = 0;
state->Tap[1].delay = 0;
state->Offset = 0;
state->iirFilter.coeff = 0.0f;
state->iirFilter.history[0] = 0.0f;
state->iirFilter.history[1] = 0.0f;
return &state->state;
}
+204
View File
@@ -0,0 +1,204 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 2009 by Chris Robinson.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <math.h>
#include <stdlib.h>
#include "alMain.h"
#include "alFilter.h"
#include "alAuxEffectSlot.h"
#include "alError.h"
#include "alu.h"
typedef struct ALmodulatorState {
// Must be first in all effects!
ALeffectState state;
enum {
SINUSOID,
SAWTOOTH,
SQUARE
} Waveform;
ALuint index;
ALuint step;
ALfloat Gain[MaxChannels];
FILTER iirFilter;
ALfloat history[1];
} ALmodulatorState;
#define WAVEFORM_FRACBITS 16
#define WAVEFORM_FRACONE (1<<WAVEFORM_FRACBITS)
#define WAVEFORM_FRACMASK (WAVEFORM_FRACONE-1)
static __inline ALfloat Sin(ALuint index)
{
return sinf(index * (F_PI*2.0f / WAVEFORM_FRACONE));
}
static __inline ALfloat Saw(ALuint index)
{
return index*(2.0f/WAVEFORM_FRACONE) - 1.0f;
}
static __inline ALfloat Square(ALuint index)
{
return ((index>>(WAVEFORM_FRACBITS-1))&1)*2.0f - 1.0f;
}
static __inline ALfloat hpFilter1P(FILTER *iir, ALuint offset, ALfloat input)
{
ALfloat *history = &iir->history[offset];
ALfloat a = iir->coeff;
ALfloat output = input;
output = output + (history[0]-output)*a;
history[0] = output;
return input - output;
}
#define DECL_TEMPLATE(func) \
static void Process##func(ALmodulatorState *state, ALuint SamplesToDo, \
const ALfloat *RESTRICT SamplesIn, \
ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE]) \
{ \
const ALuint step = state->step; \
ALuint index = state->index; \
ALfloat samp; \
ALuint i, k; \
\
for(i = 0;i < SamplesToDo;i++) \
{ \
samp = SamplesIn[i]; \
\
index += step; \
index &= WAVEFORM_FRACMASK; \
samp *= func(index); \
\
samp = hpFilter1P(&state->iirFilter, 0, samp); \
\
for(k = 0;k < MaxChannels;k++) \
SamplesOut[k][i] += state->Gain[k] * samp; \
} \
state->index = index; \
}
DECL_TEMPLATE(Sin)
DECL_TEMPLATE(Saw)
DECL_TEMPLATE(Square)
#undef DECL_TEMPLATE
static ALvoid ModulatorDestroy(ALeffectState *effect)
{
ALmodulatorState *state = (ALmodulatorState*)effect;
free(state);
}
static ALboolean ModulatorDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
{
return AL_TRUE;
(void)effect;
(void)Device;
}
static ALvoid ModulatorUpdate(ALeffectState *effect, ALCdevice *Device, const ALeffectslot *Slot)
{
ALmodulatorState *state = (ALmodulatorState*)effect;
ALfloat gain, cw, a = 0.0f;
ALuint index;
if(Slot->effect.Modulator.Waveform == AL_RING_MODULATOR_SINUSOID)
state->Waveform = SINUSOID;
else if(Slot->effect.Modulator.Waveform == AL_RING_MODULATOR_SAWTOOTH)
state->Waveform = SAWTOOTH;
else if(Slot->effect.Modulator.Waveform == AL_RING_MODULATOR_SQUARE)
state->Waveform = SQUARE;
state->step = fastf2u(Slot->effect.Modulator.Frequency*WAVEFORM_FRACONE /
Device->Frequency);
if(state->step == 0) state->step = 1;
cw = cosf(F_PI*2.0f * Slot->effect.Modulator.HighPassCutoff /
Device->Frequency);
a = (2.0f-cw) - sqrtf(powf(2.0f-cw, 2.0f) - 1.0f);
state->iirFilter.coeff = a;
gain = sqrtf(1.0f/Device->NumChan);
gain *= Slot->Gain;
for(index = 0;index < MaxChannels;index++)
state->Gain[index] = 0.0f;
for(index = 0;index < Device->NumChan;index++)
{
enum Channel chan = Device->Speaker2Chan[index];
state->Gain[chan] = gain;
}
}
static ALvoid ModulatorProcess(ALeffectState *effect, ALuint SamplesToDo, const ALfloat *RESTRICT SamplesIn, ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE])
{
ALmodulatorState *state = (ALmodulatorState*)effect;
switch(state->Waveform)
{
case SINUSOID:
ProcessSin(state, SamplesToDo, SamplesIn, SamplesOut);
break;
case SAWTOOTH:
ProcessSaw(state, SamplesToDo, SamplesIn, SamplesOut);
break;
case SQUARE:
ProcessSquare(state, SamplesToDo, SamplesIn, SamplesOut);
break;
}
}
ALeffectState *ModulatorCreate(void)
{
ALmodulatorState *state;
state = malloc(sizeof(*state));
if(!state)
return NULL;
state->state.Destroy = ModulatorDestroy;
state->state.DeviceUpdate = ModulatorDeviceUpdate;
state->state.Update = ModulatorUpdate;
state->state.Process = ModulatorProcess;
state->index = 0;
state->step = 1;
state->iirFilter.coeff = 0.0f;
state->iirFilter.history[0] = 0.0f;
return &state->state;
}
+1283
View File
File diff suppressed because it is too large Load Diff
+34 -25
View File
@@ -18,11 +18,11 @@
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <string.h>
#include <stdlib.h>
#include "config.h"
#include "alMain.h"
@@ -40,18 +40,15 @@ struct RingBuffer {
RingBuffer *CreateRingBuffer(ALsizei frame_size, ALsizei length)
{
RingBuffer *ring = calloc(1, sizeof(*ring));
RingBuffer *ring = calloc(1, sizeof(*ring) + ((length+1) * frame_size));
if(ring)
{
ring->mem = (ALubyte*)(ring+1);
ring->frame_size = frame_size;
ring->length = length+1;
ring->write_pos = 1;
ring->mem = malloc((length+1)*frame_size);
if(!ring->mem)
{
free(ring);
ring = NULL;
}
ring->read_pos = 0;
ring->write_pos = 0;
InitializeCriticalSection(&ring->cs);
}
@@ -63,45 +60,58 @@ void DestroyRingBuffer(RingBuffer *ring)
if(ring)
{
DeleteCriticalSection(&ring->cs);
free(ring->mem);
free(ring);
}
}
ALsizei RingBufferSize(RingBuffer *ring)
{
return (ring->write_pos-ring->read_pos-1+ring->length) % ring->length;
ALsizei s;
EnterCriticalSection(&ring->cs);
s = (ring->write_pos-ring->read_pos+ring->length) % ring->length;
LeaveCriticalSection(&ring->cs);
return s;
}
void WriteRingBuffer(RingBuffer *ring, const ALubyte *data, ALsizei len)
{
int remain = ring->length - ring->write_pos;
int remain;
EnterCriticalSection(&ring->cs);
if((ring->read_pos-ring->write_pos+ring->length)%ring->length < len)
ring->read_pos = (ring->write_pos+len) % ring->length;
remain = (ring->read_pos-ring->write_pos-1+ring->length) % ring->length;
if(remain < len) len = remain;
if(remain < len)
if(len > 0)
{
memcpy(ring->mem+(ring->write_pos*ring->frame_size), data, remain*ring->frame_size);
memcpy(ring->mem, data+(remain*ring->frame_size), (len-remain)*ring->frame_size);
}
else
memcpy(ring->mem+(ring->write_pos*ring->frame_size), data, len*ring->frame_size);
remain = ring->length - ring->write_pos;
if(remain < len)
{
memcpy(ring->mem+(ring->write_pos*ring->frame_size), data,
remain*ring->frame_size);
memcpy(ring->mem, data+(remain*ring->frame_size),
(len-remain)*ring->frame_size);
}
else
memcpy(ring->mem+(ring->write_pos*ring->frame_size), data,
len*ring->frame_size);
ring->write_pos += len;
ring->write_pos %= ring->length;
ring->write_pos += len;
ring->write_pos %= ring->length;
}
LeaveCriticalSection(&ring->cs);
}
void ReadRingBuffer(RingBuffer *ring, ALubyte *data, ALsizei len)
{
int remain = ring->length - ring->read_pos;
int remain;
EnterCriticalSection(&ring->cs);
remain = ring->length - ring->read_pos;
if(remain < len)
{
memcpy(data, ring->mem+(ring->read_pos*ring->frame_size), remain*ring->frame_size);
@@ -115,4 +125,3 @@ void ReadRingBuffer(RingBuffer *ring, ALubyte *data, ALsizei len)
LeaveCriticalSection(&ring->cs);
}
+33 -14
View File
@@ -18,16 +18,17 @@
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdlib.h>
#include "alMain.h"
#include "alThunk.h"
#define THREAD_STACK_SIZE (1*1024*1024) /* 1MB */
#ifdef _WIN32
#include <windows.h>
typedef struct {
ALuint (*func)(ALvoid*);
ALvoid *ptr;
@@ -47,13 +48,14 @@ static DWORD CALLBACK StarterFunc(void *ptr)
ALvoid *StartThread(ALuint (*func)(ALvoid*), ALvoid *ptr)
{
DWORD dummy;
ThreadInfo *inf = malloc(sizeof(ThreadInfo));
if(!inf) return 0;
inf->func = func;
inf->ptr = ptr;
inf->thread = CreateThread(NULL, 0, StarterFunc, inf, 0, NULL);
inf->thread = CreateThread(NULL, THREAD_STACK_SIZE, StarterFunc, inf, 0, &dummy);
if(!inf->thread)
{
free(inf);
@@ -70,6 +72,7 @@ ALuint StopThread(ALvoid *thread)
WaitForSingleObject(inf->thread, INFINITE);
GetExitCodeThread(inf->thread, &ret);
CloseHandle(inf->thread);
free(inf);
@@ -83,30 +86,44 @@ ALuint StopThread(ALvoid *thread)
typedef struct {
ALuint (*func)(ALvoid*);
ALvoid *ptr;
ALuint ret;
pthread_t thread;
} ThreadInfo;
static void *StarterFunc(void *ptr)
{
ThreadInfo *inf = (ThreadInfo*)ptr;
ALint ret;
ret = inf->func(inf->ptr);
return (void*)ret;
inf->ret = inf->func(inf->ptr);
return NULL;
}
ALvoid *StartThread(ALuint (*func)(ALvoid*), ALvoid *ptr)
{
pthread_attr_t attr;
ThreadInfo *inf = malloc(sizeof(ThreadInfo));
if(!inf) return 0;
if(!inf) return NULL;
inf->func = func;
inf->ptr = ptr;
if(pthread_create(&inf->thread, NULL, StarterFunc, inf) != 0)
if(pthread_attr_init(&attr) != 0)
{
free(inf);
return NULL;
}
if(pthread_attr_setstacksize(&attr, THREAD_STACK_SIZE) != 0)
{
pthread_attr_destroy(&attr);
free(inf);
return NULL;
}
inf->func = func;
inf->ptr = ptr;
if(pthread_create(&inf->thread, &attr, StarterFunc, inf) != 0)
{
pthread_attr_destroy(&attr);
free(inf);
return NULL;
}
pthread_attr_destroy(&attr);
return inf;
}
@@ -114,12 +131,14 @@ ALvoid *StartThread(ALuint (*func)(ALvoid*), ALvoid *ptr)
ALuint StopThread(ALvoid *thread)
{
ThreadInfo *inf = thread;
void *ret;
ALuint ret;
pthread_join(inf->thread, NULL);
ret = inf->ret;
pthread_join(inf->thread, &ret);
free(inf);
return (ALuint)ret;
return ret;
}
#endif
-912
View File
@@ -1,912 +0,0 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2007 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdlib.h>
#include <stdio.h>
#include <memory.h>
#ifdef HAVE_DLFCN_H
#include <dlfcn.h>
#endif
#include "alMain.h"
#include "AL/al.h"
#include "AL/alc.h"
#include <alsa/asoundlib.h>
typedef struct {
snd_pcm_t *pcmHandle;
snd_pcm_format_t format;
ALvoid *buffer;
ALsizei size;
int killNow;
ALvoid *thread;
} alsa_data;
typedef struct {
ALCchar *name;
int card, dev;
} DevMap;
static void *alsa_handle;
#define MAKE_FUNC(f) static typeof(f) * p##f
MAKE_FUNC(snd_strerror);
MAKE_FUNC(snd_pcm_open);
MAKE_FUNC(snd_pcm_close);
MAKE_FUNC(snd_pcm_nonblock);
MAKE_FUNC(snd_pcm_frames_to_bytes);
MAKE_FUNC(snd_pcm_hw_params_malloc);
MAKE_FUNC(snd_pcm_hw_params_free);
MAKE_FUNC(snd_pcm_hw_params_any);
MAKE_FUNC(snd_pcm_hw_params_set_access);
MAKE_FUNC(snd_pcm_hw_params_set_format);
MAKE_FUNC(snd_pcm_hw_params_set_channels);
MAKE_FUNC(snd_pcm_hw_params_set_periods_near);
MAKE_FUNC(snd_pcm_hw_params_set_rate_near);
MAKE_FUNC(snd_pcm_hw_params_set_rate);
MAKE_FUNC(snd_pcm_hw_params_set_buffer_size_near);
MAKE_FUNC(snd_pcm_hw_params_set_buffer_size_min);
MAKE_FUNC(snd_pcm_hw_params_get_access);
MAKE_FUNC(snd_pcm_hw_params);
MAKE_FUNC(snd_pcm_prepare);
MAKE_FUNC(snd_pcm_start);
MAKE_FUNC(snd_pcm_resume);
MAKE_FUNC(snd_pcm_state);
MAKE_FUNC(snd_pcm_avail_update);
MAKE_FUNC(snd_pcm_areas_silence);
MAKE_FUNC(snd_pcm_mmap_begin);
MAKE_FUNC(snd_pcm_mmap_commit);
MAKE_FUNC(snd_pcm_writei);
MAKE_FUNC(snd_pcm_drain);
MAKE_FUNC(snd_pcm_info_malloc);
MAKE_FUNC(snd_pcm_info_free);
MAKE_FUNC(snd_pcm_info_set_device);
MAKE_FUNC(snd_pcm_info_set_subdevice);
MAKE_FUNC(snd_pcm_info_set_stream);
MAKE_FUNC(snd_pcm_info_get_name);
MAKE_FUNC(snd_ctl_pcm_next_device);
MAKE_FUNC(snd_ctl_pcm_info);
MAKE_FUNC(snd_ctl_open);
MAKE_FUNC(snd_ctl_close);
MAKE_FUNC(snd_ctl_card_info_malloc);
MAKE_FUNC(snd_ctl_card_info_free);
MAKE_FUNC(snd_ctl_card_info);
MAKE_FUNC(snd_ctl_card_info_get_name);
MAKE_FUNC(snd_card_next);
#undef MAKE_FUNC
#define MAX_DEVICES 16
#define MAX_ALL_DEVICES 32
static DevMap allDevNameMap[MAX_ALL_DEVICES];
static ALCchar *alsaDeviceList[MAX_DEVICES];
static ALCchar *alsaCaptureDeviceList[MAX_DEVICES];
static int xrun_recovery(snd_pcm_t *handle, int err)
{
if (err == -EPIPE)
{ /* under-run */
err = psnd_pcm_prepare(handle);
if(err >= 0)
err = psnd_pcm_start(handle);
if (err < 0)
AL_PRINT("prepare failed: %s\n", psnd_strerror(err));
}
else if (err == -ESTRPIPE)
{
while ((err = psnd_pcm_resume(handle)) == -EAGAIN)
usleep(1); /* wait until the suspend flag is released */
if (err < 0)
{
err = psnd_pcm_prepare(handle);
if(err >= 0)
err = psnd_pcm_start(handle);
if (err < 0)
AL_PRINT("prepare failed: %s\n", psnd_strerror(err));
}
}
return err;
}
static ALuint ALSAProc(ALvoid *ptr)
{
ALCdevice *pDevice = (ALCdevice*)ptr;
alsa_data *data = (alsa_data*)pDevice->ExtraData;
const snd_pcm_channel_area_t *areas = NULL;
snd_pcm_sframes_t avail, commitres;
snd_pcm_uframes_t offset, frames;
char *WritePtr;
int WriteCnt;
int err;
while(!data->killNow)
{
snd_pcm_state_t state = psnd_pcm_state(data->pcmHandle);
if(state == SND_PCM_STATE_XRUN)
{
err = xrun_recovery(data->pcmHandle, -EPIPE);
if (err < 0)
{
AL_PRINT("XRUN recovery failed: %s\n", psnd_strerror(err));
break;
}
}
else if (state == SND_PCM_STATE_SUSPENDED)
{
err = xrun_recovery(data->pcmHandle, -ESTRPIPE);
if (err < 0)
{
AL_PRINT("SUSPEND recovery failed: %s\n", psnd_strerror(err));
break;
}
}
avail = psnd_pcm_avail_update(data->pcmHandle);
if(avail < 0)
{
err = xrun_recovery(data->pcmHandle, avail);
if (err < 0)
{
AL_PRINT("available update failed: %s\n", psnd_strerror(err));
break;
}
}
// make sure there's frames to process
if(avail == 0)
{
usleep(1000);
continue;
}
// it is possible that contiguous areas are smaller, thus we use a loop
while (avail > 0)
{
frames = avail;
err = psnd_pcm_mmap_begin(data->pcmHandle, &areas, &offset, &frames);
if (err < 0)
{
err = xrun_recovery(data->pcmHandle, err);
if (err < 0)
AL_PRINT("mmap begin error: %s\n", psnd_strerror(err));
break;
}
SuspendContext(NULL);
WritePtr = (char*)areas->addr + (offset * areas->step / 8);
WriteCnt = psnd_pcm_frames_to_bytes(data->pcmHandle, frames);
aluMixData(pDevice->Context, WritePtr, WriteCnt, pDevice->Format);
ProcessContext(NULL);
commitres = psnd_pcm_mmap_commit(data->pcmHandle, offset, frames);
if (commitres < 0 || (commitres-frames) != 0)
{
AL_PRINT("mmap commit error: %s\n",
psnd_strerror(commitres >= 0 ? -EPIPE : commitres));
break;
}
avail -= frames;
}
}
return 0;
}
static ALuint ALSANoMMapProc(ALvoid *ptr)
{
ALCdevice *pDevice = (ALCdevice*)ptr;
alsa_data *data = (alsa_data*)pDevice->ExtraData;
snd_pcm_sframes_t avail;
char *WritePtr;
while(!data->killNow)
{
SuspendContext(NULL);
aluMixData(pDevice->Context, data->buffer, data->size, pDevice->Format);
ProcessContext(NULL);
WritePtr = data->buffer;
avail = (snd_pcm_uframes_t)data->size / psnd_pcm_frames_to_bytes(data->pcmHandle, 1);
while(avail > 0)
{
int ret = psnd_pcm_writei(data->pcmHandle, WritePtr, avail);
switch (ret)
{
case -EAGAIN:
continue;
case -ESTRPIPE:
do {
ret = psnd_pcm_resume(data->pcmHandle);
} while(ret == -EAGAIN);
break;
case -EPIPE:
break;
default:
if (ret >= 0)
{
WritePtr += psnd_pcm_frames_to_bytes(data->pcmHandle, ret);
avail -= ret;
}
break;
}
if (ret < 0)
{
ret = psnd_pcm_prepare(data->pcmHandle);
if(ret < 0)
break;
}
}
}
return 0;
}
static void fill_silence(snd_pcm_t *pcmHandle, snd_pcm_format_t alsaFormat, int channels)
{
const snd_pcm_channel_area_t *areas = NULL;
snd_pcm_sframes_t avail, commitres;
snd_pcm_uframes_t offset, frames;
int err;
avail = psnd_pcm_avail_update(pcmHandle);
if(avail < 0)
{
err = xrun_recovery(pcmHandle, avail);
if (err < 0)
{
AL_PRINT("available update failed: %s\n", psnd_strerror(err));
return;
}
}
// it is possible that contiguous areas are smaller, thus we use a loop
while (avail > 0)
{
frames = avail;
err = psnd_pcm_mmap_begin(pcmHandle, &areas, &offset, &frames);
if (err < 0)
{
err = xrun_recovery(pcmHandle, err);
if (err < 0)
{
AL_PRINT("mmap begin error: %s\n", psnd_strerror(err));
break;
}
continue;
}
psnd_pcm_areas_silence(areas, offset, channels, frames, alsaFormat);
commitres = psnd_pcm_mmap_commit(pcmHandle, offset, frames);
if (commitres < 0 || (commitres-frames) != 0)
{
AL_PRINT("mmap commit error: %s\n",
psnd_strerror(commitres >= 0 ? -EPIPE : commitres));
break;
}
avail -= frames;
}
}
static ALCboolean alsa_open_playback(ALCdevice *device, const ALCchar *deviceName)
{
snd_pcm_uframes_t bufferSizeInFrames;
snd_pcm_hw_params_t *p = NULL;
snd_pcm_access_t access;
unsigned int periods;
alsa_data *data;
char driver[64];
char *err;
int i;
strncpy(driver, GetConfigValue("alsa", "device", "default"), sizeof(driver)-1);
driver[sizeof(driver)-1] = 0;
if(deviceName)
{
size_t idx;
for(idx = 0;idx < MAX_ALL_DEVICES;idx++)
{
if(allDevNameMap[idx].name &&
strcmp(deviceName, allDevNameMap[idx].name) == 0)
{
device->szDeviceName = allDevNameMap[idx].name;
if(idx > 0)
sprintf(driver, "hw:%d,%d", allDevNameMap[idx].card, allDevNameMap[idx].dev);
goto open_alsa;
}
}
for(idx = 0;idx < MAX_DEVICES;idx++)
{
if(alsaDeviceList[idx] &&
strcmp(deviceName, alsaDeviceList[idx]) == 0)
{
device->szDeviceName = alsaDeviceList[idx];
if(idx > 0)
sprintf(driver, "hw:%zd,0", idx-1);
goto open_alsa;
}
}
return ALC_FALSE;
}
else
device->szDeviceName = alsaDeviceList[0];
open_alsa:
data = (alsa_data*)calloc(1, sizeof(alsa_data));
i = psnd_pcm_open(&data->pcmHandle, driver, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
if(i < 0)
{
usleep(200000);
i = psnd_pcm_open(&data->pcmHandle, driver, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
}
if(i >= 0)
{
i = psnd_pcm_nonblock(data->pcmHandle, 0);
if(i < 0)
psnd_pcm_close(data->pcmHandle);
}
if(i < 0)
{
free(data);
AL_PRINT("Could not open playback device '%s': %s\n", driver, psnd_strerror(i));
return ALC_FALSE;
}
switch(aluBytesFromFormat(device->Format))
{
case 1:
data->format = SND_PCM_FORMAT_U8;
break;
case 2:
data->format = SND_PCM_FORMAT_S16;
break;
default:
data->format = SND_PCM_FORMAT_UNKNOWN;
AL_PRINT("Unknown format?! %x\n", device->Format);
}
periods = GetConfigValueInt("alsa", "periods", 4);
if((int)periods <= 0)
periods = 4;
bufferSizeInFrames = device->UpdateFreq / periods;
psnd_pcm_hw_params_malloc(&p);
#define ok(func, str) (i=(func),((i<0)?(err=(str)),0:1))
/* start with the largest configuration space possible */
if(!(ok(psnd_pcm_hw_params_any(data->pcmHandle, p), "any") &&
/* set interleaved access */
(ok(psnd_pcm_hw_params_set_access(data->pcmHandle, p, SND_PCM_ACCESS_MMAP_INTERLEAVED), "set access") ||
ok(psnd_pcm_hw_params_set_access(data->pcmHandle, p, SND_PCM_ACCESS_RW_INTERLEAVED), "set access")) &&
/* set format (implicitly sets sample bits) */
ok(psnd_pcm_hw_params_set_format(data->pcmHandle, p, data->format), "set format") &&
/* set channels (implicitly sets frame bits) */
ok(psnd_pcm_hw_params_set_channels(data->pcmHandle, p, device->Channels), "set channels") &&
/* set periods (implicitly constrains period/buffer parameters) */
ok(psnd_pcm_hw_params_set_periods_near(data->pcmHandle, p, &periods, NULL), "set periods near") &&
/* set rate (implicitly constrains period/buffer parameters) */
ok(psnd_pcm_hw_params_set_rate_near(data->pcmHandle, p, &device->Frequency, NULL), "set rate near") &&
/* set buffer size in frame units (implicitly sets period size/bytes/time and buffer time/bytes) */
ok(psnd_pcm_hw_params_set_buffer_size_near(data->pcmHandle, p, &bufferSizeInFrames), "set buffer size near") &&
/* install and prepare hardware configuration */
ok(psnd_pcm_hw_params(data->pcmHandle, p), "set params")))
{
AL_PRINT("%s failed: %s\n", err, psnd_strerror(i));
psnd_pcm_hw_params_free(p);
psnd_pcm_close(data->pcmHandle);
free(data);
return ALC_FALSE;
}
#undef ok
if((i=psnd_pcm_hw_params_get_access(p, &access)) < 0)
{
AL_PRINT("get_access failed: %s\n", psnd_strerror(i));
psnd_pcm_hw_params_free(p);
psnd_pcm_close(data->pcmHandle);
free(data);
return ALC_FALSE;
}
psnd_pcm_hw_params_free(p);
device->MaxNoOfSources = 256;
device->UpdateFreq = bufferSizeInFrames;
data->size = psnd_pcm_frames_to_bytes(data->pcmHandle, device->UpdateFreq);
if(access == SND_PCM_ACCESS_RW_INTERLEAVED)
{
data->buffer = malloc(data->size);
if(!data->buffer)
{
AL_PRINT("buffer malloc failed\n");
psnd_pcm_close(data->pcmHandle);
free(data);
return ALC_FALSE;
}
}
else
{
i = psnd_pcm_prepare(data->pcmHandle);
if(i < 0)
{
AL_PRINT("prepare error: %s\n", psnd_strerror(i));
psnd_pcm_close(data->pcmHandle);
free(data->buffer);
free(data);
return ALC_FALSE;
}
fill_silence(data->pcmHandle, data->format, device->Channels);
i = psnd_pcm_start(data->pcmHandle);
if(i < 0)
{
AL_PRINT("start error: %s\n", psnd_strerror(i));
psnd_pcm_close(data->pcmHandle);
free(data->buffer);
free(data);
return ALC_FALSE;
}
}
device->ExtraData = data;
if(access == SND_PCM_ACCESS_RW_INTERLEAVED)
data->thread = StartThread(ALSANoMMapProc, device);
else
data->thread = StartThread(ALSAProc, device);
if(data->thread == NULL)
{
psnd_pcm_close(data->pcmHandle);
device->ExtraData = NULL;
free(data->buffer);
free(data);
return ALC_FALSE;
}
return ALC_TRUE;
}
static void alsa_close_playback(ALCdevice *device)
{
alsa_data *data = (alsa_data*)device->ExtraData;
data->killNow = 1;
StopThread(data->thread);
psnd_pcm_close(data->pcmHandle);
free(data->buffer);
free(data);
device->ExtraData = NULL;
}
static ALCboolean alsa_open_capture(ALCdevice *pDevice, const ALCchar *deviceName, ALCuint frequency, ALCenum format, ALCsizei SampleSize)
{
snd_pcm_format_t alsaFormat;
snd_pcm_hw_params_t *p;
unsigned int periods = 4;
snd_pcm_uframes_t bufferSizeInFrames;
alsa_data *data;
char driver[64];
char *err;
int i;
strncpy(driver, GetConfigValue("alsa", "capture", "default"), sizeof(driver)-1);
driver[sizeof(driver)-1] = 0;
if(deviceName)
{
size_t idx;
for(idx = 0;idx < MAX_DEVICES;idx++)
{
if(alsaCaptureDeviceList[idx] &&
strcmp(deviceName, alsaCaptureDeviceList[idx]) == 0)
{
pDevice->szDeviceName = alsaCaptureDeviceList[idx];
if(idx > 0)
sprintf(driver, "hw:%zd,0", idx-1);
goto open_alsa;
}
}
return ALC_FALSE;
}
else
pDevice->szDeviceName = alsaCaptureDeviceList[0];
open_alsa:
data = (alsa_data*)calloc(1, sizeof(alsa_data));
i = psnd_pcm_open(&data->pcmHandle, driver, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
if(i < 0)
{
usleep(200000);
i = psnd_pcm_open(&data->pcmHandle, driver, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
}
if(i >= 0)
{
i = psnd_pcm_nonblock(data->pcmHandle, 0);
if(i < 0)
psnd_pcm_close(data->pcmHandle);
}
if(i < 0)
{
free(data);
AL_PRINT("Could not open capture device '%s': %s\n", driver, psnd_strerror(i));
return ALC_FALSE;
}
switch(aluBytesFromFormat(format))
{
case 1:
alsaFormat = SND_PCM_FORMAT_U8;
break;
case 2:
alsaFormat = SND_PCM_FORMAT_S16;
break;
default:
alsaFormat = SND_PCM_FORMAT_UNKNOWN;
AL_PRINT("Unknown format?! %x\n", format);
}
bufferSizeInFrames = SampleSize / periods;
psnd_pcm_hw_params_malloc(&p);
#define ok(func, str) (i=(func),((i<0)?(err=(str)),0:1))
/* start with the largest configuration space possible */
if(!(ok(psnd_pcm_hw_params_any(data->pcmHandle, p), "any") &&
/* set interleaved access */
ok(psnd_pcm_hw_params_set_access(data->pcmHandle, p, SND_PCM_ACCESS_MMAP_INTERLEAVED), "set access") &&
/* set format (implicitly sets sample bits) */
ok(psnd_pcm_hw_params_set_format(data->pcmHandle, p, alsaFormat), "set format") &&
/* set channels (implicitly sets frame bits) */
ok(psnd_pcm_hw_params_set_channels(data->pcmHandle, p, pDevice->Channels), "set channels") &&
/* set periods (implicitly constrains period/buffer parameters) */
ok(psnd_pcm_hw_params_set_periods_near(data->pcmHandle, p, &periods, NULL), "set periods near") &&
/* set rate (implicitly constrains period/buffer parameters) */
ok(psnd_pcm_hw_params_set_rate(data->pcmHandle, p, frequency, 0), "set rate") &&
/* set buffer size in frame units (implicitly sets period size/bytes/time and buffer time/bytes) */
ok(psnd_pcm_hw_params_set_buffer_size_min(data->pcmHandle, p, &bufferSizeInFrames), "set buffer size min") &&
/* install and prepare hardware configuration */
ok(psnd_pcm_hw_params(data->pcmHandle, p), "set params")))
{
AL_PRINT("%s failed: %s\n", err, psnd_strerror(i));
psnd_pcm_hw_params_free(p);
psnd_pcm_close(data->pcmHandle);
free(data);
return ALC_FALSE;
}
#undef ok
psnd_pcm_hw_params_free(p);
i = psnd_pcm_prepare(data->pcmHandle);
if(i < 0)
{
AL_PRINT("prepare error: %s\n", psnd_strerror(i));
psnd_pcm_close(data->pcmHandle);
free(data);
return ALC_FALSE;
}
pDevice->ExtraData = data;
return ALC_TRUE;
}
static void alsa_close_capture(ALCdevice *pDevice)
{
alsa_data *data = (alsa_data*)pDevice->ExtraData;
psnd_pcm_close(data->pcmHandle);
free(data);
pDevice->ExtraData = NULL;
}
static void alsa_start_capture(ALCdevice *pDevice)
{
alsa_data *data = (alsa_data*)pDevice->ExtraData;
psnd_pcm_start(data->pcmHandle);
}
static void alsa_stop_capture(ALCdevice *pDevice)
{
alsa_data *data = (alsa_data*)pDevice->ExtraData;
psnd_pcm_drain(data->pcmHandle);
}
static void alsa_capture_samples(ALCdevice *pDevice, ALCvoid *pBuffer, ALCuint lSamples)
{
alsa_data *data = (alsa_data*)pDevice->ExtraData;
const snd_pcm_channel_area_t *areas = NULL;
snd_pcm_sframes_t frames, commitres;
snd_pcm_uframes_t size, offset;
int err;
frames = psnd_pcm_avail_update(data->pcmHandle);
if(frames < 0)
{
err = xrun_recovery(data->pcmHandle, frames);
if (err < 0)
AL_PRINT("available update failed: %s\n", psnd_strerror(err));
else
frames = psnd_pcm_avail_update(data->pcmHandle);
}
if (frames < (snd_pcm_sframes_t)lSamples)
{
SetALCError(ALC_INVALID_VALUE);
return;
}
// it is possible that contiguous areas are smaller, thus we use a loop
while (lSamples > 0)
{
char *Pointer;
int Count;
size = lSamples;
err = psnd_pcm_mmap_begin(data->pcmHandle, &areas, &offset, &size);
if (err < 0)
{
err = xrun_recovery(data->pcmHandle, err);
if (err < 0)
{
AL_PRINT("mmap begin error: %s\n", psnd_strerror(err));
break;
}
continue;
}
Pointer = (char*)areas->addr + (offset * areas->step / 8);
Count = size * pDevice->FrameSize;
memcpy(pBuffer, Pointer, Count);
pBuffer = (char*)pBuffer + Count;
commitres = psnd_pcm_mmap_commit(data->pcmHandle, offset, size);
if (commitres < 0 || (commitres-size) != 0)
{
AL_PRINT("mmap commit error: %s\n",
psnd_strerror(commitres >= 0 ? -EPIPE : commitres));
break;
}
lSamples -= size;
}
}
static ALCuint alsa_available_samples(ALCdevice *pDevice)
{
alsa_data *data = (alsa_data*)pDevice->ExtraData;
snd_pcm_sframes_t frames = psnd_pcm_avail_update(data->pcmHandle);
if(frames < 0)
{
int err = xrun_recovery(data->pcmHandle, frames);
if (err < 0)
AL_PRINT("available update failed: %s\n", psnd_strerror(err));
else
frames = psnd_pcm_avail_update(data->pcmHandle);
if(frames < 0) /* ew.. */
SetALCError(ALC_INVALID_DEVICE);
}
return max(frames, 0);
}
BackendFuncs alsa_funcs = {
alsa_open_playback,
alsa_close_playback,
alsa_open_capture,
alsa_close_capture,
alsa_start_capture,
alsa_stop_capture,
alsa_capture_samples,
alsa_available_samples
};
void alc_alsa_init(BackendFuncs *func_list)
{
snd_ctl_t *handle;
int card, err, dev, idx = 1;
snd_ctl_card_info_t *info;
snd_pcm_info_t *pcminfo;
snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK;
char name[128];
char *str;
*func_list = alsa_funcs;
#ifdef HAVE_DLFCN_H
alsa_handle = dlopen("libasound.so.2", RTLD_NOW);
if(!alsa_handle)
return;
dlerror();
#define LOAD_FUNC(f) do { \
p##f = (typeof(f)*)dlsym(alsa_handle, #f); \
if((str=dlerror()) != NULL) \
{ \
dlclose(alsa_handle); \
alsa_handle = NULL; \
AL_PRINT("Could not load %s from libasound.so.2: %s\n", #f, str); \
return; \
} \
} while(0)
#else
str = NULL;
alsa_handle = NULL;
#define LOAD_FUNC(f) p##f = f
#endif
LOAD_FUNC(snd_strerror);
LOAD_FUNC(snd_pcm_open);
LOAD_FUNC(snd_pcm_close);
LOAD_FUNC(snd_pcm_nonblock);
LOAD_FUNC(snd_pcm_frames_to_bytes);
LOAD_FUNC(snd_pcm_hw_params_malloc);
LOAD_FUNC(snd_pcm_hw_params_free);
LOAD_FUNC(snd_pcm_hw_params_any);
LOAD_FUNC(snd_pcm_hw_params_set_access);
LOAD_FUNC(snd_pcm_hw_params_set_format);
LOAD_FUNC(snd_pcm_hw_params_set_channels);
LOAD_FUNC(snd_pcm_hw_params_set_periods_near);
LOAD_FUNC(snd_pcm_hw_params_set_rate_near);
LOAD_FUNC(snd_pcm_hw_params_set_rate);
LOAD_FUNC(snd_pcm_hw_params_set_buffer_size_near);
LOAD_FUNC(snd_pcm_hw_params_set_buffer_size_min);
LOAD_FUNC(snd_pcm_hw_params_get_access);
LOAD_FUNC(snd_pcm_hw_params);
LOAD_FUNC(snd_pcm_prepare);
LOAD_FUNC(snd_pcm_start);
LOAD_FUNC(snd_pcm_resume);
LOAD_FUNC(snd_pcm_state);
LOAD_FUNC(snd_pcm_avail_update);
LOAD_FUNC(snd_pcm_areas_silence);
LOAD_FUNC(snd_pcm_mmap_begin);
LOAD_FUNC(snd_pcm_mmap_commit);
LOAD_FUNC(snd_pcm_writei);
LOAD_FUNC(snd_pcm_drain);
LOAD_FUNC(snd_pcm_info_malloc);
LOAD_FUNC(snd_pcm_info_free);
LOAD_FUNC(snd_pcm_info_set_device);
LOAD_FUNC(snd_pcm_info_set_subdevice);
LOAD_FUNC(snd_pcm_info_set_stream);
LOAD_FUNC(snd_pcm_info_get_name);
LOAD_FUNC(snd_ctl_pcm_next_device);
LOAD_FUNC(snd_ctl_pcm_info);
LOAD_FUNC(snd_ctl_open);
LOAD_FUNC(snd_ctl_close);
LOAD_FUNC(snd_ctl_card_info_malloc);
LOAD_FUNC(snd_ctl_card_info_free);
LOAD_FUNC(snd_ctl_card_info);
LOAD_FUNC(snd_ctl_card_info_get_name);
LOAD_FUNC(snd_card_next);
#undef LOAD_FUNC
psnd_ctl_card_info_malloc(&info);
psnd_pcm_info_malloc(&pcminfo);
card = -1;
if(psnd_card_next(&card) < 0 || card < 0)
AL_PRINT("no playback cards found...\n");
else
{
alsaDeviceList[0] = AppendDeviceList("ALSA Software on default");
allDevNameMap[0].name = AppendAllDeviceList("ALSA Software on default");
}
while (card >= 0) {
sprintf(name, "hw:%d", card);
if ((err = psnd_ctl_open(&handle, name, 0)) < 0) {
AL_PRINT("control open (%i): %s\n", card, psnd_strerror(err));
goto next_card;
}
if ((err = psnd_ctl_card_info(handle, info)) < 0) {
AL_PRINT("control hardware info (%i): %s\n", card, psnd_strerror(err));
psnd_ctl_close(handle);
goto next_card;
}
if(card < MAX_DEVICES-1) {
snprintf(name, sizeof(name), "ALSA Software on %s",
psnd_ctl_card_info_get_name(info));
alsaDeviceList[card+1] = AppendDeviceList(name);
}
dev = -1;
while (idx < MAX_ALL_DEVICES) {
const char *cname, *dname;
if (psnd_ctl_pcm_next_device(handle, &dev)<0)
AL_PRINT("snd_ctl_pcm_next_device failed\n");
if (dev < 0)
break;
psnd_pcm_info_set_device(pcminfo, dev);
psnd_pcm_info_set_subdevice(pcminfo, 0);
psnd_pcm_info_set_stream(pcminfo, stream);
if ((err = psnd_ctl_pcm_info(handle, pcminfo)) < 0) {
if (err != -ENOENT)
AL_PRINT("control digital audio info (%i): %s\n", card, psnd_strerror(err));
continue;
}
cname = psnd_ctl_card_info_get_name(info);
dname = psnd_pcm_info_get_name(pcminfo);
snprintf(name, sizeof(name), "ALSA Software on %s [%s]",
cname, dname);
allDevNameMap[idx].name = AppendAllDeviceList(name);
allDevNameMap[idx].card = card;
allDevNameMap[idx].dev = dev;
idx++;
}
psnd_ctl_close(handle);
next_card:
if(psnd_card_next(&card) < 0) {
AL_PRINT("snd_card_next failed\n");
break;
}
}
stream = SND_PCM_STREAM_CAPTURE;
card = -1;
if(psnd_card_next(&card) < 0 || card < 0) {
AL_PRINT("no capture cards found...\n");
psnd_pcm_info_free(pcminfo);
psnd_ctl_card_info_free(info);
return;
}
alsaCaptureDeviceList[0] = AppendCaptureDeviceList("ALSA Capture on default");
while (card >= 0) {
sprintf(name, "hw:%d", card);
handle = NULL;
if ((err = psnd_ctl_open(&handle, name, 0)) < 0) {
AL_PRINT("control open (%i): %s\n", card, psnd_strerror(err));
}
if (err >= 0 && (err = psnd_ctl_card_info(handle, info)) < 0) {
AL_PRINT("control hardware info (%i): %s\n", card, psnd_strerror(err));
psnd_ctl_close(handle);
}
else if (err >= 0 && card < MAX_DEVICES-1)
{
snprintf(name, sizeof(name), "ALSA Capture on %s",
psnd_ctl_card_info_get_name(info));
alsaCaptureDeviceList[card+1] = AppendCaptureDeviceList(name);
}
if(handle) psnd_ctl_close(handle);
if(psnd_card_next(&card) < 0) {
AL_PRINT("snd_card_next failed\n");
break;
}
}
psnd_pcm_info_free(pcminfo);
psnd_ctl_card_info_free(info);
}
+1371
View File
File diff suppressed because it is too large Load Diff
+706
View File
@@ -0,0 +1,706 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2007 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include "alMain.h"
#include "alu.h"
#include <CoreServices/CoreServices.h>
#include <unistd.h>
#include <AudioUnit/AudioUnit.h>
#include <AudioToolbox/AudioToolbox.h>
typedef struct {
AudioUnit audioUnit;
ALuint frameSize;
ALdouble sampleRateRatio; // Ratio of hardware sample rate / requested sample rate
AudioStreamBasicDescription format; // This is the OpenAL format as a CoreAudio ASBD
AudioConverterRef audioConverter; // Sample rate converter if needed
AudioBufferList *bufferList; // Buffer for data coming from the input device
ALCvoid *resampleBuffer; // Buffer for returned RingBuffer data when resampling
RingBuffer *ring;
} ca_data;
static const ALCchar ca_device[] = "CoreAudio Default";
static void destroy_buffer_list(AudioBufferList* list)
{
if(list)
{
UInt32 i;
for(i = 0;i < list->mNumberBuffers;i++)
free(list->mBuffers[i].mData);
free(list);
}
}
static AudioBufferList* allocate_buffer_list(UInt32 channelCount, UInt32 byteSize)
{
AudioBufferList *list;
list = calloc(1, sizeof(AudioBufferList) + sizeof(AudioBuffer));
if(list)
{
list->mNumberBuffers = 1;
list->mBuffers[0].mNumberChannels = channelCount;
list->mBuffers[0].mDataByteSize = byteSize;
list->mBuffers[0].mData = malloc(byteSize);
if(list->mBuffers[0].mData == NULL)
{
free(list);
list = NULL;
}
}
return list;
}
static OSStatus ca_callback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData)
{
ALCdevice *device = (ALCdevice*)inRefCon;
ca_data *data = (ca_data*)device->ExtraData;
aluMixData(device, ioData->mBuffers[0].mData,
ioData->mBuffers[0].mDataByteSize / data->frameSize);
return noErr;
}
static OSStatus ca_capture_conversion_callback(AudioConverterRef inAudioConverter, UInt32 *ioNumberDataPackets,
AudioBufferList *ioData, AudioStreamPacketDescription **outDataPacketDescription, void* inUserData)
{
ALCdevice *device = (ALCdevice*)inUserData;
ca_data *data = (ca_data*)device->ExtraData;
// Read from the ring buffer and store temporarily in a large buffer
ReadRingBuffer(data->ring, data->resampleBuffer, (ALsizei)(*ioNumberDataPackets));
// Set the input data
ioData->mNumberBuffers = 1;
ioData->mBuffers[0].mNumberChannels = data->format.mChannelsPerFrame;
ioData->mBuffers[0].mData = data->resampleBuffer;
ioData->mBuffers[0].mDataByteSize = (*ioNumberDataPackets) * data->format.mBytesPerFrame;
return noErr;
}
static OSStatus ca_capture_callback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber,
UInt32 inNumberFrames, AudioBufferList *ioData)
{
ALCdevice *device = (ALCdevice*)inRefCon;
ca_data *data = (ca_data*)device->ExtraData;
AudioUnitRenderActionFlags flags = 0;
OSStatus err;
// fill the bufferList with data from the input device
err = AudioUnitRender(data->audioUnit, &flags, inTimeStamp, 1, inNumberFrames, data->bufferList);
if(err != noErr)
{
ERR("AudioUnitRender error: %d\n", err);
return err;
}
WriteRingBuffer(data->ring, data->bufferList->mBuffers[0].mData, inNumberFrames);
return noErr;
}
static ALCenum ca_open_playback(ALCdevice *device, const ALCchar *deviceName)
{
ComponentDescription desc;
Component comp;
ca_data *data;
OSStatus err;
if(!deviceName)
deviceName = ca_device;
else if(strcmp(deviceName, ca_device) != 0)
return ALC_INVALID_VALUE;
/* open the default output unit */
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_DefaultOutput;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
comp = FindNextComponent(NULL, &desc);
if(comp == NULL)
{
ERR("FindNextComponent failed\n");
return ALC_INVALID_VALUE;
}
data = calloc(1, sizeof(*data));
err = OpenAComponent(comp, &data->audioUnit);
if(err != noErr)
{
ERR("OpenAComponent failed\n");
free(data);
return ALC_INVALID_VALUE;
}
/* init and start the default audio unit... */
err = AudioUnitInitialize(data->audioUnit);
if(err != noErr)
{
ERR("AudioUnitInitialize failed\n");
CloseComponent(data->audioUnit);
free(data);
return ALC_INVALID_VALUE;
}
device->DeviceName = strdup(deviceName);
device->ExtraData = data;
return ALC_NO_ERROR;
}
static void ca_close_playback(ALCdevice *device)
{
ca_data *data = (ca_data*)device->ExtraData;
AudioUnitUninitialize(data->audioUnit);
CloseComponent(data->audioUnit);
free(data);
device->ExtraData = NULL;
}
static ALCboolean ca_reset_playback(ALCdevice *device)
{
ca_data *data = (ca_data*)device->ExtraData;
AudioStreamBasicDescription streamFormat;
AURenderCallbackStruct input;
OSStatus err;
UInt32 size;
err = AudioUnitUninitialize(data->audioUnit);
if(err != noErr)
ERR("-- AudioUnitUninitialize failed.\n");
/* retrieve default output unit's properties (output side) */
size = sizeof(AudioStreamBasicDescription);
err = AudioUnitGetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &streamFormat, &size);
if(err != noErr || size != sizeof(AudioStreamBasicDescription))
{
ERR("AudioUnitGetProperty failed\n");
return ALC_FALSE;
}
#if 0
TRACE("Output streamFormat of default output unit -\n");
TRACE(" streamFormat.mFramesPerPacket = %d\n", streamFormat.mFramesPerPacket);
TRACE(" streamFormat.mChannelsPerFrame = %d\n", streamFormat.mChannelsPerFrame);
TRACE(" streamFormat.mBitsPerChannel = %d\n", streamFormat.mBitsPerChannel);
TRACE(" streamFormat.mBytesPerPacket = %d\n", streamFormat.mBytesPerPacket);
TRACE(" streamFormat.mBytesPerFrame = %d\n", streamFormat.mBytesPerFrame);
TRACE(" streamFormat.mSampleRate = %5.0f\n", streamFormat.mSampleRate);
#endif
/* set default output unit's input side to match output side */
err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, size);
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
return ALC_FALSE;
}
if(device->Frequency != streamFormat.mSampleRate)
{
device->UpdateSize = (ALuint)((ALuint64)device->UpdateSize *
streamFormat.mSampleRate /
device->Frequency);
device->Frequency = streamFormat.mSampleRate;
}
/* FIXME: How to tell what channels are what in the output device, and how
* to specify what we're giving? eg, 6.0 vs 5.1 */
switch(streamFormat.mChannelsPerFrame)
{
case 1:
device->FmtChans = DevFmtMono;
break;
case 2:
device->FmtChans = DevFmtStereo;
break;
case 4:
device->FmtChans = DevFmtQuad;
break;
case 6:
device->FmtChans = DevFmtX51;
break;
case 7:
device->FmtChans = DevFmtX61;
break;
case 8:
device->FmtChans = DevFmtX71;
break;
default:
ERR("Unhandled channel count (%d), using Stereo\n", streamFormat.mChannelsPerFrame);
device->FmtChans = DevFmtStereo;
streamFormat.mChannelsPerFrame = 2;
break;
}
SetDefaultWFXChannelOrder(device);
/* use channel count and sample rate from the default output unit's current
* parameters, but reset everything else */
streamFormat.mFramesPerPacket = 1;
streamFormat.mFormatFlags = 0;
switch(device->FmtType)
{
case DevFmtUByte:
device->FmtType = DevFmtByte;
/* fall-through */
case DevFmtByte:
streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
streamFormat.mBitsPerChannel = 8;
break;
case DevFmtUShort:
device->FmtType = DevFmtShort;
/* fall-through */
case DevFmtShort:
streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
streamFormat.mBitsPerChannel = 16;
break;
case DevFmtUInt:
device->FmtType = DevFmtInt;
/* fall-through */
case DevFmtInt:
streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
streamFormat.mBitsPerChannel = 32;
break;
case DevFmtFloat:
streamFormat.mFormatFlags = kLinearPCMFormatFlagIsFloat;
streamFormat.mBitsPerChannel = 32;
break;
}
streamFormat.mBytesPerFrame = streamFormat.mChannelsPerFrame *
streamFormat.mBitsPerChannel / 8;
streamFormat.mBytesPerPacket = streamFormat.mBytesPerFrame;
streamFormat.mFormatID = kAudioFormatLinearPCM;
streamFormat.mFormatFlags |= kAudioFormatFlagsNativeEndian |
kLinearPCMFormatFlagIsPacked;
err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, sizeof(AudioStreamBasicDescription));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
return ALC_FALSE;
}
/* setup callback */
data->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
input.inputProc = ca_callback;
input.inputProcRefCon = device;
err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &input, sizeof(AURenderCallbackStruct));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
return ALC_FALSE;
}
/* init the default audio unit... */
err = AudioUnitInitialize(data->audioUnit);
if(err != noErr)
{
ERR("AudioUnitInitialize failed\n");
return ALC_FALSE;
}
return ALC_TRUE;
}
static ALCboolean ca_start_playback(ALCdevice *device)
{
ca_data *data = (ca_data*)device->ExtraData;
OSStatus err;
err = AudioOutputUnitStart(data->audioUnit);
if(err != noErr)
{
ERR("AudioOutputUnitStart failed\n");
return ALC_FALSE;
}
return ALC_TRUE;
}
static void ca_stop_playback(ALCdevice *device)
{
ca_data *data = (ca_data*)device->ExtraData;
OSStatus err;
err = AudioOutputUnitStop(data->audioUnit);
if(err != noErr)
ERR("AudioOutputUnitStop failed\n");
}
static ALCenum ca_open_capture(ALCdevice *device, const ALCchar *deviceName)
{
AudioStreamBasicDescription requestedFormat; // The application requested format
AudioStreamBasicDescription hardwareFormat; // The hardware format
AudioStreamBasicDescription outputFormat; // The AudioUnit output format
AURenderCallbackStruct input;
ComponentDescription desc;
AudioDeviceID inputDevice;
UInt32 outputFrameCount;
UInt32 propertySize;
UInt32 enableIO;
Component comp;
ca_data *data;
OSStatus err;
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_HALOutput;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
// Search for component with given description
comp = FindNextComponent(NULL, &desc);
if(comp == NULL)
{
ERR("FindNextComponent failed\n");
return ALC_INVALID_VALUE;
}
data = calloc(1, sizeof(*data));
device->ExtraData = data;
// Open the component
err = OpenAComponent(comp, &data->audioUnit);
if(err != noErr)
{
ERR("OpenAComponent failed\n");
goto error;
}
// Turn off AudioUnit output
enableIO = 0;
err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &enableIO, sizeof(ALuint));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
goto error;
}
// Turn on AudioUnit input
enableIO = 1;
err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &enableIO, sizeof(ALuint));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
goto error;
}
// Get the default input device
propertySize = sizeof(AudioDeviceID);
err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice, &propertySize, &inputDevice);
if(err != noErr)
{
ERR("AudioHardwareGetProperty failed\n");
goto error;
}
if(inputDevice == kAudioDeviceUnknown)
{
ERR("No input device found\n");
goto error;
}
// Track the input device
err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDevice, sizeof(AudioDeviceID));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
goto error;
}
// set capture callback
input.inputProc = ca_capture_callback;
input.inputProcRefCon = device;
err = AudioUnitSetProperty(data->audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &input, sizeof(AURenderCallbackStruct));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
goto error;
}
// Initialize the device
err = AudioUnitInitialize(data->audioUnit);
if(err != noErr)
{
ERR("AudioUnitInitialize failed\n");
goto error;
}
// Get the hardware format
propertySize = sizeof(AudioStreamBasicDescription);
err = AudioUnitGetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &hardwareFormat, &propertySize);
if(err != noErr || propertySize != sizeof(AudioStreamBasicDescription))
{
ERR("AudioUnitGetProperty failed\n");
goto error;
}
// Set up the requested format description
switch(device->FmtType)
{
case DevFmtUByte:
requestedFormat.mBitsPerChannel = 8;
requestedFormat.mFormatFlags = kAudioFormatFlagIsPacked;
break;
case DevFmtShort:
requestedFormat.mBitsPerChannel = 16;
requestedFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
break;
case DevFmtInt:
requestedFormat.mBitsPerChannel = 32;
requestedFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
break;
case DevFmtFloat:
requestedFormat.mBitsPerChannel = 32;
requestedFormat.mFormatFlags = kAudioFormatFlagIsPacked;
break;
case DevFmtByte:
case DevFmtUShort:
case DevFmtUInt:
ERR("%s samples not supported\n", DevFmtTypeString(device->FmtType));
goto error;
}
switch(device->FmtChans)
{
case DevFmtMono:
requestedFormat.mChannelsPerFrame = 1;
break;
case DevFmtStereo:
requestedFormat.mChannelsPerFrame = 2;
break;
case DevFmtQuad:
case DevFmtX51:
case DevFmtX51Side:
case DevFmtX61:
case DevFmtX71:
ERR("%s not supported\n", DevFmtChannelsString(device->FmtChans));
goto error;
}
requestedFormat.mBytesPerFrame = requestedFormat.mChannelsPerFrame * requestedFormat.mBitsPerChannel / 8;
requestedFormat.mBytesPerPacket = requestedFormat.mBytesPerFrame;
requestedFormat.mSampleRate = device->Frequency;
requestedFormat.mFormatID = kAudioFormatLinearPCM;
requestedFormat.mReserved = 0;
requestedFormat.mFramesPerPacket = 1;
// save requested format description for later use
data->format = requestedFormat;
data->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
// Use intermediate format for sample rate conversion (outputFormat)
// Set sample rate to the same as hardware for resampling later
outputFormat = requestedFormat;
outputFormat.mSampleRate = hardwareFormat.mSampleRate;
// Determine sample rate ratio for resampling
data->sampleRateRatio = outputFormat.mSampleRate / device->Frequency;
// The output format should be the requested format, but using the hardware sample rate
// This is because the AudioUnit will automatically scale other properties, except for sample rate
err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, (void *)&outputFormat, sizeof(outputFormat));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
goto error;
}
// Set the AudioUnit output format frame count
outputFrameCount = device->UpdateSize * data->sampleRateRatio;
err = AudioUnitSetProperty(data->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Output, 0, &outputFrameCount, sizeof(outputFrameCount));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed: %d\n", err);
goto error;
}
// Set up sample converter
err = AudioConverterNew(&outputFormat, &requestedFormat, &data->audioConverter);
if(err != noErr)
{
ERR("AudioConverterNew failed: %d\n", err);
goto error;
}
// Create a buffer for use in the resample callback
data->resampleBuffer = malloc(device->UpdateSize * data->frameSize * data->sampleRateRatio);
// Allocate buffer for the AudioUnit output
data->bufferList = allocate_buffer_list(outputFormat.mChannelsPerFrame, device->UpdateSize * data->frameSize * data->sampleRateRatio);
if(data->bufferList == NULL)
goto error;
data->ring = CreateRingBuffer(data->frameSize, (device->UpdateSize * data->sampleRateRatio) * device->NumUpdates);
if(data->ring == NULL)
goto error;
return ALC_NO_ERROR;
error:
DestroyRingBuffer(data->ring);
free(data->resampleBuffer);
destroy_buffer_list(data->bufferList);
if(data->audioConverter)
AudioConverterDispose(data->audioConverter);
if(data->audioUnit)
CloseComponent(data->audioUnit);
free(data);
device->ExtraData = NULL;
return ALC_INVALID_VALUE;
}
static void ca_close_capture(ALCdevice *device)
{
ca_data *data = (ca_data*)device->ExtraData;
DestroyRingBuffer(data->ring);
free(data->resampleBuffer);
destroy_buffer_list(data->bufferList);
AudioConverterDispose(data->audioConverter);
CloseComponent(data->audioUnit);
free(data);
device->ExtraData = NULL;
}
static void ca_start_capture(ALCdevice *device)
{
ca_data *data = (ca_data*)device->ExtraData;
OSStatus err = AudioOutputUnitStart(data->audioUnit);
if(err != noErr)
ERR("AudioOutputUnitStart failed\n");
}
static void ca_stop_capture(ALCdevice *device)
{
ca_data *data = (ca_data*)device->ExtraData;
OSStatus err = AudioOutputUnitStop(data->audioUnit);
if(err != noErr)
ERR("AudioOutputUnitStop failed\n");
}
static ALCenum ca_capture_samples(ALCdevice *device, ALCvoid *buffer, ALCuint samples)
{
ca_data *data = (ca_data*)device->ExtraData;
AudioBufferList *list;
UInt32 frameCount;
OSStatus err;
// If no samples are requested, just return
if(samples == 0)
return ALC_NO_ERROR;
// Allocate a temporary AudioBufferList to use as the return resamples data
list = alloca(sizeof(AudioBufferList) + sizeof(AudioBuffer));
// Point the resampling buffer to the capture buffer
list->mNumberBuffers = 1;
list->mBuffers[0].mNumberChannels = data->format.mChannelsPerFrame;
list->mBuffers[0].mDataByteSize = samples * data->frameSize;
list->mBuffers[0].mData = buffer;
// Resample into another AudioBufferList
frameCount = samples;
err = AudioConverterFillComplexBuffer(data->audioConverter, ca_capture_conversion_callback,
device, &frameCount, list, NULL);
if(err != noErr)
{
ERR("AudioConverterFillComplexBuffer error: %d\n", err);
return ALC_INVALID_VALUE;
}
return ALC_NO_ERROR;
}
static ALCuint ca_available_samples(ALCdevice *device)
{
ca_data *data = device->ExtraData;
return RingBufferSize(data->ring) / data->sampleRateRatio;
}
static const BackendFuncs ca_funcs = {
ca_open_playback,
ca_close_playback,
ca_reset_playback,
ca_start_playback,
ca_stop_playback,
ca_open_capture,
ca_close_capture,
ca_start_capture,
ca_stop_capture,
ca_capture_samples,
ca_available_samples,
ALCdevice_LockDefault,
ALCdevice_UnlockDefault,
ALCdevice_GetLatencyDefault
};
ALCboolean alc_ca_init(BackendFuncs *func_list)
{
*func_list = ca_funcs;
return ALC_TRUE;
}
void alc_ca_deinit(void)
{
}
void alc_ca_probe(enum DevProbe type)
{
switch(type)
{
case ALL_DEVICE_PROBE:
AppendAllDevicesList(ca_device);
break;
case CAPTURE_DEVICE_PROBE:
AppendCaptureDeviceList(ca_device);
break;
}
}
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/**
* OpenAL cross platform audio library
* Copyright (C) 2011 by Chris Robinson
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdlib.h>
#include "alMain.h"
#include "alu.h"
static ALCenum loopback_open_playback(ALCdevice *device, const ALCchar *deviceName)
{
device->DeviceName = strdup(deviceName);
return ALC_NO_ERROR;
}
static void loopback_close_playback(ALCdevice *device)
{
(void)device;
}
static ALCboolean loopback_reset_playback(ALCdevice *device)
{
SetDefaultWFXChannelOrder(device);
return ALC_TRUE;
}
static ALCboolean loopback_start_playback(ALCdevice *device)
{
return ALC_TRUE;
(void)device;
}
static void loopback_stop_playback(ALCdevice *device)
{
(void)device;
}
static const BackendFuncs loopback_funcs = {
loopback_open_playback,
loopback_close_playback,
loopback_reset_playback,
loopback_start_playback,
loopback_stop_playback,
NULL,
NULL,
NULL,
NULL,
NULL,
NULL,
ALCdevice_LockDefault,
ALCdevice_UnlockDefault,
ALCdevice_GetLatencyDefault
};
ALCboolean alc_loopback_init(BackendFuncs *func_list)
{
*func_list = loopback_funcs;
return ALC_TRUE;
}
void alc_loopback_deinit(void)
{
}
void alc_loopback_probe(enum DevProbe type)
{
(void)type;
}
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/**
* OpenAL cross platform audio library
* Copyright (C) 2010 by Chris Robinson
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdlib.h>
#ifdef HAVE_WINDOWS_H
#include <windows.h>
#endif
#include "alMain.h"
#include "alu.h"
typedef struct {
volatile int killNow;
ALvoid *thread;
} null_data;
static const ALCchar nullDevice[] = "No Output";
static ALuint NullProc(ALvoid *ptr)
{
ALCdevice *Device = (ALCdevice*)ptr;
null_data *data = (null_data*)Device->ExtraData;
ALuint now, start;
ALuint64 avail, done;
const ALuint restTime = (ALuint64)Device->UpdateSize * 1000 /
Device->Frequency / 2;
done = 0;
start = timeGetTime();
while(!data->killNow && Device->Connected)
{
now = timeGetTime();
avail = (ALuint64)(now-start) * Device->Frequency / 1000;
if(avail < done)
{
/* Timer wrapped (50 days???). Add the remainder of the cycle to
* the available count and reset the number of samples done */
avail += ((ALuint64)1<<32)*Device->Frequency/1000 - done;
done = 0;
}
if(avail-done < Device->UpdateSize)
{
Sleep(restTime);
continue;
}
while(avail-done >= Device->UpdateSize)
{
aluMixData(Device, NULL, Device->UpdateSize);
done += Device->UpdateSize;
}
}
return 0;
}
static ALCenum null_open_playback(ALCdevice *device, const ALCchar *deviceName)
{
null_data *data;
if(!deviceName)
deviceName = nullDevice;
else if(strcmp(deviceName, nullDevice) != 0)
return ALC_INVALID_VALUE;
data = (null_data*)calloc(1, sizeof(*data));
device->DeviceName = strdup(deviceName);
device->ExtraData = data;
return ALC_NO_ERROR;
}
static void null_close_playback(ALCdevice *device)
{
null_data *data = (null_data*)device->ExtraData;
free(data);
device->ExtraData = NULL;
}
static ALCboolean null_reset_playback(ALCdevice *device)
{
SetDefaultWFXChannelOrder(device);
return ALC_TRUE;
}
static ALCboolean null_start_playback(ALCdevice *device)
{
null_data *data = (null_data*)device->ExtraData;
data->thread = StartThread(NullProc, device);
if(data->thread == NULL)
return ALC_FALSE;
return ALC_TRUE;
}
static void null_stop_playback(ALCdevice *device)
{
null_data *data = (null_data*)device->ExtraData;
if(!data->thread)
return;
data->killNow = 1;
StopThread(data->thread);
data->thread = NULL;
data->killNow = 0;
}
static const BackendFuncs null_funcs = {
null_open_playback,
null_close_playback,
null_reset_playback,
null_start_playback,
null_stop_playback,
NULL,
NULL,
NULL,
NULL,
NULL,
NULL,
ALCdevice_LockDefault,
ALCdevice_UnlockDefault,
ALCdevice_GetLatencyDefault
};
ALCboolean alc_null_init(BackendFuncs *func_list)
{
*func_list = null_funcs;
return ALC_TRUE;
}
void alc_null_deinit(void)
{
}
void alc_null_probe(enum DevProbe type)
{
switch(type)
{
case ALL_DEVICE_PROBE:
AppendAllDevicesList(nullDevice);
break;
case CAPTURE_DEVICE_PROBE:
break;
}
}
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/*
* Copyright (C) 2011 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
/* This is an OpenAL backend for Android using the native audio APIs based on
* OpenSL ES 1.0.1. It is based on source code for the native-audio sample app
* bundled with NDK.
*/
#include "config.h"
#include <stdlib.h>
#include "alMain.h"
#include "alu.h"
#include <SLES/OpenSLES.h>
#if 1
#include <SLES/OpenSLES_Android.h>
#else
extern SLAPIENTRY const SLInterfaceID SL_IID_ANDROIDSIMPLEBUFFERQUEUE;
struct SLAndroidSimpleBufferQueueItf_;
typedef const struct SLAndroidSimpleBufferQueueItf_ * const * SLAndroidSimpleBufferQueueItf;
typedef void (*slAndroidSimpleBufferQueueCallback)(SLAndroidSimpleBufferQueueItf caller, void *pContext);
typedef struct SLAndroidSimpleBufferQueueState_ {
SLuint32 count;
SLuint32 index;
} SLAndroidSimpleBufferQueueState;
struct SLAndroidSimpleBufferQueueItf_ {
SLresult (*Enqueue) (
SLAndroidSimpleBufferQueueItf self,
const void *pBuffer,
SLuint32 size
);
SLresult (*Clear) (
SLAndroidSimpleBufferQueueItf self
);
SLresult (*GetState) (
SLAndroidSimpleBufferQueueItf self,
SLAndroidSimpleBufferQueueState *pState
);
SLresult (*RegisterCallback) (
SLAndroidSimpleBufferQueueItf self,
slAndroidSimpleBufferQueueCallback callback,
void* pContext
);
};
#define SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE ((SLuint32) 0x800007BD)
typedef struct SLDataLocator_AndroidSimpleBufferQueue {
SLuint32 locatorType;
SLuint32 numBuffers;
} SLDataLocator_AndroidSimpleBufferQueue;
#endif
/* Helper macros */
#define SLObjectItf_Realize(a,b) ((*(a))->Realize((a),(b)))
#define SLObjectItf_GetInterface(a,b,c) ((*(a))->GetInterface((a),(b),(c)))
#define SLObjectItf_Destroy(a) ((*(a))->Destroy((a)))
#define SLEngineItf_CreateOutputMix(a,b,c,d,e) ((*(a))->CreateOutputMix((a),(b),(c),(d),(e)))
#define SLEngineItf_CreateAudioPlayer(a,b,c,d,e,f,g) ((*(a))->CreateAudioPlayer((a),(b),(c),(d),(e),(f),(g)))
#define SLPlayItf_SetPlayState(a,b) ((*(a))->SetPlayState((a),(b)))
typedef struct {
/* engine interfaces */
SLObjectItf engineObject;
SLEngineItf engine;
/* output mix interfaces */
SLObjectItf outputMix;
/* buffer queue player interfaces */
SLObjectItf bufferQueueObject;
void *buffer;
ALuint bufferSize;
ALuint frameSize;
} osl_data;
static const ALCchar opensl_device[] = "OpenSL";
static SLuint32 GetChannelMask(enum DevFmtChannels chans)
{
switch(chans)
{
case DevFmtMono: return SL_SPEAKER_FRONT_CENTER;
case DevFmtStereo: return SL_SPEAKER_FRONT_LEFT|SL_SPEAKER_FRONT_RIGHT;
case DevFmtQuad: return SL_SPEAKER_FRONT_LEFT|SL_SPEAKER_FRONT_RIGHT|
SL_SPEAKER_BACK_LEFT|SL_SPEAKER_BACK_RIGHT;
case DevFmtX51: return SL_SPEAKER_FRONT_LEFT|SL_SPEAKER_FRONT_RIGHT|
SL_SPEAKER_FRONT_CENTER|SL_SPEAKER_LOW_FREQUENCY|
SL_SPEAKER_BACK_LEFT|SL_SPEAKER_BACK_RIGHT;
case DevFmtX61: return SL_SPEAKER_FRONT_LEFT|SL_SPEAKER_FRONT_RIGHT|
SL_SPEAKER_FRONT_CENTER|SL_SPEAKER_LOW_FREQUENCY|
SL_SPEAKER_BACK_CENTER|
SL_SPEAKER_SIDE_LEFT|SL_SPEAKER_SIDE_RIGHT;
case DevFmtX71: return SL_SPEAKER_FRONT_LEFT|SL_SPEAKER_FRONT_RIGHT|
SL_SPEAKER_FRONT_CENTER|SL_SPEAKER_LOW_FREQUENCY|
SL_SPEAKER_BACK_LEFT|SL_SPEAKER_BACK_RIGHT|
SL_SPEAKER_SIDE_LEFT|SL_SPEAKER_SIDE_RIGHT;
case DevFmtX51Side: return SL_SPEAKER_FRONT_LEFT|SL_SPEAKER_FRONT_RIGHT|
SL_SPEAKER_FRONT_CENTER|SL_SPEAKER_LOW_FREQUENCY|
SL_SPEAKER_SIDE_LEFT|SL_SPEAKER_SIDE_RIGHT;
}
return 0;
}
static const char *res_str(SLresult result)
{
switch(result)
{
case SL_RESULT_SUCCESS: return "Success";
case SL_RESULT_PRECONDITIONS_VIOLATED: return "Preconditions violated";
case SL_RESULT_PARAMETER_INVALID: return "Parameter invalid";
case SL_RESULT_MEMORY_FAILURE: return "Memory failure";
case SL_RESULT_RESOURCE_ERROR: return "Resource error";
case SL_RESULT_RESOURCE_LOST: return "Resource lost";
case SL_RESULT_IO_ERROR: return "I/O error";
case SL_RESULT_BUFFER_INSUFFICIENT: return "Buffer insufficient";
case SL_RESULT_CONTENT_CORRUPTED: return "Content corrupted";
case SL_RESULT_CONTENT_UNSUPPORTED: return "Content unsupported";
case SL_RESULT_CONTENT_NOT_FOUND: return "Content not found";
case SL_RESULT_PERMISSION_DENIED: return "Permission denied";
case SL_RESULT_FEATURE_UNSUPPORTED: return "Feature unsupported";
case SL_RESULT_INTERNAL_ERROR: return "Internal error";
case SL_RESULT_UNKNOWN_ERROR: return "Unknown error";
case SL_RESULT_OPERATION_ABORTED: return "Operation aborted";
case SL_RESULT_CONTROL_LOST: return "Control lost";
case SL_RESULT_READONLY: return "ReadOnly";
case SL_RESULT_ENGINEOPTION_UNSUPPORTED: return "Engine option unsupported";
case SL_RESULT_SOURCE_SINK_INCOMPATIBLE: return "Source/Sink incompatible";
}
return "Unknown error code";
}
#define PRINTERR(x, s) do { \
if((x) != SL_RESULT_SUCCESS) \
ERR("%s: %s\n", (s), res_str((x))); \
} while(0)
/* this callback handler is called every time a buffer finishes playing */
static void opensl_callback(SLAndroidSimpleBufferQueueItf bq, void *context)
{
ALCdevice *Device = context;
osl_data *data = Device->ExtraData;
SLresult result;
aluMixData(Device, data->buffer, data->bufferSize/data->frameSize);
result = (*bq)->Enqueue(bq, data->buffer, data->bufferSize);
PRINTERR(result, "bq->Enqueue");
}
static ALCenum opensl_open_playback(ALCdevice *Device, const ALCchar *deviceName)
{
osl_data *data = NULL;
SLresult result;
if(!deviceName)
deviceName = opensl_device;
else if(strcmp(deviceName, opensl_device) != 0)
return ALC_INVALID_VALUE;
data = calloc(1, sizeof(*data));
if(!data)
return ALC_OUT_OF_MEMORY;
// create engine
result = slCreateEngine(&data->engineObject, 0, NULL, 0, NULL, NULL);
PRINTERR(result, "slCreateEngine");
if(SL_RESULT_SUCCESS == result)
{
result = SLObjectItf_Realize(data->engineObject, SL_BOOLEAN_FALSE);
PRINTERR(result, "engine->Realize");
}
if(SL_RESULT_SUCCESS == result)
{
result = SLObjectItf_GetInterface(data->engineObject, SL_IID_ENGINE, &data->engine);
PRINTERR(result, "engine->GetInterface");
}
if(SL_RESULT_SUCCESS == result)
{
result = SLEngineItf_CreateOutputMix(data->engine, &data->outputMix, 0, NULL, NULL);
PRINTERR(result, "engine->CreateOutputMix");
}
if(SL_RESULT_SUCCESS == result)
{
result = SLObjectItf_Realize(data->outputMix, SL_BOOLEAN_FALSE);
PRINTERR(result, "outputMix->Realize");
}
if(SL_RESULT_SUCCESS != result)
{
if(data->outputMix != NULL)
SLObjectItf_Destroy(data->outputMix);
data->outputMix = NULL;
if(data->engineObject != NULL)
SLObjectItf_Destroy(data->engineObject);
data->engineObject = NULL;
data->engine = NULL;
free(data);
return ALC_INVALID_VALUE;
}
Device->DeviceName = strdup(deviceName);
Device->ExtraData = data;
return ALC_NO_ERROR;
}
static void opensl_close_playback(ALCdevice *Device)
{
osl_data *data = Device->ExtraData;
if(data->bufferQueueObject != NULL)
SLObjectItf_Destroy(data->bufferQueueObject);
data->bufferQueueObject = NULL;
SLObjectItf_Destroy(data->outputMix);
data->outputMix = NULL;
SLObjectItf_Destroy(data->engineObject);
data->engineObject = NULL;
data->engine = NULL;
free(data);
Device->ExtraData = NULL;
}
static ALCboolean opensl_reset_playback(ALCdevice *Device)
{
osl_data *data = Device->ExtraData;
SLDataLocator_AndroidSimpleBufferQueue loc_bufq;
SLDataLocator_OutputMix loc_outmix;
SLDataFormat_PCM format_pcm;
SLDataSource audioSrc;
SLDataSink audioSnk;
SLInterfaceID id;
SLboolean req;
SLresult result;
Device->UpdateSize = (ALuint64)Device->UpdateSize * 44100 / Device->Frequency;
Device->UpdateSize = Device->UpdateSize * Device->NumUpdates / 2;
Device->NumUpdates = 2;
Device->Frequency = 44100;
Device->FmtChans = DevFmtStereo;
Device->FmtType = DevFmtShort;
SetDefaultWFXChannelOrder(Device);
id = SL_IID_ANDROIDSIMPLEBUFFERQUEUE;
req = SL_BOOLEAN_TRUE;
loc_bufq.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE;
loc_bufq.numBuffers = Device->NumUpdates;
format_pcm.formatType = SL_DATAFORMAT_PCM;
format_pcm.numChannels = ChannelsFromDevFmt(Device->FmtChans);
format_pcm.samplesPerSec = Device->Frequency * 1000;
format_pcm.bitsPerSample = BytesFromDevFmt(Device->FmtType) * 8;
format_pcm.containerSize = format_pcm.bitsPerSample;
format_pcm.channelMask = GetChannelMask(Device->FmtChans);
format_pcm.endianness = SL_BYTEORDER_NATIVE;
audioSrc.pLocator = &loc_bufq;
audioSrc.pFormat = &format_pcm;
loc_outmix.locatorType = SL_DATALOCATOR_OUTPUTMIX;
loc_outmix.outputMix = data->outputMix;
audioSnk.pLocator = &loc_outmix;
audioSnk.pFormat = NULL;
if(data->bufferQueueObject != NULL)
SLObjectItf_Destroy(data->bufferQueueObject);
data->bufferQueueObject = NULL;
result = SLEngineItf_CreateAudioPlayer(data->engine, &data->bufferQueueObject, &audioSrc, &audioSnk, 1, &id, &req);
PRINTERR(result, "engine->CreateAudioPlayer");
if(SL_RESULT_SUCCESS == result)
{
result = SLObjectItf_Realize(data->bufferQueueObject, SL_BOOLEAN_FALSE);
PRINTERR(result, "bufferQueue->Realize");
}
if(SL_RESULT_SUCCESS != result)
{
if(data->bufferQueueObject != NULL)
SLObjectItf_Destroy(data->bufferQueueObject);
data->bufferQueueObject = NULL;
return ALC_FALSE;
}
return ALC_TRUE;
}
static ALCboolean opensl_start_playback(ALCdevice *Device)
{
osl_data *data = Device->ExtraData;
SLAndroidSimpleBufferQueueItf bufferQueue;
SLPlayItf player;
SLresult result;
ALuint i;
result = SLObjectItf_GetInterface(data->bufferQueueObject, SL_IID_BUFFERQUEUE, &bufferQueue);
PRINTERR(result, "bufferQueue->GetInterface");
if(SL_RESULT_SUCCESS == result)
{
result = (*bufferQueue)->RegisterCallback(bufferQueue, opensl_callback, Device);
PRINTERR(result, "bufferQueue->RegisterCallback");
}
if(SL_RESULT_SUCCESS == result)
{
data->frameSize = FrameSizeFromDevFmt(Device->FmtChans, Device->FmtType);
data->bufferSize = Device->UpdateSize * data->frameSize;
data->buffer = calloc(1, data->bufferSize);
if(!data->buffer)
{
result = SL_RESULT_MEMORY_FAILURE;
PRINTERR(result, "calloc");
}
}
/* enqueue the first buffer to kick off the callbacks */
for(i = 0;i < Device->NumUpdates;i++)
{
if(SL_RESULT_SUCCESS == result)
{
result = (*bufferQueue)->Enqueue(bufferQueue, data->buffer, data->bufferSize);
PRINTERR(result, "bufferQueue->Enqueue");
}
}
if(SL_RESULT_SUCCESS == result)
{
result = SLObjectItf_GetInterface(data->bufferQueueObject, SL_IID_PLAY, &player);
PRINTERR(result, "bufferQueue->GetInterface");
}
if(SL_RESULT_SUCCESS == result)
{
result = SLPlayItf_SetPlayState(player, SL_PLAYSTATE_PLAYING);
PRINTERR(result, "player->SetPlayState");
}
if(SL_RESULT_SUCCESS != result)
{
if(data->bufferQueueObject != NULL)
SLObjectItf_Destroy(data->bufferQueueObject);
data->bufferQueueObject = NULL;
free(data->buffer);
data->buffer = NULL;
data->bufferSize = 0;
return ALC_FALSE;
}
return ALC_TRUE;
}
static void opensl_stop_playback(ALCdevice *Device)
{
osl_data *data = Device->ExtraData;
free(data->buffer);
data->buffer = NULL;
data->bufferSize = 0;
}
static const BackendFuncs opensl_funcs = {
opensl_open_playback,
opensl_close_playback,
opensl_reset_playback,
opensl_start_playback,
opensl_stop_playback,
NULL,
NULL,
NULL,
NULL,
NULL,
NULL,
ALCdevice_LockDefault,
ALCdevice_UnlockDefault,
ALCdevice_GetLatencyDefault
};
ALCboolean alc_opensl_init(BackendFuncs *func_list)
{
*func_list = opensl_funcs;
return ALC_TRUE;
}
void alc_opensl_deinit(void)
{
}
void alc_opensl_probe(enum DevProbe type)
{
switch(type)
{
case ALL_DEVICE_PROBE:
AppendAllDevicesList(opensl_device);
break;
case CAPTURE_DEVICE_PROBE:
break;
}
}
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/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2007 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <sys/ioctl.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <stdlib.h>
#include <stdio.h>
#include <memory.h>
#include <unistd.h>
#include <errno.h>
#include <math.h>
#include "alMain.h"
#include "alu.h"
#include <sys/soundcard.h>
/*
* The OSS documentation talks about SOUND_MIXER_READ, but the header
* only contains MIXER_READ. Play safe. Same for WRITE.
*/
#ifndef SOUND_MIXER_READ
#define SOUND_MIXER_READ MIXER_READ
#endif
#ifndef SOUND_MIXER_WRITE
#define SOUND_MIXER_WRITE MIXER_WRITE
#endif
static const ALCchar oss_device[] = "OSS Default";
static const char *oss_driver = "/dev/dsp";
static const char *oss_capture = "/dev/dsp";
typedef struct {
int fd;
volatile int killNow;
ALvoid *thread;
ALubyte *mix_data;
int data_size;
RingBuffer *ring;
int doCapture;
} oss_data;
static int log2i(ALCuint x)
{
int y = 0;
while (x > 1)
{
x >>= 1;
y++;
}
return y;
}
static ALuint OSSProc(ALvoid *ptr)
{
ALCdevice *Device = (ALCdevice*)ptr;
oss_data *data = (oss_data*)Device->ExtraData;
ALint frameSize;
ssize_t wrote;
SetRTPriority();
frameSize = FrameSizeFromDevFmt(Device->FmtChans, Device->FmtType);
while(!data->killNow && Device->Connected)
{
ALint len = data->data_size;
ALubyte *WritePtr = data->mix_data;
aluMixData(Device, WritePtr, len/frameSize);
while(len > 0 && !data->killNow)
{
wrote = write(data->fd, WritePtr, len);
if(wrote < 0)
{
if(errno != EAGAIN && errno != EWOULDBLOCK && errno != EINTR)
{
ERR("write failed: %s\n", strerror(errno));
ALCdevice_Lock(Device);
aluHandleDisconnect(Device);
ALCdevice_Unlock(Device);
break;
}
Sleep(1);
continue;
}
len -= wrote;
WritePtr += wrote;
}
}
return 0;
}
static ALuint OSSCaptureProc(ALvoid *ptr)
{
ALCdevice *Device = (ALCdevice*)ptr;
oss_data *data = (oss_data*)Device->ExtraData;
int frameSize;
int amt;
SetRTPriority();
frameSize = FrameSizeFromDevFmt(Device->FmtChans, Device->FmtType);
while(!data->killNow)
{
amt = read(data->fd, data->mix_data, data->data_size);
if(amt < 0)
{
ERR("read failed: %s\n", strerror(errno));
ALCdevice_Lock(Device);
aluHandleDisconnect(Device);
ALCdevice_Unlock(Device);
break;
}
if(amt == 0)
{
Sleep(1);
continue;
}
if(data->doCapture)
WriteRingBuffer(data->ring, data->mix_data, amt/frameSize);
}
return 0;
}
static ALCenum oss_open_playback(ALCdevice *device, const ALCchar *deviceName)
{
oss_data *data;
if(!deviceName)
deviceName = oss_device;
else if(strcmp(deviceName, oss_device) != 0)
return ALC_INVALID_VALUE;
data = (oss_data*)calloc(1, sizeof(oss_data));
data->killNow = 0;
data->fd = open(oss_driver, O_WRONLY);
if(data->fd == -1)
{
free(data);
ERR("Could not open %s: %s\n", oss_driver, strerror(errno));
return ALC_INVALID_VALUE;
}
device->DeviceName = strdup(deviceName);
device->ExtraData = data;
return ALC_NO_ERROR;
}
static void oss_close_playback(ALCdevice *device)
{
oss_data *data = (oss_data*)device->ExtraData;
close(data->fd);
free(data);
device->ExtraData = NULL;
}
static ALCboolean oss_reset_playback(ALCdevice *device)
{
oss_data *data = (oss_data*)device->ExtraData;
int numFragmentsLogSize;
int log2FragmentSize;
unsigned int periods;
audio_buf_info info;
ALuint frameSize;
int numChannels;
int ossFormat;
int ossSpeed;
char *err;
switch(device->FmtType)
{
case DevFmtByte:
ossFormat = AFMT_S8;
break;
case DevFmtUByte:
ossFormat = AFMT_U8;
break;
case DevFmtUShort:
case DevFmtInt:
case DevFmtUInt:
case DevFmtFloat:
device->FmtType = DevFmtShort;
/* fall-through */
case DevFmtShort:
ossFormat = AFMT_S16_NE;
break;
}
periods = device->NumUpdates;
numChannels = ChannelsFromDevFmt(device->FmtChans);
frameSize = numChannels * BytesFromDevFmt(device->FmtType);
ossSpeed = device->Frequency;
log2FragmentSize = log2i(device->UpdateSize * frameSize);
/* according to the OSS spec, 16 bytes are the minimum */
if (log2FragmentSize < 4)
log2FragmentSize = 4;
/* Subtract one period since the temp mixing buffer counts as one. Still
* need at least two on the card, though. */
if(periods > 2) periods--;
numFragmentsLogSize = (periods << 16) | log2FragmentSize;
#define CHECKERR(func) if((func) < 0) { \
err = #func; \
goto err; \
}
/* Don't fail if SETFRAGMENT fails. We can handle just about anything
* that's reported back via GETOSPACE */
ioctl(data->fd, SNDCTL_DSP_SETFRAGMENT, &numFragmentsLogSize);
CHECKERR(ioctl(data->fd, SNDCTL_DSP_SETFMT, &ossFormat));
CHECKERR(ioctl(data->fd, SNDCTL_DSP_CHANNELS, &numChannels));
CHECKERR(ioctl(data->fd, SNDCTL_DSP_SPEED, &ossSpeed));
CHECKERR(ioctl(data->fd, SNDCTL_DSP_GETOSPACE, &info));
if(0)
{
err:
ERR("%s failed: %s\n", err, strerror(errno));
return ALC_FALSE;
}
#undef CHECKERR
if((int)ChannelsFromDevFmt(device->FmtChans) != numChannels)
{
ERR("Failed to set %s, got %d channels instead\n", DevFmtChannelsString(device->FmtChans), numChannels);
return ALC_FALSE;
}
if(!((ossFormat == AFMT_S8 && device->FmtType == DevFmtByte) ||
(ossFormat == AFMT_U8 && device->FmtType == DevFmtUByte) ||
(ossFormat == AFMT_S16_NE && device->FmtType == DevFmtShort)))
{
ERR("Failed to set %s samples, got OSS format %#x\n", DevFmtTypeString(device->FmtType), ossFormat);
return ALC_FALSE;
}
device->Frequency = ossSpeed;
device->UpdateSize = info.fragsize / frameSize;
device->NumUpdates = info.fragments + 1;
SetDefaultChannelOrder(device);
return ALC_TRUE;
}
static ALCboolean oss_start_playback(ALCdevice *device)
{
oss_data *data = (oss_data*)device->ExtraData;
data->data_size = device->UpdateSize * FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
data->mix_data = calloc(1, data->data_size);
data->thread = StartThread(OSSProc, device);
if(data->thread == NULL)
{
free(data->mix_data);
data->mix_data = NULL;
return ALC_FALSE;
}
return ALC_TRUE;
}
static void oss_stop_playback(ALCdevice *device)
{
oss_data *data = (oss_data*)device->ExtraData;
if(!data->thread)
return;
data->killNow = 1;
StopThread(data->thread);
data->thread = NULL;
data->killNow = 0;
if(ioctl(data->fd, SNDCTL_DSP_RESET) != 0)
ERR("Error resetting device: %s\n", strerror(errno));
free(data->mix_data);
data->mix_data = NULL;
}
static ALCenum oss_open_capture(ALCdevice *device, const ALCchar *deviceName)
{
int numFragmentsLogSize;
int log2FragmentSize;
unsigned int periods;
audio_buf_info info;
ALuint frameSize;
int numChannels;
oss_data *data;
int ossFormat;
int ossSpeed;
char *err;
if(!deviceName)
deviceName = oss_device;
else if(strcmp(deviceName, oss_device) != 0)
return ALC_INVALID_VALUE;
data = (oss_data*)calloc(1, sizeof(oss_data));
data->killNow = 0;
data->fd = open(oss_capture, O_RDONLY);
if(data->fd == -1)
{
free(data);
ERR("Could not open %s: %s\n", oss_capture, strerror(errno));
return ALC_INVALID_VALUE;
}
switch(device->FmtType)
{
case DevFmtByte:
ossFormat = AFMT_S8;
break;
case DevFmtUByte:
ossFormat = AFMT_U8;
break;
case DevFmtShort:
ossFormat = AFMT_S16_NE;
break;
case DevFmtUShort:
case DevFmtInt:
case DevFmtUInt:
case DevFmtFloat:
free(data);
ERR("%s capture samples not supported\n", DevFmtTypeString(device->FmtType));
return ALC_INVALID_VALUE;
}
periods = 4;
numChannels = ChannelsFromDevFmt(device->FmtChans);
frameSize = numChannels * BytesFromDevFmt(device->FmtType);
ossSpeed = device->Frequency;
log2FragmentSize = log2i(device->UpdateSize * device->NumUpdates *
frameSize / periods);
/* according to the OSS spec, 16 bytes are the minimum */
if (log2FragmentSize < 4)
log2FragmentSize = 4;
numFragmentsLogSize = (periods << 16) | log2FragmentSize;
#define CHECKERR(func) if((func) < 0) { \
err = #func; \
goto err; \
}
CHECKERR(ioctl(data->fd, SNDCTL_DSP_SETFRAGMENT, &numFragmentsLogSize));
CHECKERR(ioctl(data->fd, SNDCTL_DSP_SETFMT, &ossFormat));
CHECKERR(ioctl(data->fd, SNDCTL_DSP_CHANNELS, &numChannels));
CHECKERR(ioctl(data->fd, SNDCTL_DSP_SPEED, &ossSpeed));
CHECKERR(ioctl(data->fd, SNDCTL_DSP_GETISPACE, &info));
if(0)
{
err:
ERR("%s failed: %s\n", err, strerror(errno));
close(data->fd);
free(data);
return ALC_INVALID_VALUE;
}
#undef CHECKERR
if((int)ChannelsFromDevFmt(device->FmtChans) != numChannels)
{
ERR("Failed to set %s, got %d channels instead\n", DevFmtChannelsString(device->FmtChans), numChannels);
close(data->fd);
free(data);
return ALC_INVALID_VALUE;
}
if(!((ossFormat == AFMT_S8 && device->FmtType == DevFmtByte) ||
(ossFormat == AFMT_U8 && device->FmtType == DevFmtUByte) ||
(ossFormat == AFMT_S16_NE && device->FmtType == DevFmtShort)))
{
ERR("Failed to set %s samples, got OSS format %#x\n", DevFmtTypeString(device->FmtType), ossFormat);
close(data->fd);
free(data);
return ALC_INVALID_VALUE;
}
data->ring = CreateRingBuffer(frameSize, device->UpdateSize * device->NumUpdates);
if(!data->ring)
{
ERR("Ring buffer create failed\n");
close(data->fd);
free(data);
return ALC_OUT_OF_MEMORY;
}
data->data_size = info.fragsize;
data->mix_data = calloc(1, data->data_size);
device->ExtraData = data;
data->thread = StartThread(OSSCaptureProc, device);
if(data->thread == NULL)
{
device->ExtraData = NULL;
free(data->mix_data);
free(data);
return ALC_OUT_OF_MEMORY;
}
device->DeviceName = strdup(deviceName);
return ALC_NO_ERROR;
}
static void oss_close_capture(ALCdevice *device)
{
oss_data *data = (oss_data*)device->ExtraData;
data->killNow = 1;
StopThread(data->thread);
close(data->fd);
DestroyRingBuffer(data->ring);
free(data->mix_data);
free(data);
device->ExtraData = NULL;
}
static void oss_start_capture(ALCdevice *Device)
{
oss_data *data = (oss_data*)Device->ExtraData;
data->doCapture = 1;
}
static void oss_stop_capture(ALCdevice *Device)
{
oss_data *data = (oss_data*)Device->ExtraData;
data->doCapture = 0;
}
static ALCenum oss_capture_samples(ALCdevice *Device, ALCvoid *pBuffer, ALCuint lSamples)
{
oss_data *data = (oss_data*)Device->ExtraData;
ReadRingBuffer(data->ring, pBuffer, lSamples);
return ALC_NO_ERROR;
}
static ALCuint oss_available_samples(ALCdevice *Device)
{
oss_data *data = (oss_data*)Device->ExtraData;
return RingBufferSize(data->ring);
}
static const BackendFuncs oss_funcs = {
oss_open_playback,
oss_close_playback,
oss_reset_playback,
oss_start_playback,
oss_stop_playback,
oss_open_capture,
oss_close_capture,
oss_start_capture,
oss_stop_capture,
oss_capture_samples,
oss_available_samples,
ALCdevice_LockDefault,
ALCdevice_UnlockDefault,
ALCdevice_GetLatencyDefault
};
ALCboolean alc_oss_init(BackendFuncs *func_list)
{
ConfigValueStr("oss", "device", &oss_driver);
ConfigValueStr("oss", "capture", &oss_capture);
*func_list = oss_funcs;
return ALC_TRUE;
}
void alc_oss_deinit(void)
{
}
void alc_oss_probe(enum DevProbe type)
{
switch(type)
{
case ALL_DEVICE_PROBE:
{
#ifdef HAVE_STAT
struct stat buf;
if(stat(oss_driver, &buf) == 0)
#endif
AppendAllDevicesList(oss_device);
}
break;
case CAPTURE_DEVICE_PROBE:
{
#ifdef HAVE_STAT
struct stat buf;
if(stat(oss_capture, &buf) == 0)
#endif
AppendCaptureDeviceList(oss_device);
}
break;
}
}
+472
View File
@@ -0,0 +1,472 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2007 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include "alMain.h"
#include "alu.h"
#include <portaudio.h>
static const ALCchar pa_device[] = "PortAudio Default";
#ifdef HAVE_DYNLOAD
static void *pa_handle;
#define MAKE_FUNC(x) static typeof(x) * p##x
MAKE_FUNC(Pa_Initialize);
MAKE_FUNC(Pa_Terminate);
MAKE_FUNC(Pa_GetErrorText);
MAKE_FUNC(Pa_StartStream);
MAKE_FUNC(Pa_StopStream);
MAKE_FUNC(Pa_OpenStream);
MAKE_FUNC(Pa_CloseStream);
MAKE_FUNC(Pa_GetDefaultOutputDevice);
MAKE_FUNC(Pa_GetStreamInfo);
#undef MAKE_FUNC
#define Pa_Initialize pPa_Initialize
#define Pa_Terminate pPa_Terminate
#define Pa_GetErrorText pPa_GetErrorText
#define Pa_StartStream pPa_StartStream
#define Pa_StopStream pPa_StopStream
#define Pa_OpenStream pPa_OpenStream
#define Pa_CloseStream pPa_CloseStream
#define Pa_GetDefaultOutputDevice pPa_GetDefaultOutputDevice
#define Pa_GetStreamInfo pPa_GetStreamInfo
#endif
static ALCboolean pa_load(void)
{
PaError err;
#ifdef HAVE_DYNLOAD
if(!pa_handle)
{
#ifdef _WIN32
# define PALIB "portaudio.dll"
#elif defined(__APPLE__) && defined(__MACH__)
# define PALIB "libportaudio.2.dylib"
#elif defined(__OpenBSD__)
# define PALIB "libportaudio.so"
#else
# define PALIB "libportaudio.so.2"
#endif
pa_handle = LoadLib(PALIB);
if(!pa_handle)
return ALC_FALSE;
#define LOAD_FUNC(f) do { \
p##f = GetSymbol(pa_handle, #f); \
if(p##f == NULL) \
{ \
CloseLib(pa_handle); \
pa_handle = NULL; \
return ALC_FALSE; \
} \
} while(0)
LOAD_FUNC(Pa_Initialize);
LOAD_FUNC(Pa_Terminate);
LOAD_FUNC(Pa_GetErrorText);
LOAD_FUNC(Pa_StartStream);
LOAD_FUNC(Pa_StopStream);
LOAD_FUNC(Pa_OpenStream);
LOAD_FUNC(Pa_CloseStream);
LOAD_FUNC(Pa_GetDefaultOutputDevice);
LOAD_FUNC(Pa_GetStreamInfo);
#undef LOAD_FUNC
if((err=Pa_Initialize()) != paNoError)
{
ERR("Pa_Initialize() returned an error: %s\n", Pa_GetErrorText(err));
CloseLib(pa_handle);
pa_handle = NULL;
return ALC_FALSE;
}
}
#else
if((err=Pa_Initialize()) != paNoError)
{
ERR("Pa_Initialize() returned an error: %s\n", Pa_GetErrorText(err));
return ALC_FALSE;
}
#endif
return ALC_TRUE;
}
typedef struct {
PaStream *stream;
PaStreamParameters params;
ALuint update_size;
RingBuffer *ring;
} pa_data;
static int pa_callback(const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo *timeInfo,
const PaStreamCallbackFlags statusFlags, void *userData)
{
ALCdevice *device = (ALCdevice*)userData;
(void)inputBuffer;
(void)timeInfo;
(void)statusFlags;
aluMixData(device, outputBuffer, framesPerBuffer);
return 0;
}
static int pa_capture_cb(const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo *timeInfo,
const PaStreamCallbackFlags statusFlags, void *userData)
{
ALCdevice *device = (ALCdevice*)userData;
pa_data *data = (pa_data*)device->ExtraData;
(void)outputBuffer;
(void)timeInfo;
(void)statusFlags;
WriteRingBuffer(data->ring, inputBuffer, framesPerBuffer);
return 0;
}
static ALCenum pa_open_playback(ALCdevice *device, const ALCchar *deviceName)
{
pa_data *data;
PaError err;
if(!deviceName)
deviceName = pa_device;
else if(strcmp(deviceName, pa_device) != 0)
return ALC_INVALID_VALUE;
data = (pa_data*)calloc(1, sizeof(pa_data));
data->update_size = device->UpdateSize;
data->params.device = -1;
if(!ConfigValueInt("port", "device", &data->params.device) ||
data->params.device < 0)
data->params.device = Pa_GetDefaultOutputDevice();
data->params.suggestedLatency = (device->UpdateSize*device->NumUpdates) /
(float)device->Frequency;
data->params.hostApiSpecificStreamInfo = NULL;
data->params.channelCount = ((device->FmtChans == DevFmtMono) ? 1 : 2);
switch(device->FmtType)
{
case DevFmtByte:
data->params.sampleFormat = paInt8;
break;
case DevFmtUByte:
data->params.sampleFormat = paUInt8;
break;
case DevFmtUShort:
/* fall-through */
case DevFmtShort:
data->params.sampleFormat = paInt16;
break;
case DevFmtUInt:
/* fall-through */
case DevFmtInt:
data->params.sampleFormat = paInt32;
break;
case DevFmtFloat:
data->params.sampleFormat = paFloat32;
break;
}
retry_open:
err = Pa_OpenStream(&data->stream, NULL, &data->params, device->Frequency,
device->UpdateSize, paNoFlag, pa_callback, device);
if(err != paNoError)
{
if(data->params.sampleFormat == paFloat32)
{
data->params.sampleFormat = paInt16;
goto retry_open;
}
ERR("Pa_OpenStream() returned an error: %s\n", Pa_GetErrorText(err));
free(data);
return ALC_INVALID_VALUE;
}
device->ExtraData = data;
device->DeviceName = strdup(deviceName);
return ALC_NO_ERROR;
}
static void pa_close_playback(ALCdevice *device)
{
pa_data *data = (pa_data*)device->ExtraData;
PaError err;
err = Pa_CloseStream(data->stream);
if(err != paNoError)
ERR("Error closing stream: %s\n", Pa_GetErrorText(err));
free(data);
device->ExtraData = NULL;
}
static ALCboolean pa_reset_playback(ALCdevice *device)
{
pa_data *data = (pa_data*)device->ExtraData;
const PaStreamInfo *streamInfo;
streamInfo = Pa_GetStreamInfo(data->stream);
device->Frequency = streamInfo->sampleRate;
device->UpdateSize = data->update_size;
if(data->params.sampleFormat == paInt8)
device->FmtType = DevFmtByte;
else if(data->params.sampleFormat == paUInt8)
device->FmtType = DevFmtUByte;
else if(data->params.sampleFormat == paInt16)
device->FmtType = DevFmtShort;
else if(data->params.sampleFormat == paInt32)
device->FmtType = DevFmtInt;
else if(data->params.sampleFormat == paFloat32)
device->FmtType = DevFmtFloat;
else
{
ERR("Unexpected sample format: 0x%lx\n", data->params.sampleFormat);
return ALC_FALSE;
}
if(data->params.channelCount == 2)
device->FmtChans = DevFmtStereo;
else if(data->params.channelCount == 1)
device->FmtChans = DevFmtMono;
else
{
ERR("Unexpected channel count: %u\n", data->params.channelCount);
return ALC_FALSE;
}
SetDefaultChannelOrder(device);
return ALC_TRUE;
}
static ALCboolean pa_start_playback(ALCdevice *device)
{
pa_data *data = (pa_data*)device->ExtraData;
PaError err;
err = Pa_StartStream(data->stream);
if(err != paNoError)
{
ERR("Pa_StartStream() returned an error: %s\n", Pa_GetErrorText(err));
return ALC_FALSE;
}
return ALC_TRUE;
}
static void pa_stop_playback(ALCdevice *device)
{
pa_data *data = (pa_data*)device->ExtraData;
PaError err;
err = Pa_StopStream(data->stream);
if(err != paNoError)
ERR("Error stopping stream: %s\n", Pa_GetErrorText(err));
}
static ALCenum pa_open_capture(ALCdevice *device, const ALCchar *deviceName)
{
ALuint frame_size;
pa_data *data;
PaError err;
if(!deviceName)
deviceName = pa_device;
else if(strcmp(deviceName, pa_device) != 0)
return ALC_INVALID_VALUE;
data = (pa_data*)calloc(1, sizeof(pa_data));
if(data == NULL)
return ALC_OUT_OF_MEMORY;
frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
data->ring = CreateRingBuffer(frame_size, device->UpdateSize*device->NumUpdates);
if(data->ring == NULL)
goto error;
data->params.device = -1;
if(!ConfigValueInt("port", "capture", &data->params.device) ||
data->params.device < 0)
data->params.device = Pa_GetDefaultOutputDevice();
data->params.suggestedLatency = 0.0f;
data->params.hostApiSpecificStreamInfo = NULL;
switch(device->FmtType)
{
case DevFmtByte:
data->params.sampleFormat = paInt8;
break;
case DevFmtUByte:
data->params.sampleFormat = paUInt8;
break;
case DevFmtShort:
data->params.sampleFormat = paInt16;
break;
case DevFmtInt:
data->params.sampleFormat = paInt32;
break;
case DevFmtFloat:
data->params.sampleFormat = paFloat32;
break;
case DevFmtUInt:
case DevFmtUShort:
ERR("%s samples not supported\n", DevFmtTypeString(device->FmtType));
goto error;
}
data->params.channelCount = ChannelsFromDevFmt(device->FmtChans);
err = Pa_OpenStream(&data->stream, &data->params, NULL, device->Frequency,
paFramesPerBufferUnspecified, paNoFlag, pa_capture_cb, device);
if(err != paNoError)
{
ERR("Pa_OpenStream() returned an error: %s\n", Pa_GetErrorText(err));
goto error;
}
device->DeviceName = strdup(deviceName);
device->ExtraData = data;
return ALC_NO_ERROR;
error:
DestroyRingBuffer(data->ring);
free(data);
return ALC_INVALID_VALUE;
}
static void pa_close_capture(ALCdevice *device)
{
pa_data *data = (pa_data*)device->ExtraData;
PaError err;
err = Pa_CloseStream(data->stream);
if(err != paNoError)
ERR("Error closing stream: %s\n", Pa_GetErrorText(err));
free(data);
device->ExtraData = NULL;
}
static void pa_start_capture(ALCdevice *device)
{
pa_data *data = device->ExtraData;
PaError err;
err = Pa_StartStream(data->stream);
if(err != paNoError)
ERR("Error starting stream: %s\n", Pa_GetErrorText(err));
}
static void pa_stop_capture(ALCdevice *device)
{
pa_data *data = (pa_data*)device->ExtraData;
PaError err;
err = Pa_StopStream(data->stream);
if(err != paNoError)
ERR("Error stopping stream: %s\n", Pa_GetErrorText(err));
}
static ALCenum pa_capture_samples(ALCdevice *device, ALCvoid *buffer, ALCuint samples)
{
pa_data *data = device->ExtraData;
ReadRingBuffer(data->ring, buffer, samples);
return ALC_NO_ERROR;
}
static ALCuint pa_available_samples(ALCdevice *device)
{
pa_data *data = device->ExtraData;
return RingBufferSize(data->ring);
}
static const BackendFuncs pa_funcs = {
pa_open_playback,
pa_close_playback,
pa_reset_playback,
pa_start_playback,
pa_stop_playback,
pa_open_capture,
pa_close_capture,
pa_start_capture,
pa_stop_capture,
pa_capture_samples,
pa_available_samples,
ALCdevice_LockDefault,
ALCdevice_UnlockDefault,
ALCdevice_GetLatencyDefault
};
ALCboolean alc_pa_init(BackendFuncs *func_list)
{
if(!pa_load())
return ALC_FALSE;
*func_list = pa_funcs;
return ALC_TRUE;
}
void alc_pa_deinit(void)
{
#ifdef HAVE_DYNLOAD
if(pa_handle)
{
Pa_Terminate();
CloseLib(pa_handle);
pa_handle = NULL;
}
#else
Pa_Terminate();
#endif
}
void alc_pa_probe(enum DevProbe type)
{
switch(type)
{
case ALL_DEVICE_PROBE:
AppendAllDevicesList(pa_device);
break;
case CAPTURE_DEVICE_PROBE:
AppendCaptureDeviceList(pa_device);
break;
}
}
File diff suppressed because it is too large Load Diff
+296
View File
@@ -0,0 +1,296 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2007 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include "alMain.h"
#include "alu.h"
#include <sndio.h>
static const ALCchar sndio_device[] = "SndIO Default";
static ALCboolean sndio_load(void)
{
return ALC_TRUE;
}
typedef struct {
struct sio_hdl *sndHandle;
ALvoid *mix_data;
ALsizei data_size;
volatile int killNow;
ALvoid *thread;
} sndio_data;
static ALuint sndio_proc(ALvoid *ptr)
{
ALCdevice *device = ptr;
sndio_data *data = device->ExtraData;
ALsizei frameSize;
size_t wrote;
SetRTPriority();
frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
while(!data->killNow && device->Connected)
{
ALsizei len = data->data_size;
ALubyte *WritePtr = data->mix_data;
aluMixData(device, WritePtr, len/frameSize);
while(len > 0 && !data->killNow)
{
wrote = sio_write(data->sndHandle, WritePtr, len);
if(wrote == 0)
{
ERR("sio_write failed\n");
ALCdevice_Lock(device);
aluHandleDisconnect(device);
ALCdevice_Unlock(device);
break;
}
len -= wrote;
WritePtr += wrote;
}
}
return 0;
}
static ALCenum sndio_open_playback(ALCdevice *device, const ALCchar *deviceName)
{
sndio_data *data;
if(!deviceName)
deviceName = sndio_device;
else if(strcmp(deviceName, sndio_device) != 0)
return ALC_INVALID_VALUE;
data = calloc(1, sizeof(*data));
data->killNow = 0;
data->sndHandle = sio_open(NULL, SIO_PLAY, 0);
if(data->sndHandle == NULL)
{
free(data);
ERR("Could not open device\n");
return ALC_INVALID_VALUE;
}
device->DeviceName = strdup(deviceName);
device->ExtraData = data;
return ALC_NO_ERROR;
}
static void sndio_close_playback(ALCdevice *device)
{
sndio_data *data = device->ExtraData;
sio_close(data->sndHandle);
free(data);
device->ExtraData = NULL;
}
static ALCboolean sndio_reset_playback(ALCdevice *device)
{
sndio_data *data = device->ExtraData;
struct sio_par par;
sio_initpar(&par);
par.rate = device->Frequency;
par.pchan = ((device->FmtChans != DevFmtMono) ? 2 : 1);
switch(device->FmtType)
{
case DevFmtByte:
par.bits = 8;
par.sig = 1;
break;
case DevFmtUByte:
par.bits = 8;
par.sig = 0;
break;
case DevFmtFloat:
case DevFmtShort:
par.bits = 16;
par.sig = 1;
break;
case DevFmtUShort:
par.bits = 16;
par.sig = 0;
break;
case DevFmtInt:
par.bits = 32;
par.sig = 1;
break;
case DevFmtUInt:
par.bits = 32;
par.sig = 0;
break;
}
par.le = SIO_LE_NATIVE;
par.round = device->UpdateSize;
par.appbufsz = device->UpdateSize * (device->NumUpdates-1);
if(!par.appbufsz) par.appbufsz = device->UpdateSize;
if(!sio_setpar(data->sndHandle, &par) || !sio_getpar(data->sndHandle, &par))
{
ERR("Failed to set device parameters\n");
return ALC_FALSE;
}
if(par.bits != par.bps*8)
{
ERR("Padded samples not supported (%u of %u bits)\n", par.bits, par.bps*8);
return ALC_FALSE;
}
device->Frequency = par.rate;
device->FmtChans = ((par.pchan==1) ? DevFmtMono : DevFmtStereo);
if(par.bits == 8 && par.sig == 1)
device->FmtType = DevFmtByte;
else if(par.bits == 8 && par.sig == 0)
device->FmtType = DevFmtUByte;
else if(par.bits == 16 && par.sig == 1)
device->FmtType = DevFmtShort;
else if(par.bits == 16 && par.sig == 0)
device->FmtType = DevFmtUShort;
else if(par.bits == 32 && par.sig == 1)
device->FmtType = DevFmtInt;
else if(par.bits == 32 && par.sig == 0)
device->FmtType = DevFmtUInt;
else
{
ERR("Unhandled sample format: %s %u-bit\n", (par.sig?"signed":"unsigned"), par.bits);
return ALC_FALSE;
}
device->UpdateSize = par.round;
device->NumUpdates = (par.bufsz/par.round) + 1;
SetDefaultChannelOrder(device);
return ALC_TRUE;
}
static ALCboolean sndio_start_playback(ALCdevice *device)
{
sndio_data *data = device->ExtraData;
if(!sio_start(data->sndHandle))
{
ERR("Error starting playback\n");
return ALC_FALSE;
}
data->data_size = device->UpdateSize * FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
data->mix_data = calloc(1, data->data_size);
data->thread = StartThread(sndio_proc, device);
if(data->thread == NULL)
{
sio_stop(data->sndHandle);
free(data->mix_data);
data->mix_data = NULL;
return ALC_FALSE;
}
return ALC_TRUE;
}
static void sndio_stop_playback(ALCdevice *device)
{
sndio_data *data = device->ExtraData;
if(!data->thread)
return;
data->killNow = 1;
StopThread(data->thread);
data->thread = NULL;
data->killNow = 0;
if(!sio_stop(data->sndHandle))
ERR("Error stopping device\n");
free(data->mix_data);
data->mix_data = NULL;
}
static const BackendFuncs sndio_funcs = {
sndio_open_playback,
sndio_close_playback,
sndio_reset_playback,
sndio_start_playback,
sndio_stop_playback,
NULL,
NULL,
NULL,
NULL,
NULL,
NULL,
ALCdevice_LockDefault,
ALCdevice_UnlockDefault,
ALCdevice_GetLatencyDefault
};
ALCboolean alc_sndio_init(BackendFuncs *func_list)
{
if(!sndio_load())
return ALC_FALSE;
*func_list = sndio_funcs;
return ALC_TRUE;
}
void alc_sndio_deinit(void)
{
}
void alc_sndio_probe(enum DevProbe type)
{
switch(type)
{
case ALL_DEVICE_PROBE:
AppendAllDevicesList(sndio_device);
break;
case CAPTURE_DEVICE_PROBE:
break;
}
}
+287
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@@ -0,0 +1,287 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2007 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <sys/ioctl.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <stdlib.h>
#include <stdio.h>
#include <memory.h>
#include <unistd.h>
#include <errno.h>
#include <math.h>
#include "alMain.h"
#include "alu.h"
#include <sys/audioio.h>
static const ALCchar solaris_device[] = "Solaris Default";
static const char *solaris_driver = "/dev/audio";
typedef struct {
int fd;
volatile int killNow;
ALvoid *thread;
ALubyte *mix_data;
int data_size;
} solaris_data;
static ALuint SolarisProc(ALvoid *ptr)
{
ALCdevice *Device = (ALCdevice*)ptr;
solaris_data *data = (solaris_data*)Device->ExtraData;
ALint frameSize;
int wrote;
SetRTPriority();
frameSize = FrameSizeFromDevFmt(Device->FmtChans, Device->FmtType);
while(!data->killNow && Device->Connected)
{
ALint len = data->data_size;
ALubyte *WritePtr = data->mix_data;
aluMixData(Device, WritePtr, len/frameSize);
while(len > 0 && !data->killNow)
{
wrote = write(data->fd, WritePtr, len);
if(wrote < 0)
{
if(errno != EAGAIN && errno != EWOULDBLOCK && errno != EINTR)
{
ERR("write failed: %s\n", strerror(errno));
ALCdevice_Lock(Device);
aluHandleDisconnect(Device);
ALCdevice_Unlock(Device);
break;
}
Sleep(1);
continue;
}
len -= wrote;
WritePtr += wrote;
}
}
return 0;
}
static ALCenum solaris_open_playback(ALCdevice *device, const ALCchar *deviceName)
{
solaris_data *data;
if(!deviceName)
deviceName = solaris_device;
else if(strcmp(deviceName, solaris_device) != 0)
return ALC_INVALID_VALUE;
data = (solaris_data*)calloc(1, sizeof(solaris_data));
data->killNow = 0;
data->fd = open(solaris_driver, O_WRONLY);
if(data->fd == -1)
{
free(data);
ERR("Could not open %s: %s\n", solaris_driver, strerror(errno));
return ALC_INVALID_VALUE;
}
device->DeviceName = strdup(deviceName);
device->ExtraData = data;
return ALC_NO_ERROR;
}
static void solaris_close_playback(ALCdevice *device)
{
solaris_data *data = (solaris_data*)device->ExtraData;
close(data->fd);
free(data);
device->ExtraData = NULL;
}
static ALCboolean solaris_reset_playback(ALCdevice *device)
{
solaris_data *data = (solaris_data*)device->ExtraData;
audio_info_t info;
ALuint frameSize;
int numChannels;
AUDIO_INITINFO(&info);
info.play.sample_rate = device->Frequency;
if(device->FmtChans != DevFmtMono)
device->FmtChans = DevFmtStereo;
numChannels = ChannelsFromDevFmt(device->FmtChans);
info.play.channels = numChannels;
switch(device->FmtType)
{
case DevFmtByte:
info.play.precision = 8;
info.play.encoding = AUDIO_ENCODING_LINEAR;
break;
case DevFmtUByte:
info.play.precision = 8;
info.play.encoding = AUDIO_ENCODING_LINEAR8;
break;
case DevFmtUShort:
case DevFmtInt:
case DevFmtUInt:
case DevFmtFloat:
device->FmtType = DevFmtShort;
/* fall-through */
case DevFmtShort:
info.play.precision = 16;
info.play.encoding = AUDIO_ENCODING_LINEAR;
break;
}
frameSize = numChannels * BytesFromDevFmt(device->FmtType);
info.play.buffer_size = device->UpdateSize*device->NumUpdates * frameSize;
if(ioctl(data->fd, AUDIO_SETINFO, &info) < 0)
{
ERR("ioctl failed: %s\n", strerror(errno));
return ALC_FALSE;
}
if(ChannelsFromDevFmt(device->FmtChans) != info.play.channels)
{
ERR("Could not set %d channels, got %d instead\n", ChannelsFromDevFmt(device->FmtChans), info.play.channels);
return ALC_FALSE;
}
if(!((info.play.precision == 8 && info.play.encoding == AUDIO_ENCODING_LINEAR8 && device->FmtType == DevFmtUByte) ||
(info.play.precision == 8 && info.play.encoding == AUDIO_ENCODING_LINEAR && device->FmtType == DevFmtByte) ||
(info.play.precision == 16 && info.play.encoding == AUDIO_ENCODING_LINEAR && device->FmtType == DevFmtShort) ||
(info.play.precision == 32 && info.play.encoding == AUDIO_ENCODING_LINEAR && device->FmtType == DevFmtInt)))
{
ERR("Could not set %s samples, got %d (0x%x)\n", DevFmtTypeString(device->FmtType),
info.play.precision, info.play.encoding);
return ALC_FALSE;
}
device->Frequency = info.play.sample_rate;
device->UpdateSize = (info.play.buffer_size/device->NumUpdates) + 1;
SetDefaultChannelOrder(device);
return ALC_TRUE;
}
static ALCboolean solaris_start_playback(ALCdevice *device)
{
solaris_data *data = (solaris_data*)device->ExtraData;
data->data_size = device->UpdateSize * FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
data->mix_data = calloc(1, data->data_size);
data->thread = StartThread(SolarisProc, device);
if(data->thread == NULL)
{
free(data->mix_data);
data->mix_data = NULL;
return ALC_FALSE;
}
return ALC_TRUE;
}
static void solaris_stop_playback(ALCdevice *device)
{
solaris_data *data = (solaris_data*)device->ExtraData;
if(!data->thread)
return;
data->killNow = 1;
StopThread(data->thread);
data->thread = NULL;
data->killNow = 0;
if(ioctl(data->fd, AUDIO_DRAIN) < 0)
ERR("Error draining device: %s\n", strerror(errno));
free(data->mix_data);
data->mix_data = NULL;
}
static const BackendFuncs solaris_funcs = {
solaris_open_playback,
solaris_close_playback,
solaris_reset_playback,
solaris_start_playback,
solaris_stop_playback,
NULL,
NULL,
NULL,
NULL,
NULL,
NULL,
ALCdevice_LockDefault,
ALCdevice_UnlockDefault,
ALCdevice_GetLatencyDefault
};
ALCboolean alc_solaris_init(BackendFuncs *func_list)
{
ConfigValueStr("solaris", "device", &solaris_driver);
*func_list = solaris_funcs;
return ALC_TRUE;
}
void alc_solaris_deinit(void)
{
}
void alc_solaris_probe(enum DevProbe type)
{
switch(type)
{
case ALL_DEVICE_PROBE:
{
#ifdef HAVE_STAT
struct stat buf;
if(stat(solaris_driver, &buf) == 0)
#endif
AppendAllDevicesList(solaris_device);
}
break;
case CAPTURE_DEVICE_PROBE:
break;
}
}
+370
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/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2007 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdlib.h>
#include <stdio.h>
#include <memory.h>
#ifdef HAVE_WINDOWS_H
#include <windows.h>
#endif
#include "alMain.h"
#include "alu.h"
typedef struct {
FILE *f;
long DataStart;
ALvoid *buffer;
ALuint size;
volatile int killNow;
ALvoid *thread;
} wave_data;
static const ALCchar waveDevice[] = "Wave File Writer";
static const ALubyte SUBTYPE_PCM[] = {
0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x10, 0x00, 0x80, 0x00, 0x00, 0xaa,
0x00, 0x38, 0x9b, 0x71
};
static const ALubyte SUBTYPE_FLOAT[] = {
0x03, 0x00, 0x00, 0x00, 0x00, 0x00, 0x10, 0x00, 0x80, 0x00, 0x00, 0xaa,
0x00, 0x38, 0x9b, 0x71
};
static const ALuint channel_masks[] = {
0, /* invalid */
0x4, /* Mono */
0x1 | 0x2, /* Stereo */
0, /* 3 channel */
0x1 | 0x2 | 0x10 | 0x20, /* Quad */
0, /* 5 channel */
0x1 | 0x2 | 0x4 | 0x8 | 0x10 | 0x20, /* 5.1 */
0x1 | 0x2 | 0x4 | 0x8 | 0x100 | 0x200 | 0x400, /* 6.1 */
0x1 | 0x2 | 0x4 | 0x8 | 0x10 | 0x20 | 0x200 | 0x400, /* 7.1 */
};
static void fwrite16le(ALushort val, FILE *f)
{
fputc(val&0xff, f);
fputc((val>>8)&0xff, f);
}
static void fwrite32le(ALuint val, FILE *f)
{
fputc(val&0xff, f);
fputc((val>>8)&0xff, f);
fputc((val>>16)&0xff, f);
fputc((val>>24)&0xff, f);
}
static ALuint WaveProc(ALvoid *ptr)
{
ALCdevice *Device = (ALCdevice*)ptr;
wave_data *data = (wave_data*)Device->ExtraData;
ALuint frameSize;
ALuint now, start;
ALuint64 avail, done;
size_t fs;
const ALuint restTime = (ALuint64)Device->UpdateSize * 1000 /
Device->Frequency / 2;
frameSize = FrameSizeFromDevFmt(Device->FmtChans, Device->FmtType);
done = 0;
start = timeGetTime();
while(!data->killNow && Device->Connected)
{
now = timeGetTime();
avail = (ALuint64)(now-start) * Device->Frequency / 1000;
if(avail < done)
{
/* Timer wrapped (50 days???). Add the remainder of the cycle to
* the available count and reset the number of samples done */
avail += ((ALuint64)1<<32)*Device->Frequency/1000 - done;
done = 0;
}
if(avail-done < Device->UpdateSize)
{
Sleep(restTime);
continue;
}
while(avail-done >= Device->UpdateSize)
{
aluMixData(Device, data->buffer, Device->UpdateSize);
done += Device->UpdateSize;
if(!IS_LITTLE_ENDIAN)
{
ALuint bytesize = BytesFromDevFmt(Device->FmtType);
ALubyte *bytes = data->buffer;
ALuint i;
if(bytesize == 1)
{
for(i = 0;i < data->size;i++)
fputc(bytes[i], data->f);
}
else if(bytesize == 2)
{
for(i = 0;i < data->size;i++)
fputc(bytes[i^1], data->f);
}
else if(bytesize == 4)
{
for(i = 0;i < data->size;i++)
fputc(bytes[i^3], data->f);
}
}
else
{
fs = fwrite(data->buffer, frameSize, Device->UpdateSize,
data->f);
fs = fs;
}
if(ferror(data->f))
{
ERR("Error writing to file\n");
ALCdevice_Lock(Device);
aluHandleDisconnect(Device);
ALCdevice_Unlock(Device);
break;
}
}
}
return 0;
}
static ALCenum wave_open_playback(ALCdevice *device, const ALCchar *deviceName)
{
wave_data *data;
const char *fname;
fname = GetConfigValue("wave", "file", "");
if(!fname[0])
return ALC_INVALID_VALUE;
if(!deviceName)
deviceName = waveDevice;
else if(strcmp(deviceName, waveDevice) != 0)
return ALC_INVALID_VALUE;
data = (wave_data*)calloc(1, sizeof(wave_data));
data->f = fopen(fname, "wb");
if(!data->f)
{
free(data);
ERR("Could not open file '%s': %s\n", fname, strerror(errno));
return ALC_INVALID_VALUE;
}
device->DeviceName = strdup(deviceName);
device->ExtraData = data;
return ALC_NO_ERROR;
}
static void wave_close_playback(ALCdevice *device)
{
wave_data *data = (wave_data*)device->ExtraData;
fclose(data->f);
free(data);
device->ExtraData = NULL;
}
static ALCboolean wave_reset_playback(ALCdevice *device)
{
wave_data *data = (wave_data*)device->ExtraData;
ALuint channels=0, bits=0;
size_t val;
fseek(data->f, 0, SEEK_SET);
clearerr(data->f);
switch(device->FmtType)
{
case DevFmtByte:
device->FmtType = DevFmtUByte;
break;
case DevFmtUShort:
device->FmtType = DevFmtShort;
break;
case DevFmtUInt:
device->FmtType = DevFmtInt;
break;
case DevFmtUByte:
case DevFmtShort:
case DevFmtInt:
case DevFmtFloat:
break;
}
bits = BytesFromDevFmt(device->FmtType) * 8;
channels = ChannelsFromDevFmt(device->FmtChans);
fprintf(data->f, "RIFF");
fwrite32le(0xFFFFFFFF, data->f); // 'RIFF' header len; filled in at close
fprintf(data->f, "WAVE");
fprintf(data->f, "fmt ");
fwrite32le(40, data->f); // 'fmt ' header len; 40 bytes for EXTENSIBLE
// 16-bit val, format type id (extensible: 0xFFFE)
fwrite16le(0xFFFE, data->f);
// 16-bit val, channel count
fwrite16le(channels, data->f);
// 32-bit val, frequency
fwrite32le(device->Frequency, data->f);
// 32-bit val, bytes per second
fwrite32le(device->Frequency * channels * bits / 8, data->f);
// 16-bit val, frame size
fwrite16le(channels * bits / 8, data->f);
// 16-bit val, bits per sample
fwrite16le(bits, data->f);
// 16-bit val, extra byte count
fwrite16le(22, data->f);
// 16-bit val, valid bits per sample
fwrite16le(bits, data->f);
// 32-bit val, channel mask
fwrite32le(channel_masks[channels], data->f);
// 16 byte GUID, sub-type format
val = fwrite(((bits==32) ? SUBTYPE_FLOAT : SUBTYPE_PCM), 1, 16, data->f);
val = val;
fprintf(data->f, "data");
fwrite32le(0xFFFFFFFF, data->f); // 'data' header len; filled in at close
if(ferror(data->f))
{
ERR("Error writing header: %s\n", strerror(errno));
return ALC_FALSE;
}
data->DataStart = ftell(data->f);
SetDefaultWFXChannelOrder(device);
return ALC_TRUE;
}
static ALCboolean wave_start_playback(ALCdevice *device)
{
wave_data *data = (wave_data*)device->ExtraData;
data->size = device->UpdateSize * FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
data->buffer = malloc(data->size);
if(!data->buffer)
{
ERR("Buffer malloc failed\n");
return ALC_FALSE;
}
data->thread = StartThread(WaveProc, device);
if(data->thread == NULL)
{
free(data->buffer);
data->buffer = NULL;
return ALC_FALSE;
}
return ALC_TRUE;
}
static void wave_stop_playback(ALCdevice *device)
{
wave_data *data = (wave_data*)device->ExtraData;
ALuint dataLen;
long size;
if(!data->thread)
return;
data->killNow = 1;
StopThread(data->thread);
data->thread = NULL;
data->killNow = 0;
free(data->buffer);
data->buffer = NULL;
size = ftell(data->f);
if(size > 0)
{
dataLen = size - data->DataStart;
if(fseek(data->f, data->DataStart-4, SEEK_SET) == 0)
fwrite32le(dataLen, data->f); // 'data' header len
if(fseek(data->f, 4, SEEK_SET) == 0)
fwrite32le(size-8, data->f); // 'WAVE' header len
}
}
static const BackendFuncs wave_funcs = {
wave_open_playback,
wave_close_playback,
wave_reset_playback,
wave_start_playback,
wave_stop_playback,
NULL,
NULL,
NULL,
NULL,
NULL,
NULL,
ALCdevice_LockDefault,
ALCdevice_UnlockDefault,
ALCdevice_GetLatencyDefault
};
ALCboolean alc_wave_init(BackendFuncs *func_list)
{
*func_list = wave_funcs;
return ALC_TRUE;
}
void alc_wave_deinit(void)
{
}
void alc_wave_probe(enum DevProbe type)
{
if(!ConfigValueExists("wave", "file"))
return;
switch(type)
{
case ALL_DEVICE_PROBE:
AppendAllDevicesList(waveDevice);
break;
case CAPTURE_DEVICE_PROBE:
break;
}
}
+774
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/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2007 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdlib.h>
#include <stdio.h>
#include <memory.h>
#include <windows.h>
#include <mmsystem.h>
#include "alMain.h"
#include "alu.h"
#ifndef WAVE_FORMAT_IEEE_FLOAT
#define WAVE_FORMAT_IEEE_FLOAT 0x0003
#endif
typedef struct {
// MMSYSTEM Device
volatile ALboolean killNow;
HANDLE WaveThreadEvent;
HANDLE WaveThread;
DWORD WaveThreadID;
volatile LONG WaveBuffersCommitted;
WAVEHDR WaveBuffer[4];
union {
HWAVEIN In;
HWAVEOUT Out;
} WaveHandle;
WAVEFORMATEX Format;
RingBuffer *Ring;
} WinMMData;
static ALCchar **PlaybackDeviceList;
static ALuint NumPlaybackDevices;
static ALCchar **CaptureDeviceList;
static ALuint NumCaptureDevices;
static void ProbePlaybackDevices(void)
{
ALuint i;
for(i = 0;i < NumPlaybackDevices;i++)
free(PlaybackDeviceList[i]);
NumPlaybackDevices = waveOutGetNumDevs();
PlaybackDeviceList = realloc(PlaybackDeviceList, sizeof(ALCchar*) * NumPlaybackDevices);
for(i = 0;i < NumPlaybackDevices;i++)
{
WAVEOUTCAPS WaveCaps;
PlaybackDeviceList[i] = NULL;
if(waveOutGetDevCaps(i, &WaveCaps, sizeof(WaveCaps)) == MMSYSERR_NOERROR)
{
char name[1024];
ALuint count, j;
count = 0;
do {
if(count == 0)
snprintf(name, sizeof(name), "%s", WaveCaps.szPname);
else
snprintf(name, sizeof(name), "%s #%d", WaveCaps.szPname, count+1);
count++;
for(j = 0;j < i;j++)
{
if(strcmp(name, PlaybackDeviceList[j]) == 0)
break;
}
} while(j != i);
PlaybackDeviceList[i] = strdup(name);
}
}
}
static void ProbeCaptureDevices(void)
{
ALuint i;
for(i = 0;i < NumCaptureDevices;i++)
free(CaptureDeviceList[i]);
NumCaptureDevices = waveInGetNumDevs();
CaptureDeviceList = realloc(CaptureDeviceList, sizeof(ALCchar*) * NumCaptureDevices);
for(i = 0;i < NumCaptureDevices;i++)
{
WAVEINCAPS WaveInCaps;
CaptureDeviceList[i] = NULL;
if(waveInGetDevCaps(i, &WaveInCaps, sizeof(WAVEINCAPS)) == MMSYSERR_NOERROR)
{
char name[1024];
ALuint count, j;
count = 0;
do {
if(count == 0)
snprintf(name, sizeof(name), "%s", WaveInCaps.szPname);
else
snprintf(name, sizeof(name), "%s #%d", WaveInCaps.szPname, count+1);
count++;
for(j = 0;j < i;j++)
{
if(strcmp(name, CaptureDeviceList[j]) == 0)
break;
}
} while(j != i);
CaptureDeviceList[i] = strdup(name);
}
}
}
/*
WaveOutProc
Posts a message to 'PlaybackThreadProc' everytime a WaveOut Buffer is completed and
returns to the application (for more data)
*/
static void CALLBACK WaveOutProc(HWAVEOUT device, UINT msg, DWORD_PTR instance, DWORD_PTR param1, DWORD_PTR param2)
{
ALCdevice *Device = (ALCdevice*)instance;
WinMMData *data = Device->ExtraData;
(void)device;
(void)param2;
if(msg != WOM_DONE)
return;
InterlockedDecrement(&data->WaveBuffersCommitted);
PostThreadMessage(data->WaveThreadID, msg, 0, param1);
}
/*
PlaybackThreadProc
Used by "MMSYSTEM" Device. Called when a WaveOut buffer has used up its
audio data.
*/
static DWORD WINAPI PlaybackThreadProc(LPVOID param)
{
ALCdevice *Device = (ALCdevice*)param;
WinMMData *data = Device->ExtraData;
LPWAVEHDR WaveHdr;
ALuint FrameSize;
MSG msg;
FrameSize = FrameSizeFromDevFmt(Device->FmtChans, Device->FmtType);
SetRTPriority();
while(GetMessage(&msg, NULL, 0, 0))
{
if(msg.message != WOM_DONE)
continue;
if(data->killNow)
{
if(data->WaveBuffersCommitted == 0)
break;
continue;
}
WaveHdr = ((LPWAVEHDR)msg.lParam);
aluMixData(Device, WaveHdr->lpData, WaveHdr->dwBufferLength/FrameSize);
// Send buffer back to play more data
waveOutWrite(data->WaveHandle.Out, WaveHdr, sizeof(WAVEHDR));
InterlockedIncrement(&data->WaveBuffersCommitted);
}
// Signal Wave Thread completed event
if(data->WaveThreadEvent)
SetEvent(data->WaveThreadEvent);
ExitThread(0);
return 0;
}
/*
WaveInProc
Posts a message to 'CaptureThreadProc' everytime a WaveIn Buffer is completed and
returns to the application (with more data)
*/
static void CALLBACK WaveInProc(HWAVEIN device, UINT msg, DWORD_PTR instance, DWORD_PTR param1, DWORD_PTR param2)
{
ALCdevice *Device = (ALCdevice*)instance;
WinMMData *data = Device->ExtraData;
(void)device;
(void)param2;
if(msg != WIM_DATA)
return;
InterlockedDecrement(&data->WaveBuffersCommitted);
PostThreadMessage(data->WaveThreadID, msg, 0, param1);
}
/*
CaptureThreadProc
Used by "MMSYSTEM" Device. Called when a WaveIn buffer had been filled with new
audio data.
*/
static DWORD WINAPI CaptureThreadProc(LPVOID param)
{
ALCdevice *Device = (ALCdevice*)param;
WinMMData *data = Device->ExtraData;
LPWAVEHDR WaveHdr;
ALuint FrameSize;
MSG msg;
FrameSize = FrameSizeFromDevFmt(Device->FmtChans, Device->FmtType);
while(GetMessage(&msg, NULL, 0, 0))
{
if(msg.message != WIM_DATA)
continue;
/* Don't wait for other buffers to finish before quitting. We're
* closing so we don't need them. */
if(data->killNow)
break;
WaveHdr = ((LPWAVEHDR)msg.lParam);
WriteRingBuffer(data->Ring, (ALubyte*)WaveHdr->lpData, WaveHdr->dwBytesRecorded/FrameSize);
// Send buffer back to capture more data
waveInAddBuffer(data->WaveHandle.In, WaveHdr, sizeof(WAVEHDR));
InterlockedIncrement(&data->WaveBuffersCommitted);
}
// Signal Wave Thread completed event
if(data->WaveThreadEvent)
SetEvent(data->WaveThreadEvent);
ExitThread(0);
return 0;
}
static ALCenum WinMMOpenPlayback(ALCdevice *Device, const ALCchar *deviceName)
{
WinMMData *data = NULL;
UINT DeviceID = 0;
MMRESULT res;
ALuint i = 0;
if(!PlaybackDeviceList)
ProbePlaybackDevices();
// Find the Device ID matching the deviceName if valid
for(i = 0;i < NumPlaybackDevices;i++)
{
if(PlaybackDeviceList[i] &&
(!deviceName || strcmp(deviceName, PlaybackDeviceList[i]) == 0))
{
DeviceID = i;
break;
}
}
if(i == NumPlaybackDevices)
return ALC_INVALID_VALUE;
data = calloc(1, sizeof(*data));
if(!data)
return ALC_OUT_OF_MEMORY;
Device->ExtraData = data;
retry_open:
memset(&data->Format, 0, sizeof(WAVEFORMATEX));
if(Device->FmtType == DevFmtFloat)
{
data->Format.wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
data->Format.wBitsPerSample = 32;
}
else
{
data->Format.wFormatTag = WAVE_FORMAT_PCM;
if(Device->FmtType == DevFmtUByte || Device->FmtType == DevFmtByte)
data->Format.wBitsPerSample = 8;
else
data->Format.wBitsPerSample = 16;
}
data->Format.nChannels = ((Device->FmtChans == DevFmtMono) ? 1 : 2);
data->Format.nBlockAlign = data->Format.wBitsPerSample *
data->Format.nChannels / 8;
data->Format.nSamplesPerSec = Device->Frequency;
data->Format.nAvgBytesPerSec = data->Format.nSamplesPerSec *
data->Format.nBlockAlign;
data->Format.cbSize = 0;
if((res=waveOutOpen(&data->WaveHandle.Out, DeviceID, &data->Format, (DWORD_PTR)&WaveOutProc, (DWORD_PTR)Device, CALLBACK_FUNCTION)) != MMSYSERR_NOERROR)
{
if(Device->FmtType == DevFmtFloat)
{
Device->FmtType = DevFmtShort;
goto retry_open;
}
ERR("waveOutOpen failed: %u\n", res);
goto failure;
}
data->WaveThreadEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
if(data->WaveThreadEvent == NULL)
{
ERR("CreateEvent failed: %lu\n", GetLastError());
goto failure;
}
Device->DeviceName = strdup(PlaybackDeviceList[DeviceID]);
return ALC_NO_ERROR;
failure:
if(data->WaveThreadEvent)
CloseHandle(data->WaveThreadEvent);
if(data->WaveHandle.Out)
waveOutClose(data->WaveHandle.Out);
free(data);
Device->ExtraData = NULL;
return ALC_INVALID_VALUE;
}
static void WinMMClosePlayback(ALCdevice *device)
{
WinMMData *data = (WinMMData*)device->ExtraData;
// Close the Wave device
CloseHandle(data->WaveThreadEvent);
data->WaveThreadEvent = 0;
waveOutClose(data->WaveHandle.Out);
data->WaveHandle.Out = 0;
free(data);
device->ExtraData = NULL;
}
static ALCboolean WinMMResetPlayback(ALCdevice *device)
{
WinMMData *data = (WinMMData*)device->ExtraData;
device->UpdateSize = (ALuint)((ALuint64)device->UpdateSize *
data->Format.nSamplesPerSec /
device->Frequency);
device->UpdateSize = (device->UpdateSize*device->NumUpdates + 3) / 4;
device->NumUpdates = 4;
device->Frequency = data->Format.nSamplesPerSec;
if(data->Format.wFormatTag == WAVE_FORMAT_IEEE_FLOAT)
{
if(data->Format.wBitsPerSample == 32)
device->FmtType = DevFmtFloat;
else
{
ERR("Unhandled IEEE float sample depth: %d\n", data->Format.wBitsPerSample);
return ALC_FALSE;
}
}
else if(data->Format.wFormatTag == WAVE_FORMAT_PCM)
{
if(data->Format.wBitsPerSample == 16)
device->FmtType = DevFmtShort;
else if(data->Format.wBitsPerSample == 8)
device->FmtType = DevFmtUByte;
else
{
ERR("Unhandled PCM sample depth: %d\n", data->Format.wBitsPerSample);
return ALC_FALSE;
}
}
else
{
ERR("Unhandled format tag: 0x%04x\n", data->Format.wFormatTag);
return ALC_FALSE;
}
if(data->Format.nChannels == 2)
device->FmtChans = DevFmtStereo;
else if(data->Format.nChannels == 1)
device->FmtChans = DevFmtMono;
else
{
ERR("Unhandled channel count: %d\n", data->Format.nChannels);
return ALC_FALSE;
}
SetDefaultWFXChannelOrder(device);
return ALC_TRUE;
}
static ALCboolean WinMMStartPlayback(ALCdevice *device)
{
WinMMData *data = (WinMMData*)device->ExtraData;
ALbyte *BufferData;
ALint BufferSize;
ALuint i;
data->WaveThread = CreateThread(NULL, 0, (LPTHREAD_START_ROUTINE)PlaybackThreadProc, (LPVOID)device, 0, &data->WaveThreadID);
if(data->WaveThread == NULL)
return ALC_FALSE;
data->WaveBuffersCommitted = 0;
// Create 4 Buffers
BufferSize = device->UpdateSize*device->NumUpdates / 4;
BufferSize *= FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
BufferData = calloc(4, BufferSize);
for(i = 0;i < 4;i++)
{
memset(&data->WaveBuffer[i], 0, sizeof(WAVEHDR));
data->WaveBuffer[i].dwBufferLength = BufferSize;
data->WaveBuffer[i].lpData = ((i==0) ? (LPSTR)BufferData :
(data->WaveBuffer[i-1].lpData +
data->WaveBuffer[i-1].dwBufferLength));
waveOutPrepareHeader(data->WaveHandle.Out, &data->WaveBuffer[i], sizeof(WAVEHDR));
waveOutWrite(data->WaveHandle.Out, &data->WaveBuffer[i], sizeof(WAVEHDR));
InterlockedIncrement(&data->WaveBuffersCommitted);
}
return ALC_TRUE;
}
static void WinMMStopPlayback(ALCdevice *device)
{
WinMMData *data = (WinMMData*)device->ExtraData;
void *buffer = NULL;
int i;
if(data->WaveThread == NULL)
return;
// Set flag to stop processing headers
data->killNow = AL_TRUE;
// Wait for signal that Wave Thread has been destroyed
WaitForSingleObjectEx(data->WaveThreadEvent, 5000, FALSE);
CloseHandle(data->WaveThread);
data->WaveThread = 0;
data->killNow = AL_FALSE;
// Release the wave buffers
for(i = 0;i < 4;i++)
{
waveOutUnprepareHeader(data->WaveHandle.Out, &data->WaveBuffer[i], sizeof(WAVEHDR));
if(i == 0) buffer = data->WaveBuffer[i].lpData;
data->WaveBuffer[i].lpData = NULL;
}
free(buffer);
}
static ALCenum WinMMOpenCapture(ALCdevice *Device, const ALCchar *deviceName)
{
ALbyte *BufferData = NULL;
DWORD CapturedDataSize;
WinMMData *data = NULL;
UINT DeviceID = 0;
ALint BufferSize;
MMRESULT res;
ALuint i;
if(!CaptureDeviceList)
ProbeCaptureDevices();
// Find the Device ID matching the deviceName if valid
for(i = 0;i < NumCaptureDevices;i++)
{
if(CaptureDeviceList[i] &&
(!deviceName || strcmp(deviceName, CaptureDeviceList[i]) == 0))
{
DeviceID = i;
break;
}
}
if(i == NumCaptureDevices)
return ALC_INVALID_VALUE;
switch(Device->FmtChans)
{
case DevFmtMono:
case DevFmtStereo:
break;
case DevFmtQuad:
case DevFmtX51:
case DevFmtX51Side:
case DevFmtX61:
case DevFmtX71:
return ALC_INVALID_ENUM;
}
switch(Device->FmtType)
{
case DevFmtUByte:
case DevFmtShort:
case DevFmtInt:
case DevFmtFloat:
break;
case DevFmtByte:
case DevFmtUShort:
case DevFmtUInt:
return ALC_INVALID_ENUM;
}
data = calloc(1, sizeof(*data));
if(!data)
return ALC_OUT_OF_MEMORY;
Device->ExtraData = data;
memset(&data->Format, 0, sizeof(WAVEFORMATEX));
data->Format.wFormatTag = ((Device->FmtType == DevFmtFloat) ?
WAVE_FORMAT_IEEE_FLOAT : WAVE_FORMAT_PCM);
data->Format.nChannels = ChannelsFromDevFmt(Device->FmtChans);
data->Format.wBitsPerSample = BytesFromDevFmt(Device->FmtType) * 8;
data->Format.nBlockAlign = data->Format.wBitsPerSample *
data->Format.nChannels / 8;
data->Format.nSamplesPerSec = Device->Frequency;
data->Format.nAvgBytesPerSec = data->Format.nSamplesPerSec *
data->Format.nBlockAlign;
data->Format.cbSize = 0;
if((res=waveInOpen(&data->WaveHandle.In, DeviceID, &data->Format, (DWORD_PTR)&WaveInProc, (DWORD_PTR)Device, CALLBACK_FUNCTION)) != MMSYSERR_NOERROR)
{
ERR("waveInOpen failed: %u\n", res);
goto failure;
}
data->WaveThreadEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
if(data->WaveThreadEvent == NULL)
{
ERR("CreateEvent failed: %lu\n", GetLastError());
goto failure;
}
// Allocate circular memory buffer for the captured audio
CapturedDataSize = Device->UpdateSize*Device->NumUpdates;
// Make sure circular buffer is at least 100ms in size
if(CapturedDataSize < (data->Format.nSamplesPerSec / 10))
CapturedDataSize = data->Format.nSamplesPerSec / 10;
data->Ring = CreateRingBuffer(data->Format.nBlockAlign, CapturedDataSize);
if(!data->Ring)
goto failure;
data->WaveBuffersCommitted = 0;
// Create 4 Buffers of 50ms each
BufferSize = data->Format.nAvgBytesPerSec / 20;
BufferSize -= (BufferSize % data->Format.nBlockAlign);
BufferData = calloc(4, BufferSize);
if(!BufferData)
goto failure;
for(i = 0;i < 4;i++)
{
memset(&data->WaveBuffer[i], 0, sizeof(WAVEHDR));
data->WaveBuffer[i].dwBufferLength = BufferSize;
data->WaveBuffer[i].lpData = ((i==0) ? (LPSTR)BufferData :
(data->WaveBuffer[i-1].lpData +
data->WaveBuffer[i-1].dwBufferLength));
data->WaveBuffer[i].dwFlags = 0;
data->WaveBuffer[i].dwLoops = 0;
waveInPrepareHeader(data->WaveHandle.In, &data->WaveBuffer[i], sizeof(WAVEHDR));
waveInAddBuffer(data->WaveHandle.In, &data->WaveBuffer[i], sizeof(WAVEHDR));
InterlockedIncrement(&data->WaveBuffersCommitted);
}
data->WaveThread = CreateThread(NULL, 0, (LPTHREAD_START_ROUTINE)CaptureThreadProc, (LPVOID)Device, 0, &data->WaveThreadID);
if (data->WaveThread == NULL)
goto failure;
Device->DeviceName = strdup(CaptureDeviceList[DeviceID]);
return ALC_NO_ERROR;
failure:
if(data->WaveThread)
CloseHandle(data->WaveThread);
if(BufferData)
{
for(i = 0;i < 4;i++)
waveInUnprepareHeader(data->WaveHandle.In, &data->WaveBuffer[i], sizeof(WAVEHDR));
free(BufferData);
}
if(data->Ring)
DestroyRingBuffer(data->Ring);
if(data->WaveThreadEvent)
CloseHandle(data->WaveThreadEvent);
if(data->WaveHandle.In)
waveInClose(data->WaveHandle.In);
free(data);
Device->ExtraData = NULL;
return ALC_INVALID_VALUE;
}
static void WinMMCloseCapture(ALCdevice *Device)
{
WinMMData *data = (WinMMData*)Device->ExtraData;
void *buffer = NULL;
int i;
/* Tell the processing thread to quit and wait for it to do so. */
data->killNow = AL_TRUE;
PostThreadMessage(data->WaveThreadID, WM_QUIT, 0, 0);
WaitForSingleObjectEx(data->WaveThreadEvent, 5000, FALSE);
/* Make sure capture is stopped and all pending buffers are flushed. */
waveInReset(data->WaveHandle.In);
CloseHandle(data->WaveThread);
data->WaveThread = 0;
// Release the wave buffers
for(i = 0;i < 4;i++)
{
waveInUnprepareHeader(data->WaveHandle.In, &data->WaveBuffer[i], sizeof(WAVEHDR));
if(i == 0) buffer = data->WaveBuffer[i].lpData;
data->WaveBuffer[i].lpData = NULL;
}
free(buffer);
DestroyRingBuffer(data->Ring);
data->Ring = NULL;
// Close the Wave device
CloseHandle(data->WaveThreadEvent);
data->WaveThreadEvent = 0;
waveInClose(data->WaveHandle.In);
data->WaveHandle.In = 0;
free(data);
Device->ExtraData = NULL;
}
static void WinMMStartCapture(ALCdevice *Device)
{
WinMMData *data = (WinMMData*)Device->ExtraData;
waveInStart(data->WaveHandle.In);
}
static void WinMMStopCapture(ALCdevice *Device)
{
WinMMData *data = (WinMMData*)Device->ExtraData;
waveInStop(data->WaveHandle.In);
}
static ALCenum WinMMCaptureSamples(ALCdevice *Device, ALCvoid *Buffer, ALCuint Samples)
{
WinMMData *data = (WinMMData*)Device->ExtraData;
ReadRingBuffer(data->Ring, Buffer, Samples);
return ALC_NO_ERROR;
}
static ALCuint WinMMAvailableSamples(ALCdevice *Device)
{
WinMMData *data = (WinMMData*)Device->ExtraData;
return RingBufferSize(data->Ring);
}
static const BackendFuncs WinMMFuncs = {
WinMMOpenPlayback,
WinMMClosePlayback,
WinMMResetPlayback,
WinMMStartPlayback,
WinMMStopPlayback,
WinMMOpenCapture,
WinMMCloseCapture,
WinMMStartCapture,
WinMMStopCapture,
WinMMCaptureSamples,
WinMMAvailableSamples,
ALCdevice_LockDefault,
ALCdevice_UnlockDefault,
ALCdevice_GetLatencyDefault
};
ALCboolean alcWinMMInit(BackendFuncs *FuncList)
{
*FuncList = WinMMFuncs;
return ALC_TRUE;
}
void alcWinMMDeinit()
{
ALuint i;
for(i = 0;i < NumPlaybackDevices;i++)
free(PlaybackDeviceList[i]);
free(PlaybackDeviceList);
PlaybackDeviceList = NULL;
NumPlaybackDevices = 0;
for(i = 0;i < NumCaptureDevices;i++)
free(CaptureDeviceList[i]);
free(CaptureDeviceList);
CaptureDeviceList = NULL;
NumCaptureDevices = 0;
}
void alcWinMMProbe(enum DevProbe type)
{
ALuint i;
switch(type)
{
case ALL_DEVICE_PROBE:
ProbePlaybackDevices();
for(i = 0;i < NumPlaybackDevices;i++)
{
if(PlaybackDeviceList[i])
AppendAllDevicesList(PlaybackDeviceList[i]);
}
break;
case CAPTURE_DEVICE_PROBE:
ProbeCaptureDevices();
for(i = 0;i < NumCaptureDevices;i++)
{
if(CaptureDeviceList[i])
AppendCaptureDeviceList(CaptureDeviceList[i]);
}
break;
}
}
+185 -201
View File
@@ -1,201 +1,185 @@
/*-
* Copyright (c) 2005 Boris Mikhaylov
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files (the
* "Software"), to deal in the Software without restriction, including
* without limitation the rights to use, copy, modify, merge, publish,
* distribute, sublicense, and/or sell copies of the Software, and to
* permit persons to whom the Software is furnished to do so, subject to
* the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
#include <math.h>
#include "bs2b.h"
#ifndef M_PI
#define M_PI 3.14159265358979323846
#endif
/* Single pole IIR filter.
* O[n] = a0*I[n] + a1*I[n-1] + b1*O[n-1]
*/
/* Lowpass filter */
#define lo_filter(in, out_1) (bs2b->a0_lo*(in) + bs2b->b1_lo*(out_1))
/* Highboost filter */
#define hi_filter(in, in_1, out_1) (bs2b->a0_hi*(in) + bs2b->a1_hi*(in_1) + bs2b->b1_hi*(out_1))
/* Set up all data. */
static void init(struct bs2b *bs2b)
{
double Fc_lo, Fc_hi;
double G_lo, G_hi;
double x;
if ((bs2b->srate > 192000) || (bs2b->srate < 2000))
bs2b->srate = BS2B_DEFAULT_SRATE;
switch(bs2b->level)
{
case BS2B_LOW_CLEVEL: /* Low crossfeed level */
Fc_lo = 360.0;
Fc_hi = 501.0;
G_lo = 0.398107170553497;
G_hi = 0.205671765275719;
break;
case BS2B_MIDDLE_CLEVEL: /* Middle crossfeed level */
Fc_lo = 500.0;
Fc_hi = 711.0;
G_lo = 0.459726988530872;
G_hi = 0.228208484414988;
break;
case BS2B_HIGH_CLEVEL: /* High crossfeed level (virtual speakers are closer to itself) */
Fc_lo = 700.0;
Fc_hi = 1021.0;
G_lo = 0.530884444230988;
G_hi = 0.250105790667544;
break;
case BS2B_LOW_ECLEVEL: /* Low easy crossfeed level */
Fc_lo = 360.0;
Fc_hi = 494.0;
G_lo = 0.316227766016838;
G_hi = 0.168236228897329;
break;
case BS2B_MIDDLE_ECLEVEL: /* Middle easy crossfeed level */
Fc_lo = 500.0;
Fc_hi = 689.0;
G_lo = 0.354813389233575;
G_hi = 0.187169483835901;
break;
default: /* High easy crossfeed level */
bs2b->level = BS2B_HIGH_ECLEVEL;
Fc_lo = 700.0;
Fc_hi = 975.0;
G_lo = 0.398107170553497;
G_hi = 0.205671765275719;
break;
} /* switch */
/* $fc = $Fc / $s;
* $d = 1 / 2 / pi / $fc;
* $x = exp(-1 / $d);
*/
x = exp(-2.0 * M_PI * Fc_lo / bs2b->srate);
bs2b->b1_lo = x;
bs2b->a0_lo = G_lo * (1.0 - x);
x = exp(-2.0 * M_PI * Fc_hi / bs2b->srate);
bs2b->b1_hi = x;
bs2b->a0_hi = 1.0 - G_hi * (1.0 - x);
bs2b->a1_hi = -x;
bs2b->gain = 1.0 / (1.0 - G_hi + G_lo);
bs2b_clear(bs2b);
} /* init */
/* Exported functions.
* See descriptions in "bs2b.h"
*/
void bs2b_set_level(struct bs2b *bs2b, int level)
{
if(level == bs2b->level)
return;
bs2b->level = level;
init(bs2b);
} /* bs2b_set_level */
int bs2b_get_level(struct bs2b *bs2b)
{
return bs2b->level;
} /* bs2b_get_level */
void bs2b_set_srate(struct bs2b *bs2b, int srate)
{
if (srate == bs2b->srate)
return;
bs2b->srate = srate;
init(bs2b);
} /* bs2b_set_srate */
int bs2b_get_srate(struct bs2b *bs2b)
{
return bs2b->srate;
} /* bs2b_get_srate */
void bs2b_clear(struct bs2b *bs2b)
{
int loopv = sizeof(bs2b->last_sample);
while (loopv)
{
((char *)&bs2b->last_sample)[--loopv] = 0;
}
} /* bs2b_clear */
int bs2b_is_clear(struct bs2b *bs2b)
{
int loopv = sizeof(bs2b->last_sample);
while (loopv)
{
if (((char *)&bs2b->last_sample)[--loopv] != 0)
return 0;
}
return 1;
} /* bs2b_is_clear */
void bs2b_cross_feed(struct bs2b *bs2b, float *sample)
{
/* Lowpass filter */
bs2b->last_sample.lo[0] = lo_filter(sample[0], bs2b->last_sample.lo[0]);
bs2b->last_sample.lo[1] = lo_filter(sample[1], bs2b->last_sample.lo[1]);
/* Highboost filter */
bs2b->last_sample.hi[0] = hi_filter(sample[0], bs2b->last_sample.asis[0], bs2b->last_sample.hi[0]);
bs2b->last_sample.hi[1] = hi_filter(sample[1], bs2b->last_sample.asis[1], bs2b->last_sample.hi[1]);
bs2b->last_sample.asis[0] = sample[0];
bs2b->last_sample.asis[1] = sample[1];
/* Crossfeed */
sample[0] = bs2b->last_sample.hi[0] + bs2b->last_sample.lo[1];
sample[1] = bs2b->last_sample.hi[1] + bs2b->last_sample.lo[0];
/* Bass boost cause allpass attenuation */
sample[0] *= bs2b->gain;
sample[1] *= bs2b->gain;
/* Clipping of overloaded samples */
#if 0
if (sample[0] > 1.0)
sample[0] = 1.0;
if (sample[0] < -1.0)
sample[0] = -1.0;
if (sample[1] > 1.0)
sample[1] = 1.0;
if (sample[1] < -1.0)
sample[1] = -1.0;
#endif
} /* bs2b_cross_feed */
/*-
* Copyright (c) 2005 Boris Mikhaylov
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files (the
* "Software"), to deal in the Software without restriction, including
* without limitation the rights to use, copy, modify, merge, publish,
* distribute, sublicense, and/or sell copies of the Software, and to
* permit persons to whom the Software is furnished to do so, subject to
* the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
#include "config.h"
#include <math.h>
#include <string.h>
#include "bs2b.h"
#ifndef M_PI
#define M_PI 3.14159265358979323846
#endif
/* Single pole IIR filter.
* O[n] = a0*I[n] + a1*I[n-1] + b1*O[n-1]
*/
/* Lowpass filter */
#define lo_filter(in, out_1) (bs2b->a0_lo*(in) + bs2b->b1_lo*(out_1))
/* Highboost filter */
#define hi_filter(in, in_1, out_1) (bs2b->a0_hi*(in) + bs2b->a1_hi*(in_1) + bs2b->b1_hi*(out_1))
/* Set up all data. */
static void init(struct bs2b *bs2b)
{
double Fc_lo, Fc_hi;
double G_lo, G_hi;
double x;
if ((bs2b->srate > 192000) || (bs2b->srate < 2000))
bs2b->srate = BS2B_DEFAULT_SRATE;
switch(bs2b->level)
{
case BS2B_LOW_CLEVEL: /* Low crossfeed level */
Fc_lo = 360.0;
Fc_hi = 501.0;
G_lo = 0.398107170553497;
G_hi = 0.205671765275719;
break;
case BS2B_MIDDLE_CLEVEL: /* Middle crossfeed level */
Fc_lo = 500.0;
Fc_hi = 711.0;
G_lo = 0.459726988530872;
G_hi = 0.228208484414988;
break;
case BS2B_HIGH_CLEVEL: /* High crossfeed level (virtual speakers are closer to itself) */
Fc_lo = 700.0;
Fc_hi = 1021.0;
G_lo = 0.530884444230988;
G_hi = 0.250105790667544;
break;
case BS2B_LOW_ECLEVEL: /* Low easy crossfeed level */
Fc_lo = 360.0;
Fc_hi = 494.0;
G_lo = 0.316227766016838;
G_hi = 0.168236228897329;
break;
case BS2B_MIDDLE_ECLEVEL: /* Middle easy crossfeed level */
Fc_lo = 500.0;
Fc_hi = 689.0;
G_lo = 0.354813389233575;
G_hi = 0.187169483835901;
break;
default: /* High easy crossfeed level */
bs2b->level = BS2B_HIGH_ECLEVEL;
Fc_lo = 700.0;
Fc_hi = 975.0;
G_lo = 0.398107170553497;
G_hi = 0.205671765275719;
break;
} /* switch */
/* $fc = $Fc / $s;
* $d = 1 / 2 / pi / $fc;
* $x = exp(-1 / $d);
*/
x = exp(-2.0 * M_PI * Fc_lo / bs2b->srate);
bs2b->b1_lo = x;
bs2b->a0_lo = G_lo * (1.0 - x);
x = exp(-2.0 * M_PI * Fc_hi / bs2b->srate);
bs2b->b1_hi = x;
bs2b->a0_hi = 1.0 - G_hi * (1.0 - x);
bs2b->a1_hi = -x;
bs2b->gain = 1.0f / (float)(1.0 - G_hi + G_lo);
} /* init */
/* Exported functions.
* See descriptions in "bs2b.h"
*/
void bs2b_set_level(struct bs2b *bs2b, int level)
{
if(level == bs2b->level)
return;
bs2b->level = level;
init(bs2b);
} /* bs2b_set_level */
int bs2b_get_level(struct bs2b *bs2b)
{
return bs2b->level;
} /* bs2b_get_level */
void bs2b_set_srate(struct bs2b *bs2b, int srate)
{
if (srate == bs2b->srate)
return;
bs2b->srate = srate;
init(bs2b);
} /* bs2b_set_srate */
int bs2b_get_srate(struct bs2b *bs2b)
{
return bs2b->srate;
} /* bs2b_get_srate */
void bs2b_clear(struct bs2b *bs2b)
{
memset(&bs2b->last_sample, 0, sizeof(bs2b->last_sample));
} /* bs2b_clear */
void bs2b_cross_feed(struct bs2b *bs2b, float *sample)
{
/* Lowpass filter */
bs2b->last_sample.lo[0] = lo_filter(sample[0], bs2b->last_sample.lo[0]);
bs2b->last_sample.lo[1] = lo_filter(sample[1], bs2b->last_sample.lo[1]);
/* Highboost filter */
bs2b->last_sample.hi[0] = hi_filter(sample[0], bs2b->last_sample.asis[0], bs2b->last_sample.hi[0]);
bs2b->last_sample.hi[1] = hi_filter(sample[1], bs2b->last_sample.asis[1], bs2b->last_sample.hi[1]);
bs2b->last_sample.asis[0] = sample[0];
bs2b->last_sample.asis[1] = sample[1];
/* Crossfeed */
sample[0] = (float)(bs2b->last_sample.hi[0] + bs2b->last_sample.lo[1]);
sample[1] = (float)(bs2b->last_sample.hi[1] + bs2b->last_sample.lo[0]);
/* Bass boost cause allpass attenuation */
sample[0] *= bs2b->gain;
sample[1] *= bs2b->gain;
/* Clipping of overloaded samples */
#if 0
if (sample[0] > 1.0)
sample[0] = 1.0;
if (sample[0] < -1.0)
sample[0] = -1.0;
if (sample[1] > 1.0)
sample[1] = 1.0;
if (sample[1] < -1.0)
sample[1] = -1.0;
#endif
} /* bs2b_cross_feed */
-285
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@@ -1,285 +0,0 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2007 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include <stdlib.h>
#include <stdio.h>
#include <memory.h>
#include <windows.h>
#include <mmsystem.h>
#include <dsound.h>
#include "alMain.h"
#include "AL/al.h"
#include "AL/alc.h"
typedef struct {
// DirectSound Playback Device
LPDIRECTSOUND lpDS;
LPDIRECTSOUNDBUFFER DSpbuffer;
LPDIRECTSOUNDBUFFER DSsbuffer;
MMRESULT ulDSTimerID;
DWORD OldWriteCursor;
} DSoundData;
static ALCchar *DeviceList[16];
static void CALLBACK DirectSoundProc(UINT uID,UINT uReserved,DWORD_PTR dwUser,DWORD_PTR dwReserved1,DWORD_PTR dwReserved2)
{
ALCdevice *pDevice = (ALCdevice *)dwUser;
DSoundData *pData = pDevice->ExtraData;
DWORD PlayCursor,WriteCursor;
BYTE *WritePtr1,*WritePtr2;
DWORD WriteCnt1,WriteCnt2;
WAVEFORMATEX OutputType;
DWORD BytesPlayed;
DWORD BufSize;
HRESULT DSRes;
(void)uID;
(void)uReserved;
(void)dwReserved1;
(void)dwReserved2;
BufSize = pDevice->UpdateFreq * 2 * pDevice->FrameSize;
// Get current play and write cursors
IDirectSoundBuffer_GetCurrentPosition(pData->DSsbuffer,&PlayCursor,&WriteCursor);
if (!pData->OldWriteCursor) pData->OldWriteCursor=WriteCursor-PlayCursor;
// Get the output format and figure the number of bytes played (block aligned)
IDirectSoundBuffer_GetFormat(pData->DSsbuffer,&OutputType,sizeof(WAVEFORMATEX),NULL);
BytesPlayed=((((WriteCursor<pData->OldWriteCursor)?(BufSize+WriteCursor-pData->OldWriteCursor):(WriteCursor-pData->OldWriteCursor))/OutputType.nBlockAlign)*OutputType.nBlockAlign);
// Lock output buffer started at 40msec in front of the old write cursor (15msec in front of the actual write cursor)
DSRes=IDirectSoundBuffer_Lock(pData->DSsbuffer,(pData->OldWriteCursor+(OutputType.nSamplesPerSec/25)*OutputType.nBlockAlign)%BufSize,BytesPlayed,(LPVOID*)&WritePtr1,&WriteCnt1,(LPVOID*)&WritePtr2,&WriteCnt2,0);
// If the buffer is lost, restore it, play and lock
if (DSRes==DSERR_BUFFERLOST)
{
IDirectSoundBuffer_Restore(pData->DSsbuffer);
IDirectSoundBuffer_Play(pData->DSsbuffer,0,0,DSBPLAY_LOOPING);
DSRes=IDirectSoundBuffer_Lock(pData->DSsbuffer,(pData->OldWriteCursor+(OutputType.nSamplesPerSec/25)*OutputType.nBlockAlign)%BufSize,BytesPlayed,(LPVOID*)&WritePtr1,&WriteCnt1,(LPVOID*)&WritePtr2,&WriteCnt2,0);
}
// Successfully locked the output buffer
if (DSRes==DS_OK)
{
// If we have an active context, mix data directly into output buffer otherwise fill with silence
SuspendContext(NULL);
if (WritePtr1)
aluMixData(pDevice->Context, WritePtr1, WriteCnt1, pDevice->Format);
if (WritePtr2)
aluMixData(pDevice->Context, WritePtr2, WriteCnt2, pDevice->Format);
ProcessContext(NULL);
// Unlock output buffer only when successfully locked
IDirectSoundBuffer_Unlock(pData->DSsbuffer,WritePtr1,WriteCnt1,WritePtr2,WriteCnt2);
}
// Update old write cursor location
pData->OldWriteCursor=((pData->OldWriteCursor+BytesPlayed)%BufSize);
}
static ALCboolean DSoundOpenPlayback(ALCdevice *device, const ALCchar *deviceName)
{
DSBUFFERDESC DSBDescription;
DSoundData *pData = NULL;
WAVEFORMATEX OutputType;
HRESULT hr;
if(deviceName)
{
int i;
for(i = 0;DeviceList[i];i++)
{
if(strcmp(deviceName, DeviceList[i]) == 0)
{
device->szDeviceName = DeviceList[i];
break;
}
}
if(!DeviceList[i])
return ALC_FALSE;
}
else
device->szDeviceName = DeviceList[0];
//Platform specific
memset(&OutputType, 0, sizeof(WAVEFORMATEX));
OutputType.wFormatTag = WAVE_FORMAT_PCM;
OutputType.nChannels = device->Channels;
OutputType.wBitsPerSample = aluBytesFromFormat(device->Format) * 8;
OutputType.nBlockAlign = OutputType.nChannels*OutputType.wBitsPerSample/8;
OutputType.nSamplesPerSec = device->Frequency;
OutputType.nAvgBytesPerSec = OutputType.nSamplesPerSec*OutputType.nBlockAlign;
OutputType.cbSize = 0;
//Initialise requested device
pData = calloc(1, sizeof(DSoundData));
if(!pData)
{
SetALCError(ALC_OUT_OF_MEMORY);
return ALC_FALSE;
}
//Init COM
CoInitialize(NULL);
//DirectSound Init code
hr = CoCreateInstance(&CLSID_DirectSound, NULL, CLSCTX_INPROC_SERVER, &IID_IDirectSound, (LPVOID*)&pData->lpDS);
if(SUCCEEDED(hr))
hr = IDirectSound_Initialize(pData->lpDS, NULL);
if(SUCCEEDED(hr))
hr = IDirectSound_SetCooperativeLevel(pData->lpDS, GetForegroundWindow(), DSSCL_PRIORITY);
if(SUCCEEDED(hr))
{
memset(&DSBDescription,0,sizeof(DSBUFFERDESC));
DSBDescription.dwSize=sizeof(DSBUFFERDESC);
DSBDescription.dwFlags=DSBCAPS_PRIMARYBUFFER;
hr = IDirectSound_CreateSoundBuffer(pData->lpDS, &DSBDescription, &pData->DSpbuffer, NULL);
}
if(SUCCEEDED(hr))
hr = IDirectSoundBuffer_SetFormat(pData->DSpbuffer,&OutputType);
if(SUCCEEDED(hr))
{
memset(&DSBDescription,0,sizeof(DSBUFFERDESC));
DSBDescription.dwSize=sizeof(DSBUFFERDESC);
DSBDescription.dwFlags=DSBCAPS_GLOBALFOCUS|DSBCAPS_GETCURRENTPOSITION2;
DSBDescription.dwBufferBytes=device->UpdateFreq * 2 * device->FrameSize;
DSBDescription.lpwfxFormat=&OutputType;
hr = IDirectSound_CreateSoundBuffer(pData->lpDS, &DSBDescription, &pData->DSsbuffer, NULL);
}
if(SUCCEEDED(hr))
hr = IDirectSoundBuffer_Play(pData->DSsbuffer, 0, 0, DSBPLAY_LOOPING);
if(FAILED(hr))
{
if (pData->DSsbuffer)
IDirectSoundBuffer_Release(pData->DSsbuffer);
if (pData->DSpbuffer)
IDirectSoundBuffer_Release(pData->DSpbuffer);
if (pData->lpDS)
IDirectSound_Release(pData->lpDS);
free(pData);
return ALC_FALSE;
}
pData->ulDSTimerID = timeSetEvent(25, 0, (LPTIMECALLBACK)DirectSoundProc, (DWORD)device, (UINT)TIME_CALLBACK_FUNCTION|TIME_PERIODIC);
device->MaxNoOfSources = 256;
device->ExtraData = pData;
return ALC_TRUE;
}
static void DSoundClosePlayback(ALCdevice *device)
{
DSoundData *pData = device->ExtraData;
// Stop and release the DS timer
if (pData->ulDSTimerID)
timeKillEvent(pData->ulDSTimerID);
// Wait ... just in case any timer events happen
Sleep(100);
SuspendContext(NULL);
if (pData->DSsbuffer)
IDirectSoundBuffer_Release(pData->DSsbuffer);
if (pData->DSpbuffer)
IDirectSoundBuffer_Release(pData->DSpbuffer);
IDirectSound_Release(pData->lpDS);
//Deinit COM
CoUninitialize();
ProcessContext(NULL);
free(pData);
device->ExtraData = NULL;
}
static ALCboolean DSoundOpenCapture(ALCdevice *pDevice, const ALCchar *deviceName, ALCuint frequency, ALCenum format, ALCsizei SampleSize)
{
(void)pDevice;
(void)deviceName;
(void)frequency;
(void)format;
(void)SampleSize;
return ALC_FALSE;
}
static void DSoundCloseCapture(ALCdevice *pDevice)
{
(void)pDevice;
}
static void DSoundStartCapture(ALCdevice *pDevice)
{
(void)pDevice;
}
static void DSoundStopCapture(ALCdevice *pDevice)
{
(void)pDevice;
}
static void DSoundCaptureSamples(ALCdevice *pDevice, ALCvoid *pBuffer, ALCuint lSamples)
{
(void)pDevice;
(void)pBuffer;
(void)lSamples;
}
static ALCuint DSoundAvailableSamples(ALCdevice *pDevice)
{
(void)pDevice;
return 0;
}
BackendFuncs DSoundFuncs = {
DSoundOpenPlayback,
DSoundClosePlayback,
DSoundOpenCapture,
DSoundCloseCapture,
DSoundStartCapture,
DSoundStopCapture,
DSoundCaptureSamples,
DSoundAvailableSamples
};
void alcDSoundInit(BackendFuncs *FuncList)
{
*FuncList = DSoundFuncs;
DeviceList[0] = AppendDeviceList("DirectSound Software");
AppendAllDeviceList(DeviceList[0]);
}
+674
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@@ -0,0 +1,674 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 2011 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdlib.h>
#include <time.h>
#include <errno.h>
#include <stdarg.h>
#ifdef HAVE_MALLOC_H
#include <malloc.h>
#endif
#if defined(HAVE_GUIDDEF_H) || defined(HAVE_INITGUID_H)
#define INITGUID
#include <windows.h>
#ifdef HAVE_GUIDDEF_H
#include <guiddef.h>
#else
#include <initguid.h>
#endif
DEFINE_GUID(KSDATAFORMAT_SUBTYPE_PCM, 0x00000001, 0x0000, 0x0010, 0x80,0x00, 0x00,0xaa,0x00,0x38,0x9b,0x71);
DEFINE_GUID(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, 0x00000003, 0x0000, 0x0010, 0x80,0x00, 0x00,0xaa,0x00,0x38,0x9b,0x71);
DEFINE_GUID(IID_IDirectSoundNotify, 0xb0210783, 0x89cd, 0x11d0, 0xaf,0x08, 0x00,0xa0,0xc9,0x25,0xcd,0x16);
DEFINE_GUID(CLSID_MMDeviceEnumerator, 0xbcde0395, 0xe52f, 0x467c, 0x8e,0x3d, 0xc4,0x57,0x92,0x91,0x69,0x2e);
DEFINE_GUID(IID_IMMDeviceEnumerator, 0xa95664d2, 0x9614, 0x4f35, 0xa7,0x46, 0xde,0x8d,0xb6,0x36,0x17,0xe6);
DEFINE_GUID(IID_IAudioClient, 0x1cb9ad4c, 0xdbfa, 0x4c32, 0xb1,0x78, 0xc2,0xf5,0x68,0xa7,0x03,0xb2);
DEFINE_GUID(IID_IAudioRenderClient, 0xf294acfc, 0x3146, 0x4483, 0xa7,0xbf, 0xad,0xdc,0xa7,0xc2,0x60,0xe2);
#ifdef HAVE_MMDEVAPI
#include <devpropdef.h>
DEFINE_DEVPROPKEY(DEVPKEY_Device_FriendlyName, 0xa45c254e, 0xdf1c, 0x4efd, 0x80,0x20, 0x67,0xd1,0x46,0xa8,0x50,0xe0, 14);
#endif
#endif
#ifdef HAVE_DLFCN_H
#include <dlfcn.h>
#endif
#ifdef HAVE_CPUID_H
#include <cpuid.h>
#endif
#ifdef HAVE_FLOAT_H
#include <float.h>
#endif
#ifdef HAVE_IEEEFP_H
#include <ieeefp.h>
#endif
#include "alMain.h"
ALuint CPUCapFlags = 0;
void FillCPUCaps(ALuint capfilter)
{
ALuint caps = 0;
/* FIXME: We really should get this for all available CPUs in case different
* CPUs have different caps (is that possible on one machine?). */
#if defined(HAVE_CPUID_H) && (defined(__i386__) || defined(__x86_64__) || \
defined(_M_IX86) || defined(_M_X64))
union {
unsigned int regs[4];
char str[sizeof(unsigned int[4])];
} cpuinf[3];
if(!__get_cpuid(0, &cpuinf[0].regs[0], &cpuinf[0].regs[1], &cpuinf[0].regs[2], &cpuinf[0].regs[3]))
ERR("Failed to get CPUID\n");
else
{
unsigned int maxfunc = cpuinf[0].regs[0];
unsigned int maxextfunc = 0;
if(__get_cpuid(0x80000000, &cpuinf[0].regs[0], &cpuinf[0].regs[1], &cpuinf[0].regs[2], &cpuinf[0].regs[3]))
maxextfunc = cpuinf[0].regs[0];
TRACE("Detected max CPUID function: 0x%x (ext. 0x%x)\n", maxfunc, maxextfunc);
TRACE("Vendor ID: \"%.4s%.4s%.4s\"\n", cpuinf[0].str+4, cpuinf[0].str+12, cpuinf[0].str+8);
if(maxextfunc >= 0x80000004 &&
__get_cpuid(0x80000002, &cpuinf[0].regs[0], &cpuinf[0].regs[1], &cpuinf[0].regs[2], &cpuinf[0].regs[3]) &&
__get_cpuid(0x80000003, &cpuinf[1].regs[0], &cpuinf[1].regs[1], &cpuinf[1].regs[2], &cpuinf[1].regs[3]) &&
__get_cpuid(0x80000004, &cpuinf[2].regs[0], &cpuinf[2].regs[1], &cpuinf[2].regs[2], &cpuinf[2].regs[3]))
TRACE("Name: \"%.16s%.16s%.16s\"\n", cpuinf[0].str, cpuinf[1].str, cpuinf[2].str);
if(maxfunc >= 1 &&
__get_cpuid(1, &cpuinf[0].regs[0], &cpuinf[0].regs[1], &cpuinf[0].regs[2], &cpuinf[0].regs[3]))
{
#ifdef bit_SSE
if((cpuinf[0].regs[3]&bit_SSE))
caps |= CPU_CAP_SSE;
#endif
}
}
#elif defined(HAVE_WINDOWS_H)
HMODULE k32 = GetModuleHandleA("kernel32.dll");
BOOL (WINAPI*IsProcessorFeaturePresent)(DWORD ProcessorFeature);
IsProcessorFeaturePresent = (BOOL(WINAPI*)(DWORD))GetProcAddress(k32, "IsProcessorFeaturePresent");
if(!IsProcessorFeaturePresent)
ERR("IsProcessorFeaturePresent not available; CPU caps not detected\n");
else
{
if(IsProcessorFeaturePresent(PF_XMMI_INSTRUCTIONS_AVAILABLE))
caps |= CPU_CAP_SSE;
}
#endif
#ifdef HAVE_NEON
/* Assume Neon support if compiled with it */
caps |= CPU_CAP_NEON;
#endif
TRACE("Got caps:%s%s%s\n", ((caps&CPU_CAP_SSE)?((capfilter&CPU_CAP_SSE)?" SSE":" (SSE)"):""),
((caps&CPU_CAP_NEON)?((capfilter&CPU_CAP_NEON)?" Neon":" (Neon)"):""),
((!caps)?" -none-":""));
CPUCapFlags = caps & capfilter;
}
void *al_malloc(size_t alignment, size_t size)
{
#if defined(HAVE_ALIGNED_ALLOC)
size = (size+(alignment-1))&~(alignment-1);
return aligned_alloc(alignment, size);
#elif defined(HAVE_POSIX_MEMALIGN)
void *ret;
if(posix_memalign(&ret, alignment, size) == 0)
return ret;
return NULL;
#elif defined(HAVE__ALIGNED_MALLOC)
return _aligned_malloc(size, alignment);
#else
char *ret = malloc(size+alignment);
if(ret != NULL)
{
*(ret++) = 0x00;
while(((ALintptrEXT)ret&(alignment-1)) != 0)
*(ret++) = 0x55;
}
return ret;
#endif
}
void *al_calloc(size_t alignment, size_t size)
{
void *ret = al_malloc(alignment, size);
if(ret) memset(ret, 0, size);
return ret;
}
void al_free(void *ptr)
{
#if defined(HAVE_ALIGNED_ALLOC) || defined(HAVE_POSIX_MEMALIGN)
free(ptr);
#elif defined(HAVE__ALIGNED_MALLOC)
_aligned_free(ptr);
#else
if(ptr != NULL)
{
char *finder = ptr;
do {
--finder;
} while(*finder == 0x55);
free(finder);
}
#endif
}
#if (defined(HAVE___CONTROL87_2) || defined(HAVE__CONTROLFP)) && (defined(__x86_64__) || defined(_M_X64))
/* Win64 doesn't allow us to set the precision control. */
#undef _MCW_PC
#define _MCW_PC 0
#endif
void SetMixerFPUMode(FPUCtl *ctl)
{
#if defined(__GNUC__) && (defined(__i386__) || defined(__x86_64__))
unsigned short fpuState;
__asm__ __volatile__("fnstcw %0" : "=m" (*&fpuState));
ctl->state = fpuState;
fpuState &= ~0x300; /* clear precision to single */
fpuState |= 0xC00; /* set round-to-zero */
__asm__ __volatile__("fldcw %0" : : "m" (*&fpuState));
#ifdef HAVE_SSE
if((CPUCapFlags&CPU_CAP_SSE))
{
int sseState;
__asm__ __volatile__("stmxcsr %0" : "=m" (*&sseState));
ctl->sse_state = sseState;
sseState |= 0x0C00; /* set round-to-zero */
sseState |= 0x8000; /* set flush-to-zero */
__asm__ __volatile__("ldmxcsr %0" : : "m" (*&sseState));
}
#endif
#elif defined(HAVE___CONTROL87_2)
int mode;
__control87_2(0, 0, &ctl->state, NULL);
__control87_2(_RC_CHOP|_PC_24, _MCW_RC|_MCW_PC, &mode, NULL);
#ifdef HAVE_SSE
if((CPUCapFlags&CPU_CAP_SSE))
{
__control87_2(0, 0, NULL, &ctl->sse_state);
__control87_2(_RC_CHOP|_DN_FLUSH, _MCW_RC|_MCW_DN, NULL, &mode);
}
#endif
#elif defined(HAVE__CONTROLFP)
ctl->state = _controlfp(0, 0);
(void)_controlfp(_RC_CHOP|_PC_24, _MCW_RC|_MCW_PC);
#elif defined(HAVE_FESETROUND)
ctl->state = fegetround();
#ifdef FE_TOWARDZERO
fesetround(FE_TOWARDZERO);
#endif
#endif
}
void RestoreFPUMode(const FPUCtl *ctl)
{
#if defined(__GNUC__) && (defined(__i386__) || defined(__x86_64__))
unsigned short fpuState = ctl->state;
__asm__ __volatile__("fldcw %0" : : "m" (*&fpuState));
#ifdef HAVE_SSE
if((CPUCapFlags&CPU_CAP_SSE))
__asm__ __volatile__("ldmxcsr %0" : : "m" (*&ctl->sse_state));
#endif
#elif defined(HAVE___CONTROL87_2)
int mode;
__control87_2(ctl->state, _MCW_RC|_MCW_PC, &mode, NULL);
#ifdef HAVE_SSE
if((CPUCapFlags&CPU_CAP_SSE))
__control87_2(ctl->sse_state, _MCW_RC|_MCW_DN, NULL, &mode);
#endif
#elif defined(HAVE__CONTROLFP)
_controlfp(ctl->state, _MCW_RC|_MCW_PC);
#elif defined(HAVE_FESETROUND)
fesetround(ctl->state);
#endif
}
#ifdef _WIN32
void pthread_once(pthread_once_t *once, void (*callback)(void))
{
LONG ret;
while((ret=InterlockedExchange(once, 1)) == 1)
sched_yield();
if(ret == 0)
callback();
InterlockedExchange(once, 2);
}
int pthread_key_create(pthread_key_t *key, void (*callback)(void*))
{
*key = TlsAlloc();
if(callback)
InsertUIntMapEntry(&TlsDestructor, *key, callback);
return 0;
}
int pthread_key_delete(pthread_key_t key)
{
InsertUIntMapEntry(&TlsDestructor, key, NULL);
TlsFree(key);
return 0;
}
void *pthread_getspecific(pthread_key_t key)
{ return TlsGetValue(key); }
int pthread_setspecific(pthread_key_t key, void *val)
{
TlsSetValue(key, val);
return 0;
}
void *LoadLib(const char *name)
{ return LoadLibraryA(name); }
void CloseLib(void *handle)
{ FreeLibrary((HANDLE)handle); }
void *GetSymbol(void *handle, const char *name)
{
void *ret;
ret = (void*)GetProcAddress((HANDLE)handle, name);
if(ret == NULL)
ERR("Failed to load %s\n", name);
return ret;
}
WCHAR *strdupW(const WCHAR *str)
{
const WCHAR *n;
WCHAR *ret;
size_t len;
n = str;
while(*n) n++;
len = n - str;
ret = calloc(sizeof(WCHAR), len+1);
if(ret != NULL)
memcpy(ret, str, sizeof(WCHAR)*len);
return ret;
}
#else
#include <pthread.h>
#ifdef HAVE_PTHREAD_NP_H
#include <pthread_np.h>
#endif
#include <sched.h>
void InitializeCriticalSection(CRITICAL_SECTION *cs)
{
pthread_mutexattr_t attrib;
int ret;
ret = pthread_mutexattr_init(&attrib);
assert(ret == 0);
ret = pthread_mutexattr_settype(&attrib, PTHREAD_MUTEX_RECURSIVE);
#ifdef HAVE_PTHREAD_NP_H
if(ret != 0)
ret = pthread_mutexattr_setkind_np(&attrib, PTHREAD_MUTEX_RECURSIVE);
#endif
assert(ret == 0);
ret = pthread_mutex_init(cs, &attrib);
assert(ret == 0);
pthread_mutexattr_destroy(&attrib);
}
void DeleteCriticalSection(CRITICAL_SECTION *cs)
{
int ret;
ret = pthread_mutex_destroy(cs);
assert(ret == 0);
}
void EnterCriticalSection(CRITICAL_SECTION *cs)
{
int ret;
ret = pthread_mutex_lock(cs);
assert(ret == 0);
}
void LeaveCriticalSection(CRITICAL_SECTION *cs)
{
int ret;
ret = pthread_mutex_unlock(cs);
assert(ret == 0);
}
/* NOTE: This wrapper isn't quite accurate as it returns an ALuint, as opposed
* to the expected DWORD. Both are defined as unsigned 32-bit types, however.
* Additionally, Win32 is supposed to measure the time since Windows started,
* as opposed to the actual time. */
ALuint timeGetTime(void)
{
#if _POSIX_TIMERS > 0
struct timespec ts;
int ret = -1;
#if defined(_POSIX_MONOTONIC_CLOCK) && (_POSIX_MONOTONIC_CLOCK >= 0)
#if _POSIX_MONOTONIC_CLOCK == 0
static int hasmono = 0;
if(hasmono > 0 || (hasmono == 0 &&
(hasmono=sysconf(_SC_MONOTONIC_CLOCK)) > 0))
#endif
ret = clock_gettime(CLOCK_MONOTONIC, &ts);
#endif
if(ret != 0)
ret = clock_gettime(CLOCK_REALTIME, &ts);
assert(ret == 0);
return ts.tv_nsec/1000000 + ts.tv_sec*1000;
#else
struct timeval tv;
int ret;
ret = gettimeofday(&tv, NULL);
assert(ret == 0);
return tv.tv_usec/1000 + tv.tv_sec*1000;
#endif
}
void Sleep(ALuint t)
{
struct timespec tv, rem;
tv.tv_nsec = (t*1000000)%1000000000;
tv.tv_sec = t/1000;
while(nanosleep(&tv, &rem) == -1 && errno == EINTR)
tv = rem;
}
#ifdef HAVE_DLFCN_H
void *LoadLib(const char *name)
{
const char *err;
void *handle;
dlerror();
handle = dlopen(name, RTLD_NOW);
if((err=dlerror()) != NULL)
handle = NULL;
return handle;
}
void CloseLib(void *handle)
{ dlclose(handle); }
void *GetSymbol(void *handle, const char *name)
{
const char *err;
void *sym;
dlerror();
sym = dlsym(handle, name);
if((err=dlerror()) != NULL)
{
WARN("Failed to load %s: %s\n", name, err);
sym = NULL;
}
return sym;
}
#endif
#endif
void al_print(const char *type, const char *func, const char *fmt, ...)
{
char str[256];
int i;
i = snprintf(str, sizeof(str), "AL lib: %s %s: ", type, func);
if(i > 0 && (unsigned int)i < sizeof(str))
{
va_list ap;
va_start(ap, fmt);
vsnprintf(str+i, sizeof(str)-i, fmt, ap);
va_end(ap);
}
str[sizeof(str)-1] = 0;
fprintf(LogFile, "%s", str);
fflush(LogFile);
}
void SetRTPriority(void)
{
ALboolean failed = AL_FALSE;
#ifdef _WIN32
if(RTPrioLevel > 0)
failed = !SetThreadPriority(GetCurrentThread(), THREAD_PRIORITY_TIME_CRITICAL);
#elif defined(HAVE_PTHREAD_SETSCHEDPARAM) && !defined(__OpenBSD__)
if(RTPrioLevel > 0)
{
struct sched_param param;
/* Use the minimum real-time priority possible for now (on Linux this
* should be 1 for SCHED_RR) */
param.sched_priority = sched_get_priority_min(SCHED_RR);
failed = !!pthread_setschedparam(pthread_self(), SCHED_RR, &param);
}
#else
/* Real-time priority not available */
failed = (RTPrioLevel>0);
#endif
if(failed)
ERR("Failed to set priority level for thread\n");
}
static void Lock(volatile ALenum *l)
{
while(ExchangeInt(l, AL_TRUE) == AL_TRUE)
sched_yield();
}
static void Unlock(volatile ALenum *l)
{
ExchangeInt(l, AL_FALSE);
}
void RWLockInit(RWLock *lock)
{
lock->read_count = 0;
lock->write_count = 0;
lock->read_lock = AL_FALSE;
lock->read_entry_lock = AL_FALSE;
lock->write_lock = AL_FALSE;
}
void ReadLock(RWLock *lock)
{
Lock(&lock->read_entry_lock);
Lock(&lock->read_lock);
if(IncrementRef(&lock->read_count) == 1)
Lock(&lock->write_lock);
Unlock(&lock->read_lock);
Unlock(&lock->read_entry_lock);
}
void ReadUnlock(RWLock *lock)
{
if(DecrementRef(&lock->read_count) == 0)
Unlock(&lock->write_lock);
}
void WriteLock(RWLock *lock)
{
if(IncrementRef(&lock->write_count) == 1)
Lock(&lock->read_lock);
Lock(&lock->write_lock);
}
void WriteUnlock(RWLock *lock)
{
Unlock(&lock->write_lock);
if(DecrementRef(&lock->write_count) == 0)
Unlock(&lock->read_lock);
}
void InitUIntMap(UIntMap *map, ALsizei limit)
{
map->array = NULL;
map->size = 0;
map->maxsize = 0;
map->limit = limit;
RWLockInit(&map->lock);
}
void ResetUIntMap(UIntMap *map)
{
WriteLock(&map->lock);
free(map->array);
map->array = NULL;
map->size = 0;
map->maxsize = 0;
WriteUnlock(&map->lock);
}
ALenum InsertUIntMapEntry(UIntMap *map, ALuint key, ALvoid *value)
{
ALsizei pos = 0;
WriteLock(&map->lock);
if(map->size > 0)
{
ALsizei low = 0;
ALsizei high = map->size - 1;
while(low < high)
{
ALsizei mid = low + (high-low)/2;
if(map->array[mid].key < key)
low = mid + 1;
else
high = mid;
}
if(map->array[low].key < key)
low++;
pos = low;
}
if(pos == map->size || map->array[pos].key != key)
{
if(map->size == map->limit)
{
WriteUnlock(&map->lock);
return AL_OUT_OF_MEMORY;
}
if(map->size == map->maxsize)
{
ALvoid *temp = NULL;
ALsizei newsize;
newsize = (map->maxsize ? (map->maxsize<<1) : 4);
if(newsize >= map->maxsize)
temp = realloc(map->array, newsize*sizeof(map->array[0]));
if(!temp)
{
WriteUnlock(&map->lock);
return AL_OUT_OF_MEMORY;
}
map->array = temp;
map->maxsize = newsize;
}
if(pos < map->size)
memmove(&map->array[pos+1], &map->array[pos],
(map->size-pos)*sizeof(map->array[0]));
map->size++;
}
map->array[pos].key = key;
map->array[pos].value = value;
WriteUnlock(&map->lock);
return AL_NO_ERROR;
}
ALvoid *RemoveUIntMapKey(UIntMap *map, ALuint key)
{
ALvoid *ptr = NULL;
WriteLock(&map->lock);
if(map->size > 0)
{
ALsizei low = 0;
ALsizei high = map->size - 1;
while(low < high)
{
ALsizei mid = low + (high-low)/2;
if(map->array[mid].key < key)
low = mid + 1;
else
high = mid;
}
if(map->array[low].key == key)
{
ptr = map->array[low].value;
if(low < map->size-1)
memmove(&map->array[low], &map->array[low+1],
(map->size-1-low)*sizeof(map->array[0]));
map->size--;
}
}
WriteUnlock(&map->lock);
return ptr;
}
ALvoid *LookupUIntMapKey(UIntMap *map, ALuint key)
{
ALvoid *ptr = NULL;
ReadLock(&map->lock);
if(map->size > 0)
{
ALsizei low = 0;
ALsizei high = map->size - 1;
while(low < high)
{
ALsizei mid = low + (high-low)/2;
if(map->array[mid].key < key)
low = mid + 1;
else
high = mid;
}
if(map->array[low].key == key)
ptr = map->array[low].value;
}
ReadUnlock(&map->lock);
return ptr;
}
+804
View File
@@ -0,0 +1,804 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 2011 by Chris Robinson
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdlib.h>
#include <ctype.h>
#include "AL/al.h"
#include "AL/alc.h"
#include "alMain.h"
#include "alSource.h"
#include "alu.h"
#ifndef PATH_MAX
#define PATH_MAX 4096
#endif
/* Current data set limits defined by the makehrtf utility. */
#define MIN_IR_SIZE (8)
#define MAX_IR_SIZE (128)
#define MOD_IR_SIZE (8)
#define MIN_EV_COUNT (5)
#define MAX_EV_COUNT (128)
#define MIN_AZ_COUNT (1)
#define MAX_AZ_COUNT (128)
struct Hrtf {
ALuint sampleRate;
ALuint irSize;
ALubyte evCount;
const ALubyte *azCount;
const ALushort *evOffset;
const ALshort *coeffs;
const ALubyte *delays;
struct Hrtf *next;
};
static const ALchar magicMarker00[8] = "MinPHR00";
static const ALchar magicMarker01[8] = "MinPHR01";
/* Define the default HRTF:
* ALubyte defaultAzCount [DefaultHrtf.evCount]
* ALushort defaultEvOffset [DefaultHrtf.evCount]
* ALshort defaultCoeffs [DefaultHrtf.irCount * defaultHrtf.irSize]
* ALubyte defaultDelays [DefaultHrtf.irCount]
*
* struct Hrtf DefaultHrtf
*/
#include "hrtf_tables.inc"
static struct Hrtf *LoadedHrtfs = NULL;
/* Calculate the elevation indices given the polar elevation in radians.
* This will return two indices between 0 and (Hrtf->evCount - 1) and an
* interpolation factor between 0.0 and 1.0.
*/
static void CalcEvIndices(const struct Hrtf *Hrtf, ALfloat ev, ALuint *evidx, ALfloat *evmu)
{
ev = (F_PI_2 + ev) * (Hrtf->evCount-1) / F_PI;
evidx[0] = fastf2u(ev);
evidx[1] = minu(evidx[0] + 1, Hrtf->evCount-1);
*evmu = ev - evidx[0];
}
/* Calculate the azimuth indices given the polar azimuth in radians. This
* will return two indices between 0 and (Hrtf->azCount[ei] - 1) and an
* interpolation factor between 0.0 and 1.0.
*/
static void CalcAzIndices(const struct Hrtf *Hrtf, ALuint evidx, ALfloat az, ALuint *azidx, ALfloat *azmu)
{
az = (F_PI*2.0f + az) * Hrtf->azCount[evidx] / (F_PI*2.0f);
azidx[0] = fastf2u(az) % Hrtf->azCount[evidx];
azidx[1] = (azidx[0] + 1) % Hrtf->azCount[evidx];
*azmu = az - floorf(az);
}
/* Calculates the normalized HRTF transition factor (delta) from the changes
* in gain and listener to source angle between updates. The result is a
* normalized delta factor that can be used to calculate moving HRIR stepping
* values.
*/
ALfloat CalcHrtfDelta(ALfloat oldGain, ALfloat newGain, const ALfloat olddir[3], const ALfloat newdir[3])
{
ALfloat gainChange, angleChange, change;
// Calculate the normalized dB gain change.
newGain = maxf(newGain, 0.0001f);
oldGain = maxf(oldGain, 0.0001f);
gainChange = fabsf(log10f(newGain / oldGain) / log10f(0.0001f));
// Calculate the normalized listener to source angle change when there is
// enough gain to notice it.
angleChange = 0.0f;
if(gainChange > 0.0001f || newGain > 0.0001f)
{
// No angle change when the directions are equal or degenerate (when
// both have zero length).
if(newdir[0]-olddir[0] || newdir[1]-olddir[1] || newdir[2]-olddir[2])
angleChange = acosf(olddir[0]*newdir[0] +
olddir[1]*newdir[1] +
olddir[2]*newdir[2]) / F_PI;
}
// Use the largest of the two changes for the delta factor, and apply a
// significance shaping function to it.
change = maxf(angleChange * 25.0f, gainChange) * 2.0f;
return minf(change, 1.0f);
}
/* Calculates static HRIR coefficients and delays for the given polar
* elevation and azimuth in radians. Linear interpolation is used to
* increase the apparent resolution of the HRIR data set. The coefficients
* are also normalized and attenuated by the specified gain.
*/
void GetLerpedHrtfCoeffs(const struct Hrtf *Hrtf, ALfloat elevation, ALfloat azimuth, ALfloat gain, ALfloat (*coeffs)[2], ALuint *delays)
{
ALuint evidx[2], azidx[2];
ALuint lidx[4], ridx[4];
ALfloat mu[3], blend[4];
ALuint i;
// Claculate elevation indices and interpolation factor.
CalcEvIndices(Hrtf, elevation, evidx, &mu[2]);
// Calculate azimuth indices and interpolation factor for the first
// elevation.
CalcAzIndices(Hrtf, evidx[0], azimuth, azidx, &mu[0]);
// Calculate the first set of linear HRIR indices for left and right
// channels.
lidx[0] = Hrtf->evOffset[evidx[0]] + azidx[0];
lidx[1] = Hrtf->evOffset[evidx[0]] + azidx[1];
ridx[0] = Hrtf->evOffset[evidx[0]] + ((Hrtf->azCount[evidx[0]]-azidx[0]) % Hrtf->azCount[evidx[0]]);
ridx[1] = Hrtf->evOffset[evidx[0]] + ((Hrtf->azCount[evidx[0]]-azidx[1]) % Hrtf->azCount[evidx[0]]);
// Calculate azimuth indices and interpolation factor for the second
// elevation.
CalcAzIndices(Hrtf, evidx[1], azimuth, azidx, &mu[1]);
// Calculate the second set of linear HRIR indices for left and right
// channels.
lidx[2] = Hrtf->evOffset[evidx[1]] + azidx[0];
lidx[3] = Hrtf->evOffset[evidx[1]] + azidx[1];
ridx[2] = Hrtf->evOffset[evidx[1]] + ((Hrtf->azCount[evidx[1]]-azidx[0]) % Hrtf->azCount[evidx[1]]);
ridx[3] = Hrtf->evOffset[evidx[1]] + ((Hrtf->azCount[evidx[1]]-azidx[1]) % Hrtf->azCount[evidx[1]]);
/* Calculate 4 blending weights for 2D bilinear interpolation. */
blend[0] = (1.0f-mu[0]) * (1.0f-mu[2]);
blend[1] = ( mu[0]) * (1.0f-mu[2]);
blend[2] = (1.0f-mu[1]) * ( mu[2]);
blend[3] = ( mu[1]) * ( mu[2]);
/* Calculate the HRIR delays using linear interpolation. */
delays[0] = fastf2u(Hrtf->delays[lidx[0]]*blend[0] + Hrtf->delays[lidx[1]]*blend[1] +
Hrtf->delays[lidx[2]]*blend[2] + Hrtf->delays[lidx[3]]*blend[3] +
0.5f) << HRTFDELAY_BITS;
delays[1] = fastf2u(Hrtf->delays[ridx[0]]*blend[0] + Hrtf->delays[ridx[1]]*blend[1] +
Hrtf->delays[ridx[2]]*blend[2] + Hrtf->delays[ridx[3]]*blend[3] +
0.5f) << HRTFDELAY_BITS;
/* Calculate the sample offsets for the HRIR indices. */
lidx[0] *= Hrtf->irSize;
lidx[1] *= Hrtf->irSize;
lidx[2] *= Hrtf->irSize;
lidx[3] *= Hrtf->irSize;
ridx[0] *= Hrtf->irSize;
ridx[1] *= Hrtf->irSize;
ridx[2] *= Hrtf->irSize;
ridx[3] *= Hrtf->irSize;
/* Calculate the normalized and attenuated HRIR coefficients using linear
* interpolation when there is enough gain to warrant it. Zero the
* coefficients if gain is too low.
*/
if(gain > 0.0001f)
{
gain *= 1.0f/32767.0f;
for(i = 0;i < Hrtf->irSize;i++)
{
coeffs[i][0] = (Hrtf->coeffs[lidx[0]+i]*blend[0] +
Hrtf->coeffs[lidx[1]+i]*blend[1] +
Hrtf->coeffs[lidx[2]+i]*blend[2] +
Hrtf->coeffs[lidx[3]+i]*blend[3]) * gain;
coeffs[i][1] = (Hrtf->coeffs[ridx[0]+i]*blend[0] +
Hrtf->coeffs[ridx[1]+i]*blend[1] +
Hrtf->coeffs[ridx[2]+i]*blend[2] +
Hrtf->coeffs[ridx[3]+i]*blend[3]) * gain;
}
}
else
{
for(i = 0;i < Hrtf->irSize;i++)
{
coeffs[i][0] = 0.0f;
coeffs[i][1] = 0.0f;
}
}
}
/* Calculates the moving HRIR target coefficients, target delays, and
* stepping values for the given polar elevation and azimuth in radians.
* Linear interpolation is used to increase the apparent resolution of the
* HRIR data set. The coefficients are also normalized and attenuated by the
* specified gain. Stepping resolution and count is determined using the
* given delta factor between 0.0 and 1.0.
*/
ALuint GetMovingHrtfCoeffs(const struct Hrtf *Hrtf, ALfloat elevation, ALfloat azimuth, ALfloat gain, ALfloat delta, ALint counter, ALfloat (*coeffs)[2], ALuint *delays, ALfloat (*coeffStep)[2], ALint *delayStep)
{
ALuint evidx[2], azidx[2];
ALuint lidx[4], ridx[4];
ALfloat mu[3], blend[4];
ALfloat left, right;
ALfloat step;
ALuint i;
// Claculate elevation indices and interpolation factor.
CalcEvIndices(Hrtf, elevation, evidx, &mu[2]);
// Calculate azimuth indices and interpolation factor for the first
// elevation.
CalcAzIndices(Hrtf, evidx[0], azimuth, azidx, &mu[0]);
// Calculate the first set of linear HRIR indices for left and right
// channels.
lidx[0] = Hrtf->evOffset[evidx[0]] + azidx[0];
lidx[1] = Hrtf->evOffset[evidx[0]] + azidx[1];
ridx[0] = Hrtf->evOffset[evidx[0]] + ((Hrtf->azCount[evidx[0]]-azidx[0]) % Hrtf->azCount[evidx[0]]);
ridx[1] = Hrtf->evOffset[evidx[0]] + ((Hrtf->azCount[evidx[0]]-azidx[1]) % Hrtf->azCount[evidx[0]]);
// Calculate azimuth indices and interpolation factor for the second
// elevation.
CalcAzIndices(Hrtf, evidx[1], azimuth, azidx, &mu[1]);
// Calculate the second set of linear HRIR indices for left and right
// channels.
lidx[2] = Hrtf->evOffset[evidx[1]] + azidx[0];
lidx[3] = Hrtf->evOffset[evidx[1]] + azidx[1];
ridx[2] = Hrtf->evOffset[evidx[1]] + ((Hrtf->azCount[evidx[1]]-azidx[0]) % Hrtf->azCount[evidx[1]]);
ridx[3] = Hrtf->evOffset[evidx[1]] + ((Hrtf->azCount[evidx[1]]-azidx[1]) % Hrtf->azCount[evidx[1]]);
// Calculate the stepping parameters.
delta = maxf(floorf(delta*(Hrtf->sampleRate*0.015f) + 0.5f), 1.0f);
step = 1.0f / delta;
/* Calculate 4 blending weights for 2D bilinear interpolation. */
blend[0] = (1.0f-mu[0]) * (1.0f-mu[2]);
blend[1] = ( mu[0]) * (1.0f-mu[2]);
blend[2] = (1.0f-mu[1]) * ( mu[2]);
blend[3] = ( mu[1]) * ( mu[2]);
/* Calculate the HRIR delays using linear interpolation. Then calculate
* the delay stepping values using the target and previous running
* delays.
*/
left = (ALfloat)(delays[0] - (delayStep[0] * counter));
right = (ALfloat)(delays[1] - (delayStep[1] * counter));
delays[0] = fastf2u(Hrtf->delays[lidx[0]]*blend[0] + Hrtf->delays[lidx[1]]*blend[1] +
Hrtf->delays[lidx[2]]*blend[2] + Hrtf->delays[lidx[3]]*blend[3] +
0.5f) << HRTFDELAY_BITS;
delays[1] = fastf2u(Hrtf->delays[ridx[0]]*blend[0] + Hrtf->delays[ridx[1]]*blend[1] +
Hrtf->delays[ridx[2]]*blend[2] + Hrtf->delays[ridx[3]]*blend[3] +
0.5f) << HRTFDELAY_BITS;
delayStep[0] = fastf2i(step * (delays[0] - left));
delayStep[1] = fastf2i(step * (delays[1] - right));
/* Calculate the sample offsets for the HRIR indices. */
lidx[0] *= Hrtf->irSize;
lidx[1] *= Hrtf->irSize;
lidx[2] *= Hrtf->irSize;
lidx[3] *= Hrtf->irSize;
ridx[0] *= Hrtf->irSize;
ridx[1] *= Hrtf->irSize;
ridx[2] *= Hrtf->irSize;
ridx[3] *= Hrtf->irSize;
/* Calculate the normalized and attenuated target HRIR coefficients using
* linear interpolation when there is enough gain to warrant it. Zero
* the target coefficients if gain is too low. Then calculate the
* coefficient stepping values using the target and previous running
* coefficients.
*/
if(gain > 0.0001f)
{
gain *= 1.0f/32767.0f;
for(i = 0;i < HRIR_LENGTH;i++)
{
left = coeffs[i][0] - (coeffStep[i][0] * counter);
right = coeffs[i][1] - (coeffStep[i][1] * counter);
coeffs[i][0] = (Hrtf->coeffs[lidx[0]+i]*blend[0] +
Hrtf->coeffs[lidx[1]+i]*blend[1] +
Hrtf->coeffs[lidx[2]+i]*blend[2] +
Hrtf->coeffs[lidx[3]+i]*blend[3]) * gain;
coeffs[i][1] = (Hrtf->coeffs[ridx[0]+i]*blend[0] +
Hrtf->coeffs[ridx[1]+i]*blend[1] +
Hrtf->coeffs[ridx[2]+i]*blend[2] +
Hrtf->coeffs[ridx[3]+i]*blend[3]) * gain;
coeffStep[i][0] = step * (coeffs[i][0] - left);
coeffStep[i][1] = step * (coeffs[i][1] - right);
}
}
else
{
for(i = 0;i < HRIR_LENGTH;i++)
{
left = coeffs[i][0] - (coeffStep[i][0] * counter);
right = coeffs[i][1] - (coeffStep[i][1] * counter);
coeffs[i][0] = 0.0f;
coeffs[i][1] = 0.0f;
coeffStep[i][0] = step * -left;
coeffStep[i][1] = step * -right;
}
}
/* The stepping count is the number of samples necessary for the HRIR to
* complete its transition. The mixer will only apply stepping for this
* many samples.
*/
return fastf2u(delta);
}
static struct Hrtf *LoadHrtf00(FILE *f, ALuint deviceRate)
{
const ALubyte maxDelay = SRC_HISTORY_LENGTH-1;
struct Hrtf *Hrtf = NULL;
ALboolean failed = AL_FALSE;
ALuint rate = 0, irCount = 0;
ALushort irSize = 0;
ALubyte evCount = 0;
ALubyte *azCount = NULL;
ALushort *evOffset = NULL;
ALshort *coeffs = NULL;
ALubyte *delays = NULL;
ALuint i, j;
rate = fgetc(f);
rate |= fgetc(f)<<8;
rate |= fgetc(f)<<16;
rate |= fgetc(f)<<24;
irCount = fgetc(f);
irCount |= fgetc(f)<<8;
irSize = fgetc(f);
irSize |= fgetc(f)<<8;
evCount = fgetc(f);
if(rate != deviceRate)
{
ERR("HRIR rate does not match device rate: rate=%d (%d)\n",
rate, deviceRate);
failed = AL_TRUE;
}
if(irSize < MIN_IR_SIZE || irSize > MAX_IR_SIZE || (irSize%MOD_IR_SIZE))
{
ERR("Unsupported HRIR size: irSize=%d (%d to %d by %d)\n",
irSize, MIN_IR_SIZE, MAX_IR_SIZE, MOD_IR_SIZE);
failed = AL_TRUE;
}
if(evCount < MIN_EV_COUNT || evCount > MAX_EV_COUNT)
{
ERR("Unsupported elevation count: evCount=%d (%d to %d)\n",
evCount, MIN_EV_COUNT, MAX_EV_COUNT);
failed = AL_TRUE;
}
if(failed)
return NULL;
azCount = malloc(sizeof(azCount[0])*evCount);
evOffset = malloc(sizeof(evOffset[0])*evCount);
if(azCount == NULL || evOffset == NULL)
{
ERR("Out of memory.\n");
failed = AL_TRUE;
}
if(!failed)
{
evOffset[0] = fgetc(f);
evOffset[0] |= fgetc(f)<<8;
for(i = 1;i < evCount;i++)
{
evOffset[i] = fgetc(f);
evOffset[i] |= fgetc(f)<<8;
if(evOffset[i] <= evOffset[i-1])
{
ERR("Invalid evOffset: evOffset[%d]=%d (last=%d)\n",
i, evOffset[i], evOffset[i-1]);
failed = AL_TRUE;
}
azCount[i-1] = evOffset[i] - evOffset[i-1];
if(azCount[i-1] < MIN_AZ_COUNT || azCount[i-1] > MAX_AZ_COUNT)
{
ERR("Unsupported azimuth count: azCount[%d]=%d (%d to %d)\n",
i-1, azCount[i-1], MIN_AZ_COUNT, MAX_AZ_COUNT);
failed = AL_TRUE;
}
}
if(irCount <= evOffset[i-1])
{
ERR("Invalid evOffset: evOffset[%d]=%d (irCount=%d)\n",
i-1, evOffset[i-1], irCount);
failed = AL_TRUE;
}
azCount[i-1] = irCount - evOffset[i-1];
if(azCount[i-1] < MIN_AZ_COUNT || azCount[i-1] > MAX_AZ_COUNT)
{
ERR("Unsupported azimuth count: azCount[%d]=%d (%d to %d)\n",
i-1, azCount[i-1], MIN_AZ_COUNT, MAX_AZ_COUNT);
failed = AL_TRUE;
}
}
if(!failed)
{
coeffs = malloc(sizeof(coeffs[0])*irSize*irCount);
delays = malloc(sizeof(delays[0])*irCount);
if(coeffs == NULL || delays == NULL)
{
ERR("Out of memory.\n");
failed = AL_TRUE;
}
}
if(!failed)
{
for(i = 0;i < irCount*irSize;i+=irSize)
{
for(j = 0;j < irSize;j++)
{
ALshort coeff;
coeff = fgetc(f);
coeff |= fgetc(f)<<8;
coeffs[i+j] = coeff;
}
}
for(i = 0;i < irCount;i++)
{
delays[i] = fgetc(f);
if(delays[i] > maxDelay)
{
ERR("Invalid delays[%d]: %d (%d)\n", i, delays[i], maxDelay);
failed = AL_TRUE;
}
}
if(feof(f))
{
ERR("Premature end of data\n");
failed = AL_TRUE;
}
}
if(!failed)
{
Hrtf = malloc(sizeof(struct Hrtf));
if(Hrtf == NULL)
{
ERR("Out of memory.\n");
failed = AL_TRUE;
}
}
if(!failed)
{
Hrtf->sampleRate = rate;
Hrtf->irSize = irSize;
Hrtf->evCount = evCount;
Hrtf->azCount = azCount;
Hrtf->evOffset = evOffset;
Hrtf->coeffs = coeffs;
Hrtf->delays = delays;
Hrtf->next = NULL;
return Hrtf;
}
free(azCount);
free(evOffset);
free(coeffs);
free(delays);
return NULL;
}
static struct Hrtf *LoadHrtf01(FILE *f, ALuint deviceRate)
{
const ALubyte maxDelay = SRC_HISTORY_LENGTH-1;
struct Hrtf *Hrtf = NULL;
ALboolean failed = AL_FALSE;
ALuint rate = 0, irCount = 0;
ALubyte irSize = 0, evCount = 0;
ALubyte *azCount = NULL;
ALushort *evOffset = NULL;
ALshort *coeffs = NULL;
ALubyte *delays = NULL;
ALuint i, j;
rate = fgetc(f);
rate |= fgetc(f)<<8;
rate |= fgetc(f)<<16;
rate |= fgetc(f)<<24;
irSize = fgetc(f);
evCount = fgetc(f);
if(rate != deviceRate)
{
ERR("HRIR rate does not match device rate: rate=%d (%d)\n",
rate, deviceRate);
failed = AL_TRUE;
}
if(irSize < MIN_IR_SIZE || irSize > MAX_IR_SIZE || (irSize%MOD_IR_SIZE))
{
ERR("Unsupported HRIR size: irSize=%d (%d to %d by %d)\n",
irSize, MIN_IR_SIZE, MAX_IR_SIZE, MOD_IR_SIZE);
failed = AL_TRUE;
}
if(evCount < MIN_EV_COUNT || evCount > MAX_EV_COUNT)
{
ERR("Unsupported elevation count: evCount=%d (%d to %d)\n",
evCount, MIN_EV_COUNT, MAX_EV_COUNT);
failed = AL_TRUE;
}
if(failed)
return NULL;
azCount = malloc(sizeof(azCount[0])*evCount);
evOffset = malloc(sizeof(evOffset[0])*evCount);
if(azCount == NULL || evOffset == NULL)
{
ERR("Out of memory.\n");
failed = AL_TRUE;
}
if(!failed)
{
for(i = 0;i < evCount;i++)
{
azCount[i] = fgetc(f);
if(azCount[i] < MIN_AZ_COUNT || azCount[i] > MAX_AZ_COUNT)
{
ERR("Unsupported azimuth count: azCount[%d]=%d (%d to %d)\n",
i, azCount[i], MIN_AZ_COUNT, MAX_AZ_COUNT);
failed = AL_TRUE;
}
}
}
if(!failed)
{
evOffset[0] = 0;
irCount = azCount[0];
for(i = 1;i < evCount;i++)
{
evOffset[i] = evOffset[i-1] + azCount[i-1];
irCount += azCount[i];
}
coeffs = malloc(sizeof(coeffs[0])*irSize*irCount);
delays = malloc(sizeof(delays[0])*irCount);
if(coeffs == NULL || delays == NULL)
{
ERR("Out of memory.\n");
failed = AL_TRUE;
}
}
if(!failed)
{
for(i = 0;i < irCount*irSize;i+=irSize)
{
for(j = 0;j < irSize;j++)
{
ALshort coeff;
coeff = fgetc(f);
coeff |= fgetc(f)<<8;
coeffs[i+j] = coeff;
}
}
for(i = 0;i < irCount;i++)
{
delays[i] = fgetc(f);
if(delays[i] > maxDelay)
{
ERR("Invalid delays[%d]: %d (%d)\n", i, delays[i], maxDelay);
failed = AL_TRUE;
}
}
if(feof(f))
{
ERR("Premature end of data\n");
failed = AL_TRUE;
}
}
if(!failed)
{
Hrtf = malloc(sizeof(struct Hrtf));
if(Hrtf == NULL)
{
ERR("Out of memory.\n");
failed = AL_TRUE;
}
}
if(!failed)
{
Hrtf->sampleRate = rate;
Hrtf->irSize = irSize;
Hrtf->evCount = evCount;
Hrtf->azCount = azCount;
Hrtf->evOffset = evOffset;
Hrtf->coeffs = coeffs;
Hrtf->delays = delays;
Hrtf->next = NULL;
return Hrtf;
}
free(azCount);
free(evOffset);
free(coeffs);
free(delays);
return NULL;
}
static struct Hrtf *LoadHrtf(ALuint deviceRate)
{
const char *fnamelist = NULL;
if(!ConfigValueStr(NULL, "hrtf_tables", &fnamelist))
return NULL;
while(*fnamelist != '\0')
{
struct Hrtf *Hrtf = NULL;
char fname[PATH_MAX];
ALchar magic[8];
ALuint i;
FILE *f;
while(isspace(*fnamelist) || *fnamelist == ',')
fnamelist++;
i = 0;
while(*fnamelist != '\0' && *fnamelist != ',')
{
const char *next = strpbrk(fnamelist, "%,");
while(fnamelist != next && *fnamelist && i < sizeof(fname))
fname[i++] = *(fnamelist++);
if(!next || *next == ',')
break;
/* *next == '%' */
next++;
if(*next == 'r')
{
int wrote = snprintf(&fname[i], sizeof(fname)-i, "%u", deviceRate);
i += minu(wrote, sizeof(fname)-i);
next++;
}
else if(*next == '%')
{
if(i < sizeof(fname))
fname[i++] = '%';
next++;
}
else
ERR("Invalid marker '%%%c'\n", *next);
fnamelist = next;
}
i = minu(i, sizeof(fname)-1);
fname[i] = '\0';
while(i > 0 && isspace(fname[i-1]))
i--;
fname[i] = '\0';
if(fname[0] == '\0')
continue;
TRACE("Loading %s...\n", fname);
f = fopen(fname, "rb");
if(f == NULL)
{
ERR("Could not open %s\n", fname);
continue;
}
if(fread(magic, 1, sizeof(magic), f) != sizeof(magic))
ERR("Failed to read header from %s\n", fname);
else
{
if(memcmp(magic, magicMarker00, sizeof(magicMarker00)) == 0)
{
TRACE("Detected data set format v0\n");
Hrtf = LoadHrtf00(f, deviceRate);
}
else if(memcmp(magic, magicMarker01, sizeof(magicMarker01)) == 0)
{
TRACE("Detected data set format v1\n");
Hrtf = LoadHrtf01(f, deviceRate);
}
else
ERR("Invalid header in %s: \"%.8s\"\n", fname, magic);
}
fclose(f);
f = NULL;
if(Hrtf)
{
Hrtf->next = LoadedHrtfs;
LoadedHrtfs = Hrtf;
TRACE("Loaded HRTF support for format: %s %uhz\n",
DevFmtChannelsString(DevFmtStereo), Hrtf->sampleRate);
return Hrtf;
}
ERR("Failed to load %s\n", fname);
}
return NULL;
}
const struct Hrtf *GetHrtf(ALCdevice *device)
{
if(device->FmtChans == DevFmtStereo)
{
struct Hrtf *Hrtf = LoadedHrtfs;
while(Hrtf != NULL)
{
if(device->Frequency == Hrtf->sampleRate)
return Hrtf;
Hrtf = Hrtf->next;
}
Hrtf = LoadHrtf(device->Frequency);
if(Hrtf != NULL)
return Hrtf;
if(device->Frequency == DefaultHrtf.sampleRate)
return &DefaultHrtf;
}
ERR("Incompatible format: %s %uhz\n",
DevFmtChannelsString(device->FmtChans), device->Frequency);
return NULL;
}
void FreeHrtfs(void)
{
struct Hrtf *Hrtf = NULL;
while((Hrtf=LoadedHrtfs) != NULL)
{
LoadedHrtfs = Hrtf->next;
free((void*)Hrtf->azCount);
free((void*)Hrtf->evOffset);
free((void*)Hrtf->coeffs);
free((void*)Hrtf->delays);
free(Hrtf);
}
}
ALuint GetHrtfIrSize (const struct Hrtf *Hrtf)
{
return Hrtf->irSize;
}
+848
View File
@@ -0,0 +1,848 @@
/* Elevation metrics */
static const ALubyte defaultAzCount[19] = { 1, 12, 24, 36, 45, 56, 60, 72, 72, 72, 72, 72, 60, 56, 45, 36, 24, 12, 1, };
static const ALushort defaultEvOffset[19] = { 0, 1, 13, 37, 73, 118, 174, 234, 306, 378, 450, 522, 594, 654, 710, 755, 791, 815, 827, };
/* HRIR Coefficients */
static const ALshort defaultCoeffs[26496] =
{
+6117, +8354, +3531, +2903, +1118, +342, +874, -822, -2447, -1309, +579, -133, -617, -1093, -998, -416, -244, -202, -84, -134, -375, -200, -416, -331, -408, -614, -489, -332, -207, -97, -7, -145,
+6958, +8678, +3013, +3274, +1145, +25, +847, -771, -2501, -1666, +534, +86, -597, -917, -967, -916, -367, -80, -198, -215, -475, -312, -442, -372, -399, -554, -517, -249, -103, -75, -5, -176,
+6349, +8083, +2935, +2854, +1109, +344, +851, -891, -2351, -1252, +506, -191, -559, -861, -745, -485, -327, -77, -17, -172, -405, -189, -402, -350, -459, -599, -449, -315, -184, -77, +13, -134,
+5910, +7494, +2698, +2625, +992, +373, +884, -825, -2197, -962, +698, -127, -425, -759, -697, -286, -213, -171, -64, -127, -336, -131, -355, -271, -368, -553, -428, -299, -179, -74, +4, -137,
+5808, +7380, +2660, +2585, +1001, +453, +998, -698, -2078, -854, +773, -100, -470, -857, -730, -241, -185, -179, -63, -114, -325, -112, -322, -220, -300, -507, -404, -297, -216, -122, -35, -168,
+5957, +7591, +2687, +2503, +871, +340, +913, -821, -2282, -1053, +775, -16, -459, -904, -784, -240, -114, -100, -12, -59, -285, -105, -334, -253, -340, -532, -400, -251, -127, -17, +33, -156,
+6515, +8209, +2752, +2639, +892, +326, +895, -1074, -2683, -1313, +854, -53, -682, -1135, -939, -316, -155, -153, -32, -64, -346, -124, -339, -249, -345, -543, -400, -232, -129, -39, +38, -120,
+6887, +8888, +3060, +2770, +926, +146, +833, -1245, -2800, -1317, +584, -231, -619, -1119, -983, -413, -244, -71, -9, -143, -405, -183, -399, -307, -420, -595, -439, -302, -153, -55, +34, -129,
+7340, +9090, +2964, +3130, +1162, +115, +612, -1577, -2702, -692, +458, -1019, -864, -943, -587, -190, -434, -226, +41, -162, -398, -144, -463, -353, -533, -708, -464, -374, -270, -103, +2, -187,
+8348, +9865, +2233, +2503, +594, +154, +1277, -1477, -3002, -433, +1280, -991, -1387, -1385, -643, +20, -514, -498, -16, -150, -545, -145, -486, -329, -539, -846, -509, -326, -250, -110, +9, -211,
+8668, +10241, +2446, +2335, +160, +133, +1408, -1244, -3403, -895, +1742, -269, -1341, -2123, -1100, +207, -94, -571, -238, -198, -607, -211, -581, -326, -492, -792, -539, -293, -202, -119, -28, -265,
+8261, +10052, +3261, +2775, +344, +65, +1000, -1138, -3661, -1615, +1337, +126, -290, -1970, -1899, -55, +143, -202, -175, -248, -492, -207, -591, -284, -495, -901, -578, -409, -197, -51, +38, -151,
+7749, +9628, +3084, +3049, +870, +16, +727, -1159, -2880, -1819, +969, +83, -717, -1031, -1759, -1055, -39, -9, -26, -31, -379, -324, -646, -479, -344, -651, -608, -341, -196, -48, +22, -160,
+7872, +8889, +2424, +3846, +1113, -405, +870, -636, -2562, -2175, +535, +438, -627, -691, -988, -1594, -378, +123, -383, -287, -574, -435, -436, -421, -370, -475, -575, -138, +9, -79, -10, -210,
+7007, +8290, +2692, +3507, +1037, -125, +838, -876, -2265, -1597, +225, -49, -510, -580, -573, -987, -677, +32, +42, -236, -535, -281, -374, -349, -433, -522, -408, -257, -182, -133, +15, -132,
+6475, +7694, +2419, +2913, +1152, +342, +794, -920, -2186, -1265, +352, -196, -457, -569, -494, -665, -413, +113, +33, -239, -428, -173, -379, -384, -517, -559, -408, -308, -159, -59, +31, -118,
+5981, +7105, +2165, +2692, +1134, +347, +767, -876, -2030, -957, +421, -294, -318, -441, -466, -416, -281, -39, -27, -194, -378, -156, -345, -274, -384, -498, -440, -344, -158, -51, +21, -134,
+5468, +6465, +2050, +2460, +952, +446, +858, -776, -1849, -646, +711, -106, -171, -377, -412, -225, -208, -139, -49, -122, -280, -59, -288, -217, -330, -482, -372, -276, -153, -54, +11, -123,
+5247, +6086, +1760, +2189, +842, +506, +1054, -433, -1470, -298, +965, +30, -189, -498, -477, -163, -126, -106, -24, -94, -283, -72, -247, -143, -266, -434, -302, -196, -110, -68, -52, -213,
+5232, +6232, +1996, +2383, +983, +625, +1098, -519, -1614, -432, +848, -66, -286, -585, -461, -110, -157, -165, -46, -92, -261, -20, -217, -109, -190, -394, -333, -281, -239, -151, -66, -182,
+5323, +6297, +1900, +2226, +867, +582, +1114, -514, -1700, -507, +886, -53, -362, -695, -545, -42, +42, -2, +24, -89, -290, -90, -302, -188, -238, -341, -202, -173, -185, -163, -115, -218,
+5574, +6674, +1997, +2183, +707, +407, +936, -776, -2047, -829, +914, +130, -285, -696, -573, -93, +9, -5, +50, +20, -186, -16, -247, -182, -271, -441, -315, -175, -47, +63, +58, -176,
+6045, +7217, +2109, +2333, +722, +394, +912, -1008, -2484, -1201, +940, +97, -533, -929, -717, -112, +69, +18, +107, +96, -204, -31, -262, -175, -230, -397, -254, -106, -72, -21, +17, -112,
+6856, +7935, +1962, +2469, +717, +365, +900, -1353, -2894, -1309, +1170, +26, -784, -1168, -861, -220, -61, -113, +16, +18, -323, -43, -253, -171, -282, -468, -311, -130, -56, +10, +78, -94,
+7514, +8789, +1982, +2483, +712, +262, +907, -1695, -3191, -1325, +1153, -23, -806, -1333, -975, -279, -115, -59, -35, -42, -373, -130, -392, -273, -384, -542, -334, -150, -38, +4, +93, -104,
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+16167, +11377, -5735, -169, +103, +84, -542, -1299, -1254, +1362, +1544, -1556, +422, -670, -103, -248, -1015, -1157, -377, +88, -926, +197, -106, +201, +13, +190, -354, -777, -517, -923, -458, +58,
+15468, +10670, -5248, +0, +346, +442, -693, -1493, -1502, +1200, +1468, -1209, +625, -745, +218, -158, -964, -895, -623, +11, -688, +93, -44, +335, -4, +183, -327, -757, -508, -710, -494, -108,
+11864, +10338, -3078, -621, +1333, +599, -268, -1143, -1003, +1077, +1244, -764, -17, -195, +147, -60, -510, -758, -332, -70, -719, -16, -15, +316, +287, +250, -355, -886, -747, -549, -54, +215,
};
/* HRIR Delays */
static const ALubyte defaultDelays[828] =
{
12, 12, 13, 14, 14, 14, 13, 12, 11, 10, 10, 10, 11, 12, 13, 14, 15, 15, 16, 16, 16, 15, 15, 14, 13, 12, 11, 10, 9, 8, 8, 8, 8, 8, 9, 10, 11, 12, 13, 14, 15, 16, 16, 17, 18, 18, 18, 18, 18, 17, 16, 16, 15, 14, 13, 12, 11, 10, 9, 8, 7, 7, 6, 6, 6, 6, 6, 7, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 18, 19, 20, 20, 20, 20, 20, 19, 19, 18, 17, 16, 15, 14, 13, 12, 11, 10, 9, 8, 8, 7, 6, 5, 5, 5, 4, 4, 4, 5, 5, 6, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 19, 20, 21, 21, 22, 22, 22, 22, 22, 21, 21, 20, 19, 19, 18, 17, 16, 15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 6, 5, 4, 4, 3, 3, 3, 3, 3, 3, 3, 4, 4, 5, 6, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 22, 23, 24, 24, 24, 24, 24, 23, 22, 22, 21, 20, 19, 18, 17, 16, 15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 5, 4, 4, 3, 3, 2, 2, 2, 2, 2, 2, 2, 3, 3, 4, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 19, 20, 21, 22, 23, 24, 24, 25, 26, 26, 26, 26, 26, 25, 24, 24, 23, 22, 21, 20, 19, 19, 18, 17, 16, 15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 6, 5, 4, 3, 3, 2, 2, 1, 1, 1, 1, 1, 1, 1, 1, 1, 2, 2, 3, 3, 4, 5, 6, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 24, 25, 26, 26, 27, 28, 28, 28, 27, 26, 26, 25, 24, 23, 22, 21, 20, 19, 18, 17, 16, 15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 5, 4, 4, 3, 2, 2, 1, 1, 1, 0, 0, 0, 0, 0, 1, 1, 1, 2, 2, 3, 4, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 29, 28, 27, 26, 25, 24, 23, 22, 21, 20, 19, 18, 17, 16, 15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 5, 4, 4, 3, 2, 2, 1, 1, 0, 0, 0, 0, 0, 0, 0, 1, 1, 2, 2, 3, 4, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 24, 25, 26, 26, 27, 28, 28, 28, 27, 26, 26, 25, 24, 23, 22, 21, 20, 19, 18, 17, 16, 15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 5, 4, 4, 3, 2, 2, 1, 1, 1, 0, 0, 0, 0, 0, 1, 1, 1, 2, 2, 3, 4, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 19, 20, 21, 22, 23, 24, 24, 25, 26, 26, 26, 26, 26, 25, 24, 24, 23, 22, 21, 20, 19, 19, 18, 17, 16, 15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 6, 5, 4, 3, 3, 2, 2, 1, 1, 1, 1, 1, 1, 1, 1, 1, 2, 2, 3, 3, 4, 5, 6, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 22, 23, 24, 24, 24, 24, 24, 23, 22, 22, 21, 20, 19, 18, 17, 16, 15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 5, 4, 4, 3, 3, 2, 2, 2, 2, 2, 2, 2, 3, 3, 4, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 19, 20, 21, 21, 22, 22, 22, 22, 22, 21, 21, 20, 19, 19, 18, 17, 16, 15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 6, 5, 4, 4, 3, 3, 3, 3, 3, 3, 3, 4, 4, 5, 6, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 18, 19, 20, 20, 20, 20, 20, 19, 19, 18, 17, 16, 15, 14, 13, 12, 11, 10, 9, 8, 8, 7, 6, 5, 5, 5, 4, 4, 4, 5, 5, 6, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 16, 17, 18, 18, 18, 18, 18, 17, 16, 16, 15, 14, 13, 12, 11, 10, 9, 8, 7, 7, 6, 6, 6, 6, 6, 7, 7, 8, 9, 10, 11, 12, 13, 14, 15, 15, 16, 16, 16, 15, 15, 14, 13, 12, 11, 10, 9, 8, 8, 8, 8, 8, 9, 10, 11, 12, 13, 14, 14, 14, 13, 12, 11, 10, 10, 10, 11, 12,
};
/* Default HRTF Definition */
static const struct Hrtf DefaultHrtf = {
44100, 32, 19, defaultAzCount, defaultEvOffset,
defaultCoeffs, defaultDelays, NULL
};
+421
View File
@@ -0,0 +1,421 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2007 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <math.h>
#include <stdlib.h>
#include <string.h>
#include <ctype.h>
#include <assert.h>
#include "alMain.h"
#include "AL/al.h"
#include "AL/alc.h"
#include "alSource.h"
#include "alBuffer.h"
#include "alListener.h"
#include "alAuxEffectSlot.h"
#include "alu.h"
#include "bs2b.h"
static __inline ALfloat Sample_ALbyte(ALbyte val)
{ return val * (1.0f/127.0f); }
static __inline ALfloat Sample_ALshort(ALshort val)
{ return val * (1.0f/32767.0f); }
static __inline ALfloat Sample_ALfloat(ALfloat val)
{ return val; }
#define DECL_TEMPLATE(T) \
static void Load_##T(ALfloat *dst, const T *src, ALuint srcstep, ALuint samples)\
{ \
ALuint i; \
for(i = 0;i < samples;i++) \
dst[i] = Sample_##T(src[i*srcstep]); \
}
DECL_TEMPLATE(ALbyte)
DECL_TEMPLATE(ALshort)
DECL_TEMPLATE(ALfloat)
#undef DECL_TEMPLATE
static void LoadData(ALfloat *dst, const ALvoid *src, ALuint srcstep, enum FmtType srctype, ALuint samples)
{
switch(srctype)
{
case FmtByte:
Load_ALbyte(dst, src, srcstep, samples);
break;
case FmtShort:
Load_ALshort(dst, src, srcstep, samples);
break;
case FmtFloat:
Load_ALfloat(dst, src, srcstep, samples);
break;
}
}
static void SilenceData(ALfloat *dst, ALuint samples)
{
ALuint i;
for(i = 0;i < samples;i++)
dst[i] = 0.0f;
}
static void Filter2P(FILTER *filter, ALuint chan, ALfloat *RESTRICT dst,
const ALfloat *RESTRICT src, ALuint numsamples)
{
ALuint i;
for(i = 0;i < numsamples;i++)
dst[i] = lpFilter2P(filter, chan, src[i]);
dst[i] = lpFilter2PC(filter, chan, src[i]);
}
ALvoid MixSource(ALsource *Source, ALCdevice *Device, ALuint SamplesToDo)
{
ALbufferlistitem *BufferListItem;
ALuint DataPosInt, DataPosFrac;
ALuint BuffersPlayed;
ALboolean Looping;
ALuint increment;
enum Resampler Resampler;
ALenum State;
ALuint OutPos;
ALuint NumChannels;
ALuint SampleSize;
ALint64 DataSize64;
ALuint chan, j;
/* Get source info */
State = Source->state;
BuffersPlayed = Source->BuffersPlayed;
DataPosInt = Source->position;
DataPosFrac = Source->position_fraction;
Looping = Source->Looping;
increment = Source->Params.Step;
Resampler = (increment==FRACTIONONE) ? PointResampler : Source->Resampler;
NumChannels = Source->NumChannels;
SampleSize = Source->SampleSize;
/* Get current buffer queue item */
BufferListItem = Source->queue;
for(j = 0;j < BuffersPlayed;j++)
BufferListItem = BufferListItem->next;
OutPos = 0;
do {
const ALuint BufferPrePadding = ResamplerPrePadding[Resampler];
const ALuint BufferPadding = ResamplerPadding[Resampler];
ALuint SrcBufferSize, DstBufferSize;
/* Figure out how many buffer bytes will be needed */
DataSize64 = SamplesToDo-OutPos+1;
DataSize64 *= increment;
DataSize64 += DataPosFrac+FRACTIONMASK;
DataSize64 >>= FRACTIONBITS;
DataSize64 += BufferPadding+BufferPrePadding;
SrcBufferSize = (ALuint)mini64(DataSize64, BUFFERSIZE);
/* Figure out how many samples we can actually mix from this. */
DataSize64 = SrcBufferSize;
DataSize64 -= BufferPadding+BufferPrePadding;
DataSize64 <<= FRACTIONBITS;
DataSize64 -= increment;
DataSize64 -= DataPosFrac;
DstBufferSize = (ALuint)((DataSize64+(increment-1)) / increment);
DstBufferSize = minu(DstBufferSize, (SamplesToDo-OutPos));
/* Some mixers like having a multiple of 4, so try to give that unless
* this is the last update. */
if(OutPos+DstBufferSize < SamplesToDo)
DstBufferSize &= ~3;
for(chan = 0;chan < NumChannels;chan++)
{
ALfloat *SrcData = Device->SampleData1;
ALfloat *ResampledData = Device->SampleData2;
ALuint SrcDataSize = 0;
if(Source->SourceType == AL_STATIC)
{
const ALbuffer *ALBuffer = Source->queue->buffer;
const ALubyte *Data = ALBuffer->data;
ALuint DataSize;
ALuint pos;
/* If current pos is beyond the loop range, do not loop */
if(Looping == AL_FALSE || DataPosInt >= (ALuint)ALBuffer->LoopEnd)
{
Looping = AL_FALSE;
if(DataPosInt >= BufferPrePadding)
pos = DataPosInt - BufferPrePadding;
else
{
DataSize = BufferPrePadding - DataPosInt;
DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
SilenceData(&SrcData[SrcDataSize], DataSize);
SrcDataSize += DataSize;
pos = 0;
}
/* Copy what's left to play in the source buffer, and clear the
* rest of the temp buffer */
DataSize = minu(SrcBufferSize - SrcDataSize, ALBuffer->SampleLen - pos);
LoadData(&SrcData[SrcDataSize], &Data[(pos*NumChannels + chan)*SampleSize],
NumChannels, ALBuffer->FmtType, DataSize);
SrcDataSize += DataSize;
SilenceData(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize);
SrcDataSize += SrcBufferSize - SrcDataSize;
}
else
{
ALuint LoopStart = ALBuffer->LoopStart;
ALuint LoopEnd = ALBuffer->LoopEnd;
if(DataPosInt >= LoopStart)
{
pos = DataPosInt-LoopStart;
while(pos < BufferPrePadding)
pos += LoopEnd-LoopStart;
pos -= BufferPrePadding;
pos += LoopStart;
}
else if(DataPosInt >= BufferPrePadding)
pos = DataPosInt - BufferPrePadding;
else
{
DataSize = BufferPrePadding - DataPosInt;
DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
SilenceData(&SrcData[SrcDataSize], DataSize);
SrcDataSize += DataSize;
pos = 0;
}
/* Copy what's left of this loop iteration, then copy repeats
* of the loop section */
DataSize = LoopEnd - pos;
DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
LoadData(&SrcData[SrcDataSize], &Data[(pos*NumChannels + chan)*SampleSize],
NumChannels, ALBuffer->FmtType, DataSize);
SrcDataSize += DataSize;
DataSize = LoopEnd-LoopStart;
while(SrcBufferSize > SrcDataSize)
{
DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
LoadData(&SrcData[SrcDataSize], &Data[(LoopStart*NumChannels + chan)*SampleSize],
NumChannels, ALBuffer->FmtType, DataSize);
SrcDataSize += DataSize;
}
}
}
else
{
/* Crawl the buffer queue to fill in the temp buffer */
ALbufferlistitem *tmpiter = BufferListItem;
ALuint pos;
if(DataPosInt >= BufferPrePadding)
pos = DataPosInt - BufferPrePadding;
else
{
pos = BufferPrePadding - DataPosInt;
while(pos > 0)
{
if(!tmpiter->prev && !Looping)
{
ALuint DataSize = minu(SrcBufferSize - SrcDataSize, pos);
SilenceData(&SrcData[SrcDataSize], DataSize);
SrcDataSize += DataSize;
pos = 0;
break;
}
if(tmpiter->prev)
tmpiter = tmpiter->prev;
else
{
while(tmpiter->next)
tmpiter = tmpiter->next;
}
if(tmpiter->buffer)
{
if((ALuint)tmpiter->buffer->SampleLen > pos)
{
pos = tmpiter->buffer->SampleLen - pos;
break;
}
pos -= tmpiter->buffer->SampleLen;
}
}
}
while(tmpiter && SrcBufferSize > SrcDataSize)
{
const ALbuffer *ALBuffer;
if((ALBuffer=tmpiter->buffer) != NULL)
{
const ALubyte *Data = ALBuffer->data;
ALuint DataSize = ALBuffer->SampleLen;
/* Skip the data already played */
if(DataSize <= pos)
pos -= DataSize;
else
{
Data += (pos*NumChannels + chan)*SampleSize;
DataSize -= pos;
pos -= pos;
DataSize = minu(SrcBufferSize - SrcDataSize, DataSize);
LoadData(&SrcData[SrcDataSize], Data, NumChannels,
ALBuffer->FmtType, DataSize);
SrcDataSize += DataSize;
}
}
tmpiter = tmpiter->next;
if(!tmpiter && Looping)
tmpiter = Source->queue;
else if(!tmpiter)
{
SilenceData(&SrcData[SrcDataSize], SrcBufferSize - SrcDataSize);
SrcDataSize += SrcBufferSize - SrcDataSize;
}
}
}
/* Now resample, then filter and mix to the appropriate outputs. */
Source->Params.Resample(&SrcData[BufferPrePadding], DataPosFrac,
increment, ResampledData, DstBufferSize);
{
DirectParams *directparms = &Source->Params.Direct;
Filter2P(&directparms->iirFilter, chan, SrcData, ResampledData,
DstBufferSize);
Source->Params.DryMix(directparms, SrcData, chan, OutPos,
SamplesToDo, DstBufferSize);
}
for(j = 0;j < Device->NumAuxSends;j++)
{
SendParams *sendparms = &Source->Params.Send[j];
if(!sendparms->Slot)
continue;
Filter2P(&sendparms->iirFilter, chan, SrcData, ResampledData,
DstBufferSize);
Source->Params.WetMix(sendparms, SrcData, OutPos,
SamplesToDo, DstBufferSize);
}
}
/* Update positions */
for(j = 0;j < DstBufferSize;j++)
{
DataPosFrac += increment;
DataPosInt += DataPosFrac>>FRACTIONBITS;
DataPosFrac &= FRACTIONMASK;
}
OutPos += DstBufferSize;
/* Handle looping sources */
while(1)
{
const ALbuffer *ALBuffer;
ALuint DataSize = 0;
ALuint LoopStart = 0;
ALuint LoopEnd = 0;
if((ALBuffer=BufferListItem->buffer) != NULL)
{
DataSize = ALBuffer->SampleLen;
LoopStart = ALBuffer->LoopStart;
LoopEnd = ALBuffer->LoopEnd;
if(LoopEnd > DataPosInt)
break;
}
if(Looping && Source->SourceType == AL_STATIC)
{
DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart;
break;
}
if(DataSize > DataPosInt)
break;
if(BufferListItem->next)
{
BufferListItem = BufferListItem->next;
BuffersPlayed++;
}
else if(Looping)
{
BufferListItem = Source->queue;
BuffersPlayed = 0;
}
else
{
State = AL_STOPPED;
BufferListItem = Source->queue;
BuffersPlayed = Source->BuffersInQueue;
DataPosInt = 0;
DataPosFrac = 0;
break;
}
DataPosInt -= DataSize;
}
} while(State == AL_PLAYING && OutPos < SamplesToDo);
/* Update source info */
Source->state = State;
Source->BuffersPlayed = BuffersPlayed;
Source->position = DataPosInt;
Source->position_fraction = DataPosFrac;
Source->Hrtf.Offset += OutPos;
if(State == AL_PLAYING)
Source->Hrtf.Counter = maxu(Source->Hrtf.Counter, OutPos) - OutPos;
else
{
Source->Hrtf.Counter = 0;
Source->Hrtf.Moving = AL_FALSE;
}
}
+131
View File
@@ -0,0 +1,131 @@
#include "config.h"
#include <assert.h>
#include "alMain.h"
#include "alu.h"
#include "alSource.h"
#include "alAuxEffectSlot.h"
static __inline ALfloat point32(const ALfloat *vals, ALuint frac)
{ return vals[0]; (void)frac; }
static __inline ALfloat lerp32(const ALfloat *vals, ALuint frac)
{ return lerp(vals[0], vals[1], frac * (1.0f/FRACTIONONE)); }
static __inline ALfloat cubic32(const ALfloat *vals, ALuint frac)
{ return cubic(vals[-1], vals[0], vals[1], vals[2], frac * (1.0f/FRACTIONONE)); }
void Resample_copy32_C(const ALfloat *data, ALuint frac,
ALuint increment, ALfloat *RESTRICT OutBuffer, ALuint BufferSize)
{
(void)frac;
assert(increment==FRACTIONONE);
memcpy(OutBuffer, data, (BufferSize+1)*sizeof(ALfloat));
}
#define DECL_TEMPLATE(Sampler) \
void Resample_##Sampler##_C(const ALfloat *data, ALuint frac, \
ALuint increment, ALfloat *RESTRICT OutBuffer, ALuint BufferSize) \
{ \
ALuint pos = 0; \
ALuint i; \
\
for(i = 0;i < BufferSize+1;i++) \
{ \
OutBuffer[i] = Sampler(data + pos, frac); \
\
frac += increment; \
pos += frac>>FRACTIONBITS; \
frac &= FRACTIONMASK; \
} \
}
DECL_TEMPLATE(point32)
DECL_TEMPLATE(lerp32)
DECL_TEMPLATE(cubic32)
#undef DECL_TEMPLATE
static __inline void ApplyCoeffsStep(ALuint Offset, ALfloat (*RESTRICT Values)[2],
const ALuint IrSize,
ALfloat (*RESTRICT Coeffs)[2],
const ALfloat (*RESTRICT CoeffStep)[2],
ALfloat left, ALfloat right)
{
ALuint c;
for(c = 0;c < IrSize;c++)
{
const ALuint off = (Offset+c)&HRIR_MASK;
Values[off][0] += Coeffs[c][0] * left;
Values[off][1] += Coeffs[c][1] * right;
Coeffs[c][0] += CoeffStep[c][0];
Coeffs[c][1] += CoeffStep[c][1];
}
}
static __inline void ApplyCoeffs(ALuint Offset, ALfloat (*RESTRICT Values)[2],
const ALuint IrSize,
ALfloat (*RESTRICT Coeffs)[2],
ALfloat left, ALfloat right)
{
ALuint c;
for(c = 0;c < IrSize;c++)
{
const ALuint off = (Offset+c)&HRIR_MASK;
Values[off][0] += Coeffs[c][0] * left;
Values[off][1] += Coeffs[c][1] * right;
}
}
#define SUFFIX C
#include "mixer_inc.c"
#undef SUFFIX
void MixDirect_C(const DirectParams *params, const ALfloat *RESTRICT data, ALuint srcchan,
ALuint OutPos, ALuint SamplesToDo, ALuint BufferSize)
{
ALfloat (*RESTRICT DryBuffer)[BUFFERSIZE] = params->OutBuffer;
ALfloat *RESTRICT ClickRemoval = params->ClickRemoval;
ALfloat *RESTRICT PendingClicks = params->PendingClicks;
ALfloat DrySend;
ALuint pos;
ALuint c;
for(c = 0;c < MaxChannels;c++)
{
DrySend = params->Gains[srcchan][c];
if(DrySend < 0.00001f)
continue;
if(OutPos == 0)
ClickRemoval[c] -= data[0]*DrySend;
for(pos = 0;pos < BufferSize;pos++)
DryBuffer[c][OutPos+pos] += data[pos]*DrySend;
if(OutPos+pos == SamplesToDo)
PendingClicks[c] += data[pos]*DrySend;
}
}
void MixSend_C(const SendParams *params, const ALfloat *RESTRICT data,
ALuint OutPos, ALuint SamplesToDo, ALuint BufferSize)
{
ALeffectslot *Slot = params->Slot;
ALfloat (*RESTRICT WetBuffer)[BUFFERSIZE] = Slot->WetBuffer;
ALfloat *RESTRICT WetClickRemoval = Slot->ClickRemoval;
ALfloat *RESTRICT WetPendingClicks = Slot->PendingClicks;
ALfloat WetSend = params->Gain;
ALuint pos;
if(WetSend < 0.00001f)
return;
if(OutPos == 0)
WetClickRemoval[0] -= data[0] * WetSend;
for(pos = 0;pos < BufferSize;pos++)
WetBuffer[0][OutPos+pos] += data[pos] * WetSend;
if(OutPos+pos == SamplesToDo)
WetPendingClicks[0] += data[pos] * WetSend;
}
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#ifndef MIXER_DEFS_H
#define MIXER_DEFS_H
#include "AL/alc.h"
#include "AL/al.h"
#include "alMain.h"
struct DirectParams;
struct SendParams;
/* C resamplers */
void Resample_copy32_C(const ALfloat *src, ALuint frac, ALuint increment, ALfloat *RESTRICT dst, ALuint dstlen);
void Resample_point32_C(const ALfloat *src, ALuint frac, ALuint increment, ALfloat *RESTRICT dst, ALuint dstlen);
void Resample_lerp32_C(const ALfloat *src, ALuint frac, ALuint increment, ALfloat *RESTRICT dst, ALuint dstlen);
void Resample_cubic32_C(const ALfloat *src, ALuint frac, ALuint increment, ALfloat *RESTRICT dst, ALuint dstlen);
/* C mixers */
void MixDirect_Hrtf_C(const struct DirectParams*,const ALfloat*RESTRICT,ALuint,ALuint,ALuint,ALuint);
void MixDirect_C(const struct DirectParams*,const ALfloat*RESTRICT,ALuint,ALuint,ALuint,ALuint);
void MixSend_C(const struct SendParams*,const ALfloat*RESTRICT,ALuint,ALuint,ALuint);
/* SSE mixers */
void MixDirect_Hrtf_SSE(const struct DirectParams*,const ALfloat*RESTRICT,ALuint,ALuint,ALuint,ALuint);
void MixDirect_SSE(const struct DirectParams*,const ALfloat*RESTRICT,ALuint,ALuint,ALuint,ALuint);
void MixSend_SSE(const struct SendParams*,const ALfloat*RESTRICT,ALuint,ALuint,ALuint);
/* Neon mixers */
void MixDirect_Hrtf_Neon(const struct DirectParams*,const ALfloat*RESTRICT,ALuint,ALuint,ALuint,ALuint);
#endif /* MIXER_DEFS_H */
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#include "config.h"
#include "alMain.h"
#include "alSource.h"
#include "mixer_defs.h"
#ifdef __GNUC__
#define LIKELY(x) __builtin_expect(!!(x), 1)
#define UNLIKELY(x) __builtin_expect(!!(x), 0)
#else
#define LIKELY(x) (x)
#define UNLIKELY(x) (x)
#endif
#define REAL_MERGE2(a,b) a##b
#define MERGE2(a,b) REAL_MERGE2(a,b)
#define MixDirect_Hrtf MERGE2(MixDirect_Hrtf_,SUFFIX)
static __inline void ApplyCoeffsStep(ALuint Offset, ALfloat (*RESTRICT Values)[2],
const ALuint irSize,
ALfloat (*RESTRICT Coeffs)[2],
const ALfloat (*RESTRICT CoeffStep)[2],
ALfloat left, ALfloat right);
static __inline void ApplyCoeffs(ALuint Offset, ALfloat (*RESTRICT Values)[2],
const ALuint irSize,
ALfloat (*RESTRICT Coeffs)[2],
ALfloat left, ALfloat right);
void MixDirect_Hrtf(const DirectParams *params, const ALfloat *RESTRICT data, ALuint srcchan,
ALuint OutPos, ALuint SamplesToDo, ALuint BufferSize)
{
ALfloat (*RESTRICT DryBuffer)[BUFFERSIZE] = params->OutBuffer;
ALfloat *RESTRICT ClickRemoval = params->ClickRemoval;
ALfloat *RESTRICT PendingClicks = params->PendingClicks;
const ALuint IrSize = params->Hrtf.Params.IrSize;
const ALint *RESTRICT DelayStep = params->Hrtf.Params.DelayStep;
const ALfloat (*RESTRICT CoeffStep)[2] = params->Hrtf.Params.CoeffStep;
const ALfloat (*RESTRICT TargetCoeffs)[2] = params->Hrtf.Params.Coeffs[srcchan];
const ALuint *RESTRICT TargetDelay = params->Hrtf.Params.Delay[srcchan];
ALfloat *RESTRICT History = params->Hrtf.State->History[srcchan];
ALfloat (*RESTRICT Values)[2] = params->Hrtf.State->Values[srcchan];
ALint Counter = maxu(params->Hrtf.State->Counter, OutPos) - OutPos;
ALuint Offset = params->Hrtf.State->Offset + OutPos;
ALIGN(16) ALfloat Coeffs[HRIR_LENGTH][2];
ALuint Delay[2];
ALfloat left, right;
ALuint pos;
ALuint c;
pos = 0;
for(c = 0;c < IrSize;c++)
{
Coeffs[c][0] = TargetCoeffs[c][0] - (CoeffStep[c][0]*Counter);
Coeffs[c][1] = TargetCoeffs[c][1] - (CoeffStep[c][1]*Counter);
}
Delay[0] = TargetDelay[0] - (DelayStep[0]*Counter);
Delay[1] = TargetDelay[1] - (DelayStep[1]*Counter);
if(LIKELY(OutPos == 0))
{
History[Offset&SRC_HISTORY_MASK] = data[pos];
left = lerp(History[(Offset-(Delay[0]>>HRTFDELAY_BITS))&SRC_HISTORY_MASK],
History[(Offset-(Delay[0]>>HRTFDELAY_BITS)-1)&SRC_HISTORY_MASK],
(Delay[0]&HRTFDELAY_MASK)*(1.0f/HRTFDELAY_FRACONE));
right = lerp(History[(Offset-(Delay[1]>>HRTFDELAY_BITS))&SRC_HISTORY_MASK],
History[(Offset-(Delay[1]>>HRTFDELAY_BITS)-1)&SRC_HISTORY_MASK],
(Delay[1]&HRTFDELAY_MASK)*(1.0f/HRTFDELAY_FRACONE));
ClickRemoval[FrontLeft] -= Values[(Offset+1)&HRIR_MASK][0] +
Coeffs[0][0] * left;
ClickRemoval[FrontRight] -= Values[(Offset+1)&HRIR_MASK][1] +
Coeffs[0][1] * right;
}
for(pos = 0;pos < BufferSize && Counter > 0;pos++)
{
History[Offset&SRC_HISTORY_MASK] = data[pos];
left = lerp(History[(Offset-(Delay[0]>>HRTFDELAY_BITS))&SRC_HISTORY_MASK],
History[(Offset-(Delay[0]>>HRTFDELAY_BITS)-1)&SRC_HISTORY_MASK],
(Delay[0]&HRTFDELAY_MASK)*(1.0f/HRTFDELAY_FRACONE));
right = lerp(History[(Offset-(Delay[1]>>HRTFDELAY_BITS))&SRC_HISTORY_MASK],
History[(Offset-(Delay[1]>>HRTFDELAY_BITS)-1)&SRC_HISTORY_MASK],
(Delay[1]&HRTFDELAY_MASK)*(1.0f/HRTFDELAY_FRACONE));
Delay[0] += DelayStep[0];
Delay[1] += DelayStep[1];
Values[(Offset+IrSize)&HRIR_MASK][0] = 0.0f;
Values[(Offset+IrSize)&HRIR_MASK][1] = 0.0f;
Offset++;
ApplyCoeffsStep(Offset, Values, IrSize, Coeffs, CoeffStep, left, right);
DryBuffer[FrontLeft][OutPos] += Values[Offset&HRIR_MASK][0];
DryBuffer[FrontRight][OutPos] += Values[Offset&HRIR_MASK][1];
OutPos++;
Counter--;
}
Delay[0] >>= HRTFDELAY_BITS;
Delay[1] >>= HRTFDELAY_BITS;
for(;pos < BufferSize;pos++)
{
History[Offset&SRC_HISTORY_MASK] = data[pos];
left = History[(Offset-Delay[0])&SRC_HISTORY_MASK];
right = History[(Offset-Delay[1])&SRC_HISTORY_MASK];
Values[(Offset+IrSize)&HRIR_MASK][0] = 0.0f;
Values[(Offset+IrSize)&HRIR_MASK][1] = 0.0f;
Offset++;
ApplyCoeffs(Offset, Values, IrSize, Coeffs, left, right);
DryBuffer[FrontLeft][OutPos] += Values[Offset&HRIR_MASK][0];
DryBuffer[FrontRight][OutPos] += Values[Offset&HRIR_MASK][1];
OutPos++;
}
if(LIKELY(OutPos == SamplesToDo))
{
History[Offset&SRC_HISTORY_MASK] = data[pos];
left = History[(Offset-Delay[0])&SRC_HISTORY_MASK];
right = History[(Offset-Delay[1])&SRC_HISTORY_MASK];
PendingClicks[FrontLeft] += Values[(Offset+1)&HRIR_MASK][0] +
Coeffs[0][0] * left;
PendingClicks[FrontRight] += Values[(Offset+1)&HRIR_MASK][1] +
Coeffs[0][1] * right;
}
}
#undef MixDirect_Hrtf
#undef MERGE2
#undef REAL_MERGE2
#undef UNLIKELY
#undef LIKELY
+61
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#include "config.h"
#ifdef HAVE_ARM_NEON_H
#include <arm_neon.h>
#endif
#include "AL/al.h"
#include "AL/alc.h"
#include "alMain.h"
#include "alu.h"
static __inline void ApplyCoeffsStep(ALuint Offset, ALfloat (*RESTRICT Values)[2],
const ALuint IrSize,
ALfloat (*RESTRICT Coeffs)[2],
const ALfloat (*RESTRICT CoeffStep)[2],
ALfloat left, ALfloat right)
{
ALuint c;
for(c = 0;c < IrSize;c++)
{
const ALuint off = (Offset+c)&HRIR_MASK;
Values[off][0] += Coeffs[c][0] * left;
Values[off][1] += Coeffs[c][1] * right;
Coeffs[c][0] += CoeffStep[c][0];
Coeffs[c][1] += CoeffStep[c][1];
}
}
static __inline void ApplyCoeffs(ALuint Offset, ALfloat (*RESTRICT Values)[2],
const ALuint IrSize,
ALfloat (*RESTRICT Coeffs)[2],
ALfloat left, ALfloat right)
{
ALuint c;
float32x4_t leftright4;
{
float32x2_t leftright2 = vdup_n_f32(0.0);
leftright2 = vset_lane_f32(left, leftright2, 0);
leftright2 = vset_lane_f32(right, leftright2, 1);
leftright4 = vcombine_f32(leftright2, leftright2);
}
for(c = 0;c < IrSize;c += 2)
{
const ALuint o0 = (Offset+c)&HRIR_MASK;
const ALuint o1 = (o0+1)&HRIR_MASK;
float32x4_t vals = vcombine_f32(vld1_f32((float32_t*)&Values[o0][0]),
vld1_f32((float32_t*)&Values[o1][0]));
float32x4_t coefs = vld1q_f32((float32_t*)&Coeffs[c][0]);
vals = vmlaq_f32(vals, coefs, leftright4);
vst1_f32((float32_t*)&Values[o0][0], vget_low_f32(vals));
vst1_f32((float32_t*)&Values[o1][0], vget_high_f32(vals));
}
}
#define SUFFIX Neon
#include "mixer_inc.c"
#undef SUFFIX
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#include "config.h"
#ifdef HAVE_XMMINTRIN_H
#include <xmmintrin.h>
#endif
#include "AL/al.h"
#include "AL/alc.h"
#include "alMain.h"
#include "alu.h"
#include "alSource.h"
#include "alAuxEffectSlot.h"
#include "mixer_defs.h"
static __inline void ApplyCoeffsStep(ALuint Offset, ALfloat (*RESTRICT Values)[2],
const ALuint IrSize,
ALfloat (*RESTRICT Coeffs)[2],
const ALfloat (*RESTRICT CoeffStep)[2],
ALfloat left, ALfloat right)
{
const __m128 lrlr = { left, right, left, right };
__m128 coeffs, deltas, imp0, imp1;
__m128 vals = _mm_setzero_ps();
ALuint i;
if((Offset&1))
{
const ALuint o0 = Offset&HRIR_MASK;
const ALuint o1 = (Offset+IrSize-1)&HRIR_MASK;
coeffs = _mm_load_ps(&Coeffs[0][0]);
deltas = _mm_load_ps(&CoeffStep[0][0]);
vals = _mm_loadl_pi(vals, (__m64*)&Values[o0][0]);
imp0 = _mm_mul_ps(lrlr, coeffs);
coeffs = _mm_add_ps(coeffs, deltas);
vals = _mm_add_ps(imp0, vals);
_mm_store_ps(&Coeffs[0][0], coeffs);
_mm_storel_pi((__m64*)&Values[o0][0], vals);
for(i = 1;i < IrSize-1;i += 2)
{
const ALuint o2 = (Offset+i)&HRIR_MASK;
coeffs = _mm_load_ps(&Coeffs[i+1][0]);
deltas = _mm_load_ps(&CoeffStep[i+1][0]);
vals = _mm_load_ps(&Values[o2][0]);
imp1 = _mm_mul_ps(lrlr, coeffs);
coeffs = _mm_add_ps(coeffs, deltas);
imp0 = _mm_shuffle_ps(imp0, imp1, _MM_SHUFFLE(1, 0, 3, 2));
vals = _mm_add_ps(imp0, vals);
_mm_store_ps(&Coeffs[i+1][0], coeffs);
_mm_store_ps(&Values[o2][0], vals);
imp0 = imp1;
}
vals = _mm_loadl_pi(vals, (__m64*)&Values[o1][0]);
imp0 = _mm_movehl_ps(imp0, imp0);
vals = _mm_add_ps(imp0, vals);
_mm_storel_pi((__m64*)&Values[o1][0], vals);
}
else
{
for(i = 0;i < IrSize;i += 2)
{
const ALuint o = (Offset + i)&HRIR_MASK;
coeffs = _mm_load_ps(&Coeffs[i][0]);
deltas = _mm_load_ps(&CoeffStep[i][0]);
vals = _mm_load_ps(&Values[o][0]);
imp0 = _mm_mul_ps(lrlr, coeffs);
coeffs = _mm_add_ps(coeffs, deltas);
vals = _mm_add_ps(imp0, vals);
_mm_store_ps(&Coeffs[i][0], coeffs);
_mm_store_ps(&Values[o][0], vals);
}
}
}
static __inline void ApplyCoeffs(ALuint Offset, ALfloat (*RESTRICT Values)[2],
const ALuint IrSize,
ALfloat (*RESTRICT Coeffs)[2],
ALfloat left, ALfloat right)
{
const __m128 lrlr = { left, right, left, right };
__m128 vals = _mm_setzero_ps();
__m128 coeffs;
ALuint i;
if((Offset&1))
{
const ALuint o0 = Offset&HRIR_MASK;
const ALuint o1 = (Offset+IrSize-1)&HRIR_MASK;
__m128 imp0, imp1;
coeffs = _mm_load_ps(&Coeffs[0][0]);
vals = _mm_loadl_pi(vals, (__m64*)&Values[o0][0]);
imp0 = _mm_mul_ps(lrlr, coeffs);
vals = _mm_add_ps(imp0, vals);
_mm_storel_pi((__m64*)&Values[o0][0], vals);
for(i = 1;i < IrSize-1;i += 2)
{
const ALuint o2 = (Offset+i)&HRIR_MASK;
coeffs = _mm_load_ps(&Coeffs[i+1][0]);
vals = _mm_load_ps(&Values[o2][0]);
imp1 = _mm_mul_ps(lrlr, coeffs);
imp0 = _mm_shuffle_ps(imp0, imp1, _MM_SHUFFLE(1, 0, 3, 2));
vals = _mm_add_ps(imp0, vals);
_mm_store_ps(&Values[o2][0], vals);
imp0 = imp1;
}
vals = _mm_loadl_pi(vals, (__m64*)&Values[o1][0]);
imp0 = _mm_movehl_ps(imp0, imp0);
vals = _mm_add_ps(imp0, vals);
_mm_storel_pi((__m64*)&Values[o1][0], vals);
}
else
{
for(i = 0;i < IrSize;i += 2)
{
const ALuint o = (Offset + i)&HRIR_MASK;
coeffs = _mm_load_ps(&Coeffs[i][0]);
vals = _mm_load_ps(&Values[o][0]);
vals = _mm_add_ps(vals, _mm_mul_ps(lrlr, coeffs));
_mm_store_ps(&Values[o][0], vals);
}
}
}
#define SUFFIX SSE
#include "mixer_inc.c"
#undef SUFFIX
void MixDirect_SSE(const DirectParams *params, const ALfloat *RESTRICT data, ALuint srcchan,
ALuint OutPos, ALuint SamplesToDo, ALuint BufferSize)
{
ALfloat (*RESTRICT DryBuffer)[BUFFERSIZE] = params->OutBuffer;
ALfloat *RESTRICT ClickRemoval = params->ClickRemoval;
ALfloat *RESTRICT PendingClicks = params->PendingClicks;
ALfloat DrySend;
ALuint pos;
ALuint c;
for(c = 0;c < MaxChannels;c++)
{
__m128 gain;
DrySend = params->Gains[srcchan][c];
if(DrySend < 0.00001f)
continue;
if(OutPos == 0)
ClickRemoval[c] -= data[0]*DrySend;
gain = _mm_set1_ps(DrySend);
for(pos = 0;pos < BufferSize-3;pos += 4)
{
const __m128 val4 = _mm_load_ps(&data[pos]);
__m128 dry4 = _mm_load_ps(&DryBuffer[c][OutPos+pos]);
dry4 = _mm_add_ps(dry4, _mm_mul_ps(val4, gain));
_mm_store_ps(&DryBuffer[c][OutPos+pos], dry4);
}
for(;pos < BufferSize;pos++)
DryBuffer[c][OutPos+pos] += data[pos]*DrySend;
if(OutPos+pos == SamplesToDo)
PendingClicks[c] += data[pos]*DrySend;
}
}
void MixSend_SSE(const SendParams *params, const ALfloat *RESTRICT data,
ALuint OutPos, ALuint SamplesToDo, ALuint BufferSize)
{
ALeffectslot *Slot = params->Slot;
ALfloat (*RESTRICT WetBuffer)[BUFFERSIZE] = Slot->WetBuffer;
ALfloat *RESTRICT WetClickRemoval = Slot->ClickRemoval;
ALfloat *RESTRICT WetPendingClicks = Slot->PendingClicks;
const ALfloat WetGain = params->Gain;
__m128 gain;
ALuint pos;
if(WetGain < 0.00001f)
return;
if(OutPos == 0)
WetClickRemoval[0] -= data[0] * WetGain;
gain = _mm_set1_ps(WetGain);
for(pos = 0;pos < BufferSize-3;pos+=4)
{
const __m128 val4 = _mm_load_ps(&data[pos]);
__m128 wet4 = _mm_load_ps(&WetBuffer[0][OutPos+pos]);
wet4 = _mm_add_ps(wet4, _mm_mul_ps(val4, gain));
_mm_store_ps(&WetBuffer[0][OutPos+pos], wet4);
}
for(;pos < BufferSize;pos++)
WetBuffer[0][OutPos+pos] += data[pos] * WetGain;
if(OutPos+pos == SamplesToDo)
WetPendingClicks[0] += data[pos] * WetGain;
}
-487
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@@ -1,487 +0,0 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2007 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include <sys/ioctl.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <stdlib.h>
#include <stdio.h>
#include <memory.h>
#include <unistd.h>
#include <errno.h>
#include <math.h>
#include "alMain.h"
#include "AL/al.h"
#include "AL/alc.h"
#include <sys/soundcard.h>
/*
* The OSS documentation talks about SOUND_MIXER_READ, but the header
* only contains MIXER_READ. Play safe. Same for WRITE.
*/
#ifndef SOUND_MIXER_READ
#define SOUND_MIXER_READ MIXER_READ
#endif
#ifndef SOUND_MIXER_WRITE
#define SOUND_MIXER_WRITE MIXER_WRITE
#endif
static char *oss_device;
static char *oss_device_capture;
typedef struct {
int fd;
int killNow;
ALvoid *thread;
ALubyte *mix_data;
int data_size;
int silence;
RingBuffer *ring;
int doCapture;
} oss_data;
static int log2i(ALCuint x)
{
int y = 0;
while (x > 1)
{
x >>= 1;
y++;
}
return y;
}
static ALuint OSSProc(ALvoid *ptr)
{
ALCdevice *pDevice = (ALCdevice*)ptr;
oss_data *data = (oss_data*)pDevice->ExtraData;
int remaining = 0;
int wrote;
while(!data->killNow)
{
int len = data->data_size - remaining;
if(len > 0)
{
SuspendContext(NULL);
aluMixData(pDevice->Context, data->mix_data+remaining, len, pDevice->Format);
ProcessContext(NULL);
}
remaining += len;
wrote = write(data->fd, data->mix_data, remaining);
if(wrote < 0)
{
AL_PRINT("write failed: %s\n", strerror(errno));
remaining = 0;
}
else if(wrote > 0)
{
remaining -= wrote;
if(remaining > 0)
memmove(data->mix_data, data->mix_data+wrote, remaining);
}
else
usleep(1000);
}
return 0;
}
static ALuint OSSCaptureProc(ALvoid *ptr)
{
ALCdevice *pDevice = (ALCdevice*)ptr;
oss_data *data = (oss_data*)pDevice->ExtraData;
int frameSize;
int amt;
frameSize = aluBytesFromFormat(pDevice->Format);
frameSize *= aluChannelsFromFormat(pDevice->Format);
while(!data->killNow)
{
amt = read(data->fd, data->mix_data, data->data_size);
if(amt < 0)
{
AL_PRINT("read failed: %s\n", strerror(errno));
break;
}
if(amt == 0)
{
usleep(1000);
continue;
}
if(data->doCapture)
WriteRingBuffer(data->ring, data->mix_data, amt/frameSize);
}
return 0;
}
static ALCboolean oss_open_playback(ALCdevice *device, const ALCchar *deviceName)
{
int numFragmentsLogSize;
int log2FragmentSize;
unsigned int periods;
audio_buf_info info;
char driver[64];
int numChannels;
oss_data *data;
int ossFormat;
int ossSpeed;
char *err;
int i;
strncpy(driver, GetConfigValue("oss", "device", "/dev/dsp"), sizeof(driver)-1);
driver[sizeof(driver)-1] = 0;
if(deviceName)
{
if(strcmp(deviceName, oss_device))
return ALC_FALSE;
device->szDeviceName = oss_device;
}
else
device->szDeviceName = oss_device;
data = (oss_data*)calloc(1, sizeof(oss_data));
data->killNow = 0;
data->fd = open(driver, O_WRONLY);
if(data->fd == -1)
{
free(data);
AL_PRINT("Could not open %s: %s\n", driver, strerror(errno));
return ALC_FALSE;
}
switch(aluBytesFromFormat(device->Format))
{
case 1:
data->silence = 0x80;
ossFormat = AFMT_U8;
break;
case 2:
data->silence = 0;
ossFormat = AFMT_S16_NE;
break;
default:
ossFormat = -1;
AL_PRINT("Unknown format?! %x\n", device->Format);
}
periods = GetConfigValueInt("oss", "periods", 4);
if((int)periods <= 0)
periods = 4;
numChannels = device->Channels;
ossSpeed = device->Frequency;
log2FragmentSize = log2i(device->UpdateFreq * device->FrameSize / periods);
/* according to the OSS spec, 16 bytes are the minimum */
if (log2FragmentSize < 4)
log2FragmentSize = 4;
numFragmentsLogSize = (periods << 16) | log2FragmentSize;
#define ok(func, str) (i=(func),((i<0)?(err=(str)),0:1))
if (!(ok(ioctl(data->fd, SNDCTL_DSP_SETFRAGMENT, &numFragmentsLogSize), "set fragment") &&
ok(ioctl(data->fd, SNDCTL_DSP_SETFMT, &ossFormat), "set format") &&
ok(ioctl(data->fd, SNDCTL_DSP_CHANNELS, &numChannels), "set channels") &&
ok(ioctl(data->fd, SNDCTL_DSP_SPEED, &ossSpeed), "set speed") &&
ok(ioctl(data->fd, SNDCTL_DSP_GETOSPACE, &info), "get space")))
{
AL_PRINT("%s failed: %s\n", err, strerror(errno));
close(data->fd);
free(data);
return ALC_FALSE;
}
#undef ok
device->Frequency = ossSpeed;
if((int)device->Channels != numChannels)
{
AL_PRINT("Could not set %d channels, got %d instead\n", device->Channels, numChannels);
close(data->fd);
free(data);
return ALC_FALSE;
}
if(!((ossFormat == AFMT_U8 && aluBytesFromFormat(device->Format) == 1) ||
(ossFormat == AFMT_S16_NE && aluBytesFromFormat(device->Format) == 2)))
{
AL_PRINT("Could not set %d-bit output, got format %#x\n", aluBytesFromFormat(device->Format)*8, ossFormat);
close(data->fd);
free(data);
return ALC_FALSE;
}
device->UpdateFreq = info.fragsize / device->FrameSize;
data->data_size = device->UpdateFreq * device->FrameSize;
data->mix_data = calloc(1, data->data_size);
device->MaxNoOfSources = 256;
device->ExtraData = data;
data->thread = StartThread(OSSProc, device);
if(data->thread == NULL)
{
device->ExtraData = NULL;
free(data->mix_data);
free(data);
return ALC_FALSE;
}
return ALC_TRUE;
}
static void oss_close_playback(ALCdevice *device)
{
oss_data *data = (oss_data*)device->ExtraData;
data->killNow = 1;
StopThread(data->thread);
close(data->fd);
free(data->mix_data);
free(data);
device->ExtraData = NULL;
}
static ALCboolean oss_open_capture(ALCdevice *device, const ALCchar *deviceName, ALCuint frequency, ALCenum format, ALCsizei SampleSize)
{
int numFragmentsLogSize;
int log2FragmentSize;
unsigned int periods;
audio_buf_info info;
int numChannels;
char driver[64];
oss_data *data;
int ossFormat;
int ossSpeed;
char *err;
int i;
strncpy(driver, GetConfigValue("oss", "capture", "/dev/dsp"), sizeof(driver)-1);
driver[sizeof(driver)-1] = 0;
if(deviceName)
{
if(strcmp(deviceName, oss_device_capture))
return ALC_FALSE;
device->szDeviceName = oss_device_capture;
}
else
device->szDeviceName = oss_device_capture;
data = (oss_data*)calloc(1, sizeof(oss_data));
data->killNow = 0;
data->fd = open(driver, O_RDONLY);
if(data->fd == -1)
{
free(data);
AL_PRINT("Could not open %s: %s\n", driver, strerror(errno));
return ALC_FALSE;
}
switch(aluBytesFromFormat(format))
{
case 1:
data->silence = 0x80;
ossFormat = AFMT_U8;
break;
case 2:
data->silence = 0;
ossFormat = AFMT_S16_NE;
break;
default:
ossFormat = -1;
AL_PRINT("Unknown format?! %x\n", device->Format);
}
periods = 4;
numChannels = device->Channels;
ossSpeed = frequency;
log2FragmentSize = log2i(device->UpdateFreq * device->FrameSize / periods);
/* according to the OSS spec, 16 bytes are the minimum */
if (log2FragmentSize < 4)
log2FragmentSize = 4;
numFragmentsLogSize = (periods << 16) | log2FragmentSize;
#define ok(func, str) (i=(func),((i<0)?(err=(str)),0:1))
if (!(ok(ioctl(data->fd, SNDCTL_DSP_SETFRAGMENT, &numFragmentsLogSize), "set fragment") &&
ok(ioctl(data->fd, SNDCTL_DSP_SETFMT, &ossFormat), "set format") &&
ok(ioctl(data->fd, SNDCTL_DSP_CHANNELS, &numChannels), "set channels") &&
ok(ioctl(data->fd, SNDCTL_DSP_SPEED, &ossSpeed), "set speed") &&
ok(ioctl(data->fd, SNDCTL_DSP_GETISPACE, &info), "get space")))
{
AL_PRINT("%s failed: %s\n", err, strerror(errno));
close(data->fd);
free(data);
return ALC_FALSE;
}
#undef ok
device->Channels = numChannels;
device->Frequency = ossSpeed;
device->Format = 0;
if(ossFormat == AFMT_U8)
{
if(device->Channels == 1)
{
device->Format = AL_FORMAT_MONO8;
device->FrameSize = 1;
}
else if(device->Channels == 2)
{
device->Format = AL_FORMAT_STEREO8;
device->FrameSize = 2;
}
else if(device->Channels == 4)
{
device->Format = AL_FORMAT_QUAD8;
device->FrameSize = 4;
}
}
else if(ossFormat == AFMT_S16_NE)
{
if(device->Channels == 1)
{
device->Format = AL_FORMAT_MONO16;
device->FrameSize = 2;
}
else if(device->Channels == 2)
{
device->Format = AL_FORMAT_STEREO16;
device->FrameSize = 4;
}
else if(device->Channels == 4)
{
device->Format = AL_FORMAT_QUAD16;
device->FrameSize = 8;
}
}
if(!device->Format)
{
AL_PRINT("returned unknown format: %#x %d!\n", ossFormat, numChannels);
close(data->fd);
free(data);
return ALC_FALSE;
}
data->ring = CreateRingBuffer(device->FrameSize, SampleSize);
if(!data->ring)
{
AL_PRINT("ring buffer create failed\n");
close(data->fd);
free(data);
return ALC_FALSE;
}
device->UpdateFreq = info.fragsize / device->FrameSize;
data->data_size = device->UpdateFreq * device->FrameSize * info.fragments;
data->mix_data = calloc(1, data->data_size);
device->ExtraData = data;
data->thread = StartThread(OSSCaptureProc, device);
if(data->thread == NULL)
{
device->ExtraData = NULL;
free(data->mix_data);
free(data);
return ALC_FALSE;
}
return ALC_TRUE;
}
static void oss_close_capture(ALCdevice *device)
{
oss_data *data = (oss_data*)device->ExtraData;
data->killNow = 1;
StopThread(data->thread);
close(data->fd);
DestroyRingBuffer(data->ring);
free(data->mix_data);
free(data);
device->ExtraData = NULL;
}
static void oss_start_capture(ALCdevice *pDevice)
{
oss_data *data = (oss_data*)pDevice->ExtraData;
data->doCapture = 1;
}
static void oss_stop_capture(ALCdevice *pDevice)
{
oss_data *data = (oss_data*)pDevice->ExtraData;
data->doCapture = 0;
}
static void oss_capture_samples(ALCdevice *pDevice, ALCvoid *pBuffer, ALCuint lSamples)
{
oss_data *data = (oss_data*)pDevice->ExtraData;
if(lSamples <= (ALCuint)RingBufferSize(data->ring))
ReadRingBuffer(data->ring, pBuffer, lSamples);
else
SetALCError(ALC_INVALID_VALUE);
}
static ALCuint oss_available_samples(ALCdevice *pDevice)
{
oss_data *data = (oss_data*)pDevice->ExtraData;
return RingBufferSize(data->ring);
}
BackendFuncs oss_funcs = {
oss_open_playback,
oss_close_playback,
oss_open_capture,
oss_close_capture,
oss_start_capture,
oss_stop_capture,
oss_capture_samples,
oss_available_samples
};
void alc_oss_init(BackendFuncs *func_list)
{
*func_list = oss_funcs;
oss_device = AppendDeviceList("OSS Software");
AppendAllDeviceList(oss_device);
oss_device_capture = AppendCaptureDeviceList("OSS Capture");
}
+450
View File
@@ -0,0 +1,450 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2010 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <math.h>
#include <stdlib.h>
#include <string.h>
#include <ctype.h>
#include <assert.h>
#include "alMain.h"
#include "AL/al.h"
#include "AL/alc.h"
#include "alu.h"
static void SetSpeakerArrangement(const char *name, ALfloat SpeakerAngle[MaxChannels],
enum Channel Speaker2Chan[MaxChannels], ALint chans)
{
char *confkey, *next;
char *layout_str;
char *sep, *end;
enum Channel val;
const char *str;
int i;
if(!ConfigValueStr(NULL, name, &str) && !ConfigValueStr(NULL, "layout", &str))
return;
layout_str = strdup(str);
next = confkey = layout_str;
while(next && *next)
{
confkey = next;
next = strchr(confkey, ',');
if(next)
{
*next = 0;
do {
next++;
} while(isspace(*next) || *next == ',');
}
sep = strchr(confkey, '=');
if(!sep || confkey == sep)
{
ERR("Malformed speaker key: %s\n", confkey);
continue;
}
end = sep - 1;
while(isspace(*end) && end != confkey)
end--;
*(++end) = 0;
if(strcmp(confkey, "fl") == 0 || strcmp(confkey, "front-left") == 0)
val = FrontLeft;
else if(strcmp(confkey, "fr") == 0 || strcmp(confkey, "front-right") == 0)
val = FrontRight;
else if(strcmp(confkey, "fc") == 0 || strcmp(confkey, "front-center") == 0)
val = FrontCenter;
else if(strcmp(confkey, "bl") == 0 || strcmp(confkey, "back-left") == 0)
val = BackLeft;
else if(strcmp(confkey, "br") == 0 || strcmp(confkey, "back-right") == 0)
val = BackRight;
else if(strcmp(confkey, "bc") == 0 || strcmp(confkey, "back-center") == 0)
val = BackCenter;
else if(strcmp(confkey, "sl") == 0 || strcmp(confkey, "side-left") == 0)
val = SideLeft;
else if(strcmp(confkey, "sr") == 0 || strcmp(confkey, "side-right") == 0)
val = SideRight;
else
{
ERR("Unknown speaker for %s: \"%s\"\n", name, confkey);
continue;
}
*(sep++) = 0;
while(isspace(*sep))
sep++;
for(i = 0;i < chans;i++)
{
if(Speaker2Chan[i] == val)
{
long angle = strtol(sep, NULL, 10);
if(angle >= -180 && angle <= 180)
SpeakerAngle[i] = angle * F_PI/180.0f;
else
ERR("Invalid angle for speaker \"%s\": %ld\n", confkey, angle);
break;
}
}
}
free(layout_str);
layout_str = NULL;
for(i = 0;i < chans;i++)
{
int min = i;
int i2;
for(i2 = i+1;i2 < chans;i2++)
{
if(SpeakerAngle[i2] < SpeakerAngle[min])
min = i2;
}
if(min != i)
{
ALfloat tmpf;
enum Channel tmpc;
tmpf = SpeakerAngle[i];
SpeakerAngle[i] = SpeakerAngle[min];
SpeakerAngle[min] = tmpf;
tmpc = Speaker2Chan[i];
Speaker2Chan[i] = Speaker2Chan[min];
Speaker2Chan[min] = tmpc;
}
}
}
/**
* ComputeAngleGains
*
* Sets channel gains based on a given source's angle and its half-width. The
* angle and hwidth parameters are in radians.
*/
ALvoid ComputeAngleGains(const ALCdevice *device, ALfloat angle, ALfloat hwidth, ALfloat ingain, ALfloat *gains)
{
ALfloat tmpgains[MaxChannels] = { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f };
enum Channel Speaker2Chan[MaxChannels];
ALfloat SpeakerAngle[MaxChannels];
ALfloat langle, rangle;
ALfloat a;
ALuint i;
for(i = 0;i < device->NumChan;i++)
Speaker2Chan[i] = device->Speaker2Chan[i];
for(i = 0;i < device->NumChan;i++)
SpeakerAngle[i] = device->SpeakerAngle[i];
/* Some easy special-cases first... */
if(device->NumChan == 1 || hwidth >= F_PI)
{
/* Full coverage for all speakers. */
for(i = 0;i < device->NumChan;i++)
{
enum Channel chan = Speaker2Chan[i];
gains[chan] = ingain;
}
return;
}
if(hwidth <= 0.0f)
{
/* Infinitely small sound point. */
for(i = 0;i < device->NumChan-1;i++)
{
if(angle >= SpeakerAngle[i] && angle < SpeakerAngle[i+1])
{
/* Sound is between speakers i and i+1 */
a = (angle-SpeakerAngle[i]) /
(SpeakerAngle[i+1]-SpeakerAngle[i]);
gains[Speaker2Chan[i]] = sqrtf(1.0f-a) * ingain;
gains[Speaker2Chan[i+1]] = sqrtf( a) * ingain;
return;
}
}
/* Sound is between last and first speakers */
if(angle < SpeakerAngle[0])
angle += F_PI*2.0f;
a = (angle-SpeakerAngle[i]) /
(F_PI*2.0f + SpeakerAngle[0]-SpeakerAngle[i]);
gains[Speaker2Chan[i]] = sqrtf(1.0f-a) * ingain;
gains[Speaker2Chan[0]] = sqrtf( a) * ingain;
return;
}
if(fabsf(angle)+hwidth > F_PI)
{
/* The coverage area would go outside of -pi...+pi. Instead, rotate the
* speaker angles so it would be as if angle=0, and keep them wrapped
* within -pi...+pi. */
if(angle > 0.0f)
{
ALuint done = 0;
ALuint i = 0;
while(i < device->NumChan && device->SpeakerAngle[i]-angle < -F_PI)
i++;
for(done = 0;i < device->NumChan;done++)
{
SpeakerAngle[done] = device->SpeakerAngle[i]-angle;
Speaker2Chan[done] = device->Speaker2Chan[i];
i++;
}
for(i = 0;done < device->NumChan;i++)
{
SpeakerAngle[done] = device->SpeakerAngle[i]-angle + F_PI*2.0f;
Speaker2Chan[done] = device->Speaker2Chan[i];
done++;
}
}
else
{
/* NOTE: '< device->NumChan' on the iterators is correct here since
* we need to handle index 0. Because the iterators are unsigned,
* they'll underflow and wrap to become 0xFFFFFFFF, which will
* break as expected. */
ALuint done = device->NumChan-1;
ALuint i = device->NumChan-1;
while(i < device->NumChan && device->SpeakerAngle[i]-angle > F_PI)
i--;
for(done = device->NumChan-1;i < device->NumChan;done--)
{
SpeakerAngle[done] = device->SpeakerAngle[i]-angle;
Speaker2Chan[done] = device->Speaker2Chan[i];
i--;
}
for(i = device->NumChan-1;done < device->NumChan;i--)
{
SpeakerAngle[done] = device->SpeakerAngle[i]-angle - F_PI*2.0f;
Speaker2Chan[done] = device->Speaker2Chan[i];
done--;
}
}
angle = 0.0f;
}
langle = angle - hwidth;
rangle = angle + hwidth;
/* First speaker */
i = 0;
do {
ALuint last = device->NumChan-1;
enum Channel chan = Speaker2Chan[i];
if(SpeakerAngle[i] >= langle && SpeakerAngle[i] <= rangle)
{
tmpgains[chan] = 1.0f;
continue;
}
if(SpeakerAngle[i] < langle && SpeakerAngle[i+1] > langle)
{
a = (langle-SpeakerAngle[i]) /
(SpeakerAngle[i+1]-SpeakerAngle[i]);
tmpgains[chan] = lerp(tmpgains[chan], 1.0f, 1.0f-a);
}
if(SpeakerAngle[i] > rangle)
{
a = (F_PI*2.0f + rangle-SpeakerAngle[last]) /
(F_PI*2.0f + SpeakerAngle[i]-SpeakerAngle[last]);
tmpgains[chan] = lerp(tmpgains[chan], 1.0f, a);
}
else if(SpeakerAngle[last] < rangle)
{
a = (rangle-SpeakerAngle[last]) /
(F_PI*2.0f + SpeakerAngle[i]-SpeakerAngle[last]);
tmpgains[chan] = lerp(tmpgains[chan], 1.0f, a);
}
} while(0);
for(i = 1;i < device->NumChan-1;i++)
{
enum Channel chan = Speaker2Chan[i];
if(SpeakerAngle[i] >= langle && SpeakerAngle[i] <= rangle)
{
tmpgains[chan] = 1.0f;
continue;
}
if(SpeakerAngle[i] < langle && SpeakerAngle[i+1] > langle)
{
a = (langle-SpeakerAngle[i]) /
(SpeakerAngle[i+1]-SpeakerAngle[i]);
tmpgains[chan] = lerp(tmpgains[chan], 1.0f, 1.0f-a);
}
if(SpeakerAngle[i] > rangle && SpeakerAngle[i-1] < rangle)
{
a = (rangle-SpeakerAngle[i-1]) /
(SpeakerAngle[i]-SpeakerAngle[i-1]);
tmpgains[chan] = lerp(tmpgains[chan], 1.0f, a);
}
}
/* Last speaker */
i = device->NumChan-1;
do {
enum Channel chan = Speaker2Chan[i];
if(SpeakerAngle[i] >= langle && SpeakerAngle[i] <= rangle)
{
tmpgains[Speaker2Chan[i]] = 1.0f;
continue;
}
if(SpeakerAngle[i] > rangle && SpeakerAngle[i-1] < rangle)
{
a = (rangle-SpeakerAngle[i-1]) /
(SpeakerAngle[i]-SpeakerAngle[i-1]);
tmpgains[chan] = lerp(tmpgains[chan], 1.0f, a);
}
if(SpeakerAngle[i] < langle)
{
a = (langle-SpeakerAngle[i]) /
(F_PI*2.0f + SpeakerAngle[0]-SpeakerAngle[i]);
tmpgains[chan] = lerp(tmpgains[chan], 1.0f, 1.0f-a);
}
else if(SpeakerAngle[0] > langle)
{
a = (F_PI*2.0f + langle-SpeakerAngle[i]) /
(F_PI*2.0f + SpeakerAngle[0]-SpeakerAngle[i]);
tmpgains[chan] = lerp(tmpgains[chan], 1.0f, 1.0f-a);
}
} while(0);
for(i = 0;i < device->NumChan;i++)
{
enum Channel chan = device->Speaker2Chan[i];
gains[chan] = sqrtf(tmpgains[chan]) * ingain;
}
}
ALvoid aluInitPanning(ALCdevice *Device)
{
const char *layoutname = NULL;
enum Channel *Speaker2Chan;
ALfloat *SpeakerAngle;
Speaker2Chan = Device->Speaker2Chan;
SpeakerAngle = Device->SpeakerAngle;
switch(Device->FmtChans)
{
case DevFmtMono:
Device->NumChan = 1;
Speaker2Chan[0] = FrontCenter;
SpeakerAngle[0] = F_PI/180.0f * 0.0f;
layoutname = NULL;
break;
case DevFmtStereo:
Device->NumChan = 2;
Speaker2Chan[0] = FrontLeft;
Speaker2Chan[1] = FrontRight;
SpeakerAngle[0] = F_PI/180.0f * -90.0f;
SpeakerAngle[1] = F_PI/180.0f * 90.0f;
layoutname = "layout_stereo";
break;
case DevFmtQuad:
Device->NumChan = 4;
Speaker2Chan[0] = BackLeft;
Speaker2Chan[1] = FrontLeft;
Speaker2Chan[2] = FrontRight;
Speaker2Chan[3] = BackRight;
SpeakerAngle[0] = F_PI/180.0f * -135.0f;
SpeakerAngle[1] = F_PI/180.0f * -45.0f;
SpeakerAngle[2] = F_PI/180.0f * 45.0f;
SpeakerAngle[3] = F_PI/180.0f * 135.0f;
layoutname = "layout_quad";
break;
case DevFmtX51:
Device->NumChan = 5;
Speaker2Chan[0] = BackLeft;
Speaker2Chan[1] = FrontLeft;
Speaker2Chan[2] = FrontCenter;
Speaker2Chan[3] = FrontRight;
Speaker2Chan[4] = BackRight;
SpeakerAngle[0] = F_PI/180.0f * -110.0f;
SpeakerAngle[1] = F_PI/180.0f * -30.0f;
SpeakerAngle[2] = F_PI/180.0f * 0.0f;
SpeakerAngle[3] = F_PI/180.0f * 30.0f;
SpeakerAngle[4] = F_PI/180.0f * 110.0f;
layoutname = "layout_surround51";
break;
case DevFmtX51Side:
Device->NumChan = 5;
Speaker2Chan[0] = SideLeft;
Speaker2Chan[1] = FrontLeft;
Speaker2Chan[2] = FrontCenter;
Speaker2Chan[3] = FrontRight;
Speaker2Chan[4] = SideRight;
SpeakerAngle[0] = F_PI/180.0f * -90.0f;
SpeakerAngle[1] = F_PI/180.0f * -30.0f;
SpeakerAngle[2] = F_PI/180.0f * 0.0f;
SpeakerAngle[3] = F_PI/180.0f * 30.0f;
SpeakerAngle[4] = F_PI/180.0f * 90.0f;
layoutname = "layout_side51";
break;
case DevFmtX61:
Device->NumChan = 6;
Speaker2Chan[0] = SideLeft;
Speaker2Chan[1] = FrontLeft;
Speaker2Chan[2] = FrontCenter;
Speaker2Chan[3] = FrontRight;
Speaker2Chan[4] = SideRight;
Speaker2Chan[5] = BackCenter;
SpeakerAngle[0] = F_PI/180.0f * -90.0f;
SpeakerAngle[1] = F_PI/180.0f * -30.0f;
SpeakerAngle[2] = F_PI/180.0f * 0.0f;
SpeakerAngle[3] = F_PI/180.0f * 30.0f;
SpeakerAngle[4] = F_PI/180.0f * 90.0f;
SpeakerAngle[5] = F_PI/180.0f * 180.0f;
layoutname = "layout_surround61";
break;
case DevFmtX71:
Device->NumChan = 7;
Speaker2Chan[0] = BackLeft;
Speaker2Chan[1] = SideLeft;
Speaker2Chan[2] = FrontLeft;
Speaker2Chan[3] = FrontCenter;
Speaker2Chan[4] = FrontRight;
Speaker2Chan[5] = SideRight;
Speaker2Chan[6] = BackRight;
SpeakerAngle[0] = F_PI/180.0f * -150.0f;
SpeakerAngle[1] = F_PI/180.0f * -90.0f;
SpeakerAngle[2] = F_PI/180.0f * -30.0f;
SpeakerAngle[3] = F_PI/180.0f * 0.0f;
SpeakerAngle[4] = F_PI/180.0f * 30.0f;
SpeakerAngle[5] = F_PI/180.0f * 90.0f;
SpeakerAngle[6] = F_PI/180.0f * 150.0f;
layoutname = "layout_surround71";
break;
}
if(layoutname && Device->Type != Loopback)
SetSpeakerArrangement(layoutname, SpeakerAngle, Speaker2Chan, Device->NumChan);
}
-420
View File
@@ -1,420 +0,0 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2007 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include <stdlib.h>
#include <stdio.h>
#include <memory.h>
#include <windows.h>
#include <mmsystem.h>
#include "alMain.h"
#include "AL/al.h"
#include "AL/alc.h"
typedef struct {
// MMSYSTEM Capture Device
ALboolean bWaveInShutdown;
HANDLE hWaveInHdrEvent;
HANDLE hWaveInThreadEvent;
HANDLE hWaveInThread;
DWORD ulWaveInThreadID;
ALint lWaveInBuffersCommitted;
HWAVEIN hWaveInHandle;
WAVEHDR WaveInBuffer[4];
ALCchar *pCapturedSampleData;
ALuint ulCapturedDataSize;
ALuint ulReadCapturedDataPos;
ALuint ulWriteCapturedDataPos;
} WinMMData;
static ALCchar *CaptureDeviceList[16];
/*
WaveInProc
Posts a message to 'CaptureThreadProc' everytime a WaveIn Buffer is completed and
returns to the application (with more data)
*/
static void CALLBACK WaveInProc(HWAVEIN hDevice,UINT uMsg,DWORD_PTR dwInstance,DWORD_PTR dwParam1,DWORD_PTR dwParam2)
{
ALCdevice *pDevice = (ALCdevice *)dwInstance;
WinMMData *pData = pDevice->ExtraData;
(void)hDevice;
(void)dwParam2;
if ((uMsg==WIM_DATA) && (pDevice))
{
// Decrement number of buffers in use
pData->lWaveInBuffersCommitted--;
if (pData->bWaveInShutdown == AL_FALSE)
{
// Notify Wave Processor Thread that a Wave Header has returned
PostThreadMessage(pData->ulWaveInThreadID,uMsg,0,dwParam1);
}
else
{
if (pData->lWaveInBuffersCommitted == 0)
{
// Signal Wave Buffers Returned event
if (pData->hWaveInHdrEvent)
SetEvent(pData->hWaveInHdrEvent);
// Post 'Quit' Message to WaveIn Processor Thread
PostThreadMessage(pData->ulWaveInThreadID,WM_QUIT,0,0);
}
}
}
}
/*
CaptureThreadProc
Used by "MMSYSTEM" Device. Called when a WaveIn buffer had been filled with new
audio data.
*/
DWORD WINAPI CaptureThreadProc(LPVOID lpParameter)
{
ALCdevice *pDevice = (ALCdevice*)lpParameter;
WinMMData *pData = pDevice->ExtraData;
ALuint ulOffset, ulMaxSize, ulSection;
LPWAVEHDR pWaveHdr;
MSG msg;
while (GetMessage(&msg, NULL, 0, 0))
{
if ((msg.message==WIM_DATA)&&(!pData->bWaveInShutdown))
{
SuspendContext(NULL);
pWaveHdr = ((LPWAVEHDR)msg.lParam);
// Calculate offset in local buffer to write data to
ulOffset = pData->ulWriteCapturedDataPos % pData->ulCapturedDataSize;
if ((ulOffset + pWaveHdr->dwBytesRecorded) > pData->ulCapturedDataSize)
{
ulSection = pData->ulCapturedDataSize - ulOffset;
memcpy(pData->pCapturedSampleData + ulOffset, pWaveHdr->lpData, ulSection);
memcpy(pData->pCapturedSampleData, pWaveHdr->lpData + ulSection, pWaveHdr->dwBytesRecorded - ulSection);
}
else
{
memcpy(pData->pCapturedSampleData + ulOffset, pWaveHdr->lpData, pWaveHdr->dwBytesRecorded);
}
pData->ulWriteCapturedDataPos += pWaveHdr->dwBytesRecorded;
if (pData->ulWriteCapturedDataPos > (pData->ulReadCapturedDataPos + pData->ulCapturedDataSize))
{
// Application has not read enough audio data from the capture buffer so data has been
// overwritten. Reset ReadPosition.
pData->ulReadCapturedDataPos = pData->ulWriteCapturedDataPos - pData->ulCapturedDataSize;
}
// To prevent an over-flow prevent the offset values from getting too large
ulMaxSize = pData->ulCapturedDataSize << 4;
if ((pData->ulReadCapturedDataPos > ulMaxSize) && (pData->ulWriteCapturedDataPos > ulMaxSize))
{
pData->ulReadCapturedDataPos -= ulMaxSize;
pData->ulWriteCapturedDataPos -= ulMaxSize;
}
// Send buffer back to capture more data
waveInAddBuffer(pData->hWaveInHandle,pWaveHdr,sizeof(WAVEHDR));
pData->lWaveInBuffersCommitted++;
ProcessContext(NULL);
}
}
// Signal Wave Thread completed event
if (pData->hWaveInThreadEvent)
SetEvent(pData->hWaveInThreadEvent);
ExitThread(0);
return 0;
}
static ALCboolean WinMMOpenPlayback(ALCdevice *device, const ALCchar *deviceName)
{
(void)device;
(void)deviceName;
return ALC_FALSE;
}
static void WinMMClosePlayback(ALCdevice *device)
{
(void)device;
}
static ALCboolean WinMMOpenCapture(ALCdevice *pDevice, const ALCchar *deviceName, ALCuint frequency, ALCenum format, ALCsizei SampleSize)
{
WAVEFORMATEX wfexCaptureFormat;
WinMMData *pData = NULL;
ALint lDeviceID = 0;
ALint lBufferSize;
ALint i;
(void)format;
// Find the Device ID matching the deviceName if valid
if (deviceName)
{
for(i = 0;CaptureDeviceList[i];i++)
{
if (!strcmp(deviceName, CaptureDeviceList[i]))
{
lDeviceID = i;
break;
}
}
if(!CaptureDeviceList[i])
return ALC_FALSE;
}
pDevice->szDeviceName = CaptureDeviceList[lDeviceID];
pData = calloc(1, sizeof(*pData));
if(!pData)
{
SetALCError(ALC_OUT_OF_MEMORY);
return ALC_FALSE;
}
memset(&wfexCaptureFormat, 0, sizeof(WAVEFORMATEX));
wfexCaptureFormat.wFormatTag = WAVE_FORMAT_PCM;
wfexCaptureFormat.nChannels = pDevice->Channels;
wfexCaptureFormat.wBitsPerSample = pDevice->FrameSize / pDevice->Channels * 8;
wfexCaptureFormat.nBlockAlign = pDevice->FrameSize;
wfexCaptureFormat.nSamplesPerSec = frequency;
wfexCaptureFormat.nAvgBytesPerSec = wfexCaptureFormat.nSamplesPerSec *
pDevice->FrameSize;
wfexCaptureFormat.cbSize = 0;
if (waveInOpen(&pData->hWaveInHandle, lDeviceID, &wfexCaptureFormat, (DWORD_PTR)&WaveInProc, (DWORD_PTR)pDevice, CALLBACK_FUNCTION) != MMSYSERR_NOERROR)
goto failure;
pData->hWaveInHdrEvent = CreateEvent(NULL, AL_TRUE, AL_FALSE, "WaveInAllHeadersReturned");
if (pData->hWaveInHdrEvent == NULL)
goto failure;
pData->hWaveInThreadEvent = CreateEvent(NULL, AL_TRUE, AL_FALSE, "WaveInThreadDestroyed");
if (pData->hWaveInThreadEvent == NULL)
goto failure;
pData->hWaveInThread = CreateThread(NULL, 0, (LPTHREAD_START_ROUTINE)CaptureThreadProc, (LPVOID)pDevice, 0, &pData->ulWaveInThreadID);
if (pData->hWaveInThread == NULL)
goto failure;
// Allocate circular memory buffer for the captured audio
pData->ulCapturedDataSize = SampleSize * wfexCaptureFormat.nBlockAlign;
// Make sure circular buffer is at least 100ms in size (and an exact multiple of
// the block alignment
if (pData->ulCapturedDataSize < (wfexCaptureFormat.nAvgBytesPerSec / 10))
{
pData->ulCapturedDataSize = wfexCaptureFormat.nAvgBytesPerSec / 10;
pData->ulCapturedDataSize -= (pData->ulCapturedDataSize % wfexCaptureFormat.nBlockAlign);
}
pData->pCapturedSampleData = (ALCchar*)malloc(pData->ulCapturedDataSize);
pData->lWaveInBuffersCommitted=0;
// Create 4 Buffers of 50ms each
lBufferSize = wfexCaptureFormat.nAvgBytesPerSec / 20;
lBufferSize -= (lBufferSize % wfexCaptureFormat.nBlockAlign);
for (i=0;i<4;i++)
{
memset(&pData->WaveInBuffer[i], 0, sizeof(WAVEHDR));
pData->WaveInBuffer[i].dwBufferLength = lBufferSize;
pData->WaveInBuffer[i].lpData = calloc(1,pData->WaveInBuffer[i].dwBufferLength);
pData->WaveInBuffer[i].dwFlags = 0;
pData->WaveInBuffer[i].dwLoops = 0;
waveInPrepareHeader(pData->hWaveInHandle, &pData->WaveInBuffer[i], sizeof(WAVEHDR));
waveInAddBuffer(pData->hWaveInHandle, &pData->WaveInBuffer[i], sizeof(WAVEHDR));
pData->lWaveInBuffersCommitted++;
}
pData->ulReadCapturedDataPos = 0;
pData->ulWriteCapturedDataPos = 0;
pDevice->ExtraData = pData;
return ALC_TRUE;
failure:
if (pData->hWaveInThreadEvent)
CloseHandle(pData->hWaveInThreadEvent);
if (pData->hWaveInHdrEvent)
CloseHandle(pData->hWaveInHdrEvent);
if (pData->hWaveInHandle)
waveInClose(pData->hWaveInHandle);
free(pData);
return ALC_FALSE;
}
static void WinMMCloseCapture(ALCdevice *pDevice)
{
WinMMData *pData = (WinMMData*)pDevice->ExtraData;
int i;
// Call waveOutReset to shutdown wave device
pData->bWaveInShutdown = AL_TRUE;
waveInReset(pData->hWaveInHandle);
// Wait for signal that all Wave Buffers have returned
WaitForSingleObjectEx(pData->hWaveInHdrEvent, 5000, FALSE);
// Wait for signal that Wave Thread has been destroyed
WaitForSingleObjectEx(pData->hWaveInThreadEvent, 5000, FALSE);
// Release the wave buffers
for (i=0;i<4;i++)
{
waveInUnprepareHeader(pData->hWaveInHandle, &pData->WaveInBuffer[i], sizeof(WAVEHDR));
free(pData->WaveInBuffer[i].lpData);
}
// Free Audio Buffer data
free(pData->pCapturedSampleData);
pData->pCapturedSampleData = NULL;
// Close the Wave device
waveInClose(pData->hWaveInHandle);
pData->hWaveInHandle = 0;
CloseHandle(pData->hWaveInThread);
pData->hWaveInThread = 0;
if (pData->hWaveInHdrEvent)
{
CloseHandle(pData->hWaveInHdrEvent);
pData->hWaveInHdrEvent = 0;
}
if (pData->hWaveInThreadEvent)
{
CloseHandle(pData->hWaveInThreadEvent);
pData->hWaveInThreadEvent = 0;
}
free(pData);
pDevice->ExtraData = NULL;
}
static void WinMMStartCapture(ALCdevice *pDevice)
{
WinMMData *pData = (WinMMData*)pDevice->ExtraData;
waveInStart(pData->hWaveInHandle);
}
static void WinMMStopCapture(ALCdevice *pDevice)
{
WinMMData *pData = (WinMMData*)pDevice->ExtraData;
waveInStop(pData->hWaveInHandle);
}
static void WinMMCaptureSamples(ALCdevice *pDevice, ALCvoid *pBuffer, ALCuint lSamples)
{
WinMMData *pData = (WinMMData*)pDevice->ExtraData;
ALuint ulSamples = (unsigned long)lSamples;
ALuint ulBytes, ulBytesToCopy;
ALuint ulCapturedSamples;
ALuint ulReadOffset;
// Check that we have the requested numbers of Samples
ulCapturedSamples = (pData->ulWriteCapturedDataPos - pData->ulReadCapturedDataPos) / pDevice->FrameSize;
if(ulSamples > ulCapturedSamples)
{
SetALCError(ALC_INVALID_VALUE);
return;
}
ulBytes = ulSamples * pDevice->FrameSize;
// Get Read Offset
ulReadOffset = (pData->ulReadCapturedDataPos % pData->ulCapturedDataSize);
// Check for wrap-around condition
if ((ulReadOffset + ulBytes) > pData->ulCapturedDataSize)
{
// Copy data from last Read position to end of data
ulBytesToCopy = pData->ulCapturedDataSize - ulReadOffset;
memcpy(pBuffer, pData->pCapturedSampleData + ulReadOffset, ulBytesToCopy);
// Copy rest of the data from the start of the captured data
memcpy(((char *)pBuffer) + ulBytesToCopy, pData->pCapturedSampleData, ulBytes - ulBytesToCopy);
}
else
{
// Copy data from the read position in the captured data
memcpy(pBuffer, pData->pCapturedSampleData + ulReadOffset, ulBytes);
}
// Update Read Position
pData->ulReadCapturedDataPos += ulBytes;
}
static ALCuint WinMMAvailableSamples(ALCdevice *pDevice)
{
WinMMData *pData = (WinMMData*)pDevice->ExtraData;
ALCuint lCapturedBytes = (pData->ulWriteCapturedDataPos - pData->ulReadCapturedDataPos);
return lCapturedBytes / pDevice->FrameSize;
}
BackendFuncs WinMMFuncs = {
WinMMOpenPlayback,
WinMMClosePlayback,
WinMMOpenCapture,
WinMMCloseCapture,
WinMMStartCapture,
WinMMStopCapture,
WinMMCaptureSamples,
WinMMAvailableSamples
};
void alcWinMMInit(BackendFuncs *FuncList)
{
ALint lNumDevs;
ALint lLoop;
*FuncList = WinMMFuncs;
lNumDevs = waveInGetNumDevs();
for (lLoop = 0; lLoop < lNumDevs; lLoop++)
{
WAVEINCAPS WaveInCaps;
if(waveInGetDevCaps(lLoop, &WaveInCaps, sizeof(WAVEINCAPS)) == MMSYSERR_NOERROR)
{
char name[128];
snprintf(name, sizeof(name), "WaveIn on %s", WaveInCaps.szPname);
CaptureDeviceList[lLoop] = AppendCaptureDeviceList(name);
}
}
}
+687 -128
View File
@@ -1,53 +1,160 @@
# CMake build file list for OpenAL
CMAKE_MINIMUM_REQUIRED(VERSION 2.4)
IF(COMMAND CMAKE_POLICY)
CMAKE_POLICY(SET CMP0003 NEW)
ENDIF(COMMAND CMAKE_POLICY)
SET(CMAKE_MODULE_PATH "${CMAKE_SOURCE_DIR}/cmake")
INCLUDE(CheckFunctionExists)
INCLUDE(CheckLibraryExists)
INCLUDE(CheckSharedFunctionExists)
INCLUDE(CheckIncludeFile)
INCLUDE(CheckIncludeFiles)
INCLUDE(CheckSymbolExists)
INCLUDE(CheckCCompilerFlag)
INCLUDE(CheckCSourceCompiles)
INCLUDE(CheckTypeSize)
INCLUDE(FindPkgConfig)
PROJECT(OpenAL)
PROJECT(OpenAL C)
SET(CMAKE_ALLOW_LOOSE_LOOP_CONSTRUCTS TRUE)
OPTION(ALSA "ALSA backend" ON)
OPTION(OSS "OSS backend" ON)
OPTION(DSOUND "DirectSound backend" ON)
OPTION(WINMM "Windows Multimedia backend" ON)
OPTION(SSE "Check for SSE CPU extensions" ON)
OPTION(NEON "Check for ARM Neon CPU extensions" ON)
OPTION(DLOPEN "Use the dlopen API for loading optional libs" ON)
OPTION(REQUIRE_SSE "Require SSE CPU extensions" OFF)
OPTION(REQUIRE_NEON "Require ARM Neon CPU extensions" OFF)
OPTION(DEBUG "Build lib in debug mode" OFF)
OPTION(NODEBUG "Disable all debug info for optimizations" OFF)
OPTION(WERROR "Treat compile warnings as errors" OFF)
OPTION(ALSA "Check for ALSA backend" ON)
OPTION(OSS "Check for OSS backend" ON)
OPTION(SOLARIS "Check for Solaris backend" ON)
OPTION(SNDIO "Check for SndIO backend" ON)
OPTION(MMDEVAPI "Check for MMDevApi" ON)
OPTION(DSOUND "Check for DirectSound backend" ON)
OPTION(WINMM "Check for Windows Multimedia backend" ON)
OPTION(PORTAUDIO "Check for PortAudio backend" ON)
OPTION(PULSEAUDIO "Check for PulseAudio backend" ON)
OPTION(COREAUDIO "Check for CoreAudio backend" ON)
OPTION(OPENSL "Check for OpenSL backend" ON)
OPTION(WAVE "Enable Wave Writer backend" ON)
OPTION(REQUIRE_ALSA "Require ALSA backend" OFF)
OPTION(REQUIRE_OSS "Require OSS backend" OFF)
OPTION(REQUIRE_SOLARIS "Require Solaris backend" OFF)
OPTION(REQUIRE_SNDIO "Require SndIO backend" OFF)
OPTION(REQUIRE_MMDEVAPI "Require MMDevApi" OFF)
OPTION(REQUIRE_DSOUND "Require DirectSound backend" OFF)
OPTION(REQUIRE_WINMM "Require Windows Multimedia backend" OFF)
OPTION(REQUIRE_PORTAUDIO "Require PortAudio backend" OFF)
OPTION(REQUIRE_PULSEAUDIO "Require PulseAudio backend" OFF)
OPTION(REQUIRE_COREAUDIO "Require CoreAudio backend" OFF)
OPTION(REQUIRE_OPENSL "Require OpenSL backend" OFF)
OPTION(DLOPEN "Check for the dlopen API for loading optional libs" ON)
OPTION(WERROR "Treat compile warnings as errors" OFF)
OPTION(UTILS "Build and install utility programs" ON)
OPTION(EXAMPLES "Build and install example programs" ON)
OPTION(ALSOFT_CONFIG "Install alsoft.conf sample configuration file" ON)
IF(WIN32)
SET(LIBNAME OpenAL32)
ADD_DEFINITIONS("-D_WIN32 -D_WIN32_WINNT=0x0501")
ELSE()
SET(LIBNAME openal)
ENDIF()
IF(NOT LIBTYPE)
SET(LIBTYPE SHARED)
ENDIF()
SET(LIB_MAJOR_VERSION "1")
SET(LIB_MINOR_VERSION "1")
SET(LIB_BUILD_VERSION "93")
SET(LIB_VERSION "${LIB_MAJOR_VERSION}.${LIB_MINOR_VERSION}.${LIB_BUILD_VERSION}")
SET(LIB_MINOR_VERSION "15")
SET(LIB_VERSION "${LIB_MAJOR_VERSION}.${LIB_MINOR_VERSION}")
SET(EXPORT_DECL "")
SET(ALIGN_DECL "")
SET(RESTRICT_DECL "")
CHECK_TYPE_SIZE("long" SIZEOF_LONG)
CHECK_TYPE_SIZE("long long" SIZEOF_LONG_LONG)
CHECK_TYPE_SIZE("unsigned int" SIZEOF_UINT)
CHECK_TYPE_SIZE("void*" SIZEOF_VOIDP)
CHECK_C_SOURCE_COMPILES("int *restrict foo;
int main() {return 0;}" HAVE_RESTRICT)
IF(HAVE_RESTRICT)
SET(RESTRICT_DECL "restrict")
ELSE()
CHECK_C_SOURCE_COMPILES("int *__restrict foo;
int main() {return 0;}" HAVE___RESTRICT)
IF(HAVE___RESTRICT)
SET(RESTRICT_DECL "__restrict")
ENDIF()
ENDIF()
# Add definitions, compiler switches, etc.
INCLUDE_DIRECTORIES(OpenAL32/Include include "${OpenAL_BINARY_DIR}")
IF("${MSVC}")
IF(NOT CMAKE_BUILD_TYPE)
SET(CMAKE_BUILD_TYPE RelWithDebInfo CACHE STRING
"Choose the type of build, options are: Debug Release RelWithDebInfo MinSizeRel."
FORCE)
ENDIF()
IF(NOT CMAKE_DEBUG_POSTFIX)
SET(CMAKE_DEBUG_POSTFIX "" CACHE STRING
"Library postfix for debug builds. Normally left blank."
FORCE)
ENDIF()
IF(MSVC)
# ???
SET(CMAKE_C_FLAGS_RELWITHDEBINFO "${CMAKE_C_FLAGS_RELWITHDEBINFO} -D_DEBUG")
SET(CMAKE_C_FLAGS_MINSIZEREL "${CMAKE_C_FLAGS_MINSIZEREL} -DNDEBUG")
SET(CMAKE_C_FLAGS_RELEASE "${CMAKE_C_FLAGS_RELEASE} -DNDEBUG")
SET(CMAKE_C_FLAGS_DEBUG "${CMAKE_C_FLAGS_DEBUG} -D_DEBUG")
ADD_DEFINITIONS(-D_CRT_SECURE_NO_WARNINGS)
ADD_DEFINITIONS(-D_CRT_NONSTDC_NO_DEPRECATE)
IF(NOT DXSDK_DIR)
STRING(REGEX REPLACE "\\\\" "/" DXSDK_DIR "$ENV{DXSDK_DIR}")
ELSE()
STRING(REGEX REPLACE "\\\\" "/" DXSDK_DIR "${DXSDK_DIR}")
ENDIF()
IF(DXSDK_DIR)
MESSAGE(STATUS "Using DirectX SDK directory: ${DXSDK_DIR}")
SET(CMAKE_REQUIRED_INCLUDES ${CMAKE_REQUIRED_INCLUDES} "${DXSDK_DIR}/Include")
INCLUDE_DIRECTORIES("${DXSDK_DIR}/Include")
LINK_DIRECTORIES("${DXSDK_DIR}/Lib")
ENDIF()
OPTION(FORCE_STATIC_VCRT "Force /MT for static VC runtimes" OFF)
IF(FORCE_STATIC_VCRT)
FOREACH(flag_var
CMAKE_C_FLAGS CMAKE_C_FLAGS_DEBUG CMAKE_C_FLAGS_RELEASE
CMAKE_C_FLAGS_MINSIZEREL CMAKE_C_FLAGS_RELWITHDEBINFO)
IF(${flag_var} MATCHES "/MD")
STRING(REGEX REPLACE "/MD" "/MT" ${flag_var} "${${flag_var}}")
ENDIF()
ENDFOREACH(flag_var)
ENDIF()
ELSE()
ADD_DEFINITIONS(-Wall)
ADD_DEFINITIONS(-Winline -Wall)
CHECK_C_COMPILER_FLAG(-Wextra HAVE_W_EXTRA)
IF("${HAVE_W_EXTRA}")
IF(HAVE_W_EXTRA)
ADD_DEFINITIONS(-Wextra)
ENDIF()
@@ -55,222 +162,598 @@ ELSE()
ADD_DEFINITIONS(-Werror)
ENDIF()
IF(DEBUG)
ADD_DEFINITIONS(-g3)
ELSEIF(NODEBUG)
ADD_DEFINITIONS(-O2 -funroll-loops -fomit-frame-pointer)
SET(CMAKE_C_FLAGS_RELWITHDEBINFO "-g -O2 -D_DEBUG" CACHE STRING
"Flags used by the compiler during Release with Debug Info builds."
FORCE)
SET(CMAKE_C_FLAGS_MINSIZEREL "-Os -DNDEBUG" CACHE STRING
"Flags used by the compiler during release minsize builds."
FORCE)
SET(CMAKE_C_FLAGS_RELEASE "-O2 -fomit-frame-pointer -DNDEBUG" CACHE STRING
"Flags used by the compiler during release builds"
FORCE)
SET(CMAKE_C_FLAGS_DEBUG "-g3 -D_DEBUG" CACHE STRING
"Flags used by the compiler during debug builds."
FORCE)
CHECK_C_SOURCE_COMPILES("int foo() __attribute__((destructor));
int main() {return 0;}" HAVE_GCC_DESTRUCTOR)
ENDIF()
# Set visibility/export options if available
IF(WIN32)
SET(EXPORT_DECL "__declspec(dllexport)")
SET(ALIGN_DECL "__declspec(align(x))")
OPTION(WINE "Enable use of Wine headers when compiling" OFF)
IF(WINE)
FIND_PATH(WINE_INCLUDE_DIR library.h
PATHS
/usr/include/wine
/usr/local/include/wine
CMAKE_FIND_ROOT_PATH_BOTH)
IF(WINE_INCLUDE_DIR)
MESSAGE(STATUS "Found Wine header files - ${WINE_INCLUDE_DIR}" )
INCLUDE_DIRECTORIES("${WINE_INCLUDE_DIR}/windows")
SET(CMAKE_REQUIRED_INCLUDES ${CMAKE_REQUIRED_INCLUDES} "${WINE_INCLUDE_DIR}/windows")
ELSE()
MESSAGE(STATUS "Could not find Wine header files" )
ENDIF()
ENDIF()
ELSE()
SET(OLD_REQUIRED_FLAGS "${CMAKE_REQUIRED_FLAGS}")
# Yes GCC, really don't accept visibility modes you don't support
SET(CMAKE_REQUIRED_FLAGS "${OLD_REQUIRED_FLAGS} -Wattributes -Werror")
CHECK_C_SOURCE_COMPILES("int foo() __attribute__((visibility(\"protected\")));
int main() {return 0;}" HAVE_GCC_PROTECTED_VISIBILITY)
IF(HAVE_GCC_PROTECTED_VISIBILITY)
SET(EXPORT_DECL "__attribute__((visibility(\"protected\")))")
ELSE()
ADD_DEFINITIONS(-g -O2 -funroll-loops)
CHECK_C_SOURCE_COMPILES("int foo() __attribute__((visibility(\"default\")));
int main() {return 0;}" HAVE_GCC_DEFAULT_VISIBILITY)
IF(HAVE_GCC_DEFAULT_VISIBILITY)
SET(EXPORT_DECL "__attribute__((visibility(\"default\")))")
ENDIF()
ENDIF()
# The mixer doesn't like GCC's strict aliasing optimizations. Make sure
# it's turned off
CHECK_C_COMPILER_FLAG(-fstrict-aliasing HAVE_STRICT_ALIASING)
IF("${HAVE_STRICT_ALIASING}")
ADD_DEFINITIONS(-fno-strict-aliasing)
ENDIF()
# Set visibility options if available
IF(NOT "${WIN32}")
CHECK_C_SOURCE_COMPILES("int foo() __attribute__((destructor));
int main() {return 0;}" HAVE_GCC_DESTRUCTOR)
CHECK_C_COMPILER_FLAG(-fvisibility=hidden HAVE_VISIBILITY_SWITCH)
IF("${HAVE_VISIBILITY_SWITCH}")
CHECK_C_SOURCE_COMPILES("int foo() __attribute__((visibility(\"default\")));
int main() {return 0;}" HAVE_GCC_VISIBILITY)
IF("${HAVE_GCC_VISIBILITY}")
ADD_DEFINITIONS(-fvisibility=hidden -DHAVE_GCC_VISIBILITY)
IF(HAVE_GCC_PROTECTED_VISIBILITY OR HAVE_GCC_DEFAULT_VISIBILITY)
CHECK_C_COMPILER_FLAG(-fvisibility=internal HAVE_VISIBILITY_INTERNAL_SWITCH)
IF(HAVE_VISIBILITY_INTERNAL_SWITCH)
ADD_DEFINITIONS(-fvisibility=internal)
ELSE()
CHECK_C_COMPILER_FLAG(-fvisibility=hidden HAVE_VISIBILITY_HIDDEN_SWITCH)
IF(HAVE_VISIBILITY_HIDDEN_SWITCH)
ADD_DEFINITIONS(-fvisibility=hidden)
ENDIF()
ENDIF()
ENDIF()
CHECK_C_SOURCE_COMPILES("int foo __attribute__((aligned(16)));
int main() {return 0;}" HAVE_ATTRIBUTE_ALIGNED)
IF(HAVE_ATTRIBUTE_ALIGNED)
SET(ALIGN_DECL "__attribute__((aligned(x)))")
ENDIF()
SET(CMAKE_REQUIRED_FLAGS "${OLD_REQUIRED_FLAGS}")
ENDIF()
IF(DEBUG)
ADD_DEFINITIONS(-D_DEBUG)
ELSEIF(NODEBUG)
ADD_DEFINITIONS(-DNDEBUG)
SET(SSE_SWITCH "")
CHECK_C_COMPILER_FLAG(-msse HAVE_MSSE_SWITCH)
IF(HAVE_MSSE_SWITCH)
SET(SSE_SWITCH "-msse")
ENDIF()
CHECK_LIBRARY_EXISTS(m sqrtf "" HAVE_SQRTF)
IF("${HAVE_SQRTF}")
CHECK_C_SOURCE_COMPILES("int foo(const char *str, ...) __attribute__((format(printf, 1, 2)));
int main() {return 0;}" HAVE_GCC_FORMAT)
CHECK_INCLUDE_FILE(malloc.h HAVE_MALLOC_H)
CHECK_INCLUDE_FILE(cpuid.h HAVE_CPUID_H)
CHECK_INCLUDE_FILE(fenv.h HAVE_FENV_H)
CHECK_INCLUDE_FILE(float.h HAVE_FLOAT_H)
CHECK_INCLUDE_FILE(ieeefp.h HAVE_IEEEFP_H)
CHECK_INCLUDE_FILE(guiddef.h HAVE_GUIDDEF_H)
IF(NOT HAVE_GUIDDEF_H)
CHECK_INCLUDE_FILE(initguid.h HAVE_INITGUID_H)
ENDIF()
# Some systems need libm for some of the following math functions to work
CHECK_LIBRARY_EXISTS(m pow "" HAVE_LIBM)
IF(HAVE_LIBM)
SET(EXTRA_LIBS m ${EXTRA_LIBS})
SET(CMAKE_REQUIRED_LIBRARIES ${CMAKE_REQUIRED_LIBRARIES} m)
ENDIF()
CHECK_FUNCTION_EXISTS(strtof HAVE_STRTOF)
CHECK_SYMBOL_EXISTS(aligned_alloc stdlib.h HAVE_ALIGNED_ALLOC)
CHECK_SYMBOL_EXISTS(posix_memalign stdlib.h HAVE_POSIX_MEMALIGN)
CHECK_SYMBOL_EXISTS(_aligned_malloc malloc.h HAVE__ALIGNED_MALLOC)
CHECK_SYMBOL_EXISTS(lrintf math.h HAVE_LRINTF)
IF(HAVE_FENV_H)
CHECK_SYMBOL_EXISTS(fesetround fenv.h HAVE_FESETROUND)
ENDIF()
IF(HAVE_FLOAT_H)
CHECK_SYMBOL_EXISTS(_controlfp float.h HAVE__CONTROLFP)
CHECK_SYMBOL_EXISTS(__control87_2 float.h HAVE___CONTROL87_2)
ENDIF()
CHECK_FUNCTION_EXISTS(strtof HAVE_STRTOF)
CHECK_FUNCTION_EXISTS(stat HAVE_STAT)
CHECK_FUNCTION_EXISTS(strcasecmp HAVE_STRCASECMP)
IF(NOT "${HAVE_STRCASECMP}")
IF(NOT HAVE_STRCASECMP)
CHECK_FUNCTION_EXISTS(_stricmp HAVE__STRICMP)
IF(NOT "${HAVE__STRICMP}")
IF(NOT HAVE__STRICMP)
MESSAGE(FATAL_ERROR "No case-insensitive compare function found, please report!")
ENDIF()
ADD_DEFINITIONS(-Dstrcasecmp=_stricmp)
ENDIF()
CHECK_FUNCTION_EXISTS(strncasecmp HAVE_STRNCASECMP)
IF(NOT HAVE_STRNCASECMP)
CHECK_FUNCTION_EXISTS(_strnicmp HAVE__STRNICMP)
IF(NOT HAVE__STRNICMP)
MESSAGE(FATAL_ERROR "No case-insensitive size-limitted compare function found, please report!")
ENDIF()
ADD_DEFINITIONS(-Dstrncasecmp=_strnicmp)
ENDIF()
CHECK_FUNCTION_EXISTS(snprintf HAVE_SNPRINTF)
IF(NOT "${HAVE_SNPRINTF}")
IF(NOT HAVE_SNPRINTF)
CHECK_FUNCTION_EXISTS(_snprintf HAVE__SNPRINTF)
IF(NOT "${HAVE__SNPRINTF}")
IF(NOT HAVE__SNPRINTF)
MESSAGE(FATAL_ERROR "No snprintf function found, please report!")
ENDIF()
ADD_DEFINITIONS(-Dsnprintf=_snprintf)
ENDIF()
CHECK_FUNCTION_EXISTS(vsnprintf HAVE_VSNPRINTF)
IF(NOT HAVE_VSNPRINTF)
CHECK_FUNCTION_EXISTS(_vsnprintf HAVE__VSNPRINTF)
IF(NOT HAVE__VSNPRINTF)
MESSAGE(FATAL_ERROR "No vsnprintf function found, please report!")
ENDIF()
ADD_DEFINITIONS(-Dvsnprintf=_vsnprintf)
ENDIF()
CHECK_SYMBOL_EXISTS(isfinite math.h HAVE_ISFINITE)
IF(NOT HAVE_ISFINITE)
CHECK_FUNCTION_EXISTS(finite HAVE_FINITE)
IF(NOT HAVE_FINITE)
CHECK_FUNCTION_EXISTS(_finite HAVE__FINITE)
IF(NOT HAVE__FINITE)
MESSAGE(FATAL_ERROR "No isfinite function found, please report!")
ENDIF()
ADD_DEFINITIONS(-Disfinite=_finite)
ELSE()
ADD_DEFINITIONS(-Disfinite=finite)
ENDIF()
ENDIF()
CHECK_SYMBOL_EXISTS(isnan math.h HAVE_ISNAN)
IF(NOT HAVE_ISNAN)
CHECK_FUNCTION_EXISTS(_isnan HAVE__ISNAN)
IF(NOT HAVE__ISNAN)
MESSAGE(FATAL_ERROR "No isnan function found, please report!")
ENDIF()
ADD_DEFINITIONS(-Disnan=_isnan)
ENDIF()
# Check for the dlopen API (for dynamicly loading backend libs)
IF(DLOPEN)
CHECK_INCLUDE_FILE(dlfcn.h HAVE_DLFCN_H)
IF("${HAVE_DLFCN_H}")
IF(HAVE_DLFCN_H)
CHECK_LIBRARY_EXISTS(dl dlopen "" HAVE_LIBDL)
IF("${HAVE_LIBDL}")
IF(HAVE_LIBDL)
SET(EXTRA_LIBS dl ${EXTRA_LIBS})
ENDIF()
ENDIF()
ENDIF()
# Check if we have Windows headers
CHECK_INCLUDE_FILE(windows.h HAVE_WINDOWS_H)
IF(NOT "${HAVE_WINDOWS_H}")
# We need pthreads outside of Windows
CHECK_INCLUDE_FILE(pthread.h HAVE_PTHREAD_H)
IF(NOT "${HAVE_PTHREAD_H}")
MESSAGE(FATAL_ERROR "PThreads is required for non-Windows builds!")
CHECK_INCLUDE_FILE(windows.h HAVE_WINDOWS_H -D_WIN32_WINNT=0x0501)
IF(NOT HAVE_WINDOWS_H)
CHECK_FUNCTION_EXISTS(gettimeofday HAVE_GETTIMEOFDAY)
IF(NOT HAVE_GETTIMEOFDAY)
MESSAGE(FATAL_ERROR "No timing function found!")
ENDIF()
# _GNU_SOURCE is needed on some systems for extra attributes, and
# _REENTRANT is needed for libc thread-safety
ADD_DEFINITIONS(-D_GNU_SOURCE=1 -D_REENTRANT)
CHECK_FUNCTION_EXISTS(nanosleep HAVE_NANOSLEEP)
IF(NOT HAVE_NANOSLEEP)
MESSAGE(FATAL_ERROR "No sleep function found!")
ENDIF()
CHECK_C_COMPILER_FLAG(-pthread HAVE_PTHREAD)
IF(HAVE_PTHREAD)
ADD_DEFINITIONS(-pthread)
SET(EXTRA_LIBS ${EXTRA_LIBS} -pthread)
ENDIF()
# We need pthreads outside of Windows
CHECK_INCLUDE_FILE(pthread.h HAVE_PTHREAD_H)
IF(NOT HAVE_PTHREAD_H)
MESSAGE(FATAL_ERROR "PThreads is required for non-Windows builds!")
ENDIF()
# Some systems need pthread_np.h to get recursive mutexes
CHECK_INCLUDE_FILES("pthread.h;pthread_np.h" HAVE_PTHREAD_NP_H)
# _GNU_SOURCE is needed on some systems for extra attributes
ADD_DEFINITIONS(-D_GNU_SOURCE=1)
CHECK_LIBRARY_EXISTS(pthread pthread_create "" HAVE_LIBPTHREAD)
IF("${HAVE_LIBPTHREAD}")
IF(HAVE_LIBPTHREAD)
SET(EXTRA_LIBS pthread ${EXTRA_LIBS})
ENDIF()
CHECK_LIBRARY_EXISTS(pthread pthread_setschedparam "" HAVE_PTHREAD_SETSCHEDPARAM)
CHECK_LIBRARY_EXISTS(rt clock_gettime "" HAVE_LIBRT)
IF(HAVE_LIBRT)
SET(EXTRA_LIBS rt ${EXTRA_LIBS})
ENDIF()
ENDIF()
# Check for a 64-bit type
CHECK_INCLUDE_FILE(stdint.h HAVE_STDINT_H)
IF(NOT "${HAVE_STDINT_H}")
IF("${HAVE_WINDOWS_H}")
CHECK_C_SOURCE_COMPILES("\#include <windows.h>
IF(NOT HAVE_STDINT_H)
IF(HAVE_WINDOWS_H)
CHECK_C_SOURCE_COMPILES("\#define _WIN32_WINNT 0x0501
\#include <windows.h>
__int64 foo;
int main() {return 0;}" HAVE___INT64)
ENDIF()
IF(NOT "${HAVE___INT64}")
IF(NOT "${SIZEOF_LONG}" MATCHES "8")
IF(NOT "${SIZEOF_LONG_LONG}" MATCHES "8")
IF(NOT HAVE___INT64)
IF(NOT SIZEOF_LONG MATCHES "8")
IF(NOT SIZEOF_LONG_LONG MATCHES "8")
MESSAGE(FATAL_ERROR "No 64-bit types found, please report!")
ENDIF()
ENDIF()
ENDIF()
ENDIF()
# Windows needs winmm for timeGetTime, even if the backend is disabled
CHECK_SHARED_FUNCTION_EXISTS(timeGetTime "windows.h;mmsystem.h" winmm "" HAVE_LIBWINMM)
IF(HAVE_LIBWINMM)
SET(EXTRA_LIBS winmm ${EXTRA_LIBS})
SET(PKG_CONFIG_LIBS ${PKG_CONFIG_LIBS} -lwinmm)
ENDIF()
SET(OPENAL_OBJS OpenAL32/alBuffer.c
SET(OPENAL_OBJS OpenAL32/alAuxEffectSlot.c
OpenAL32/alBuffer.c
OpenAL32/alEffect.c
OpenAL32/alError.c
OpenAL32/alExtension.c
OpenAL32/alFilter.c
OpenAL32/alListener.c
OpenAL32/alSource.c
OpenAL32/alState.c
OpenAL32/alThunk.c
OpenAL32/OpenAL32.c
)
SET(ALC_OBJS Alc/ALc.c
Alc/ALu.c
Alc/alcConfig.c
Alc/alcDedicated.c
Alc/alcEcho.c
Alc/alcModulator.c
Alc/alcReverb.c
Alc/alcRing.c
Alc/alcThread.c
Alc/bs2b.c
Alc/helpers.c
Alc/panning.c
Alc/hrtf.c
Alc/mixer.c
Alc/mixer_c.c
)
SET(CPU_EXTS "Default")
SET(HAVE_SSE 0)
SET(HAVE_NEON 0)
# Check for SSE support
IF(SSE AND ALIGN_DECL)
CHECK_INCLUDE_FILE(xmmintrin.h HAVE_XMMINTRIN_H "${SSE_SWITCH}")
IF(HAVE_XMMINTRIN_H)
SET(HAVE_SSE 1)
SET(ALC_OBJS ${ALC_OBJS} Alc/mixer_sse.c)
IF(SSE_SWITCH)
SET_SOURCE_FILES_PROPERTIES(Alc/mixer_sse.c PROPERTIES
COMPILE_FLAGS "${SSE_SWITCH}")
ENDIF()
SET(CPU_EXTS "${CPU_EXTS}, SSE")
ENDIF()
ENDIF()
IF(REQUIRE_SSE AND NOT HAVE_SSE)
MESSAGE(FATAL_ERROR "Failed to enabled required SSE CPU extensions")
ENDIF()
# Check for ARM Neon support
IF(NEON)
CHECK_INCLUDE_FILE(arm_neon.h HAVE_ARM_NEON_H)
IF(HAVE_ARM_NEON_H)
SET(HAVE_NEON 1)
SET(ALC_OBJS ${ALC_OBJS} Alc/mixer_neon.c)
SET(CPU_EXTS "${CPU_EXTS}, Neon")
ENDIF()
ENDIF()
IF(REQUIRE_NEON AND NOT HAVE_NEON)
MESSAGE(FATAL_ERROR "Failed to enabled required ARM Neon CPU extensions")
ENDIF()
SET(ALC_OBJS ${ALC_OBJS}
# Default backends, always available
Alc/backends/loopback.c
Alc/backends/null.c
)
SET(BACKENDS "")
SET(HAVE_ALSA 0)
SET(HAVE_OSS 0)
SET(HAVE_SOLARIS 0)
SET(HAVE_SNDIO 0)
SET(HAVE_DSOUND 0)
SET(HAVE_MMDEVAPI 0)
SET(HAVE_WINMM 0)
SET(HAVE_PORTAUDIO 0)
SET(HAVE_PULSEAUDIO 0)
SET(HAVE_COREAUDIO 0)
SET(HAVE_OPENSL 0)
SET(HAVE_WAVE 0)
# Check ALSA backend
IF(ALSA)
CHECK_INCLUDE_FILE(alsa/asoundlib.h HAVE_ALSA_ASOUNDLIB_H)
IF("${HAVE_ALSA_ASOUNDLIB_H}")
CHECK_LIBRARY_EXISTS(asound snd_pcm_open "" HAVE_LIBASOUND)
IF("${HAVE_DLFCN_H}" OR "${HAVE_LIBASOUND}")
IF(HAVE_ALSA_ASOUNDLIB_H)
CHECK_SHARED_FUNCTION_EXISTS(snd_pcm_open "alsa/asoundlib.h" asound "" HAVE_LIBASOUND)
IF(HAVE_LIBASOUND OR HAVE_DLFCN_H OR WIN32)
SET(HAVE_ALSA 1)
SET(ALC_OBJS ${ALC_OBJS} Alc/alsa.c)
IF("${HAVE_DLFCN_H}")
SET(BACKENDS ${BACKENDS} ALSA)
SET(ALC_OBJS ${ALC_OBJS} Alc/backends/alsa.c)
IF(HAVE_DLFCN_H OR WIN32)
SET(BACKENDS "${BACKENDS} ALSA,")
ELSE()
SET(BACKENDS ${BACKENDS} ALSA\(linked\))
SET(BACKENDS "${BACKENDS} ALSA \(linked\),")
SET(EXTRA_LIBS asound ${EXTRA_LIBS})
ENDIF()
ENDIF()
ENDIF()
ENDIF()
IF(REQUIRE_ALSA AND NOT HAVE_ALSA)
MESSAGE(FATAL_ERROR "Failed to enabled required ALSA backend")
ENDIF()
# Check OSS backend
IF(OSS)
CHECK_INCLUDE_FILE(sys/soundcard.h HAVE_SYS_SOUNDCARD_H)
IF("${HAVE_SYS_SOUNDCARD_H}")
IF(HAVE_SYS_SOUNDCARD_H)
SET(HAVE_OSS 1)
SET(ALC_OBJS ${ALC_OBJS} Alc/oss.c)
SET(BACKENDS ${BACKENDS} OSS)
SET(ALC_OBJS ${ALC_OBJS} Alc/backends/oss.c)
SET(BACKENDS "${BACKENDS} OSS,")
ENDIF()
ENDIF()
IF(REQUIRE_OSS AND NOT HAVE_OSS)
MESSAGE(FATAL_ERROR "Failed to enabled required OSS backend")
ENDIF()
# Check Solaris backend
IF(SOLARIS)
CHECK_INCLUDE_FILE(sys/audioio.h HAVE_SYS_AUDIOIO_H)
IF(HAVE_SYS_AUDIOIO_H)
SET(HAVE_SOLARIS 1)
SET(ALC_OBJS ${ALC_OBJS} Alc/backends/solaris.c)
SET(BACKENDS "${BACKENDS} Solaris,")
ENDIF()
ENDIF()
IF(REQUIRE_SOLARIS AND NOT HAVE_SOLARIS)
MESSAGE(FATAL_ERROR "Failed to enabled required Solaris backend")
ENDIF()
# Check SndIO backend
IF(SNDIO)
CHECK_INCLUDE_FILE(sndio.h HAVE_SNDIO_H)
IF(HAVE_SNDIO_H)
CHECK_SHARED_FUNCTION_EXISTS(sio_open "sndio.h" sndio "" HAVE_LIBSNDIO)
IF(HAVE_LIBSNDIO)
SET(HAVE_SNDIO 1)
SET(ALC_OBJS ${ALC_OBJS} Alc/backends/sndio.c)
SET(BACKENDS "${BACKENDS} SndIO \(linked\),")
SET(EXTRA_LIBS sndio ${EXTRA_LIBS})
ENDIF()
ENDIF()
ENDIF()
IF(REQUIRE_SNDIO AND NOT HAVE_SNDIO)
MESSAGE(FATAL_ERROR "Failed to enabled required SndIO backend")
ENDIF()
# Check for MMDevApi backend
IF(HAVE_WINDOWS_H)
IF(MMDEVAPI)
CHECK_INCLUDE_FILE(mmdeviceapi.h HAVE_MMDEVICEAPI_H)
IF(HAVE_MMDEVICEAPI_H)
SET(HAVE_MMDEVAPI 1)
SET(ALC_OBJS ${ALC_OBJS} Alc/backends/mmdevapi.c)
SET(BACKENDS "${BACKENDS} MMDevApi,")
ENDIF()
ENDIF()
ENDIF()
IF(REQUIRE_MMDEVAPI AND NOT HAVE_MMDEVAPI)
MESSAGE(FATAL_ERROR "Failed to enabled required MMDevApi backend")
ENDIF()
# Check DSound/MMSystem backend
IF("${HAVE_WINDOWS_H}")
SET(CMAKE_REQUIRED_LIBRARIES ole32)
CHECK_C_SOURCE_COMPILES("int main() {return 0;}" HAVE_LIBOLE32)
SET(CMAKE_REQUIRED_LIBRARIES "")
# CHECK_LIBRARY_EXISTS(ole32 CoInitialize "" HAVE_LIBOLE32)
IF("${HAVE_LIBOLE32}")
SET(EXTRA_LIBS ole32 ${EXTRA_LIBS})
ENDIF()
IF(DSOUND)
CHECK_INCLUDE_FILE(dsound.h HAVE_DSOUND_H)
IF("${HAVE_DSOUND_H}")
IF(DSOUND)
CHECK_INCLUDE_FILE(dsound.h HAVE_DSOUND_H)
IF(HAVE_DSOUND_H)
CHECK_SHARED_FUNCTION_EXISTS(DirectSoundCreate "dsound.h" dsound "" HAVE_LIBDSOUND)
IF(HAVE_LIBDSOUND OR HAVE_DLFCN_H OR WIN32)
SET(HAVE_DSOUND 1)
SET(ALC_OBJS ${ALC_OBJS} Alc/dsound.c)
SET(BACKENDS ${BACKENDS} DirectSound)
SET(ALC_OBJS ${ALC_OBJS} Alc/backends/dsound.c)
CHECK_LIBRARY_EXISTS(dxguid IID_IDirectSound "" HAVE_LIBDXGUID)
IF("${HAVE_LIBDXGUID}")
SET(EXTRA_LIBS dxguid ${EXTRA_LIBS})
ENDIF()
ENDIF()
ENDIF()
IF(WINMM)
CHECK_INCLUDE_FILES("windows.h;mmsystem.h" HAVE_MMSYSTEM_H)
IF("${HAVE_MMSYSTEM_H}")
SET(HAVE_WINMM 1)
SET(ALC_OBJS ${ALC_OBJS} Alc/winmm.c)
SET(BACKENDS ${BACKENDS} WinMM)
SET(CMAKE_REQUIRED_LIBRARIES winmm)
CHECK_C_SOURCE_COMPILES("int main() {return 0;}" HAVE_LIBWINMM)
SET(CMAKE_REQUIRED_LIBRARIES "")
# CHECK_LIBRARY_EXISTS(winmm waveInOpen "" HAVE_LIBWINMM)
IF("${HAVE_LIBWINMM}")
SET(EXTRA_LIBS winmm ${EXTRA_LIBS})
IF(HAVE_DLFCN_H OR WIN32)
SET(BACKENDS "${BACKENDS} DirectSound,")
ELSE()
SET(BACKENDS "${BACKENDS} DirectSound \(linked\),")
SET(EXTRA_LIBS dsound ${EXTRA_LIBS})
ENDIF()
ENDIF()
ENDIF()
ENDIF()
IF(REQUIRE_DSOUND AND NOT HAVE_DSOUND)
MESSAGE(FATAL_ERROR "Failed to enabled required DSound backend")
ENDIF()
IF(HAVE_WINDOWS_H)
IF(WINMM)
CHECK_INCLUDE_FILES("windows.h;mmsystem.h" HAVE_MMSYSTEM_H -D_WIN32_WINNT=0x0501)
IF(HAVE_MMSYSTEM_H AND HAVE_LIBWINMM)
SET(HAVE_WINMM 1)
SET(ALC_OBJS ${ALC_OBJS} Alc/backends/winmm.c)
SET(BACKENDS "${BACKENDS} WinMM,")
ENDIF()
ENDIF()
ENDIF()
IF(REQUIRE_WINMM AND NOT HAVE_WINMM)
MESSAGE(FATAL_ERROR "Failed to enabled required WinMM backend")
ENDIF()
# Check PortAudio backend
IF(PORTAUDIO)
CHECK_INCLUDE_FILE(portaudio.h HAVE_PORTAUDIO_H)
IF(HAVE_PORTAUDIO_H)
CHECK_SHARED_FUNCTION_EXISTS(Pa_Initialize "portaudio.h" portaudio "" HAVE_LIBPORTAUDIO)
IF(HAVE_LIBPORTAUDIO OR HAVE_DLFCN_H OR WIN32)
SET(HAVE_PORTAUDIO 1)
SET(ALC_OBJS ${ALC_OBJS} Alc/backends/portaudio.c)
IF(HAVE_DLFCN_H OR WIN32)
SET(BACKENDS "${BACKENDS} PortAudio,")
ELSE()
SET(BACKENDS "${BACKENDS} PortAudio \(linked\),")
SET(EXTRA_LIBS portaudio ${EXTRA_LIBS})
ENDIF()
ENDIF()
ENDIF()
ENDIF()
IF(REQUIRE_PORTAUDIO AND NOT HAVE_PORTAUDIO)
MESSAGE(FATAL_ERROR "Failed to enabled required PortAudio backend")
ENDIF()
# Check PulseAudio backend
IF(PULSEAUDIO)
CHECK_INCLUDE_FILE(pulse/pulseaudio.h HAVE_PULSE_PULSEAUDIO_H)
IF(HAVE_PULSE_PULSEAUDIO_H)
CHECK_SHARED_FUNCTION_EXISTS(pa_context_new "pulse/pulseaudio.h" pulse "" HAVE_LIBPULSE)
IF(HAVE_LIBPULSE OR HAVE_DLFCN_H OR WIN32)
SET(HAVE_PULSEAUDIO 1)
SET(ALC_OBJS ${ALC_OBJS} Alc/backends/pulseaudio.c)
IF(HAVE_DLFCN_H OR WIN32)
SET(BACKENDS "${BACKENDS} PulseAudio,")
ELSE()
SET(BACKENDS "${BACKENDS} PulseAudio \(linked\),")
SET(EXTRA_LIBS pulse ${EXTRA_LIBS})
ENDIF()
ENDIF()
ENDIF()
ENDIF()
IF(REQUIRE_PULSEAUDIO AND NOT HAVE_PULSEAUDIO)
MESSAGE(FATAL_ERROR "Failed to enabled required PulseAudio backend")
ENDIF()
# Check CoreAudio backend
IF(COREAUDIO)
CHECK_INCLUDE_FILE(/System/Library/Frameworks/CoreAudio.framework/Headers/CoreAudio.h HAVE_COREAUDIO_FRAMEWORK)
IF(HAVE_COREAUDIO_FRAMEWORK)
SET(HAVE_COREAUDIO 1)
SET(ALC_OBJS ${ALC_OBJS} Alc/backends/coreaudio.c)
SET(BACKENDS "${BACKENDS} CoreAudio,")
SET(EXTRA_LIBS /System/Library/Frameworks/CoreAudio.framework ${EXTRA_LIBS})
SET(EXTRA_LIBS /System/Library/Frameworks/AudioUnit.framework ${EXTRA_LIBS})
SET(EXTRA_LIBS /System/Library/Frameworks/ApplicationServices.framework ${EXTRA_LIBS})
# Some versions of OSX may need the AudioToolbox framework. Add it if
# it's found.
FIND_LIBRARY(AUDIOTOOLBOX_LIBRARY
NAMES AudioToolbox
PATHS ~/Library/Frameworks
/Library/Frameworks
/System/Library/Frameworks
)
IF(AUDIOTOOLBOX_LIBRARY)
SET(EXTRA_LIBS ${AUDIOTOOLBOX_LIBRARY} ${EXTRA_LIBS})
ENDIF()
ENDIF()
ENDIF()
IF(REQUIRE_COREAUDIO AND NOT HAVE_COREAUDIO)
MESSAGE(FATAL_ERROR "Failed to enabled required CoreAudio backend")
ENDIF()
# Check for OpenSL (Android) backend
IF(OPENSL)
CHECK_INCLUDE_FILES("SLES/OpenSLES.h;SLES/OpenSLES_Android.h" HAVE_SLES_OPENSLES_ANDROID_H)
IF(HAVE_SLES_OPENSLES_ANDROID_H)
CHECK_SHARED_FUNCTION_EXISTS(slCreateEngine "SLES/OpenSLES.h" OpenSLES "" HAVE_LIBOPENSLES)
IF(HAVE_LIBOPENSLES)
SET(HAVE_OPENSL 1)
SET(ALC_OBJS ${ALC_OBJS} Alc/backends/opensl.c)
SET(BACKENDS "${BACKENDS} OpenSL,")
SET(EXTRA_LIBS OpenSLES ${EXTRA_LIBS})
ENDIF()
ENDIF()
ENDIF()
IF(REQUIRE_OPENSL AND NOT HAVE_OPENSL)
MESSAGE(FATAL_ERROR "Failed to enabled required OpenSL backend")
ENDIF()
# Optionally enable the Wave Writer backend
IF(WAVE)
SET(HAVE_WAVE 1)
SET(ALC_OBJS ${ALC_OBJS} Alc/backends/wave.c)
SET(BACKENDS "${BACKENDS} WaveFile,")
ENDIF()
# This is always available
SET(BACKENDS "${BACKENDS} Null")
IF(EXAMPLES)
# Might be able to use earlier versions, but these definitely work
PKG_CHECK_MODULES(FFMPEG libavcodec>=53.61.100 libavformat>=53.32.100 libavutil>=51.35.100)
ENDIF()
IF(LIBTYPE STREQUAL "STATIC")
ADD_DEFINITIONS(-DAL_LIBTYPE_STATIC)
SET(PKG_CONFIG_CFLAGS -DAL_LIBTYPE_STATIC ${PKG_CONFIG_CFLAGS})
ENDIF()
# Needed for openal.pc.in
SET(prefix ${CMAKE_INSTALL_PREFIX})
SET(exec_prefix "\${prefix}")
SET(libdir "\${exec_prefix}/lib${LIB_SUFFIX}")
SET(bindir "\${exec_prefix}/bin")
SET(includedir "\${prefix}/include")
SET(PACKAGE_VERSION "${LIB_VERSION}")
# End configuration
CONFIGURE_FILE(
"${OpenAL_SOURCE_DIR}/config.h.in"
"${OpenAL_BINARY_DIR}/config.h")
CONFIGURE_FILE(
"${OpenAL_SOURCE_DIR}/openal.pc.in"
"${OpenAL_BINARY_DIR}/openal.pc"
@ONLY)
ADD_DEFINITIONS(-DAL_BUILD_LIBRARY)
# Build a shared library
IF("${WIN32}")
SET(LIBNAME openal32)
ELSE()
SET(LIBNAME openal)
ENDIF()
ADD_LIBRARY(${LIBNAME} SHARED ${OPENAL_OBJS} ${ALC_OBJS})
SET_TARGET_PROPERTIES(${LIBNAME} PROPERTIES VERSION ${LIB_VERSION}
# Build a library
ADD_LIBRARY(${LIBNAME} ${LIBTYPE} ${OPENAL_OBJS} ${ALC_OBJS})
SET_TARGET_PROPERTIES(${LIBNAME} PROPERTIES DEFINE_SYMBOL AL_BUILD_LIBRARY
COMPILE_FLAGS -DAL_ALEXT_PROTOTYPES
VERSION ${LIB_VERSION}.0
SOVERSION ${LIB_MAJOR_VERSION})
IF("${WIN32}")
IF(WIN32 AND NOT LIBTYPE STREQUAL "STATIC")
SET_TARGET_PROPERTIES(${LIBNAME} PROPERTIES PREFIX "")
ENDIF()
@@ -279,15 +762,91 @@ TARGET_LINK_LIBRARIES(${LIBNAME} ${EXTRA_LIBS})
# Add an install target here
INSTALL(TARGETS ${LIBNAME}
RUNTIME DESTINATION bin
LIBRARY DESTINATION lib
ARCHIVE DESTINATION lib
LIBRARY DESTINATION "lib${LIB_SUFFIX}"
ARCHIVE DESTINATION "lib${LIB_SUFFIX}"
)
INSTALL(FILES include/AL/al.h
include/AL/alc.h
include/AL/alext.h
include/AL/efx.h
include/AL/efx-creative.h
include/AL/efx-presets.h
DESTINATION include/AL
)
INSTALL(FILES "${OpenAL_BINARY_DIR}/openal.pc"
DESTINATION "lib${LIB_SUFFIX}/pkgconfig")
MESSAGE(STATUS "")
MESSAGE(STATUS "Building OpenAL with support for the following backends:")
MESSAGE(STATUS " ${BACKENDS}")
MESSAGE(STATUS "")
MESSAGE(STATUS "Building with support for CPU extensions:")
MESSAGE(STATUS " ${CPU_EXTS}")
MESSAGE(STATUS "")
IF(WIN32)
IF(NOT HAVE_DSOUND)
MESSAGE(STATUS "WARNING: Building the Windows version without DirectSound output")
MESSAGE(STATUS " This is probably NOT what you want!")
MESSAGE(STATUS "")
ENDIF()
ENDIF()
# Install alsoft.conf configuration file
IF(ALSOFT_CONFIG)
INSTALL(FILES alsoftrc.sample
DESTINATION share/openal
)
MESSAGE(STATUS "Installing sample configuration")
MESSAGE(STATUS "")
ENDIF()
IF(UTILS)
ADD_EXECUTABLE(openal-info utils/openal-info.c)
TARGET_LINK_LIBRARIES(openal-info ${LIBNAME})
ADD_EXECUTABLE(makehrtf utils/makehrtf.c)
IF(HAVE_LIBM)
TARGET_LINK_LIBRARIES(makehrtf m)
ENDIF()
INSTALL(TARGETS openal-info makehrtf
RUNTIME DESTINATION bin
LIBRARY DESTINATION "lib${LIB_SUFFIX}"
ARCHIVE DESTINATION "lib${LIB_SUFFIX}"
)
MESSAGE(STATUS "Building utility programs")
MESSAGE(STATUS "")
ENDIF()
IF(EXAMPLES)
IF(FFMPEG_FOUND)
ADD_EXECUTABLE(alstream examples/common/alhelpers.c
examples/common/alffmpeg.c
examples/alstream.c)
TARGET_LINK_LIBRARIES(alstream ${FFMPEG_LIBRARIES} ${LIBNAME})
SET_TARGET_PROPERTIES(alstream PROPERTIES COMPILE_FLAGS "${FFMPEG_CFLAGS}")
ADD_EXECUTABLE(alreverb examples/common/alhelpers.c
examples/common/alffmpeg.c
examples/alreverb.c)
TARGET_LINK_LIBRARIES(alreverb ${FFMPEG_LIBRARIES} ${LIBNAME})
SET_TARGET_PROPERTIES(alreverb PROPERTIES COMPILE_FLAGS "${FFMPEG_CFLAGS}")
ADD_EXECUTABLE(allatency examples/common/alhelpers.c
examples/common/alffmpeg.c
examples/allatency.c)
TARGET_LINK_LIBRARIES(allatency ${FFMPEG_LIBRARIES} ${LIBNAME})
SET_TARGET_PROPERTIES(allatency PROPERTIES COMPILE_FLAGS "${FFMPEG_CFLAGS}")
INSTALL(TARGETS alstream alreverb allatency
RUNTIME DESTINATION bin
LIBRARY DESTINATION "lib${LIB_SUFFIX}"
ARCHIVE DESTINATION "lib${LIB_SUFFIX}"
)
MESSAGE(STATUS "Building ffmpeg example programs")
MESSAGE(STATUS "")
ENDIF()
ENDIF()
+61
View File
@@ -0,0 +1,61 @@
#ifndef _AL_AUXEFFECTSLOT_H_
#define _AL_AUXEFFECTSLOT_H_
#include "alMain.h"
#include "alEffect.h"
#ifdef __cplusplus
extern "C" {
#endif
typedef struct ALeffectState {
ALvoid (*Destroy)(struct ALeffectState *State);
ALboolean (*DeviceUpdate)(struct ALeffectState *State, ALCdevice *Device);
ALvoid (*Update)(struct ALeffectState *State, ALCdevice *Device, const struct ALeffectslot *Slot);
ALvoid (*Process)(struct ALeffectState *State, ALuint SamplesToDo, const ALfloat *RESTRICT SamplesIn, ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE]);
} ALeffectState;
typedef struct ALeffectslot
{
ALeffect effect;
volatile ALfloat Gain;
volatile ALboolean AuxSendAuto;
volatile ALenum NeedsUpdate;
ALeffectState *EffectState;
ALIGN(16) ALfloat WetBuffer[1][BUFFERSIZE];
ALfloat ClickRemoval[1];
ALfloat PendingClicks[1];
RefCount ref;
/* Self ID */
ALuint id;
} ALeffectslot;
ALenum InitEffectSlot(ALeffectslot *slot);
ALvoid ReleaseALAuxiliaryEffectSlots(ALCcontext *Context);
ALeffectState *NoneCreate(void);
ALeffectState *ReverbCreate(void);
ALeffectState *EchoCreate(void);
ALeffectState *ModulatorCreate(void);
ALeffectState *DedicatedCreate(void);
#define ALeffectState_Destroy(a) ((a)->Destroy((a)))
#define ALeffectState_DeviceUpdate(a,b) ((a)->DeviceUpdate((a),(b)))
#define ALeffectState_Update(a,b,c) ((a)->Update((a),(b),(c)))
#define ALeffectState_Process(a,b,c,d) ((a)->Process((a),(b),(c),(d)))
ALenum InitializeEffect(ALCdevice *Device, ALeffectslot *EffectSlot, ALeffect *effect);
#ifdef __cplusplus
}
#endif
#endif
+84 -14
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@@ -1,29 +1,99 @@
#ifndef _AL_BUFFER_H_
#define _AL_BUFFER_H_
#include "AL/al.h"
#include "alMain.h"
#ifdef __cplusplus
extern "C" {
#endif
#define UNUSED 0
#define PENDING 1
#define PROCESSED 2
/* User formats */
enum UserFmtType {
UserFmtByte = AL_BYTE_SOFT,
UserFmtUByte = AL_UNSIGNED_BYTE_SOFT,
UserFmtShort = AL_SHORT_SOFT,
UserFmtUShort = AL_UNSIGNED_SHORT_SOFT,
UserFmtInt = AL_INT_SOFT,
UserFmtUInt = AL_UNSIGNED_INT_SOFT,
UserFmtFloat = AL_FLOAT_SOFT,
UserFmtDouble = AL_DOUBLE_SOFT,
UserFmtByte3 = AL_BYTE3_SOFT,
UserFmtUByte3 = AL_UNSIGNED_BYTE3_SOFT,
UserFmtMulaw,
UserFmtAlaw,
UserFmtIMA4,
};
enum UserFmtChannels {
UserFmtMono = AL_MONO_SOFT,
UserFmtStereo = AL_STEREO_SOFT,
UserFmtRear = AL_REAR_SOFT,
UserFmtQuad = AL_QUAD_SOFT,
UserFmtX51 = AL_5POINT1_SOFT, /* (WFX order) */
UserFmtX61 = AL_6POINT1_SOFT, /* (WFX order) */
UserFmtX71 = AL_7POINT1_SOFT, /* (WFX order) */
};
typedef struct ALbuffer_struct
ALuint BytesFromUserFmt(enum UserFmtType type);
ALuint ChannelsFromUserFmt(enum UserFmtChannels chans);
static __inline ALuint FrameSizeFromUserFmt(enum UserFmtChannels chans,
enum UserFmtType type)
{
ALenum format;
ALenum eOriginalFormat;
ALshort *data;
ALsizei size;
ALsizei frequency;
ALenum state;
ALuint refcount; // Number of sources using this buffer (deletion can only occur when this is 0)
struct ALbuffer_struct *next;
return ChannelsFromUserFmt(chans) * BytesFromUserFmt(type);
}
/* Storable formats */
enum FmtType {
FmtByte = UserFmtByte,
FmtShort = UserFmtShort,
FmtFloat = UserFmtFloat,
};
enum FmtChannels {
FmtMono = UserFmtMono,
FmtStereo = UserFmtStereo,
FmtRear = UserFmtRear,
FmtQuad = UserFmtQuad,
FmtX51 = UserFmtX51,
FmtX61 = UserFmtX61,
FmtX71 = UserFmtX71,
};
ALuint BytesFromFmt(enum FmtType type);
ALuint ChannelsFromFmt(enum FmtChannels chans);
static __inline ALuint FrameSizeFromFmt(enum FmtChannels chans, enum FmtType type)
{
return ChannelsFromFmt(chans) * BytesFromFmt(type);
}
typedef struct ALbuffer
{
ALvoid *data;
ALsizei Frequency;
ALenum Format;
ALsizei SampleLen;
enum FmtChannels FmtChannels;
enum FmtType FmtType;
enum UserFmtChannels OriginalChannels;
enum UserFmtType OriginalType;
ALsizei OriginalSize;
ALsizei LoopStart;
ALsizei LoopEnd;
/* Number of times buffer was attached to a source (deletion can only occur when 0) */
RefCount ref;
RWLock lock;
/* Self ID */
ALuint id;
} ALbuffer;
ALvoid ReleaseALBuffers(ALvoid);
ALvoid ReleaseALBuffers(ALCdevice *device);
#ifdef __cplusplus
}
+114
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@@ -0,0 +1,114 @@
#ifndef _AL_EFFECT_H_
#define _AL_EFFECT_H_
#include "alMain.h"
#ifdef __cplusplus
extern "C" {
#endif
enum {
EAXREVERB = 0,
REVERB,
ECHO,
MODULATOR,
DEDICATED,
MAX_EFFECTS
};
extern ALboolean DisabledEffects[MAX_EFFECTS];
extern ALfloat ReverbBoost;
extern ALboolean EmulateEAXReverb;
typedef struct ALeffect
{
// Effect type (AL_EFFECT_NULL, ...)
ALenum type;
struct {
// Shared Reverb Properties
ALfloat Density;
ALfloat Diffusion;
ALfloat Gain;
ALfloat GainHF;
ALfloat DecayTime;
ALfloat DecayHFRatio;
ALfloat ReflectionsGain;
ALfloat ReflectionsDelay;
ALfloat LateReverbGain;
ALfloat LateReverbDelay;
ALfloat AirAbsorptionGainHF;
ALfloat RoomRolloffFactor;
ALboolean DecayHFLimit;
// Additional EAX Reverb Properties
ALfloat GainLF;
ALfloat DecayLFRatio;
ALfloat ReflectionsPan[3];
ALfloat LateReverbPan[3];
ALfloat EchoTime;
ALfloat EchoDepth;
ALfloat ModulationTime;
ALfloat ModulationDepth;
ALfloat HFReference;
ALfloat LFReference;
} Reverb;
struct {
ALfloat Delay;
ALfloat LRDelay;
ALfloat Damping;
ALfloat Feedback;
ALfloat Spread;
} Echo;
struct {
ALfloat Frequency;
ALfloat HighPassCutoff;
ALint Waveform;
} Modulator;
struct {
ALfloat Gain;
} Dedicated;
void (*SetParami)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALint val);
void (*SetParamiv)(struct ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals);
void (*SetParamf)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val);
void (*SetParamfv)(struct ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals);
void (*GetParami)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALint *val);
void (*GetParamiv)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals);
void (*GetParamf)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val);
void (*GetParamfv)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals);
/* Self ID */
ALuint id;
} ALeffect;
#define ALeffect_SetParami(x, c, p, v) ((x)->SetParami((x),(c),(p),(v)))
#define ALeffect_SetParamiv(x, c, p, v) ((x)->SetParamiv((x),(c),(p),(v)))
#define ALeffect_SetParamf(x, c, p, v) ((x)->SetParamf((x),(c),(p),(v)))
#define ALeffect_SetParamfv(x, c, p, v) ((x)->SetParamfv((x),(c),(p),(v)))
#define ALeffect_GetParami(x, c, p, v) ((x)->GetParami((x),(c),(p),(v)))
#define ALeffect_GetParamiv(x, c, p, v) ((x)->GetParamiv((x),(c),(p),(v)))
#define ALeffect_GetParamf(x, c, p, v) ((x)->GetParamf((x),(c),(p),(v)))
#define ALeffect_GetParamfv(x, c, p, v) ((x)->GetParamfv((x),(c),(p),(v)))
static __inline ALboolean IsReverbEffect(ALenum type)
{ return type == AL_EFFECT_REVERB || type == AL_EFFECT_EAXREVERB; }
ALenum InitEffect(ALeffect *effect);
ALvoid ReleaseALEffects(ALCdevice *device);
ALvoid LoadReverbPreset(const char *name, ALeffect *effect);
#ifdef __cplusplus
}
#endif
#endif
+4 -2
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@@ -1,13 +1,15 @@
#ifndef _AL_ERROR_H_
#define _AL_ERROR_H_
#include "AL/al.h"
#include "alMain.h"
#ifdef __cplusplus
extern "C" {
#endif
ALvoid alSetError(ALenum errorCode);
extern ALboolean TrapALError;
ALvoid alSetError(ALCcontext *Context, ALenum errorCode);
#ifdef __cplusplus
}
-33
View File
@@ -1,33 +0,0 @@
#ifndef _AL_EXTENSION_H_
#define _AL_EXTENSION_H_
#include "AL/al.h"
#include "AL/alc.h"
#ifdef __cplusplus
extern "C" {
#endif
typedef struct ALextension_struct
{
ALchar *extName;
ALvoid *address;
} ALextension;
typedef struct ALfunction_struct
{
ALchar *funcName;
ALvoid *address;
} ALfunction;
typedef struct ALenum_struct
{
ALchar *enumName;
ALenum value;
} ALenums;
#ifdef __cplusplus
}
#endif
#endif
+90
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@@ -0,0 +1,90 @@
#ifndef _AL_FILTER_H_
#define _AL_FILTER_H_
#include "alMain.h"
#ifdef __cplusplus
extern "C" {
#endif
#define LOWPASSFREQREF (5000)
typedef struct {
ALfloat coeff;
#ifndef _MSC_VER
ALfloat history[0];
#else
ALfloat history[1];
#endif
} FILTER;
static __inline ALfloat lpFilter2P(FILTER *iir, ALuint offset, ALfloat input)
{
ALfloat *history = &iir->history[offset*2];
ALfloat a = iir->coeff;
ALfloat output = input;
output = output + (history[0]-output)*a;
history[0] = output;
output = output + (history[1]-output)*a;
history[1] = output;
return output;
}
static __inline ALfloat lpFilter2PC(const FILTER *iir, ALuint offset, ALfloat input)
{
const ALfloat *history = &iir->history[offset*2];
ALfloat a = iir->coeff;
ALfloat output = input;
output = output + (history[0]-output)*a;
output = output + (history[1]-output)*a;
return output;
}
/* Calculates the low-pass filter coefficient given the pre-scaled gain and
* cos(w) value. Note that g should be pre-scaled (sqr(gain) for one-pole,
* sqrt(gain) for four-pole, etc) */
ALfloat lpCoeffCalc(ALfloat g, ALfloat cw);
typedef struct ALfilter {
// Filter type (AL_FILTER_NULL, ...)
ALenum type;
ALfloat Gain;
ALfloat GainHF;
void (*SetParami)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALint val);
void (*SetParamiv)(struct ALfilter *filter, ALCcontext *context, ALenum param, const ALint *vals);
void (*SetParamf)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val);
void (*SetParamfv)(struct ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals);
void (*GetParami)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALint *val);
void (*GetParamiv)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALint *vals);
void (*GetParamf)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *val);
void (*GetParamfv)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals);
/* Self ID */
ALuint id;
} ALfilter;
#define ALfilter_SetParami(x, c, p, v) ((x)->SetParami((x),(c),(p),(v)))
#define ALfilter_SetParamiv(x, c, p, v) ((x)->SetParamiv((x),(c),(p),(v)))
#define ALfilter_SetParamf(x, c, p, v) ((x)->SetParamf((x),(c),(p),(v)))
#define ALfilter_SetParamfv(x, c, p, v) ((x)->SetParamfv((x),(c),(p),(v)))
#define ALfilter_GetParami(x, c, p, v) ((x)->GetParami((x),(c),(p),(v)))
#define ALfilter_GetParamiv(x, c, p, v) ((x)->GetParamiv((x),(c),(p),(v)))
#define ALfilter_GetParamf(x, c, p, v) ((x)->GetParamf((x),(c),(p),(v)))
#define ALfilter_GetParamfv(x, c, p, v) ((x)->GetParamfv((x),(c),(p),(v)))
ALvoid ReleaseALFilters(ALCdevice *device);
#ifdef __cplusplus
}
#endif
#endif
+13 -8
View File
@@ -1,19 +1,24 @@
#ifndef _AL_LISTENER_H_
#define _AL_LISTENER_H_
#include "AL/al.h"
#include "alMain.h"
#ifdef __cplusplus
extern "C" {
#endif
typedef struct ALlistener_struct
{
ALfloat Position[3];
ALfloat Velocity[3];
ALfloat Forward[3];
ALfloat Up[3];
ALfloat Gain;
typedef struct ALlistener {
volatile ALfloat Position[3];
volatile ALfloat Velocity[3];
volatile ALfloat Forward[3];
volatile ALfloat Up[3];
volatile ALfloat Gain;
volatile ALfloat MetersPerUnit;
struct {
ALfloat Matrix[4][4];
ALfloat Velocity[3];
} Params;
} ALlistener;
#ifdef __cplusplus
+786 -119
View File
@@ -2,197 +2,747 @@
#define AL_MAIN_H
#include <string.h>
#include <stdio.h>
#include <stdarg.h>
#include <assert.h>
#include <math.h>
#include "alu.h"
#ifdef HAVE_FENV_H
#include <fenv.h>
#endif
#include "AL/al.h"
#include "AL/alc.h"
#include "AL/alext.h"
#ifndef AL_SOFT_deferred_updates
#define AL_SOFT_deferred_updates 1
#define AL_DEFERRED_UPDATES_SOFT 0xC002
typedef ALvoid (AL_APIENTRY*LPALDEFERUPDATESSOFT)(void);
typedef ALvoid (AL_APIENTRY*LPALPROCESSUPDATESSOFT)(void);
#ifdef AL_ALEXT_PROTOTYPES
AL_API ALvoid AL_APIENTRY alDeferUpdatesSOFT(void);
AL_API ALvoid AL_APIENTRY alProcessUpdatesSOFT(void);
#endif
#endif
#if defined(HAVE_STDINT_H)
#include <stdint.h>
typedef int64_t ALint64;
typedef uint64_t ALuint64;
#elif defined(HAVE___INT64)
typedef __int64 ALint64;
typedef unsigned __int64 ALuint64;
#elif (SIZEOF_LONG == 8)
typedef long ALint64;
typedef unsigned long ALuint64;
#elif (SIZEOF_LONG_LONG == 8)
typedef long long ALint64;
typedef unsigned long long ALuint64;
#endif
typedef ptrdiff_t ALintptrEXT;
typedef ptrdiff_t ALsizeiptrEXT;
#define MAKEU64(x,y) (((ALuint64)(x)<<32)|(ALuint64)(y))
#ifdef HAVE_GCC_FORMAT
#define PRINTF_STYLE(x, y) __attribute__((format(printf, (x), (y))))
#else
#define PRINTF_STYLE(x, y)
#endif
static const union {
ALuint u;
ALubyte b[sizeof(ALuint)];
} EndianTest = { 1 };
#define IS_LITTLE_ENDIAN (EndianTest.b[0] == 1)
#define COUNTOF(x) (sizeof((x))/sizeof((x)[0]))
#ifdef _WIN32
#define WIN32_LEAN_AND_MEAN
#include <windows.h>
//#define strcasecmp _stricmp
typedef DWORD pthread_key_t;
int pthread_key_create(pthread_key_t *key, void (*callback)(void*));
int pthread_key_delete(pthread_key_t key);
void *pthread_getspecific(pthread_key_t key);
int pthread_setspecific(pthread_key_t key, void *val);
#define HAVE_DYNLOAD 1
void *LoadLib(const char *name);
void CloseLib(void *handle);
void *GetSymbol(void *handle, const char *name);
WCHAR *strdupW(const WCHAR *str);
typedef LONG pthread_once_t;
#define PTHREAD_ONCE_INIT 0
void pthread_once(pthread_once_t *once, void (*callback)(void));
static __inline int sched_yield(void)
{ SwitchToThread(); return 0; }
#else
#include <unistd.h>
#include <assert.h>
#include <pthread.h>
#include <sys/time.h>
#include <time.h>
#include <errno.h>
#define IsBadWritePtr(a,b) (0)
#define IsBadWritePtr(a,b) ((a) == NULL && (b) != 0)
typedef pthread_mutex_t CRITICAL_SECTION;
static inline void EnterCriticalSection(CRITICAL_SECTION *cs)
{
int ret;
ret = pthread_mutex_lock(cs);
assert(ret == 0);
}
static inline void LeaveCriticalSection(CRITICAL_SECTION *cs)
{
int ret;
ret = pthread_mutex_unlock(cs);
assert(ret == 0);
}
static inline void InitializeCriticalSection(CRITICAL_SECTION *cs)
{
pthread_mutexattr_t attrib;
int ret;
void InitializeCriticalSection(CRITICAL_SECTION *cs);
void DeleteCriticalSection(CRITICAL_SECTION *cs);
void EnterCriticalSection(CRITICAL_SECTION *cs);
void LeaveCriticalSection(CRITICAL_SECTION *cs);
ret = pthread_mutexattr_init(&attrib);
assert(ret == 0);
ALuint timeGetTime(void);
void Sleep(ALuint t);
ret = pthread_mutexattr_settype(&attrib, PTHREAD_MUTEX_RECURSIVE);
assert(ret == 0);
ret = pthread_mutex_init(cs, &attrib);
assert(ret == 0);
pthread_mutexattr_destroy(&attrib);
}
static inline void DeleteCriticalSection(CRITICAL_SECTION *cs)
{
int ret;
ret = pthread_mutex_destroy(cs);
assert(ret == 0);
}
#define min(x,y) (((x)<(y))?(x):(y))
#define max(x,y) (((x)>(y))?(x):(y))
#if defined(HAVE_DLFCN_H)
#define HAVE_DYNLOAD 1
void *LoadLib(const char *name);
void CloseLib(void *handle);
void *GetSymbol(void *handle, const char *name);
#endif
#include "alListener.h"
#ifdef __cplusplus
extern "C"
{
#endif
extern CRITICAL_SECTION _alMutex;
typedef void *volatile XchgPtr;
extern char _alDebug[256];
#if defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 1))
typedef ALuint RefCount;
static __inline RefCount IncrementRef(volatile RefCount *ptr)
{ return __sync_add_and_fetch(ptr, 1); }
static __inline RefCount DecrementRef(volatile RefCount *ptr)
{ return __sync_sub_and_fetch(ptr, 1); }
#define AL_PRINT(...) do { \
int _al_print_i; \
char *_al_print_fn = strrchr(__FILE__, '/'); \
if(!_al_print_fn) _al_print_fn = __FILE__; \
else _al_print_fn += 1; \
_al_print_i = snprintf(_alDebug, sizeof(_alDebug), "AL lib: %s:%d: ", _al_print_fn, __LINE__); \
if(_al_print_i < (int)sizeof(_alDebug) && _al_print_i > 0) \
snprintf(_alDebug+_al_print_i, sizeof(_alDebug)-_al_print_i, __VA_ARGS__); \
_alDebug[sizeof(_alDebug)-1] = 0; \
fprintf(stderr, "%s", _alDebug); \
} while(0)
static __inline int ExchangeInt(volatile int *ptr, int newval)
{
return __sync_lock_test_and_set(ptr, newval);
}
static __inline void *ExchangePtr(XchgPtr *ptr, void *newval)
{
return __sync_lock_test_and_set(ptr, newval);
}
static __inline ALboolean CompExchangeInt(volatile int *ptr, int oldval, int newval)
{
return __sync_bool_compare_and_swap(ptr, oldval, newval);
}
static __inline ALboolean CompExchangePtr(XchgPtr *ptr, void *oldval, void *newval)
{
return __sync_bool_compare_and_swap(ptr, oldval, newval);
}
#elif defined(__GNUC__) && (defined(__i386__) || defined(__x86_64__))
#define AL_FORMAT_MONO_FLOAT32 0x10010
#define AL_FORMAT_STEREO_FLOAT32 0x10011
static __inline int xaddl(volatile int *dest, int incr)
{
int ret;
__asm__ __volatile__("lock; xaddl %0,(%1)"
: "=r" (ret)
: "r" (dest), "0" (incr)
: "memory");
return ret;
}
#define AL_FORMAT_MONO_IMA4 0x1300
#define AL_FORMAT_STEREO_IMA4 0x1301
typedef int RefCount;
static __inline RefCount IncrementRef(volatile RefCount *ptr)
{ return xaddl(ptr, 1)+1; }
static __inline RefCount DecrementRef(volatile RefCount *ptr)
{ return xaddl(ptr, -1)-1; }
#define AL_FORMAT_51CHN8 0x120A
#define AL_FORMAT_51CHN16 0x120B
#define AL_FORMAT_51CHN32 0x120C
#define AL_FORMAT_61CHN8 0x120D
#define AL_FORMAT_61CHN16 0x120E
#define AL_FORMAT_61CHN32 0x120F
#define AL_FORMAT_71CHN8 0x1210
#define AL_FORMAT_71CHN16 0x1211
#define AL_FORMAT_71CHN32 0x1212
#define AL_FORMAT_QUAD8 0x1204
#define AL_FORMAT_QUAD16 0x1205
#define AL_FORMAT_QUAD32 0x1206
#define AL_FORMAT_REAR8 0x1207
#define AL_FORMAT_REAR16 0x1208
#define AL_FORMAT_REAR32 0x1209
static __inline int ExchangeInt(volatile int *dest, int newval)
{
int ret;
__asm__ __volatile__("lock; xchgl %0,(%1)"
: "=r" (ret)
: "r" (dest), "0" (newval)
: "memory");
return ret;
}
#define SWMIXER_OUTPUT_RATE 44100
static __inline ALboolean CompExchangeInt(volatile int *dest, int oldval, int newval)
{
int ret;
__asm__ __volatile__("lock; cmpxchgl %2,(%1)"
: "=a" (ret)
: "r" (dest), "r" (newval), "0" (oldval)
: "memory");
return ret == oldval;
}
static __inline void *ExchangePtr(XchgPtr *dest, void *newval)
{
void *ret;
__asm__ __volatile__(
#ifdef __i386__
"lock; xchgl %0,(%1)"
#else
"lock; xchgq %0,(%1)"
#endif
: "=r" (ret)
: "r" (dest), "0" (newval)
: "memory"
);
return ret;
}
static __inline ALboolean CompExchangePtr(XchgPtr *dest, void *oldval, void *newval)
{
void *ret;
__asm__ __volatile__(
#ifdef __i386__
"lock; cmpxchgl %2,(%1)"
#else
"lock; cmpxchgq %2,(%1)"
#endif
: "=a" (ret)
: "r" (dest), "r" (newval), "0" (oldval)
: "memory"
);
return ret == oldval;
}
#elif defined(_WIN32)
typedef LONG RefCount;
static __inline RefCount IncrementRef(volatile RefCount *ptr)
{ return InterlockedIncrement(ptr); }
static __inline RefCount DecrementRef(volatile RefCount *ptr)
{ return InterlockedDecrement(ptr); }
extern ALbyte LONG_size_does_not_match_int[(sizeof(LONG)==sizeof(int))?1:-1];
static __inline int ExchangeInt(volatile int *ptr, int newval)
{
union {
volatile int *i;
volatile LONG *l;
} u = { ptr };
return InterlockedExchange(u.l, newval);
}
static __inline void *ExchangePtr(XchgPtr *ptr, void *newval)
{
return InterlockedExchangePointer(ptr, newval);
}
static __inline ALboolean CompExchangeInt(volatile int *ptr, int oldval, int newval)
{
union {
volatile int *i;
volatile LONG *l;
} u = { ptr };
return InterlockedCompareExchange(u.l, newval, oldval) == oldval;
}
static __inline ALboolean CompExchangePtr(XchgPtr *ptr, void *oldval, void *newval)
{
return InterlockedCompareExchangePointer(ptr, newval, oldval) == oldval;
}
#elif defined(__APPLE__)
#include <libkern/OSAtomic.h>
typedef int32_t RefCount;
static __inline RefCount IncrementRef(volatile RefCount *ptr)
{ return OSAtomicIncrement32Barrier(ptr); }
static __inline RefCount DecrementRef(volatile RefCount *ptr)
{ return OSAtomicDecrement32Barrier(ptr); }
static __inline int ExchangeInt(volatile int *ptr, int newval)
{
/* Really? No regular old atomic swap? */
int oldval;
do {
oldval = *ptr;
} while(!OSAtomicCompareAndSwap32Barrier(oldval, newval, ptr));
return oldval;
}
static __inline void *ExchangePtr(XchgPtr *ptr, void *newval)
{
void *oldval;
do {
oldval = *ptr;
} while(!OSAtomicCompareAndSwapPtrBarrier(oldval, newval, ptr));
return oldval;
}
static __inline ALboolean CompExchangeInt(volatile int *ptr, int oldval, int newval)
{
return OSAtomicCompareAndSwap32Barrier(oldval, newval, ptr);
}
static __inline ALboolean CompExchangePtr(XchgPtr *ptr, void *oldval, void *newval)
{
return OSAtomicCompareAndSwapPtrBarrier(oldval, newval, ptr);
}
#else
#error "No atomic functions available on this platform!"
typedef ALuint RefCount;
#endif
#define SPEEDOFSOUNDMETRESPERSEC (343.3f)
typedef struct {
ALCboolean (*OpenPlayback)(ALCdevice*, const ALCchar*);
void (*ClosePlayback)(ALCdevice*);
volatile RefCount read_count;
volatile RefCount write_count;
volatile ALenum read_lock;
volatile ALenum read_entry_lock;
volatile ALenum write_lock;
} RWLock;
ALCboolean (*OpenCapture)(ALCdevice*, const ALCchar*, ALCuint, ALCenum, ALCsizei);
void RWLockInit(RWLock *lock);
void ReadLock(RWLock *lock);
void ReadUnlock(RWLock *lock);
void WriteLock(RWLock *lock);
void WriteUnlock(RWLock *lock);
typedef struct UIntMap {
struct {
ALuint key;
ALvoid *value;
} *array;
ALsizei size;
ALsizei maxsize;
ALsizei limit;
RWLock lock;
} UIntMap;
extern UIntMap TlsDestructor;
void InitUIntMap(UIntMap *map, ALsizei limit);
void ResetUIntMap(UIntMap *map);
ALenum InsertUIntMapEntry(UIntMap *map, ALuint key, ALvoid *value);
ALvoid *RemoveUIntMapKey(UIntMap *map, ALuint key);
ALvoid *LookupUIntMapKey(UIntMap *map, ALuint key);
static __inline void LockUIntMapRead(UIntMap *map)
{ ReadLock(&map->lock); }
static __inline void UnlockUIntMapRead(UIntMap *map)
{ ReadUnlock(&map->lock); }
static __inline void LockUIntMapWrite(UIntMap *map)
{ WriteLock(&map->lock); }
static __inline void UnlockUIntMapWrite(UIntMap *map)
{ WriteUnlock(&map->lock); }
#ifdef __cplusplus
extern "C" {
#endif
struct Hrtf;
#define DEFAULT_OUTPUT_RATE (44100)
#define MIN_OUTPUT_RATE (8000)
// Find the next power-of-2 for non-power-of-2 numbers.
static __inline ALuint NextPowerOf2(ALuint value)
{
if(value > 0)
{
value--;
value |= value>>1;
value |= value>>2;
value |= value>>4;
value |= value>>8;
value |= value>>16;
}
return value+1;
}
/* Fast float-to-int conversion. Assumes the FPU is already in round-to-zero
* mode. */
static __inline ALint fastf2i(ALfloat f)
{
#ifdef HAVE_LRINTF
return lrintf(f);
#elif defined(_MSC_VER) && defined(_M_IX86)
ALint i;
__asm fld f
__asm fistp i
return i;
#else
return (ALint)f;
#endif
}
/* Fast float-to-uint conversion. Assumes the FPU is already in round-to-zero
* mode. */
static __inline ALuint fastf2u(ALfloat f)
{ return fastf2i(f); }
enum DevProbe {
ALL_DEVICE_PROBE,
CAPTURE_DEVICE_PROBE
};
typedef struct {
ALCenum (*OpenPlayback)(ALCdevice*, const ALCchar*);
void (*ClosePlayback)(ALCdevice*);
ALCboolean (*ResetPlayback)(ALCdevice*);
ALCboolean (*StartPlayback)(ALCdevice*);
void (*StopPlayback)(ALCdevice*);
ALCenum (*OpenCapture)(ALCdevice*, const ALCchar*);
void (*CloseCapture)(ALCdevice*);
void (*StartCapture)(ALCdevice*);
void (*StopCapture)(ALCdevice*);
void (*CaptureSamples)(ALCdevice*, void*, ALCuint);
ALCenum (*CaptureSamples)(ALCdevice*, void*, ALCuint);
ALCuint (*AvailableSamples)(ALCdevice*);
void (*Lock)(ALCdevice*);
void (*Unlock)(ALCdevice*);
ALint64 (*GetLatency)(ALCdevice*);
} BackendFuncs;
void alc_alsa_init(BackendFuncs *func_list);
void alc_oss_init(BackendFuncs *func_list);
void alcDSoundInit(BackendFuncs *func_list);
void alcWinMMInit(BackendFuncs *FuncList);
struct BackendInfo {
const char *name;
ALCboolean (*Init)(BackendFuncs*);
void (*Deinit)(void);
void (*Probe)(enum DevProbe);
BackendFuncs Funcs;
};
ALCboolean alc_alsa_init(BackendFuncs *func_list);
void alc_alsa_deinit(void);
void alc_alsa_probe(enum DevProbe type);
ALCboolean alc_oss_init(BackendFuncs *func_list);
void alc_oss_deinit(void);
void alc_oss_probe(enum DevProbe type);
ALCboolean alc_solaris_init(BackendFuncs *func_list);
void alc_solaris_deinit(void);
void alc_solaris_probe(enum DevProbe type);
ALCboolean alc_sndio_init(BackendFuncs *func_list);
void alc_sndio_deinit(void);
void alc_sndio_probe(enum DevProbe type);
ALCboolean alcMMDevApiInit(BackendFuncs *func_list);
void alcMMDevApiDeinit(void);
void alcMMDevApiProbe(enum DevProbe type);
ALCboolean alcDSoundInit(BackendFuncs *func_list);
void alcDSoundDeinit(void);
void alcDSoundProbe(enum DevProbe type);
ALCboolean alcWinMMInit(BackendFuncs *FuncList);
void alcWinMMDeinit(void);
void alcWinMMProbe(enum DevProbe type);
ALCboolean alc_pa_init(BackendFuncs *func_list);
void alc_pa_deinit(void);
void alc_pa_probe(enum DevProbe type);
ALCboolean alc_wave_init(BackendFuncs *func_list);
void alc_wave_deinit(void);
void alc_wave_probe(enum DevProbe type);
ALCboolean alc_pulse_init(BackendFuncs *func_list);
void alc_pulse_deinit(void);
void alc_pulse_probe(enum DevProbe type);
ALCboolean alc_ca_init(BackendFuncs *func_list);
void alc_ca_deinit(void);
void alc_ca_probe(enum DevProbe type);
ALCboolean alc_opensl_init(BackendFuncs *func_list);
void alc_opensl_deinit(void);
void alc_opensl_probe(enum DevProbe type);
ALCboolean alc_null_init(BackendFuncs *func_list);
void alc_null_deinit(void);
void alc_null_probe(enum DevProbe type);
ALCboolean alc_loopback_init(BackendFuncs *func_list);
void alc_loopback_deinit(void);
void alc_loopback_probe(enum DevProbe type);
enum DistanceModel {
InverseDistanceClamped = AL_INVERSE_DISTANCE_CLAMPED,
LinearDistanceClamped = AL_LINEAR_DISTANCE_CLAMPED,
ExponentDistanceClamped = AL_EXPONENT_DISTANCE_CLAMPED,
InverseDistance = AL_INVERSE_DISTANCE,
LinearDistance = AL_LINEAR_DISTANCE,
ExponentDistance = AL_EXPONENT_DISTANCE,
DisableDistance = AL_NONE,
DefaultDistanceModel = InverseDistanceClamped
};
enum Resampler {
PointResampler,
LinearResampler,
CubicResampler,
ResamplerMax,
};
enum Channel {
FrontLeft = 0,
FrontRight,
FrontCenter,
LFE,
BackLeft,
BackRight,
BackCenter,
SideLeft,
SideRight,
MaxChannels,
};
/* Device formats */
enum DevFmtType {
DevFmtByte = ALC_BYTE_SOFT,
DevFmtUByte = ALC_UNSIGNED_BYTE_SOFT,
DevFmtShort = ALC_SHORT_SOFT,
DevFmtUShort = ALC_UNSIGNED_SHORT_SOFT,
DevFmtInt = ALC_INT_SOFT,
DevFmtUInt = ALC_UNSIGNED_INT_SOFT,
DevFmtFloat = ALC_FLOAT_SOFT,
DevFmtTypeDefault = DevFmtFloat
};
enum DevFmtChannels {
DevFmtMono = ALC_MONO_SOFT,
DevFmtStereo = ALC_STEREO_SOFT,
DevFmtQuad = ALC_QUAD_SOFT,
DevFmtX51 = ALC_5POINT1_SOFT,
DevFmtX61 = ALC_6POINT1_SOFT,
DevFmtX71 = ALC_7POINT1_SOFT,
/* Similar to 5.1, except using the side channels instead of back */
DevFmtX51Side = 0x80000000,
DevFmtChannelsDefault = DevFmtStereo
};
ALuint BytesFromDevFmt(enum DevFmtType type);
ALuint ChannelsFromDevFmt(enum DevFmtChannels chans);
static __inline ALuint FrameSizeFromDevFmt(enum DevFmtChannels chans,
enum DevFmtType type)
{
return ChannelsFromDevFmt(chans) * BytesFromDevFmt(type);
}
extern const struct EffectList {
const char *name;
int type;
const char *ename;
ALenum val;
} EffectList[];
enum DeviceType {
Playback,
Capture,
Loopback
};
/* Size for temporary storage of buffer data, in ALfloats. Larger values need
* more memory, while smaller values may need more iterations. The value needs
* to be a sensible size, however, as it constrains the max stepping value used
* for mixing, as well as the maximum number of samples per mixing iteration.
*
* The mixer requires being able to do two samplings per mixing loop. With the
* cubic resampler (which requires 3 padding samples), this limits a 2048
* buffer size to about 2044. This means that buffer_freq*source_pitch cannot
* exceed device_freq*2044 for a 32-bit buffer.
*/
#ifndef BUFFERSIZE
#define BUFFERSIZE 2048
#endif
struct ALCdevice_struct
{
ALboolean InUse;
ALboolean IsCaptureDevice;
volatile RefCount ref;
ALCboolean Connected;
enum DeviceType Type;
CRITICAL_SECTION Mutex;
ALuint Frequency;
ALuint UpdateFreq;
ALuint FrameSize;
ALuint Channels;
ALenum Format;
ALuint UpdateSize;
ALuint NumUpdates;
enum DevFmtChannels FmtChans;
enum DevFmtType FmtType;
ALCchar *szDeviceName;
ALCchar *DeviceName;
volatile ALCenum LastError;
// Maximum number of sources that can be created
ALuint MaxNoOfSources;
// Maximum number of slots that can be created
ALuint AuxiliaryEffectSlotMax;
// Context created on this device
ALCcontext *Context;
ALCuint NumMonoSources;
ALCuint NumStereoSources;
ALuint NumAuxSends;
// Map of Buffers for this device
UIntMap BufferMap;
// Map of Effects for this device
UIntMap EffectMap;
// Map of Filters for this device
UIntMap FilterMap;
/* HRTF filter tables */
const struct Hrtf *Hrtf;
// Stereo-to-binaural filter
struct bs2b *Bs2b;
ALCint Bs2bLevel;
// Device flags
ALuint Flags;
ALuint ChannelOffsets[MaxChannels];
enum Channel Speaker2Chan[MaxChannels];
ALfloat SpeakerAngle[MaxChannels];
ALuint NumChan;
/* Temp storage used for mixing. +1 for the predictive sample. */
ALIGN(16) ALfloat SampleData1[BUFFERSIZE+1];
ALIGN(16) ALfloat SampleData2[BUFFERSIZE+1];
// Dry path buffer mix
ALIGN(16) ALfloat DryBuffer[MaxChannels][BUFFERSIZE];
ALIGN(16) ALfloat ClickRemoval[MaxChannels];
ALIGN(16) ALfloat PendingClicks[MaxChannels];
/* Default effect slot */
struct ALeffectslot *DefaultSlot;
// Contexts created on this device
ALCcontext *volatile ContextList;
BackendFuncs *Funcs;
void *ExtraData; // For the backend's use
ALCdevice *volatile next;
};
#define ALCdevice_OpenPlayback(a,b) ((a)->Funcs->OpenPlayback((a), (b)))
#define ALCdevice_ClosePlayback(a) ((a)->Funcs->ClosePlayback((a)))
#define ALCdevice_OpenCapture(a,b,c,d,e) ((a)->Funcs->OpenCapture((a), (b), (c), (d), (e)))
#define ALCdevice_ResetPlayback(a) ((a)->Funcs->ResetPlayback((a)))
#define ALCdevice_StartPlayback(a) ((a)->Funcs->StartPlayback((a)))
#define ALCdevice_StopPlayback(a) ((a)->Funcs->StopPlayback((a)))
#define ALCdevice_OpenCapture(a,b) ((a)->Funcs->OpenCapture((a), (b)))
#define ALCdevice_CloseCapture(a) ((a)->Funcs->CloseCapture((a)))
#define ALCdevice_StartCapture(a) ((a)->Funcs->StartCapture((a)))
#define ALCdevice_StopCapture(a) ((a)->Funcs->StopCapture((a)))
#define ALCdevice_CaptureSamples(a,b,c) ((a)->Funcs->CaptureSamples((a), (b), (c)))
#define ALCdevice_AvailableSamples(a) ((a)->Funcs->AvailableSamples((a)))
#define ALCdevice_Lock(a) ((a)->Funcs->Lock((a)))
#define ALCdevice_Unlock(a) ((a)->Funcs->Unlock((a)))
#define ALCdevice_GetLatency(a) ((a)->Funcs->GetLatency((a)))
// Frequency was requested by the app or config file
#define DEVICE_FREQUENCY_REQUEST (1<<1)
// Channel configuration was requested by the config file
#define DEVICE_CHANNELS_REQUEST (1<<2)
// Sample type was requested by the config file
#define DEVICE_SAMPLE_TYPE_REQUEST (1<<3)
// Stereo sources cover 120-degree angles around +/-90
#define DEVICE_WIDE_STEREO (1<<16)
// Specifies if the device is currently running
#define DEVICE_RUNNING (1<<31)
/* Invalid channel offset */
#define INVALID_OFFSET (~0u)
#define LookupBuffer(m, k) ((struct ALbuffer*)LookupUIntMapKey(&(m)->BufferMap, (k)))
#define LookupEffect(m, k) ((struct ALeffect*)LookupUIntMapKey(&(m)->EffectMap, (k)))
#define LookupFilter(m, k) ((struct ALfilter*)LookupUIntMapKey(&(m)->FilterMap, (k)))
#define RemoveBuffer(m, k) ((struct ALbuffer*)RemoveUIntMapKey(&(m)->BufferMap, (k)))
#define RemoveEffect(m, k) ((struct ALeffect*)RemoveUIntMapKey(&(m)->EffectMap, (k)))
#define RemoveFilter(m, k) ((struct ALfilter*)RemoveUIntMapKey(&(m)->FilterMap, (k)))
struct ALCcontext_struct
{
ALlistener Listener;
volatile RefCount ref;
struct ALsource *Source;
ALuint SourceCount;
struct ALlistener *Listener;
ALenum LastError;
ALboolean InUse;
UIntMap SourceMap;
UIntMap EffectSlotMap;
ALuint Frequency;
ALenum LastError;
ALenum DistanceModel;
volatile ALenum UpdateSources;
ALfloat DopplerFactor;
ALfloat DopplerVelocity;
ALfloat flSpeedOfSound;
volatile enum DistanceModel DistanceModel;
volatile ALboolean SourceDistanceModel;
ALint lNumMonoSources;
ALint lNumStereoSources;
volatile ALfloat DopplerFactor;
volatile ALfloat DopplerVelocity;
volatile ALfloat SpeedOfSound;
volatile ALenum DeferUpdates;
struct ALsource **ActiveSources;
ALsizei ActiveSourceCount;
ALsizei MaxActiveSources;
struct ALeffectslot **ActiveEffectSlots;
ALsizei ActiveEffectSlotCount;
ALsizei MaxActiveEffectSlots;
ALCdevice *Device;
ALCchar ExtensionList[1024];
const ALCchar *ExtensionList;
struct bs2b *bs2b;
ALCcontext *next;
ALCcontext *volatile next;
};
ALCchar *AppendDeviceList(char *name);
ALCchar *AppendAllDeviceList(char *name);
ALCchar *AppendCaptureDeviceList(char *name);
#define LookupSource(m, k) ((struct ALsource*)LookupUIntMapKey(&(m)->SourceMap, (k)))
#define LookupEffectSlot(m, k) ((struct ALeffectslot*)LookupUIntMapKey(&(m)->EffectSlotMap, (k)))
#define RemoveSource(m, k) ((struct ALsource*)RemoveUIntMapKey(&(m)->SourceMap, (k)))
#define RemoveEffectSlot(m, k) ((struct ALeffectslot*)RemoveUIntMapKey(&(m)->EffectSlotMap, (k)))
ALCvoid SetALCError(ALenum errorCode);
ALCvoid SuspendContext(ALCcontext *context);
ALCvoid ProcessContext(ALCcontext *context);
ALCcontext *GetContextRef(void);
void ALCcontext_IncRef(ALCcontext *context);
void ALCcontext_DecRef(ALCcontext *context);
void AppendAllDevicesList(const ALCchar *name);
void AppendCaptureDeviceList(const ALCchar *name);
void ALCdevice_LockDefault(ALCdevice *device);
void ALCdevice_UnlockDefault(ALCdevice *device);
ALint64 ALCdevice_GetLatencyDefault(ALCdevice *device);
static __inline void LockContext(ALCcontext *context)
{ ALCdevice_Lock(context->Device); }
static __inline void UnlockContext(ALCcontext *context)
{ ALCdevice_Unlock(context->Device); }
void *al_malloc(size_t alignment, size_t size);
void *al_calloc(size_t alignment, size_t size);
void al_free(void *ptr);
typedef struct {
int state;
#ifdef HAVE_SSE
int sse_state;
#endif
} FPUCtl;
void SetMixerFPUMode(FPUCtl *ctl);
void RestoreFPUMode(const FPUCtl *ctl);
ALvoid *StartThread(ALuint (*func)(ALvoid*), ALvoid *ptr);
ALuint StopThread(ALvoid *thread);
@@ -206,9 +756,126 @@ void ReadRingBuffer(RingBuffer *ring, ALubyte *data, ALsizei len);
void ReadALConfig(void);
void FreeALConfig(void);
int ConfigValueExists(const char *blockName, const char *keyName);
const char *GetConfigValue(const char *blockName, const char *keyName, const char *def);
int GetConfigValueInt(const char *blockName, const char *keyName, int def);
float GetConfigValueFloat(const char *blockName, const char *keyName, float def);
int GetConfigValueBool(const char *blockName, const char *keyName, int def);
int ConfigValueStr(const char *blockName, const char *keyName, const char **ret);
int ConfigValueInt(const char *blockName, const char *keyName, int *ret);
int ConfigValueUInt(const char *blockName, const char *keyName, unsigned int *ret);
int ConfigValueFloat(const char *blockName, const char *keyName, float *ret);
void SetRTPriority(void);
void SetDefaultChannelOrder(ALCdevice *device);
void SetDefaultWFXChannelOrder(ALCdevice *device);
const ALCchar *DevFmtTypeString(enum DevFmtType type);
const ALCchar *DevFmtChannelsString(enum DevFmtChannels chans);
#define HRIR_BITS (7)
#define HRIR_LENGTH (1<<HRIR_BITS)
#define HRIR_MASK (HRIR_LENGTH-1)
#define HRTFDELAY_BITS (20)
#define HRTFDELAY_FRACONE (1<<HRTFDELAY_BITS)
#define HRTFDELAY_MASK (HRTFDELAY_FRACONE-1)
const struct Hrtf *GetHrtf(ALCdevice *device);
void FreeHrtfs(void);
ALuint GetHrtfIrSize (const struct Hrtf *Hrtf);
ALfloat CalcHrtfDelta(ALfloat oldGain, ALfloat newGain, const ALfloat olddir[3], const ALfloat newdir[3]);
void GetLerpedHrtfCoeffs(const struct Hrtf *Hrtf, ALfloat elevation, ALfloat azimuth, ALfloat gain, ALfloat (*coeffs)[2], ALuint *delays);
ALuint GetMovingHrtfCoeffs(const struct Hrtf *Hrtf, ALfloat elevation, ALfloat azimuth, ALfloat gain, ALfloat delta, ALint counter, ALfloat (*coeffs)[2], ALuint *delays, ALfloat (*coeffStep)[2], ALint *delayStep);
void al_print(const char *type, const char *func, const char *fmt, ...) PRINTF_STYLE(3,4);
#define AL_PRINT(T, ...) al_print((T), __FUNCTION__, __VA_ARGS__)
extern FILE *LogFile;
enum LogLevel {
NoLog,
LogError,
LogWarning,
LogTrace,
LogRef
};
extern enum LogLevel LogLevel;
#define TRACEREF(...) do { \
if(LogLevel >= LogRef) \
AL_PRINT("(--)", __VA_ARGS__); \
} while(0)
#define TRACE(...) do { \
if(LogLevel >= LogTrace) \
AL_PRINT("(II)", __VA_ARGS__); \
} while(0)
#define WARN(...) do { \
if(LogLevel >= LogWarning) \
AL_PRINT("(WW)", __VA_ARGS__); \
} while(0)
#define ERR(...) do { \
if(LogLevel >= LogError) \
AL_PRINT("(EE)", __VA_ARGS__); \
} while(0)
extern ALint RTPrioLevel;
extern ALuint CPUCapFlags;
enum {
CPU_CAP_SSE = 1<<0,
CPU_CAP_NEON = 1<<1,
};
void FillCPUCaps(ALuint capfilter);
/**
* Starts a try block. Must not be nested within another try block within the
* same function.
*/
#define al_try do { \
int _al_err=0; \
_al_try_label: \
if(_al_err == 0)
/**
* After a try or another catch block, runs the next block if the given value
* was thrown.
*/
#define al_catch(val) else if(_al_err == (val))
/**
* After a try or catch block, runs the next block for any value thrown and not
* caught.
*/
#define al_catchany() else
/** Marks the end of the final catch (or the try) block. */
#define al_endtry } while(0)
/**
* The given integer value is "thrown" so as to be caught by a catch block.
* Must be called in a try block within the same function. The value must not
* be 0.
*/
#define al_throw(e) do { \
_al_err = (e); \
assert(_al_err != 0); \
goto _al_try_label; \
} while(0)
/** Sets an AL error on the given context, before throwing the error code. */
#define al_throwerr(ctx, err) do { \
alSetError((ctx), (err)); \
al_throw((err)); \
} while(0)
/**
* Throws an AL_INVALID_VALUE error with the given ctx if the given condition
* is false.
*/
#define CHECK_VALUE(ctx, cond) do { \
if(!(cond)) \
al_throwerr((ctx), AL_INVALID_VALUE); \
} while(0)
#ifdef __cplusplus
}
+159 -40
View File
@@ -1,68 +1,187 @@
#ifndef _AL_SOURCE_H_
#define _AL_SOURCE_H_
#define AL_NUM_SOURCE_PARAMS 128
#define MAX_SENDS 4
#include "AL/al.h"
#include "alMain.h"
#include "alu.h"
#include "alFilter.h"
#ifdef __cplusplus
extern "C" {
#endif
#define SRC_HISTORY_BITS (6)
#define SRC_HISTORY_LENGTH (1<<SRC_HISTORY_BITS)
#define SRC_HISTORY_MASK (SRC_HISTORY_LENGTH-1)
extern enum Resampler DefaultResampler;
extern const ALsizei ResamplerPadding[ResamplerMax];
extern const ALsizei ResamplerPrePadding[ResamplerMax];
typedef struct ALbufferlistitem
{
ALuint buffer;
ALuint bufferstate;
ALuint flag;
struct ALbuffer *buffer;
struct ALbufferlistitem *next;
struct ALbufferlistitem *prev;
} ALbufferlistitem;
typedef struct HrtfState {
ALboolean Moving;
ALuint Counter;
ALIGN(16) ALfloat History[MaxChannels][SRC_HISTORY_LENGTH];
ALIGN(16) ALfloat Values[MaxChannels][HRIR_LENGTH][2];
ALuint Offset;
} HrtfState;
typedef struct HrtfParams {
ALfloat Gain;
ALfloat Dir[3];
ALIGN(16) ALfloat Coeffs[MaxChannels][HRIR_LENGTH][2];
ALIGN(16) ALfloat CoeffStep[HRIR_LENGTH][2];
ALuint Delay[MaxChannels][2];
ALint DelayStep[2];
ALuint IrSize;
} HrtfParams;
typedef struct DirectParams {
ALfloat (*OutBuffer)[BUFFERSIZE];
ALfloat *ClickRemoval;
ALfloat *PendingClicks;
struct {
HrtfParams Params;
HrtfState *State;
} Hrtf;
/* A mixing matrix. First subscript is the channel number of the input data
* (regardless of channel configuration) and the second is the channel
* target (eg. FrontLeft). Not used with HRTF. */
ALfloat Gains[MaxChannels][MaxChannels];
/* A low-pass filter, using 2 chained one-pole filters. */
FILTER iirFilter;
ALfloat history[MaxChannels*2];
} DirectParams;
typedef struct SendParams {
struct ALeffectslot *Slot;
/* Gain control, which applies to all input channels to a single (mono)
* output buffer. */
ALfloat Gain;
/* A low-pass filter, using 2 chained one-pole filters. */
FILTER iirFilter;
ALfloat history[MaxChannels*2];
} SendParams;
typedef struct ALsource
{
ALfloat flPitch;
ALfloat flGain;
ALfloat flOuterGain;
ALfloat flMinGain;
ALfloat flMaxGain;
ALfloat flInnerAngle;
ALfloat flOuterAngle;
ALfloat flRefDistance;
ALfloat flMaxDistance;
ALfloat flRollOffFactor;
ALfloat vPosition[3];
ALfloat vVelocity[3];
ALfloat vOrientation[3];
ALboolean bHeadRelative;
ALboolean bLooping;
/** Source properties. */
volatile ALfloat Pitch;
volatile ALfloat Gain;
volatile ALfloat OuterGain;
volatile ALfloat MinGain;
volatile ALfloat MaxGain;
volatile ALfloat InnerAngle;
volatile ALfloat OuterAngle;
volatile ALfloat RefDistance;
volatile ALfloat MaxDistance;
volatile ALfloat RollOffFactor;
volatile ALfloat Position[3];
volatile ALfloat Velocity[3];
volatile ALfloat Orientation[3];
volatile ALboolean HeadRelative;
volatile ALboolean Looping;
volatile enum DistanceModel DistanceModel;
volatile ALboolean DirectChannels;
ALuint ulBufferID;
volatile ALboolean DryGainHFAuto;
volatile ALboolean WetGainAuto;
volatile ALboolean WetGainHFAuto;
volatile ALfloat OuterGainHF;
ALboolean inuse;
ALboolean play;
ALenum state;
ALuint position;
ALuint position_fraction;
struct ALbufferlistitem *queue; // Linked list of buffers in queue
ALuint BuffersInQueue; // Number of buffers in queue
ALuint BuffersProcessed; // Number of buffers already processed (played)
volatile ALfloat AirAbsorptionFactor;
volatile ALfloat RoomRolloffFactor;
volatile ALfloat DopplerFactor;
ALuint TotalBufferDataSize; // Total amount of data contained in the buffers queued for this source
ALuint BuffersPlayed; // Number of buffers played on this loop
ALuint BufferPosition; // Read position in audio data of current buffer
enum Resampler Resampler;
// Index to itself
ALuint source;
/**
* Last user-specified offset, and the offset type (bytes, samples, or
* seconds).
*/
ALdouble Offset;
ALenum OffsetType;
ALint lBytesPlayed;
/** Source type (static, streaming, or undetermined) */
volatile ALint SourceType;
ALint lOffset;
ALint lOffsetType;
/** Source state (initial, playing, paused, or stopped) */
volatile ALenum state;
ALenum new_state;
// Source Type (Static, Streaming, or Undetermined)
ALint lSourceType;
/**
* Source offset in samples, relative to the currently playing buffer, NOT
* the whole queue, and the fractional (fixed-point) offset to the next
* sample.
*/
ALuint position;
ALuint position_fraction;
struct ALsource *next;
/** Source Buffer Queue info. */
ALbufferlistitem *queue;
ALuint BuffersInQueue;
ALuint BuffersPlayed;
/** Current buffer sample info. */
ALuint NumChannels;
ALuint SampleSize;
/** Direct filter and auxiliary send info. */
ALfloat DirectGain;
ALfloat DirectGainHF;
struct {
struct ALeffectslot *Slot;
ALfloat Gain;
ALfloat GainHF;
} Send[MAX_SENDS];
/** HRTF info. */
HrtfState Hrtf;
/** Current target parameters used for mixing. */
struct {
ResamplerFunc Resample;
DryMixerFunc DryMix;
WetMixerFunc WetMix;
ALint Step;
DirectParams Direct;
SendParams Send[MAX_SENDS];
} Params;
/** Source needs to update its mixing parameters. */
volatile ALenum NeedsUpdate;
/** Method to update mixing parameters. */
ALvoid (*Update)(struct ALsource *self, const ALCcontext *context);
/** Self ID */
ALuint id;
} ALsource;
#define ALsource_Update(s,a) ((s)->Update(s,a))
ALvoid SetSourceState(ALsource *Source, ALCcontext *Context, ALenum state);
ALboolean ApplyOffset(ALsource *Source);
ALvoid ReleaseALSources(ALCcontext *Context);
#ifdef __cplusplus
}
-14
View File
@@ -1,14 +0,0 @@
#ifndef _AL_STATE_H_
#define _AL_STATE_H_
#include "AL/al.h"
#ifdef __cplusplus
extern "C" {
#endif
#ifdef __cplusplus
}
#endif
#endif
+8 -30
View File
@@ -1,42 +1,20 @@
#ifndef _AL_THUNK_H_
#define _AL_THUNK_H_
#ifndef ALTHUNK_H
#define ALTHUNK_H
#include "config.h"
#include "AL/al.h"
#include "AL/alc.h"
#include "alMain.h"
#ifdef __cplusplus
extern "C" {
#endif
void alThunkInit(void);
void alThunkExit(void);
ALuint alThunkAddEntry(ALvoid * ptr);
void alThunkRemoveEntry(ALuint index);
ALvoid *alThunkLookupEntry(ALuint index);
#if (SIZEOF_VOIDP > SIZEOF_UINT)
#define ALTHUNK_INIT() alThunkInit()
#define ALTHUNK_EXIT() alThunkExit()
#define ALTHUNK_ADDENTRY(p) alThunkAddEntry(p)
#define ALTHUNK_REMOVEENTRY(i) alThunkRemoveEntry(i)
#define ALTHUNK_LOOKUPENTRY(i) alThunkLookupEntry(i)
#else
#define ALTHUNK_INIT()
#define ALTHUNK_EXIT()
#define ALTHUNK_ADDENTRY(p) ((ALuint)p)
#define ALTHUNK_REMOVEENTRY(i)
#define ALTHUNK_LOOKUPENTRY(i) ((ALvoid*)(i))
#endif // (SIZEOF_VOIDP > SIZEOF_INT)
void ThunkInit(void);
void ThunkExit(void);
ALenum NewThunkEntry(ALuint *index);
void FreeThunkEntry(ALuint index);
#ifdef __cplusplus
}
#endif
#endif //_AL_THUNK_H_
#endif //ALTHUNK_H
+112 -5
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@@ -1,16 +1,123 @@
#ifndef _ALU_H_
#define _ALU_H_
#include "AL/al.h"
#include "AL/alc.h"
#include "alMain.h"
#include <limits.h>
#include <math.h>
#ifdef HAVE_FLOAT_H
#include <float.h>
#endif
#ifdef HAVE_IEEEFP_H
#include <ieeefp.h>
#endif
#define F_PI (3.14159265358979323846f) /* pi */
#define F_PI_2 (1.57079632679489661923f) /* pi/2 */
#ifndef FLT_EPSILON
#define FLT_EPSILON (1.19209290e-07f)
#endif
#ifdef __cplusplus
extern "C" {
#endif
__inline ALuint aluBytesFromFormat(ALenum format);
__inline ALuint aluChannelsFromFormat(ALenum format);
ALvoid aluMixData(ALCcontext *context,ALvoid *buffer,ALsizei size,ALenum format);
struct ALsource;
struct ALbuffer;
struct DirectParams;
struct SendParams;
typedef void (*ResamplerFunc)(const ALfloat *src, ALuint frac, ALuint increment,
ALfloat *RESTRICT dst, ALuint dstlen);
typedef ALvoid (*DryMixerFunc)(const struct DirectParams *params,
const ALfloat *RESTRICT data, ALuint srcchan,
ALuint OutPos, ALuint SamplesToDo,
ALuint BufferSize);
typedef ALvoid (*WetMixerFunc)(const struct SendParams *params,
const ALfloat *RESTRICT data,
ALuint OutPos, ALuint SamplesToDo,
ALuint BufferSize);
#define SPEEDOFSOUNDMETRESPERSEC (343.3f)
#define AIRABSORBGAINHF (0.99426f) /* -0.05dB */
#define FRACTIONBITS (14)
#define FRACTIONONE (1<<FRACTIONBITS)
#define FRACTIONMASK (FRACTIONONE-1)
static __inline ALfloat minf(ALfloat a, ALfloat b)
{ return ((a > b) ? b : a); }
static __inline ALfloat maxf(ALfloat a, ALfloat b)
{ return ((a > b) ? a : b); }
static __inline ALfloat clampf(ALfloat val, ALfloat min, ALfloat max)
{ return minf(max, maxf(min, val)); }
static __inline ALuint minu(ALuint a, ALuint b)
{ return ((a > b) ? b : a); }
static __inline ALuint maxu(ALuint a, ALuint b)
{ return ((a > b) ? a : b); }
static __inline ALuint clampu(ALuint val, ALuint min, ALuint max)
{ return minu(max, maxu(min, val)); }
static __inline ALint mini(ALint a, ALint b)
{ return ((a > b) ? b : a); }
static __inline ALint maxi(ALint a, ALint b)
{ return ((a > b) ? a : b); }
static __inline ALint clampi(ALint val, ALint min, ALint max)
{ return mini(max, maxi(min, val)); }
static __inline ALint64 mini64(ALint64 a, ALint64 b)
{ return ((a > b) ? b : a); }
static __inline ALint64 maxi64(ALint64 a, ALint64 b)
{ return ((a > b) ? a : b); }
static __inline ALint64 clampi64(ALint64 val, ALint64 min, ALint64 max)
{ return mini64(max, maxi64(min, val)); }
static __inline ALuint64 minu64(ALuint64 a, ALuint64 b)
{ return ((a > b) ? b : a); }
static __inline ALuint64 maxu64(ALuint64 a, ALuint64 b)
{ return ((a > b) ? a : b); }
static __inline ALuint64 clampu64(ALuint64 val, ALuint64 min, ALuint64 max)
{ return minu64(max, maxu64(min, val)); }
static __inline ALfloat lerp(ALfloat val1, ALfloat val2, ALfloat mu)
{
return val1 + (val2-val1)*mu;
}
static __inline ALfloat cubic(ALfloat val0, ALfloat val1, ALfloat val2, ALfloat val3, ALfloat mu)
{
ALfloat mu2 = mu*mu;
ALfloat a0 = -0.5f*val0 + 1.5f*val1 + -1.5f*val2 + 0.5f*val3;
ALfloat a1 = val0 + -2.5f*val1 + 2.0f*val2 + -0.5f*val3;
ALfloat a2 = -0.5f*val0 + 0.5f*val2;
ALfloat a3 = val1;
return a0*mu*mu2 + a1*mu2 + a2*mu + a3;
}
ALvoid aluInitPanning(ALCdevice *Device);
ALvoid ComputeAngleGains(const ALCdevice *device, ALfloat angle, ALfloat hwidth, ALfloat ingain, ALfloat *gains);
ALvoid CalcSourceParams(struct ALsource *ALSource, const ALCcontext *ALContext);
ALvoid CalcNonAttnSourceParams(struct ALsource *ALSource, const ALCcontext *ALContext);
ALvoid MixSource(struct ALsource *Source, ALCdevice *Device, ALuint SamplesToDo);
ALvoid aluMixData(ALCdevice *device, ALvoid *buffer, ALsizei size);
/* Caller must lock the device. */
ALvoid aluHandleDisconnect(ALCdevice *device);
extern ALfloat ConeScale;
extern ALfloat ZScale;
#ifdef __cplusplus
}
+106 -109
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@@ -1,109 +1,106 @@
/*-
* Copyright (c) 2005 Boris Mikhaylov
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files (the
* "Software"), to deal in the Software without restriction, including
* without limitation the rights to use, copy, modify, merge, publish,
* distribute, sublicense, and/or sell copies of the Software, and to
* permit persons to whom the Software is furnished to do so, subject to
* the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
#ifndef BS2B_H
#define BS2B_H
/* Number of crossfeed levels */
#define BS2B_CLEVELS 3
/* Normal crossfeed levels */
#define BS2B_HIGH_CLEVEL 3
#define BS2B_MIDDLE_CLEVEL 2
#define BS2B_LOW_CLEVEL 1
/* Easy crossfeed levels */
#define BS2B_HIGH_ECLEVEL BS2B_HIGH_CLEVEL + BS2B_CLEVELS
#define BS2B_MIDDLE_ECLEVEL BS2B_MIDDLE_CLEVEL + BS2B_CLEVELS
#define BS2B_LOW_ECLEVEL BS2B_LOW_CLEVEL + BS2B_CLEVELS
/* Default crossfeed levels */
#define BS2B_DEFAULT_CLEVEL BS2B_HIGH_ECLEVEL
/* Default sample rate (Hz) */
#define BS2B_DEFAULT_SRATE 44100
#ifdef __cplusplus
extern "C" {
#endif /* __cplusplus */
struct bs2b {
int level; /* Crossfeed level */
int srate; /* Sample rate (Hz) */
/* Lowpass IIR filter coefficients */
double a0_lo;
double b1_lo;
/* Highboost IIR filter coefficients */
double a0_hi;
double a1_hi;
double b1_hi;
/* Global gain against overloading */
double gain;
/* Buffer of last filtered sample.
* [0] - first channel, [1] - second channel
*/
struct t_last_sample {
double asis[2];
double lo[2];
double hi[2];
} last_sample;
};
/* Clear buffers and set new coefficients with new crossfeed level value.
* level - crossfeed level of *LEVEL values.
*/
void bs2b_set_level(struct bs2b *bs2b, int level);
/* Return current crossfeed level value */
int bs2b_get_level(struct bs2b *bs2b);
/* Clear buffers and set new coefficients with new sample rate value.
* srate - sample rate by Hz.
*/
void bs2b_set_srate(struct bs2b *bs2b, int srate);
/* Return current sample rate value */
int bs2b_get_srate(struct bs2b *bs2b);
/* Clear buffer */
void bs2b_clear(struct bs2b *bs2b);
/* Return 1 if buffer is clear */
int bs2b_is_clear(struct bs2b *bs2b);
/* Crossfeeds one stereo sample that are pointed by sample.
* [0] - first channel, [1] - second channel.
* Returns crossfided samle by sample pointer.
*/
/* sample poits to floats */
void bs2b_cross_feed(struct bs2b *bs2b, float *sample);
#ifdef __cplusplus
} /* extern "C" */
#endif /* __cplusplus */
#endif /* BS2B_H */
/*-
* Copyright (c) 2005 Boris Mikhaylov
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files (the
* "Software"), to deal in the Software without restriction, including
* without limitation the rights to use, copy, modify, merge, publish,
* distribute, sublicense, and/or sell copies of the Software, and to
* permit persons to whom the Software is furnished to do so, subject to
* the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
#ifndef BS2B_H
#define BS2B_H
/* Number of crossfeed levels */
#define BS2B_CLEVELS 3
/* Normal crossfeed levels */
#define BS2B_HIGH_CLEVEL 3
#define BS2B_MIDDLE_CLEVEL 2
#define BS2B_LOW_CLEVEL 1
/* Easy crossfeed levels */
#define BS2B_HIGH_ECLEVEL BS2B_HIGH_CLEVEL + BS2B_CLEVELS
#define BS2B_MIDDLE_ECLEVEL BS2B_MIDDLE_CLEVEL + BS2B_CLEVELS
#define BS2B_LOW_ECLEVEL BS2B_LOW_CLEVEL + BS2B_CLEVELS
/* Default crossfeed levels */
#define BS2B_DEFAULT_CLEVEL BS2B_HIGH_ECLEVEL
/* Default sample rate (Hz) */
#define BS2B_DEFAULT_SRATE 44100
#ifdef __cplusplus
extern "C" {
#endif /* __cplusplus */
struct bs2b {
int level; /* Crossfeed level */
int srate; /* Sample rate (Hz) */
/* Lowpass IIR filter coefficients */
double a0_lo;
double b1_lo;
/* Highboost IIR filter coefficients */
double a0_hi;
double a1_hi;
double b1_hi;
/* Global gain against overloading */
float gain;
/* Buffer of last filtered sample.
* [0] - first channel, [1] - second channel
*/
struct t_last_sample {
double asis[2];
double lo[2];
double hi[2];
} last_sample;
};
/* Clear buffers and set new coefficients with new crossfeed level value.
* level - crossfeed level of *LEVEL values.
*/
void bs2b_set_level(struct bs2b *bs2b, int level);
/* Return current crossfeed level value */
int bs2b_get_level(struct bs2b *bs2b);
/* Clear buffers and set new coefficients with new sample rate value.
* srate - sample rate by Hz.
*/
void bs2b_set_srate(struct bs2b *bs2b, int srate);
/* Return current sample rate value */
int bs2b_get_srate(struct bs2b *bs2b);
/* Clear buffer */
void bs2b_clear(struct bs2b *bs2b);
/* Crossfeeds one stereo sample that are pointed by sample.
* [0] - first channel, [1] - second channel.
* Returns crossfided samle by sample pointer.
*/
/* sample poits to floats */
void bs2b_cross_feed(struct bs2b *bs2b, float *sample);
#ifdef __cplusplus
} /* extern "C" */
#endif /* __cplusplus */
#endif /* BS2B_H */
-63
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@@ -1,63 +0,0 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2007 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "alMain.h"
#include "alBuffer.h"
#include "alThunk.h"
CRITICAL_SECTION _alMutex;
#ifdef _WIN32
BOOL APIENTRY DllMain(HANDLE hModule,DWORD ul_reason_for_call,LPVOID lpReserved)
{
(void)lpReserved;
// Perform actions based on the reason for calling.
switch(ul_reason_for_call)
{
case DLL_PROCESS_ATTACH:
DisableThreadLibraryCalls(hModule);
break;
case DLL_PROCESS_DETACH:
ReleaseALBuffers();
FreeALConfig();
ALTHUNK_EXIT();
DeleteCriticalSection(&_alMutex);
break;
}
return TRUE;
}
#else
#ifdef HAVE_GCC_DESTRUCTOR
static void my_deinit() __attribute__((destructor));
static void my_deinit()
{
static ALenum once = AL_FALSE;
if(once) return;
once = AL_TRUE;
ReleaseALBuffers();
FreeALConfig();
ALTHUNK_EXIT();
DeleteCriticalSection(&_alMutex);
}
#endif
#endif
+609
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@@ -0,0 +1,609 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2007 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdlib.h>
#include <math.h>
#include "AL/al.h"
#include "AL/alc.h"
#include "alMain.h"
#include "alAuxEffectSlot.h"
#include "alThunk.h"
#include "alError.h"
#include "alSource.h"
static ALenum AddEffectSlotArray(ALCcontext *Context, ALsizei count, const ALuint *slots);
static ALvoid RemoveEffectSlotArray(ALCcontext *Context, ALeffectslot *slot);
AL_API ALvoid AL_APIENTRY alGenAuxiliaryEffectSlots(ALsizei n, ALuint *effectslots)
{
ALCcontext *Context;
ALsizei cur = 0;
Context = GetContextRef();
if(!Context) return;
al_try
{
ALenum err;
CHECK_VALUE(Context, n >= 0);
for(cur = 0;cur < n;cur++)
{
ALeffectslot *slot = al_calloc(16, sizeof(ALeffectslot));
err = AL_OUT_OF_MEMORY;
if(!slot || (err=InitEffectSlot(slot)) != AL_NO_ERROR)
{
al_free(slot);
al_throwerr(Context, err);
break;
}
err = NewThunkEntry(&slot->id);
if(err == AL_NO_ERROR)
err = InsertUIntMapEntry(&Context->EffectSlotMap, slot->id, slot);
if(err != AL_NO_ERROR)
{
FreeThunkEntry(slot->id);
ALeffectState_Destroy(slot->EffectState);
al_free(slot);
al_throwerr(Context, err);
}
effectslots[cur] = slot->id;
}
err = AddEffectSlotArray(Context, n, effectslots);
if(err != AL_NO_ERROR)
al_throwerr(Context, err);
}
al_catchany()
{
if(cur > 0)
alDeleteAuxiliaryEffectSlots(cur, effectslots);
}
al_endtry;
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alDeleteAuxiliaryEffectSlots(ALsizei n, const ALuint *effectslots)
{
ALCcontext *Context;
ALeffectslot *slot;
ALsizei i;
Context = GetContextRef();
if(!Context) return;
al_try
{
CHECK_VALUE(Context, n >= 0);
for(i = 0;i < n;i++)
{
if((slot=LookupEffectSlot(Context, effectslots[i])) == NULL)
al_throwerr(Context, AL_INVALID_NAME);
if(slot->ref != 0)
al_throwerr(Context, AL_INVALID_OPERATION);
}
// All effectslots are valid
for(i = 0;i < n;i++)
{
if((slot=RemoveEffectSlot(Context, effectslots[i])) == NULL)
continue;
FreeThunkEntry(slot->id);
RemoveEffectSlotArray(Context, slot);
ALeffectState_Destroy(slot->EffectState);
memset(slot, 0, sizeof(*slot));
al_free(slot);
}
}
al_endtry;
ALCcontext_DecRef(Context);
}
AL_API ALboolean AL_APIENTRY alIsAuxiliaryEffectSlot(ALuint effectslot)
{
ALCcontext *Context;
ALboolean result;
Context = GetContextRef();
if(!Context) return AL_FALSE;
result = (LookupEffectSlot(Context, effectslot) ? AL_TRUE : AL_FALSE);
ALCcontext_DecRef(Context);
return result;
}
AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSloti(ALuint effectslot, ALenum param, ALint value)
{
ALCcontext *Context;
ALeffectslot *Slot;
ALeffect *effect = NULL;
ALenum err;
Context = GetContextRef();
if(!Context) return;
al_try
{
ALCdevice *device = Context->Device;
if((Slot=LookupEffectSlot(Context, effectslot)) == NULL)
al_throwerr(Context, AL_INVALID_NAME);
switch(param)
{
case AL_EFFECTSLOT_EFFECT:
CHECK_VALUE(Context, value == 0 || (effect=LookupEffect(device, value)) != NULL);
err = InitializeEffect(device, Slot, effect);
if(err != AL_NO_ERROR)
al_throwerr(Context, err);
Context->UpdateSources = AL_TRUE;
break;
case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO:
CHECK_VALUE(Context, value == AL_TRUE || value == AL_FALSE);
Slot->AuxSendAuto = value;
Context->UpdateSources = AL_TRUE;
break;
default:
al_throwerr(Context, AL_INVALID_ENUM);
}
}
al_endtry;
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotiv(ALuint effectslot, ALenum param, const ALint *values)
{
ALCcontext *Context;
switch(param)
{
case AL_EFFECTSLOT_EFFECT:
case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO:
alAuxiliaryEffectSloti(effectslot, param, values[0]);
return;
}
Context = GetContextRef();
if(!Context) return;
al_try
{
if(LookupEffectSlot(Context, effectslot) == NULL)
al_throwerr(Context, AL_INVALID_NAME);
switch(param)
{
default:
al_throwerr(Context, AL_INVALID_ENUM);
}
}
al_endtry;
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotf(ALuint effectslot, ALenum param, ALfloat value)
{
ALCcontext *Context;
ALeffectslot *Slot;
Context = GetContextRef();
if(!Context) return;
al_try
{
if((Slot=LookupEffectSlot(Context, effectslot)) == NULL)
al_throwerr(Context, AL_INVALID_NAME);
switch(param)
{
case AL_EFFECTSLOT_GAIN:
CHECK_VALUE(Context, value >= 0.0f && value <= 1.0f);
Slot->Gain = value;
Slot->NeedsUpdate = AL_TRUE;
break;
default:
al_throwerr(Context, AL_INVALID_ENUM);
}
}
al_endtry;
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotfv(ALuint effectslot, ALenum param, const ALfloat *values)
{
ALCcontext *Context;
switch(param)
{
case AL_EFFECTSLOT_GAIN:
alAuxiliaryEffectSlotf(effectslot, param, values[0]);
return;
}
Context = GetContextRef();
if(!Context) return;
al_try
{
if(LookupEffectSlot(Context, effectslot) == NULL)
al_throwerr(Context, AL_INVALID_NAME);
switch(param)
{
default:
al_throwerr(Context, AL_INVALID_ENUM);
}
}
al_endtry;
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSloti(ALuint effectslot, ALenum param, ALint *value)
{
ALCcontext *Context;
ALeffectslot *Slot;
Context = GetContextRef();
if(!Context) return;
al_try
{
if((Slot=LookupEffectSlot(Context, effectslot)) == NULL)
al_throwerr(Context, AL_INVALID_NAME);
switch(param)
{
case AL_EFFECTSLOT_EFFECT:
*value = Slot->effect.id;
break;
case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO:
*value = Slot->AuxSendAuto;
break;
default:
al_throwerr(Context, AL_INVALID_ENUM);
}
}
al_endtry;
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotiv(ALuint effectslot, ALenum param, ALint *values)
{
ALCcontext *Context;
switch(param)
{
case AL_EFFECTSLOT_EFFECT:
case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO:
alGetAuxiliaryEffectSloti(effectslot, param, values);
return;
}
Context = GetContextRef();
if(!Context) return;
al_try
{
if(LookupEffectSlot(Context, effectslot) == NULL)
al_throwerr(Context, AL_INVALID_NAME);
switch(param)
{
default:
al_throwerr(Context, AL_INVALID_ENUM);
}
}
al_endtry;
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotf(ALuint effectslot, ALenum param, ALfloat *value)
{
ALCcontext *Context;
ALeffectslot *Slot;
Context = GetContextRef();
if(!Context) return;
al_try
{
if((Slot=LookupEffectSlot(Context, effectslot)) == NULL)
al_throwerr(Context, AL_INVALID_NAME);
switch(param)
{
case AL_EFFECTSLOT_GAIN:
*value = Slot->Gain;
break;
default:
al_throwerr(Context, AL_INVALID_ENUM);
}
}
al_endtry;
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotfv(ALuint effectslot, ALenum param, ALfloat *values)
{
ALCcontext *Context;
switch(param)
{
case AL_EFFECTSLOT_GAIN:
alGetAuxiliaryEffectSlotf(effectslot, param, values);
return;
}
Context = GetContextRef();
if(!Context) return;
al_try
{
if(LookupEffectSlot(Context, effectslot) == NULL)
al_throwerr(Context, AL_INVALID_NAME);
switch(param)
{
default:
al_throwerr(Context, AL_INVALID_ENUM);
}
}
al_endtry;
ALCcontext_DecRef(Context);
}
static ALvoid NoneDestroy(ALeffectState *State)
{ free(State); }
static ALboolean NoneDeviceUpdate(ALeffectState *State, ALCdevice *Device)
{
return AL_TRUE;
(void)State;
(void)Device;
}
static ALvoid NoneUpdate(ALeffectState *State, ALCdevice *Device, const ALeffectslot *Slot)
{
(void)State;
(void)Device;
(void)Slot;
}
static ALvoid NoneProcess(ALeffectState *State, ALuint SamplesToDo, const ALfloat *RESTRICT SamplesIn, ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE])
{
(void)State;
(void)SamplesToDo;
(void)SamplesIn;
(void)SamplesOut;
}
ALeffectState *NoneCreate(void)
{
ALeffectState *state;
state = calloc(1, sizeof(*state));
if(!state)
return NULL;
state->Destroy = NoneDestroy;
state->DeviceUpdate = NoneDeviceUpdate;
state->Update = NoneUpdate;
state->Process = NoneProcess;
return state;
}
static ALvoid RemoveEffectSlotArray(ALCcontext *Context, ALeffectslot *slot)
{
ALeffectslot **slotlist, **slotlistend;
LockContext(Context);
slotlist = Context->ActiveEffectSlots;
slotlistend = slotlist + Context->ActiveEffectSlotCount;
while(slotlist != slotlistend)
{
if(*slotlist == slot)
{
*slotlist = *(--slotlistend);
Context->ActiveEffectSlotCount--;
break;
}
slotlist++;
}
UnlockContext(Context);
}
static ALenum AddEffectSlotArray(ALCcontext *Context, ALsizei count, const ALuint *slots)
{
ALsizei i;
LockContext(Context);
if(count > Context->MaxActiveEffectSlots-Context->ActiveEffectSlotCount)
{
ALsizei newcount;
void *temp = NULL;
newcount = Context->MaxActiveEffectSlots ? (Context->MaxActiveEffectSlots<<1) : 1;
if(newcount > Context->MaxActiveEffectSlots)
temp = realloc(Context->ActiveEffectSlots,
newcount * sizeof(*Context->ActiveEffectSlots));
if(!temp)
{
UnlockContext(Context);
return AL_OUT_OF_MEMORY;
}
Context->ActiveEffectSlots = temp;
Context->MaxActiveEffectSlots = newcount;
}
for(i = 0;i < count;i++)
{
ALeffectslot *slot = LookupEffectSlot(Context, slots[i]);
assert(slot != NULL);
Context->ActiveEffectSlots[Context->ActiveEffectSlotCount++] = slot;
}
UnlockContext(Context);
return AL_NO_ERROR;
}
ALenum InitializeEffect(ALCdevice *Device, ALeffectslot *EffectSlot, ALeffect *effect)
{
ALenum newtype = (effect ? effect->type : AL_EFFECT_NULL);
ALeffectState *State = NULL;
ALenum err = AL_NO_ERROR;
ALCdevice_Lock(Device);
if(newtype == AL_EFFECT_NULL && EffectSlot->effect.type != AL_EFFECT_NULL)
{
State = NoneCreate();
if(!State) err = AL_OUT_OF_MEMORY;
}
else if(newtype == AL_EFFECT_EAXREVERB || newtype == AL_EFFECT_REVERB)
{
if(EffectSlot->effect.type != AL_EFFECT_EAXREVERB && EffectSlot->effect.type != AL_EFFECT_REVERB)
{
State = ReverbCreate();
if(!State) err = AL_OUT_OF_MEMORY;
}
}
else if(newtype == AL_EFFECT_ECHO && EffectSlot->effect.type != AL_EFFECT_ECHO)
{
State = EchoCreate();
if(!State) err = AL_OUT_OF_MEMORY;
}
else if(newtype == AL_EFFECT_RING_MODULATOR && EffectSlot->effect.type != AL_EFFECT_RING_MODULATOR)
{
State = ModulatorCreate();
if(!State) err = AL_OUT_OF_MEMORY;
}
else if(newtype == AL_EFFECT_DEDICATED_DIALOGUE || newtype == AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT)
{
if(EffectSlot->effect.type != AL_EFFECT_DEDICATED_DIALOGUE && EffectSlot->effect.type != AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT)
{
State = DedicatedCreate();
if(!State) err = AL_OUT_OF_MEMORY;
}
}
if(err != AL_NO_ERROR)
{
ALCdevice_Unlock(Device);
return err;
}
if(State)
{
FPUCtl oldMode;
SetMixerFPUMode(&oldMode);
if(ALeffectState_DeviceUpdate(State, Device) == AL_FALSE)
{
RestoreFPUMode(&oldMode);
ALCdevice_Unlock(Device);
ALeffectState_Destroy(State);
return AL_OUT_OF_MEMORY;
}
State = ExchangePtr((XchgPtr*)&EffectSlot->EffectState, State);
if(!effect)
memset(&EffectSlot->effect, 0, sizeof(EffectSlot->effect));
else
memcpy(&EffectSlot->effect, effect, sizeof(*effect));
/* FIXME: This should be done asynchronously, but since the EffectState
* object was changed, it needs an update before its Process method can
* be called. */
EffectSlot->NeedsUpdate = AL_FALSE;
ALeffectState_Update(EffectSlot->EffectState, Device, EffectSlot);
ALCdevice_Unlock(Device);
RestoreFPUMode(&oldMode);
ALeffectState_Destroy(State);
State = NULL;
}
else
{
if(!effect)
memset(&EffectSlot->effect, 0, sizeof(EffectSlot->effect));
else
memcpy(&EffectSlot->effect, effect, sizeof(*effect));
ALCdevice_Unlock(Device);
EffectSlot->NeedsUpdate = AL_TRUE;
}
return AL_NO_ERROR;
}
ALenum InitEffectSlot(ALeffectslot *slot)
{
ALint i, c;
if(!(slot->EffectState=NoneCreate()))
return AL_OUT_OF_MEMORY;
slot->Gain = 1.0;
slot->AuxSendAuto = AL_TRUE;
slot->NeedsUpdate = AL_FALSE;
for(c = 0;c < 1;c++)
{
for(i = 0;i < BUFFERSIZE;i++)
slot->WetBuffer[c][i] = 0.0f;
slot->ClickRemoval[c] = 0.0f;
slot->PendingClicks[c] = 0.0f;
}
slot->ref = 0;
return AL_NO_ERROR;
}
ALvoid ReleaseALAuxiliaryEffectSlots(ALCcontext *Context)
{
ALsizei pos;
for(pos = 0;pos < Context->EffectSlotMap.size;pos++)
{
ALeffectslot *temp = Context->EffectSlotMap.array[pos].value;
Context->EffectSlotMap.array[pos].value = NULL;
ALeffectState_Destroy(temp->EffectState);
FreeThunkEntry(temp->id);
memset(temp, 0, sizeof(ALeffectslot));
al_free(temp);
}
}
+1986 -841
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+1516
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File diff suppressed because it is too large Load Diff
+39 -25
View File
@@ -18,40 +18,54 @@
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <signal.h>
#include "alMain.h"
#include "AL/alc.h"
#include "alError.h"
ALAPI ALenum ALAPIENTRY alGetError(ALvoid)
ALboolean TrapALError = AL_FALSE;
ALvoid alSetError(ALCcontext *Context, ALenum errorCode)
{
ALCcontext *Context;
ALenum errorCode;
Context = alcGetCurrentContext();
SuspendContext(Context);
if (Context)
{
errorCode = Context->LastError;
Context->LastError = AL_NO_ERROR;
}
else
errorCode = AL_INVALID_OPERATION;
ProcessContext(Context);
return errorCode;
if(TrapALError)
{
#ifdef _WIN32
/* DebugBreak will cause an exception if there is no debugger */
if(IsDebuggerPresent())
DebugBreak();
#elif defined(SIGTRAP)
raise(SIGTRAP);
#endif
}
CompExchangeInt(&Context->LastError, AL_NO_ERROR, errorCode);
}
ALvoid alSetError(ALenum errorCode)
AL_API ALenum AL_APIENTRY alGetError(void)
{
ALCcontext *Context;
ALCcontext *Context;
ALenum errorCode;
Context=alcGetCurrentContext();
SuspendContext(Context);
Context = GetContextRef();
if(!Context)
{
if(TrapALError)
{
#ifdef _WIN32
if(IsDebuggerPresent())
DebugBreak();
#elif defined(SIGTRAP)
raise(SIGTRAP);
#endif
}
return AL_INVALID_OPERATION;
}
if (Context && Context->LastError == AL_NO_ERROR)
Context->LastError = errorCode;
errorCode = ExchangeInt(&Context->LastError, AL_NO_ERROR);
ProcessContext(Context);
ALCcontext_DecRef(Context);
return errorCode;
}
+50 -239
View File
@@ -18,272 +18,83 @@
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdlib.h>
#include <string.h>
#include <ctype.h>
#include "alExtension.h"
#include "alError.h"
#include "alMain.h"
#include "alFilter.h"
#include "alEffect.h"
#include "alAuxEffectSlot.h"
#include "alSource.h"
#include "alBuffer.h"
#include "AL/al.h"
#include "AL/alc.h"
static ALfunction function[]= {
{ "alEnable", (ALvoid *) alEnable },
{ "alDisable", (ALvoid *) alDisable },
{ "alIsEnabled", (ALvoid *) alIsEnabled },
{ "alGetString", (ALvoid *) alGetString },
{ "alGetBooleanv", (ALvoid *) alGetBooleanv },
{ "alGetIntegerv", (ALvoid *) alGetIntegerv },
{ "alGetFloatv", (ALvoid *) alGetFloatv },
{ "alGetDoublev", (ALvoid *) alGetDoublev },
{ "alGetBoolean", (ALvoid *) alGetBoolean },
{ "alGetInteger", (ALvoid *) alGetInteger },
{ "alGetFloat", (ALvoid *) alGetFloat },
{ "alGetDouble", (ALvoid *) alGetDouble },
{ "alGetError", (ALvoid *) alGetError },
{ "alIsExtensionPresent", (ALvoid *) alIsExtensionPresent },
{ "alGetProcAddress", (ALvoid *) alGetProcAddress },
{ "alGetEnumValue", (ALvoid *) alGetEnumValue },
{ "alListenerf", (ALvoid *) alListenerf },
{ "alListener3f", (ALvoid *) alListener3f },
{ "alListenerfv", (ALvoid *) alListenerfv },
{ "alListeneri", (ALvoid *) alListeneri },
{ "alListener3i", (ALvoid *) alListener3i },
{ "alListeneriv", (ALvoid *) alListeneriv },
{ "alGetListenerf", (ALvoid *) alGetListenerf },
{ "alGetListener3f", (ALvoid *) alGetListener3f },
{ "alGetListenerfv", (ALvoid *) alGetListenerfv },
{ "alGetListeneri", (ALvoid *) alGetListeneri },
{ "alGetListener3i", (ALvoid *) alGetListener3i },
{ "alGetListeneriv", (ALvoid *) alGetListeneriv },
{ "alGenSources", (ALvoid *) alGenSources },
{ "alDeleteSources", (ALvoid *) alDeleteSources },
{ "alIsSource", (ALvoid *) alIsSource },
{ "alSourcef", (ALvoid *) alSourcef },
{ "alSource3f", (ALvoid *) alSource3f },
{ "alSourcefv", (ALvoid *) alSourcefv },
{ "alSourcei", (ALvoid *) alSourcei },
{ "alSource3i", (ALvoid *) alSource3i },
{ "alSourceiv", (ALvoid *) alSourceiv },
{ "alGetSourcef", (ALvoid *) alGetSourcef },
{ "alGetSource3f", (ALvoid *) alGetSource3f },
{ "alGetSourcefv", (ALvoid *) alGetSourcefv },
{ "alGetSourcei", (ALvoid *) alGetSourcei },
{ "alGetSource3i", (ALvoid *) alGetSource3i },
{ "alGetSourceiv", (ALvoid *) alGetSourceiv },
{ "alSourcePlayv", (ALvoid *) alSourcePlayv },
{ "alSourceStopv", (ALvoid *) alSourceStopv },
{ "alSourceRewindv", (ALvoid *) alSourceRewindv },
{ "alSourcePausev", (ALvoid *) alSourcePausev },
{ "alSourcePlay", (ALvoid *) alSourcePlay },
{ "alSourceStop", (ALvoid *) alSourceStop },
{ "alSourceRewind", (ALvoid *) alSourceRewind },
{ "alSourcePause", (ALvoid *) alSourcePause },
{ "alSourceQueueBuffers", (ALvoid *) alSourceQueueBuffers },
{ "alSourceUnqueueBuffers", (ALvoid *) alSourceUnqueueBuffers },
{ "alGenBuffers", (ALvoid *) alGenBuffers },
{ "alDeleteBuffers", (ALvoid *) alDeleteBuffers },
{ "alIsBuffer", (ALvoid *) alIsBuffer },
{ "alBufferData", (ALvoid *) alBufferData },
{ "alBufferf", (ALvoid *) alBufferf },
{ "alBuffer3f", (ALvoid *) alBuffer3f },
{ "alBufferfv", (ALvoid *) alBufferfv },
{ "alBufferi", (ALvoid *) alBufferi },
{ "alBuffer3i", (ALvoid *) alBuffer3i },
{ "alBufferiv", (ALvoid *) alBufferiv },
{ "alGetBufferf", (ALvoid *) alGetBufferf },
{ "alGetBuffer3f", (ALvoid *) alGetBuffer3f },
{ "alGetBufferfv", (ALvoid *) alGetBufferfv },
{ "alGetBufferi", (ALvoid *) alGetBufferi },
{ "alGetBuffer3i", (ALvoid *) alGetBuffer3i },
{ "alGetBufferiv", (ALvoid *) alGetBufferiv },
{ "alDopplerFactor", (ALvoid *) alDopplerFactor },
{ "alDopplerVelocity", (ALvoid *) alDopplerVelocity },
{ "alSpeedOfSound", (ALvoid *) alSpeedOfSound },
{ "alDistanceModel", (ALvoid *) alDistanceModel },
{ NULL, (ALvoid *) NULL } };
static ALenums enumeration[]={
// Types
{ (ALchar *)"AL_INVALID", AL_INVALID },
{ (ALchar *)"AL_NONE", AL_NONE },
{ (ALchar *)"AL_FALSE", AL_FALSE },
{ (ALchar *)"AL_TRUE", AL_TRUE },
// Source and Listener Properties
{ (ALchar *)"AL_SOURCE_RELATIVE", AL_SOURCE_RELATIVE },
{ (ALchar *)"AL_CONE_INNER_ANGLE", AL_CONE_INNER_ANGLE },
{ (ALchar *)"AL_CONE_OUTER_ANGLE", AL_CONE_OUTER_ANGLE },
{ (ALchar *)"AL_PITCH", AL_PITCH },
{ (ALchar *)"AL_POSITION", AL_POSITION },
{ (ALchar *)"AL_DIRECTION", AL_DIRECTION },
{ (ALchar *)"AL_VELOCITY", AL_VELOCITY },
{ (ALchar *)"AL_LOOPING", AL_LOOPING },
{ (ALchar *)"AL_BUFFER", AL_BUFFER },
{ (ALchar *)"AL_GAIN", AL_GAIN },
{ (ALchar *)"AL_MIN_GAIN", AL_MIN_GAIN },
{ (ALchar *)"AL_MAX_GAIN", AL_MAX_GAIN },
{ (ALchar *)"AL_ORIENTATION", AL_ORIENTATION },
{ (ALchar *)"AL_REFERENCE_DISTANCE", AL_REFERENCE_DISTANCE },
{ (ALchar *)"AL_ROLLOFF_FACTOR", AL_ROLLOFF_FACTOR },
{ (ALchar *)"AL_CONE_OUTER_GAIN", AL_CONE_OUTER_GAIN },
{ (ALchar *)"AL_MAX_DISTANCE", AL_MAX_DISTANCE },
{ (ALchar *)"AL_SEC_OFFSET", AL_SEC_OFFSET },
{ (ALchar *)"AL_SAMPLE_OFFSET", AL_SAMPLE_OFFSET },
{ (ALchar *)"AL_BYTE_OFFSET", AL_BYTE_OFFSET },
{ (ALchar *)"AL_SOURCE_TYPE", AL_SOURCE_TYPE },
{ (ALchar *)"AL_STATIC", AL_STATIC },
{ (ALchar *)"AL_STREAMING", AL_STREAMING },
{ (ALchar *)"AL_UNDETERMINED", AL_UNDETERMINED },
// Source State information
{ (ALchar *)"AL_SOURCE_STATE", AL_SOURCE_STATE },
{ (ALchar *)"AL_INITIAL", AL_INITIAL },
{ (ALchar *)"AL_PLAYING", AL_PLAYING },
{ (ALchar *)"AL_PAUSED", AL_PAUSED },
{ (ALchar *)"AL_STOPPED", AL_STOPPED },
// Queue information
{ (ALchar *)"AL_BUFFERS_QUEUED", AL_BUFFERS_QUEUED },
{ (ALchar *)"AL_BUFFERS_PROCESSED", AL_BUFFERS_PROCESSED },
// Buffer Formats
{ (ALchar *)"AL_FORMAT_MONO8", AL_FORMAT_MONO8 },
{ (ALchar *)"AL_FORMAT_MONO16", AL_FORMAT_MONO16 },
{ (ALchar *)"AL_FORMAT_MONO_FLOAT32", AL_FORMAT_MONO_FLOAT32 },
{ (ALchar *)"AL_FORMAT_STEREO8", AL_FORMAT_STEREO8 },
{ (ALchar *)"AL_FORMAT_STEREO16", AL_FORMAT_STEREO16 },
{ (ALchar *)"AL_FORMAT_STEREO_FLOAT32", AL_FORMAT_STEREO_FLOAT32 },
{ (ALchar *)"AL_FORMAT_MONO_IMA4", AL_FORMAT_MONO_IMA4 },
{ (ALchar *)"AL_FORMAT_STEREO_IMA4", AL_FORMAT_STEREO_IMA4 },
{ (ALchar *)"AL_FORMAT_QUAD8", AL_FORMAT_QUAD8 },
{ (ALchar *)"AL_FORMAT_QUAD16", AL_FORMAT_QUAD16 },
{ (ALchar *)"AL_FORMAT_QUAD32", AL_FORMAT_QUAD32 },
{ (ALchar *)"AL_FORMAT_51CHN8", AL_FORMAT_51CHN8 },
{ (ALchar *)"AL_FORMAT_51CHN16", AL_FORMAT_51CHN16 },
{ (ALchar *)"AL_FORMAT_51CHN32", AL_FORMAT_51CHN32 },
{ (ALchar *)"AL_FORMAT_61CHN8", AL_FORMAT_61CHN8 },
{ (ALchar *)"AL_FORMAT_61CHN16", AL_FORMAT_61CHN16 },
{ (ALchar *)"AL_FORMAT_61CHN32", AL_FORMAT_61CHN32 },
{ (ALchar *)"AL_FORMAT_71CHN8", AL_FORMAT_71CHN8 },
{ (ALchar *)"AL_FORMAT_71CHN16", AL_FORMAT_71CHN16 },
{ (ALchar *)"AL_FORMAT_71CHN32", AL_FORMAT_71CHN32 },
{ (ALchar *)"AL_FORMAT_REAR8", AL_FORMAT_REAR8 },
{ (ALchar *)"AL_FORMAT_REAR16", AL_FORMAT_REAR16 },
{ (ALchar *)"AL_FORMAT_REAR32", AL_FORMAT_REAR32 },
// Buffer attributes
{ (ALchar *)"AL_FREQUENCY", AL_FREQUENCY },
{ (ALchar *)"AL_BITS", AL_BITS },
{ (ALchar *)"AL_CHANNELS", AL_CHANNELS },
{ (ALchar *)"AL_SIZE", AL_SIZE },
// Buffer States (not supported yet)
{ (ALchar *)"AL_UNUSED", AL_UNUSED },
{ (ALchar *)"AL_PENDING", AL_PENDING },
{ (ALchar *)"AL_PROCESSED", AL_PROCESSED },
// AL Error Messages
{ (ALchar *)"AL_NO_ERROR", AL_NO_ERROR },
{ (ALchar *)"AL_INVALID_NAME", AL_INVALID_NAME },
{ (ALchar *)"AL_INVALID_ENUM", AL_INVALID_ENUM },
{ (ALchar *)"AL_INVALID_VALUE", AL_INVALID_VALUE },
{ (ALchar *)"AL_INVALID_OPERATION", AL_INVALID_OPERATION },
{ (ALchar *)"AL_OUT_OF_MEMORY", AL_OUT_OF_MEMORY },
// Context strings
{ (ALchar *)"AL_VENDOR", AL_VENDOR },
{ (ALchar *)"AL_VERSION", AL_VERSION },
{ (ALchar *)"AL_RENDERER", AL_RENDERER },
{ (ALchar *)"AL_EXTENSIONS", AL_EXTENSIONS },
// Global states
{ (ALchar *)"AL_DOPPLER_FACTOR", AL_DOPPLER_FACTOR },
{ (ALchar *)"AL_DOPPLER_VELOCITY", AL_DOPPLER_VELOCITY },
{ (ALchar *)"AL_DISTANCE_MODEL", AL_DISTANCE_MODEL },
{ (ALchar *)"AL_SPEED_OF_SOUND", AL_SPEED_OF_SOUND },
// Distance Models
{ (ALchar *)"AL_INVERSE_DISTANCE", AL_INVERSE_DISTANCE },
{ (ALchar *)"AL_INVERSE_DISTANCE_CLAMPED", AL_INVERSE_DISTANCE_CLAMPED },
{ (ALchar *)"AL_LINEAR_DISTANCE", AL_LINEAR_DISTANCE },
{ (ALchar *)"AL_LINEAR_DISTANCE_CLAMPED", AL_LINEAR_DISTANCE_CLAMPED },
{ (ALchar *)"AL_EXPONENT_DISTANCE", AL_EXPONENT_DISTANCE },
{ (ALchar *)"AL_EXPONENT_DISTANCE_CLAMPED", AL_EXPONENT_DISTANCE_CLAMPED },
// Default
{ (ALchar *)NULL, (ALenum ) 0 } };
const struct EffectList EffectList[] = {
{ "eaxreverb", EAXREVERB, "AL_EFFECT_EAXREVERB", AL_EFFECT_EAXREVERB },
{ "reverb", REVERB, "AL_EFFECT_REVERB", AL_EFFECT_REVERB },
{ "echo", ECHO, "AL_EFFECT_ECHO", AL_EFFECT_ECHO },
{ "modulator", MODULATOR, "AL_EFFECT_RING_MODULATOR", AL_EFFECT_RING_MODULATOR },
{ "dedicated", DEDICATED, "AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT", AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT },
{ "dedicated", DEDICATED, "AL_EFFECT_DEDICATED_DIALOGUE", AL_EFFECT_DEDICATED_DIALOGUE },
{ NULL, 0, NULL, (ALenum)0 }
};
ALAPI ALboolean ALAPIENTRY alIsExtensionPresent(const ALchar *extName)
AL_API ALboolean AL_APIENTRY alIsExtensionPresent(const ALchar *extName)
{
ALboolean bIsSupported = AL_FALSE;
ALCcontext *pContext;
char *ptr, *ext;
ALboolean ret = AL_FALSE;
ALCcontext *Context;
const char *ptr;
size_t len;
if (!extName)
Context = GetContextRef();
if(!Context) return AL_FALSE;
al_try
{
alSetError(AL_INVALID_VALUE);
return AL_FALSE;
}
CHECK_VALUE(Context, extName);
pContext = alcGetCurrentContext();
if(!pContext)
{
alSetError(AL_INVALID_OPERATION);
return AL_FALSE;
}
ext = strdup(extName);
ptr = ext;
do {
*ptr = (char)toupper(*ptr);
} while(*(ptr++));
SuspendContext(pContext);
ptr = pContext->ExtensionList;
while((ptr=strstr(ptr, ext)) != NULL)
{
if(ptr == pContext->ExtensionList || ptr[-1] == ' ')
len = strlen(extName);
ptr = Context->ExtensionList;
while(ptr && *ptr)
{
char e = ptr[strlen(ext)];
if(e == ' ' || e == 0)
if(strncasecmp(ptr, extName, len) == 0 &&
(ptr[len] == '\0' || isspace(ptr[len])))
{
bIsSupported = AL_TRUE;
ret = AL_TRUE;
break;
}
if((ptr=strchr(ptr, ' ')) != NULL)
{
do {
++ptr;
} while(isspace(*ptr));
}
}
ptr++;
}
al_endtry;
ProcessContext(pContext);
free(ext);
return bIsSupported;
ALCcontext_DecRef(Context);
return ret;
}
ALAPI ALvoid * ALAPIENTRY alGetProcAddress(const ALchar *funcName)
AL_API ALvoid* AL_APIENTRY alGetProcAddress(const ALchar *funcName)
{
ALsizei i = 0;
while(function[i].funcName &&
strcmp((char*)function[i].funcName, (char*)funcName) != 0)
i++;
return function[i].address;
if(!funcName)
return NULL;
return alcGetProcAddress(NULL, funcName);
}
/* NOTE: This function must be able to run without a context! */
ALAPI ALenum ALAPIENTRY alGetEnumValue(const ALchar *enumName)
AL_API ALenum AL_APIENTRY alGetEnumValue(const ALchar *enumName)
{
ALsizei i = 0;
while(enumeration[i].enumName &&
strcmp(enumeration[i].enumName, enumName) != 0)
i++;
return enumeration[i].value;
if(!enumName)
return (ALenum)0;
return alcGetEnumValue(NULL, enumName);
}
+468
View File
@@ -0,0 +1,468 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2007 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdlib.h>
#include "alMain.h"
#include "alu.h"
#include "alFilter.h"
#include "alThunk.h"
#include "alError.h"
static void InitFilterParams(ALfilter *filter, ALenum type);
AL_API ALvoid AL_APIENTRY alGenFilters(ALsizei n, ALuint *filters)
{
ALCcontext *Context;
ALsizei cur = 0;
Context = GetContextRef();
if(!Context) return;
al_try
{
ALCdevice *device = Context->Device;
ALenum err;
CHECK_VALUE(Context, n >= 0);
for(cur = 0;cur < n;cur++)
{
ALfilter *filter = calloc(1, sizeof(ALfilter));
if(!filter)
al_throwerr(Context, AL_OUT_OF_MEMORY);
InitFilterParams(filter, AL_FILTER_NULL);
err = NewThunkEntry(&filter->id);
if(err == AL_NO_ERROR)
err = InsertUIntMapEntry(&device->FilterMap, filter->id, filter);
if(err != AL_NO_ERROR)
{
FreeThunkEntry(filter->id);
memset(filter, 0, sizeof(ALfilter));
free(filter);
al_throwerr(Context, err);
}
filters[cur] = filter->id;
}
}
al_catchany()
{
if(cur > 0)
alDeleteFilters(cur, filters);
}
al_endtry;
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alDeleteFilters(ALsizei n, const ALuint *filters)
{
ALCcontext *Context;
ALfilter *Filter;
ALsizei i;
Context = GetContextRef();
if(!Context) return;
al_try
{
ALCdevice *device = Context->Device;
CHECK_VALUE(Context, n >= 0);
for(i = 0;i < n;i++)
{
if(filters[i] && LookupFilter(device, filters[i]) == NULL)
al_throwerr(Context, AL_INVALID_NAME);
}
for(i = 0;i < n;i++)
{
if((Filter=RemoveFilter(device, filters[i])) == NULL)
continue;
FreeThunkEntry(Filter->id);
memset(Filter, 0, sizeof(*Filter));
free(Filter);
}
}
al_endtry;
ALCcontext_DecRef(Context);
}
AL_API ALboolean AL_APIENTRY alIsFilter(ALuint filter)
{
ALCcontext *Context;
ALboolean result;
Context = GetContextRef();
if(!Context) return AL_FALSE;
result = ((!filter || LookupFilter(Context->Device, filter)) ?
AL_TRUE : AL_FALSE);
ALCcontext_DecRef(Context);
return result;
}
AL_API ALvoid AL_APIENTRY alFilteri(ALuint filter, ALenum param, ALint value)
{
ALCcontext *Context;
ALCdevice *Device;
ALfilter *ALFilter;
Context = GetContextRef();
if(!Context) return;
Device = Context->Device;
if((ALFilter=LookupFilter(Device, filter)) == NULL)
alSetError(Context, AL_INVALID_NAME);
else
{
if(param == AL_FILTER_TYPE)
{
if(value == AL_FILTER_NULL || value == AL_FILTER_LOWPASS)
InitFilterParams(ALFilter, value);
else
alSetError(Context, AL_INVALID_VALUE);
}
else
{
/* Call the appropriate handler */
ALfilter_SetParami(ALFilter, Context, param, value);
}
}
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alFilteriv(ALuint filter, ALenum param, const ALint *values)
{
ALCcontext *Context;
ALCdevice *Device;
ALfilter *ALFilter;
switch(param)
{
case AL_FILTER_TYPE:
alFilteri(filter, param, values[0]);
return;
}
Context = GetContextRef();
if(!Context) return;
Device = Context->Device;
if((ALFilter=LookupFilter(Device, filter)) == NULL)
alSetError(Context, AL_INVALID_NAME);
else
{
/* Call the appropriate handler */
ALfilter_SetParamiv(ALFilter, Context, param, values);
}
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alFilterf(ALuint filter, ALenum param, ALfloat value)
{
ALCcontext *Context;
ALCdevice *Device;
ALfilter *ALFilter;
Context = GetContextRef();
if(!Context) return;
Device = Context->Device;
if((ALFilter=LookupFilter(Device, filter)) == NULL)
alSetError(Context, AL_INVALID_NAME);
else
{
/* Call the appropriate handler */
ALfilter_SetParamf(ALFilter, Context, param, value);
}
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alFilterfv(ALuint filter, ALenum param, const ALfloat *values)
{
ALCcontext *Context;
ALCdevice *Device;
ALfilter *ALFilter;
Context = GetContextRef();
if(!Context) return;
Device = Context->Device;
if((ALFilter=LookupFilter(Device, filter)) == NULL)
alSetError(Context, AL_INVALID_NAME);
else
{
/* Call the appropriate handler */
ALfilter_SetParamfv(ALFilter, Context, param, values);
}
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alGetFilteri(ALuint filter, ALenum param, ALint *value)
{
ALCcontext *Context;
ALCdevice *Device;
ALfilter *ALFilter;
Context = GetContextRef();
if(!Context) return;
Device = Context->Device;
if((ALFilter=LookupFilter(Device, filter)) == NULL)
alSetError(Context, AL_INVALID_NAME);
else
{
if(param == AL_FILTER_TYPE)
*value = ALFilter->type;
else
{
/* Call the appropriate handler */
ALfilter_GetParami(ALFilter, Context, param, value);
}
}
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alGetFilteriv(ALuint filter, ALenum param, ALint *values)
{
ALCcontext *Context;
ALCdevice *Device;
ALfilter *ALFilter;
switch(param)
{
case AL_FILTER_TYPE:
alGetFilteri(filter, param, values);
return;
}
Context = GetContextRef();
if(!Context) return;
Device = Context->Device;
if((ALFilter=LookupFilter(Device, filter)) == NULL)
alSetError(Context, AL_INVALID_NAME);
else
{
/* Call the appropriate handler */
ALfilter_GetParamiv(ALFilter, Context, param, values);
}
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alGetFilterf(ALuint filter, ALenum param, ALfloat *value)
{
ALCcontext *Context;
ALCdevice *Device;
ALfilter *ALFilter;
Context = GetContextRef();
if(!Context) return;
Device = Context->Device;
if((ALFilter=LookupFilter(Device, filter)) == NULL)
alSetError(Context, AL_INVALID_NAME);
else
{
/* Call the appropriate handler */
ALfilter_GetParamf(ALFilter, Context, param, value);
}
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alGetFilterfv(ALuint filter, ALenum param, ALfloat *values)
{
ALCcontext *Context;
ALCdevice *Device;
ALfilter *ALFilter;
Context = GetContextRef();
if(!Context) return;
Device = Context->Device;
if((ALFilter=LookupFilter(Device, filter)) == NULL)
alSetError(Context, AL_INVALID_NAME);
else
{
/* Call the appropriate handler */
ALfilter_GetParamfv(ALFilter, Context, param, values);
}
ALCcontext_DecRef(Context);
}
ALfloat lpCoeffCalc(ALfloat g, ALfloat cw)
{
ALfloat a = 0.0f;
if(g < 0.9999f) /* 1-epsilon */
{
/* Be careful with gains < 0.001, as that causes the coefficient head
* towards 1, which will flatten the signal */
g = maxf(g, 0.001f);
a = (1 - g*cw - sqrtf(2*g*(1-cw) - g*g*(1 - cw*cw))) /
(1 - g);
}
return a;
}
static void lp_SetParami(ALfilter *filter, ALCcontext *context, ALenum param, ALint val)
{ (void)filter;(void)param;(void)val; alSetError(context, AL_INVALID_ENUM); }
static void lp_SetParamiv(ALfilter *filter, ALCcontext *context, ALenum param, const ALint *vals)
{ (void)filter;(void)param;(void)vals; alSetError(context, AL_INVALID_ENUM); }
static void lp_SetParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val)
{
switch(param)
{
case AL_LOWPASS_GAIN:
if(val >= AL_LOWPASS_MIN_GAIN && val <= AL_LOWPASS_MAX_GAIN)
filter->Gain = val;
else
alSetError(context, AL_INVALID_VALUE);
break;
case AL_LOWPASS_GAINHF:
if(val >= AL_LOWPASS_MIN_GAINHF && val <= AL_LOWPASS_MAX_GAINHF)
filter->GainHF = val;
else
alSetError(context, AL_INVALID_VALUE);
break;
default:
alSetError(context, AL_INVALID_ENUM);
break;
}
}
static void lp_SetParamfv(ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals)
{
lp_SetParamf(filter, context, param, vals[0]);
}
static void lp_GetParami(ALfilter *filter, ALCcontext *context, ALenum param, ALint *val)
{ (void)filter;(void)param;(void)val; alSetError(context, AL_INVALID_ENUM); }
static void lp_GetParamiv(ALfilter *filter, ALCcontext *context, ALenum param, ALint *vals)
{ (void)filter;(void)param;(void)vals; alSetError(context, AL_INVALID_ENUM); }
static void lp_GetParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *val)
{
switch(param)
{
case AL_LOWPASS_GAIN:
*val = filter->Gain;
break;
case AL_LOWPASS_GAINHF:
*val = filter->GainHF;
break;
default:
alSetError(context, AL_INVALID_ENUM);
break;
}
}
static void lp_GetParamfv(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals)
{
lp_GetParamf(filter, context, param, vals);
}
static void null_SetParami(ALfilter *filter, ALCcontext *context, ALenum param, ALint val)
{ (void)filter;(void)param;(void)val; alSetError(context, AL_INVALID_ENUM); }
static void null_SetParamiv(ALfilter *filter, ALCcontext *context, ALenum param, const ALint *vals)
{ (void)filter;(void)param;(void)vals; alSetError(context, AL_INVALID_ENUM); }
static void null_SetParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val)
{ (void)filter;(void)param;(void)val; alSetError(context, AL_INVALID_ENUM); }
static void null_SetParamfv(ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals)
{ (void)filter;(void)param;(void)vals; alSetError(context, AL_INVALID_ENUM); }
static void null_GetParami(ALfilter *filter, ALCcontext *context, ALenum param, ALint *val)
{ (void)filter;(void)param;(void)val; alSetError(context, AL_INVALID_ENUM); }
static void null_GetParamiv(ALfilter *filter, ALCcontext *context, ALenum param, ALint *vals)
{ (void)filter;(void)param;(void)vals; alSetError(context, AL_INVALID_ENUM); }
static void null_GetParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *val)
{ (void)filter;(void)param;(void)val; alSetError(context, AL_INVALID_ENUM); }
static void null_GetParamfv(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals)
{ (void)filter;(void)param;(void)vals; alSetError(context, AL_INVALID_ENUM); }
ALvoid ReleaseALFilters(ALCdevice *device)
{
ALsizei i;
for(i = 0;i < device->FilterMap.size;i++)
{
ALfilter *temp = device->FilterMap.array[i].value;
device->FilterMap.array[i].value = NULL;
// Release filter structure
FreeThunkEntry(temp->id);
memset(temp, 0, sizeof(ALfilter));
free(temp);
}
}
static void InitFilterParams(ALfilter *filter, ALenum type)
{
if(type == AL_FILTER_LOWPASS)
{
filter->Gain = AL_LOWPASS_DEFAULT_GAIN;
filter->GainHF = AL_LOWPASS_DEFAULT_GAINHF;
filter->SetParami = lp_SetParami;
filter->SetParamiv = lp_SetParamiv;
filter->SetParamf = lp_SetParamf;
filter->SetParamfv = lp_SetParamfv;
filter->GetParami = lp_GetParami;
filter->GetParamiv = lp_GetParamiv;
filter->GetParamf = lp_GetParamf;
filter->GetParamfv = lp_GetParamfv;
}
else
{
filter->SetParami = null_SetParami;
filter->SetParamiv = null_SetParamiv;
filter->SetParamf = null_SetParamf;
filter->SetParamfv = null_SetParamfv;
filter->GetParami = null_GetParami;
filter->GetParamiv = null_GetParamiv;
filter->GetParamf = null_GetParamf;
filter->GetParamfv = null_GetParamfv;
}
filter->type = type;
}
+381 -422
View File
@@ -18,496 +18,455 @@
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include "alMain.h"
#include "AL/alc.h"
#include "alError.h"
#include "alListener.h"
#include "alSource.h"
ALAPI ALvoid ALAPIENTRY alListenerf(ALenum eParam, ALfloat flValue)
AL_API ALvoid AL_APIENTRY alListenerf(ALenum param, ALfloat value)
{
ALCcontext *pContext;
ALCcontext *Context;
pContext = alcGetCurrentContext();
if (pContext)
Context = GetContextRef();
if(!Context) return;
al_try
{
SuspendContext(pContext);
switch (eParam)
switch(param)
{
case AL_GAIN:
CHECK_VALUE(Context, value >= 0.0f && isfinite(value));
Context->Listener->Gain = value;
Context->UpdateSources = AL_TRUE;
break;
case AL_METERS_PER_UNIT:
CHECK_VALUE(Context, value >= 0.0f && isfinite(value));
Context->Listener->MetersPerUnit = value;
Context->UpdateSources = AL_TRUE;
break;
default:
al_throwerr(Context, AL_INVALID_ENUM);
}
}
al_endtry;
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alListener3f(ALenum param, ALfloat value1, ALfloat value2, ALfloat value3)
{
ALCcontext *Context;
Context = GetContextRef();
if(!Context) return;
al_try
{
switch(param)
{
case AL_POSITION:
CHECK_VALUE(Context, isfinite(value1) && isfinite(value2) && isfinite(value3));
LockContext(Context);
Context->Listener->Position[0] = value1;
Context->Listener->Position[1] = value2;
Context->Listener->Position[2] = value3;
Context->UpdateSources = AL_TRUE;
UnlockContext(Context);
break;
case AL_VELOCITY:
CHECK_VALUE(Context, isfinite(value1) && isfinite(value2) && isfinite(value3));
LockContext(Context);
Context->Listener->Velocity[0] = value1;
Context->Listener->Velocity[1] = value2;
Context->Listener->Velocity[2] = value3;
Context->UpdateSources = AL_TRUE;
UnlockContext(Context);
break;
default:
al_throwerr(Context, AL_INVALID_ENUM);
}
}
al_endtry;
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alListenerfv(ALenum param, const ALfloat *values)
{
ALCcontext *Context;
if(values)
{
switch(param)
{
case AL_GAIN:
case AL_METERS_PER_UNIT:
alListenerf(param, values[0]);
return;
case AL_POSITION:
case AL_VELOCITY:
alListener3f(param, values[0], values[1], values[2]);
return;
}
}
Context = GetContextRef();
if(!Context) return;
al_try
{
CHECK_VALUE(Context, values);
switch(param)
{
case AL_ORIENTATION:
CHECK_VALUE(Context, isfinite(values[0]) && isfinite(values[1]) &&
isfinite(values[2]) && isfinite(values[3]) &&
isfinite(values[4]) && isfinite(values[5]));
LockContext(Context);
/* AT then UP */
Context->Listener->Forward[0] = values[0];
Context->Listener->Forward[1] = values[1];
Context->Listener->Forward[2] = values[2];
Context->Listener->Up[0] = values[3];
Context->Listener->Up[1] = values[4];
Context->Listener->Up[2] = values[5];
Context->UpdateSources = AL_TRUE;
UnlockContext(Context);
break;
default:
al_throwerr(Context, AL_INVALID_ENUM);
}
}
al_endtry;
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alListeneri(ALenum param, ALint value)
{
ALCcontext *Context;
(void)value;
Context = GetContextRef();
if(!Context) return;
al_try
{
switch(param)
{
default:
al_throwerr(Context, AL_INVALID_ENUM);
}
}
al_endtry;
ALCcontext_DecRef(Context);
}
AL_API void AL_APIENTRY alListener3i(ALenum param, ALint value1, ALint value2, ALint value3)
{
ALCcontext *Context;
switch(param)
{
case AL_POSITION:
case AL_VELOCITY:
alListener3f(param, (ALfloat)value1, (ALfloat)value2, (ALfloat)value3);
return;
}
Context = GetContextRef();
if(!Context) return;
al_try
{
switch(param)
{
default:
al_throwerr(Context, AL_INVALID_ENUM);
}
}
al_endtry;
ALCcontext_DecRef(Context);
}
AL_API void AL_APIENTRY alListeneriv(ALenum param, const ALint *values)
{
ALCcontext *Context;
if(values)
{
ALfloat fvals[6];
switch(param)
{
case AL_POSITION:
case AL_VELOCITY:
alListener3f(param, (ALfloat)values[0], (ALfloat)values[1], (ALfloat)values[2]);
return;
case AL_ORIENTATION:
fvals[0] = (ALfloat)values[0];
fvals[1] = (ALfloat)values[1];
fvals[2] = (ALfloat)values[2];
fvals[3] = (ALfloat)values[3];
fvals[4] = (ALfloat)values[4];
fvals[5] = (ALfloat)values[5];
alListenerfv(param, fvals);
return;
}
}
Context = GetContextRef();
if(!Context) return;
al_try
{
CHECK_VALUE(Context, values);
switch(param)
{
default:
al_throwerr(Context, AL_INVALID_ENUM);
}
}
al_endtry;
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alGetListenerf(ALenum param, ALfloat *value)
{
ALCcontext *Context;
Context = GetContextRef();
if(!Context) return;
al_try
{
CHECK_VALUE(Context, value);
switch(param)
{
case AL_GAIN:
*value = Context->Listener->Gain;
break;
case AL_METERS_PER_UNIT:
*value = Context->Listener->MetersPerUnit;
break;
default:
al_throwerr(Context, AL_INVALID_ENUM);
}
}
al_endtry;
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alGetListener3f(ALenum param, ALfloat *value1, ALfloat *value2, ALfloat *value3)
{
ALCcontext *Context;
Context = GetContextRef();
if(!Context) return;
al_try
{
CHECK_VALUE(Context, value1 && value2 && value3);
switch(param)
{
case AL_POSITION:
LockContext(Context);
*value1 = Context->Listener->Position[0];
*value2 = Context->Listener->Position[1];
*value3 = Context->Listener->Position[2];
UnlockContext(Context);
break;
case AL_VELOCITY:
LockContext(Context);
*value1 = Context->Listener->Velocity[0];
*value2 = Context->Listener->Velocity[1];
*value3 = Context->Listener->Velocity[2];
UnlockContext(Context);
break;
default:
al_throwerr(Context, AL_INVALID_ENUM);
}
}
al_endtry;
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alGetListenerfv(ALenum param, ALfloat *values)
{
ALCcontext *Context;
switch(param)
{
case AL_GAIN:
if (flValue >= 0.0f)
pContext->Listener.Gain = flValue;
else
alSetError(AL_INVALID_VALUE);
break;
case AL_METERS_PER_UNIT:
alGetListenerf(param, values);
return;
default:
alSetError(AL_INVALID_ENUM);
break;
}
ProcessContext(pContext);
}
else
alSetError(AL_INVALID_OPERATION);
return;
}
ALAPI ALvoid ALAPIENTRY alListener3f(ALenum eParam, ALfloat flValue1, ALfloat flValue2, ALfloat flValue3)
{
ALCcontext *pContext;
pContext = alcGetCurrentContext();
if (pContext)
{
SuspendContext(pContext);
switch(eParam)
{
case AL_POSITION:
pContext->Listener.Position[0] = flValue1;
pContext->Listener.Position[1] = flValue2;
pContext->Listener.Position[2] = flValue3;
break;
case AL_VELOCITY:
pContext->Listener.Velocity[0] = flValue1;
pContext->Listener.Velocity[1] = flValue2;
pContext->Listener.Velocity[2] = flValue3;
break;
default:
alSetError(AL_INVALID_ENUM);
break;
}
ProcessContext(pContext);
}
else
alSetError(AL_INVALID_OPERATION);
return;
}
ALAPI ALvoid ALAPIENTRY alListenerfv(ALenum eParam, const ALfloat *pflValues)
{
ALCcontext *pContext;
pContext = alcGetCurrentContext();
if (pContext)
{
SuspendContext(pContext);
if (pflValues)
{
switch (eParam)
{
case AL_GAIN:
if (pflValues[0] >= 0.0f)
pContext->Listener.Gain = pflValues[0];
else
alSetError(AL_INVALID_VALUE);
break;
case AL_POSITION:
pContext->Listener.Position[0] = pflValues[0];
pContext->Listener.Position[1] = pflValues[1];
pContext->Listener.Position[2] = pflValues[2];
break;
case AL_VELOCITY:
pContext->Listener.Velocity[0] = pflValues[0];
pContext->Listener.Velocity[1] = pflValues[1];
pContext->Listener.Velocity[2] = pflValues[2];
break;
case AL_ORIENTATION:
// AT then UP
pContext->Listener.Forward[0] = pflValues[0];
pContext->Listener.Forward[1] = pflValues[1];
pContext->Listener.Forward[2] = pflValues[2];
pContext->Listener.Up[0] = pflValues[3];
pContext->Listener.Up[1] = pflValues[4];
pContext->Listener.Up[2] = pflValues[5];
break;
default:
alSetError(AL_INVALID_ENUM);
break;
}
}
else
alSetError(AL_INVALID_VALUE);
ProcessContext(pContext);
}
else
alSetError(AL_INVALID_OPERATION);
return;
}
ALAPI ALvoid ALAPIENTRY alListeneri(ALenum eParam, ALint lValue)
{
ALCcontext *pContext;
(void)lValue;
pContext = alcGetCurrentContext();
if (pContext)
{
SuspendContext(pContext);
switch (eParam)
{
default:
alSetError(AL_INVALID_ENUM);
break;
}
ProcessContext(pContext);
}
else
alSetError(AL_INVALID_OPERATION);
return;
}
ALAPI void ALAPIENTRY alListener3i(ALenum eParam, ALint lValue1, ALint lValue2, ALint lValue3)
{
ALCcontext *pContext;
pContext = alcGetCurrentContext();
if (pContext)
{
SuspendContext(pContext);
switch(eParam)
{
case AL_POSITION:
case AL_VELOCITY:
alListener3f(eParam, (ALfloat)lValue1, (ALfloat)lValue2, (ALfloat)lValue3);
break;
default:
alSetError(AL_INVALID_ENUM);
break;
}
ProcessContext(pContext);
alGetListener3f(param, values+0, values+1, values+2);
return;
}
else
alSetError(AL_INVALID_OPERATION);
return;
}
Context = GetContextRef();
if(!Context) return;
ALAPI void ALAPIENTRY alListeneriv( ALenum eParam, const ALint* plValues )
{
ALCcontext *pContext;
ALfloat flValues[6];
pContext = alcGetCurrentContext();
if (pContext)
al_try
{
SuspendContext(pContext);
if (plValues)
CHECK_VALUE(Context, values);
switch(param)
{
switch (eParam)
{
case AL_POSITION:
case AL_VELOCITY:
flValues[0] = (ALfloat)plValues[0];
flValues[1] = (ALfloat)plValues[1];
flValues[2] = (ALfloat)plValues[2];
alListenerfv(eParam, flValues);
break;
case AL_ORIENTATION:
flValues[0] = (ALfloat)plValues[0];
flValues[1] = (ALfloat)plValues[1];
flValues[2] = (ALfloat)plValues[2];
flValues[3] = (ALfloat)plValues[3];
flValues[4] = (ALfloat)plValues[4];
flValues[5] = (ALfloat)plValues[5];
alListenerfv(eParam, flValues);
break;
default:
alSetError(AL_INVALID_ENUM);
break;
}
}
else
alSetError(AL_INVALID_VALUE);
ProcessContext(pContext);
}
else
alSetError(AL_INVALID_OPERATION);
return;
}
ALAPI ALvoid ALAPIENTRY alGetListenerf(ALenum eParam, ALfloat *pflValue)
{
ALCcontext *pContext;
pContext = alcGetCurrentContext();
if (pContext)
{
SuspendContext(pContext);
if (pflValue)
{
switch (eParam)
{
case AL_GAIN:
*pflValue = pContext->Listener.Gain;
break;
default:
alSetError(AL_INVALID_ENUM);
break;
}
}
else
alSetError(AL_INVALID_VALUE);
ProcessContext(pContext);
}
else
alSetError(AL_INVALID_OPERATION);
return;
}
ALAPI ALvoid ALAPIENTRY alGetListener3f(ALenum eParam, ALfloat *pflValue1, ALfloat *pflValue2, ALfloat *pflValue3)
{
ALCcontext *pContext;
pContext = alcGetCurrentContext();
if (pContext)
{
SuspendContext(pContext);
if ((pflValue1) && (pflValue2) && (pflValue3))
{
switch (eParam)
{
case AL_POSITION:
*pflValue1 = pContext->Listener.Position[0];
*pflValue2 = pContext->Listener.Position[1];
*pflValue3 = pContext->Listener.Position[2];
break;
case AL_VELOCITY:
*pflValue1 = pContext->Listener.Velocity[0];
*pflValue2 = pContext->Listener.Velocity[1];
*pflValue3 = pContext->Listener.Velocity[2];
break;
default:
alSetError(AL_INVALID_ENUM);
break;
}
}
else
alSetError(AL_INVALID_VALUE);
ProcessContext(pContext);
}
else
alSetError(AL_INVALID_OPERATION);
return;
}
ALAPI ALvoid ALAPIENTRY alGetListenerfv(ALenum eParam, ALfloat *pflValues)
{
ALCcontext *pContext;
pContext = alcGetCurrentContext();
if (pContext)
{
SuspendContext(pContext);
if (pflValues)
{
switch (eParam)
{
case AL_GAIN:
pflValues[0] = pContext->Listener.Gain;
break;
case AL_POSITION:
pflValues[0] = pContext->Listener.Position[0];
pflValues[1] = pContext->Listener.Position[1];
pflValues[2] = pContext->Listener.Position[2];
break;
case AL_VELOCITY:
pflValues[0] = pContext->Listener.Velocity[0];
pflValues[1] = pContext->Listener.Velocity[1];
pflValues[2] = pContext->Listener.Velocity[2];
break;
case AL_ORIENTATION:
LockContext(Context);
// AT then UP
pflValues[0] = pContext->Listener.Forward[0];
pflValues[1] = pContext->Listener.Forward[1];
pflValues[2] = pContext->Listener.Forward[2];
pflValues[3] = pContext->Listener.Up[0];
pflValues[4] = pContext->Listener.Up[1];
pflValues[5] = pContext->Listener.Up[2];
values[0] = Context->Listener->Forward[0];
values[1] = Context->Listener->Forward[1];
values[2] = Context->Listener->Forward[2];
values[3] = Context->Listener->Up[0];
values[4] = Context->Listener->Up[1];
values[5] = Context->Listener->Up[2];
UnlockContext(Context);
break;
default:
alSetError(AL_INVALID_ENUM);
break;
}
al_throwerr(Context, AL_INVALID_ENUM);
}
else
alSetError(AL_INVALID_VALUE);
ProcessContext(pContext);
}
else
alSetError(AL_INVALID_OPERATION);
al_endtry;
return;
ALCcontext_DecRef(Context);
}
ALAPI ALvoid ALAPIENTRY alGetListeneri(ALenum eParam, ALint *plValue)
AL_API ALvoid AL_APIENTRY alGetListeneri(ALenum param, ALint *value)
{
ALCcontext *pContext;
ALCcontext *Context;
pContext = alcGetCurrentContext();
if (pContext)
Context = GetContextRef();
if(!Context) return;
al_try
{
SuspendContext(pContext);
if (plValue)
CHECK_VALUE(Context, value);
switch(param)
{
switch (eParam)
{
default:
alSetError(AL_INVALID_ENUM);
break;
}
al_throwerr(Context, AL_INVALID_ENUM);
}
else
alSetError(AL_INVALID_VALUE);
ProcessContext(pContext);
}
else
alSetError(AL_INVALID_OPERATION);
al_endtry;
return;
ALCcontext_DecRef(Context);
}
ALAPI void ALAPIENTRY alGetListener3i(ALenum eParam, ALint *plValue1, ALint *plValue2, ALint *plValue3)
AL_API void AL_APIENTRY alGetListener3i(ALenum param, ALint *value1, ALint *value2, ALint *value3)
{
ALCcontext *pContext;
ALCcontext *Context;
pContext = alcGetCurrentContext();
if (pContext)
Context = GetContextRef();
if(!Context) return;
al_try
{
SuspendContext(pContext);
if ((plValue1) && (plValue2) && (plValue3))
CHECK_VALUE(Context, value1 && value2 && value3);
switch (param)
{
switch (eParam)
{
case AL_POSITION:
*plValue1 = (ALint)pContext->Listener.Position[0];
*plValue2 = (ALint)pContext->Listener.Position[1];
*plValue3 = (ALint)pContext->Listener.Position[2];
LockContext(Context);
*value1 = (ALint)Context->Listener->Position[0];
*value2 = (ALint)Context->Listener->Position[1];
*value3 = (ALint)Context->Listener->Position[2];
UnlockContext(Context);
break;
case AL_VELOCITY:
*plValue1 = (ALint)pContext->Listener.Velocity[0];
*plValue2 = (ALint)pContext->Listener.Velocity[1];
*plValue3 = (ALint)pContext->Listener.Velocity[2];
LockContext(Context);
*value1 = (ALint)Context->Listener->Velocity[0];
*value2 = (ALint)Context->Listener->Velocity[1];
*value3 = (ALint)Context->Listener->Velocity[2];
UnlockContext(Context);
break;
default:
alSetError(AL_INVALID_ENUM);
break;
}
al_throwerr(Context, AL_INVALID_ENUM);
}
else
alSetError(AL_INVALID_VALUE);
ProcessContext(pContext);
}
else
alSetError(AL_INVALID_OPERATION);
al_endtry;
return;
ALCcontext_DecRef(Context);
}
ALAPI void ALAPIENTRY alGetListeneriv(ALenum eParam, ALint* plValues)
AL_API void AL_APIENTRY alGetListeneriv(ALenum param, ALint* values)
{
ALCcontext *pContext;
ALCcontext *Context;
pContext = alcGetCurrentContext();
if (pContext)
switch(param)
{
SuspendContext(pContext);
case AL_POSITION:
case AL_VELOCITY:
alGetListener3i(param, values+0, values+1, values+2);
return;
}
if (plValues)
Context = GetContextRef();
if(!Context) return;
al_try
{
CHECK_VALUE(Context, values);
switch(param)
{
switch (eParam)
{
case AL_POSITION:
plValues[0] = (ALint)pContext->Listener.Position[0];
plValues[1] = (ALint)pContext->Listener.Position[1];
plValues[2] = (ALint)pContext->Listener.Position[2];
break;
case AL_VELOCITY:
plValues[0] = (ALint)pContext->Listener.Velocity[0];
plValues[1] = (ALint)pContext->Listener.Velocity[1];
plValues[2] = (ALint)pContext->Listener.Velocity[2];
break;
case AL_ORIENTATION:
LockContext(Context);
// AT then UP
plValues[0] = (ALint)pContext->Listener.Forward[0];
plValues[1] = (ALint)pContext->Listener.Forward[1];
plValues[2] = (ALint)pContext->Listener.Forward[2];
plValues[3] = (ALint)pContext->Listener.Up[0];
plValues[4] = (ALint)pContext->Listener.Up[1];
plValues[5] = (ALint)pContext->Listener.Up[2];
values[0] = (ALint)Context->Listener->Forward[0];
values[1] = (ALint)Context->Listener->Forward[1];
values[2] = (ALint)Context->Listener->Forward[2];
values[3] = (ALint)Context->Listener->Up[0];
values[4] = (ALint)Context->Listener->Up[1];
values[5] = (ALint)Context->Listener->Up[2];
UnlockContext(Context);
break;
default:
alSetError(AL_INVALID_ENUM);
break;
}
al_throwerr(Context, AL_INVALID_ENUM);
}
else
alSetError(AL_INVALID_VALUE);
ProcessContext(pContext);
}
else
alSetError(AL_INVALID_OPERATION);
al_endtry;
return;
ALCcontext_DecRef(Context);
}
+2567 -1831
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+415 -418
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+43 -62
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@@ -18,91 +18,72 @@
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdlib.h>
#include "alMain.h"
#include "alThunk.h"
typedef struct {
ALvoid *ptr;
ALboolean InUse;
} ThunkEntry;
static ThunkEntry *g_ThunkArray;
static ALuint g_ThunkArraySize;
static ALenum *ThunkArray;
static ALuint ThunkArraySize;
static RWLock ThunkLock;
static CRITICAL_SECTION g_ThunkLock;
void alThunkInit(void)
void ThunkInit(void)
{
InitializeCriticalSection(&g_ThunkLock);
g_ThunkArraySize = 1;
g_ThunkArray = calloc(1, g_ThunkArraySize * sizeof(ThunkEntry));
RWLockInit(&ThunkLock);
ThunkArraySize = 1;
ThunkArray = calloc(1, ThunkArraySize * sizeof(*ThunkArray));
}
void alThunkExit(void)
void ThunkExit(void)
{
free(g_ThunkArray);
g_ThunkArray = NULL;
g_ThunkArraySize = 0;
DeleteCriticalSection(&g_ThunkLock);
free(ThunkArray);
ThunkArray = NULL;
ThunkArraySize = 0;
}
ALuint alThunkAddEntry(ALvoid *ptr)
ALenum NewThunkEntry(ALuint *index)
{
ALuint index;
ALenum *NewList;
ALuint i;
EnterCriticalSection(&g_ThunkLock);
for(index = 0;index < g_ThunkArraySize;index++)
ReadLock(&ThunkLock);
for(i = 0;i < ThunkArraySize;i++)
{
if(g_ThunkArray[index].InUse == AL_FALSE)
break;
}
if(index == g_ThunkArraySize)
{
ThunkEntry *NewList;
NewList = realloc(g_ThunkArray, g_ThunkArraySize*2 * sizeof(ThunkEntry));
if(!NewList)
if(ExchangeInt(&ThunkArray[i], AL_TRUE) == AL_FALSE)
{
LeaveCriticalSection(&g_ThunkLock);
return 0;
ReadUnlock(&ThunkLock);
*index = i+1;
return AL_NO_ERROR;
}
memset(&NewList[g_ThunkArraySize], 0, g_ThunkArraySize*sizeof(ThunkEntry));
g_ThunkArraySize *= 2;
g_ThunkArray = NewList;
}
ReadUnlock(&ThunkLock);
g_ThunkArray[index].ptr = ptr;
g_ThunkArray[index].InUse = AL_TRUE;
WriteLock(&ThunkLock);
NewList = realloc(ThunkArray, ThunkArraySize*2 * sizeof(*ThunkArray));
if(!NewList)
{
WriteUnlock(&ThunkLock);
ERR("Realloc failed to increase to %u entries!\n", ThunkArraySize*2);
return AL_OUT_OF_MEMORY;
}
memset(&NewList[ThunkArraySize], 0, ThunkArraySize*sizeof(*ThunkArray));
ThunkArraySize *= 2;
ThunkArray = NewList;
LeaveCriticalSection(&g_ThunkLock);
ThunkArray[i] = AL_TRUE;
WriteUnlock(&ThunkLock);
return index+1;
*index = i+1;
return AL_NO_ERROR;
}
void alThunkRemoveEntry(ALuint index)
void FreeThunkEntry(ALuint index)
{
EnterCriticalSection(&g_ThunkLock);
if(index > 0 && index <= g_ThunkArraySize)
g_ThunkArray[index-1].InUse = AL_FALSE;
LeaveCriticalSection(&g_ThunkLock);
}
ALvoid *alThunkLookupEntry(ALuint index)
{
ALvoid *ptr = NULL;
EnterCriticalSection(&g_ThunkLock);
if(index > 0 && index <= g_ThunkArraySize)
ptr = g_ThunkArray[index-1].ptr;
LeaveCriticalSection(&g_ThunkLock);
return ptr;
ReadLock(&ThunkLock);
if(index > 0 && index <= ThunkArraySize)
ExchangeInt(&ThunkArray[index-1], AL_FALSE);
ReadUnlock(&ThunkLock);
}
+53
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@@ -0,0 +1,53 @@
Source Install
==============
To install OpenAL Soft, use your favorite shell to go into the build/
directory, and run:
cmake ..
Assuming configuration went well, you can then build it, typically using GNU
Make (KDevelop, MSVC, and others are possible depending on your system setup
and CMake configuration).
Please Note: Double check that the appropriate backends were detected. Often,
complaints of no sound, crashing, and missing devices can be solved by making
sure the correct backends are being used. CMake's output will identify which
backends were enabled.
For most systems, you will likely want to make sure ALSA, OSS, and PulseAudio
were detected (if your target system uses them). For Windows, make sure
DirectSound was detected.
Utilities
=========
The source package comes with an informational utility, openal-info, and is
built by default. It prints out information provided by the ALC and AL sub-
systems, including discovered devices, version information, and extensions.
Configuration
=============
OpenAL Soft can be configured on a per-user and per-system basis. This allows
users and sysadmins to control information provided to applications, as well
as application-agnostic behavior of the library. See alsoftrc.sample for
available settings.
Acknowledgements
================
Special thanks go to:
Creative Labs for the original source code this is based off of.
Christopher Fitzgerald for the current reverb effect implementation, and
helping with the low-pass filter.
Christian Borss for the 3D panning code the current implementation is heavilly
based on.
Ben Davis for the idea behind the current click-removal code.
+30
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@@ -0,0 +1,30 @@
# Cross-compiling requires CMake 2.6 or newer. To use it from build/, call it
# like this:
# cmake .. -DCMAKE_TOOLCHAIN_FILE=../XCompile.txt -DHOST=i686-pc-mingw32
# Where 'i686-pc-mingw32' is the host prefix for your cross-compiler. If you
# already have a toolchain file setup, you may use that instead of this file.
# the name of the target operating system
SET(CMAKE_SYSTEM_NAME Windows)
# which compilers to use for C and C++
SET(CMAKE_C_COMPILER "${HOST}-gcc")
SET(CMAKE_CXX_COMPILER "${HOST}-g++")
# here is the target environment located
SET(CMAKE_FIND_ROOT_PATH "/usr/${HOST}")
# here is where stuff gets installed to
SET(CMAKE_INSTALL_PREFIX "${CMAKE_FIND_ROOT_PATH}/usr" CACHE STRING "Install path prefix, prepended onto install directories." FORCE)
# adjust the default behaviour of the FIND_XXX() commands:
# search headers and libraries in the target environment, search
# programs in the host environment
set(CMAKE_FIND_ROOT_PATH_MODE_PROGRAM NEVER)
set(CMAKE_FIND_ROOT_PATH_MODE_LIBRARY ONLY)
set(CMAKE_FIND_ROOT_PATH_MODE_INCLUDE ONLY)
# set env vars so that pkg-config will look in the appropriate directory for
# .pc files (as there seems to be no way to force using ${HOST}-pkg-config)
set(ENV{PKG_CONFIG_LIBDIR} "${CMAKE_INSTALL_PREFIX}/lib/pkgconfig")
set(ENV{PKG_CONFIG_PATH} "")
+328
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@@ -0,0 +1,328 @@
# OpenAL config file. Options that are not under a block or are under the
# [general] block are for general, non-backend-specific options. Blocks may
# appear multiple times, and duplicated options will take the last value
# specified.
# The system-wide settings can be put in /etc/openal/alsoft.conf and user-
# specific override settings in ~/.alsoftrc.
# For Windows, these settings should go into %AppData%\alsoft.ini
# Option and block names are case-insenstive. The supplied values are only
# hints and may not be honored (though generally it'll try to get as close as
# possible). Note: options that are left unset may default to app- or system-
# specified values. These are the current available settings:
## disable-cpu-exts:
# Disables use of the listed CPU extensions. Certain methods may utilize CPU
# extensions when detected, and this option is useful for preventing those
# extensions from being used. The available extensions are: sse, neon.
# Specifying 'all' disables use of all extensions.
#disable-cpu-exts =
## channels:
# Sets the output channel configuration. If left unspecified, one will try to
# be detected from the system, and defaulting to stereo. The available values
# are: mono, stereo, quad, surround51, surround61, surround71
#channels = stereo
## sample-type:
# Sets the output sample type. Currently, all mixing is done with 32-bit float
# and converted to the output sample type as needed. Available values are:
# int8 - signed 8-bit int
# uint8 - unsigned 8-bit int
# int16 - signed 16-bit int
# uint16 - unsigned 16-bit int
# int32 - signed 32-bit int
# uint32 - unsigned 32-bit int
# float32 - 32-bit float
#sample-type = float32
## hrtf:
# Enables HRTF filters. These filters provide for better sound spatialization
# while using headphones. The default filter will only work when output is
# 44100hz stereo. While HRTF is active, the cf_level option is disabled.
# Default is disabled since stereo speaker output quality may suffer.
#hrtf = false
## hrtf_tables
# Specifies a comma-separated list of files containing HRTF data sets. The
# listed data sets can be used in place of or in addiiton to the the built-in
# set. The format of the files are described in hrtf.txt. The filenames may
# contain these markers, which will be replaced as needed:
# %r - Device sampling rate
# %% - Percent sign (%)
# So if this is set to "kemar-%r-diffuse.mhr", it will try to open
# "kemar-44100-diffuse.mhr" if the device is using 44100hz output, or
# "kemar-48000-diffuse.mhr" if the device is using 48000hz output, etc.
#hrtf_tables =
## cf_level:
# Sets the crossfeed level for stereo output. Valid values are:
# 0 - No crossfeed
# 1 - Low crossfeed
# 2 - Middle crossfeed
# 3 - High crossfeed (virtual speakers are closer to itself)
# 4 - Low easy crossfeed
# 5 - Middle easy crossfeed
# 6 - High easy crossfeed
# Users of headphones may want to try various settings. Has no effect on non-
# stereo modes.
#cf_level = 0
## wide-stereo:
# Specifies that stereo sources are given a width of about 120 degrees on each
# channel, centering on -90 (left) and +90 (right), as opposed to being points
# placed at -30 (left) and +30 (right). This can be useful for surround-sound
# to give stereo sources a more encompassing sound. Note that the sound's
# overall volume will be slightly reduced to account for the extra output.
#wide-stereo = false
## frequency:
# Sets the output frequency.
#frequency = 44100
## resampler:
# Selects the resampler used when mixing sources. Valid values are:
# point - nearest sample, no interpolation
# linear - extrapolates samples using a linear slope between samples
# cubic - extrapolates samples using a Catmull-Rom spline
# Specifying other values will result in using the default (linear).
#resampler = linear
## rt-prio:
# Sets real-time priority for the mixing thread. Not all drivers may use this
# (eg. PortAudio) as they already control the priority of the mixing thread.
# 0 and negative values will disable it. Note that this may constitute a
# security risk since a real-time priority thread can indefinitely block
# normal-priority threads if it fails to wait. As such, the default is
# disabled.
#rt-prio = 0
## period_size:
# Sets the update period size, in frames. This is the number of frames needed
# for each mixing update. Acceptable values range between 64 and 8192.
#period_size = 1024
## periods:
# Sets the number of update periods. Higher values create a larger mix ahead,
# which helps protect against skips when the CPU is under load, but increases
# the delay between a sound getting mixed and being heard. Acceptable values
# range between 2 and 16.
#periods = 4
## sources:
# Sets the maximum number of allocatable sources. Lower values may help for
# systems with apps that try to play more sounds than the CPU can handle.
#sources = 256
## drivers:
# Sets the backend driver list order, comma-seperated. Unknown backends and
# duplicated names are ignored. Unlisted backends won't be considered for use
# unless the list is ended with a comma (eg. 'oss,' will list OSS first
# followed by all other available backends, while 'oss' will list OSS only).
# Backends prepended with - won't be available for use (eg. '-oss,' will allow
# all available backends except OSS). An empty list means the default.
#drivers = pulse,alsa,core,oss,solaris,sndio,mmdevapi,dsound,winmm,port,opensl,null,wave
## excludefx:
# Sets which effects to exclude, preventing apps from using them. This can
# help for apps that try to use effects which are too CPU intensive for the
# system to handle. Available effects are: eaxreverb,reverb,echo,modulator,
# dedicated
#excludefx =
## slots:
# Sets the maximum number of Auxiliary Effect Slots an app can create. A slot
# can use a non-negligible amount of CPU time if an effect is set on it even
# if no sources are feeding it, so this may help when apps use more than the
# system can handle.
#slots = 4
## sends:
# Sets the number of auxiliary sends per source. When not specified (default),
# it allows the app to request how many it wants. The maximum value currently
# possible is 4.
#sends =
## layout:
# Sets the virtual speaker layout. Values are specified in degrees, where 0 is
# straight in front, negative goes left, and positive goes right. Unspecified
# speakers will remain at their default positions (which are dependant on the
# output format). Available speakers are back-left(bl), side-left(sl), front-
# left(fl), front-center(fc), front-right(fr), side-right(sr), back-right(br),
# and back-center(bc).
#layout =
## layout_*:
# Channel-specific layouts may be specified to override the layout option. The
# same speakers as the layout option are available, and the default settings
# are shown below.
#layout_stereo = fl=-90, fr=90
#layout_quad = fl=-45, fr=45, bl=-135, br=135
#layout_surround51 = fl=-30, fr=30, fc=0, bl=-110, br=110
#layout_surround61 = fl=-30, fr=30, fc=0, sl=-90, sr=90, bc=180
#layout_surround71 = fl=-30, fr=30, fc=0, sl=-90, sr=90, bl=-150, br=150
## default-reverb:
# A reverb preset that applies by default to all sources on send 0
# (applications that set their own slots on send 0 will override this).
# Available presets are: None, Generic, PaddedCell, Room, Bathroom,
# Livingroom, Stoneroom, Auditorium, ConcertHall, Cave, Arena, Hangar,
# CarpetedHallway, Hallway, StoneCorridor, Alley, Forest, City, Moutains,
# Quarry, Plain, ParkingLot, SewerPipe, Underwater, Drugged, Dizzy, Psychotic.
#default-reverb =
## trap-alc-error:
# Generates a SIGTRAP signal when an ALC device error is generated, on systems
# that support it. This helps when debugging, while trying to find the cause
# of a device error. On Windows, a breakpoint exception is generated.
#trap-alc-error = false
## trap-al-error:
# Generates a SIGTRAP signal when an AL context error is generated, on systems
# that support it. This helps when debugging, while trying to find the cause
# of a context error. On Windows, a breakpoint exception is generated.
#trap-al-error = false
##
## Reverb effect stuff (includes EAX reverb)
##
[reverb]
## boost:
# A global amplification for reverb output, expressed in decibels. The value
# is logarithmic, so +6 will be a scale of (approximately) 2x, +12 will be a
# scale of 4x, etc. Similarly, -6 will be about half, and -12 about 1/4th. A
# value of 0 means no change.
#boost = 0
## emulate-eax:
# Allows the standard reverb effect to be used in place of EAX reverb. EAX
# reverb processing is a bit more CPU intensive than standard, so this option
# allows a simpler effect to be used at the loss of some quality.
#emulate-eax = false
##
## ALSA backend stuff
##
[alsa]
## device:
# Sets the device name for the default playback device.
#device = default
## device-prefix:
# Sets the prefix used by the discovered (non-default) playback devices. This
# will be appended with "CARD=c,DEV=d", where c is the card id and d is the
# device index for the requested device name.
#device-prefix = plughw:
## device-prefix-*:
# Card- and device-specific prefixes may be used to override the device-prefix
# option. The option may specify the card id (eg, device-prefix-NVidia), or
# the card id and device index (eg, device-prefix-NVidia-0). The card id is
# case-sensitive.
#device-prefix- =
## capture:
# Sets the device name for the default capture device.
#capture = default
## capture-prefix:
# Sets the prefix used by the discovered (non-default) capture devices. This
# will be appended with "CARD=c,DEV=d", where c is the card id and d is the
# device number for the requested device name.
#capture-prefix = plughw:
## capture-prefix-*:
# Card- and device-specific prefixes may be used to override the
# capture-prefix option. The option may specify the card id (eg,
# capture-prefix-NVidia), or the card id and device index (eg,
# capture-prefix-NVidia-0). The card id is case-sensitive.
#capture-prefix- =
## mmap:
# Sets whether to try using mmap mode (helps reduce latencies and CPU
# consumption). If mmap isn't available, it will automatically fall back to
# non-mmap mode. True, yes, on, and non-0 values will attempt to use mmap. 0
# and anything else will force mmap off.
#mmap = true
##
## OSS backend stuff
##
[oss]
## device:
# Sets the device name for OSS output.
#device = /dev/dsp
## capture:
# Sets the device name for OSS capture.
#capture = /dev/dsp
##
## Solaris backend stuff
##
[solaris]
## device:
# Sets the device name for Solaris output.
#device = /dev/audio
##
## MMDevApi backend stuff
##
[mmdevapi]
##
## DirectSound backend stuff
##
[dsound]
##
## Windows Multimedia backend stuff
##
[winmm]
##
## PortAudio backend stuff
##
[port]
## device:
# Sets the device index for output. Negative values will use the default as
# given by PortAudio itself.
#device = -1
## capture:
# Sets the device index for capture. Negative values will use the default as
# given by PortAudio itself.
#capture = -1
##
## PulseAudio backend stuff
##
[pulse]
## spawn-server:
# Attempts to spawn a PulseAudio server when requesting to open a PulseAudio
# device. Setting autospawn to false in PulseAudio's client.conf will still
# prevent autospawning even if this is set to true.
#spawn-server = true
## allow-moves:
# Allows PulseAudio to move active streams to different devices. Note that the
# device specifier seen by applications will not be updated when this occurs,
# and neither will the AL device configuration (sample rate, format, etc).
#allow-moves = false
##
## Wave File Writer stuff
##
[wave]
## file:
# Sets the filename of the wave file to write to. An empty name prevents the
# backend from opening, even when explicitly requested.
# THIS WILL OVERWRITE EXISTING FILES WITHOUT QUESTION!
#file =
View File
+59
View File
@@ -0,0 +1,59 @@
# - Check if the C source code provided in the SOURCE argument compiles.
# CHECK_C_SOURCE_COMPILES(SOURCE VAR)
#
# FLAG - compiler flag to check
# VAR - variable to store whether the source code compiled
#
# The following variables may be set before calling this macro to
# modify the way the check is run:
#
# CMAKE_REQUIRED_FLAGS = string of compile command line flags
# CMAKE_REQUIRED_DEFINITIONS = list of macros to define (-DFOO=bar)
# CMAKE_REQUIRED_INCLUDES = list of include directories
# CMAKE_REQUIRED_LIBRARIES = list of libraries to link
MACRO(CHECK_C_COMPILER_FLAG FLAG VAR)
IF("${VAR}" MATCHES "^${VAR}$")
SET(MACRO_CHECK_FUNCTION_DEFINITIONS
"${FLAG} ${CMAKE_REQUIRED_FLAGS}")
IF(CMAKE_REQUIRED_LIBRARIES)
SET(CHECK_C_COMPILER_FLAG_ADD_LIBRARIES
"-DLINK_LIBRARIES:STRING=${CMAKE_REQUIRED_LIBRARIES}")
ELSE(CMAKE_REQUIRED_LIBRARIES)
SET(CHECK_C_COMPILER_FLAG_ADD_LIBRARIES)
ENDIF(CMAKE_REQUIRED_LIBRARIES)
IF(CMAKE_REQUIRED_INCLUDES)
SET(CHECK_C_COMPILER_FLAG_ADD_INCLUDES
"-DINCLUDE_DIRECTORIES:STRING=${CMAKE_REQUIRED_INCLUDES}")
ELSE(CMAKE_REQUIRED_INCLUDES)
SET(CHECK_C_COMPILER_FLAG_ADD_INCLUDES)
ENDIF(CMAKE_REQUIRED_INCLUDES)
FILE(WRITE "${CMAKE_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeTmp/src.c"
"int main() {return 0;}\n")
MESSAGE(STATUS "Checking if C compiler supports ${FLAG}")
TRY_COMPILE(${VAR}
${CMAKE_BINARY_DIR}
${CMAKE_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeTmp/src.c
COMPILE_DEFINITIONS ${CMAKE_REQUIRED_DEFINITIONS}
CMAKE_FLAGS -DCOMPILE_DEFINITIONS:STRING=${MACRO_CHECK_FUNCTION_DEFINITIONS}
"${CHECK_C_COMPILER_FLAG_ADD_LIBRARIES}"
"${CHECK_C_COMPILER_FLAG_ADD_INCLUDES}"
OUTPUT_VARIABLE OUTPUT)
IF(${VAR})
SET(${VAR} 1 CACHE INTERNAL "Test ${VAR}")
MESSAGE(STATUS "Checking if C compiler supports ${FLAG} - Success")
FILE(APPEND ${CMAKE_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeOutput.log
"Performing C SOURCE FILE Test ${VAR} succeded with the following output:\n"
"${OUTPUT}\n"
"Source file was:\n${SOURCE}\n")
ELSE(${VAR})
MESSAGE(STATUS "Checking if C compiler supports ${FLAG} - Failed")
SET(${VAR} "" CACHE INTERNAL "Test ${VAR}")
FILE(APPEND ${CMAKE_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeError.log
"Performing C SOURCE FILE Test ${VAR} failed with the following output:\n"
"${OUTPUT}\n"
"Source file was:\n${SOURCE}\n")
ENDIF(${VAR})
ENDIF("${VAR}" MATCHES "^${VAR}$")
ENDMACRO(CHECK_C_COMPILER_FLAG)
+92
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@@ -0,0 +1,92 @@
# - Check if a symbol exists as a function, variable, or macro
# CHECK_SYMBOL_EXISTS(<symbol> <files> <variable>)
#
# Check that the <symbol> is available after including given header
# <files> and store the result in a <variable>. Specify the list
# of files in one argument as a semicolon-separated list.
#
# If the header files define the symbol as a macro it is considered
# available and assumed to work. If the header files declare the
# symbol as a function or variable then the symbol must also be
# available for linking. If the symbol is a type or enum value
# it will not be recognized (consider using CheckTypeSize or
# CheckCSourceCompiles).
#
# The following variables may be set before calling this macro to
# modify the way the check is run:
#
# CMAKE_REQUIRED_FLAGS = string of compile command line flags
# CMAKE_REQUIRED_DEFINITIONS = list of macros to define (-DFOO=bar)
# CMAKE_REQUIRED_INCLUDES = list of include directories
# CMAKE_REQUIRED_LIBRARIES = list of libraries to link
#=============================================================================
# Copyright 2003-2011 Kitware, Inc.
#
# Distributed under the OSI-approved BSD License (the "License");
# see accompanying file Copyright.txt for details.
#
# This software is distributed WITHOUT ANY WARRANTY; without even the
# implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
# See the License for more information.
#=============================================================================
# (To distribute this file outside of CMake, substitute the full
# License text for the above reference.)
MACRO(CHECK_SHARED_FUNCTION_EXISTS SYMBOL FILES LIBRARY LOCATION VARIABLE)
IF("${VARIABLE}" MATCHES "^${VARIABLE}$")
SET(CMAKE_CONFIGURABLE_FILE_CONTENT "/* */\n")
SET(MACRO_CHECK_SYMBOL_EXISTS_FLAGS ${CMAKE_REQUIRED_FLAGS})
IF(CMAKE_REQUIRED_LIBRARIES)
SET(CHECK_SYMBOL_EXISTS_LIBS
"-DLINK_LIBRARIES:STRING=${CMAKE_REQUIRED_LIBRARIES};${LIBRARY}")
ELSE(CMAKE_REQUIRED_LIBRARIES)
SET(CHECK_SYMBOL_EXISTS_LIBS
"-DLINK_LIBRARIES:STRING=${LIBRARY}")
ENDIF(CMAKE_REQUIRED_LIBRARIES)
IF(CMAKE_REQUIRED_INCLUDES)
SET(CMAKE_SYMBOL_EXISTS_INCLUDES
"-DINCLUDE_DIRECTORIES:STRING=${CMAKE_REQUIRED_INCLUDES}")
ELSE(CMAKE_REQUIRED_INCLUDES)
SET(CMAKE_SYMBOL_EXISTS_INCLUDES)
ENDIF(CMAKE_REQUIRED_INCLUDES)
FOREACH(FILE ${FILES})
SET(CMAKE_CONFIGURABLE_FILE_CONTENT
"${CMAKE_CONFIGURABLE_FILE_CONTENT}#include <${FILE}>\n")
ENDFOREACH(FILE)
SET(CMAKE_CONFIGURABLE_FILE_CONTENT
"${CMAKE_CONFIGURABLE_FILE_CONTENT}\nvoid cmakeRequireSymbol(int dummy,...){(void)dummy;}\nint main()\n{\n cmakeRequireSymbol(0,&${SYMBOL});\n return 0;\n}\n")
CONFIGURE_FILE("${CMAKE_ROOT}/Modules/CMakeConfigurableFile.in"
"${CMAKE_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeTmp/CheckSymbolExists.c" @ONLY)
MESSAGE(STATUS "Looking for ${SYMBOL} in ${LIBRARY}")
TRY_COMPILE(${VARIABLE}
${CMAKE_BINARY_DIR}
${CMAKE_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeTmp/CheckSymbolExists.c
COMPILE_DEFINITIONS ${CMAKE_REQUIRED_DEFINITIONS}
CMAKE_FLAGS
-DCOMPILE_DEFINITIONS:STRING=${MACRO_CHECK_SYMBOL_EXISTS_FLAGS}
-DLINK_DIRECTORIES:STRING=${LOCATION}
"${CHECK_SYMBOL_EXISTS_LIBS}"
"${CMAKE_SYMBOL_EXISTS_INCLUDES}"
OUTPUT_VARIABLE OUTPUT)
IF(${VARIABLE})
MESSAGE(STATUS "Looking for ${SYMBOL} in ${LIBRARY} - found")
SET(${VARIABLE} 1 CACHE INTERNAL "Have symbol ${SYMBOL} in ${LIBRARY}")
FILE(APPEND ${CMAKE_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeOutput.log
"Determining if the ${SYMBOL} "
"exist in ${LIBRARY} passed with the following output:\n"
"${OUTPUT}\nFile ${CMAKE_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeTmp/CheckSymbolExists.c:\n"
"${CMAKE_CONFIGURABLE_FILE_CONTENT}\n")
ELSE(${VARIABLE})
MESSAGE(STATUS "Looking for ${SYMBOL} in ${LIBRARY} - not found.")
SET(${VARIABLE} "" CACHE INTERNAL "Have symbol ${SYMBOL} in ${LIBRARY}")
FILE(APPEND ${CMAKE_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeError.log
"Determining if the ${SYMBOL} "
"exist in ${LIBRARY} failed with the following output:\n"
"${OUTPUT}\nFile ${CMAKE_BINARY_DIR}${CMAKE_FILES_DIRECTORY}/CMakeTmp/CheckSymbolExists.c:\n"
"${CMAKE_CONFIGURABLE_FILE_CONTENT}\n")
ENDIF(${VARIABLE})
ENDIF("${VARIABLE}" MATCHES "^${VARIABLE}$")
ENDMACRO(CHECK_SHARED_FUNCTION_EXISTS)
+111 -16
View File
@@ -1,5 +1,33 @@
#ifndef CONFIG_H
#define CONFIG_H
/* API declaration export attribute */
#define AL_API ${EXPORT_DECL}
#define ALC_API ${EXPORT_DECL}
/* Define to the library version */
#define ALSOFT_VERSION "${LIB_VERSION}"
/* Define any available alignment declaration */
#define ALIGN(x) ${ALIGN_DECL}
#ifdef __MINGW32__
#define align(x) aligned(x)
#endif
/* Define to the appropriate 'restrict' keyword */
#define RESTRICT ${RESTRICT_DECL}
/* Define if we have the C11 aligned_alloc function */
#cmakedefine HAVE_ALIGNED_ALLOC
/* Define if we have the posix_memalign function */
#cmakedefine HAVE_POSIX_MEMALIGN
/* Define if we have the _aligned_malloc function */
#cmakedefine HAVE__ALIGNED_MALLOC
/* Define if we have SSE CPU extensions */
#cmakedefine HAVE_SSE
/* Define if we have ARM Neon CPU extensions */
#cmakedefine HAVE_NEON
/* Define if we have the ALSA backend */
#cmakedefine HAVE_ALSA
@@ -7,24 +35,45 @@
/* Define if we have the OSS backend */
#cmakedefine HAVE_OSS
/* Define if we have the Solaris backend */
#cmakedefine HAVE_SOLARIS
/* Define if we have the SndIO backend */
#cmakedefine HAVE_SNDIO
/* Define if we have the MMDevApi backend */
#cmakedefine HAVE_MMDEVAPI
/* Define if we have the DSound backend */
#cmakedefine HAVE_DSOUND
/* Define if we have the Windows Multimedia backend */
#cmakedefine HAVE_WINMM
/* Define if we have dlfcn.h */
#cmakedefine HAVE_DLFCN_H
/* Define if we have the PortAudio backend */
#cmakedefine HAVE_PORTAUDIO
/* Define if we have the sqrtf function */
#cmakedefine HAVE_SQRTF
/* Define if we have the PulseAudio backend */
#cmakedefine HAVE_PULSEAUDIO
/* Define if we have the CoreAudio backend */
#cmakedefine HAVE_COREAUDIO
/* Define if we have the OpenSL backend */
#cmakedefine HAVE_OPENSL
/* Define if we have the Wave Writer backend */
#cmakedefine HAVE_WAVE
/* Define if we have the stat function */
#cmakedefine HAVE_STAT
/* Define if we have the lrintf function */
#cmakedefine HAVE_LRINTF
/* Define if we have the strtof function */
#cmakedefine HAVE_STRTOF
/* Define if we have stdint.h */
#cmakedefine HAVE_STDINT_H
/* Define if we have the __int64 type */
#cmakedefine HAVE___INT64
@@ -34,13 +83,59 @@
/* Define to the size of a long long int type */
#cmakedefine SIZEOF_LONG_LONG ${SIZEOF_LONG_LONG}
/* Define to the size of an unsigned int type */
#cmakedefine SIZEOF_UINT ${SIZEOF_UINT}
/* Define to the size of a void pointer type */
#cmakedefine SIZEOF_VOIDP ${SIZEOF_VOIDP}
/* Define if we have GCC's destructor attribute */
#cmakedefine HAVE_GCC_DESTRUCTOR
#endif
/* Define if we have GCC's format attribute */
#cmakedefine HAVE_GCC_FORMAT
/* Define if we have stdint.h */
#cmakedefine HAVE_STDINT_H
/* Define if we have windows.h */
#cmakedefine HAVE_WINDOWS_H
/* Define if we have dlfcn.h */
#cmakedefine HAVE_DLFCN_H
/* Define if we have pthread_np.h */
#cmakedefine HAVE_PTHREAD_NP_H
/* Define if we have xmmintrin.h */
#cmakedefine HAVE_XMMINTRIN_H
/* Define if we have arm_neon.h */
#cmakedefine HAVE_ARM_NEON_H
/* Define if we have malloc.h */
#cmakedefine HAVE_MALLOC_H
/* Define if we have cpuid.h */
#cmakedefine HAVE_CPUID_H
/* Define if we have guiddef.h */
#cmakedefine HAVE_GUIDDEF_H
/* Define if we have initguid.h */
#cmakedefine HAVE_INITGUID_H
/* Define if we have ieeefp.h */
#cmakedefine HAVE_IEEEFP_H
/* Define if we have float.h */
#cmakedefine HAVE_FLOAT_H
/* Define if we have fenv.h */
#cmakedefine HAVE_FENV_H
/* Define if we have fesetround() */
#cmakedefine HAVE_FESETROUND
/* Define if we have _controlfp() */
#cmakedefine HAVE__CONTROLFP
/* Define if we have __control87_2() */
#cmakedefine HAVE___CONTROL87_2
/* Define if we have pthread_setschedparam() */
#cmakedefine HAVE_PTHREAD_SETSCHEDPARAM
+71
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@@ -0,0 +1,71 @@
Useful Environment Variables
Below is a list of environment variables that can be set to aid with running or
debugging apps that use OpenAL Soft. They should be set before the app is run.
*** Logging ***
ALSOFT_LOGLEVEL
Specifies the amount of logging OpenAL Soft will write out:
0 - Effectively disables all logging
1 - Prints out errors only
2 - Prints out warnings and errors
3 - Prints out additional information, as well as warnings and errors
4 - Same as 3, but also device and context reference count changes. This will
print out *a lot* of info, and is generally not useful unless you're trying
to track a reference leak within the library.
ALSOFT_LOGFILE
Specifies a filename that logged output will be written to. Note that the file
will be first cleared when logging is initialized.
*** Overrides ***
ALSOFT_CONF
Specifies an additional configuration file to load settings from. These
settings will take precedence over the global and user configs, but not other
environment variable settings.
ALSOFT_DRIVERS
Overrides the drivers config option. This specifies which backend drivers to
consider or not consider for use. Please see the drivers option in
alsoftrc.sample for a list of available drivers.
ALSOFT_DEFAULT_REVERB
Specifies the default reverb preset to apply to sources. Please see the
default-reverb option in alsoftrc.sample for additional information and a list
of available presets.
ALSOFT_TRAP_AL_ERROR
Set to "true" or "1" to force trapping AL errors. Like the trap-al-error config
option, this will raise a SIGTRAP signal (or a breakpoint exception under
Windows) when a context-level error is generated. Useful when run under a
debugger as it will break execution right when the error occurs, making it
easier to track the cause.
ALSOFT_TRAP_ALC_ERROR
Set to "true" or "1" to force trapping ALC errors. Like the trap-alc-error
config option, this will raise a SIGTRAP signal (or a breakpoint exception
under Windows) when a device-level error is generated. Useful when run under a
debugger as it will break execution right when the error occurs, making it
easier to track the cause.
ALSOFT_TRAP_ERROR
Set to "true" or "1" to force trapping both ALC and AL errors.
*** Compatibility ***
__ALSOFT_HALF_ANGLE_CONES
Older versions of OpenAL Soft incorrectly calculated the cone angles to range
between 0 and 180 degrees, instead of the expected range of 0 to 360 degrees.
Setting this to "true" or "1" restores the old buggy behavior, for apps that
were written to expect the incorrect range.
__ALSOFT_REVERSE_Z
Applications that don't natively use OpenAL's coordinate system have to convert
to it before passing in 3D coordinates. Depending on how exactly this is done,
it can cause correct output for stereo but incorrect Z panning for surround
sound (i.e., sounds that are supposed to be behind you sound like they're in
front, and vice-versa). Setting this to "true" or "1" will negate the localized
Z coordinate to attempt to fix output for apps that have incorrect front/back
panning.
+213
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@@ -0,0 +1,213 @@
/*
* OpenAL Source Latency Example
*
* Copyright (c) 2012 by Chris Robinson <chris.kcat@gmail.com>
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/* This file contains an example for checking the latency of a sound. */
#include <stdio.h>
#include <assert.h>
#ifdef _WIN32
#define WIN32_LEAN_AND_MEAN
#include <windows.h>
#else
#include <unistd.h>
#define Sleep(x) usleep((x)*1000)
#endif
#include "AL/al.h"
#include "AL/alc.h"
#include "AL/alext.h"
#include "common/alhelpers.h"
#include "common/alffmpeg.h"
LPALBUFFERSAMPLESSOFT alBufferSamplesSOFT = wrap_BufferSamples;
LPALISBUFFERFORMATSUPPORTEDSOFT alIsBufferFormatSupportedSOFT;
LPALSOURCEDSOFT alSourcedSOFT;
LPALSOURCE3DSOFT alSource3dSOFT;
LPALSOURCEDVSOFT alSourcedvSOFT;
LPALGETSOURCEDSOFT alGetSourcedSOFT;
LPALGETSOURCE3DSOFT alGetSource3dSOFT;
LPALGETSOURCEDVSOFT alGetSourcedvSOFT;
LPALSOURCEI64SOFT alSourcei64SOFT;
LPALSOURCE3I64SOFT alSource3i64SOFT;
LPALSOURCEI64VSOFT alSourcei64vSOFT;
LPALGETSOURCEI64SOFT alGetSourcei64SOFT;
LPALGETSOURCE3I64SOFT alGetSource3i64SOFT;
LPALGETSOURCEI64VSOFT alGetSourcei64vSOFT;
/* LoadBuffer loads the named audio file into an OpenAL buffer object, and
* returns the new buffer ID. */
static ALuint LoadSound(const char *filename)
{
ALenum err, format, type, channels;
ALuint rate, buffer;
size_t datalen;
void *data;
FilePtr audiofile;
StreamPtr sound;
/* Open the file and get the first stream from it */
audiofile = openAVFile(filename);
sound = getAVAudioStream(audiofile, 0);
if(!sound)
{
fprintf(stderr, "Could not open audio in %s\n", filename);
closeAVFile(audiofile);
return 0;
}
/* Get the sound format, and figure out the OpenAL format */
if(getAVAudioInfo(sound, &rate, &channels, &type) != 0)
{
fprintf(stderr, "Error getting audio info for %s\n", filename);
closeAVFile(audiofile);
return 0;
}
format = GetFormat(channels, type, alIsBufferFormatSupportedSOFT);
if(format == AL_NONE)
{
fprintf(stderr, "Unsupported format (%s, %s) for %s\n",
ChannelsName(channels), TypeName(type), filename);
closeAVFile(audiofile);
return 0;
}
/* Decode the whole audio stream to a buffer. */
data = decodeAVAudioStream(sound, &datalen);
if(!data)
{
fprintf(stderr, "Failed to read audio from %s\n", filename);
closeAVFile(audiofile);
return 0;
}
/* Buffer the audio data into a new buffer object, then free the data and
* close the file. */
buffer = 0;
alGenBuffers(1, &buffer);
alBufferSamplesSOFT(buffer, rate, format, BytesToFrames(datalen, channels, type),
channels, type, data);
free(data);
closeAVFile(audiofile);
/* Check if an error occured, and clean up if so. */
err = alGetError();
if(err != AL_NO_ERROR)
{
fprintf(stderr, "OpenAL Error: %s\n", alGetString(err));
if(alIsBuffer(buffer))
alDeleteBuffers(1, &buffer);
return 0;
}
return buffer;
}
int main(int argc, char **argv)
{
ALuint source, buffer;
ALdouble offsets[2];
ALenum state;
/* Print out usage if no file was specified */
if(argc < 2)
{
fprintf(stderr, "Usage: %s <filename>\n", argv[0]);
return 1;
}
/* Initialize OpenAL with the default device, and check for EFX support. */
if(InitAL() != 0)
return 1;
if(!alIsExtensionPresent("AL_SOFT_source_latency"))
{
fprintf(stderr, "Error: AL_SOFT_source_latency not supported\n");
CloseAL();
return 1;
}
/* Define a macro to help load the function pointers. */
#define LOAD_PROC(x) ((x) = alGetProcAddress(#x))
LOAD_PROC(alSourcedSOFT);
LOAD_PROC(alSource3dSOFT);
LOAD_PROC(alSourcedvSOFT);
LOAD_PROC(alGetSourcedSOFT);
LOAD_PROC(alGetSource3dSOFT);
LOAD_PROC(alGetSourcedvSOFT);
LOAD_PROC(alSourcei64SOFT);
LOAD_PROC(alSource3i64SOFT);
LOAD_PROC(alSourcei64vSOFT);
LOAD_PROC(alGetSourcei64SOFT);
LOAD_PROC(alGetSource3i64SOFT);
LOAD_PROC(alGetSourcei64vSOFT);
if(alIsExtensionPresent("AL_SOFT_buffer_samples"))
{
LOAD_PROC(alBufferSamplesSOFT);
LOAD_PROC(alIsBufferFormatSupportedSOFT);
}
#undef LOAD_PROC
/* Load the sound into a buffer. */
buffer = LoadSound(argv[1]);
if(!buffer)
{
CloseAL();
return 1;
}
/* Create the source to play the sound with. */
source = 0;
alGenSources(1, &source);
alSourcei(source, AL_BUFFER, buffer);
assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
/* Play the sound until it finishes. */
alSourcePlay(source);
do {
Sleep(10);
alGetSourcei(source, AL_SOURCE_STATE, &state);
/* Get the source offset and latency. AL_SEC_OFFSET_LATENCY_SOFT will
* place the offset (in seconds) in offsets[0], and the time until that
* offset will be heard (in seconds) in offsets[1]. */
alGetSourcedvSOFT(source, AL_SEC_OFFSET_LATENCY_SOFT, offsets);
printf("\rOffset: %f - Latency:%3u ms ", offsets[0], (ALuint)(offsets[1]*1000));
fflush(stdout);
} while(alGetError() == AL_NO_ERROR && state == AL_PLAYING);
printf("\n");
/* All done. Delete resources, and close OpenAL. */
alDeleteSources(1, &source);
alDeleteBuffers(1, &buffer);
CloseAL();
return 0;
}
+337
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@@ -0,0 +1,337 @@
/*
* OpenAL Reverb Example
*
* Copyright (c) 2012 by Chris Robinson <chris.kcat@gmail.com>
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/* This file contains an example for applying reverb to a sound. */
#include <stdio.h>
#include <assert.h>
#ifdef _WIN32
#define WIN32_LEAN_AND_MEAN
#include <windows.h>
#else
#include <unistd.h>
#define Sleep(x) usleep((x)*1000)
#endif
#include "AL/al.h"
#include "AL/alc.h"
#include "AL/alext.h"
#include "AL/efx-presets.h"
#include "common/alhelpers.h"
#include "common/alffmpeg.h"
LPALBUFFERSAMPLESSOFT alBufferSamplesSOFT = wrap_BufferSamples;
LPALISBUFFERFORMATSUPPORTEDSOFT alIsBufferFormatSupportedSOFT;
/* Effect object functions */
LPALGENEFFECTS alGenEffects;
LPALDELETEEFFECTS alDeleteEffects;
LPALISEFFECT alIsEffect;
LPALEFFECTI alEffecti;
LPALEFFECTIV alEffectiv;
LPALEFFECTF alEffectf;
LPALEFFECTFV alEffectfv;
LPALGETEFFECTI alGetEffecti;
LPALGETEFFECTIV alGetEffectiv;
LPALGETEFFECTF alGetEffectf;
LPALGETEFFECTFV alGetEffectfv;
/* Auxiliary Effect Slot object functions */
LPALGENAUXILIARYEFFECTSLOTS alGenAuxiliaryEffectSlots;
LPALDELETEAUXILIARYEFFECTSLOTS alDeleteAuxiliaryEffectSlots;
LPALISAUXILIARYEFFECTSLOT alIsAuxiliaryEffectSlot;
LPALAUXILIARYEFFECTSLOTI alAuxiliaryEffectSloti;
LPALAUXILIARYEFFECTSLOTIV alAuxiliaryEffectSlotiv;
LPALAUXILIARYEFFECTSLOTF alAuxiliaryEffectSlotf;
LPALAUXILIARYEFFECTSLOTFV alAuxiliaryEffectSlotfv;
LPALGETAUXILIARYEFFECTSLOTI alGetAuxiliaryEffectSloti;
LPALGETAUXILIARYEFFECTSLOTIV alGetAuxiliaryEffectSlotiv;
LPALGETAUXILIARYEFFECTSLOTF alGetAuxiliaryEffectSlotf;
LPALGETAUXILIARYEFFECTSLOTFV alGetAuxiliaryEffectSlotfv;
/* LoadEffect loads the given reverb properties into a new OpenAL effect
* object, and returns the new effect ID. */
static ALuint LoadEffect(const EFXEAXREVERBPROPERTIES *reverb)
{
ALuint effect = 0;
ALenum err;
/* Create the effect object and check if we can do EAX reverb. */
alGenEffects(1, &effect);
if(alGetEnumValue("AL_EFFECT_EAXREVERB") != 0)
{
printf("Using EAX Reverb\n");
/* EAX Reverb is available. Set the EAX effect type then load the
* reverb properties. */
alEffecti(effect, AL_EFFECT_TYPE, AL_EFFECT_EAXREVERB);
alEffectf(effect, AL_EAXREVERB_DENSITY, reverb->flDensity);
alEffectf(effect, AL_EAXREVERB_DIFFUSION, reverb->flDiffusion);
alEffectf(effect, AL_EAXREVERB_GAIN, reverb->flGain);
alEffectf(effect, AL_EAXREVERB_GAINHF, reverb->flGainHF);
alEffectf(effect, AL_EAXREVERB_GAINLF, reverb->flGainLF);
alEffectf(effect, AL_EAXREVERB_DECAY_TIME, reverb->flDecayTime);
alEffectf(effect, AL_EAXREVERB_DECAY_HFRATIO, reverb->flDecayHFRatio);
alEffectf(effect, AL_EAXREVERB_DECAY_LFRATIO, reverb->flDecayLFRatio);
alEffectf(effect, AL_EAXREVERB_REFLECTIONS_GAIN, reverb->flReflectionsGain);
alEffectf(effect, AL_EAXREVERB_REFLECTIONS_DELAY, reverb->flReflectionsDelay);
alEffectfv(effect, AL_EAXREVERB_REFLECTIONS_PAN, reverb->flReflectionsPan);
alEffectf(effect, AL_EAXREVERB_LATE_REVERB_GAIN, reverb->flLateReverbGain);
alEffectf(effect, AL_EAXREVERB_LATE_REVERB_DELAY, reverb->flLateReverbDelay);
alEffectfv(effect, AL_EAXREVERB_LATE_REVERB_PAN, reverb->flLateReverbPan);
alEffectf(effect, AL_EAXREVERB_ECHO_TIME, reverb->flEchoTime);
alEffectf(effect, AL_EAXREVERB_ECHO_DEPTH, reverb->flEchoDepth);
alEffectf(effect, AL_EAXREVERB_MODULATION_TIME, reverb->flModulationTime);
alEffectf(effect, AL_EAXREVERB_MODULATION_DEPTH, reverb->flModulationDepth);
alEffectf(effect, AL_EAXREVERB_AIR_ABSORPTION_GAINHF, reverb->flAirAbsorptionGainHF);
alEffectf(effect, AL_EAXREVERB_HFREFERENCE, reverb->flHFReference);
alEffectf(effect, AL_EAXREVERB_LFREFERENCE, reverb->flLFReference);
alEffectf(effect, AL_EAXREVERB_ROOM_ROLLOFF_FACTOR, reverb->flRoomRolloffFactor);
alEffecti(effect, AL_EAXREVERB_DECAY_HFLIMIT, reverb->iDecayHFLimit);
}
else
{
printf("Using Standard Reverb\n");
/* No EAX Reverb. Set the standard reverb effect type then load the
* available reverb properties. */
alEffecti(effect, AL_EFFECT_TYPE, AL_EFFECT_REVERB);
alEffectf(effect, AL_REVERB_DENSITY, reverb->flDensity);
alEffectf(effect, AL_REVERB_DIFFUSION, reverb->flDiffusion);
alEffectf(effect, AL_REVERB_GAIN, reverb->flGain);
alEffectf(effect, AL_REVERB_GAINHF, reverb->flGainHF);
alEffectf(effect, AL_REVERB_DECAY_TIME, reverb->flDecayTime);
alEffectf(effect, AL_REVERB_DECAY_HFRATIO, reverb->flDecayHFRatio);
alEffectf(effect, AL_REVERB_REFLECTIONS_GAIN, reverb->flReflectionsGain);
alEffectf(effect, AL_REVERB_REFLECTIONS_DELAY, reverb->flReflectionsDelay);
alEffectf(effect, AL_REVERB_LATE_REVERB_GAIN, reverb->flLateReverbGain);
alEffectf(effect, AL_REVERB_LATE_REVERB_DELAY, reverb->flLateReverbDelay);
alEffectf(effect, AL_REVERB_AIR_ABSORPTION_GAINHF, reverb->flAirAbsorptionGainHF);
alEffectf(effect, AL_REVERB_ROOM_ROLLOFF_FACTOR, reverb->flRoomRolloffFactor);
alEffecti(effect, AL_REVERB_DECAY_HFLIMIT, reverb->iDecayHFLimit);
}
/* Check if an error occured, and clean up if so. */
err = alGetError();
if(err != AL_NO_ERROR)
{
fprintf(stderr, "OpenAL error: %s\n", alGetString(err));
if(alIsEffect(effect))
alDeleteEffects(1, &effect);
return 0;
}
return effect;
}
/* LoadBuffer loads the named audio file into an OpenAL buffer object, and
* returns the new buffer ID. */
static ALuint LoadSound(const char *filename)
{
ALenum err, format, type, channels;
ALuint rate, buffer;
size_t datalen;
void *data;
FilePtr audiofile;
StreamPtr sound;
/* Open the file and get the first stream from it */
audiofile = openAVFile(filename);
sound = getAVAudioStream(audiofile, 0);
if(!sound)
{
fprintf(stderr, "Could not open audio in %s\n", filename);
closeAVFile(audiofile);
return 0;
}
/* Get the sound format, and figure out the OpenAL format */
if(getAVAudioInfo(sound, &rate, &channels, &type) != 0)
{
fprintf(stderr, "Error getting audio info for %s\n", filename);
closeAVFile(audiofile);
return 0;
}
format = GetFormat(channels, type, alIsBufferFormatSupportedSOFT);
if(format == AL_NONE)
{
fprintf(stderr, "Unsupported format (%s, %s) for %s\n",
ChannelsName(channels), TypeName(type), filename);
closeAVFile(audiofile);
return 0;
}
/* Decode the whole audio stream to a buffer. */
data = decodeAVAudioStream(sound, &datalen);
if(!data)
{
fprintf(stderr, "Failed to read audio from %s\n", filename);
closeAVFile(audiofile);
return 0;
}
/* Buffer the audio data into a new buffer object, then free the data and
* close the file. */
buffer = 0;
alGenBuffers(1, &buffer);
alBufferSamplesSOFT(buffer, rate, format, BytesToFrames(datalen, channels, type),
channels, type, data);
free(data);
closeAVFile(audiofile);
/* Check if an error occured, and clean up if so. */
err = alGetError();
if(err != AL_NO_ERROR)
{
fprintf(stderr, "OpenAL Error: %s\n", alGetString(err));
if(alIsBuffer(buffer))
alDeleteBuffers(1, &buffer);
return 0;
}
return buffer;
}
int main(int argc, char **argv)
{
EFXEAXREVERBPROPERTIES reverb = EFX_REVERB_PRESET_GENERIC;
ALuint source, buffer, effect, slot;
ALenum state;
/* Print out usage if no file was specified */
if(argc < 2)
{
fprintf(stderr, "Usage: %s <filename>\n", argv[0]);
return 1;
}
/* Initialize OpenAL with the default device, and check for EFX support. */
if(InitAL() != 0)
return 1;
if(!alcIsExtensionPresent(alcGetContextsDevice(alcGetCurrentContext()), "ALC_EXT_EFX"))
{
fprintf(stderr, "Error: EFX not supported\n");
CloseAL();
return 1;
}
/* Define a macro to help load the function pointers. */
#define LOAD_PROC(x) ((x) = alGetProcAddress(#x))
LOAD_PROC(alGenEffects);
LOAD_PROC(alDeleteEffects);
LOAD_PROC(alIsEffect);
LOAD_PROC(alEffecti);
LOAD_PROC(alEffectiv);
LOAD_PROC(alEffectf);
LOAD_PROC(alEffectfv);
LOAD_PROC(alGetEffecti);
LOAD_PROC(alGetEffectiv);
LOAD_PROC(alGetEffectf);
LOAD_PROC(alGetEffectfv);
LOAD_PROC(alGenAuxiliaryEffectSlots);
LOAD_PROC(alDeleteAuxiliaryEffectSlots);
LOAD_PROC(alIsAuxiliaryEffectSlot);
LOAD_PROC(alAuxiliaryEffectSloti);
LOAD_PROC(alAuxiliaryEffectSlotiv);
LOAD_PROC(alAuxiliaryEffectSlotf);
LOAD_PROC(alAuxiliaryEffectSlotfv);
LOAD_PROC(alGetAuxiliaryEffectSloti);
LOAD_PROC(alGetAuxiliaryEffectSlotiv);
LOAD_PROC(alGetAuxiliaryEffectSlotf);
LOAD_PROC(alGetAuxiliaryEffectSlotfv);
if(alIsExtensionPresent("AL_SOFT_buffer_samples"))
{
LOAD_PROC(alBufferSamplesSOFT);
LOAD_PROC(alIsBufferFormatSupportedSOFT);
}
#undef LOAD_PROC
/* Load the sound into a buffer. */
buffer = LoadSound(argv[1]);
if(!buffer)
{
CloseAL();
return 1;
}
/* Load the reverb into an effect. */
effect = LoadEffect(&reverb);
if(!effect)
{
alDeleteBuffers(1, &buffer);
CloseAL();
return 1;
}
/* Create the effect slot object. This is what "plays" an effect on sources
* that connect to it. */
slot = 0;
alGenAuxiliaryEffectSlots(1, &slot);
/* Tell the effect slot to use the loaded effect object. Note that the this
* effectively copies the effect properties. You can modify or delete the
* effect object afterward without affecting the effect slot.
*/
alAuxiliaryEffectSloti(slot, AL_EFFECTSLOT_EFFECT, effect);
assert(alGetError()==AL_NO_ERROR && "Failed to set effect slot");
/* Create the source to play the sound with. */
source = 0;
alGenSources(1, &source);
alSourcei(source, AL_BUFFER, buffer);
/* Connect the source to the effect slot. This tells the source to use the
* effect slot 'slot', on send #0 with the AL_FILTER_NULL filter object.
*/
alSource3i(source, AL_AUXILIARY_SEND_FILTER, slot, 0, AL_FILTER_NULL);
assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
/* Play the sound until it finishes. */
alSourcePlay(source);
do {
Sleep(10);
alGetSourcei(source, AL_SOURCE_STATE, &state);
} while(alGetError() == AL_NO_ERROR && state == AL_PLAYING);
/* All done. Delete resources, and close OpenAL. */
alDeleteSources(1, &source);
alDeleteAuxiliaryEffectSlots(1, &slot);
alDeleteEffects(1, &effect);
alDeleteBuffers(1, &buffer);
CloseAL();
return 0;
}
+347
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@@ -0,0 +1,347 @@
/*
* OpenAL Audio Stream Example
*
* Copyright (c) 2011 by Chris Robinson <chris.kcat@gmail.com>
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/* This file contains a relatively simple streaming audio player. */
#include <string.h>
#include <stdlib.h>
#include <stdio.h>
#include <signal.h>
#include <assert.h>
#include "AL/al.h"
#include "AL/alc.h"
#include "AL/alext.h"
#include "common/alhelpers.h"
#include "common/alffmpeg.h"
LPALBUFFERSAMPLESSOFT alBufferSamplesSOFT = wrap_BufferSamples;
LPALISBUFFERFORMATSUPPORTEDSOFT alIsBufferFormatSupportedSOFT = NULL;
/* Define the number of buffers and buffer size (in samples) to use. 4 buffers
* with 8192 samples each gives a nice per-chunk size, and lets the queue last
* for almost 3/4ths of a second for a 44.1khz stream. */
#define NUM_BUFFERS 4
#define BUFFER_SIZE 8192
typedef struct StreamPlayer {
/* These are the buffers and source to play out through OpenAL with */
ALuint buffers[NUM_BUFFERS];
ALuint source;
/* Handles for the audio stream */
FilePtr file;
StreamPtr stream;
/* A temporary data buffer for readAVAudioData to write to and pass to
* OpenAL with */
ALbyte *data;
ALsizei datasize;
/* The format of the output stream */
ALenum format;
ALenum channels;
ALenum type;
ALuint rate;
} StreamPlayer;
static StreamPlayer *NewPlayer(void);
static void DeletePlayer(StreamPlayer *player);
static int OpenPlayerFile(StreamPlayer *player, const char *filename);
static void ClosePlayerFile(StreamPlayer *player);
static int StartPlayer(StreamPlayer *player);
static int UpdatePlayer(StreamPlayer *player);
/* Creates a new player object, and allocates the needed OpenAL source and
* buffer objects. Error checking is simplified for the purposes of this
* example, and will cause an abort if needed. */
static StreamPlayer *NewPlayer(void)
{
StreamPlayer *player;
player = malloc(sizeof(*player));
assert(player != NULL);
memset(player, 0, sizeof(*player));
/* Generate the buffers and source */
alGenBuffers(NUM_BUFFERS, player->buffers);
assert(alGetError() == AL_NO_ERROR && "Could not create buffers");
alGenSources(1, &player->source);
assert(alGetError() == AL_NO_ERROR && "Could not create source");
/* Set parameters so mono sources play out the front-center speaker and
* won't distance attenuate. */
alSource3i(player->source, AL_POSITION, 0, 0, -1);
alSourcei(player->source, AL_SOURCE_RELATIVE, AL_TRUE);
alSourcei(player->source, AL_ROLLOFF_FACTOR, 0);
assert(alGetError() == AL_NO_ERROR && "Could not set source parameters");
return player;
}
/* Destroys a player object, deleting the source and buffers. No error handling
* since these calls shouldn't fail with a properly-made player object. */
static void DeletePlayer(StreamPlayer *player)
{
ClosePlayerFile(player);
alDeleteSources(1, &player->source);
alDeleteBuffers(NUM_BUFFERS, player->buffers);
if(alGetError() != AL_NO_ERROR)
fprintf(stderr, "Failed to delete object IDs\n");
memset(player, 0, sizeof(*player));
free(player);
}
/* Opens the first audio stream of the named file. If a file is already open,
* it will be closed first. */
static int OpenPlayerFile(StreamPlayer *player, const char *filename)
{
ClosePlayerFile(player);
/* Open the file and get the first stream from it */
player->file = openAVFile(filename);
player->stream = getAVAudioStream(player->file, 0);
if(!player->stream)
{
fprintf(stderr, "Could not open audio in %s\n", filename);
goto error;
}
/* Get the stream format, and figure out the OpenAL format */
if(getAVAudioInfo(player->stream, &player->rate, &player->channels,
&player->type) != 0)
{
fprintf(stderr, "Error getting audio info for %s\n", filename);
goto error;
}
player->format = GetFormat(player->channels, player->type, alIsBufferFormatSupportedSOFT);
if(player->format == 0)
{
fprintf(stderr, "Unsupported format (%s, %s) for %s\n",
ChannelsName(player->channels), TypeName(player->type),
filename);
goto error;
}
/* Allocate enough space for the temp buffer, given the format */
player->datasize = FramesToBytes(BUFFER_SIZE, player->channels,
player->type);
player->data = malloc(player->datasize);
if(player->data == NULL)
{
fprintf(stderr, "Error allocating %d bytes\n", player->datasize);
goto error;
}
return 1;
error:
closeAVFile(player->file);
player->file = NULL;
player->stream = NULL;
player->datasize = 0;
return 0;
}
/* Closes the audio file stream */
static void ClosePlayerFile(StreamPlayer *player)
{
closeAVFile(player->file);
player->file = NULL;
player->stream = NULL;
free(player->data);
player->data = NULL;
player->datasize = 0;
}
/* Prebuffers some audio from the file, and starts playing the source */
static int StartPlayer(StreamPlayer *player)
{
size_t i, got;
/* Rewind the source position and clear the buffer queue */
alSourceRewind(player->source);
alSourcei(player->source, AL_BUFFER, 0);
/* Fill the buffer queue */
for(i = 0;i < NUM_BUFFERS;i++)
{
/* Get some data to give it to the buffer */
got = readAVAudioData(player->stream, player->data, player->datasize);
if(got == 0) break;
alBufferSamplesSOFT(player->buffers[i], player->rate, player->format,
BytesToFrames(got, player->channels, player->type),
player->channels, player->type, player->data);
}
if(alGetError() != AL_NO_ERROR)
{
fprintf(stderr, "Error buffering for playback\n");
return 0;
}
/* Now queue and start playback! */
alSourceQueueBuffers(player->source, i, player->buffers);
alSourcePlay(player->source);
if(alGetError() != AL_NO_ERROR)
{
fprintf(stderr, "Error starting playback\n");
return 0;
}
return 1;
}
static int UpdatePlayer(StreamPlayer *player)
{
ALint processed, state;
/* Get relevant source info */
alGetSourcei(player->source, AL_SOURCE_STATE, &state);
alGetSourcei(player->source, AL_BUFFERS_PROCESSED, &processed);
if(alGetError() != AL_NO_ERROR)
{
fprintf(stderr, "Error checking source state\n");
return 0;
}
/* Unqueue and handle each processed buffer */
while(processed > 0)
{
ALuint bufid;
size_t got;
alSourceUnqueueBuffers(player->source, 1, &bufid);
processed--;
/* Read the next chunk of data, refill the buffer, and queue it
* back on the source */
got = readAVAudioData(player->stream, player->data, player->datasize);
if(got > 0)
{
alBufferSamplesSOFT(bufid, player->rate, player->format,
BytesToFrames(got, player->channels, player->type),
player->channels, player->type, player->data);
alSourceQueueBuffers(player->source, 1, &bufid);
}
if(alGetError() != AL_NO_ERROR)
{
fprintf(stderr, "Error buffering data\n");
return 0;
}
}
/* Make sure the source hasn't underrun */
if(state != AL_PLAYING && state != AL_PAUSED)
{
ALint queued;
/* If no buffers are queued, playback is finished */
alGetSourcei(player->source, AL_BUFFERS_QUEUED, &queued);
if(queued == 0)
return 0;
alSourcePlay(player->source);
if(alGetError() != AL_NO_ERROR)
{
fprintf(stderr, "Error restarting playback\n");
return 0;
}
}
return 1;
}
int main(int argc, char **argv)
{
StreamPlayer *player;
int i;
/* Print out usage if no file was specified */
if(argc < 2)
{
fprintf(stderr, "Usage: %s <filenames...>\n", argv[0]);
return 1;
}
if(InitAL() != 0)
return 1;
if(alIsExtensionPresent("AL_SOFT_buffer_samples"))
{
printf("AL_SOFT_buffer_samples supported!\n");
alBufferSamplesSOFT = alGetProcAddress("alBufferSamplesSOFT");
alIsBufferFormatSupportedSOFT = alGetProcAddress("alIsBufferFormatSupportedSOFT");
}
else
printf("AL_SOFT_buffer_samples not supported\n");
player = NewPlayer();
/* Play each file listed on the command line */
for(i = 1;i < argc;i++)
{
if(!OpenPlayerFile(player, argv[i]))
continue;
printf("Playing %s (%s, %s, %dhz)\n", argv[i],
TypeName(player->type), ChannelsName(player->channels),
player->rate);
fflush(stdout);
if(!StartPlayer(player))
{
ClosePlayerFile(player);
continue;
}
while(UpdatePlayer(player))
Sleep(10);
/* All done with this file. Close it and go to the next */
ClosePlayerFile(player);
}
printf("Done.\n");
/* All files done. Delete the player, and close OpenAL */
DeletePlayer(player);
player = NULL;
CloseAL();
return 0;
}
+638
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@@ -0,0 +1,638 @@
/*
* FFmpeg Decoder Helpers
*
* Copyright (c) 2011 by Chris Robinson <chris.kcat@gmail.com>
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/* This file contains routines for helping to decode audio using libavformat
* and libavcodec (ffmpeg). There's very little OpenAL-specific code here. */
#include <string.h>
#include <stdlib.h>
#include <stdio.h>
#include <signal.h>
#include <assert.h>
#include "AL/al.h"
#include "AL/alc.h"
#include "AL/alext.h"
#include "alhelpers.h"
#include "alffmpeg.h"
static size_t NextPowerOf2(size_t value)
{
size_t powerOf2 = 1;
if(value)
{
value--;
while(value)
{
value >>= 1;
powerOf2 <<= 1;
}
}
return powerOf2;
}
struct MemData {
char *buffer;
size_t length;
size_t pos;
};
static int MemData_read(void *opaque, uint8_t *buf, int buf_size)
{
struct MemData *membuf = (struct MemData*)opaque;
int rem = membuf->length - membuf->pos;
if(rem > buf_size)
rem = buf_size;
memcpy(buf, &membuf->buffer[membuf->pos], rem);
membuf->pos += rem;
return rem;
}
static int MemData_write(void *opaque, uint8_t *buf, int buf_size)
{
struct MemData *membuf = (struct MemData*)opaque;
int rem = membuf->length - membuf->pos;
if(rem > buf_size)
rem = buf_size;
memcpy(&membuf->buffer[membuf->pos], buf, rem);
membuf->pos += rem;
return rem;
}
static int64_t MemData_seek(void *opaque, int64_t offset, int whence)
{
struct MemData *membuf = (struct MemData*)opaque;
whence &= ~AVSEEK_FORCE;
switch(whence)
{
case SEEK_SET:
if(offset < 0 || (uint64_t)offset > membuf->length)
return -1;
membuf->pos = offset;
break;
case SEEK_CUR:
if((offset >= 0 && (uint64_t)offset > membuf->length-membuf->pos) ||
(offset < 0 && (uint64_t)(-offset) > membuf->pos))
return -1;
membuf->pos += offset;
break;
case SEEK_END:
if(offset > 0 || (uint64_t)(-offset) > membuf->length)
return -1;
membuf->pos = membuf->length + offset;
break;
case AVSEEK_SIZE:
return membuf->length;
default:
return -1;
}
return membuf->pos;
}
struct PacketList {
AVPacket pkt;
struct PacketList *next;
};
struct MyStream {
AVCodecContext *CodecCtx;
int StreamIdx;
struct PacketList *Packets;
AVFrame *Frame;
const uint8_t *FrameData;
size_t FrameDataSize;
FilePtr parent;
};
struct MyFile {
AVFormatContext *FmtCtx;
StreamPtr *Streams;
size_t StreamsSize;
struct MemData membuf;
};
static int done_init = 0;
FilePtr openAVFile(const char *fname)
{
FilePtr file;
/* We need to make sure ffmpeg is initialized. Optionally silence warning
* output from the lib */
if(!done_init) {av_register_all();
av_log_set_level(AV_LOG_ERROR);
done_init = 1;}
file = (FilePtr)calloc(1, sizeof(*file));
if(file && avformat_open_input(&file->FmtCtx, fname, NULL, NULL) == 0)
{
/* After opening, we must search for the stream information because not
* all formats will have it in stream headers */
if(avformat_find_stream_info(file->FmtCtx, NULL) >= 0)
return file;
avformat_close_input(&file->FmtCtx);
}
free(file);
return NULL;
}
FilePtr openAVData(const char *name, char *buffer, size_t buffer_len)
{
FilePtr file;
if(!done_init) {av_register_all();
av_log_set_level(AV_LOG_ERROR);
done_init = 1;}
if(!name)
name = "";
file = (FilePtr)calloc(1, sizeof(*file));
if(file && (file->FmtCtx=avformat_alloc_context()) != NULL)
{
file->membuf.buffer = buffer;
file->membuf.length = buffer_len;
file->membuf.pos = 0;
file->FmtCtx->pb = avio_alloc_context(NULL, 0, 0, &file->membuf,
MemData_read, MemData_write,
MemData_seek);
if(file->FmtCtx->pb && avformat_open_input(&file->FmtCtx, name, NULL, NULL) == 0)
{
if(avformat_find_stream_info(file->FmtCtx, NULL) >= 0)
return file;
avformat_close_input(&file->FmtCtx);
}
if(file->FmtCtx)
avformat_free_context(file->FmtCtx);
file->FmtCtx = NULL;
}
free(file);
return NULL;
}
FilePtr openAVCustom(const char *name, void *user_data,
int (*read_packet)(void *user_data, uint8_t *buf, int buf_size),
int (*write_packet)(void *user_data, uint8_t *buf, int buf_size),
int64_t (*seek)(void *user_data, int64_t offset, int whence))
{
FilePtr file;
if(!done_init) {av_register_all();
av_log_set_level(AV_LOG_ERROR);
done_init = 1;}
if(!name)
name = "";
file = (FilePtr)calloc(1, sizeof(*file));
if(file && (file->FmtCtx=avformat_alloc_context()) != NULL)
{
file->FmtCtx->pb = avio_alloc_context(NULL, 0, 0, user_data,
read_packet, write_packet, seek);
if(file->FmtCtx->pb && avformat_open_input(&file->FmtCtx, name, NULL, NULL) == 0)
{
if(avformat_find_stream_info(file->FmtCtx, NULL) >= 0)
return file;
avformat_close_input(&file->FmtCtx);
}
if(file->FmtCtx)
avformat_free_context(file->FmtCtx);
file->FmtCtx = NULL;
}
free(file);
return NULL;
}
void clearAVAudioData(StreamPtr stream)
{
while(stream->Packets)
{
struct PacketList *self;
self = stream->Packets;
stream->Packets = self->next;
av_free_packet(&self->pkt);
av_free(self);
}
}
void closeAVFile(FilePtr file)
{
size_t i;
if(!file) return;
for(i = 0;i < file->StreamsSize;i++)
{
StreamPtr stream = file->Streams[i];
while(stream->Packets)
{
struct PacketList *self;
self = stream->Packets;
stream->Packets = self->next;
av_free_packet(&self->pkt);
av_free(self);
}
avcodec_close(stream->CodecCtx);
av_free(stream->Frame);
free(stream);
}
free(file->Streams);
avformat_close_input(&file->FmtCtx);
free(file);
}
int getAVFileInfo(FilePtr file, int *numaudiostreams)
{
unsigned int i;
int audiocount = 0;
if(!file) return 1;
for(i = 0;i < file->FmtCtx->nb_streams;i++)
{
if(file->FmtCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO)
audiocount++;
}
*numaudiostreams = audiocount;
return 0;
}
StreamPtr getAVAudioStream(FilePtr file, int streamnum)
{
unsigned int i;
if(!file) return NULL;
for(i = 0;i < file->FmtCtx->nb_streams;i++)
{
if(file->FmtCtx->streams[i]->codec->codec_type != AVMEDIA_TYPE_AUDIO)
continue;
if(streamnum == 0)
{
StreamPtr stream;
AVCodec *codec;
void *temp;
size_t j;
/* Found the requested stream. Check if a handle to this stream
* already exists and return it if it does */
for(j = 0;j < file->StreamsSize;j++)
{
if(file->Streams[j]->StreamIdx == (int)i)
return file->Streams[j];
}
/* Doesn't yet exist. Now allocate a new stream object and fill in
* its info */
stream = (StreamPtr)calloc(1, sizeof(*stream));
if(!stream) return NULL;
stream->parent = file;
stream->CodecCtx = file->FmtCtx->streams[i]->codec;
stream->StreamIdx = i;
/* Try to find the codec for the given codec ID, and open it */
codec = avcodec_find_decoder(stream->CodecCtx->codec_id);
if(!codec || avcodec_open2(stream->CodecCtx, codec, NULL) < 0)
{
free(stream);
return NULL;
}
/* Allocate space for the decoded data to be stored in before it
* gets passed to the app */
stream->Frame = avcodec_alloc_frame();
if(!stream->Frame)
{
avcodec_close(stream->CodecCtx);
free(stream);
return NULL;
}
stream->FrameData = NULL;
stream->FrameDataSize = 0;
/* Append the new stream object to the stream list. The original
* pointer will remain valid if realloc fails, so we need to use
* another pointer to watch for errors and not leak memory */
temp = realloc(file->Streams, (file->StreamsSize+1) *
sizeof(*file->Streams));
if(!temp)
{
avcodec_close(stream->CodecCtx);
av_free(stream->Frame);
free(stream);
return NULL;
}
file->Streams = (StreamPtr*)temp;
file->Streams[file->StreamsSize++] = stream;
return stream;
}
streamnum--;
}
return NULL;
}
int getAVAudioInfo(StreamPtr stream, ALuint *rate, ALenum *channels, ALenum *type)
{
if(!stream || stream->CodecCtx->codec_type != AVMEDIA_TYPE_AUDIO)
return 1;
/* Get the sample type for OpenAL given the format detected by ffmpeg. */
if(stream->CodecCtx->sample_fmt == AV_SAMPLE_FMT_U8)
*type = AL_UNSIGNED_BYTE_SOFT;
else if(stream->CodecCtx->sample_fmt == AV_SAMPLE_FMT_S16)
*type = AL_SHORT_SOFT;
else if(stream->CodecCtx->sample_fmt == AV_SAMPLE_FMT_S32)
*type = AL_INT_SOFT;
else if(stream->CodecCtx->sample_fmt == AV_SAMPLE_FMT_FLT)
*type = AL_FLOAT_SOFT;
else if(stream->CodecCtx->sample_fmt == AV_SAMPLE_FMT_DBL)
*type = AL_DOUBLE_SOFT;
else
{
fprintf(stderr, "Unsupported ffmpeg sample format: %s\n",
av_get_sample_fmt_name(stream->CodecCtx->sample_fmt));
return 1;
}
/* Get the OpenAL channel configuration using the channel layout detected
* by ffmpeg. NOTE: some file types may not specify a channel layout. In
* that case, one must be guessed based on the channel count. */
if(stream->CodecCtx->channel_layout == AV_CH_LAYOUT_MONO)
*channels = AL_MONO_SOFT;
else if(stream->CodecCtx->channel_layout == AV_CH_LAYOUT_STEREO)
*channels = AL_STEREO_SOFT;
else if(stream->CodecCtx->channel_layout == AV_CH_LAYOUT_QUAD)
*channels = AL_QUAD_SOFT;
else if(stream->CodecCtx->channel_layout == AV_CH_LAYOUT_5POINT1_BACK)
*channels = AL_5POINT1_SOFT;
else if(stream->CodecCtx->channel_layout == AV_CH_LAYOUT_7POINT1)
*channels = AL_7POINT1_SOFT;
else if(stream->CodecCtx->channel_layout == 0)
{
/* Unknown channel layout. Try to guess. */
if(stream->CodecCtx->channels == 1)
*channels = AL_MONO_SOFT;
else if(stream->CodecCtx->channels == 2)
*channels = AL_STEREO_SOFT;
else
{
fprintf(stderr, "Unsupported ffmpeg raw channel count: %d\n",
stream->CodecCtx->channels);
return 1;
}
}
else
{
char str[1024];
av_get_channel_layout_string(str, sizeof(str), stream->CodecCtx->channels,
stream->CodecCtx->channel_layout);
fprintf(stderr, "Unsupported ffmpeg channel layout: %s\n", str);
return 1;
}
*rate = stream->CodecCtx->sample_rate;
return 0;
}
/* Used by getAV*Data to search for more compressed data, and buffer it in the
* correct stream. It won't buffer data for streams that the app doesn't have a
* handle for. */
static int getNextPacket(FilePtr file, int streamidx)
{
struct PacketList *packet;
packet = (struct PacketList*)av_malloc(sizeof(*packet));
packet->next = NULL;
next_packet:
while(av_read_frame(file->FmtCtx, &packet->pkt) >= 0)
{
StreamPtr *iter = file->Streams;
StreamPtr *iter_end = iter + file->StreamsSize;
/* Check each stream the user has a handle for, looking for the one
* this packet belongs to */
while(iter != iter_end)
{
if((*iter)->StreamIdx == packet->pkt.stream_index)
{
struct PacketList **last;
last = &(*iter)->Packets;
while(*last != NULL)
last = &(*last)->next;
*last = packet;
if((*iter)->StreamIdx == streamidx)
return 1;
packet = (struct PacketList*)av_malloc(sizeof(*packet));
packet->next = NULL;
goto next_packet;
}
iter++;
}
/* Free the packet and look for another */
av_free_packet(&packet->pkt);
}
av_free(packet);
return 0;
}
uint8_t *getAVAudioData(StreamPtr stream, size_t *length)
{
int got_frame;
int len;
if(length) *length = 0;
if(!stream || stream->CodecCtx->codec_type != AVMEDIA_TYPE_AUDIO)
return NULL;
next_packet:
if(!stream->Packets && !getNextPacket(stream->parent, stream->StreamIdx))
return NULL;
/* Decode some data, and check for errors */
avcodec_get_frame_defaults(stream->Frame);
while((len=avcodec_decode_audio4(stream->CodecCtx, stream->Frame,
&got_frame, &stream->Packets->pkt)) < 0)
{
struct PacketList *self;
/* Error? Drop it and try the next, I guess... */
self = stream->Packets;
stream->Packets = self->next;
av_free_packet(&self->pkt);
av_free(self);
if(!stream->Packets)
goto next_packet;
}
if(len < stream->Packets->pkt.size)
{
/* Move the unread data to the front and clear the end bits */
int remaining = stream->Packets->pkt.size - len;
memmove(stream->Packets->pkt.data, &stream->Packets->pkt.data[len],
remaining);
memset(&stream->Packets->pkt.data[remaining], 0,
stream->Packets->pkt.size - remaining);
stream->Packets->pkt.size -= len;
}
else
{
struct PacketList *self;
self = stream->Packets;
stream->Packets = self->next;
av_free_packet(&self->pkt);
av_free(self);
}
if(!got_frame || stream->Frame->nb_samples == 0)
goto next_packet;
/* Set the output buffer size */
*length = av_samples_get_buffer_size(NULL, stream->CodecCtx->channels,
stream->Frame->nb_samples,
stream->CodecCtx->sample_fmt, 1);
return stream->Frame->data[0];
}
size_t readAVAudioData(StreamPtr stream, void *data, size_t length)
{
size_t dec = 0;
if(!stream || stream->CodecCtx->codec_type != AVMEDIA_TYPE_AUDIO)
return 0;
while(dec < length)
{
/* If there's no decoded data, find some */
if(stream->FrameDataSize == 0)
{
stream->FrameData = getAVAudioData(stream, &stream->FrameDataSize);
if(!stream->FrameData)
break;
}
if(stream->FrameDataSize > 0)
{
/* Get the amount of bytes remaining to be written, and clamp to
* the amount of decoded data we have */
size_t rem = length-dec;
if(rem > stream->FrameDataSize)
rem = stream->FrameDataSize;
/* Copy the data to the app's buffer and increment */
if(data != NULL)
{
memcpy(data, stream->FrameData, rem);
data = (char*)data + rem;
}
dec += rem;
/* If there's any decoded data left, move it to the front of the
* buffer for next time */
stream->FrameData += rem;
stream->FrameDataSize -= rem;
}
}
/* Return the number of bytes we were able to get */
return dec;
}
void *decodeAVAudioStream(StreamPtr stream, size_t *length)
{
char *outbuf = NULL;
size_t buflen = 0;
void *inbuf;
size_t got;
*length = 0;
if(!stream || stream->CodecCtx->codec_type != AVMEDIA_TYPE_AUDIO)
return NULL;
while((inbuf=getAVAudioData(stream, &got)) != NULL && got > 0)
{
void *ptr;
ptr = realloc(outbuf, NextPowerOf2(buflen+got));
if(ptr == NULL)
break;
outbuf = (char*)ptr;
memcpy(&outbuf[buflen], inbuf, got);
buflen += got;
}
outbuf = (char*)realloc(outbuf, buflen);
*length = buflen;
return outbuf;
}
+66
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#ifndef ALFFMPEG_H
#define ALFFMPEG_H
#ifdef __cplusplus
extern "C" {
#endif /* __cplusplus */
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
/* Opaque handles to files and streams. Apps don't need to concern themselves
* with the internals */
typedef struct MyFile *FilePtr;
typedef struct MyStream *StreamPtr;
/* Opens a file with ffmpeg and sets up the streams' information */
FilePtr openAVFile(const char *fname);
/* Opens a named file image with ffmpeg and sets up the streams' information */
FilePtr openAVData(const char *name, char *buffer, size_t buffer_len);
/* Opens a named data stream with ffmpeg, using the specified data pointer and
* callbacks, and sets up the streams' information */
FilePtr openAVCustom(const char *name, void *user_data,
int (*read_packet)(void *user_data, uint8_t *buf, int buf_size),
int (*write_packet)(void *user_data, uint8_t *buf, int buf_size),
int64_t (*seek)(void *user_data, int64_t offset, int whence));
/* Closes/frees an opened file and any of its streams */
void closeAVFile(FilePtr file);
/* Reports certain information from the file, eg, the number of audio
* streams. Returns 0 on success. */
int getAVFileInfo(FilePtr file, int *numaudiostreams);
/* Retrieves a handle for the given audio stream number (generally 0, but some
* files can have multiple audio streams in one file). */
StreamPtr getAVAudioStream(FilePtr file, int streamnum);
/* Returns information about the given audio stream. Returns 0 on success. */
int getAVAudioInfo(StreamPtr stream, ALuint *rate, ALenum *channels, ALenum *type);
/* Returns a pointer to the next available packet of decoded audio. Any data
* from a previously-decoded packet is dropped. The size (in bytes) of the
* returned data buffer is stored in 'length', and the returned pointer is only
* valid until the next call to getAVAudioData or readAVAudioData. */
uint8_t *getAVAudioData(StreamPtr stream, size_t *length);
/* The "meat" function. Decodes audio and writes, at most, length bytes into
* the provided data buffer. Will only return less for end-of-stream or error
* conditions. Returns the number of bytes written. */
size_t readAVAudioData(StreamPtr stream, void *data, size_t length);
/* Decodes all remaining data from the stream and returns a buffer containing
* the audio data, with the size stored in 'length'. The returned pointer must
* be freed with a call to free(). Note that since this decodes the whole
* stream, using it on lengthy streams (eg, music) will use a lot of memory.
* Such streams are better handled using getAVAudioData or readAVAudioData to
* keep smaller chunks in memory at any given time. */
void *decodeAVAudioStream(StreamPtr stream, size_t *length);
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif /* ALFFMPEG_H */
+327
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/*
* OpenAL Helpers
*
* Copyright (c) 2011 by Chris Robinson <chris.kcat@gmail.com>
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/* This file contains routines to help with some menial OpenAL-related tasks,
* such as opening a device and setting up a context, closing the device and
* destroying its context, converting between frame counts and byte lengths,
* finding an appropriate buffer format, and getting readable strings for
* channel configs and sample types. */
#include <stdio.h>
#include "AL/al.h"
#include "AL/alc.h"
#include "AL/alext.h"
#include "alhelpers.h"
/* InitAL opens the default device and sets up a context using default
* attributes, making the program ready to call OpenAL functions. */
int InitAL(void)
{
ALCdevice *device;
ALCcontext *ctx;
/* Open and initialize a device with default settings */
device = alcOpenDevice(NULL);
if(!device)
{
fprintf(stderr, "Could not open a device!\n");
return 1;
}
ctx = alcCreateContext(device, NULL);
if(ctx == NULL || alcMakeContextCurrent(ctx) == ALC_FALSE)
{
if(ctx != NULL)
alcDestroyContext(ctx);
alcCloseDevice(device);
fprintf(stderr, "Could not set a context!\n");
return 1;
}
printf("Opened \"%s\"\n", alcGetString(device, ALC_DEVICE_SPECIFIER));
return 0;
}
/* CloseAL closes the device belonging to the current context, and destroys the
* context. */
void CloseAL(void)
{
ALCdevice *device;
ALCcontext *ctx;
ctx = alcGetCurrentContext();
if(ctx == NULL)
return;
device = alcGetContextsDevice(ctx);
alcMakeContextCurrent(NULL);
alcDestroyContext(ctx);
alcCloseDevice(device);
}
/* GetFormat retrieves a compatible buffer format given the channel config and
* sample type. If an alIsBufferFormatSupportedSOFT-compatible function is
* provided, it will be called to find the closest-matching format from
* AL_SOFT_buffer_samples. Returns AL_NONE (0) if no supported format can be
* found. */
ALenum GetFormat(ALenum channels, ALenum type, LPALISBUFFERFORMATSUPPORTEDSOFT palIsBufferFormatSupportedSOFT)
{
ALenum format = AL_NONE;
/* If using AL_SOFT_buffer_samples, try looking through its formats */
if(palIsBufferFormatSupportedSOFT)
{
/* AL_SOFT_buffer_samples is more lenient with matching formats. The
* specified sample type does not need to match the returned format,
* but it is nice to try to get something close. */
if(type == AL_UNSIGNED_BYTE_SOFT || type == AL_BYTE_SOFT)
{
if(channels == AL_MONO_SOFT) format = AL_MONO8_SOFT;
else if(channels == AL_STEREO_SOFT) format = AL_STEREO8_SOFT;
else if(channels == AL_QUAD_SOFT) format = AL_QUAD8_SOFT;
else if(channels == AL_5POINT1_SOFT) format = AL_5POINT1_8_SOFT;
else if(channels == AL_6POINT1_SOFT) format = AL_6POINT1_8_SOFT;
else if(channels == AL_7POINT1_SOFT) format = AL_7POINT1_8_SOFT;
}
else if(type == AL_UNSIGNED_SHORT_SOFT || type == AL_SHORT_SOFT)
{
if(channels == AL_MONO_SOFT) format = AL_MONO16_SOFT;
else if(channels == AL_STEREO_SOFT) format = AL_STEREO16_SOFT;
else if(channels == AL_QUAD_SOFT) format = AL_QUAD16_SOFT;
else if(channels == AL_5POINT1_SOFT) format = AL_5POINT1_16_SOFT;
else if(channels == AL_6POINT1_SOFT) format = AL_6POINT1_16_SOFT;
else if(channels == AL_7POINT1_SOFT) format = AL_7POINT1_16_SOFT;
}
else if(type == AL_UNSIGNED_BYTE3_SOFT || type == AL_BYTE3_SOFT ||
type == AL_UNSIGNED_INT_SOFT || type == AL_INT_SOFT ||
type == AL_FLOAT_SOFT || type == AL_DOUBLE_SOFT)
{
if(channels == AL_MONO_SOFT) format = AL_MONO32F_SOFT;
else if(channels == AL_STEREO_SOFT) format = AL_STEREO32F_SOFT;
else if(channels == AL_QUAD_SOFT) format = AL_QUAD32F_SOFT;
else if(channels == AL_5POINT1_SOFT) format = AL_5POINT1_32F_SOFT;
else if(channels == AL_6POINT1_SOFT) format = AL_6POINT1_32F_SOFT;
else if(channels == AL_7POINT1_SOFT) format = AL_7POINT1_32F_SOFT;
}
if(format != AL_NONE && !palIsBufferFormatSupportedSOFT(format))
format = AL_NONE;
/* A matching format was not found or supported. Try 32-bit float. */
if(format == AL_NONE)
{
if(channels == AL_MONO_SOFT) format = AL_MONO32F_SOFT;
else if(channels == AL_STEREO_SOFT) format = AL_STEREO32F_SOFT;
else if(channels == AL_QUAD_SOFT) format = AL_QUAD32F_SOFT;
else if(channels == AL_5POINT1_SOFT) format = AL_5POINT1_32F_SOFT;
else if(channels == AL_6POINT1_SOFT) format = AL_6POINT1_32F_SOFT;
else if(channels == AL_7POINT1_SOFT) format = AL_7POINT1_32F_SOFT;
if(format != AL_NONE && !palIsBufferFormatSupportedSOFT(format))
format = AL_NONE;
}
/* 32-bit float not supported. Try 16-bit int. */
if(format == AL_NONE)
{
if(channels == AL_MONO_SOFT) format = AL_MONO16_SOFT;
else if(channels == AL_STEREO_SOFT) format = AL_STEREO16_SOFT;
else if(channels == AL_QUAD_SOFT) format = AL_QUAD16_SOFT;
else if(channels == AL_5POINT1_SOFT) format = AL_5POINT1_16_SOFT;
else if(channels == AL_6POINT1_SOFT) format = AL_6POINT1_16_SOFT;
else if(channels == AL_7POINT1_SOFT) format = AL_7POINT1_16_SOFT;
if(format != AL_NONE && !palIsBufferFormatSupportedSOFT(format))
format = AL_NONE;
}
/* 16-bit int not supported. Try 8-bit int. */
if(format == AL_NONE)
{
if(channels == AL_MONO_SOFT) format = AL_MONO8_SOFT;
else if(channels == AL_STEREO_SOFT) format = AL_STEREO8_SOFT;
else if(channels == AL_QUAD_SOFT) format = AL_QUAD8_SOFT;
else if(channels == AL_5POINT1_SOFT) format = AL_5POINT1_8_SOFT;
else if(channels == AL_6POINT1_SOFT) format = AL_6POINT1_8_SOFT;
else if(channels == AL_7POINT1_SOFT) format = AL_7POINT1_8_SOFT;
if(format != AL_NONE && !palIsBufferFormatSupportedSOFT(format))
format = AL_NONE;
}
return format;
}
/* We use the AL_EXT_MCFORMATS extension to provide output of Quad, 5.1,
* and 7.1 channel configs, AL_EXT_FLOAT32 for 32-bit float samples, and
* AL_EXT_DOUBLE for 64-bit float samples. */
if(type == AL_UNSIGNED_BYTE_SOFT)
{
if(channels == AL_MONO_SOFT)
format = AL_FORMAT_MONO8;
else if(channels == AL_STEREO_SOFT)
format = AL_FORMAT_STEREO8;
else if(alIsExtensionPresent("AL_EXT_MCFORMATS"))
{
if(channels == AL_QUAD_SOFT)
format = alGetEnumValue("AL_FORMAT_QUAD8");
else if(channels == AL_5POINT1_SOFT)
format = alGetEnumValue("AL_FORMAT_51CHN8");
else if(channels == AL_6POINT1_SOFT)
format = alGetEnumValue("AL_FORMAT_61CHN8");
else if(channels == AL_7POINT1_SOFT)
format = alGetEnumValue("AL_FORMAT_71CHN8");
}
}
else if(type == AL_SHORT_SOFT)
{
if(channels == AL_MONO_SOFT)
format = AL_FORMAT_MONO16;
else if(channels == AL_STEREO_SOFT)
format = AL_FORMAT_STEREO16;
else if(alIsExtensionPresent("AL_EXT_MCFORMATS"))
{
if(channels == AL_QUAD_SOFT)
format = alGetEnumValue("AL_FORMAT_QUAD16");
else if(channels == AL_5POINT1_SOFT)
format = alGetEnumValue("AL_FORMAT_51CHN16");
else if(channels == AL_6POINT1_SOFT)
format = alGetEnumValue("AL_FORMAT_61CHN16");
else if(channels == AL_7POINT1_SOFT)
format = alGetEnumValue("AL_FORMAT_71CHN16");
}
}
else if(type == AL_FLOAT_SOFT && alIsExtensionPresent("AL_EXT_FLOAT32"))
{
if(channels == AL_MONO_SOFT)
format = alGetEnumValue("AL_FORMAT_MONO_FLOAT32");
else if(channels == AL_STEREO_SOFT)
format = alGetEnumValue("AL_FORMAT_STEREO_FLOAT32");
else if(alIsExtensionPresent("AL_EXT_MCFORMATS"))
{
if(channels == AL_QUAD_SOFT)
format = alGetEnumValue("AL_FORMAT_QUAD32");
else if(channels == AL_5POINT1_SOFT)
format = alGetEnumValue("AL_FORMAT_51CHN32");
else if(channels == AL_6POINT1_SOFT)
format = alGetEnumValue("AL_FORMAT_61CHN32");
else if(channels == AL_7POINT1_SOFT)
format = alGetEnumValue("AL_FORMAT_71CHN32");
}
}
else if(type == AL_DOUBLE_SOFT && alIsExtensionPresent("AL_EXT_DOUBLE"))
{
if(channels == AL_MONO_SOFT)
format = alGetEnumValue("AL_FORMAT_MONO_DOUBLE");
else if(channels == AL_STEREO_SOFT)
format = alGetEnumValue("AL_FORMAT_STEREO_DOUBLE");
}
/* NOTE: It seems OSX returns -1 from alGetEnumValue for unknown enums, as
* opposed to 0. Correct it. */
if(format == -1)
format = 0;
return format;
}
void AL_APIENTRY wrap_BufferSamples(ALuint buffer, ALuint samplerate,
ALenum internalformat, ALsizei samples,
ALenum channels, ALenum type,
const ALvoid *data)
{
alBufferData(buffer, internalformat, data,
FramesToBytes(samples, channels, type),
samplerate);
}
const char *ChannelsName(ALenum chans)
{
switch(chans)
{
case AL_MONO_SOFT: return "Mono";
case AL_STEREO_SOFT: return "Stereo";
case AL_REAR_SOFT: return "Rear";
case AL_QUAD_SOFT: return "Quadraphonic";
case AL_5POINT1_SOFT: return "5.1 Surround";
case AL_6POINT1_SOFT: return "6.1 Surround";
case AL_7POINT1_SOFT: return "7.1 Surround";
}
return "Unknown Channels";
}
const char *TypeName(ALenum type)
{
switch(type)
{
case AL_BYTE_SOFT: return "S8";
case AL_UNSIGNED_BYTE_SOFT: return "U8";
case AL_SHORT_SOFT: return "S16";
case AL_UNSIGNED_SHORT_SOFT: return "U16";
case AL_INT_SOFT: return "S32";
case AL_UNSIGNED_INT_SOFT: return "U32";
case AL_FLOAT_SOFT: return "Float32";
case AL_DOUBLE_SOFT: return "Float64";
}
return "Unknown Type";
}
ALsizei FramesToBytes(ALsizei size, ALenum channels, ALenum type)
{
switch(channels)
{
case AL_MONO_SOFT: size *= 1; break;
case AL_STEREO_SOFT: size *= 2; break;
case AL_REAR_SOFT: size *= 2; break;
case AL_QUAD_SOFT: size *= 4; break;
case AL_5POINT1_SOFT: size *= 6; break;
case AL_6POINT1_SOFT: size *= 7; break;
case AL_7POINT1_SOFT: size *= 8; break;
}
switch(type)
{
case AL_BYTE_SOFT: size *= sizeof(ALbyte); break;
case AL_UNSIGNED_BYTE_SOFT: size *= sizeof(ALubyte); break;
case AL_SHORT_SOFT: size *= sizeof(ALshort); break;
case AL_UNSIGNED_SHORT_SOFT: size *= sizeof(ALushort); break;
case AL_INT_SOFT: size *= sizeof(ALint); break;
case AL_UNSIGNED_INT_SOFT: size *= sizeof(ALuint); break;
case AL_FLOAT_SOFT: size *= sizeof(ALfloat); break;
case AL_DOUBLE_SOFT: size *= sizeof(ALdouble); break;
}
return size;
}
ALsizei BytesToFrames(ALsizei size, ALenum channels, ALenum type)
{
return size / FramesToBytes(1, channels, type);
}
+47
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#ifndef ALHELPERS_H
#define ALHELPERS_H
#ifndef _WIN32
#include <unistd.h>
#define Sleep(x) usleep((x)*1000)
#else
#define WIN32_LEAN_AND_MEAN
#include <windows.h>
#endif
#ifdef __cplusplus
extern "C" {
#endif /* __cplusplus */
/* Some helper functions to get the name from the channel and type enums. */
const char *ChannelsName(ALenum chans);
const char *TypeName(ALenum type);
/* Helpers to convert frame counts and byte lengths. */
ALsizei FramesToBytes(ALsizei size, ALenum channels, ALenum type);
ALsizei BytesToFrames(ALsizei size, ALenum channels, ALenum type);
/* Retrieves a compatible buffer format given the channel configuration and
* sample type. If an alIsBufferFormatSupportedSOFT-compatible function is
* provided, it will be called to find the closest-matching format from
* AL_SOFT_buffer_samples. Returns AL_NONE (0) if no supported format can be
* found. */
ALenum GetFormat(ALenum channels, ALenum type, LPALISBUFFERFORMATSUPPORTEDSOFT palIsBufferFormatSupportedSOFT);
/* Loads samples into a buffer using the standard alBufferData call, but with a
* LPALBUFFERSAMPLESSOFT-compatible prototype. Assumes internalformat is valid
* for alBufferData, and that channels and type match it. */
void AL_APIENTRY wrap_BufferSamples(ALuint buffer, ALuint samplerate,
ALenum internalformat, ALsizei samples,
ALenum channels, ALenum type,
const ALvoid *data);
/* Easy device init/deinit functions. InitAL returns 0 on success. */
int InitAL(void);
void CloseAL(void);
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif /* ALHELPERS_H */
+74
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@@ -0,0 +1,74 @@
HRTF Support
============
Starting with OpenAL Soft 1.14, HRTFs can be used to enable enhanced
spatialization for both 3D (mono) and multi-channel sources, when used with
headphones/stereo output. This can be enabled using the 'hrtf' config option.
For multi-channel sources this creates a virtual speaker effect, making it
sound as if speakers provide a discrete position for each channel around the
listener. For mono sources this provides much more versatility in the perceived
placement of sounds, making it seem as though they are coming from all around,
including above and below the listener, instead of just to the front, back, and
sides.
The built-in data set is based on the KEMAR HRTF data provided by MIT, which
can be found at <http://sound.media.mit.edu/resources/KEMAR.html>. It's only
available when using 44100hz playback.
External HRTF Data Sets
=======================
OpenAL Soft also provides an option to use user-specified data sets, in
addition to or in place of the built-in set. This allows users to provide their
own data sets, which could be better suited for their heads, or to work with
stereo speakers instead of headphones, or to support more playback sample
rates, for example.
The file format is specified below. It uses little-endian byte order.
==
ALchar magic[8] = "MinPHR01";
ALuint sampleRate;
ALubyte hrirSize; /* Can be 8 to 128 in steps of 8. */
ALubyte evCount; /* Can be 5 to 128. */
ALubyte azCount[evCount]; /* Each can be 1 to 128. */
/* NOTE: hrirCount is the sum of all azCounts */
ALshort coefficients[hrirCount][hrirSize];
ALubyte delays[hrirCount]; /* Each can be 0 to 63. */
==
The data is described as thus:
The file first starts with the 8-byte marker, "MinPHR01", to identify it as an
HRTF data set. This is followed by an unsigned 32-bit integer, specifying the
sample rate the data set is designed for (OpenAL Soft will not use it if the
output device's playback rate doesn't match).
Afterward, an unsigned 8-bit integer specifies how many sample points (or
finite impulse response filter coefficients) make up each HRIR.
The following unsigned 8-bit integer specifies the number of elevations used
by the data set. The elevations start at the bottom (-90 degrees), and
increment upwards. Following this is an array of unsigned 8-bit integers, one
for each elevation which specifies the number of azimuths (and thus HRIRs) that
make up each elevation. Azimuths start clockwise from the front, constructing
a full circle for the left ear only. The right ear uses the same HRIRs but in
reverse (ie, left = angle, right = 360-angle).
The actual coefficients follow. Each coefficient is a signed 16-bit sample,
with each HRIR being a consecutive number of sample points. The HRIRs must be
minimum-phase. This allows the use of a smaller filter length, reducing
computation. For reference, the built-in data set uses a 32-point filter while
even the smallest data set provided by MIT used a 128-sample filter (a 4x
reduction by applying minimum-phase reconstruction). Theoretically, one could
further reduce the minimum-phase version down to a 16-point filter with only a
small reduction in quality.
After the coefficients is an array of unsigned 8-bit delay values, one for
each HRIR. This is the propagation delay (in samples) a signal must wait before
being convolved with the corresponding minimum-phase HRIR filter.
+515 -583
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+107 -151
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@@ -5,16 +5,11 @@
extern "C" {
#endif
#if defined(_WIN32) && !defined(_XBOX)
/* _OPENAL32LIB is deprecated */
#if defined(AL_BUILD_LIBRARY) || defined (_OPENAL32LIB)
#define ALC_API __declspec(dllexport)
#else
#ifndef ALC_API
#if defined(AL_LIBTYPE_STATIC)
#define ALC_API
#elif defined(_WIN32)
#define ALC_API __declspec(dllimport)
#endif
#else
#if defined(AL_BUILD_LIBRARY) && defined(HAVE_GCC_VISIBILITY)
#define ALC_API __attribute__((visibility("default")))
#else
#define ALC_API extern
#endif
@@ -26,25 +21,20 @@ extern "C" {
#define ALC_APIENTRY
#endif
#if defined(TARGET_OS_MAC) && TARGET_OS_MAC
#pragma export on
#endif
/*
* The ALCAPI, ALCAPIENTRY, and ALC_INVALID macros are deprecated, but are
* included for applications porting code from AL 1.0
*/
#define ALCAPI ALC_API
#define ALCAPIENTRY ALC_APIENTRY
#define ALC_INVALID 0
/** Deprecated macro. */
#define ALCAPI ALC_API
#define ALCAPIENTRY ALC_APIENTRY
#define ALC_INVALID 0
/** Supported ALC version? */
#define ALC_VERSION_0_1 1
#define ALC_VERSION_0_1 1
/** Opaque device handle */
typedef struct ALCdevice_struct ALCdevice;
/** Opaque context handle */
typedef struct ALCcontext_struct ALCcontext;
/** 8-bit boolean */
typedef char ALCboolean;
@@ -52,7 +42,7 @@ typedef char ALCboolean;
typedef char ALCchar;
/** signed 8-bit 2's complement integer */
typedef char ALCbyte;
typedef signed char ALCbyte;
/** unsigned 8-bit integer */
typedef unsigned char ALCubyte;
@@ -87,192 +77,158 @@ typedef void ALCvoid;
/* Enumerant values begin at column 50. No tabs. */
/* Boolean False. */
/** Boolean False. */
#define ALC_FALSE 0
/* Boolean True. */
/** Boolean True. */
#define ALC_TRUE 1
/**
* followed by <int> Hz
*/
/** Context attribute: <int> Hz. */
#define ALC_FREQUENCY 0x1007
/**
* followed by <int> Hz
*/
/** Context attribute: <int> Hz. */
#define ALC_REFRESH 0x1008
/**
* followed by AL_TRUE, AL_FALSE
*/
/** Context attribute: AL_TRUE or AL_FALSE. */
#define ALC_SYNC 0x1009
/**
* followed by <int> Num of requested Mono (3D) Sources
*/
/** Context attribute: <int> requested Mono (3D) Sources. */
#define ALC_MONO_SOURCES 0x1010
/**
* followed by <int> Num of requested Stereo Sources
*/
/** Context attribute: <int> requested Stereo Sources. */
#define ALC_STEREO_SOURCES 0x1011
/**
* errors
*/
/** No error. */
#define ALC_NO_ERROR 0
/**
* No error
*/
#define ALC_NO_ERROR ALC_FALSE
/**
* No device
*/
/** Invalid device handle. */
#define ALC_INVALID_DEVICE 0xA001
/**
* invalid context ID
*/
/** Invalid context handle. */
#define ALC_INVALID_CONTEXT 0xA002
/**
* bad enum
*/
/** Invalid enum parameter passed to an ALC call. */
#define ALC_INVALID_ENUM 0xA003
/**
* bad value
*/
/** Invalid value parameter passed to an ALC call. */
#define ALC_INVALID_VALUE 0xA004
/**
* Out of memory.
*/
/** Out of memory. */
#define ALC_OUT_OF_MEMORY 0xA005
/**
* The Specifier string for default device
*/
#define ALC_DEFAULT_DEVICE_SPECIFIER 0x1004
#define ALC_DEVICE_SPECIFIER 0x1005
#define ALC_EXTENSIONS 0x1006
/** Runtime ALC version. */
#define ALC_MAJOR_VERSION 0x1000
#define ALC_MINOR_VERSION 0x1001
/** Context attribute list properties. */
#define ALC_ATTRIBUTES_SIZE 0x1002
#define ALC_ALL_ATTRIBUTES 0x1003
/** String for the default device specifier. */
#define ALC_DEFAULT_DEVICE_SPECIFIER 0x1004
/**
* ALC_ENUMERATE_ALL_EXT enums
* String for the given device's specifier.
*
* If device handle is NULL, it is instead a null-char separated list of
* strings of known device specifiers (list ends with an empty string).
*/
#define ALC_DEFAULT_ALL_DEVICES_SPECIFIER 0x1012
#define ALC_ALL_DEVICES_SPECIFIER 0x1013
#define ALC_DEVICE_SPECIFIER 0x1005
/** String for space-separated list of ALC extensions. */
#define ALC_EXTENSIONS 0x1006
/** Capture extension */
#define ALC_EXT_CAPTURE 1
/**
* Capture extension
* String for the given capture device's specifier.
*
* If device handle is NULL, it is instead a null-char separated list of
* strings of known capture device specifiers (list ends with an empty string).
*/
#define ALC_CAPTURE_DEVICE_SPECIFIER 0x310
/** String for the default capture device specifier. */
#define ALC_CAPTURE_DEFAULT_DEVICE_SPECIFIER 0x311
/** Number of sample frames available for capture. */
#define ALC_CAPTURE_SAMPLES 0x312
/*
* Context Management
/** Enumerate All extension */
#define ALC_ENUMERATE_ALL_EXT 1
/** String for the default extended device specifier. */
#define ALC_DEFAULT_ALL_DEVICES_SPECIFIER 0x1012
/**
* String for the given extended device's specifier.
*
* If device handle is NULL, it is instead a null-char separated list of
* strings of known extended device specifiers (list ends with an empty string).
*/
ALC_API ALCcontext * ALC_APIENTRY alcCreateContext( ALCdevice *device, const ALCint* attrlist );
ALC_API ALCboolean ALC_APIENTRY alcMakeContextCurrent( ALCcontext *context );
ALC_API void ALC_APIENTRY alcProcessContext( ALCcontext *context );
ALC_API void ALC_APIENTRY alcSuspendContext( ALCcontext *context );
ALC_API void ALC_APIENTRY alcDestroyContext( ALCcontext *context );
ALC_API ALCcontext * ALC_APIENTRY alcGetCurrentContext( void );
ALC_API ALCdevice* ALC_APIENTRY alcGetContextsDevice( ALCcontext *context );
#define ALC_ALL_DEVICES_SPECIFIER 0x1013
/*
* Device Management
*/
ALC_API ALCdevice * ALC_APIENTRY alcOpenDevice( const ALCchar *devicename );
/** Context management. */
ALC_API ALCcontext* ALC_APIENTRY alcCreateContext(ALCdevice *device, const ALCint* attrlist);
ALC_API ALCboolean ALC_APIENTRY alcMakeContextCurrent(ALCcontext *context);
ALC_API void ALC_APIENTRY alcProcessContext(ALCcontext *context);
ALC_API void ALC_APIENTRY alcSuspendContext(ALCcontext *context);
ALC_API void ALC_APIENTRY alcDestroyContext(ALCcontext *context);
ALC_API ALCcontext* ALC_APIENTRY alcGetCurrentContext(void);
ALC_API ALCdevice* ALC_APIENTRY alcGetContextsDevice(ALCcontext *context);
ALC_API ALCboolean ALC_APIENTRY alcCloseDevice( ALCdevice *device );
/** Device management. */
ALC_API ALCdevice* ALC_APIENTRY alcOpenDevice(const ALCchar *devicename);
ALC_API ALCboolean ALC_APIENTRY alcCloseDevice(ALCdevice *device);
/*
/**
* Error support.
* Obtain the most recent Context error
*
* Obtain the most recent Device error.
*/
ALC_API ALCenum ALC_APIENTRY alcGetError( ALCdevice *device );
ALC_API ALCenum ALC_APIENTRY alcGetError(ALCdevice *device);
/*
/**
* Extension support.
*
* Query for the presence of an extension, and obtain any appropriate
* function pointers and enum values.
*/
ALC_API ALCboolean ALC_APIENTRY alcIsExtensionPresent( ALCdevice *device, const ALCchar *extname );
ALC_API ALCboolean ALC_APIENTRY alcIsExtensionPresent(ALCdevice *device, const ALCchar *extname);
ALC_API void* ALC_APIENTRY alcGetProcAddress(ALCdevice *device, const ALCchar *funcname);
ALC_API ALCenum ALC_APIENTRY alcGetEnumValue(ALCdevice *device, const ALCchar *enumname);
ALC_API void * ALC_APIENTRY alcGetProcAddress( ALCdevice *device, const ALCchar *funcname );
/** Query function. */
ALC_API const ALCchar* ALC_APIENTRY alcGetString(ALCdevice *device, ALCenum param);
ALC_API void ALC_APIENTRY alcGetIntegerv(ALCdevice *device, ALCenum param, ALCsizei size, ALCint *values);
ALC_API ALCenum ALC_APIENTRY alcGetEnumValue( ALCdevice *device, const ALCchar *enumname );
/** Capture function. */
ALC_API ALCdevice* ALC_APIENTRY alcCaptureOpenDevice(const ALCchar *devicename, ALCuint frequency, ALCenum format, ALCsizei buffersize);
ALC_API ALCboolean ALC_APIENTRY alcCaptureCloseDevice(ALCdevice *device);
ALC_API void ALC_APIENTRY alcCaptureStart(ALCdevice *device);
ALC_API void ALC_APIENTRY alcCaptureStop(ALCdevice *device);
ALC_API void ALC_APIENTRY alcCaptureSamples(ALCdevice *device, ALCvoid *buffer, ALCsizei samples);
/*
* Query functions
*/
ALC_API const ALCchar * ALC_APIENTRY alcGetString( ALCdevice *device, ALCenum param );
ALC_API void ALC_APIENTRY alcGetIntegerv( ALCdevice *device, ALCenum param, ALCsizei size, ALCint *data );
/*
* Capture functions
*/
ALC_API ALCdevice* ALC_APIENTRY alcCaptureOpenDevice( const ALCchar *devicename, ALCuint frequency, ALCenum format, ALCsizei buffersize );
ALC_API ALCboolean ALC_APIENTRY alcCaptureCloseDevice( ALCdevice *device );
ALC_API void ALC_APIENTRY alcCaptureStart( ALCdevice *device );
ALC_API void ALC_APIENTRY alcCaptureStop( ALCdevice *device );
ALC_API void ALC_APIENTRY alcCaptureSamples( ALCdevice *device, ALCvoid *buffer, ALCsizei samples );
/*
* Pointer-to-function types, useful for dynamically getting ALC entry points.
*/
typedef ALCcontext * (ALC_APIENTRY *LPALCCREATECONTEXT) (ALCdevice *device, const ALCint *attrlist);
typedef ALCboolean (ALC_APIENTRY *LPALCMAKECONTEXTCURRENT)( ALCcontext *context );
typedef void (ALC_APIENTRY *LPALCPROCESSCONTEXT)( ALCcontext *context );
typedef void (ALC_APIENTRY *LPALCSUSPENDCONTEXT)( ALCcontext *context );
typedef void (ALC_APIENTRY *LPALCDESTROYCONTEXT)( ALCcontext *context );
typedef ALCcontext * (ALC_APIENTRY *LPALCGETCURRENTCONTEXT)( void );
typedef ALCdevice * (ALC_APIENTRY *LPALCGETCONTEXTSDEVICE)( ALCcontext *context );
typedef ALCdevice * (ALC_APIENTRY *LPALCOPENDEVICE)( const ALCchar *devicename );
typedef ALCboolean (ALC_APIENTRY *LPALCCLOSEDEVICE)( ALCdevice *device );
typedef ALCenum (ALC_APIENTRY *LPALCGETERROR)( ALCdevice *device );
typedef ALCboolean (ALC_APIENTRY *LPALCISEXTENSIONPRESENT)( ALCdevice *device, const ALCchar *extname );
typedef void * (ALC_APIENTRY *LPALCGETPROCADDRESS)(ALCdevice *device, const ALCchar *funcname );
typedef ALCenum (ALC_APIENTRY *LPALCGETENUMVALUE)(ALCdevice *device, const ALCchar *enumname );
typedef const ALCchar* (ALC_APIENTRY *LPALCGETSTRING)( ALCdevice *device, ALCenum param );
typedef void (ALC_APIENTRY *LPALCGETINTEGERV)( ALCdevice *device, ALCenum param, ALCsizei size, ALCint *dest );
typedef ALCdevice * (ALC_APIENTRY *LPALCCAPTUREOPENDEVICE)( const ALCchar *devicename, ALCuint frequency, ALCenum format, ALCsizei buffersize );
typedef ALCboolean (ALC_APIENTRY *LPALCCAPTURECLOSEDEVICE)( ALCdevice *device );
typedef void (ALC_APIENTRY *LPALCCAPTURESTART)( ALCdevice *device );
typedef void (ALC_APIENTRY *LPALCCAPTURESTOP)( ALCdevice *device );
typedef void (ALC_APIENTRY *LPALCCAPTURESAMPLES)( ALCdevice *device, ALCvoid *buffer, ALCsizei samples );
#if defined(TARGET_OS_MAC) && TARGET_OS_MAC
#pragma export off
#endif
/** Pointer-to-function type, useful for dynamically getting ALC entry points. */
typedef ALCcontext* (ALC_APIENTRY *LPALCCREATECONTEXT)(ALCdevice *device, const ALCint *attrlist);
typedef ALCboolean (ALC_APIENTRY *LPALCMAKECONTEXTCURRENT)(ALCcontext *context);
typedef void (ALC_APIENTRY *LPALCPROCESSCONTEXT)(ALCcontext *context);
typedef void (ALC_APIENTRY *LPALCSUSPENDCONTEXT)(ALCcontext *context);
typedef void (ALC_APIENTRY *LPALCDESTROYCONTEXT)(ALCcontext *context);
typedef ALCcontext* (ALC_APIENTRY *LPALCGETCURRENTCONTEXT)(void);
typedef ALCdevice* (ALC_APIENTRY *LPALCGETCONTEXTSDEVICE)(ALCcontext *context);
typedef ALCdevice* (ALC_APIENTRY *LPALCOPENDEVICE)(const ALCchar *devicename);
typedef ALCboolean (ALC_APIENTRY *LPALCCLOSEDEVICE)(ALCdevice *device);
typedef ALCenum (ALC_APIENTRY *LPALCGETERROR)(ALCdevice *device);
typedef ALCboolean (ALC_APIENTRY *LPALCISEXTENSIONPRESENT)(ALCdevice *device, const ALCchar *extname);
typedef void* (ALC_APIENTRY *LPALCGETPROCADDRESS)(ALCdevice *device, const ALCchar *funcname);
typedef ALCenum (ALC_APIENTRY *LPALCGETENUMVALUE)(ALCdevice *device, const ALCchar *enumname);
typedef const ALCchar* (ALC_APIENTRY *LPALCGETSTRING)(ALCdevice *device, ALCenum param);
typedef void (ALC_APIENTRY *LPALCGETINTEGERV)(ALCdevice *device, ALCenum param, ALCsizei size, ALCint *values);
typedef ALCdevice* (ALC_APIENTRY *LPALCCAPTUREOPENDEVICE)(const ALCchar *devicename, ALCuint frequency, ALCenum format, ALCsizei buffersize);
typedef ALCboolean (ALC_APIENTRY *LPALCCAPTURECLOSEDEVICE)(ALCdevice *device);
typedef void (ALC_APIENTRY *LPALCCAPTURESTART)(ALCdevice *device);
typedef void (ALC_APIENTRY *LPALCCAPTURESTOP)(ALCdevice *device);
typedef void (ALC_APIENTRY *LPALCCAPTURESAMPLES)(ALCdevice *device, ALCvoid *buffer, ALCsizei samples);
#if defined(__cplusplus)
}
+355
View File
@@ -0,0 +1,355 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 2008 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#ifndef AL_ALEXT_H
#define AL_ALEXT_H
#include <stddef.h>
/* Define int64_t and uint64_t types */
#if defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L
#include <inttypes.h>
#elif defined(_WIN32) && defined(__GNUC__)
#include <stdint.h>
#elif defined(_WIN32)
typedef __int64 int64_t;
typedef unsigned __int64 uint64_t;
#else
/* Fallback if nothing above works */
#include <inttypes.h>
#endif
#include "alc.h"
#include "al.h"
#ifdef __cplusplus
extern "C" {
#endif
#ifndef AL_LOKI_IMA_ADPCM_format
#define AL_LOKI_IMA_ADPCM_format 1
#define AL_FORMAT_IMA_ADPCM_MONO16_EXT 0x10000
#define AL_FORMAT_IMA_ADPCM_STEREO16_EXT 0x10001
#endif
#ifndef AL_LOKI_WAVE_format
#define AL_LOKI_WAVE_format 1
#define AL_FORMAT_WAVE_EXT 0x10002
#endif
#ifndef AL_EXT_vorbis
#define AL_EXT_vorbis 1
#define AL_FORMAT_VORBIS_EXT 0x10003
#endif
#ifndef AL_LOKI_quadriphonic
#define AL_LOKI_quadriphonic 1
#define AL_FORMAT_QUAD8_LOKI 0x10004
#define AL_FORMAT_QUAD16_LOKI 0x10005
#endif
#ifndef AL_EXT_float32
#define AL_EXT_float32 1
#define AL_FORMAT_MONO_FLOAT32 0x10010
#define AL_FORMAT_STEREO_FLOAT32 0x10011
#endif
#ifndef AL_EXT_double
#define AL_EXT_double 1
#define AL_FORMAT_MONO_DOUBLE_EXT 0x10012
#define AL_FORMAT_STEREO_DOUBLE_EXT 0x10013
#endif
#ifndef AL_EXT_MULAW
#define AL_EXT_MULAW 1
#define AL_FORMAT_MONO_MULAW_EXT 0x10014
#define AL_FORMAT_STEREO_MULAW_EXT 0x10015
#endif
#ifndef AL_EXT_ALAW
#define AL_EXT_ALAW 1
#define AL_FORMAT_MONO_ALAW_EXT 0x10016
#define AL_FORMAT_STEREO_ALAW_EXT 0x10017
#endif
#ifndef ALC_LOKI_audio_channel
#define ALC_LOKI_audio_channel 1
#define ALC_CHAN_MAIN_LOKI 0x500001
#define ALC_CHAN_PCM_LOKI 0x500002
#define ALC_CHAN_CD_LOKI 0x500003
#endif
#ifndef AL_EXT_MCFORMATS
#define AL_EXT_MCFORMATS 1
#define AL_FORMAT_QUAD8 0x1204
#define AL_FORMAT_QUAD16 0x1205
#define AL_FORMAT_QUAD32 0x1206
#define AL_FORMAT_REAR8 0x1207
#define AL_FORMAT_REAR16 0x1208
#define AL_FORMAT_REAR32 0x1209
#define AL_FORMAT_51CHN8 0x120A
#define AL_FORMAT_51CHN16 0x120B
#define AL_FORMAT_51CHN32 0x120C
#define AL_FORMAT_61CHN8 0x120D
#define AL_FORMAT_61CHN16 0x120E
#define AL_FORMAT_61CHN32 0x120F
#define AL_FORMAT_71CHN8 0x1210
#define AL_FORMAT_71CHN16 0x1211
#define AL_FORMAT_71CHN32 0x1212
#endif
#ifndef AL_EXT_MULAW_MCFORMATS
#define AL_EXT_MULAW_MCFORMATS 1
#define AL_FORMAT_MONO_MULAW 0x10014
#define AL_FORMAT_STEREO_MULAW 0x10015
#define AL_FORMAT_QUAD_MULAW 0x10021
#define AL_FORMAT_REAR_MULAW 0x10022
#define AL_FORMAT_51CHN_MULAW 0x10023
#define AL_FORMAT_61CHN_MULAW 0x10024
#define AL_FORMAT_71CHN_MULAW 0x10025
#endif
#ifndef AL_EXT_IMA4
#define AL_EXT_IMA4 1
#define AL_FORMAT_MONO_IMA4 0x1300
#define AL_FORMAT_STEREO_IMA4 0x1301
#endif
#ifndef AL_EXT_STATIC_BUFFER
#define AL_EXT_STATIC_BUFFER 1
typedef ALvoid (AL_APIENTRY*PFNALBUFFERDATASTATICPROC)(const ALint,ALenum,ALvoid*,ALsizei,ALsizei);
#ifdef AL_ALEXT_PROTOTYPES
AL_API ALvoid AL_APIENTRY alBufferDataStatic(const ALint buffer, ALenum format, ALvoid *data, ALsizei len, ALsizei freq);
#endif
#endif
#ifndef ALC_EXT_EFX
#define ALC_EXT_EFX 1
#include "efx.h"
#endif
#ifndef ALC_EXT_disconnect
#define ALC_EXT_disconnect 1
#define ALC_CONNECTED 0x313
#endif
#ifndef ALC_EXT_thread_local_context
#define ALC_EXT_thread_local_context 1
typedef ALCboolean (ALC_APIENTRY*PFNALCSETTHREADCONTEXTPROC)(ALCcontext *context);
typedef ALCcontext* (ALC_APIENTRY*PFNALCGETTHREADCONTEXTPROC)(void);
#ifdef AL_ALEXT_PROTOTYPES
ALC_API ALCboolean ALC_APIENTRY alcSetThreadContext(ALCcontext *context);
ALC_API ALCcontext* ALC_APIENTRY alcGetThreadContext(void);
#endif
#endif
#ifndef AL_EXT_source_distance_model
#define AL_EXT_source_distance_model 1
#define AL_SOURCE_DISTANCE_MODEL 0x200
#endif
#ifndef AL_SOFT_buffer_sub_data
#define AL_SOFT_buffer_sub_data 1
#define AL_BYTE_RW_OFFSETS_SOFT 0x1031
#define AL_SAMPLE_RW_OFFSETS_SOFT 0x1032
typedef ALvoid (AL_APIENTRY*PFNALBUFFERSUBDATASOFTPROC)(ALuint,ALenum,const ALvoid*,ALsizei,ALsizei);
#ifdef AL_ALEXT_PROTOTYPES
AL_API ALvoid AL_APIENTRY alBufferSubDataSOFT(ALuint buffer,ALenum format,const ALvoid *data,ALsizei offset,ALsizei length);
#endif
#endif
#ifndef AL_SOFT_loop_points
#define AL_SOFT_loop_points 1
#define AL_LOOP_POINTS_SOFT 0x2015
#endif
#ifndef AL_EXT_FOLDBACK
#define AL_EXT_FOLDBACK 1
#define AL_EXT_FOLDBACK_NAME "AL_EXT_FOLDBACK"
#define AL_FOLDBACK_EVENT_BLOCK 0x4112
#define AL_FOLDBACK_EVENT_START 0x4111
#define AL_FOLDBACK_EVENT_STOP 0x4113
#define AL_FOLDBACK_MODE_MONO 0x4101
#define AL_FOLDBACK_MODE_STEREO 0x4102
typedef void (AL_APIENTRY*LPALFOLDBACKCALLBACK)(ALenum,ALsizei);
typedef void (AL_APIENTRY*LPALREQUESTFOLDBACKSTART)(ALenum,ALsizei,ALsizei,ALfloat*,LPALFOLDBACKCALLBACK);
typedef void (AL_APIENTRY*LPALREQUESTFOLDBACKSTOP)(void);
#ifdef AL_ALEXT_PROTOTYPES
AL_API void AL_APIENTRY alRequestFoldbackStart(ALenum mode,ALsizei count,ALsizei length,ALfloat *mem,LPALFOLDBACKCALLBACK callback);
AL_API void AL_APIENTRY alRequestFoldbackStop(void);
#endif
#endif
#ifndef ALC_EXT_DEDICATED
#define ALC_EXT_DEDICATED 1
#define AL_DEDICATED_GAIN 0x0001
#define AL_EFFECT_DEDICATED_DIALOGUE 0x9001
#define AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT 0x9000
#endif
#ifndef AL_SOFT_buffer_samples
#define AL_SOFT_buffer_samples 1
/* Channel configurations */
#define AL_MONO_SOFT 0x1500
#define AL_STEREO_SOFT 0x1501
#define AL_REAR_SOFT 0x1502
#define AL_QUAD_SOFT 0x1503
#define AL_5POINT1_SOFT 0x1504
#define AL_6POINT1_SOFT 0x1505
#define AL_7POINT1_SOFT 0x1506
/* Sample types */
#define AL_BYTE_SOFT 0x1400
#define AL_UNSIGNED_BYTE_SOFT 0x1401
#define AL_SHORT_SOFT 0x1402
#define AL_UNSIGNED_SHORT_SOFT 0x1403
#define AL_INT_SOFT 0x1404
#define AL_UNSIGNED_INT_SOFT 0x1405
#define AL_FLOAT_SOFT 0x1406
#define AL_DOUBLE_SOFT 0x1407
#define AL_BYTE3_SOFT 0x1408
#define AL_UNSIGNED_BYTE3_SOFT 0x1409
/* Storage formats */
#define AL_MONO8_SOFT 0x1100
#define AL_MONO16_SOFT 0x1101
#define AL_MONO32F_SOFT 0x10010
#define AL_STEREO8_SOFT 0x1102
#define AL_STEREO16_SOFT 0x1103
#define AL_STEREO32F_SOFT 0x10011
#define AL_QUAD8_SOFT 0x1204
#define AL_QUAD16_SOFT 0x1205
#define AL_QUAD32F_SOFT 0x1206
#define AL_REAR8_SOFT 0x1207
#define AL_REAR16_SOFT 0x1208
#define AL_REAR32F_SOFT 0x1209
#define AL_5POINT1_8_SOFT 0x120A
#define AL_5POINT1_16_SOFT 0x120B
#define AL_5POINT1_32F_SOFT 0x120C
#define AL_6POINT1_8_SOFT 0x120D
#define AL_6POINT1_16_SOFT 0x120E
#define AL_6POINT1_32F_SOFT 0x120F
#define AL_7POINT1_8_SOFT 0x1210
#define AL_7POINT1_16_SOFT 0x1211
#define AL_7POINT1_32F_SOFT 0x1212
/* Buffer attributes */
#define AL_INTERNAL_FORMAT_SOFT 0x2008
#define AL_BYTE_LENGTH_SOFT 0x2009
#define AL_SAMPLE_LENGTH_SOFT 0x200A
#define AL_SEC_LENGTH_SOFT 0x200B
typedef void (AL_APIENTRY*LPALBUFFERSAMPLESSOFT)(ALuint,ALuint,ALenum,ALsizei,ALenum,ALenum,const ALvoid*);
typedef void (AL_APIENTRY*LPALBUFFERSUBSAMPLESSOFT)(ALuint,ALsizei,ALsizei,ALenum,ALenum,const ALvoid*);
typedef void (AL_APIENTRY*LPALGETBUFFERSAMPLESSOFT)(ALuint,ALsizei,ALsizei,ALenum,ALenum,ALvoid*);
typedef ALboolean (AL_APIENTRY*LPALISBUFFERFORMATSUPPORTEDSOFT)(ALenum);
#ifdef AL_ALEXT_PROTOTYPES
AL_API void AL_APIENTRY alBufferSamplesSOFT(ALuint buffer, ALuint samplerate, ALenum internalformat, ALsizei samples, ALenum channels, ALenum type, const ALvoid *data);
AL_API void AL_APIENTRY alBufferSubSamplesSOFT(ALuint buffer, ALsizei offset, ALsizei samples, ALenum channels, ALenum type, const ALvoid *data);
AL_API void AL_APIENTRY alGetBufferSamplesSOFT(ALuint buffer, ALsizei offset, ALsizei samples, ALenum channels, ALenum type, ALvoid *data);
AL_API ALboolean AL_APIENTRY alIsBufferFormatSupportedSOFT(ALenum format);
#endif
#endif
#ifndef AL_SOFT_direct_channels
#define AL_SOFT_direct_channels 1
#define AL_DIRECT_CHANNELS_SOFT 0x1033
#endif
#ifndef ALC_SOFT_loopback
#define ALC_SOFT_loopback 1
#define ALC_FORMAT_CHANNELS_SOFT 0x1990
#define ALC_FORMAT_TYPE_SOFT 0x1991
/* Sample types */
#define ALC_BYTE_SOFT 0x1400
#define ALC_UNSIGNED_BYTE_SOFT 0x1401
#define ALC_SHORT_SOFT 0x1402
#define ALC_UNSIGNED_SHORT_SOFT 0x1403
#define ALC_INT_SOFT 0x1404
#define ALC_UNSIGNED_INT_SOFT 0x1405
#define ALC_FLOAT_SOFT 0x1406
/* Channel configurations */
#define ALC_MONO_SOFT 0x1500
#define ALC_STEREO_SOFT 0x1501
#define ALC_QUAD_SOFT 0x1503
#define ALC_5POINT1_SOFT 0x1504
#define ALC_6POINT1_SOFT 0x1505
#define ALC_7POINT1_SOFT 0x1506
typedef ALCdevice* (ALC_APIENTRY*LPALCLOOPBACKOPENDEVICESOFT)(const ALCchar*);
typedef ALCboolean (ALC_APIENTRY*LPALCISRENDERFORMATSUPPORTEDSOFT)(ALCdevice*,ALCsizei,ALCenum,ALCenum);
typedef void (ALC_APIENTRY*LPALCRENDERSAMPLESSOFT)(ALCdevice*,ALCvoid*,ALCsizei);
#ifdef AL_ALEXT_PROTOTYPES
ALC_API ALCdevice* ALC_APIENTRY alcLoopbackOpenDeviceSOFT(const ALCchar *deviceName);
ALC_API ALCboolean ALC_APIENTRY alcIsRenderFormatSupportedSOFT(ALCdevice *device, ALCsizei freq, ALCenum channels, ALCenum type);
ALC_API void ALC_APIENTRY alcRenderSamplesSOFT(ALCdevice *device, ALCvoid *buffer, ALCsizei samples);
#endif
#endif
#ifndef AL_EXT_STEREO_ANGLES
#define AL_EXT_STEREO_ANGLES 1
#define AL_STEREO_ANGLES 0x1030
#endif
#ifndef AL_EXT_SOURCE_RADIUS
#define AL_EXT_SOURCE_RADIUS 1
#define AL_SOURCE_RADIUS 0x1031
#endif
#ifndef AL_SOFT_source_latency
#define AL_SOFT_source_latency 1
#define AL_SAMPLE_OFFSET_LATENCY_SOFT 0x1200
#define AL_SEC_OFFSET_LATENCY_SOFT 0x1201
typedef int64_t ALint64SOFT;
typedef uint64_t ALuint64SOFT;
typedef void (AL_APIENTRY*LPALSOURCEDSOFT)(ALuint,ALenum,ALdouble);
typedef void (AL_APIENTRY*LPALSOURCE3DSOFT)(ALuint,ALenum,ALdouble,ALdouble,ALdouble);
typedef void (AL_APIENTRY*LPALSOURCEDVSOFT)(ALuint,ALenum,const ALdouble*);
typedef void (AL_APIENTRY*LPALGETSOURCEDSOFT)(ALuint,ALenum,ALdouble*);
typedef void (AL_APIENTRY*LPALGETSOURCE3DSOFT)(ALuint,ALenum,ALdouble*,ALdouble*,ALdouble*);
typedef void (AL_APIENTRY*LPALGETSOURCEDVSOFT)(ALuint,ALenum,ALdouble*);
typedef void (AL_APIENTRY*LPALSOURCEI64SOFT)(ALuint,ALenum,ALint64SOFT);
typedef void (AL_APIENTRY*LPALSOURCE3I64SOFT)(ALuint,ALenum,ALint64SOFT,ALint64SOFT,ALint64SOFT);
typedef void (AL_APIENTRY*LPALSOURCEI64VSOFT)(ALuint,ALenum,const ALint64SOFT*);
typedef void (AL_APIENTRY*LPALGETSOURCEI64SOFT)(ALuint,ALenum,ALint64SOFT*);
typedef void (AL_APIENTRY*LPALGETSOURCE3I64SOFT)(ALuint,ALenum,ALint64SOFT*,ALint64SOFT*,ALint64SOFT*);
typedef void (AL_APIENTRY*LPALGETSOURCEI64VSOFT)(ALuint,ALenum,ALint64SOFT*);
#ifdef AL_ALEXT_PROTOTYPES
AL_API void AL_APIENTRY alSourcedSOFT(ALuint source, ALenum param, ALdouble value);
AL_API void AL_APIENTRY alSource3dSOFT(ALuint source, ALenum param, ALdouble value1, ALdouble value2, ALdouble value3);
AL_API void AL_APIENTRY alSourcedvSOFT(ALuint source, ALenum param, const ALdouble *values);
AL_API void AL_APIENTRY alGetSourcedSOFT(ALuint source, ALenum param, ALdouble *value);
AL_API void AL_APIENTRY alGetSource3dSOFT(ALuint source, ALenum param, ALdouble *value1, ALdouble *value2, ALdouble *value3);
AL_API void AL_APIENTRY alGetSourcedvSOFT(ALuint source, ALenum param, ALdouble *values);
AL_API void AL_APIENTRY alSourcei64SOFT(ALuint source, ALenum param, ALint64SOFT value);
AL_API void AL_APIENTRY alSource3i64SOFT(ALuint source, ALenum param, ALint64SOFT value1, ALint64SOFT value2, ALint64SOFT value3);
AL_API void AL_APIENTRY alSourcei64vSOFT(ALuint source, ALenum param, const ALint64SOFT *values);
AL_API void AL_APIENTRY alGetSourcei64SOFT(ALuint source, ALenum param, ALint64SOFT *value);
AL_API void AL_APIENTRY alGetSource3i64SOFT(ALuint source, ALenum param, ALint64SOFT *value1, ALint64SOFT *value2, ALint64SOFT *value3);
AL_API void AL_APIENTRY alGetSourcei64vSOFT(ALuint source, ALenum param, ALint64SOFT *values);
#endif
#endif
#ifdef __cplusplus
}
#endif
#endif
+3
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@@ -0,0 +1,3 @@
/* The tokens that would be defined here are already defined in efx.h. This
* empty file is here to provide compatibility with Windows-based projects
* that would include it. */
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/* Reverb presets for EFX */
#ifndef EFX_PRESETS_H
#define EFX_PRESETS_H
#ifndef EFXEAXREVERBPROPERTIES_DEFINED
#define EFXEAXREVERBPROPERTIES_DEFINED
typedef struct {
float flDensity;
float flDiffusion;
float flGain;
float flGainHF;
float flGainLF;
float flDecayTime;
float flDecayHFRatio;
float flDecayLFRatio;
float flReflectionsGain;
float flReflectionsDelay;
float flReflectionsPan[3];
float flLateReverbGain;
float flLateReverbDelay;
float flLateReverbPan[3];
float flEchoTime;
float flEchoDepth;
float flModulationTime;
float flModulationDepth;
float flAirAbsorptionGainHF;
float flHFReference;
float flLFReference;
float flRoomRolloffFactor;
int iDecayHFLimit;
} EFXEAXREVERBPROPERTIES, *LPEFXEAXREVERBPROPERTIES;
#endif
/* Default Presets */
#define EFX_REVERB_PRESET_GENERIC \
{ 1.0000f, 1.0000f, 0.3162f, 0.8913f, 1.0000f, 1.4900f, 0.8300f, 1.0000f, 0.0500f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0110f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_PADDEDCELL \
{ 0.1715f, 1.0000f, 0.3162f, 0.0010f, 1.0000f, 0.1700f, 0.1000f, 1.0000f, 0.2500f, 0.0010f, { 0.0000f, 0.0000f, 0.0000f }, 1.2691f, 0.0020f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_ROOM \
{ 0.4287f, 1.0000f, 0.3162f, 0.5929f, 1.0000f, 0.4000f, 0.8300f, 1.0000f, 0.1503f, 0.0020f, { 0.0000f, 0.0000f, 0.0000f }, 1.0629f, 0.0030f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_BATHROOM \
{ 0.1715f, 1.0000f, 0.3162f, 0.2512f, 1.0000f, 1.4900f, 0.5400f, 1.0000f, 0.6531f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 3.2734f, 0.0110f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_LIVINGROOM \
{ 0.9766f, 1.0000f, 0.3162f, 0.0010f, 1.0000f, 0.5000f, 0.1000f, 1.0000f, 0.2051f, 0.0030f, { 0.0000f, 0.0000f, 0.0000f }, 0.2805f, 0.0040f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_STONEROOM \
{ 1.0000f, 1.0000f, 0.3162f, 0.7079f, 1.0000f, 2.3100f, 0.6400f, 1.0000f, 0.4411f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 1.1003f, 0.0170f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_AUDITORIUM \
{ 1.0000f, 1.0000f, 0.3162f, 0.5781f, 1.0000f, 4.3200f, 0.5900f, 1.0000f, 0.4032f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 0.7170f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_CONCERTHALL \
{ 1.0000f, 1.0000f, 0.3162f, 0.5623f, 1.0000f, 3.9200f, 0.7000f, 1.0000f, 0.2427f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 0.9977f, 0.0290f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_CAVE \
{ 1.0000f, 1.0000f, 0.3162f, 1.0000f, 1.0000f, 2.9100f, 1.3000f, 1.0000f, 0.5000f, 0.0150f, { 0.0000f, 0.0000f, 0.0000f }, 0.7063f, 0.0220f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
#define EFX_REVERB_PRESET_ARENA \
{ 1.0000f, 1.0000f, 0.3162f, 0.4477f, 1.0000f, 7.2400f, 0.3300f, 1.0000f, 0.2612f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 1.0186f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_HANGAR \
{ 1.0000f, 1.0000f, 0.3162f, 0.3162f, 1.0000f, 10.0500f, 0.2300f, 1.0000f, 0.5000f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 1.2560f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_CARPETEDHALLWAY \
{ 0.4287f, 1.0000f, 0.3162f, 0.0100f, 1.0000f, 0.3000f, 0.1000f, 1.0000f, 0.1215f, 0.0020f, { 0.0000f, 0.0000f, 0.0000f }, 0.1531f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_HALLWAY \
{ 0.3645f, 1.0000f, 0.3162f, 0.7079f, 1.0000f, 1.4900f, 0.5900f, 1.0000f, 0.2458f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 1.6615f, 0.0110f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_STONECORRIDOR \
{ 1.0000f, 1.0000f, 0.3162f, 0.7612f, 1.0000f, 2.7000f, 0.7900f, 1.0000f, 0.2472f, 0.0130f, { 0.0000f, 0.0000f, 0.0000f }, 1.5758f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_ALLEY \
{ 1.0000f, 0.3000f, 0.3162f, 0.7328f, 1.0000f, 1.4900f, 0.8600f, 1.0000f, 0.2500f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 0.9954f, 0.0110f, { 0.0000f, 0.0000f, 0.0000f }, 0.1250f, 0.9500f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_FOREST \
{ 1.0000f, 0.3000f, 0.3162f, 0.0224f, 1.0000f, 1.4900f, 0.5400f, 1.0000f, 0.0525f, 0.1620f, { 0.0000f, 0.0000f, 0.0000f }, 0.7682f, 0.0880f, { 0.0000f, 0.0000f, 0.0000f }, 0.1250f, 1.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_CITY \
{ 1.0000f, 0.5000f, 0.3162f, 0.3981f, 1.0000f, 1.4900f, 0.6700f, 1.0000f, 0.0730f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 0.1427f, 0.0110f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_MOUNTAINS \
{ 1.0000f, 0.2700f, 0.3162f, 0.0562f, 1.0000f, 1.4900f, 0.2100f, 1.0000f, 0.0407f, 0.3000f, { 0.0000f, 0.0000f, 0.0000f }, 0.1919f, 0.1000f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 1.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
#define EFX_REVERB_PRESET_QUARRY \
{ 1.0000f, 1.0000f, 0.3162f, 0.3162f, 1.0000f, 1.4900f, 0.8300f, 1.0000f, 0.0000f, 0.0610f, { 0.0000f, 0.0000f, 0.0000f }, 1.7783f, 0.0250f, { 0.0000f, 0.0000f, 0.0000f }, 0.1250f, 0.7000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_PLAIN \
{ 1.0000f, 0.2100f, 0.3162f, 0.1000f, 1.0000f, 1.4900f, 0.5000f, 1.0000f, 0.0585f, 0.1790f, { 0.0000f, 0.0000f, 0.0000f }, 0.1089f, 0.1000f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 1.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_PARKINGLOT \
{ 1.0000f, 1.0000f, 0.3162f, 1.0000f, 1.0000f, 1.6500f, 1.5000f, 1.0000f, 0.2082f, 0.0080f, { 0.0000f, 0.0000f, 0.0000f }, 0.2652f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
#define EFX_REVERB_PRESET_SEWERPIPE \
{ 0.3071f, 0.8000f, 0.3162f, 0.3162f, 1.0000f, 2.8100f, 0.1400f, 1.0000f, 1.6387f, 0.0140f, { 0.0000f, 0.0000f, 0.0000f }, 3.2471f, 0.0210f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_UNDERWATER \
{ 0.3645f, 1.0000f, 0.3162f, 0.0100f, 1.0000f, 1.4900f, 0.1000f, 1.0000f, 0.5963f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 7.0795f, 0.0110f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 1.1800f, 0.3480f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_DRUGGED \
{ 0.4287f, 0.5000f, 0.3162f, 1.0000f, 1.0000f, 8.3900f, 1.3900f, 1.0000f, 0.8760f, 0.0020f, { 0.0000f, 0.0000f, 0.0000f }, 3.1081f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 1.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
#define EFX_REVERB_PRESET_DIZZY \
{ 0.3645f, 0.6000f, 0.3162f, 0.6310f, 1.0000f, 17.2300f, 0.5600f, 1.0000f, 0.1392f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 0.4937f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 1.0000f, 0.8100f, 0.3100f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
#define EFX_REVERB_PRESET_PSYCHOTIC \
{ 0.0625f, 0.5000f, 0.3162f, 0.8404f, 1.0000f, 7.5600f, 0.9100f, 1.0000f, 0.4864f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 2.4378f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 4.0000f, 1.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
/* Castle Presets */
#define EFX_REVERB_PRESET_CASTLE_SMALLROOM \
{ 1.0000f, 0.8900f, 0.3162f, 0.3981f, 0.1000f, 1.2200f, 0.8300f, 0.3100f, 0.8913f, 0.0220f, { 0.0000f, 0.0000f, 0.0000f }, 1.9953f, 0.0110f, { 0.0000f, 0.0000f, 0.0000f }, 0.1380f, 0.0800f, 0.2500f, 0.0000f, 0.9943f, 5168.6001f, 139.5000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_CASTLE_SHORTPASSAGE \
{ 1.0000f, 0.8900f, 0.3162f, 0.3162f, 0.1000f, 2.3200f, 0.8300f, 0.3100f, 0.8913f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0230f, { 0.0000f, 0.0000f, 0.0000f }, 0.1380f, 0.0800f, 0.2500f, 0.0000f, 0.9943f, 5168.6001f, 139.5000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_CASTLE_MEDIUMROOM \
{ 1.0000f, 0.9300f, 0.3162f, 0.2818f, 0.1000f, 2.0400f, 0.8300f, 0.4600f, 0.6310f, 0.0220f, { 0.0000f, 0.0000f, 0.0000f }, 1.5849f, 0.0110f, { 0.0000f, 0.0000f, 0.0000f }, 0.1550f, 0.0300f, 0.2500f, 0.0000f, 0.9943f, 5168.6001f, 139.5000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_CASTLE_LARGEROOM \
{ 1.0000f, 0.8200f, 0.3162f, 0.2818f, 0.1259f, 2.5300f, 0.8300f, 0.5000f, 0.4467f, 0.0340f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0160f, { 0.0000f, 0.0000f, 0.0000f }, 0.1850f, 0.0700f, 0.2500f, 0.0000f, 0.9943f, 5168.6001f, 139.5000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_CASTLE_LONGPASSAGE \
{ 1.0000f, 0.8900f, 0.3162f, 0.3981f, 0.1000f, 3.4200f, 0.8300f, 0.3100f, 0.8913f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 1.4125f, 0.0230f, { 0.0000f, 0.0000f, 0.0000f }, 0.1380f, 0.0800f, 0.2500f, 0.0000f, 0.9943f, 5168.6001f, 139.5000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_CASTLE_HALL \
{ 1.0000f, 0.8100f, 0.3162f, 0.2818f, 0.1778f, 3.1400f, 0.7900f, 0.6200f, 0.1778f, 0.0560f, { 0.0000f, 0.0000f, 0.0000f }, 1.1220f, 0.0240f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5168.6001f, 139.5000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_CASTLE_CUPBOARD \
{ 1.0000f, 0.8900f, 0.3162f, 0.2818f, 0.1000f, 0.6700f, 0.8700f, 0.3100f, 1.4125f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 3.5481f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 0.1380f, 0.0800f, 0.2500f, 0.0000f, 0.9943f, 5168.6001f, 139.5000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_CASTLE_COURTYARD \
{ 1.0000f, 0.4200f, 0.3162f, 0.4467f, 0.1995f, 2.1300f, 0.6100f, 0.2300f, 0.2239f, 0.1600f, { 0.0000f, 0.0000f, 0.0000f }, 0.7079f, 0.0360f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.3700f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
#define EFX_REVERB_PRESET_CASTLE_ALCOVE \
{ 1.0000f, 0.8900f, 0.3162f, 0.5012f, 0.1000f, 1.6400f, 0.8700f, 0.3100f, 1.0000f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 1.4125f, 0.0340f, { 0.0000f, 0.0000f, 0.0000f }, 0.1380f, 0.0800f, 0.2500f, 0.0000f, 0.9943f, 5168.6001f, 139.5000f, 0.0000f, 0x1 }
/* Factory Presets */
#define EFX_REVERB_PRESET_FACTORY_SMALLROOM \
{ 0.3645f, 0.8200f, 0.3162f, 0.7943f, 0.5012f, 1.7200f, 0.6500f, 1.3100f, 0.7079f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 1.7783f, 0.0240f, { 0.0000f, 0.0000f, 0.0000f }, 0.1190f, 0.0700f, 0.2500f, 0.0000f, 0.9943f, 3762.6001f, 362.5000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_FACTORY_SHORTPASSAGE \
{ 0.3645f, 0.6400f, 0.2512f, 0.7943f, 0.5012f, 2.5300f, 0.6500f, 1.3100f, 1.0000f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0380f, { 0.0000f, 0.0000f, 0.0000f }, 0.1350f, 0.2300f, 0.2500f, 0.0000f, 0.9943f, 3762.6001f, 362.5000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_FACTORY_MEDIUMROOM \
{ 0.4287f, 0.8200f, 0.2512f, 0.7943f, 0.5012f, 2.7600f, 0.6500f, 1.3100f, 0.2818f, 0.0220f, { 0.0000f, 0.0000f, 0.0000f }, 1.4125f, 0.0230f, { 0.0000f, 0.0000f, 0.0000f }, 0.1740f, 0.0700f, 0.2500f, 0.0000f, 0.9943f, 3762.6001f, 362.5000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_FACTORY_LARGEROOM \
{ 0.4287f, 0.7500f, 0.2512f, 0.7079f, 0.6310f, 4.2400f, 0.5100f, 1.3100f, 0.1778f, 0.0390f, { 0.0000f, 0.0000f, 0.0000f }, 1.1220f, 0.0230f, { 0.0000f, 0.0000f, 0.0000f }, 0.2310f, 0.0700f, 0.2500f, 0.0000f, 0.9943f, 3762.6001f, 362.5000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_FACTORY_LONGPASSAGE \
{ 0.3645f, 0.6400f, 0.2512f, 0.7943f, 0.5012f, 4.0600f, 0.6500f, 1.3100f, 1.0000f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0370f, { 0.0000f, 0.0000f, 0.0000f }, 0.1350f, 0.2300f, 0.2500f, 0.0000f, 0.9943f, 3762.6001f, 362.5000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_FACTORY_HALL \
{ 0.4287f, 0.7500f, 0.3162f, 0.7079f, 0.6310f, 7.4300f, 0.5100f, 1.3100f, 0.0631f, 0.0730f, { 0.0000f, 0.0000f, 0.0000f }, 0.8913f, 0.0270f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0700f, 0.2500f, 0.0000f, 0.9943f, 3762.6001f, 362.5000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_FACTORY_CUPBOARD \
{ 0.3071f, 0.6300f, 0.2512f, 0.7943f, 0.5012f, 0.4900f, 0.6500f, 1.3100f, 1.2589f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 1.9953f, 0.0320f, { 0.0000f, 0.0000f, 0.0000f }, 0.1070f, 0.0700f, 0.2500f, 0.0000f, 0.9943f, 3762.6001f, 362.5000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_FACTORY_COURTYARD \
{ 0.3071f, 0.5700f, 0.3162f, 0.3162f, 0.6310f, 2.3200f, 0.2900f, 0.5600f, 0.2239f, 0.1400f, { 0.0000f, 0.0000f, 0.0000f }, 0.3981f, 0.0390f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.2900f, 0.2500f, 0.0000f, 0.9943f, 3762.6001f, 362.5000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_FACTORY_ALCOVE \
{ 0.3645f, 0.5900f, 0.2512f, 0.7943f, 0.5012f, 3.1400f, 0.6500f, 1.3100f, 1.4125f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 1.0000f, 0.0380f, { 0.0000f, 0.0000f, 0.0000f }, 0.1140f, 0.1000f, 0.2500f, 0.0000f, 0.9943f, 3762.6001f, 362.5000f, 0.0000f, 0x1 }
/* Ice Palace Presets */
#define EFX_REVERB_PRESET_ICEPALACE_SMALLROOM \
{ 1.0000f, 0.8400f, 0.3162f, 0.5623f, 0.2818f, 1.5100f, 1.5300f, 0.2700f, 0.8913f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 1.4125f, 0.0110f, { 0.0000f, 0.0000f, 0.0000f }, 0.1640f, 0.1400f, 0.2500f, 0.0000f, 0.9943f, 12428.5000f, 99.6000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_ICEPALACE_SHORTPASSAGE \
{ 1.0000f, 0.7500f, 0.3162f, 0.5623f, 0.2818f, 1.7900f, 1.4600f, 0.2800f, 0.5012f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 1.1220f, 0.0190f, { 0.0000f, 0.0000f, 0.0000f }, 0.1770f, 0.0900f, 0.2500f, 0.0000f, 0.9943f, 12428.5000f, 99.6000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_ICEPALACE_MEDIUMROOM \
{ 1.0000f, 0.8700f, 0.3162f, 0.5623f, 0.4467f, 2.2200f, 1.5300f, 0.3200f, 0.3981f, 0.0390f, { 0.0000f, 0.0000f, 0.0000f }, 1.1220f, 0.0270f, { 0.0000f, 0.0000f, 0.0000f }, 0.1860f, 0.1200f, 0.2500f, 0.0000f, 0.9943f, 12428.5000f, 99.6000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_ICEPALACE_LARGEROOM \
{ 1.0000f, 0.8100f, 0.3162f, 0.5623f, 0.4467f, 3.1400f, 1.5300f, 0.3200f, 0.2512f, 0.0390f, { 0.0000f, 0.0000f, 0.0000f }, 1.0000f, 0.0270f, { 0.0000f, 0.0000f, 0.0000f }, 0.2140f, 0.1100f, 0.2500f, 0.0000f, 0.9943f, 12428.5000f, 99.6000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_ICEPALACE_LONGPASSAGE \
{ 1.0000f, 0.7700f, 0.3162f, 0.5623f, 0.3981f, 3.0100f, 1.4600f, 0.2800f, 0.7943f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0250f, { 0.0000f, 0.0000f, 0.0000f }, 0.1860f, 0.0400f, 0.2500f, 0.0000f, 0.9943f, 12428.5000f, 99.6000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_ICEPALACE_HALL \
{ 1.0000f, 0.7600f, 0.3162f, 0.4467f, 0.5623f, 5.4900f, 1.5300f, 0.3800f, 0.1122f, 0.0540f, { 0.0000f, 0.0000f, 0.0000f }, 0.6310f, 0.0520f, { 0.0000f, 0.0000f, 0.0000f }, 0.2260f, 0.1100f, 0.2500f, 0.0000f, 0.9943f, 12428.5000f, 99.6000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_ICEPALACE_CUPBOARD \
{ 1.0000f, 0.8300f, 0.3162f, 0.5012f, 0.2239f, 0.7600f, 1.5300f, 0.2600f, 1.1220f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 1.9953f, 0.0160f, { 0.0000f, 0.0000f, 0.0000f }, 0.1430f, 0.0800f, 0.2500f, 0.0000f, 0.9943f, 12428.5000f, 99.6000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_ICEPALACE_COURTYARD \
{ 1.0000f, 0.5900f, 0.3162f, 0.2818f, 0.3162f, 2.0400f, 1.2000f, 0.3800f, 0.3162f, 0.1730f, { 0.0000f, 0.0000f, 0.0000f }, 0.3162f, 0.0430f, { 0.0000f, 0.0000f, 0.0000f }, 0.2350f, 0.4800f, 0.2500f, 0.0000f, 0.9943f, 12428.5000f, 99.6000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_ICEPALACE_ALCOVE \
{ 1.0000f, 0.8400f, 0.3162f, 0.5623f, 0.2818f, 2.7600f, 1.4600f, 0.2800f, 1.1220f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 0.8913f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 0.1610f, 0.0900f, 0.2500f, 0.0000f, 0.9943f, 12428.5000f, 99.6000f, 0.0000f, 0x1 }
/* Space Station Presets */
#define EFX_REVERB_PRESET_SPACESTATION_SMALLROOM \
{ 0.2109f, 0.7000f, 0.3162f, 0.7079f, 0.8913f, 1.7200f, 0.8200f, 0.5500f, 0.7943f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 1.4125f, 0.0130f, { 0.0000f, 0.0000f, 0.0000f }, 0.1880f, 0.2600f, 0.2500f, 0.0000f, 0.9943f, 3316.1001f, 458.2000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_SPACESTATION_SHORTPASSAGE \
{ 0.2109f, 0.8700f, 0.3162f, 0.6310f, 0.8913f, 3.5700f, 0.5000f, 0.5500f, 1.0000f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 1.1220f, 0.0160f, { 0.0000f, 0.0000f, 0.0000f }, 0.1720f, 0.2000f, 0.2500f, 0.0000f, 0.9943f, 3316.1001f, 458.2000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_SPACESTATION_MEDIUMROOM \
{ 0.2109f, 0.7500f, 0.3162f, 0.6310f, 0.8913f, 3.0100f, 0.5000f, 0.5500f, 0.3981f, 0.0340f, { 0.0000f, 0.0000f, 0.0000f }, 1.1220f, 0.0350f, { 0.0000f, 0.0000f, 0.0000f }, 0.2090f, 0.3100f, 0.2500f, 0.0000f, 0.9943f, 3316.1001f, 458.2000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_SPACESTATION_LARGEROOM \
{ 0.3645f, 0.8100f, 0.3162f, 0.6310f, 0.8913f, 3.8900f, 0.3800f, 0.6100f, 0.3162f, 0.0560f, { 0.0000f, 0.0000f, 0.0000f }, 0.8913f, 0.0350f, { 0.0000f, 0.0000f, 0.0000f }, 0.2330f, 0.2800f, 0.2500f, 0.0000f, 0.9943f, 3316.1001f, 458.2000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_SPACESTATION_LONGPASSAGE \
{ 0.4287f, 0.8200f, 0.3162f, 0.6310f, 0.8913f, 4.6200f, 0.6200f, 0.5500f, 1.0000f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0310f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.2300f, 0.2500f, 0.0000f, 0.9943f, 3316.1001f, 458.2000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_SPACESTATION_HALL \
{ 0.4287f, 0.8700f, 0.3162f, 0.6310f, 0.8913f, 7.1100f, 0.3800f, 0.6100f, 0.1778f, 0.1000f, { 0.0000f, 0.0000f, 0.0000f }, 0.6310f, 0.0470f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.2500f, 0.2500f, 0.0000f, 0.9943f, 3316.1001f, 458.2000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_SPACESTATION_CUPBOARD \
{ 0.1715f, 0.5600f, 0.3162f, 0.7079f, 0.8913f, 0.7900f, 0.8100f, 0.5500f, 1.4125f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 1.7783f, 0.0180f, { 0.0000f, 0.0000f, 0.0000f }, 0.1810f, 0.3100f, 0.2500f, 0.0000f, 0.9943f, 3316.1001f, 458.2000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_SPACESTATION_ALCOVE \
{ 0.2109f, 0.7800f, 0.3162f, 0.7079f, 0.8913f, 1.1600f, 0.8100f, 0.5500f, 1.4125f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 1.0000f, 0.0180f, { 0.0000f, 0.0000f, 0.0000f }, 0.1920f, 0.2100f, 0.2500f, 0.0000f, 0.9943f, 3316.1001f, 458.2000f, 0.0000f, 0x1 }
/* Wooden Galleon Presets */
#define EFX_REVERB_PRESET_WOODEN_SMALLROOM \
{ 1.0000f, 1.0000f, 0.3162f, 0.1122f, 0.3162f, 0.7900f, 0.3200f, 0.8700f, 1.0000f, 0.0320f, { 0.0000f, 0.0000f, 0.0000f }, 0.8913f, 0.0290f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 4705.0000f, 99.6000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_WOODEN_SHORTPASSAGE \
{ 1.0000f, 1.0000f, 0.3162f, 0.1259f, 0.3162f, 1.7500f, 0.5000f, 0.8700f, 0.8913f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 0.6310f, 0.0240f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 4705.0000f, 99.6000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_WOODEN_MEDIUMROOM \
{ 1.0000f, 1.0000f, 0.3162f, 0.1000f, 0.2818f, 1.4700f, 0.4200f, 0.8200f, 0.8913f, 0.0490f, { 0.0000f, 0.0000f, 0.0000f }, 0.8913f, 0.0290f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 4705.0000f, 99.6000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_WOODEN_LARGEROOM \
{ 1.0000f, 1.0000f, 0.3162f, 0.0891f, 0.2818f, 2.6500f, 0.3300f, 0.8200f, 0.8913f, 0.0660f, { 0.0000f, 0.0000f, 0.0000f }, 0.7943f, 0.0490f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 4705.0000f, 99.6000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_WOODEN_LONGPASSAGE \
{ 1.0000f, 1.0000f, 0.3162f, 0.1000f, 0.3162f, 1.9900f, 0.4000f, 0.7900f, 1.0000f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 0.4467f, 0.0360f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 4705.0000f, 99.6000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_WOODEN_HALL \
{ 1.0000f, 1.0000f, 0.3162f, 0.0794f, 0.2818f, 3.4500f, 0.3000f, 0.8200f, 0.8913f, 0.0880f, { 0.0000f, 0.0000f, 0.0000f }, 0.7943f, 0.0630f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 4705.0000f, 99.6000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_WOODEN_CUPBOARD \
{ 1.0000f, 1.0000f, 0.3162f, 0.1413f, 0.3162f, 0.5600f, 0.4600f, 0.9100f, 1.1220f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 1.1220f, 0.0280f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 4705.0000f, 99.6000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_WOODEN_COURTYARD \
{ 1.0000f, 0.6500f, 0.3162f, 0.0794f, 0.3162f, 1.7900f, 0.3500f, 0.7900f, 0.5623f, 0.1230f, { 0.0000f, 0.0000f, 0.0000f }, 0.1000f, 0.0320f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 4705.0000f, 99.6000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_WOODEN_ALCOVE \
{ 1.0000f, 1.0000f, 0.3162f, 0.1259f, 0.3162f, 1.2200f, 0.6200f, 0.9100f, 1.1220f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 0.7079f, 0.0240f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 4705.0000f, 99.6000f, 0.0000f, 0x1 }
/* Sports Presets */
#define EFX_REVERB_PRESET_SPORT_EMPTYSTADIUM \
{ 1.0000f, 1.0000f, 0.3162f, 0.4467f, 0.7943f, 6.2600f, 0.5100f, 1.1000f, 0.0631f, 0.1830f, { 0.0000f, 0.0000f, 0.0000f }, 0.3981f, 0.0380f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_SPORT_SQUASHCOURT \
{ 1.0000f, 0.7500f, 0.3162f, 0.3162f, 0.7943f, 2.2200f, 0.9100f, 1.1600f, 0.4467f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 0.7943f, 0.0110f, { 0.0000f, 0.0000f, 0.0000f }, 0.1260f, 0.1900f, 0.2500f, 0.0000f, 0.9943f, 7176.8999f, 211.2000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_SPORT_SMALLSWIMMINGPOOL \
{ 1.0000f, 0.7000f, 0.3162f, 0.7943f, 0.8913f, 2.7600f, 1.2500f, 1.1400f, 0.6310f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 0.7943f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 0.1790f, 0.1500f, 0.8950f, 0.1900f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
#define EFX_REVERB_PRESET_SPORT_LARGESWIMMINGPOOL \
{ 1.0000f, 0.8200f, 0.3162f, 0.7943f, 1.0000f, 5.4900f, 1.3100f, 1.1400f, 0.4467f, 0.0390f, { 0.0000f, 0.0000f, 0.0000f }, 0.5012f, 0.0490f, { 0.0000f, 0.0000f, 0.0000f }, 0.2220f, 0.5500f, 1.1590f, 0.2100f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
#define EFX_REVERB_PRESET_SPORT_GYMNASIUM \
{ 1.0000f, 0.8100f, 0.3162f, 0.4467f, 0.8913f, 3.1400f, 1.0600f, 1.3500f, 0.3981f, 0.0290f, { 0.0000f, 0.0000f, 0.0000f }, 0.5623f, 0.0450f, { 0.0000f, 0.0000f, 0.0000f }, 0.1460f, 0.1400f, 0.2500f, 0.0000f, 0.9943f, 7176.8999f, 211.2000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_SPORT_FULLSTADIUM \
{ 1.0000f, 1.0000f, 0.3162f, 0.0708f, 0.7943f, 5.2500f, 0.1700f, 0.8000f, 0.1000f, 0.1880f, { 0.0000f, 0.0000f, 0.0000f }, 0.2818f, 0.0380f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_SPORT_STADIUMTANNOY \
{ 1.0000f, 0.7800f, 0.3162f, 0.5623f, 0.5012f, 2.5300f, 0.8800f, 0.6800f, 0.2818f, 0.2300f, { 0.0000f, 0.0000f, 0.0000f }, 0.5012f, 0.0630f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.2000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
/* Prefab Presets */
#define EFX_REVERB_PRESET_PREFAB_WORKSHOP \
{ 0.4287f, 1.0000f, 0.3162f, 0.1413f, 0.3981f, 0.7600f, 1.0000f, 1.0000f, 1.0000f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 1.1220f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
#define EFX_REVERB_PRESET_PREFAB_SCHOOLROOM \
{ 0.4022f, 0.6900f, 0.3162f, 0.6310f, 0.5012f, 0.9800f, 0.4500f, 0.1800f, 1.4125f, 0.0170f, { 0.0000f, 0.0000f, 0.0000f }, 1.4125f, 0.0150f, { 0.0000f, 0.0000f, 0.0000f }, 0.0950f, 0.1400f, 0.2500f, 0.0000f, 0.9943f, 7176.8999f, 211.2000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_PREFAB_PRACTISEROOM \
{ 0.4022f, 0.8700f, 0.3162f, 0.3981f, 0.5012f, 1.1200f, 0.5600f, 0.1800f, 1.2589f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 1.4125f, 0.0110f, { 0.0000f, 0.0000f, 0.0000f }, 0.0950f, 0.1400f, 0.2500f, 0.0000f, 0.9943f, 7176.8999f, 211.2000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_PREFAB_OUTHOUSE \
{ 1.0000f, 0.8200f, 0.3162f, 0.1122f, 0.1585f, 1.3800f, 0.3800f, 0.3500f, 0.8913f, 0.0240f, { 0.0000f, 0.0000f, -0.0000f }, 0.6310f, 0.0440f, { 0.0000f, 0.0000f, 0.0000f }, 0.1210f, 0.1700f, 0.2500f, 0.0000f, 0.9943f, 2854.3999f, 107.5000f, 0.0000f, 0x0 }
#define EFX_REVERB_PRESET_PREFAB_CARAVAN \
{ 1.0000f, 1.0000f, 0.3162f, 0.0891f, 0.1259f, 0.4300f, 1.5000f, 1.0000f, 1.0000f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 1.9953f, 0.0120f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
/* Dome and Pipe Presets */
#define EFX_REVERB_PRESET_DOME_TOMB \
{ 1.0000f, 0.7900f, 0.3162f, 0.3548f, 0.2239f, 4.1800f, 0.2100f, 0.1000f, 0.3868f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 1.6788f, 0.0220f, { 0.0000f, 0.0000f, 0.0000f }, 0.1770f, 0.1900f, 0.2500f, 0.0000f, 0.9943f, 2854.3999f, 20.0000f, 0.0000f, 0x0 }
#define EFX_REVERB_PRESET_PIPE_SMALL \
{ 1.0000f, 1.0000f, 0.3162f, 0.3548f, 0.2239f, 5.0400f, 0.1000f, 0.1000f, 0.5012f, 0.0320f, { 0.0000f, 0.0000f, 0.0000f }, 2.5119f, 0.0150f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 2854.3999f, 20.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_DOME_SAINTPAULS \
{ 1.0000f, 0.8700f, 0.3162f, 0.3548f, 0.2239f, 10.4800f, 0.1900f, 0.1000f, 0.1778f, 0.0900f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0420f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.1200f, 0.2500f, 0.0000f, 0.9943f, 2854.3999f, 20.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_PIPE_LONGTHIN \
{ 0.2560f, 0.9100f, 0.3162f, 0.4467f, 0.2818f, 9.2100f, 0.1800f, 0.1000f, 0.7079f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 0.7079f, 0.0220f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 2854.3999f, 20.0000f, 0.0000f, 0x0 }
#define EFX_REVERB_PRESET_PIPE_LARGE \
{ 1.0000f, 1.0000f, 0.3162f, 0.3548f, 0.2239f, 8.4500f, 0.1000f, 0.1000f, 0.3981f, 0.0460f, { 0.0000f, 0.0000f, 0.0000f }, 1.5849f, 0.0320f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 2854.3999f, 20.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_PIPE_RESONANT \
{ 0.1373f, 0.9100f, 0.3162f, 0.4467f, 0.2818f, 6.8100f, 0.1800f, 0.1000f, 0.7079f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 1.0000f, 0.0220f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 2854.3999f, 20.0000f, 0.0000f, 0x0 }
/* Outdoors Presets */
#define EFX_REVERB_PRESET_OUTDOORS_BACKYARD \
{ 1.0000f, 0.4500f, 0.3162f, 0.2512f, 0.5012f, 1.1200f, 0.3400f, 0.4600f, 0.4467f, 0.0690f, { 0.0000f, 0.0000f, -0.0000f }, 0.7079f, 0.0230f, { 0.0000f, 0.0000f, 0.0000f }, 0.2180f, 0.3400f, 0.2500f, 0.0000f, 0.9943f, 4399.1001f, 242.9000f, 0.0000f, 0x0 }
#define EFX_REVERB_PRESET_OUTDOORS_ROLLINGPLAINS \
{ 1.0000f, 0.0000f, 0.3162f, 0.0112f, 0.6310f, 2.1300f, 0.2100f, 0.4600f, 0.1778f, 0.3000f, { 0.0000f, 0.0000f, -0.0000f }, 0.4467f, 0.0190f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 1.0000f, 0.2500f, 0.0000f, 0.9943f, 4399.1001f, 242.9000f, 0.0000f, 0x0 }
#define EFX_REVERB_PRESET_OUTDOORS_DEEPCANYON \
{ 1.0000f, 0.7400f, 0.3162f, 0.1778f, 0.6310f, 3.8900f, 0.2100f, 0.4600f, 0.3162f, 0.2230f, { 0.0000f, 0.0000f, -0.0000f }, 0.3548f, 0.0190f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 1.0000f, 0.2500f, 0.0000f, 0.9943f, 4399.1001f, 242.9000f, 0.0000f, 0x0 }
#define EFX_REVERB_PRESET_OUTDOORS_CREEK \
{ 1.0000f, 0.3500f, 0.3162f, 0.1778f, 0.5012f, 2.1300f, 0.2100f, 0.4600f, 0.3981f, 0.1150f, { 0.0000f, 0.0000f, -0.0000f }, 0.1995f, 0.0310f, { 0.0000f, 0.0000f, 0.0000f }, 0.2180f, 0.3400f, 0.2500f, 0.0000f, 0.9943f, 4399.1001f, 242.9000f, 0.0000f, 0x0 }
#define EFX_REVERB_PRESET_OUTDOORS_VALLEY \
{ 1.0000f, 0.2800f, 0.3162f, 0.0282f, 0.1585f, 2.8800f, 0.2600f, 0.3500f, 0.1413f, 0.2630f, { 0.0000f, 0.0000f, -0.0000f }, 0.3981f, 0.1000f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.3400f, 0.2500f, 0.0000f, 0.9943f, 2854.3999f, 107.5000f, 0.0000f, 0x0 }
/* Mood Presets */
#define EFX_REVERB_PRESET_MOOD_HEAVEN \
{ 1.0000f, 0.9400f, 0.3162f, 0.7943f, 0.4467f, 5.0400f, 1.1200f, 0.5600f, 0.2427f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0290f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0800f, 2.7420f, 0.0500f, 0.9977f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_MOOD_HELL \
{ 1.0000f, 0.5700f, 0.3162f, 0.3548f, 0.4467f, 3.5700f, 0.4900f, 2.0000f, 0.0000f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 1.4125f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 0.1100f, 0.0400f, 2.1090f, 0.5200f, 0.9943f, 5000.0000f, 139.5000f, 0.0000f, 0x0 }
#define EFX_REVERB_PRESET_MOOD_MEMORY \
{ 1.0000f, 0.8500f, 0.3162f, 0.6310f, 0.3548f, 4.0600f, 0.8200f, 0.5600f, 0.0398f, 0.0000f, { 0.0000f, 0.0000f, 0.0000f }, 1.1220f, 0.0000f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.4740f, 0.4500f, 0.9886f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
/* Driving Presets */
#define EFX_REVERB_PRESET_DRIVING_COMMENTATOR \
{ 1.0000f, 0.0000f, 3.1623f, 0.5623f, 0.5012f, 2.4200f, 0.8800f, 0.6800f, 0.1995f, 0.0930f, { 0.0000f, 0.0000f, 0.0000f }, 0.2512f, 0.0170f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 1.0000f, 0.2500f, 0.0000f, 0.9886f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_DRIVING_PITGARAGE \
{ 0.4287f, 0.5900f, 0.3162f, 0.7079f, 0.5623f, 1.7200f, 0.9300f, 0.8700f, 0.5623f, 0.0000f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0160f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.1100f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
#define EFX_REVERB_PRESET_DRIVING_INCAR_RACER \
{ 0.0832f, 0.8000f, 0.3162f, 1.0000f, 0.7943f, 0.1700f, 2.0000f, 0.4100f, 1.7783f, 0.0070f, { 0.0000f, 0.0000f, 0.0000f }, 0.7079f, 0.0150f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 10268.2002f, 251.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_DRIVING_INCAR_SPORTS \
{ 0.0832f, 0.8000f, 0.3162f, 0.6310f, 1.0000f, 0.1700f, 0.7500f, 0.4100f, 1.0000f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 0.5623f, 0.0000f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 10268.2002f, 251.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_DRIVING_INCAR_LUXURY \
{ 0.2560f, 1.0000f, 0.3162f, 0.1000f, 0.5012f, 0.1300f, 0.4100f, 0.4600f, 0.7943f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 1.5849f, 0.0100f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 10268.2002f, 251.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_DRIVING_FULLGRANDSTAND \
{ 1.0000f, 1.0000f, 0.3162f, 0.2818f, 0.6310f, 3.0100f, 1.3700f, 1.2800f, 0.3548f, 0.0900f, { 0.0000f, 0.0000f, 0.0000f }, 0.1778f, 0.0490f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 10420.2002f, 250.0000f, 0.0000f, 0x0 }
#define EFX_REVERB_PRESET_DRIVING_EMPTYGRANDSTAND \
{ 1.0000f, 1.0000f, 0.3162f, 1.0000f, 0.7943f, 4.6200f, 1.7500f, 1.4000f, 0.2082f, 0.0900f, { 0.0000f, 0.0000f, 0.0000f }, 0.2512f, 0.0490f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.0000f, 0.9943f, 10420.2002f, 250.0000f, 0.0000f, 0x0 }
#define EFX_REVERB_PRESET_DRIVING_TUNNEL \
{ 1.0000f, 0.8100f, 0.3162f, 0.3981f, 0.8913f, 3.4200f, 0.9400f, 1.3100f, 0.7079f, 0.0510f, { 0.0000f, 0.0000f, 0.0000f }, 0.7079f, 0.0470f, { 0.0000f, 0.0000f, 0.0000f }, 0.2140f, 0.0500f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 155.3000f, 0.0000f, 0x1 }
/* City Presets */
#define EFX_REVERB_PRESET_CITY_STREETS \
{ 1.0000f, 0.7800f, 0.3162f, 0.7079f, 0.8913f, 1.7900f, 1.1200f, 0.9100f, 0.2818f, 0.0460f, { 0.0000f, 0.0000f, 0.0000f }, 0.1995f, 0.0280f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.2000f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_CITY_SUBWAY \
{ 1.0000f, 0.7400f, 0.3162f, 0.7079f, 0.8913f, 3.0100f, 1.2300f, 0.9100f, 0.7079f, 0.0460f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0280f, { 0.0000f, 0.0000f, 0.0000f }, 0.1250f, 0.2100f, 0.2500f, 0.0000f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_CITY_MUSEUM \
{ 1.0000f, 0.8200f, 0.3162f, 0.1778f, 0.1778f, 3.2800f, 1.4000f, 0.5700f, 0.2512f, 0.0390f, { 0.0000f, 0.0000f, -0.0000f }, 0.8913f, 0.0340f, { 0.0000f, 0.0000f, 0.0000f }, 0.1300f, 0.1700f, 0.2500f, 0.0000f, 0.9943f, 2854.3999f, 107.5000f, 0.0000f, 0x0 }
#define EFX_REVERB_PRESET_CITY_LIBRARY \
{ 1.0000f, 0.8200f, 0.3162f, 0.2818f, 0.0891f, 2.7600f, 0.8900f, 0.4100f, 0.3548f, 0.0290f, { 0.0000f, 0.0000f, -0.0000f }, 0.8913f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 0.1300f, 0.1700f, 0.2500f, 0.0000f, 0.9943f, 2854.3999f, 107.5000f, 0.0000f, 0x0 }
#define EFX_REVERB_PRESET_CITY_UNDERPASS \
{ 1.0000f, 0.8200f, 0.3162f, 0.4467f, 0.8913f, 3.5700f, 1.1200f, 0.9100f, 0.3981f, 0.0590f, { 0.0000f, 0.0000f, 0.0000f }, 0.8913f, 0.0370f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.1400f, 0.2500f, 0.0000f, 0.9920f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_CITY_ABANDONED \
{ 1.0000f, 0.6900f, 0.3162f, 0.7943f, 0.8913f, 3.2800f, 1.1700f, 0.9100f, 0.4467f, 0.0440f, { 0.0000f, 0.0000f, 0.0000f }, 0.2818f, 0.0240f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.2000f, 0.2500f, 0.0000f, 0.9966f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
/* Misc. Presets */
#define EFX_REVERB_PRESET_DUSTYROOM \
{ 0.3645f, 0.5600f, 0.3162f, 0.7943f, 0.7079f, 1.7900f, 0.3800f, 0.2100f, 0.5012f, 0.0020f, { 0.0000f, 0.0000f, 0.0000f }, 1.2589f, 0.0060f, { 0.0000f, 0.0000f, 0.0000f }, 0.2020f, 0.0500f, 0.2500f, 0.0000f, 0.9886f, 13046.0000f, 163.3000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_CHAPEL \
{ 1.0000f, 0.8400f, 0.3162f, 0.5623f, 1.0000f, 4.6200f, 0.6400f, 1.2300f, 0.4467f, 0.0320f, { 0.0000f, 0.0000f, 0.0000f }, 0.7943f, 0.0490f, { 0.0000f, 0.0000f, 0.0000f }, 0.2500f, 0.0000f, 0.2500f, 0.1100f, 0.9943f, 5000.0000f, 250.0000f, 0.0000f, 0x1 }
#define EFX_REVERB_PRESET_SMALLWATERROOM \
{ 1.0000f, 0.7000f, 0.3162f, 0.4477f, 1.0000f, 1.5100f, 1.2500f, 1.1400f, 0.8913f, 0.0200f, { 0.0000f, 0.0000f, 0.0000f }, 1.4125f, 0.0300f, { 0.0000f, 0.0000f, 0.0000f }, 0.1790f, 0.1500f, 0.8950f, 0.1900f, 0.9920f, 5000.0000f, 250.0000f, 0.0000f, 0x0 }
#endif /* EFX_PRESETS_H */
+761
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#ifndef AL_EFX_H
#define AL_EFX_H
#include "alc.h"
#include "al.h"
#ifdef __cplusplus
extern "C" {
#endif
#define ALC_EXT_EFX_NAME "ALC_EXT_EFX"
#define ALC_EFX_MAJOR_VERSION 0x20001
#define ALC_EFX_MINOR_VERSION 0x20002
#define ALC_MAX_AUXILIARY_SENDS 0x20003
/* Listener properties. */
#define AL_METERS_PER_UNIT 0x20004
/* Source properties. */
#define AL_DIRECT_FILTER 0x20005
#define AL_AUXILIARY_SEND_FILTER 0x20006
#define AL_AIR_ABSORPTION_FACTOR 0x20007
#define AL_ROOM_ROLLOFF_FACTOR 0x20008
#define AL_CONE_OUTER_GAINHF 0x20009
#define AL_DIRECT_FILTER_GAINHF_AUTO 0x2000A
#define AL_AUXILIARY_SEND_FILTER_GAIN_AUTO 0x2000B
#define AL_AUXILIARY_SEND_FILTER_GAINHF_AUTO 0x2000C
/* Effect properties. */
/* Reverb effect parameters */
#define AL_REVERB_DENSITY 0x0001
#define AL_REVERB_DIFFUSION 0x0002
#define AL_REVERB_GAIN 0x0003
#define AL_REVERB_GAINHF 0x0004
#define AL_REVERB_DECAY_TIME 0x0005
#define AL_REVERB_DECAY_HFRATIO 0x0006
#define AL_REVERB_REFLECTIONS_GAIN 0x0007
#define AL_REVERB_REFLECTIONS_DELAY 0x0008
#define AL_REVERB_LATE_REVERB_GAIN 0x0009
#define AL_REVERB_LATE_REVERB_DELAY 0x000A
#define AL_REVERB_AIR_ABSORPTION_GAINHF 0x000B
#define AL_REVERB_ROOM_ROLLOFF_FACTOR 0x000C
#define AL_REVERB_DECAY_HFLIMIT 0x000D
/* EAX Reverb effect parameters */
#define AL_EAXREVERB_DENSITY 0x0001
#define AL_EAXREVERB_DIFFUSION 0x0002
#define AL_EAXREVERB_GAIN 0x0003
#define AL_EAXREVERB_GAINHF 0x0004
#define AL_EAXREVERB_GAINLF 0x0005
#define AL_EAXREVERB_DECAY_TIME 0x0006
#define AL_EAXREVERB_DECAY_HFRATIO 0x0007
#define AL_EAXREVERB_DECAY_LFRATIO 0x0008
#define AL_EAXREVERB_REFLECTIONS_GAIN 0x0009
#define AL_EAXREVERB_REFLECTIONS_DELAY 0x000A
#define AL_EAXREVERB_REFLECTIONS_PAN 0x000B
#define AL_EAXREVERB_LATE_REVERB_GAIN 0x000C
#define AL_EAXREVERB_LATE_REVERB_DELAY 0x000D
#define AL_EAXREVERB_LATE_REVERB_PAN 0x000E
#define AL_EAXREVERB_ECHO_TIME 0x000F
#define AL_EAXREVERB_ECHO_DEPTH 0x0010
#define AL_EAXREVERB_MODULATION_TIME 0x0011
#define AL_EAXREVERB_MODULATION_DEPTH 0x0012
#define AL_EAXREVERB_AIR_ABSORPTION_GAINHF 0x0013
#define AL_EAXREVERB_HFREFERENCE 0x0014
#define AL_EAXREVERB_LFREFERENCE 0x0015
#define AL_EAXREVERB_ROOM_ROLLOFF_FACTOR 0x0016
#define AL_EAXREVERB_DECAY_HFLIMIT 0x0017
/* Chorus effect parameters */
#define AL_CHORUS_WAVEFORM 0x0001
#define AL_CHORUS_PHASE 0x0002
#define AL_CHORUS_RATE 0x0003
#define AL_CHORUS_DEPTH 0x0004
#define AL_CHORUS_FEEDBACK 0x0005
#define AL_CHORUS_DELAY 0x0006
/* Distortion effect parameters */
#define AL_DISTORTION_EDGE 0x0001
#define AL_DISTORTION_GAIN 0x0002
#define AL_DISTORTION_LOWPASS_CUTOFF 0x0003
#define AL_DISTORTION_EQCENTER 0x0004
#define AL_DISTORTION_EQBANDWIDTH 0x0005
/* Echo effect parameters */
#define AL_ECHO_DELAY 0x0001
#define AL_ECHO_LRDELAY 0x0002
#define AL_ECHO_DAMPING 0x0003
#define AL_ECHO_FEEDBACK 0x0004
#define AL_ECHO_SPREAD 0x0005
/* Flanger effect parameters */
#define AL_FLANGER_WAVEFORM 0x0001
#define AL_FLANGER_PHASE 0x0002
#define AL_FLANGER_RATE 0x0003
#define AL_FLANGER_DEPTH 0x0004
#define AL_FLANGER_FEEDBACK 0x0005
#define AL_FLANGER_DELAY 0x0006
/* Frequency shifter effect parameters */
#define AL_FREQUENCY_SHIFTER_FREQUENCY 0x0001
#define AL_FREQUENCY_SHIFTER_LEFT_DIRECTION 0x0002
#define AL_FREQUENCY_SHIFTER_RIGHT_DIRECTION 0x0003
/* Vocal morpher effect parameters */
#define AL_VOCAL_MORPHER_PHONEMEA 0x0001
#define AL_VOCAL_MORPHER_PHONEMEA_COARSE_TUNING 0x0002
#define AL_VOCAL_MORPHER_PHONEMEB 0x0003
#define AL_VOCAL_MORPHER_PHONEMEB_COARSE_TUNING 0x0004
#define AL_VOCAL_MORPHER_WAVEFORM 0x0005
#define AL_VOCAL_MORPHER_RATE 0x0006
/* Pitchshifter effect parameters */
#define AL_PITCH_SHIFTER_COARSE_TUNE 0x0001
#define AL_PITCH_SHIFTER_FINE_TUNE 0x0002
/* Ringmodulator effect parameters */
#define AL_RING_MODULATOR_FREQUENCY 0x0001
#define AL_RING_MODULATOR_HIGHPASS_CUTOFF 0x0002
#define AL_RING_MODULATOR_WAVEFORM 0x0003
/* Autowah effect parameters */
#define AL_AUTOWAH_ATTACK_TIME 0x0001
#define AL_AUTOWAH_RELEASE_TIME 0x0002
#define AL_AUTOWAH_RESONANCE 0x0003
#define AL_AUTOWAH_PEAK_GAIN 0x0004
/* Compressor effect parameters */
#define AL_COMPRESSOR_ONOFF 0x0001
/* Equalizer effect parameters */
#define AL_EQUALIZER_LOW_GAIN 0x0001
#define AL_EQUALIZER_LOW_CUTOFF 0x0002
#define AL_EQUALIZER_MID1_GAIN 0x0003
#define AL_EQUALIZER_MID1_CENTER 0x0004
#define AL_EQUALIZER_MID1_WIDTH 0x0005
#define AL_EQUALIZER_MID2_GAIN 0x0006
#define AL_EQUALIZER_MID2_CENTER 0x0007
#define AL_EQUALIZER_MID2_WIDTH 0x0008
#define AL_EQUALIZER_HIGH_GAIN 0x0009
#define AL_EQUALIZER_HIGH_CUTOFF 0x000A
/* Effect type */
#define AL_EFFECT_FIRST_PARAMETER 0x0000
#define AL_EFFECT_LAST_PARAMETER 0x8000
#define AL_EFFECT_TYPE 0x8001
/* Effect types, used with the AL_EFFECT_TYPE property */
#define AL_EFFECT_NULL 0x0000
#define AL_EFFECT_REVERB 0x0001
#define AL_EFFECT_CHORUS 0x0002
#define AL_EFFECT_DISTORTION 0x0003
#define AL_EFFECT_ECHO 0x0004
#define AL_EFFECT_FLANGER 0x0005
#define AL_EFFECT_FREQUENCY_SHIFTER 0x0006
#define AL_EFFECT_VOCAL_MORPHER 0x0007
#define AL_EFFECT_PITCH_SHIFTER 0x0008
#define AL_EFFECT_RING_MODULATOR 0x0009
#define AL_EFFECT_AUTOWAH 0x000A
#define AL_EFFECT_COMPRESSOR 0x000B
#define AL_EFFECT_EQUALIZER 0x000C
#define AL_EFFECT_EAXREVERB 0x8000
/* Auxiliary Effect Slot properties. */
#define AL_EFFECTSLOT_EFFECT 0x0001
#define AL_EFFECTSLOT_GAIN 0x0002
#define AL_EFFECTSLOT_AUXILIARY_SEND_AUTO 0x0003
/* NULL Auxiliary Slot ID to disable a source send. */
#define AL_EFFECTSLOT_NULL 0x0000
/* Filter properties. */
/* Lowpass filter parameters */
#define AL_LOWPASS_GAIN 0x0001
#define AL_LOWPASS_GAINHF 0x0002
/* Highpass filter parameters */
#define AL_HIGHPASS_GAIN 0x0001
#define AL_HIGHPASS_GAINLF 0x0002
/* Bandpass filter parameters */
#define AL_BANDPASS_GAIN 0x0001
#define AL_BANDPASS_GAINLF 0x0002
#define AL_BANDPASS_GAINHF 0x0003
/* Filter type */
#define AL_FILTER_FIRST_PARAMETER 0x0000
#define AL_FILTER_LAST_PARAMETER 0x8000
#define AL_FILTER_TYPE 0x8001
/* Filter types, used with the AL_FILTER_TYPE property */
#define AL_FILTER_NULL 0x0000
#define AL_FILTER_LOWPASS 0x0001
#define AL_FILTER_HIGHPASS 0x0002
#define AL_FILTER_BANDPASS 0x0003
/* Effect object function types. */
typedef void (AL_APIENTRY *LPALGENEFFECTS)(ALsizei, ALuint*);
typedef void (AL_APIENTRY *LPALDELETEEFFECTS)(ALsizei, const ALuint*);
typedef ALboolean (AL_APIENTRY *LPALISEFFECT)(ALuint);
typedef void (AL_APIENTRY *LPALEFFECTI)(ALuint, ALenum, ALint);
typedef void (AL_APIENTRY *LPALEFFECTIV)(ALuint, ALenum, const ALint*);
typedef void (AL_APIENTRY *LPALEFFECTF)(ALuint, ALenum, ALfloat);
typedef void (AL_APIENTRY *LPALEFFECTFV)(ALuint, ALenum, const ALfloat*);
typedef void (AL_APIENTRY *LPALGETEFFECTI)(ALuint, ALenum, ALint*);
typedef void (AL_APIENTRY *LPALGETEFFECTIV)(ALuint, ALenum, ALint*);
typedef void (AL_APIENTRY *LPALGETEFFECTF)(ALuint, ALenum, ALfloat*);
typedef void (AL_APIENTRY *LPALGETEFFECTFV)(ALuint, ALenum, ALfloat*);
/* Filter object function types. */
typedef void (AL_APIENTRY *LPALGENFILTERS)(ALsizei, ALuint*);
typedef void (AL_APIENTRY *LPALDELETEFILTERS)(ALsizei, const ALuint*);
typedef ALboolean (AL_APIENTRY *LPALISFILTER)(ALuint);
typedef void (AL_APIENTRY *LPALFILTERI)(ALuint, ALenum, ALint);
typedef void (AL_APIENTRY *LPALFILTERIV)(ALuint, ALenum, const ALint*);
typedef void (AL_APIENTRY *LPALFILTERF)(ALuint, ALenum, ALfloat);
typedef void (AL_APIENTRY *LPALFILTERFV)(ALuint, ALenum, const ALfloat*);
typedef void (AL_APIENTRY *LPALGETFILTERI)(ALuint, ALenum, ALint*);
typedef void (AL_APIENTRY *LPALGETFILTERIV)(ALuint, ALenum, ALint*);
typedef void (AL_APIENTRY *LPALGETFILTERF)(ALuint, ALenum, ALfloat*);
typedef void (AL_APIENTRY *LPALGETFILTERFV)(ALuint, ALenum, ALfloat*);
/* Auxiliary Effect Slot object function types. */
typedef void (AL_APIENTRY *LPALGENAUXILIARYEFFECTSLOTS)(ALsizei, ALuint*);
typedef void (AL_APIENTRY *LPALDELETEAUXILIARYEFFECTSLOTS)(ALsizei, const ALuint*);
typedef ALboolean (AL_APIENTRY *LPALISAUXILIARYEFFECTSLOT)(ALuint);
typedef void (AL_APIENTRY *LPALAUXILIARYEFFECTSLOTI)(ALuint, ALenum, ALint);
typedef void (AL_APIENTRY *LPALAUXILIARYEFFECTSLOTIV)(ALuint, ALenum, const ALint*);
typedef void (AL_APIENTRY *LPALAUXILIARYEFFECTSLOTF)(ALuint, ALenum, ALfloat);
typedef void (AL_APIENTRY *LPALAUXILIARYEFFECTSLOTFV)(ALuint, ALenum, const ALfloat*);
typedef void (AL_APIENTRY *LPALGETAUXILIARYEFFECTSLOTI)(ALuint, ALenum, ALint*);
typedef void (AL_APIENTRY *LPALGETAUXILIARYEFFECTSLOTIV)(ALuint, ALenum, ALint*);
typedef void (AL_APIENTRY *LPALGETAUXILIARYEFFECTSLOTF)(ALuint, ALenum, ALfloat*);
typedef void (AL_APIENTRY *LPALGETAUXILIARYEFFECTSLOTFV)(ALuint, ALenum, ALfloat*);
#ifdef AL_ALEXT_PROTOTYPES
AL_API ALvoid AL_APIENTRY alGenEffects(ALsizei n, ALuint *effects);
AL_API ALvoid AL_APIENTRY alDeleteEffects(ALsizei n, const ALuint *effects);
AL_API ALboolean AL_APIENTRY alIsEffect(ALuint effect);
AL_API ALvoid AL_APIENTRY alEffecti(ALuint effect, ALenum param, ALint iValue);
AL_API ALvoid AL_APIENTRY alEffectiv(ALuint effect, ALenum param, const ALint *piValues);
AL_API ALvoid AL_APIENTRY alEffectf(ALuint effect, ALenum param, ALfloat flValue);
AL_API ALvoid AL_APIENTRY alEffectfv(ALuint effect, ALenum param, const ALfloat *pflValues);
AL_API ALvoid AL_APIENTRY alGetEffecti(ALuint effect, ALenum param, ALint *piValue);
AL_API ALvoid AL_APIENTRY alGetEffectiv(ALuint effect, ALenum param, ALint *piValues);
AL_API ALvoid AL_APIENTRY alGetEffectf(ALuint effect, ALenum param, ALfloat *pflValue);
AL_API ALvoid AL_APIENTRY alGetEffectfv(ALuint effect, ALenum param, ALfloat *pflValues);
AL_API ALvoid AL_APIENTRY alGenFilters(ALsizei n, ALuint *filters);
AL_API ALvoid AL_APIENTRY alDeleteFilters(ALsizei n, const ALuint *filters);
AL_API ALboolean AL_APIENTRY alIsFilter(ALuint filter);
AL_API ALvoid AL_APIENTRY alFilteri(ALuint filter, ALenum param, ALint iValue);
AL_API ALvoid AL_APIENTRY alFilteriv(ALuint filter, ALenum param, const ALint *piValues);
AL_API ALvoid AL_APIENTRY alFilterf(ALuint filter, ALenum param, ALfloat flValue);
AL_API ALvoid AL_APIENTRY alFilterfv(ALuint filter, ALenum param, const ALfloat *pflValues);
AL_API ALvoid AL_APIENTRY alGetFilteri(ALuint filter, ALenum param, ALint *piValue);
AL_API ALvoid AL_APIENTRY alGetFilteriv(ALuint filter, ALenum param, ALint *piValues);
AL_API ALvoid AL_APIENTRY alGetFilterf(ALuint filter, ALenum param, ALfloat *pflValue);
AL_API ALvoid AL_APIENTRY alGetFilterfv(ALuint filter, ALenum param, ALfloat *pflValues);
AL_API ALvoid AL_APIENTRY alGenAuxiliaryEffectSlots(ALsizei n, ALuint *effectslots);
AL_API ALvoid AL_APIENTRY alDeleteAuxiliaryEffectSlots(ALsizei n, const ALuint *effectslots);
AL_API ALboolean AL_APIENTRY alIsAuxiliaryEffectSlot(ALuint effectslot);
AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSloti(ALuint effectslot, ALenum param, ALint iValue);
AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotiv(ALuint effectslot, ALenum param, const ALint *piValues);
AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotf(ALuint effectslot, ALenum param, ALfloat flValue);
AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotfv(ALuint effectslot, ALenum param, const ALfloat *pflValues);
AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSloti(ALuint effectslot, ALenum param, ALint *piValue);
AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotiv(ALuint effectslot, ALenum param, ALint *piValues);
AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotf(ALuint effectslot, ALenum param, ALfloat *pflValue);
AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotfv(ALuint effectslot, ALenum param, ALfloat *pflValues);
#endif
/* Filter ranges and defaults. */
/* Lowpass filter */
#define AL_LOWPASS_MIN_GAIN (0.0f)
#define AL_LOWPASS_MAX_GAIN (1.0f)
#define AL_LOWPASS_DEFAULT_GAIN (1.0f)
#define AL_LOWPASS_MIN_GAINHF (0.0f)
#define AL_LOWPASS_MAX_GAINHF (1.0f)
#define AL_LOWPASS_DEFAULT_GAINHF (1.0f)
/* Highpass filter */
#define AL_HIGHPASS_MIN_GAIN (0.0f)
#define AL_HIGHPASS_MAX_GAIN (1.0f)
#define AL_HIGHPASS_DEFAULT_GAIN (1.0f)
#define AL_HIGHPASS_MIN_GAINLF (0.0f)
#define AL_HIGHPASS_MAX_GAINLF (1.0f)
#define AL_HIGHPASS_DEFAULT_GAINLF (1.0f)
/* Bandpass filter */
#define AL_BANDPASS_MIN_GAIN (0.0f)
#define AL_BANDPASS_MAX_GAIN (1.0f)
#define AL_BANDPASS_DEFAULT_GAIN (1.0f)
#define AL_BANDPASS_MIN_GAINHF (0.0f)
#define AL_BANDPASS_MAX_GAINHF (1.0f)
#define AL_BANDPASS_DEFAULT_GAINHF (1.0f)
#define AL_BANDPASS_MIN_GAINLF (0.0f)
#define AL_BANDPASS_MAX_GAINLF (1.0f)
#define AL_BANDPASS_DEFAULT_GAINLF (1.0f)
/* Effect parameter ranges and defaults. */
/* Standard reverb effect */
#define AL_REVERB_MIN_DENSITY (0.0f)
#define AL_REVERB_MAX_DENSITY (1.0f)
#define AL_REVERB_DEFAULT_DENSITY (1.0f)
#define AL_REVERB_MIN_DIFFUSION (0.0f)
#define AL_REVERB_MAX_DIFFUSION (1.0f)
#define AL_REVERB_DEFAULT_DIFFUSION (1.0f)
#define AL_REVERB_MIN_GAIN (0.0f)
#define AL_REVERB_MAX_GAIN (1.0f)
#define AL_REVERB_DEFAULT_GAIN (0.32f)
#define AL_REVERB_MIN_GAINHF (0.0f)
#define AL_REVERB_MAX_GAINHF (1.0f)
#define AL_REVERB_DEFAULT_GAINHF (0.89f)
#define AL_REVERB_MIN_DECAY_TIME (0.1f)
#define AL_REVERB_MAX_DECAY_TIME (20.0f)
#define AL_REVERB_DEFAULT_DECAY_TIME (1.49f)
#define AL_REVERB_MIN_DECAY_HFRATIO (0.1f)
#define AL_REVERB_MAX_DECAY_HFRATIO (2.0f)
#define AL_REVERB_DEFAULT_DECAY_HFRATIO (0.83f)
#define AL_REVERB_MIN_REFLECTIONS_GAIN (0.0f)
#define AL_REVERB_MAX_REFLECTIONS_GAIN (3.16f)
#define AL_REVERB_DEFAULT_REFLECTIONS_GAIN (0.05f)
#define AL_REVERB_MIN_REFLECTIONS_DELAY (0.0f)
#define AL_REVERB_MAX_REFLECTIONS_DELAY (0.3f)
#define AL_REVERB_DEFAULT_REFLECTIONS_DELAY (0.007f)
#define AL_REVERB_MIN_LATE_REVERB_GAIN (0.0f)
#define AL_REVERB_MAX_LATE_REVERB_GAIN (10.0f)
#define AL_REVERB_DEFAULT_LATE_REVERB_GAIN (1.26f)
#define AL_REVERB_MIN_LATE_REVERB_DELAY (0.0f)
#define AL_REVERB_MAX_LATE_REVERB_DELAY (0.1f)
#define AL_REVERB_DEFAULT_LATE_REVERB_DELAY (0.011f)
#define AL_REVERB_MIN_AIR_ABSORPTION_GAINHF (0.892f)
#define AL_REVERB_MAX_AIR_ABSORPTION_GAINHF (1.0f)
#define AL_REVERB_DEFAULT_AIR_ABSORPTION_GAINHF (0.994f)
#define AL_REVERB_MIN_ROOM_ROLLOFF_FACTOR (0.0f)
#define AL_REVERB_MAX_ROOM_ROLLOFF_FACTOR (10.0f)
#define AL_REVERB_DEFAULT_ROOM_ROLLOFF_FACTOR (0.0f)
#define AL_REVERB_MIN_DECAY_HFLIMIT AL_FALSE
#define AL_REVERB_MAX_DECAY_HFLIMIT AL_TRUE
#define AL_REVERB_DEFAULT_DECAY_HFLIMIT AL_TRUE
/* EAX reverb effect */
#define AL_EAXREVERB_MIN_DENSITY (0.0f)
#define AL_EAXREVERB_MAX_DENSITY (1.0f)
#define AL_EAXREVERB_DEFAULT_DENSITY (1.0f)
#define AL_EAXREVERB_MIN_DIFFUSION (0.0f)
#define AL_EAXREVERB_MAX_DIFFUSION (1.0f)
#define AL_EAXREVERB_DEFAULT_DIFFUSION (1.0f)
#define AL_EAXREVERB_MIN_GAIN (0.0f)
#define AL_EAXREVERB_MAX_GAIN (1.0f)
#define AL_EAXREVERB_DEFAULT_GAIN (0.32f)
#define AL_EAXREVERB_MIN_GAINHF (0.0f)
#define AL_EAXREVERB_MAX_GAINHF (1.0f)
#define AL_EAXREVERB_DEFAULT_GAINHF (0.89f)
#define AL_EAXREVERB_MIN_GAINLF (0.0f)
#define AL_EAXREVERB_MAX_GAINLF (1.0f)
#define AL_EAXREVERB_DEFAULT_GAINLF (1.0f)
#define AL_EAXREVERB_MIN_DECAY_TIME (0.1f)
#define AL_EAXREVERB_MAX_DECAY_TIME (20.0f)
#define AL_EAXREVERB_DEFAULT_DECAY_TIME (1.49f)
#define AL_EAXREVERB_MIN_DECAY_HFRATIO (0.1f)
#define AL_EAXREVERB_MAX_DECAY_HFRATIO (2.0f)
#define AL_EAXREVERB_DEFAULT_DECAY_HFRATIO (0.83f)
#define AL_EAXREVERB_MIN_DECAY_LFRATIO (0.1f)
#define AL_EAXREVERB_MAX_DECAY_LFRATIO (2.0f)
#define AL_EAXREVERB_DEFAULT_DECAY_LFRATIO (1.0f)
#define AL_EAXREVERB_MIN_REFLECTIONS_GAIN (0.0f)
#define AL_EAXREVERB_MAX_REFLECTIONS_GAIN (3.16f)
#define AL_EAXREVERB_DEFAULT_REFLECTIONS_GAIN (0.05f)
#define AL_EAXREVERB_MIN_REFLECTIONS_DELAY (0.0f)
#define AL_EAXREVERB_MAX_REFLECTIONS_DELAY (0.3f)
#define AL_EAXREVERB_DEFAULT_REFLECTIONS_DELAY (0.007f)
#define AL_EAXREVERB_DEFAULT_REFLECTIONS_PAN_XYZ (0.0f)
#define AL_EAXREVERB_MIN_LATE_REVERB_GAIN (0.0f)
#define AL_EAXREVERB_MAX_LATE_REVERB_GAIN (10.0f)
#define AL_EAXREVERB_DEFAULT_LATE_REVERB_GAIN (1.26f)
#define AL_EAXREVERB_MIN_LATE_REVERB_DELAY (0.0f)
#define AL_EAXREVERB_MAX_LATE_REVERB_DELAY (0.1f)
#define AL_EAXREVERB_DEFAULT_LATE_REVERB_DELAY (0.011f)
#define AL_EAXREVERB_DEFAULT_LATE_REVERB_PAN_XYZ (0.0f)
#define AL_EAXREVERB_MIN_ECHO_TIME (0.075f)
#define AL_EAXREVERB_MAX_ECHO_TIME (0.25f)
#define AL_EAXREVERB_DEFAULT_ECHO_TIME (0.25f)
#define AL_EAXREVERB_MIN_ECHO_DEPTH (0.0f)
#define AL_EAXREVERB_MAX_ECHO_DEPTH (1.0f)
#define AL_EAXREVERB_DEFAULT_ECHO_DEPTH (0.0f)
#define AL_EAXREVERB_MIN_MODULATION_TIME (0.04f)
#define AL_EAXREVERB_MAX_MODULATION_TIME (4.0f)
#define AL_EAXREVERB_DEFAULT_MODULATION_TIME (0.25f)
#define AL_EAXREVERB_MIN_MODULATION_DEPTH (0.0f)
#define AL_EAXREVERB_MAX_MODULATION_DEPTH (1.0f)
#define AL_EAXREVERB_DEFAULT_MODULATION_DEPTH (0.0f)
#define AL_EAXREVERB_MIN_AIR_ABSORPTION_GAINHF (0.892f)
#define AL_EAXREVERB_MAX_AIR_ABSORPTION_GAINHF (1.0f)
#define AL_EAXREVERB_DEFAULT_AIR_ABSORPTION_GAINHF (0.994f)
#define AL_EAXREVERB_MIN_HFREFERENCE (1000.0f)
#define AL_EAXREVERB_MAX_HFREFERENCE (20000.0f)
#define AL_EAXREVERB_DEFAULT_HFREFERENCE (5000.0f)
#define AL_EAXREVERB_MIN_LFREFERENCE (20.0f)
#define AL_EAXREVERB_MAX_LFREFERENCE (1000.0f)
#define AL_EAXREVERB_DEFAULT_LFREFERENCE (250.0f)
#define AL_EAXREVERB_MIN_ROOM_ROLLOFF_FACTOR (0.0f)
#define AL_EAXREVERB_MAX_ROOM_ROLLOFF_FACTOR (10.0f)
#define AL_EAXREVERB_DEFAULT_ROOM_ROLLOFF_FACTOR (0.0f)
#define AL_EAXREVERB_MIN_DECAY_HFLIMIT AL_FALSE
#define AL_EAXREVERB_MAX_DECAY_HFLIMIT AL_TRUE
#define AL_EAXREVERB_DEFAULT_DECAY_HFLIMIT AL_TRUE
/* Chorus effect */
#define AL_CHORUS_WAVEFORM_SINUSOID (0)
#define AL_CHORUS_WAVEFORM_TRIANGLE (1)
#define AL_CHORUS_MIN_WAVEFORM (0)
#define AL_CHORUS_MAX_WAVEFORM (1)
#define AL_CHORUS_DEFAULT_WAVEFORM (1)
#define AL_CHORUS_MIN_PHASE (-180)
#define AL_CHORUS_MAX_PHASE (180)
#define AL_CHORUS_DEFAULT_PHASE (90)
#define AL_CHORUS_MIN_RATE (0.0f)
#define AL_CHORUS_MAX_RATE (10.0f)
#define AL_CHORUS_DEFAULT_RATE (1.1f)
#define AL_CHORUS_MIN_DEPTH (0.0f)
#define AL_CHORUS_MAX_DEPTH (1.0f)
#define AL_CHORUS_DEFAULT_DEPTH (0.1f)
#define AL_CHORUS_MIN_FEEDBACK (-1.0f)
#define AL_CHORUS_MAX_FEEDBACK (1.0f)
#define AL_CHORUS_DEFAULT_FEEDBACK (0.25f)
#define AL_CHORUS_MIN_DELAY (0.0f)
#define AL_CHORUS_MAX_DELAY (0.016f)
#define AL_CHORUS_DEFAULT_DELAY (0.016f)
/* Distortion effect */
#define AL_DISTORTION_MIN_EDGE (0.0f)
#define AL_DISTORTION_MAX_EDGE (1.0f)
#define AL_DISTORTION_DEFAULT_EDGE (0.2f)
#define AL_DISTORTION_MIN_GAIN (0.01f)
#define AL_DISTORTION_MAX_GAIN (1.0f)
#define AL_DISTORTION_DEFAULT_GAIN (0.05f)
#define AL_DISTORTION_MIN_LOWPASS_CUTOFF (80.0f)
#define AL_DISTORTION_MAX_LOWPASS_CUTOFF (24000.0f)
#define AL_DISTORTION_DEFAULT_LOWPASS_CUTOFF (8000.0f)
#define AL_DISTORTION_MIN_EQCENTER (80.0f)
#define AL_DISTORTION_MAX_EQCENTER (24000.0f)
#define AL_DISTORTION_DEFAULT_EQCENTER (3600.0f)
#define AL_DISTORTION_MIN_EQBANDWIDTH (80.0f)
#define AL_DISTORTION_MAX_EQBANDWIDTH (24000.0f)
#define AL_DISTORTION_DEFAULT_EQBANDWIDTH (3600.0f)
/* Echo effect */
#define AL_ECHO_MIN_DELAY (0.0f)
#define AL_ECHO_MAX_DELAY (0.207f)
#define AL_ECHO_DEFAULT_DELAY (0.1f)
#define AL_ECHO_MIN_LRDELAY (0.0f)
#define AL_ECHO_MAX_LRDELAY (0.404f)
#define AL_ECHO_DEFAULT_LRDELAY (0.1f)
#define AL_ECHO_MIN_DAMPING (0.0f)
#define AL_ECHO_MAX_DAMPING (0.99f)
#define AL_ECHO_DEFAULT_DAMPING (0.5f)
#define AL_ECHO_MIN_FEEDBACK (0.0f)
#define AL_ECHO_MAX_FEEDBACK (1.0f)
#define AL_ECHO_DEFAULT_FEEDBACK (0.5f)
#define AL_ECHO_MIN_SPREAD (-1.0f)
#define AL_ECHO_MAX_SPREAD (1.0f)
#define AL_ECHO_DEFAULT_SPREAD (-1.0f)
/* Flanger effect */
#define AL_FLANGER_WAVEFORM_SINUSOID (0)
#define AL_FLANGER_WAVEFORM_TRIANGLE (1)
#define AL_FLANGER_MIN_WAVEFORM (0)
#define AL_FLANGER_MAX_WAVEFORM (1)
#define AL_FLANGER_DEFAULT_WAVEFORM (1)
#define AL_FLANGER_MIN_PHASE (-180)
#define AL_FLANGER_MAX_PHASE (180)
#define AL_FLANGER_DEFAULT_PHASE (0)
#define AL_FLANGER_MIN_RATE (0.0f)
#define AL_FLANGER_MAX_RATE (10.0f)
#define AL_FLANGER_DEFAULT_RATE (0.27f)
#define AL_FLANGER_MIN_DEPTH (0.0f)
#define AL_FLANGER_MAX_DEPTH (1.0f)
#define AL_FLANGER_DEFAULT_DEPTH (1.0f)
#define AL_FLANGER_MIN_FEEDBACK (-1.0f)
#define AL_FLANGER_MAX_FEEDBACK (1.0f)
#define AL_FLANGER_DEFAULT_FEEDBACK (-0.5f)
#define AL_FLANGER_MIN_DELAY (0.0f)
#define AL_FLANGER_MAX_DELAY (0.004f)
#define AL_FLANGER_DEFAULT_DELAY (0.002f)
/* Frequency shifter effect */
#define AL_FREQUENCY_SHIFTER_MIN_FREQUENCY (0.0f)
#define AL_FREQUENCY_SHIFTER_MAX_FREQUENCY (24000.0f)
#define AL_FREQUENCY_SHIFTER_DEFAULT_FREQUENCY (0.0f)
#define AL_FREQUENCY_SHIFTER_MIN_LEFT_DIRECTION (0)
#define AL_FREQUENCY_SHIFTER_MAX_LEFT_DIRECTION (2)
#define AL_FREQUENCY_SHIFTER_DEFAULT_LEFT_DIRECTION (0)
#define AL_FREQUENCY_SHIFTER_DIRECTION_DOWN (0)
#define AL_FREQUENCY_SHIFTER_DIRECTION_UP (1)
#define AL_FREQUENCY_SHIFTER_DIRECTION_OFF (2)
#define AL_FREQUENCY_SHIFTER_MIN_RIGHT_DIRECTION (0)
#define AL_FREQUENCY_SHIFTER_MAX_RIGHT_DIRECTION (2)
#define AL_FREQUENCY_SHIFTER_DEFAULT_RIGHT_DIRECTION (0)
/* Vocal morpher effect */
#define AL_VOCAL_MORPHER_MIN_PHONEMEA (0)
#define AL_VOCAL_MORPHER_MAX_PHONEMEA (29)
#define AL_VOCAL_MORPHER_DEFAULT_PHONEMEA (0)
#define AL_VOCAL_MORPHER_MIN_PHONEMEA_COARSE_TUNING (-24)
#define AL_VOCAL_MORPHER_MAX_PHONEMEA_COARSE_TUNING (24)
#define AL_VOCAL_MORPHER_DEFAULT_PHONEMEA_COARSE_TUNING (0)
#define AL_VOCAL_MORPHER_MIN_PHONEMEB (0)
#define AL_VOCAL_MORPHER_MAX_PHONEMEB (29)
#define AL_VOCAL_MORPHER_DEFAULT_PHONEMEB (10)
#define AL_VOCAL_MORPHER_MIN_PHONEMEB_COARSE_TUNING (-24)
#define AL_VOCAL_MORPHER_MAX_PHONEMEB_COARSE_TUNING (24)
#define AL_VOCAL_MORPHER_DEFAULT_PHONEMEB_COARSE_TUNING (0)
#define AL_VOCAL_MORPHER_PHONEME_A (0)
#define AL_VOCAL_MORPHER_PHONEME_E (1)
#define AL_VOCAL_MORPHER_PHONEME_I (2)
#define AL_VOCAL_MORPHER_PHONEME_O (3)
#define AL_VOCAL_MORPHER_PHONEME_U (4)
#define AL_VOCAL_MORPHER_PHONEME_AA (5)
#define AL_VOCAL_MORPHER_PHONEME_AE (6)
#define AL_VOCAL_MORPHER_PHONEME_AH (7)
#define AL_VOCAL_MORPHER_PHONEME_AO (8)
#define AL_VOCAL_MORPHER_PHONEME_EH (9)
#define AL_VOCAL_MORPHER_PHONEME_ER (10)
#define AL_VOCAL_MORPHER_PHONEME_IH (11)
#define AL_VOCAL_MORPHER_PHONEME_IY (12)
#define AL_VOCAL_MORPHER_PHONEME_UH (13)
#define AL_VOCAL_MORPHER_PHONEME_UW (14)
#define AL_VOCAL_MORPHER_PHONEME_B (15)
#define AL_VOCAL_MORPHER_PHONEME_D (16)
#define AL_VOCAL_MORPHER_PHONEME_F (17)
#define AL_VOCAL_MORPHER_PHONEME_G (18)
#define AL_VOCAL_MORPHER_PHONEME_J (19)
#define AL_VOCAL_MORPHER_PHONEME_K (20)
#define AL_VOCAL_MORPHER_PHONEME_L (21)
#define AL_VOCAL_MORPHER_PHONEME_M (22)
#define AL_VOCAL_MORPHER_PHONEME_N (23)
#define AL_VOCAL_MORPHER_PHONEME_P (24)
#define AL_VOCAL_MORPHER_PHONEME_R (25)
#define AL_VOCAL_MORPHER_PHONEME_S (26)
#define AL_VOCAL_MORPHER_PHONEME_T (27)
#define AL_VOCAL_MORPHER_PHONEME_V (28)
#define AL_VOCAL_MORPHER_PHONEME_Z (29)
#define AL_VOCAL_MORPHER_WAVEFORM_SINUSOID (0)
#define AL_VOCAL_MORPHER_WAVEFORM_TRIANGLE (1)
#define AL_VOCAL_MORPHER_WAVEFORM_SAWTOOTH (2)
#define AL_VOCAL_MORPHER_MIN_WAVEFORM (0)
#define AL_VOCAL_MORPHER_MAX_WAVEFORM (2)
#define AL_VOCAL_MORPHER_DEFAULT_WAVEFORM (0)
#define AL_VOCAL_MORPHER_MIN_RATE (0.0f)
#define AL_VOCAL_MORPHER_MAX_RATE (10.0f)
#define AL_VOCAL_MORPHER_DEFAULT_RATE (1.41f)
/* Pitch shifter effect */
#define AL_PITCH_SHIFTER_MIN_COARSE_TUNE (-12)
#define AL_PITCH_SHIFTER_MAX_COARSE_TUNE (12)
#define AL_PITCH_SHIFTER_DEFAULT_COARSE_TUNE (12)
#define AL_PITCH_SHIFTER_MIN_FINE_TUNE (-50)
#define AL_PITCH_SHIFTER_MAX_FINE_TUNE (50)
#define AL_PITCH_SHIFTER_DEFAULT_FINE_TUNE (0)
/* Ring modulator effect */
#define AL_RING_MODULATOR_MIN_FREQUENCY (0.0f)
#define AL_RING_MODULATOR_MAX_FREQUENCY (8000.0f)
#define AL_RING_MODULATOR_DEFAULT_FREQUENCY (440.0f)
#define AL_RING_MODULATOR_MIN_HIGHPASS_CUTOFF (0.0f)
#define AL_RING_MODULATOR_MAX_HIGHPASS_CUTOFF (24000.0f)
#define AL_RING_MODULATOR_DEFAULT_HIGHPASS_CUTOFF (800.0f)
#define AL_RING_MODULATOR_SINUSOID (0)
#define AL_RING_MODULATOR_SAWTOOTH (1)
#define AL_RING_MODULATOR_SQUARE (2)
#define AL_RING_MODULATOR_MIN_WAVEFORM (0)
#define AL_RING_MODULATOR_MAX_WAVEFORM (2)
#define AL_RING_MODULATOR_DEFAULT_WAVEFORM (0)
/* Autowah effect */
#define AL_AUTOWAH_MIN_ATTACK_TIME (0.0001f)
#define AL_AUTOWAH_MAX_ATTACK_TIME (1.0f)
#define AL_AUTOWAH_DEFAULT_ATTACK_TIME (0.06f)
#define AL_AUTOWAH_MIN_RELEASE_TIME (0.0001f)
#define AL_AUTOWAH_MAX_RELEASE_TIME (1.0f)
#define AL_AUTOWAH_DEFAULT_RELEASE_TIME (0.06f)
#define AL_AUTOWAH_MIN_RESONANCE (2.0f)
#define AL_AUTOWAH_MAX_RESONANCE (1000.0f)
#define AL_AUTOWAH_DEFAULT_RESONANCE (1000.0f)
#define AL_AUTOWAH_MIN_PEAK_GAIN (0.00003f)
#define AL_AUTOWAH_MAX_PEAK_GAIN (31621.0f)
#define AL_AUTOWAH_DEFAULT_PEAK_GAIN (11.22f)
/* Compressor effect */
#define AL_COMPRESSOR_MIN_ONOFF (0)
#define AL_COMPRESSOR_MAX_ONOFF (1)
#define AL_COMPRESSOR_DEFAULT_ONOFF (1)
/* Equalizer effect */
#define AL_EQUALIZER_MIN_LOW_GAIN (0.126f)
#define AL_EQUALIZER_MAX_LOW_GAIN (7.943f)
#define AL_EQUALIZER_DEFAULT_LOW_GAIN (1.0f)
#define AL_EQUALIZER_MIN_LOW_CUTOFF (50.0f)
#define AL_EQUALIZER_MAX_LOW_CUTOFF (800.0f)
#define AL_EQUALIZER_DEFAULT_LOW_CUTOFF (200.0f)
#define AL_EQUALIZER_MIN_MID1_GAIN (0.126f)
#define AL_EQUALIZER_MAX_MID1_GAIN (7.943f)
#define AL_EQUALIZER_DEFAULT_MID1_GAIN (1.0f)
#define AL_EQUALIZER_MIN_MID1_CENTER (200.0f)
#define AL_EQUALIZER_MAX_MID1_CENTER (3000.0f)
#define AL_EQUALIZER_DEFAULT_MID1_CENTER (500.0f)
#define AL_EQUALIZER_MIN_MID1_WIDTH (0.01f)
#define AL_EQUALIZER_MAX_MID1_WIDTH (1.0f)
#define AL_EQUALIZER_DEFAULT_MID1_WIDTH (1.0f)
#define AL_EQUALIZER_MIN_MID2_GAIN (0.126f)
#define AL_EQUALIZER_MAX_MID2_GAIN (7.943f)
#define AL_EQUALIZER_DEFAULT_MID2_GAIN (1.0f)
#define AL_EQUALIZER_MIN_MID2_CENTER (1000.0f)
#define AL_EQUALIZER_MAX_MID2_CENTER (8000.0f)
#define AL_EQUALIZER_DEFAULT_MID2_CENTER (3000.0f)
#define AL_EQUALIZER_MIN_MID2_WIDTH (0.01f)
#define AL_EQUALIZER_MAX_MID2_WIDTH (1.0f)
#define AL_EQUALIZER_DEFAULT_MID2_WIDTH (1.0f)
#define AL_EQUALIZER_MIN_HIGH_GAIN (0.126f)
#define AL_EQUALIZER_MAX_HIGH_GAIN (7.943f)
#define AL_EQUALIZER_DEFAULT_HIGH_GAIN (1.0f)
#define AL_EQUALIZER_MIN_HIGH_CUTOFF (4000.0f)
#define AL_EQUALIZER_MAX_HIGH_CUTOFF (16000.0f)
#define AL_EQUALIZER_DEFAULT_HIGH_CUTOFF (6000.0f)
/* Source parameter value ranges and defaults. */
#define AL_MIN_AIR_ABSORPTION_FACTOR (0.0f)
#define AL_MAX_AIR_ABSORPTION_FACTOR (10.0f)
#define AL_DEFAULT_AIR_ABSORPTION_FACTOR (0.0f)
#define AL_MIN_ROOM_ROLLOFF_FACTOR (0.0f)
#define AL_MAX_ROOM_ROLLOFF_FACTOR (10.0f)
#define AL_DEFAULT_ROOM_ROLLOFF_FACTOR (0.0f)
#define AL_MIN_CONE_OUTER_GAINHF (0.0f)
#define AL_MAX_CONE_OUTER_GAINHF (1.0f)
#define AL_DEFAULT_CONE_OUTER_GAINHF (1.0f)
#define AL_MIN_DIRECT_FILTER_GAINHF_AUTO AL_FALSE
#define AL_MAX_DIRECT_FILTER_GAINHF_AUTO AL_TRUE
#define AL_DEFAULT_DIRECT_FILTER_GAINHF_AUTO AL_TRUE
#define AL_MIN_AUXILIARY_SEND_FILTER_GAIN_AUTO AL_FALSE
#define AL_MAX_AUXILIARY_SEND_FILTER_GAIN_AUTO AL_TRUE
#define AL_DEFAULT_AUXILIARY_SEND_FILTER_GAIN_AUTO AL_TRUE
#define AL_MIN_AUXILIARY_SEND_FILTER_GAINHF_AUTO AL_FALSE
#define AL_MAX_AUXILIARY_SEND_FILTER_GAINHF_AUTO AL_TRUE
#define AL_DEFAULT_AUXILIARY_SEND_FILTER_GAINHF_AUTO AL_TRUE
/* Listener parameter value ranges and defaults. */
#define AL_MIN_METERS_PER_UNIT FLT_MIN
#define AL_MAX_METERS_PER_UNIT FLT_MAX
#define AL_DEFAULT_METERS_PER_UNIT (1.0f)
#ifdef __cplusplus
} /* extern "C" */
#endif
#endif /* AL_EFX_H */
-483
View File
@@ -1,483 +0,0 @@
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write to the author to ask for permission. For software which is
copyrighted by the Free Software Foundation, write to the Free
Software Foundation; we sometimes make exceptions for this. Our
decision will be guided by the two goals of preserving the free status
of all derivatives of our free software and of promoting the sharing
and reuse of software generally.
NO WARRANTY
15. BECAUSE THE LIBRARY IS LICENSED FREE OF CHARGE, THERE IS NO
WARRANTY FOR THE LIBRARY, TO THE EXTENT PERMITTED BY APPLICABLE LAW.
EXCEPT WHEN OTHERWISE STATED IN WRITING THE COPYRIGHT HOLDERS AND/OR
OTHER PARTIES PROVIDE THE LIBRARY "AS IS" WITHOUT WARRANTY OF ANY
KIND, EITHER EXPRESSED OR IMPLIED, INCLUDING, BUT NOT LIMITED TO, THE
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
PURPOSE. THE ENTIRE RISK AS TO THE QUALITY AND PERFORMANCE OF THE
LIBRARY IS WITH YOU. SHOULD THE LIBRARY PROVE DEFECTIVE, YOU ASSUME
THE COST OF ALL NECESSARY SERVICING, REPAIR OR CORRECTION.
16. IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN
WRITING WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MAY MODIFY
AND/OR REDISTRIBUTE THE LIBRARY AS PERMITTED ABOVE, BE LIABLE TO YOU
FOR DAMAGES, INCLUDING ANY GENERAL, SPECIAL, INCIDENTAL OR
CONSEQUENTIAL DAMAGES ARISING OUT OF THE USE OR INABILITY TO USE THE
LIBRARY (INCLUDING BUT NOT LIMITED TO LOSS OF DATA OR DATA BEING
RENDERED INACCURATE OR LOSSES SUSTAINED BY YOU OR THIRD PARTIES OR A
FAILURE OF THE LIBRARY TO OPERATE WITH ANY OTHER SOFTWARE), EVEN IF
SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH
DAMAGES.
END OF TERMS AND CONDITIONS
Appendix: How to Apply These Terms to Your New Libraries
If you develop a new library, and you want it to be of the greatest
possible use to the public, we recommend making it free software that
everyone can redistribute and change. You can do so by permitting
redistribution under these terms (or, alternatively, under the terms of the
ordinary General Public License).
To apply these terms, attach the following notices to the library. It is
safest to attach them to the start of each source file to most effectively
convey the exclusion of warranty; and each file should have at least the
"copyright" line and a pointer to where the full notice is found.
<one line to give the library's name and a brief idea of what it does.>
Copyright (C) <year> <name of author>
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Library General Public
License as published by the Free Software Foundation; either
version 2 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Library General Public License for more details.
You should have received a copy of the GNU Library General Public
License along with this library; if not, write to the Free
Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
Also add information on how to contact you by electronic and paper mail.
You should also get your employer (if you work as a programmer) or your
school, if any, to sign a "copyright disclaimer" for the library, if
necessary. Here is a sample; alter the names:
Yoyodyne, Inc., hereby disclaims all copyright interest in the
library `Frob' (a library for tweaking knobs) written by James Random Hacker.
<signature of Ty Coon>, 1 April 1990
Ty Coon, President of Vice
That's all there is to it!
+11
View File
@@ -0,0 +1,11 @@
prefix=@prefix@
exec_prefix=@exec_prefix@
libdir=@libdir@
includedir=@includedir@
Name: OpenAL
Description: OpenAL is a cross-platform 3D audio API
Requires: @PKG_CONFIG_REQUIRES@
Version: @PACKAGE_VERSION@
Libs: -L${libdir} -l@LIBNAME@ @PKG_CONFIG_LIBS@
Cflags: -I${includedir} -I${includedir}/AL @PKG_CONFIG_CFLAGS@
-69
View File
@@ -1,69 +0,0 @@
# OpenAL config file. Options that are not under a block or are under the
# [general] block are for general, non-backend-specific options. Blocks may
# appear multiple times, and duplicated options will take the last value
# specified. The system-wide settings can be put in /etc/openal/config and
# user-specific override settings in ~/.openalrc
# Option and block names are case-insenstive. The supplied values are only
# hints and may not be honored (though generally it'll try to get as close as
# possible). These are the current available settings:
format = AL_FORMAT_STEREO16 # Sets the output format. Can be one of:
# AL_FORMAT_MONO8 (8-bit mono)
# AL_FORMAT_STEREO8 (8-bit stereo)
# AL_FORMAT_QUAD8 (8-bit 4-channel)
# AL_FORMAT_51CHN8 (8-bit 5.1 output)
# AL_FORMAT_61CHN8 (8-bit 6.1 output)
# AL_FORMAT_71CHN8 (8-bit 7.1 output)
# AL_FORMAT_MONO16 (16-bit mono)
# AL_FORMAT_STEREO16 (16-bit stereo)
# AL_FORMAT_QUAD16 (16-bit 4-channel)
# AL_FORMAT_51CHN16 (16-bit 5.1 output)
# AL_FORMAT_61CHN16 (16-bit 6.1 output)
# AL_FORMAT_71CHN16 (16-bit 7.1 output)
# Default is AL_FORMAT_STEREO16
cf_level = 0 # Sets the crossfeed level for stereo output. Valid values are:
# 0 - No crossfeed
# 1 - Low crossfeed
# 2 - Middle crossfeed
# 3 - High crossfeed (virtual speakers are closer to itself)
# 4 - Low easy crossfeed
# 5 - Middle easy crossfeed
# 6 - High easy crossfeed
# Default is 0. Users of headphones may want to try various
# settings. Has no effect on non-stereo modes.
frequency = 44100 # Sets the output frequency. Default is 44100
refresh = 0 # Sets the number of frames-per-update. Default is calculated as
# 8192*frequency/22050. Note that the actual granularity may or
# may not be less than this.
drivers = # Sets the backend driver list order, comma-seperated. Unknown
# backends and duplicated names are ignored, and unlisted backends
# won't be considered for use. An empty list means the default.
# Default is:
# alsa,oss,dsound,winmm
[alsa] # ALSA backend stuff
device = default # Sets the device name for the default playback device.
# Default is default
periods = 4 # Sets the number of update buffers. Default is 4
capture = default # Sets the device name for the default capture device.
# Default is default
[oss] # OSS backend stuff
device = /dev/dsp # Sets the device name for OSS output. Default is /dev/dsp
periods = 4 # Sets the number of update buffers. Default is 4
capture = /dev/dsp # Sets the device name for OSS capture. Default is /dev/dsp
[dsound] # DirectSound backend stuff
# Nothing yet...
[winmm] # Windows Multimedia backend stuff
# Nothing yet...
+2010
View File
File diff suppressed because it is too large Load Diff
+421
View File
@@ -0,0 +1,421 @@
# This is a makehrtf HRIR definition file. It is used to define the layout
# and source data to be processed into an OpenAL Soft compatible HRTF.
#
# This definition is used to transform an average of the left and right ear
# HRIRs from any raw dataset from the IRCAM/AKG Listen HRTF database.
#
# The datasets are available free of charge from:
#
# http://recherche.ircam.fr/equipes/salles/listen/index.html
#
# Contact for the Listen HRTF Database:
#
# Olivier Warusfel <olivier.warusfel@ircam.fr>,
# Room Acoustics Team, IRCAM
# 1, place Igor Stravinsky
# 75004 PARIS, France
rate = 44100
# The raw sets have up to 8192 samples, but 2048 seems large enough.
points = 2048
# The IRCAM sets are not as dense as the MIT set.
azimuths = 1, 6, 12, 24, 24, 24, 24, 24, 24, 24, 12, 6, 1
# No head radius was provided. Just use the average radius of 9 cm.
radius = 0.09
# The distance between the source and the listener (in meters).
distance = 1.95
# The IRCAM source azimuth is counter-clockwise, so it needs to be flipped.
# Left and right ear HRIRs (from the respective WAVE channels) are averaged.
# Repalce all occurrences of IRC_#### for the desired subject (1005 was used
# in this demonstration).
[ 3, 0 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T000_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T000_P315.wav"
[ 3, 1 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T345_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T015_P315.wav"
[ 3, 2 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T330_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T030_P315.wav"
[ 3, 3 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T315_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T045_P315.wav"
[ 3, 4 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T300_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T060_P315.wav"
[ 3, 5 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T285_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T075_P315.wav"
[ 3, 6 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T270_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T090_P315.wav"
[ 3, 7 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T255_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T105_P315.wav"
[ 3, 8 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T240_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T120_P315.wav"
[ 3, 9 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T225_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T135_P315.wav"
[ 3, 10 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T210_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T150_P315.wav"
[ 3, 11 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T195_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T165_P315.wav"
[ 3, 12 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T180_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T180_P315.wav"
[ 3, 13 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T165_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T195_P315.wav"
[ 3, 14 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T150_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T210_P315.wav"
[ 3, 15 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T135_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T225_P315.wav"
[ 3, 16 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T120_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T240_P315.wav"
[ 3, 17 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T105_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T255_P315.wav"
[ 3, 18 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T090_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T270_P315.wav"
[ 3, 19 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T075_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T285_P315.wav"
[ 3, 20 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T060_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T300_P315.wav"
[ 3, 21 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T045_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T315_P315.wav"
[ 3, 22 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T030_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T330_P315.wav"
[ 3, 23 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T015_P315.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T345_P315.wav"
[ 4, 0 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T000_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T000_P330.wav"
[ 4, 1 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T345_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T015_P330.wav"
[ 4, 2 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T330_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T030_P330.wav"
[ 4, 3 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T315_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T045_P330.wav"
[ 4, 4 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T300_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T060_P330.wav"
[ 4, 5 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T285_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T075_P330.wav"
[ 4, 6 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T270_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T090_P330.wav"
[ 4, 7 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T255_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T105_P330.wav"
[ 4, 8 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T240_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T120_P330.wav"
[ 4, 9 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T225_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T135_P330.wav"
[ 4, 10 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T210_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T150_P330.wav"
[ 4, 11 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T195_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T165_P330.wav"
[ 4, 12 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T180_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T180_P330.wav"
[ 4, 13 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T165_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T195_P330.wav"
[ 4, 14 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T150_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T210_P330.wav"
[ 4, 15 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T135_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T225_P330.wav"
[ 4, 16 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T120_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T240_P330.wav"
[ 4, 17 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T105_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T255_P330.wav"
[ 4, 18 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T090_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T270_P330.wav"
[ 4, 19 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T075_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T285_P330.wav"
[ 4, 20 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T060_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T300_P330.wav"
[ 4, 21 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T045_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T315_P330.wav"
[ 4, 22 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T030_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T330_P330.wav"
[ 4, 23 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T015_P330.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T345_P330.wav"
[ 5, 0 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T000_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T000_P345.wav"
[ 5, 1 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T345_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T015_P345.wav"
[ 5, 2 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T330_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T030_P345.wav"
[ 5, 3 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T315_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T045_P345.wav"
[ 5, 4 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T300_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T060_P345.wav"
[ 5, 5 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T285_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T075_P345.wav"
[ 5, 6 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T270_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T090_P345.wav"
[ 5, 7 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T255_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T105_P345.wav"
[ 5, 8 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T240_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T120_P345.wav"
[ 5, 9 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T225_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T135_P345.wav"
[ 5, 10 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T210_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T150_P345.wav"
[ 5, 11 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T195_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T165_P345.wav"
[ 5, 12 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T180_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T180_P345.wav"
[ 5, 13 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T165_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T195_P345.wav"
[ 5, 14 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T150_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T210_P345.wav"
[ 5, 15 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T135_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T225_P345.wav"
[ 5, 16 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T120_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T240_P345.wav"
[ 5, 17 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T105_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T255_P345.wav"
[ 5, 18 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T090_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T270_P345.wav"
[ 5, 19 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T075_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T285_P345.wav"
[ 5, 20 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T060_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T300_P345.wav"
[ 5, 21 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T045_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T315_P345.wav"
[ 5, 22 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T030_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T330_P345.wav"
[ 5, 23 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T015_P345.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T345_P345.wav"
[ 6, 0 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T000_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T000_P000.wav"
[ 6, 1 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T345_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T015_P000.wav"
[ 6, 2 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T330_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T030_P000.wav"
[ 6, 3 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T315_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T045_P000.wav"
[ 6, 4 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T300_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T060_P000.wav"
[ 6, 5 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T285_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T075_P000.wav"
[ 6, 6 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T270_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T090_P000.wav"
[ 6, 7 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T255_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T105_P000.wav"
[ 6, 8 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T240_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T120_P000.wav"
[ 6, 9 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T225_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T135_P000.wav"
[ 6, 10 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T210_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T150_P000.wav"
[ 6, 11 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T195_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T165_P000.wav"
[ 6, 12 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T180_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T180_P000.wav"
[ 6, 13 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T165_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T195_P000.wav"
[ 6, 14 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T150_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T210_P000.wav"
[ 6, 15 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T135_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T225_P000.wav"
[ 6, 16 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T120_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T240_P000.wav"
[ 6, 17 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T105_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T255_P000.wav"
[ 6, 18 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T090_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T270_P000.wav"
[ 6, 19 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T075_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T285_P000.wav"
[ 6, 20 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T060_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T300_P000.wav"
[ 6, 21 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T045_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T315_P000.wav"
[ 6, 22 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T030_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T330_P000.wav"
[ 6, 23 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T015_P000.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T345_P000.wav"
[ 7, 0 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T000_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T000_P015.wav"
[ 7, 1 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T345_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T015_P015.wav"
[ 7, 2 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T330_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T030_P015.wav"
[ 7, 3 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T315_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T045_P015.wav"
[ 7, 4 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T300_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T060_P015.wav"
[ 7, 5 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T285_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T075_P015.wav"
[ 7, 6 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T270_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T090_P015.wav"
[ 7, 7 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T255_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T105_P015.wav"
[ 7, 8 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T240_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T120_P015.wav"
[ 7, 9 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T225_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T135_P015.wav"
[ 7, 10 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T210_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T150_P015.wav"
[ 7, 11 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T195_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T165_P015.wav"
[ 7, 12 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T180_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T180_P015.wav"
[ 7, 13 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T165_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T195_P015.wav"
[ 7, 14 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T150_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T210_P015.wav"
[ 7, 15 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T135_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T225_P015.wav"
[ 7, 16 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T120_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T240_P015.wav"
[ 7, 17 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T105_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T255_P015.wav"
[ 7, 18 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T090_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T270_P015.wav"
[ 7, 19 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T075_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T285_P015.wav"
[ 7, 20 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T060_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T300_P015.wav"
[ 7, 21 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T045_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T315_P015.wav"
[ 7, 22 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T030_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T330_P015.wav"
[ 7, 23 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T015_P015.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T345_P015.wav"
[ 8, 0 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T000_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T000_P030.wav"
[ 8, 1 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T345_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T015_P030.wav"
[ 8, 2 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T330_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T030_P030.wav"
[ 8, 3 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T315_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T045_P030.wav"
[ 8, 4 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T300_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T060_P030.wav"
[ 8, 5 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T285_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T075_P030.wav"
[ 8, 6 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T270_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T090_P030.wav"
[ 8, 7 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T255_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T105_P030.wav"
[ 8, 8 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T240_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T120_P030.wav"
[ 8, 9 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T225_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T135_P030.wav"
[ 8, 10 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T210_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T150_P030.wav"
[ 8, 11 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T195_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T165_P030.wav"
[ 8, 12 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T180_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T180_P030.wav"
[ 8, 13 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T165_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T195_P030.wav"
[ 8, 14 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T150_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T210_P030.wav"
[ 8, 15 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T135_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T225_P030.wav"
[ 8, 16 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T120_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T240_P030.wav"
[ 8, 17 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T105_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T255_P030.wav"
[ 8, 18 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T090_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T270_P030.wav"
[ 8, 19 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T075_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T285_P030.wav"
[ 8, 20 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T060_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T300_P030.wav"
[ 8, 21 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T045_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T315_P030.wav"
[ 8, 22 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T030_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T330_P030.wav"
[ 8, 23 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T015_P030.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T345_P030.wav"
[ 9, 0 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T000_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T000_P045.wav"
[ 9, 1 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T345_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T015_P045.wav"
[ 9, 2 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T330_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T030_P045.wav"
[ 9, 3 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T315_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T045_P045.wav"
[ 9, 4 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T300_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T060_P045.wav"
[ 9, 5 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T285_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T075_P045.wav"
[ 9, 6 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T270_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T090_P045.wav"
[ 9, 7 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T255_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T105_P045.wav"
[ 9, 8 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T240_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T120_P045.wav"
[ 9, 9 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T225_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T135_P045.wav"
[ 9, 10 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T210_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T150_P045.wav"
[ 9, 11 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T195_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T165_P045.wav"
[ 9, 12 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T180_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T180_P045.wav"
[ 9, 13 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T165_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T195_P045.wav"
[ 9, 14 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T150_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T210_P045.wav"
[ 9, 15 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T135_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T225_P045.wav"
[ 9, 16 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T120_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T240_P045.wav"
[ 9, 17 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T105_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T255_P045.wav"
[ 9, 18 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T090_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T270_P045.wav"
[ 9, 19 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T075_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T285_P045.wav"
[ 9, 20 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T060_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T300_P045.wav"
[ 9, 21 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T045_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T315_P045.wav"
[ 9, 22 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T030_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T330_P045.wav"
[ 9, 23 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T015_P045.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T345_P045.wav"
[ 10, 0 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T000_P060.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T000_P060.wav"
[ 10, 1 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T330_P060.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T030_P060.wav"
[ 10, 2 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T300_P060.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T060_P060.wav"
[ 10, 3 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T270_P060.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T090_P060.wav"
[ 10, 4 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T240_P060.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T120_P060.wav"
[ 10, 5 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T210_P060.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T150_P060.wav"
[ 10, 6 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T180_P060.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T180_P060.wav"
[ 10, 7 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T150_P060.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T210_P060.wav"
[ 10, 8 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T120_P060.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T240_P060.wav"
[ 10, 9 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T090_P060.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T270_P060.wav"
[ 10, 10 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T060_P060.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T300_P060.wav"
[ 10, 11 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T030_P060.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T330_P060.wav"
[ 11, 0 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T000_P075.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T000_P075.wav"
[ 11, 1 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T300_P075.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T060_P075.wav"
[ 11, 2 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T240_P075.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T120_P075.wav"
[ 11, 3 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T180_P075.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T180_P075.wav"
[ 11, 4 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T120_P075.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T240_P075.wav"
[ 11, 5 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T060_P075.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T300_P075.wav"
[ 12, 0 ] = wave (0) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T000_P090.wav"
+ wave (1) : "./IRC/RAW/WAV/IRC_1005_R/IRC_1005_R_R0195_T000_P090.wav"
+819
View File
@@ -0,0 +1,819 @@
# This is a makehrtf HRIR definition file. It is used to define the layout
# and source data to be processed into an OpenAL Soft compatible HRTF.
#
# This definition is used to transform the left ear HRIRs from the full set
# of KEMAR HRIRs provided by Bill Gardner <billg@media.mit.edu> and Keith
# Martin <kdm@media.mit.edu> of MIT Media Laboratory.
#
# The data (full.tar.Z or full.zip) is available from:
#
# http://sound.media.mit.edu/resources/KEMAR.html
#
# It is copyright 1994 by MIT Media Laboratory, and provided free of charge
# with no restrictions on use so long as the authors (above) are cited.
#
# This definition is used to generate the internal HRTF table used by OpenAL
# Soft.
# The following are the dataset metrics. They must always be specified at
# start of a definition file, but their order is not important.
# Sampling rate of the HRIR data (in hertz).
rate = 44100
# The number of points to use from the HRIR data. This should be a
# sufficiently large value (to encompass the entire impulse response). It
# cannot be smaller than the truncation size (default is 32) specified on the
# command line.
points = 512
# A list of the number of azimuths measured for each elevation. There must
# be at least 5 elevations covering the 180 degrees for the dataset to be
# viable.
azimuths = 1, 12, 24, 36, 45, 56, 60, 72, 72, 72, 72, 72, 60, 56, 45, 36, 24, 12, 1
# The approximate radius (measured ear-to-ear) of the listener's head (in
# meters). This does not have to match the dataset, since makehrtf uses a
# spherical model to reconstruct the propagation delay.
radius = 0.09
# The distance between the source and the listener (in meters). This does
# have to match the dataset, but it's effect is minimal at the moment due to
# the coupled nature of OpenAL Soft's HRTF model.
distance = 1.4
# Following the metrics is the list of source HRIRs for each elevation and
# azimuth pair. They don't have to be specified in order, but the final
# composition must not be sparse. They can however begin above a number of
# elevations (as typical for HRIR measurements).
#
# The elevation and azimuth indices are used to determine the resulting polar
# coordinates following OpenAL Soft's convention (-90 degree elevation
# increasing counter-clockwise from the bottom; 0 degree azimuth increasing
# clockwise from the front).
#
# More than one HRIR can be used per source, in which case the average
# magnitude response of all references for that source is used.
#
# Source specification is of the form (~BNF):
#
# source = '[' ev_index ',' az_index ']' '=' source_ref [ '+' source_ref ]*
#
# ev_index = unsigned_integer
# az_index = unsigned_integer
# source_ref = ref_spec ':' filename
#
# ref_spec = ( wave_fmt '(' wave_parms ')' [ '@' start_sample ] ) |
# ( bin_fmt '(' bini_parms ')' [ '@' start_bytes ] ) |
# ( bin_fmt '(' binf_parms ')' [ '@' start_bytes ] ) |
# ( ascii_fmt '(' asci_parms ')' [ '@' start_elements ] ) |
# ( ascii_fmt '(' ascf_parms ')' [ '@' start_elements ] )
# filename = double_quoted_string
#
# wave_fmt = 'wave'
# wave_parms = channel
# bin_fmt = 'bin_le' | 'bin_be'
# bini_parms = 'int' ',' byte_size [ ',' bin_sig_bits ] [ ';' skip_bytes ]
# binf_parms = 'fp' ',' byte_size [ ';' skip_bytes ]
# ascii_fmt = 'ascii'
# asci_parms = 'int' ',' sig_bits [ ';' skip_elements ]
# ascf_parms = 'fp' [ ';' skip_elements ]
# start_sample = unsigned_integer
# start_bytes = unsigned_integer
# start_elements = unsigned_integer
#
# channel = unsigned_integer
# byte_size = unsigned_integer
# bin_sig_bits = signed_integer
# skip_bytes = unsigned_integer
# sig_bits = unsigned_integer
# skip_elements = unsigned_integer
#
# For bin_sig_bits, positive values mean the significant bits start at the
# MSB (padding toward the LSB) while negative values mean they start at the
# LSB.
[ 5, 0 ] = wave (0) : "./MITfull/elev-40/L-40e000a.wav"
[ 5, 1 ] = wave (0) : "./MITfull/elev-40/L-40e006a.wav"
[ 5, 2 ] = wave (0) : "./MITfull/elev-40/L-40e013a.wav"
[ 5, 3 ] = wave (0) : "./MITfull/elev-40/L-40e019a.wav"
[ 5, 4 ] = wave (0) : "./MITfull/elev-40/L-40e026a.wav"
[ 5, 5 ] = wave (0) : "./MITfull/elev-40/L-40e032a.wav"
[ 5, 6 ] = wave (0) : "./MITfull/elev-40/L-40e039a.wav"
[ 5, 7 ] = wave (0) : "./MITfull/elev-40/L-40e045a.wav"
[ 5, 8 ] = wave (0) : "./MITfull/elev-40/L-40e051a.wav"
[ 5, 9 ] = wave (0) : "./MITfull/elev-40/L-40e058a.wav"
[ 5, 10 ] = wave (0) : "./MITfull/elev-40/L-40e064a.wav"
[ 5, 11 ] = wave (0) : "./MITfull/elev-40/L-40e071a.wav"
[ 5, 12 ] = wave (0) : "./MITfull/elev-40/L-40e077a.wav"
[ 5, 13 ] = wave (0) : "./MITfull/elev-40/L-40e084a.wav"
[ 5, 14 ] = wave (0) : "./MITfull/elev-40/L-40e090a.wav"
[ 5, 15 ] = wave (0) : "./MITfull/elev-40/L-40e096a.wav"
[ 5, 16 ] = wave (0) : "./MITfull/elev-40/L-40e103a.wav"
[ 5, 17 ] = wave (0) : "./MITfull/elev-40/L-40e109a.wav"
[ 5, 18 ] = wave (0) : "./MITfull/elev-40/L-40e116a.wav"
[ 5, 19 ] = wave (0) : "./MITfull/elev-40/L-40e122a.wav"
[ 5, 20 ] = wave (0) : "./MITfull/elev-40/L-40e129a.wav"
[ 5, 21 ] = wave (0) : "./MITfull/elev-40/L-40e135a.wav"
[ 5, 22 ] = wave (0) : "./MITfull/elev-40/L-40e141a.wav"
[ 5, 23 ] = wave (0) : "./MITfull/elev-40/L-40e148a.wav"
[ 5, 24 ] = wave (0) : "./MITfull/elev-40/L-40e154a.wav"
[ 5, 25 ] = wave (0) : "./MITfull/elev-40/L-40e161a.wav"
[ 5, 26 ] = wave (0) : "./MITfull/elev-40/L-40e167a.wav"
[ 5, 27 ] = wave (0) : "./MITfull/elev-40/L-40e174a.wav"
[ 5, 28 ] = wave (0) : "./MITfull/elev-40/L-40e180a.wav"
[ 5, 29 ] = wave (0) : "./MITfull/elev-40/L-40e186a.wav"
[ 5, 30 ] = wave (0) : "./MITfull/elev-40/L-40e193a.wav"
[ 5, 31 ] = wave (0) : "./MITfull/elev-40/L-40e199a.wav"
[ 5, 32 ] = wave (0) : "./MITfull/elev-40/L-40e206a.wav"
[ 5, 33 ] = wave (0) : "./MITfull/elev-40/L-40e212a.wav"
[ 5, 34 ] = wave (0) : "./MITfull/elev-40/L-40e219a.wav"
[ 5, 35 ] = wave (0) : "./MITfull/elev-40/L-40e225a.wav"
[ 5, 36 ] = wave (0) : "./MITfull/elev-40/L-40e231a.wav"
[ 5, 37 ] = wave (0) : "./MITfull/elev-40/L-40e238a.wav"
[ 5, 38 ] = wave (0) : "./MITfull/elev-40/L-40e244a.wav"
[ 5, 39 ] = wave (0) : "./MITfull/elev-40/L-40e251a.wav"
[ 5, 40 ] = wave (0) : "./MITfull/elev-40/L-40e257a.wav"
[ 5, 41 ] = wave (0) : "./MITfull/elev-40/L-40e264a.wav"
[ 5, 42 ] = wave (0) : "./MITfull/elev-40/L-40e270a.wav"
[ 5, 43 ] = wave (0) : "./MITfull/elev-40/L-40e276a.wav"
[ 5, 44 ] = wave (0) : "./MITfull/elev-40/L-40e283a.wav"
[ 5, 45 ] = wave (0) : "./MITfull/elev-40/L-40e289a.wav"
[ 5, 46 ] = wave (0) : "./MITfull/elev-40/L-40e296a.wav"
[ 5, 47 ] = wave (0) : "./MITfull/elev-40/L-40e302a.wav"
[ 5, 48 ] = wave (0) : "./MITfull/elev-40/L-40e309a.wav"
[ 5, 49 ] = wave (0) : "./MITfull/elev-40/L-40e315a.wav"
[ 5, 50 ] = wave (0) : "./MITfull/elev-40/L-40e321a.wav"
[ 5, 51 ] = wave (0) : "./MITfull/elev-40/L-40e328a.wav"
[ 5, 52 ] = wave (0) : "./MITfull/elev-40/L-40e334a.wav"
[ 5, 53 ] = wave (0) : "./MITfull/elev-40/L-40e341a.wav"
[ 5, 54 ] = wave (0) : "./MITfull/elev-40/L-40e347a.wav"
[ 5, 55 ] = wave (0) : "./MITfull/elev-40/L-40e354a.wav"
[ 6, 0 ] = wave (0) : "./MITfull/elev-30/L-30e000a.wav"
[ 6, 1 ] = wave (0) : "./MITfull/elev-30/L-30e006a.wav"
[ 6, 2 ] = wave (0) : "./MITfull/elev-30/L-30e012a.wav"
[ 6, 3 ] = wave (0) : "./MITfull/elev-30/L-30e018a.wav"
[ 6, 4 ] = wave (0) : "./MITfull/elev-30/L-30e024a.wav"
[ 6, 5 ] = wave (0) : "./MITfull/elev-30/L-30e030a.wav"
[ 6, 6 ] = wave (0) : "./MITfull/elev-30/L-30e036a.wav"
[ 6, 7 ] = wave (0) : "./MITfull/elev-30/L-30e042a.wav"
[ 6, 8 ] = wave (0) : "./MITfull/elev-30/L-30e048a.wav"
[ 6, 9 ] = wave (0) : "./MITfull/elev-30/L-30e054a.wav"
[ 6, 10 ] = wave (0) : "./MITfull/elev-30/L-30e060a.wav"
[ 6, 11 ] = wave (0) : "./MITfull/elev-30/L-30e066a.wav"
[ 6, 12 ] = wave (0) : "./MITfull/elev-30/L-30e072a.wav"
[ 6, 13 ] = wave (0) : "./MITfull/elev-30/L-30e078a.wav"
[ 6, 14 ] = wave (0) : "./MITfull/elev-30/L-30e084a.wav"
[ 6, 15 ] = wave (0) : "./MITfull/elev-30/L-30e090a.wav"
[ 6, 16 ] = wave (0) : "./MITfull/elev-30/L-30e096a.wav"
[ 6, 17 ] = wave (0) : "./MITfull/elev-30/L-30e102a.wav"
[ 6, 18 ] = wave (0) : "./MITfull/elev-30/L-30e108a.wav"
[ 6, 19 ] = wave (0) : "./MITfull/elev-30/L-30e114a.wav"
[ 6, 20 ] = wave (0) : "./MITfull/elev-30/L-30e120a.wav"
[ 6, 21 ] = wave (0) : "./MITfull/elev-30/L-30e126a.wav"
[ 6, 22 ] = wave (0) : "./MITfull/elev-30/L-30e132a.wav"
[ 6, 23 ] = wave (0) : "./MITfull/elev-30/L-30e138a.wav"
[ 6, 24 ] = wave (0) : "./MITfull/elev-30/L-30e144a.wav"
[ 6, 25 ] = wave (0) : "./MITfull/elev-30/L-30e150a.wav"
[ 6, 26 ] = wave (0) : "./MITfull/elev-30/L-30e156a.wav"
[ 6, 27 ] = wave (0) : "./MITfull/elev-30/L-30e162a.wav"
[ 6, 28 ] = wave (0) : "./MITfull/elev-30/L-30e168a.wav"
[ 6, 29 ] = wave (0) : "./MITfull/elev-30/L-30e174a.wav"
[ 6, 30 ] = wave (0) : "./MITfull/elev-30/L-30e180a.wav"
[ 6, 31 ] = wave (0) : "./MITfull/elev-30/L-30e186a.wav"
[ 6, 32 ] = wave (0) : "./MITfull/elev-30/L-30e192a.wav"
[ 6, 33 ] = wave (0) : "./MITfull/elev-30/L-30e198a.wav"
[ 6, 34 ] = wave (0) : "./MITfull/elev-30/L-30e204a.wav"
[ 6, 35 ] = wave (0) : "./MITfull/elev-30/L-30e210a.wav"
[ 6, 36 ] = wave (0) : "./MITfull/elev-30/L-30e216a.wav"
[ 6, 37 ] = wave (0) : "./MITfull/elev-30/L-30e222a.wav"
[ 6, 38 ] = wave (0) : "./MITfull/elev-30/L-30e228a.wav"
[ 6, 39 ] = wave (0) : "./MITfull/elev-30/L-30e234a.wav"
[ 6, 40 ] = wave (0) : "./MITfull/elev-30/L-30e240a.wav"
[ 6, 41 ] = wave (0) : "./MITfull/elev-30/L-30e246a.wav"
[ 6, 42 ] = wave (0) : "./MITfull/elev-30/L-30e252a.wav"
[ 6, 43 ] = wave (0) : "./MITfull/elev-30/L-30e258a.wav"
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/*
* OpenAL Info Utility
*
* Copyright (c) 2010 by Chris Robinson <chris.kcat@gmail.com>
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include <stdio.h>
#include <string.h>
#include "AL/alc.h"
#include "AL/al.h"
#include "AL/alext.h"
#ifndef ALC_ENUMERATE_ALL_EXT
#define ALC_DEFAULT_ALL_DEVICES_SPECIFIER 0x1012
#define ALC_ALL_DEVICES_SPECIFIER 0x1013
#endif
#ifndef ALC_EXT_EFX
#define ALC_EFX_MAJOR_VERSION 0x20001
#define ALC_EFX_MINOR_VERSION 0x20002
#define ALC_MAX_AUXILIARY_SENDS 0x20003
#endif
#define MAX_WIDTH 80
static void printList(const char *list, char separator)
{
size_t col = MAX_WIDTH, len;
const char *indent = " ";
const char *next;
if(!list || *list == '\0')
{
fprintf(stdout, "\n%s!!! none !!!\n", indent);
return;
}
do {
next = strchr(list, separator);
if(next)
{
len = next-list;
do {
next++;
} while(*next == separator);
}
else
len = strlen(list);
if(len + col + 2 >= MAX_WIDTH)
{
fprintf(stdout, "\n%s", indent);
col = strlen(indent);
}
else
{
fputc(' ', stdout);
col++;
}
len = fwrite(list, 1, len, stdout);
col += len;
if(!next || *next == '\0')
break;
fputc(',', stdout);
col++;
list = next;
} while(1);
fputc('\n', stdout);
}
static void printDeviceList(const char *list)
{
if(!list || *list == '\0')
printf(" !!! none !!!\n");
else do {
printf(" %s\n", list);
list += strlen(list) + 1;
} while(*list != '\0');
}
static ALenum checkALErrors(int linenum)
{
ALenum err = alGetError();
if(err != AL_NO_ERROR)
printf("OpenAL Error: %s (0x%x), @ %d\n", alGetString(err), err, linenum);
return err;
}
#define checkALErrors() checkALErrors(__LINE__)
static ALCenum checkALCErrors(ALCdevice *device, int linenum)
{
ALCenum err = alcGetError(device);
if(err != ALC_NO_ERROR)
printf("ALC Error: %s (0x%x), @ %d\n", alcGetString(device, err), err, linenum);
return err;
}
#define checkALCErrors(x) checkALCErrors((x),__LINE__)
static void printALCInfo(ALCdevice *device)
{
ALCint major, minor;
if(device)
{
const ALCchar *devname = NULL;
printf("\n");
if(alcIsExtensionPresent(device, "ALC_ENUMERATE_ALL_EXT") != AL_FALSE)
devname = alcGetString(device, ALC_ALL_DEVICES_SPECIFIER);
if(checkALCErrors(device) != ALC_NO_ERROR || !devname)
devname = alcGetString(device, ALC_DEVICE_SPECIFIER);
printf("** Info for device \"%s\" **\n", devname);
}
alcGetIntegerv(device, ALC_MAJOR_VERSION, 1, &major);
alcGetIntegerv(device, ALC_MINOR_VERSION, 1, &minor);
if(checkALCErrors(device) == ALC_NO_ERROR)
printf("ALC version: %d.%d\n", major, minor);
if(device)
{
printf("ALC extensions:");
printList(alcGetString(device, ALC_EXTENSIONS), ' ');
checkALCErrors(device);
}
}
static void printALInfo(void)
{
printf("OpenAL vendor string: %s\n", alGetString(AL_VENDOR));
printf("OpenAL renderer string: %s\n", alGetString(AL_RENDERER));
printf("OpenAL version string: %s\n", alGetString(AL_VERSION));
printf("OpenAL extensions:");
printList(alGetString(AL_EXTENSIONS), ' ');
checkALErrors();
}
static void printEFXInfo(ALCdevice *device)
{
ALCint major, minor, sends;
static const ALchar filters[][32] = {
"AL_FILTER_LOWPASS", "AL_FILTER_HIGHPASS", "AL_FILTER_BANDPASS", ""
};
char filterNames[] = "Low-pass,High-pass,Band-pass,";
static const ALchar effects[][32] = {
"AL_EFFECT_EAXREVERB", "AL_EFFECT_REVERB", "AL_EFFECT_CHORUS",
"AL_EFFECT_DISTORTION", "AL_EFFECT_ECHO", "AL_EFFECT_FLANGER",
"AL_EFFECT_FREQUENCY_SHIFTER", "AL_EFFECT_VOCAL_MORPHER",
"AL_EFFECT_PITCH_SHIFTER", "AL_EFFECT_RING_MODULATOR",
"AL_EFFECT_AUTOWAH", "AL_EFFECT_COMPRESSOR", "AL_EFFECT_EQUALIZER", ""
};
static const ALchar dedeffects[][64] = {
"AL_EFFECT_DEDICATED_DIALOGUE",
"AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT", ""
};
char effectNames[] = "EAX Reverb,Reverb,Chorus,Distortion,Echo,Flanger,"
"Frequency Shifter,Vocal Morpher,Pitch Shifter,"
"Ring Modulator,Autowah,Compressor,Equalizer,"
"Dedicated Dialog,Dedicated LFE,";
char *current;
int i;
if(alcIsExtensionPresent(device, "ALC_EXT_EFX") == AL_FALSE)
{
printf("EFX not available\n");
return;
}
alcGetIntegerv(device, ALC_EFX_MAJOR_VERSION, 1, &major);
alcGetIntegerv(device, ALC_EFX_MINOR_VERSION, 1, &minor);
if(checkALCErrors(device) == ALC_NO_ERROR)
printf("EFX version: %d.%d\n", major, minor);
alcGetIntegerv(device, ALC_MAX_AUXILIARY_SENDS, 1, &sends);
if(checkALCErrors(device) == ALC_NO_ERROR)
printf("Max auxiliary sends: %d\n", sends);
current = filterNames;
for(i = 0;filters[i][0];i++)
{
char *next = strchr(current, ',');
ALenum val;
val = alGetEnumValue(filters[i]);
if(alGetError() != AL_NO_ERROR || val == 0 || val == -1)
memmove(current, next+1, strlen(next));
else
current = next+1;
}
printf("Supported filters:");
printList(filterNames, ',');
current = effectNames;
for(i = 0;effects[i][0];i++)
{
char *next = strchr(current, ',');
ALenum val;
val = alGetEnumValue(effects[i]);
if(alGetError() != AL_NO_ERROR || val == 0 || val == -1)
memmove(current, next+1, strlen(next));
else
current = next+1;
}
if(alcIsExtensionPresent(device, "ALC_EXT_DEDICATED"))
{
for(i = 0;dedeffects[i][0];i++)
{
char *next = strchr(current, ',');
ALenum val;
val = alGetEnumValue(dedeffects[i]);
if(alGetError() != AL_NO_ERROR || val == 0 || val == -1)
memmove(current, next+1, strlen(next));
else
current = next+1;
}
}
else
{
for(i = 0;dedeffects[i][0];i++)
{
char *next = strchr(current, ',');
memmove(current, next+1, strlen(next));
}
}
printf("Supported effects:");
printList(effectNames, ',');
}
int main(int argc, char *argv[])
{
ALCdevice *device;
ALCcontext *context;
if(argc > 1 && (strcmp(argv[1], "--help") == 0 ||
strcmp(argv[1], "-h") == 0))
{
printf("Usage: %s [playback device]\n", argv[0]);
return 0;
}
printf("Available playback devices:\n");
if(alcIsExtensionPresent(NULL, "ALC_ENUMERATE_ALL_EXT") != AL_FALSE)
printDeviceList(alcGetString(NULL, ALC_ALL_DEVICES_SPECIFIER));
else
printDeviceList(alcGetString(NULL, ALC_DEVICE_SPECIFIER));
printf("Available capture devices:\n");
printDeviceList(alcGetString(NULL, ALC_CAPTURE_DEVICE_SPECIFIER));
if(alcIsExtensionPresent(NULL, "ALC_ENUMERATE_ALL_EXT") != AL_FALSE)
printf("Default playback device: %s\n",
alcGetString(NULL, ALC_DEFAULT_ALL_DEVICES_SPECIFIER));
else
printf("Default playback device: %s\n",
alcGetString(NULL, ALC_DEFAULT_DEVICE_SPECIFIER));
printf("Default capture device: %s\n",
alcGetString(NULL, ALC_CAPTURE_DEFAULT_DEVICE_SPECIFIER));
printALCInfo(NULL);
device = alcOpenDevice((argc>1) ? argv[1] : NULL);
if(!device)
{
printf("\n!!! Failed to open %s !!!\n\n", ((argc>1) ? argv[1] : "default device"));
return 1;
}
printALCInfo(device);
context = alcCreateContext(device, NULL);
if(!context || alcMakeContextCurrent(context) == ALC_FALSE)
{
if(context)
alcDestroyContext(context);
alcCloseDevice(device);
printf("\n!!! Failed to set a context !!!\n\n");
return 1;
}
printALInfo();
printEFXInfo(device);
alcMakeContextCurrent(NULL);
alcDestroyContext(context);
alcCloseDevice(device);
return 0;
}