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1419 Commits

Author SHA1 Message Date
Chris Robinson c0cf323e1d Release 1.20.0 2019-11-28 00:45:08 -08:00
Chris Robinson 39e3484dbe Update ChangeLog 2019-11-22 22:47:36 -08:00
Chris Robinson eb8922596a Apply the full HRIR length for the B-Format decoder 2019-11-21 23:55:10 -08:00
Chris Robinson 2fd8c619b3 Rename a couple struct fields 2019-11-21 02:58:35 -08:00
Chris Robinson 7bc4a27900 Allocate storage for full HRTF coefficient lengths 2019-11-21 02:43:34 -08:00
Chris Robinson f2eab3e919 Properly get the AppData path on Windows in alsoft-config 2019-11-20 14:22:04 -08:00
Chris Robinson 6159b837cf Add a first-order ambisonic decoder for HRTF 2019-11-18 00:32:01 -08:00
Chris Robinson c86a28af5c Simplify some pi statements 2019-11-16 14:33:09 -08:00
Chris Robinson 6ad252efda Use wrappers to distinguish elevation and azimuth values 2019-11-16 14:07:31 -08:00
Chris Robinson d120e1464f Improve precision of the HRTF ambisonic decoder matrix 2019-11-13 08:39:27 -08:00
Chris Robinson 0ad512dd51 Clear the whole response HRIR before blending 2019-11-10 21:56:19 -08:00
Chris Robinson 74cbba511d Limit HRTF ambisonic decoding to second-order
The generated third-order matrix has incorrect first-order coefficients,
indicating a wonky decoder. The generated second-order matrix looks more
stable.
2019-11-09 13:04:44 -08:00
kcat 89938b95cf Merge pull request #356 from Raulshc/modulator_fix
EFX: Modulator fix
2019-11-09 06:58:29 -08:00
Raulshc c5f88ab59f EFX: Add explicit cast to a square function
MSVC 2015 and above returns the expression according to its datatype.
In this case, returns 4294967295 instead of -1.
2019-11-09 12:12:53 +01:00
Chris Robinson d2608e4bde Avoid holding HRTF accumulation samples per-source
It notably simplifies things to mix HRTF sources into an accumulation buffer
together, which the Dry buffer's Ambisonic-to-HRTF decode is then added to,
before being mixed to the Real output.
2019-11-03 00:30:33 -07:00
Chris Robinson 2741a94e58 Use rotate to reorder the default list entry 2019-10-28 23:47:30 -07:00
Chris Robinson d37a294d4c Add a config option for custom ALSA capture devices 2019-10-27 19:20:26 -07:00
Chris Robinson 205ff0080e Adjust padding in alsoft-config and remove an invalid signal 2019-10-27 18:02:25 -07:00
Chris Robinson 101afb1ee4 Convert HRTF field distances to meters when creating storage 2019-10-27 15:50:59 -07:00
Chris Robinson 697da8724f Avoid an extraneous macro and use a simplified type alias 2019-10-27 00:47:33 -07:00
Chris Robinson cac4072781 Change a few more really small gain values to 0 2019-10-26 19:57:17 -07:00
Chris Robinson 0cba99ed1b Avoid static constexpr for arrays iterated over at run-time 2019-10-25 01:43:23 -07:00
Chris Robinson 5ac1f19241 Don't dereference the end iterator 2019-10-24 21:58:35 -07:00
Chris Robinson 48347f62b5 Don't trace twice for a new HRTF data file 2019-10-24 15:28:28 -07:00
Chris Robinson f0977e5cc9 Print the found file entries after sorting 2019-10-24 15:10:37 -07:00
Chris Robinson 7bed2fe7cc Report the real ambisonic order set for HRTF rendering 2019-10-24 14:33:43 -07:00
Chris Robinson f40e69f9e3 Use a 26-point Lebedev grid for the HRTF ambisonic decode 2019-10-23 23:36:02 -07:00
Chris Robinson 0dfdebdf6d Limit the number of azimuths and elevations used in SOFA files 2019-10-22 16:30:07 -07:00
Chris Robinson 4733fc6f1e Fix azimuth limit 2019-10-22 15:23:17 -07:00
Chris Robinson a8a3acb6f6 More consistently use doubles in makemhr and loadsofa 2019-10-22 15:22:37 -07:00
Chris Robinson fcde56e1fc Increase the max elevation and azimuth count for HRTFs 2019-10-21 11:30:39 -07:00
Chris Robinson f58167e72d More sanely handle the voice state when mixing 2019-10-14 20:45:23 -07:00
Chris Robinson 33fd1f9efd Use better types for some specific sizes 2019-10-13 09:37:07 -07:00
Chris Robinson e70f98c95a Wrap the cycle amount when passing to sin() 2019-10-12 16:41:13 -07:00
Chris Robinson 630d4d573d Gracefully drain the OpenSL capture buffer on disconnect 2019-10-10 13:33:27 -07:00
Chris Robinson d2053e6784 Use one PulseAudio mainloop per device
To help avoid devices blocking on each other when handling asynchronous
messages.
2019-10-09 23:25:56 -07:00
Chris Robinson 8bf3da0cd2 Remove a useless prebuf check with PulseAudio 2019-10-09 09:45:02 -07:00
Chris Robinson 57cdac3368 Create the initial PulseAudio stream corked 2019-10-09 09:42:35 -07:00
Chris Robinson 404f3e2d08 Don't track the PulseAudio context state in devices 2019-10-09 05:01:30 -07:00
Chris Robinson b687e952ef Make C callbacks noexcept
No telling what would happen if exceptions managed to get back into presumably
C-based callers.
2019-10-09 03:29:25 -07:00
Chris Robinson a7c1245360 Use smart pointers for ALSA param handles
And simplify some related error handling
2019-10-09 02:16:08 -07:00
Chris Robinson 2842df5a02 Catch exceptions from backend start calls 2019-10-09 00:11:19 -07:00
Chris Robinson 963580c2d5 Never return null from CreateRingBuffer
Allocation failure would already throw a bad_alloc anyway, now a size overflow
throws an exception too.
2019-10-08 21:55:03 -07:00
Chris Robinson 7726a06d26 Clean up some exception messages and avoid duplicate log messages 2019-10-08 05:44:38 -07:00
Chris Robinson 360330b2ad Define some simple wrapper methods inline 2019-10-08 03:41:49 -07:00
Chris Robinson f4bc3d7ab2 Improve logging for Windows 2019-10-08 00:21:21 -07:00
Chris Robinson 52a003e9bb Avoid raw lock/unlock calls 2019-10-07 23:22:06 -07:00
Chris Robinson f0fa6c6baf Fix BackendVase typo 2019-10-07 22:42:54 -07:00
Chris Robinson 9b411cfcbe Reduce some indentation 2019-10-07 22:25:45 -07:00
Chris Robinson 02d80cd74d Use exceptions for backend open failures 2019-10-07 21:37:56 -07:00
Chris Robinson f8ff4e269b Put the pragma defines in a separate header 2019-10-07 15:26:35 -07:00
Chris Robinson f52eed0607 Update some wording in the changelog 2019-10-07 05:01:05 -07:00
Chris Robinson acb6baad90 Use std::array instead of plain arrays in a couple places 2019-10-05 21:23:31 -07:00
Chris Robinson eb89faf963 Use a span instead of a reference-to-array 2019-10-05 20:00:22 -07:00
Chris Robinson 267b79f337 Avoid duplicate structs 2019-10-05 16:11:38 -07:00
Chris Robinson 1bb93f4fc2 Avoid direct function template and alias types
It's somewhat ambiguous what they mean. Sometimes acting as a pointer, other
times having weird behavior. Pointer-to-function types are explicitly defined
as such, whereas uses of these tend to be as references (never null and not
changeable).
2019-10-03 04:22:39 -07:00
Chris Robinson b350ae3766 Remove the Offset parameter from ApplyCoeffs 2019-10-02 22:38:19 -07:00
Chris Robinson bce6889173 Remove an unnecessary struct specifier 2019-10-02 22:20:34 -07:00
Chris Robinson 324fb8d0b7 Fix a comment 2019-10-02 19:22:14 -07:00
Chris Robinson 58085f1c7b Clean up some unnecessary includes 2019-10-02 19:13:07 -07:00
Chris Robinson 50198ee30c Clean up some ALfloat -> float 2019-10-02 17:07:23 -07:00
Chris Robinson 64e2c377d8 Move ALvoice from alu.h to a separate header 2019-10-02 16:53:23 -07:00
Chris Robinson d639935e19 Move a couple types to the source they're used in 2019-10-02 15:37:33 -07:00
Chris Robinson a35cac7ce0 Rename mixvoice.cpp to voice.cpp 2019-10-02 15:29:01 -07:00
Chris Robinson 4620912f0f Don't inline the utf8 converters 2019-10-01 23:33:00 -07:00
Chris Robinson 2980adb8c0 Make sure the temporary HRIRs are properly aligned 2019-10-01 22:50:28 -07:00
Chris Robinson 69eb685dbc Check for librt earlier 2019-10-01 22:35:33 -07:00
Chris Robinson accac7950c Silence some warnings from GCC in the router 2019-10-01 22:22:46 -07:00
Chris Robinson 28d54efe72 Use al::getenv to get the router env vars 2019-10-01 22:15:40 -07:00
Chris Robinson a0a55d300f Remove an unnecessary function 2019-10-01 21:45:44 -07:00
Chris Robinson e2c1602ede Link the examples with librt if it exists 2019-10-01 19:56:39 -07:00
Chris Robinson 0063f4bfac Add some allocator fields GCC 6.3 seems to want 2019-10-01 01:49:21 -07:00
Chris Robinson 8f6fafd19a Update the changelog 2019-09-30 22:23:14 -07:00
Chris Robinson 5d7a1fa6da Clear the HRTF state values on allocation 2019-09-30 19:29:01 -07:00
Chris Robinson 4d127a2f98 Avoid infs/nans in the crest detector
It needs to be investigated why the rendered mix sometimes has such large
sample values when starting, but the compressor/limiter shouldn't generate NaNs
because of it.
2019-09-30 17:29:04 -07:00
Chris Robinson cf617760b6 Separate a couple assignments from conditionals 2019-09-30 03:03:27 -07:00
Chris Robinson 0139d8a04f Remove noexcept from a function that explicitly throws 2019-09-30 02:57:19 -07:00
Chris Robinson c833af9ecd Repack the bsinc resamplers coefficients
This puts the base coefficients and the phase deltas next to each other. This
improves caching, as the base and phase deltas are always used together while
the scales are only used for the non-fast versions.
2019-09-29 23:39:04 -07:00
Chris Robinson fb56b02041 Modify the bsinc resampler
Readjusted the bsinc12 cutoff back to -60dB. Also increased the filter's phase
count.
2019-09-29 22:43:46 -07:00
Chris Robinson 94ff2daafc Use using to avoid extraneous template instantiations 2019-09-29 16:02:25 -07:00
Chris Robinson fabb9add9b Silence an MSVC warning 2019-09-28 23:31:49 -07:00
Chris Robinson 00250042c8 Check MAX_RESAMPLER_PADDING properly to ensure it's large enough 2019-09-28 19:14:27 -07:00
Chris Robinson 7783fa738c Make the BSincTables constexpr in an anonymous namespace 2019-09-28 16:40:38 -07:00
Chris Robinson 4b746b8d37 Make MAX_RESAMPLER_PADDING specify the total padding 2019-09-28 14:35:42 -07:00
Chris Robinson 31ffb0887c Don't let a function end without a return 2019-09-28 13:56:51 -07:00
Chris Robinson 2d2e5539c0 Use FastBSinc24 for WASAPI and CoreAudio capture
Given a fixed rate, there's no downside to the fast version.
2019-09-28 03:44:13 -07:00
Chris Robinson cbc00bcffe Combine two function calls into one 2019-09-28 03:42:17 -07:00
Chris Robinson f7b574c8f2 Redo resampler strings to be safer
Now the name is guaranteed to match the type, and to be ordered as the enum
declares.
2019-09-28 03:15:48 -07:00
Chris Robinson 882b4acae8 Add "fast" variants for the bsinc resamplers
This simply omits the scale factor from the filter, similar to how up-sampling
does. The consequence of this is less smooth transitions when ramping the
pitch while down-sampling, but otherwise behaves fine.
2019-09-28 01:58:29 -07:00
Chris Robinson 00d5356b96 Remove the unnecessary FRACTIONONE from bsincgen 2019-09-28 00:01:21 -07:00
Chris Robinson 9b64e5e0db Implement a "fast" bsinc path
This takes advantage of the fact than when increment <= 1 (when not down-
sampling), the scale factor is always 0. As a result, the scale and scale-phase
deltas never contribute to the filtered output. Removing those multiply+add
operations cuts half of the work done by the inner loop.

Sounds that do need to down-sample (when played with a high pitch, or is 48khz
on 44.1khz output, for example), still go through the normal bsinc process.
2019-09-26 19:24:29 -07:00
Chris Robinson d50ca464cd Use a span for holding the source handles 2019-09-25 03:01:58 -07:00
Chris Robinson f009219523 Avoid extraneous parameters 2019-09-25 02:11:37 -07:00
Chris Robinson 5bb91b858f Use blended HRIRs for the B-Format decode 2019-09-24 22:27:12 -07:00
Chris Robinson 16a715a2b5 Fix unsigned 8-bit buffers 2019-09-24 21:49:12 -07:00
Chris Robinson 6963712d46 Use an array to match the HRTF rendering method 2019-09-24 13:51:40 -07:00
Chris Robinson 96865d6b99 Only build ex-common as needed 2019-09-24 01:16:57 -07:00
Chris Robinson bf31ce688a Add a missing header for Android 2019-09-23 19:23:33 -07:00
Chris Robinson e01b32f8e0 Use istream for makemhr input 2019-09-23 18:37:36 -07:00
Chris Robinson 34a0fad6cd Use a unique_ptr to auto-free MYSOFA_HRTF 2019-09-23 17:38:43 -07:00
Chris Robinson 4a111f7671 Update changelog 2019-09-22 21:27:59 -07:00
Chris Robinson 61ffa23e44 Fix a couple more conversion warnings 2019-09-22 21:27:10 -07:00
Chris Robinson 24db8a3f4b Make the resampler type an enum class 2019-09-22 21:19:19 -07:00
Chris Robinson 95996effaf Move the ifstream wrapper to common 2019-09-22 12:23:41 -07:00
Chris Robinson 9c95f62e95 Remove large file macros
Large file offsets aren't being utilized, and C++ seems to use to 64-bit
offsets anyway.
2019-09-22 11:17:31 -07:00
Chris Robinson 5f6a35c960 Avoid storing an integer in a pointer
C++ does not guarantee that, given an int of sufficient size, converting
int->ptr->int will result in the original value. A pointer may have more than
one integer representation. Only ptr->int->ptr round trips are well-defined.
2019-09-21 23:35:24 -07:00
Chris Robinson 9325f1d507 Split some code into separate functions 2019-09-21 21:14:11 -07:00
Chris Robinson 9993ea818c Restructure the changelog updates 2019-09-21 19:46:19 -07:00
Chris Robinson 564c953e94 Make the buffer frequency unsigned 2019-09-21 16:47:33 -07:00
Chris Robinson 146afcbd70 Explicitly mark a couple functions as inline 2019-09-21 16:12:48 -07:00
Chris Robinson c83055ff53 Initial update for the changelog 2019-09-21 15:49:18 -07:00
Chris Robinson 28e62456ac Use an array and loop instead of individual tests 2019-09-21 12:08:45 -07:00
Chris Robinson 8907a4fb04 Cleanup some router warnings 2019-09-20 15:45:59 -07:00
Chris Robinson a0a8af47ae Include VS2017 in AppVeyor builds 2019-09-20 14:56:21 -07:00
Chris Robinson 2ae1f8b8b6 Revert "Silence some unreachable code warnings on MSVC"
This reverts commit 2ab4883439.

It apparently didn't work at silencing anything.
2019-09-20 14:40:12 -07:00
Chris Robinson 2ab4883439 Silence some unreachable code warnings on MSVC 2019-09-20 14:25:06 -07:00
Chris Robinson 65eb0987e2 Remove and simplify some functions 2019-09-20 13:35:29 -07:00
Chris Robinson 79a621ac47 Simplify some REQUIRES uses 2019-09-20 10:59:12 -07:00
Chris Robinson b9daffe159 Don't clean up more than necessary on destruction 2019-09-20 10:58:29 -07:00
Chris Robinson 057b253adb Use an anonymous namespace instead of static 2019-09-19 21:01:10 -07:00
Chris Robinson c0678816fc Remove deprecated CMAKE_ALLOW_LOOSE_LOOP_CONSTRUCTS 2019-09-19 18:50:45 -07:00
Chris Robinson ffb5623ead Fix a couple more annoying conversion warnings 2019-09-18 20:13:58 -07:00
Chris Robinson a092624ed3 Fix some Windows warnings 2019-09-18 18:20:11 -07:00
Chris Robinson 9a0a5c79d8 Avoid hiding a class member function 2019-09-18 17:02:33 -07:00
Chris Robinson 24a1058952 Work around MSVC's lack of standard _Pragma support 2019-09-18 13:58:04 -07:00
Chris Robinson b8f64155e9 Silence a couple specific warning instances 2019-09-18 13:24:16 -07:00
Chris Robinson 66565ca7a3 Enable and fix some more warnings 2019-09-18 10:09:04 -07:00
Chris Robinson da80a7c2b2 Disable warnings 4268 and 4324 on MSVC
warning C4268: 'const' static/global data initialized with compiler generated
default constructor fills the object with zeros

warning C4324: structure was padded due to alignment specifier
2019-09-18 09:31:32 -07:00
Chris Robinson ab87e56b22 Enable some more MSVC warnings 2019-09-18 09:01:46 -07:00
Chris Robinson dffb874478 Some cleanup for CMakeLists.txt 2019-09-18 08:58:36 -07:00
Chris Robinson aca9f4e095 Make the bsinc l and m coefficients unsigned 2019-09-17 18:38:46 -07:00
Chris Robinson 1da7512628 Avoid an unnecessary multiply 2019-09-16 21:49:06 -07:00
Chris Robinson 62972babea Remove unnecessary long long checks 2019-09-16 18:10:01 -07:00
Chris Robinson 6398bd8373 Add preprocessor defines for the common lib 2019-09-16 15:57:05 -07:00
Chris Robinson b00128411f Add a missing include directory for the common lib 2019-09-16 15:45:20 -07:00
Chris Robinson 899a414591 Remove the last uses of the system's str[n]casecmp 2019-09-16 15:10:36 -07:00
Chris Robinson 6d93b2ba81 Use our case-insensitive compare functions in makemhr 2019-09-16 14:55:52 -07:00
Chris Robinson 7a9463f863 Make the common code a separate static library 2019-09-16 14:46:41 -07:00
Chris Robinson 10a841cd7e Avoid an unused variable 2019-09-16 14:10:48 -07:00
Chris Robinson 2c5c5a5397 Add and use custom string types and functions 2019-09-16 13:45:14 -07:00
Chris Robinson 650764775f Fix typo 2019-09-16 07:50:39 -07:00
Chris Robinson fcf850c1dd Clean up some more conversion warnings 2019-09-16 07:32:13 -07:00
Chris Robinson bf2c865d39 Clean up some more shadowing warnings 2019-09-16 07:16:31 -07:00
Chris Robinson e16e2269b6 Add a fallback if SLAndroidDataFormat_PCM_EX isn't available 2019-09-15 20:46:37 -07:00
Chris Robinson 35129d66b7 Clean up the spaghetti mess in alcCaptureSamples 2019-09-15 13:42:56 -07:00
Chris Robinson a999df8ce5 Rename ALautowah* for consistency 2019-09-15 12:58:58 -07:00
Chris Robinson 47f0d301dc Make the BackendFactory base destructor protected 2019-09-15 10:22:56 -07:00
Chris Robinson c5a3c52822 Return and pass more appropriate types for backends 2019-09-15 09:50:28 -07:00
Chris Robinson e4cc77ea23 Fix a few warnings from MSVC 2019-09-14 20:19:59 -07:00
Chris Robinson 4b8f78a8d6 Fix a few more GCC warnings 2019-09-14 19:42:54 -07:00
Chris Robinson e82c43fb38 Fix a copy-paste error 2019-09-14 19:07:30 -07:00
Chris Robinson b43cd047ba Build with -Wunused 2019-09-14 18:54:16 -07:00
Chris Robinson b212ab9ea2 Add a note about clearing complex_hilbert's imaginary input 2019-09-14 18:37:00 -07:00
Chris Robinson 807d3b64ca Enable and fix more warnings 2019-09-14 18:35:23 -07:00
Chris Robinson 2cc0600476 Avoid recalculations in the formant filter process 2019-09-14 18:35:23 -07:00
Chris Robinson d75de6ee4d Don't hold the pulse lock while not necessary 2019-09-14 18:35:23 -07:00
Chris Robinson 2c348cecb6 Fix some more implicit conversions noted by GCC 2019-09-14 18:35:23 -07:00
kcat 1c45b1791b Merge pull request #344 from Lopuska/patch-3
removed unnecessary loop
2019-09-14 18:34:53 -07:00
kcat 6644a660e8 Merge pull request #342 from Lopuska/patch-1
pass proper effects slot number to AddActiveEffectSlots
2019-09-14 18:29:48 -07:00
Lopuska 925894fb8b removed unnecessary loop
the caller in fshifter is already doing the same job by putting 0 for the imaginary part
2019-09-15 02:58:53 +02:00
Lopuska 0023614da5 pass proper effects slot number to AddActiveEffectSlots
It resolves wrong logic when creating more than 1 aux slot at time in a batch
2019-09-15 02:09:05 +02:00
Chris Robinson 4e8caea97e Enable -Wconversion warnings
This is now clean with Clang 7, let's see what other compilers give.
2019-09-14 12:50:44 -07:00
Chris Robinson 532197a650 Clean up (most) implicit conversions in the backends 2019-09-14 12:44:35 -07:00
Chris Robinson fa64b1fd6d Fix implicit conversions in the effects 2019-09-14 12:16:51 -07:00
Chris Robinson 3675dfcfef Use an unsigned index value 2019-09-14 08:50:07 -07:00
Chris Robinson 88364dcbc7 Fix more implicit conversions 2019-09-14 08:29:02 -07:00
Chris Robinson 2646f509ee Store the ambisonic order as unsigned 2019-09-13 20:04:22 -07:00
Chris Robinson 42ae95b8fa Remove a couple no-op statements 2019-09-13 14:44:52 -07:00
Chris Robinson a250b6a986 Return unsigned values from the FromDevFmt functions 2019-09-13 14:29:25 -07:00
Chris Robinson fcd3bed0c0 Clean up more implicit conversions 2019-09-13 12:51:16 -07:00
Chris Robinson f09734b707 Pass IrSize to the HRTF mixers as unsigned 2019-09-13 11:19:11 -07:00
Chris Robinson 4254888576 Make IrSize unsigned 2019-09-13 09:38:35 -07:00
Chris Robinson df306b5524 Make NumAuxSends unsigned 2019-09-13 04:15:05 -07:00
Chris Robinson 5b763e1437 Clean up implicit conversions in alu.cpp 2019-09-13 03:47:23 -07:00
Chris Robinson c1690178ec Make the resampler increment unsigned 2019-09-13 03:25:13 -07:00
Chris Robinson 5f862a5b49 Clean up sample converter implicit conversions 2019-09-12 17:45:06 -07:00
Chris Robinson 5ca8796d6a Clean up some lambda definitions 2019-09-12 17:10:33 -07:00
Chris Robinson 70b58d79fe Fix source limit check 2019-09-12 12:19:07 -07:00
Chris Robinson 6ca8fadd58 Fix counting free objects 2019-09-12 12:14:23 -07:00
Chris Robinson ac48569c69 Avoid a macro call and mark unlikely paths as unlikely 2019-09-12 11:59:36 -07:00
Chris Robinson 2e010d29cb Clean up implicit conversions in source.cpp 2019-09-12 11:33:04 -07:00
Chris Robinson c0ce03d8b9 Get rid of more implicit conversions 2019-09-12 06:29:32 -07:00
Chris Robinson b71eb4dafd Don't use [[nodiscard]] in C++11
To silence some warnings in older compilers, and fix an error with newer MSVC.
2019-09-12 04:30:52 -07:00
Chris Robinson 6699f3cf1c Use unsigned channel indices 2019-09-12 04:17:21 -07:00
Chris Robinson 4c76f32dda Avoid implicit conversions with the examples and utils 2019-09-12 03:22:34 -07:00
Chris Robinson 474d478854 Only pass -Wold-style-cast for C++ 2019-09-11 15:55:37 -07:00
Chris Robinson 07e2aa3bc0 Clean up some more implicit conversions 2019-09-11 14:33:26 -07:00
Chris Robinson 681c78d348 Allocate buffer batches separately from buffers 2019-09-11 12:28:33 -07:00
Chris Robinson 1a9f1e0869 Fix a few more C-style casts 2019-09-11 08:08:23 -07:00
Chris Robinson 14c76ca244 Fix allocator comparison operators 2019-09-11 07:32:14 -07:00
Chris Robinson a895709b6f Fix function declaration 2019-09-11 06:58:27 -07:00
Chris Robinson be0442c620 Avoid C-style casts in C++ 2019-09-11 06:47:56 -07:00
Chris Robinson 388928f3aa Fix some more implicit casts 2019-09-11 05:53:10 -07:00
Chris Robinson e4b15aeefc Fix some implicit casts 2019-09-11 04:55:54 -07:00
Chris Robinson 5b37e2339b Simplify flexible array member usage 2019-09-11 03:59:53 -07:00
Chris Robinson c6c5048416 Don't inherit for the allocator 2019-09-11 03:22:10 -07:00
Chris Robinson 65374dc5d0 Avoid dynamically allocating ChannelConverter 2019-09-10 23:01:33 -07:00
Chris Robinson 19e1cd7430 Make hq-mode the default and update ambisonics.txt 2019-09-08 01:38:20 -07:00
Chris Robinson 449c09bf03 Remove mention of an unsupported env var 2019-09-08 00:59:56 -07:00
Chris Robinson ec8b56ef3b Remove unneeded TRACEREF logging 2019-09-08 00:59:10 -07:00
Chris Robinson fcb496bb5c Avoid unnecessary explicit copy methods 2019-09-08 00:31:55 -07:00
Chris Robinson 987fd13796 Use a new voice when restarting a playing source 2019-09-06 21:20:20 -07:00
Chris Robinson e5c9643dd5 Combine two functions into one 2019-09-05 15:08:22 -07:00
Chris Robinson 97c043da07 Don't set voice properties in ApplyOffset 2019-09-05 04:00:49 -07:00
Chris Robinson ef2769af03 Use a normal vector for the voices array 2019-09-04 23:04:55 -07:00
Chris Robinson c47a6d2279 Increment the mix count during disconnect
So attempts to get the current playback offset behave correctly while
disconnecting.
2019-09-04 20:31:55 -07:00
Chris Robinson e873165019 Only ASSUME values where a variable is used 2019-09-04 19:31:30 -07:00
Chris Robinson 8940bbd034 Only use one accumulation buffer for B-Format HRTF mixing
It's all getting added together anyway and all channels are continuous inputs,
so this is fewer passes over various buffers.
2019-09-03 23:19:24 -07:00
Chris Robinson e1ee1bcde9 Clean up some formatting 2019-09-03 21:34:58 -07:00
Chris Robinson 3d7ce5a860 Use global placement new for AL object batches 2019-09-01 18:07:16 -07:00
Chris Robinson bb35e24c9b Avoid unnecessary placement new definitions 2019-09-01 17:54:17 -07:00
Chris Robinson 727217ce0a Ensure AL_STOPPED state change events are sent
With explicit calls to alSourceStop on a playing or paused source
2019-09-01 15:28:33 -07:00
Chris Robinson d15950f383 Avoid accumulating a temporary 2019-09-01 15:03:22 -07:00
Chris Robinson aee10ef606 Hold the source lock in UpdateAllSourceProps 2019-09-01 00:29:26 -07:00
Chris Robinson a15f25b07a Return a QString from GetVersionString 2019-08-31 22:28:26 -07:00
Chris Robinson 5becf4bb73 Don't include version.h in mainwindow.cpp
Should help with post-commit compile times
2019-08-31 21:47:58 -07:00
Chris Robinson 96af306b3c Update the language for travis 2019-08-31 20:24:34 -07:00
Chris Robinson 3973334a64 Store the voice fraction offset as unsigned 2019-08-31 15:49:34 -07:00
Chris Robinson a546343148 Make MixVoice a member function 2019-08-31 14:53:28 -07:00
Chris Robinson 5b6772440a Remove a couple unnecessary variables 2019-08-27 12:31:19 -07:00
Chris Robinson 3223b6a906 Update mainwindow.cpp
Use the less-error-prone pointer-to-member syntax for connect calls, and use
uniform initialization more often.
2019-08-26 19:13:31 -07:00
Chris Robinson b5a6dcc14f Remove some unnecessary local spans 2019-08-26 11:02:09 -07:00
Chris Robinson 7dbf69afa2 Use a span for effect state input 2019-08-26 09:16:20 -07:00
Chris Robinson 12e6dc0cce Fix a function's sample count type 2019-08-25 21:30:32 -07:00
Chris Robinson 4e72d7b3ec Use al::byte for a couple more buffers 2019-08-25 18:09:14 -07:00
Chris Robinson 164626a7be Pass unsigned sample count to aluMixData 2019-08-25 17:54:36 -07:00
Chris Robinson b93098f7df Use unsigned sample counts for the compressor/limiter 2019-08-25 17:24:53 -07:00
Chris Robinson 88a8bf903b Use size_t for the post-process sample length 2019-08-25 15:36:40 -07:00
Chris Robinson e165cae3cd Fade reverb over the whole update
Since the early and late panning gains fade over the course of the update, it
should match the fading done by the feedback loops to avoid percussive "blasts"
when transitioning to a long-decay low-gain environment from a short-decay
high-gain environment.
2019-08-24 16:25:56 -07:00
Chris Robinson 02d38091c8 Make a couple functions into member functions 2019-08-23 15:34:45 -07:00
Chris Robinson b7ab8b6896 Fix a variable declaration 2019-08-22 18:53:09 -07:00
Chris Robinson 7ad2ed965c Avoid reading from pointers to __m128 values 2019-08-21 07:43:28 -07:00
Chris Robinson 024d5d900a Dereference the correct buffer when destructing a source 2019-08-21 03:00:19 -07:00
Chris Robinson bb46cec0b1 Pass samplesToDo as size_t to effects 2019-08-20 14:30:04 -07:00
Chris Robinson 3e499e70fd Try to improve non-dynamic-extent span construction 2019-08-20 12:31:46 -07:00
Chris Robinson 3a6676b61e Fix NEON store call 2019-08-20 12:19:38 -07:00
Chris Robinson 80710c146a Use size_t for the mixers' fade counter and outpos 2019-08-20 12:00:24 -07:00
Chris Robinson 3fa83547e9 Use size_t for HrtfMixer functions' buffer size 2019-08-20 11:39:39 -07:00
Chris Robinson a19f65f2c8 Use size_t for reverb offsets and masks 2019-08-20 11:13:19 -07:00
Chris Robinson 9e326846f6 Simplify passing some span parameters 2019-08-20 10:36:17 -07:00
Chris Robinson a3a295f6dc Try to fix span construction for MSVC 2019-08-20 09:31:08 -07:00
Chris Robinson a7f078927d Fix MixRow definition for NEON 2019-08-20 09:20:18 -07:00
Chris Robinson 0961f4eb00 Pass a span to the Resample function 2019-08-20 08:46:12 -07:00
Chris Robinson ce76cc1441 Use unsigned for the sample and channel converters 2019-08-20 07:57:37 -07:00
Chris Robinson 2ab52968f4 Track the MixVoice sample count and fade counter as unsigned 2019-08-20 06:03:31 -07:00
Chris Robinson 7a8f81259c Use size_t for the NFC and biquad filters' sample count 2019-08-20 05:26:51 -07:00
Chris Robinson 102ef6cb33 Pass a span for the Mix function's input 2019-08-20 04:16:44 -07:00
Chris Robinson 4883091f5d Rename the Mix function input for clarity 2019-08-20 01:24:02 -07:00
Chris Robinson 8fd90334a1 Pass the MixRow buffer size as a span 2019-08-20 00:27:28 -07:00
Chris Robinson 34a61122f6 Change NUM_LINES to a size_t for MSVC 2019-08-19 09:49:01 -07:00
Chris Robinson 34331c0dfa Fix for GCC5 decaying an array to a pointer 2019-08-19 09:14:54 -07:00
Chris Robinson c0cd43d100 More logically separate temp reverb buffers 2019-08-19 08:46:57 -07:00
Chris Robinson 2d0568c048 Allow using a variable channel stride for MixRowSamples 2019-08-19 08:02:08 -07:00
Chris Robinson a9223ae601 Improve subspan default template argument 2019-08-18 23:26:49 -07:00
Chris Robinson 2bbdd329cd Use the appropriate type 2019-08-18 15:37:39 -07:00
Chris Robinson 6629b65ab9 Formatting cleanup 2019-08-18 15:25:18 -07:00
Chris Robinson 4f4ef3a410 Don't require MixRow's output to be a FloatBufferLine 2019-08-18 15:15:56 -07:00
kcat 6750907bdd Merge pull request #329 from Raulshc/fshifter
EFX: Update Frequency shifter
2019-08-18 15:15:19 -07:00
kcat 71ddb86047 Merge pull request #328 from Raulshc/double2int
Move double2int function
2019-08-18 15:15:09 -07:00
kcat c89fb550b6 Merge pull request #327 from Raulshc/alsoft_config_vmorpher
Alsoft-config: Add vocal morpher effect
2019-08-18 15:15:01 -07:00
Raulshc eaaa194163 EFX: Update Frequency shifter
Add f. shifter processing for L and R channels.
2019-08-18 19:28:00 +02:00
Raulshc 550f1dce1f Move double2int function
Move inline double2int function to alnumeric.h from pshifter.cpp
2019-08-18 19:12:38 +02:00
Raulshc 42602b9ede Alsoft-config: Add vocal morpher effect 2019-08-18 18:49:52 +02:00
Chris Robinson ff66061091 Reduce the size of reverb's temporary buffer storage
The size of ReverbState is now almost half of what it was.
2019-08-18 03:19:19 -07:00
Chris Robinson 16886bd259 Avoid an out parameter for VectorPartialScatter 2019-08-17 23:00:46 -07:00
Chris Robinson ddf7e0f07c Use a std::array for the reverb sample buffer 2019-08-17 21:23:30 -07:00
Chris Robinson 351ccf2e11 Use new/delete for context and effectslot properties 2019-08-13 22:25:59 -07:00
Chris Robinson ecab99bce9 Move update pointers to the containers they update 2019-08-13 22:06:14 -07:00
Chris Robinson 0806a003e2 Use new/delete for listener properties 2019-08-13 20:33:44 -07:00
Chris Robinson 91ff01d1ad Don't explicitly inline a particular function 2019-08-13 20:33:26 -07:00
Chris Robinson bc9f39b5ac Environment variables should override config settings 2019-08-12 17:10:04 -07:00
Chris Robinson 38a565bdf8 Move UID definitions to a separate source 2019-08-12 14:56:17 -07:00
Chris Robinson dff906c13b Remove always-true assumption 2019-08-12 14:32:04 -07:00
Chris Robinson 81d17bb80f Fix some return types 2019-08-12 14:30:47 -07:00
Chris Robinson 1aaf65abfe Add methods to get env vars as an optional 2019-08-12 03:59:52 -07:00
Chris Robinson 50d16d2422 Don't use INT_MAX for unknown span lengths
32-bit doesn't like it, for some reason. Use the largest possible length it
could be.
2019-08-11 22:48:18 -07:00
Chris Robinson f290bf2751 Move vector.h to common 2019-08-11 20:54:21 -07:00
Chris Robinson e200569cd3 Move the wstr converters to a separate header 2019-08-11 18:50:07 -07:00
Chris Robinson 7118733458 Remove multiple buffers per queue item
And simplify related code
2019-08-11 14:01:57 -07:00
Chris Robinson 70058a8a84 Move the dynload decls and defs to common 2019-08-10 21:54:30 -07:00
Chris Robinson dca2365051 Fix array access index 2019-08-10 20:31:08 -07:00
Chris Robinson e3f5bd37e6 Avoid __popcnt[64] on MSVC
It requires SSE4, and provides no fallback mechanism for CPU targets lacking
the opcode it maps to.
2019-08-10 16:35:29 -07:00
Chris Robinson 98029d64b9 Fix and clarify the peaking biquad filter 2019-08-10 14:41:55 -07:00
kcat 8f93656f53 Merge pull request #323 from Raulshc/vmorpher_decls
Add parameter DECLs for vocal morpher
2019-08-10 14:41:25 -07:00
Raulshc c4f1c95a45 Add parameter DECLs for vocal morpher 2019-08-10 18:42:02 +02:00
kcat 926b515edf Merge pull request #322 from gongminmin/SDK1903
Enable to detect Windows SDK v1903
2019-08-10 03:18:01 -07:00
Minmin Gong becd1cf1bd Enable to detect Windows SDK v1903 2019-08-09 23:04:37 -07:00
Chris Robinson f900efa7f2 Simplify and fix vocal morpher pitch calculations 2019-08-08 15:22:22 -07:00
Chris Robinson fb1fde9fb0 Simplify the weighted decay time calculation 2019-08-08 12:44:53 -07:00
Chris Robinson a4391a213d Turn a couple methods into member functions 2019-08-08 12:40:34 -07:00
Chris Robinson e22dc27218 Improve log formatting for file searches 2019-08-07 15:56:38 -07:00
Chris Robinson 0eb5e80f67 Don't store options that are set to blank/default 2019-08-07 15:43:37 -07:00
Chris Robinson 5b33b4aa97 Make the post-process methods member functions 2019-08-07 11:43:53 -07:00
Chris Robinson 7c069e29ee Update some more includes 2019-08-06 23:13:05 -07:00
kcat 50c59dab8f Merge pull request #321 from Frozen-Tofu/master
no AudioObjectPropertyAddress  on iOS
2019-08-06 23:11:55 -07:00
Li Keqing ad059fb62e no AudioObjectPropertyAddress on iOS 2019-08-07 13:29:26 +08:00
Chris Robinson 80a85febcf Update some includes 2019-08-05 18:36:39 -07:00
Chris Robinson d06afa7ca1 Move a struct definition to where it's used 2019-08-05 15:11:47 -07:00
Chris Robinson e286ec8d09 Move some declarations out of alcmain.h 2019-08-05 15:03:18 -07:00
Chris Robinson d24401c3f3 Move the meters per unit property to the listener 2019-08-05 12:15:14 -07:00
Chris Robinson 3154a915b1 Remove the ReverbSpeedOfSound hack
No other effect depends on context or listener properties, so reverb being the
only exception for speed of sound and meters per unit was putting extra work on
the effect engine for no real reason. Especially since the reverb decay time
should be the time actual time to decay irrespective of other settings.
2019-08-05 11:37:05 -07:00
Chris Robinson 3bc9490fd2 Move some structs to where they're used 2019-08-04 17:45:46 -07:00
Chris Robinson 082622951d Don't allow numeric values for the resampler option 2019-08-04 17:29:55 -07:00
Chris Robinson 2fa2c35bdc Modify LIKELY and UNLIKELY to not need extra parenthesis 2019-08-04 11:59:14 -07:00
Chris Robinson 7897de31d0 Pass spans to the source get-property methods 2019-08-03 19:36:19 -07:00
Chris Robinson 7baa07e3ad Pass spans to the source set-property methods
This avoids an extra property check to get the size since the number of values
can be checked when accessed.
2019-08-03 18:57:38 -07:00
Chris Robinson 559d1666b8 Add a Create method to FlexArray for "raw" arrays 2019-08-03 14:59:01 -07:00
Chris Robinson 13222d719d Make a couple counts size_t 2019-08-03 13:05:42 -07:00
Chris Robinson 3417436804 Avoid manually incrementing a reference count 2019-08-03 12:14:50 -07:00
Chris Robinson a7a9c00275 Turn a couple more functions into methods 2019-08-02 18:30:22 -07:00
Chris Robinson 9f223898f2 Use an unsigned voice index 2019-08-02 12:38:20 -07:00
Chris Robinson ce7c86b217 Avoid reloading the voices array when processing 2019-08-02 11:07:48 -07:00
Chris Robinson 61bb079036 Add operator* to intrusive_ptr 2019-08-01 20:15:47 -07:00
Chris Robinson 33bcced82a Use a smart pointer for holding the context's device 2019-08-01 19:44:09 -07:00
Chris Robinson 4917024c94 Reduce the AsyncEvent struct size
The "user" message length is significantly reduced to fit the struct in 256
bytes, rather than 1KB.
2019-08-01 15:54:17 -07:00
Chris Robinson 0be823320d Add and use an intrusive_ptr type 2019-08-01 15:19:37 -07:00
Chris Robinson 57e7fff6f6 Remove an unnecessary variable 2019-08-01 14:29:02 -07:00
Chris Robinson 65f7fc610e Add a common base for auto-deleting ref-counted objects
Which will also work as the basis for a future intrusive_ptr
2019-08-01 13:43:32 -07:00
Chris Robinson 380f3dc11d Cleanup alcontext.h includes 2019-08-01 09:21:56 -07:00
Chris Robinson 62534f424a Use float2 where appropriate 2019-07-31 11:05:53 -07:00
Chris Robinson 151ff51e7d Don't templatize HrirArray 2019-07-31 10:46:33 -07:00
Chris Robinson d1f72624a7 Fix a couple ASSUME statements 2019-07-31 10:28:04 -07:00
Chris Robinson 471f905fbd Use enums for the resampler and mixer template tags 2019-07-31 10:09:43 -07:00
Chris Robinson a0aa5bc80a Add iterators to ALbufferlistitem
And change some types to ALuint
2019-07-31 09:20:53 -07:00
Chris Robinson 4cd7eee01c Remove improper include 2019-07-30 21:50:47 -07:00
Chris Robinson f286c3fa38 Move another function to a ALCcontext method 2019-07-30 21:32:05 -07:00
Chris Robinson ac554de67d Turn some functions into methods 2019-07-30 14:13:05 -07:00
Chris Robinson ea76e003e7 Properly prefix ALCcontext members 2019-07-30 09:05:54 -07:00
Chris Robinson 488d1de944 More include cleanup 2019-07-29 19:59:48 -07:00
Chris Robinson 06e5454eb9 Use Transposed Direct Form II for the BS2B filters 2019-07-29 19:40:03 -07:00
Chris Robinson 4c9e18c5a0 Rename al/* sources to avoid camel-case 2019-07-29 17:54:07 -07:00
Chris Robinson c2de0782cf Minor formatting fixes 2019-07-29 16:49:56 -07:00
Chris Robinson 76d87330ec Move the event declarations to a separate header 2019-07-29 15:58:26 -07:00
Chris Robinson 0a26bab14e Rename the OpenAL32 directory to al 2019-07-29 15:40:17 -07:00
Chris Robinson 8ccb7604d3 Remove some unnecessary cmake checks 2019-07-29 13:46:25 -07:00
Chris Robinson f0408809d6 Cleanup common sources' includes 2019-07-29 09:29:35 -07:00
Chris Robinson 40e937c63a Cleanup the examples' includes 2019-07-29 08:21:38 -07:00
Chris Robinson d38d255364 More include cleanups 2019-07-28 21:29:59 -07:00
Chris Robinson 83432a7e5c Move some headers out of the Include subdirectory 2019-07-28 19:09:07 -07:00
Chris Robinson cb3e96e756 Rename Alc to alc 2019-07-28 18:56:04 -07:00
Chris Robinson 93e60919c8 Rename alMain.h to alcmain.h
And move it and alu.h to Alc/.
2019-07-28 18:33:29 -07:00
Chris Robinson bb0062625f Move the ADPCM decoders to alBuffer.cpp 2019-07-28 17:22:00 -07:00
Chris Robinson b4d56d3fdf Remove the UNUSED macro 2019-07-28 17:15:34 -07:00
Chris Robinson c8bbd75bf9 Remove a couple more cmake checks 2019-07-28 15:37:12 -07:00
Chris Robinson 12e179d539 Remove some unnecessary header checks 2019-07-28 15:22:58 -07:00
Chris Robinson 585d965c92 Remove a couple unnecessary includes 2019-07-28 15:16:13 -07:00
Chris Robinson e0a795d9d2 Clean up some more headers 2019-07-28 14:55:02 -07:00
Chris Robinson 5428d6acc3 Clean up includes a bit
Trying out the IWYU tool to only include what's necessary in a given file.
Seems to work decently (it'll miss some headers, suggest unnecessary ones, and
make nonsense suggestions for some things, but overall gives a good starting
point), and helps clean out some headers.
2019-07-28 11:28:36 -07:00
Chris Robinson 659b6d4245 Use more proper cmake to set the C/C++ standard version 2019-07-27 18:58:19 -07:00
Chris Robinson 7cfb353334 Don't explicitly check for standard functions 2019-07-26 14:02:14 -07:00
Chris Robinson b22ecc45c9 Increase the video picture queue size to 24 2019-07-26 03:44:46 -07:00
Chris Robinson 18f1139de8 Only redraw the image when necessary 2019-07-23 12:51:14 -07:00
Chris Robinson b8ac4b79e4 Only send packets as needed 2019-07-23 02:38:07 -07:00
Chris Robinson 8a9434c623 Use a local variable to track the decoded pts 2019-07-20 18:46:43 -07:00
Chris Robinson f0ed35d3d0 Set the initial clock time closer to starting playback 2019-07-20 01:10:14 -07:00
Chris Robinson 9959d661a0 Restructure codec send/receive calls
In particular, after an initial fill of the codec's internal buffer, each
receive_frame call is followed by one or more send_packet calls. For
asynchronous codecs, this has the effect of letting the codec work while the
handler thread is waiting for an AVFrame structure to become available or
waiting for more decoded data to be needed. For synchronous codecs, this
makes the send_packet calls use up time that would be spent waiting.
2019-07-19 22:55:20 -07:00
Chris Robinson 29cf87f34d Combine duplicate code into a function 2019-07-18 16:04:45 -07:00
kcat ec855215db Merge pull request #318 from Lopuska/pitchshift_for_vmorpher
pitch shift for formant filters
2019-07-18 15:42:45 -07:00
Anis 31055f48d6 pitch shift for formant filters 2019-07-18 21:42:59 +02:00
Chris Robinson 2b21a08f89 Receive video frames in a loop 2019-07-16 20:20:25 -07:00
Chris Robinson 101e641288 Fix an unused parameter warning 2019-07-16 16:19:51 -07:00
Chris Robinson ffc7258cbc Remove an unused lambda capture 2019-07-16 16:04:56 -07:00
kcat 8a9e72109e Merge pull request #317 from Lopuska/vocal_morpher_improvements
misc fixes and improvements for Vocal Morpher
2019-07-16 16:04:27 -07:00
Anis ee013521ee misc fixes and improvements for Vocal Morpher 2019-07-16 17:40:18 +02:00
Philip Muzzall 28f07d2d5e Misc fixes (#315)
* Added rc scripts for dll

* Reverted numbering scheme in CMakeLists

* Misc fixes
2019-07-14 17:39:45 -07:00
Chris Robinson b728cf7bd6 Properly include getopt.h 2019-07-14 04:05:08 -07:00
Chris Robinson 4ff7bfd2d8 Use atomics for the picture queue 2019-07-14 03:59:57 -07:00
Chris Robinson ac7eeeae79 Don't use the same mutex for the video clock 2019-07-14 03:59:57 -07:00
Chris Robinson 93c53e33f0 Receive frames directly into the picture's AVFrame 2019-07-14 03:59:57 -07:00
Chris Robinson e7e734f8b9 Don't use one texture per picture in alffplay 2019-07-14 03:59:56 -07:00
kcat d2f71ae42a Merge pull request #314 from PenguinDOOM/pr-master
Fixed makemhr and sofa-info errors
2019-07-14 03:58:38 -07:00
Penguin 4027664fc2 Fixed MSVC error. 2019-07-14 18:03:49 +09:00
Penguin 313549c76d Add missing header. 2019-07-14 18:03:32 +09:00
Chris Robinson 363e2fb73a MSVC doesn't like parenthesized type initialization 2019-07-13 16:51:19 -07:00
Chris Robinson f99474c913 Handle alffplay video using continuous rendering 2019-07-12 23:34:21 -07:00
Chris Robinson 79d572cf72 Properly comment out the sample config's options 2019-07-12 22:13:19 -07:00
Chris Robinson 2e6c7808b0 Try to improve alffplay timing again 2019-07-11 03:54:26 -07:00
Chris Robinson 99b55bc230 Add the Windows SDK for the winmm library path 2019-07-10 20:06:50 -07:00
Chris Robinson 93d0c8993d Fix OpenSL library name 2019-07-10 19:55:14 -07:00
Chris Robinson e2f2b74d6a Get rid of the custom CHECK_SHARED_FUNCTION_EXISTS function 2019-07-10 19:32:26 -07:00
Chris Robinson ac28b7d0f2 Use a find module for OpenSL 2019-07-10 18:54:43 -07:00
Chris Robinson 426c4587cc Some clean up to use uniform initialization 2019-07-10 02:13:28 -07:00
Chris Robinson 159024acc9 Improve alffplay video clock timing 2019-07-09 22:15:05 -07:00
Lopuska 4a33bbb14d vocal morpher implementation (#312)
* vocal morpher implementation

* compile fix for GCC
2019-07-09 22:14:31 -07:00
Chris Robinson 3ffb6867a3 Rework packet handling in alffplay
Turns out avcodec_send_packet is what can invoke the decode for serialized
codecs, so don't call that in the parse handler thread. The packet queue is
used to get the compressed data from the parse handler to the audio/video
threads.

Additionally, don't serialize the video frame preparation with the decode
thread.
2019-07-08 13:18:10 -07:00
Chris Robinson 2783b4c04b Somewhat simplify alffplay playback timing 2019-07-06 22:18:52 -07:00
Chris Robinson c230554851 Clamp NFC reference distance between 0.1m and 10m 2019-07-06 19:09:26 -07:00
Chris Robinson b95bf8d7c1 Update a comment about the speaker distance 2019-07-06 18:39:26 -07:00
Chris Robinson c9f6f9652a Avoid looking up source IDs multiple times 2019-07-06 14:59:26 -07:00
Chris Robinson 47246e5205 Don't stop the device before going over the attributes 2019-07-06 13:20:21 -07:00
Chris Robinson aeb7fe52f1 Use unsigned more consistently for source counts 2019-07-06 00:19:48 -07:00
Chris Robinson fa032368ae Rename HRTF Quality title to HRTF Render Method 2019-07-05 12:27:58 -07:00
Chris Robinson 3edbeefdb7 Add a setting for hrtf-mode to alsoft-config 2019-07-04 22:59:12 -07:00
Chris Robinson 474073955b Pass a span to ApplyStablizer 2019-07-04 17:00:01 -07:00
Chris Robinson 3fe5ef272f Use a span for MixParams 2019-07-04 15:02:12 -07:00
Chris Robinson 729ffe02d6 Reorder some methods for better placement 2019-07-04 14:03:27 -07:00
Chris Robinson 949507c891 Use a span for RealMixParams 2019-07-03 23:26:33 -07:00
Chris Robinson 9a51ca0a78 Pass a span to BFormatDec::process 2019-07-03 22:59:29 -07:00
Chris Robinson 0a0704071a Allocate device buffer when setting the channel counts 2019-07-03 22:32:39 -07:00
Chris Robinson 61ba455edd Don't warn about non-multiple-of-4 update sizes
It's not always possible to do anything about it, especially for backends that
aren't restricted to the period size, and it's not really a problem anyway
(still getting SIMD benefits for the vast majority of samples).
2019-07-02 01:56:59 -07:00
Minmin Gong cee8100f19 Remove noexcept from sampler functions to match the signature of SamplerT (#309)
This change fixes compilation problems on gcc 9.1 with -std=c++17.
2019-07-01 22:42:20 -07:00
Chris Robinson bc1d058d2d Add a helper to construct the optional value 2019-07-01 17:55:56 -07:00
Chris Robinson 6bb0edf0a5 Create and use a make_optional method 2019-07-01 17:25:58 -07:00
Chris Robinson 53c13de5ce Simplify some optional usage 2019-07-01 16:28:51 -07:00
Chris Robinson eb70171433 Add a few more constructor and assignment operators for optional 2019-07-01 16:27:49 -07:00
Chris Robinson 143ad16051 Use uninitialized_copy/move for optionals 2019-07-01 12:34:24 -07:00
Chris Robinson c9ffa9d466 Add C++17-like uninitialized_move methods 2019-07-01 12:33:39 -07:00
Chris Robinson e9b41d9b90 Don't unnecessarily force the output limiter on 2019-07-01 11:13:23 -07:00
Chris Robinson 871257b69e Some cleanup for optional 2019-07-01 10:44:55 -07:00
Chris Robinson 0066ac26f0 Add a missing return 2019-07-01 10:44:55 -07:00
kcat 22311169d8 Merge pull request #307 from Lopuska/patch-2
Corrected old naming
2019-07-01 10:41:20 -07:00
Lopuska a5197177be Corrected old naming 2019-07-01 14:58:32 +02:00
Chris Robinson 0c2edd10dd Remove BOM markers and set UTF-8 codepage for rc files 2019-07-01 00:28:06 -07:00
Chris Robinson 90b1bc7b7a Make sure a variable is set before use 2019-06-30 23:32:09 -07:00
Chris Robinson 01300d9735 Convert rc files to UTF-8
MinGW chokes on them being UTF-16
2019-06-30 22:29:21 -07:00
Chris Robinson 0aeaf06173 Properly set DisableVideo to true 2019-06-30 22:02:40 -07:00
Philip Muzzall 7affe3d78d Added rc scripts for dll (#306)
* Added rc scripts for dll

* Reverted numbering scheme in CMakeLists
2019-06-30 22:01:04 -07:00
Chris Robinson e9bf7e4b15 Add an option to disable video in alffplay 2019-06-30 18:17:23 -07:00
Chris Robinson c21b7e2461 Use explicit storage types for some enums 2019-06-30 17:41:43 -07:00
Chris Robinson f458642237 Use optionals where methods may not return a valid value 2019-06-30 16:57:10 -07:00
Chris Robinson 49ceae681b Return optionals from the remaining ConfigValue* methods 2019-06-30 16:40:08 -07:00
Chris Robinson 3658dafdcb Use an optional for ConfigValueStr 2019-06-30 16:40:08 -07:00
Chris Robinson db026454f2 Fix Y channel offset for the UHJ encoder 2019-06-30 16:40:08 -07:00
Chris Robinson 0fc30151f2 Use an optional for ConfigValueUInt 2019-06-30 16:40:08 -07:00
Chris Robinson 51f53afe12 Use an optional for ConfigValueInt 2019-06-30 16:40:08 -07:00
Chris Robinson 689f70ce6d Add a simple optional<> implementation 2019-06-30 16:40:08 -07:00
kcat 15ac839054 Merge pull request #305 from Lopuska/patch-1
avoid extra local member declaration
2019-06-30 16:39:36 -07:00
Lopuska 541a9383a7 avoid extra local member declaration 2019-06-30 23:17:13 +02:00
Chris Robinson ada86d2fcf Hold references in the global device and context lists 2019-06-29 22:38:38 -07:00
Chris Robinson 7303e22d89 Use a FlexArray for a device's contexts 2019-06-29 21:32:36 -07:00
Chris Robinson cbcee69ed1 Add an empty() method to FlexArray 2019-06-29 18:53:20 -07:00
Chris Robinson c7797fa10f Declare variables closer to their use 2019-06-29 15:04:20 -07:00
Chris Robinson 811d0738fa Put some static functions in an anonymous namespace 2019-06-29 12:16:38 -07:00
Chris Robinson b905a224ee Avoid a generic function for specialized behavior 2019-06-29 11:45:55 -07:00
Chris Robinson 6790a9b44f Use a bool for the TrapALError flag 2019-06-29 11:22:29 -07:00
Chris Robinson 12911cc533 Use a range-for loop to concatenate available backends 2019-06-29 10:23:29 -07:00
Chris Robinson 3ede66ae45 Don't keep retrieving the backend factories 2019-06-28 18:54:31 -07:00
kcat c238619a8b Merge pull request #301 from gongminmin/FixGcc9
Fix the error "a reinterpret_cast is not a constant expression" in GCC 9
2019-06-27 23:29:48 -07:00
Minmin Gong 9326b98e2d Fix the error "a reinterpret_cast is not a constant expression" on GCC 9 2019-06-27 22:29:42 -07:00
Chris Robinson 2598ee6f0a Properly set extra fields in MakeExtensible 2019-06-27 09:47:31 -07:00
Chris Robinson 341c68c9a1 Trace the requested and required formats for WASAPI 2019-06-27 09:41:52 -07:00
Chris Robinson 360869f23b Use a span for some post-mixing processing 2019-06-27 08:49:04 -07:00
Chris Robinson 48191cbf0a Use a span instead of explicit current and end pointers 2019-06-25 17:10:46 -07:00
Chris Robinson edd75b3b4e Don't return a blank span when offset==size() 2019-06-25 14:40:25 -07:00
Chris Robinson ab3c2ea777 Don't bother with a passthru filter method
There's not really a case where it'll process less than 2 samples, and it was
wrong anyway.
2019-06-24 21:18:25 -07:00
Chris Robinson f5cb6ac20e Use a reference instead of the same array index 2019-06-24 13:33:09 -07:00
Chris Robinson eb5ff42d15 Correctly index the send params 2019-06-24 13:21:06 -07:00
Chris Robinson e8cef0cdae Clean up all unused HRTFs when going through them 2019-06-24 12:55:36 -07:00
Chris Robinson b46fc572f5 Use spans to handle pulseaudio capture buffers 2019-06-23 17:08:52 -07:00
Chris Robinson ee983bda75 Document the hrtf-mode config option 2019-06-21 12:43:33 -07:00
kcat 281e61e4cc Merge pull request #298 from TeamHypersomnia/master
Fix ambiguous calls to destroy_at
2019-06-21 12:42:55 -07:00
geneotech 499aa65a4b Fix ambiguous calls to destroy_at 2019-06-21 18:25:09 +02:00
Chris Robinson 723ab7f15a Don't inline ifstream destructor 2019-06-18 23:04:24 -07:00
Chris Robinson 0fa984027c Simplify al_print 2019-06-18 22:57:48 -07:00
Chris Robinson 706df72d18 Rename HrtfParams to HrtfFilter 2019-06-18 06:20:35 -07:00
Chris Robinson 2b19c53ece Use al::byte instead of ALbyte for raw bytes 2019-06-17 20:26:00 -07:00
Chris Robinson 55785c0a16 Use al::make_unique for ChannelConverter 2019-06-17 20:15:37 -07:00
Chris Robinson 5a884572f0 Use a span instead of pointer+size parameters 2019-06-17 20:05:38 -07:00
Chris Robinson 21b8571f50 Avoid an implied else if check 2019-06-16 22:45:07 -07:00
Chris Robinson a009b9502a Avoid manually looping to destroy orphaned async events 2019-06-16 18:58:56 -07:00
Chris Robinson 0a532729ba Reorganize how some device fields are set and reset 2019-06-15 23:10:11 -07:00
Chris Robinson 8670fca3dc Fix BS2B output 2019-06-12 22:51:09 -07:00
Chris Robinson a2ba230e05 Combine two macros into one 2019-06-11 22:29:39 -07:00
Chris Robinson a478fd4b25 Fix unsigned short/int sample converters
And add const/noexcept in some places
2019-06-11 14:59:06 -07:00
Chris Robinson 97d56dd424 Use C++ I/O to check for NEON support 2019-06-10 22:29:58 -07:00
Chris Robinson 1a14946306 Get rid of an unnecessary constructor 2019-06-10 21:56:09 -07:00
Chris Robinson c22d537d93 Remove some extern "C" blocks 2019-06-10 21:45:33 -07:00
Chris Robinson 8bb42c2f98 Don't inline some file IO methods 2019-06-10 01:51:14 -07:00
Chris Robinson ec6fdff0c6 Make the voice count unsigned 2019-06-09 19:27:15 -07:00
Chris Robinson bc8f206ee1 Use a FlexArray for the context's voices 2019-06-09 18:13:54 -07:00
Chris Robinson 90d25e5187 Make sure the bitfield indices are constants 2019-06-09 02:20:30 -07:00
Chris Robinson 2e154069c6 Use a bitfield for the device flags 2019-06-08 23:49:15 -07:00
Chris Robinson c9ba7ba193 Add a bitfield class for indexed, auto-sized flags 2019-06-08 23:33:59 -07:00
Chris Robinson b6ce793f84 Use a span for the complex_fft/hilbert functions 2019-06-08 16:05:18 -07:00
Chris Robinson 1569b79c5d Fix for GCC 5.4 2019-06-08 02:17:08 -07:00
Chris Robinson 91b7e8142c Simplify DistanceComp somewhat 2019-06-08 01:39:28 -07:00
Chris Robinson 7988bc6e91 Add and use proper types for FlexArray 2019-06-07 23:42:31 -07:00
Chris Robinson 7537bb3492 Don't warn about standard functions with MSVC 2019-06-06 15:57:15 -07:00
Chris Robinson 87ccdf02a7 Avoid strcpy 2019-06-06 04:39:30 -07:00
Chris Robinson 56faf66887 Disable MSVC warning 4200 again 2019-06-06 03:32:28 -07:00
Chris Robinson 39c2b18cd4 Remove a couple unused functions 2019-06-06 00:42:59 -07:00
Chris Robinson 585b0cf3be Remove the DEF_ALIGN macro 2019-06-06 00:37:00 -07:00
Chris Robinson a7be531049 Remove the unused ALIGN macro 2019-06-06 00:24:13 -07:00
Chris Robinson d9d9e70ed8 Don't disable some MSVC warnings 2019-06-06 00:18:25 -07:00
Chris Robinson 5acae56ecd Use a FlexArray for ringbuffer storage 2019-06-05 23:29:13 -07:00
Chris Robinson 24df52c042 Remove the per-voice ChannelsPerOrder field 2019-06-05 23:00:28 -07:00
Chris Robinson f9da06fc6a Use a span for the effect state's output target 2019-06-05 19:58:58 -07:00
Chris Robinson 1ce310c6d1 Make some more channel counts unsigned 2019-06-05 19:26:54 -07:00
Chris Robinson 410a5ca621 Make RealMixParams channel count unsigned 2019-06-05 18:54:17 -07:00
Chris Robinson 5f26205f8f Properly destroy other objects 2019-06-05 17:25:08 -07:00
Chris Robinson f27e73989c Properly destroy the limiter's extra fields 2019-06-05 17:09:15 -07:00
Chris Robinson 142721df17 Add methods to construct and destruct objects in-place 2019-06-05 16:38:53 -07:00
Chris Robinson b2735331c0 Use a 16-sample base delay for the B-Format decoder IRs 2019-06-04 20:27:32 -07:00
Chris Robinson 9d861406c0 Avoid a separate struct for the bandsplitter all-pass 2019-06-04 17:50:36 -07:00
Chris Robinson 4522a51ea2 Don't log the function or prefix
It's ultimately unnecessary since the message is an indicator about where it
was logged from. The message itself is generally more important than where it
was from, too.
2019-06-04 01:37:36 -07:00
Chris Robinson f0bc9d8a9b Improve alignment handling for the alignment allocator 2019-06-03 22:58:56 -07:00
Chris Robinson c76fb714cc Restructure voice data members
This should improve access patters by packing each buffer channel's data
together, which is more inline with its use.
2019-06-03 22:24:26 -07:00
Chris Robinson 53e1415a67 Allow selecting the ambisonic order for basic HRTF rendering 2019-06-02 20:38:43 -07:00
Chris Robinson 8c4a9a5a32 Properly search for prebuilt native-tools 2019-06-01 11:21:43 -07:00
Chris Robinson 20ce461096 Make sure the T60 filter gains are properly clamped 2019-05-31 11:58:48 -07:00
Chris Robinson 6ee49cad52 Once more for MSVC 2019-05-30 18:55:24 -07:00
Chris Robinson 4bae4cbafb Another attempt to fix MSVC 2015 2019-05-30 18:39:51 -07:00
Chris Robinson a123c87ba5 Avoid some MSVC workarounds that didn't seem to work 2019-05-30 14:38:06 -07:00
Chris Robinson 1c8dfb55d8 Improve span constructor requirements
Particularly, properly account for the const-ness of the data returned by it.
2019-05-30 11:46:48 -07:00
Chris Robinson 76e7c8b244 Try to work around some MSVC short-comings 2019-05-30 10:52:28 -07:00
Chris Robinson 1961828d0b Avoid potentially ambiguous span copy constructor 2019-05-30 10:06:12 -07:00
Chris Robinson 153f133435 Try to fix "ambiguous" initializations with older compilers 2019-05-29 23:41:09 -07:00
Chris Robinson dbdf516dbf Use a span for the voice's buffer references 2019-05-29 23:06:24 -07:00
Chris Robinson 2909f263fd Use span<FloatBufferLine> for EffectState::process output 2019-05-29 22:31:36 -07:00
Chris Robinson 893ffe9a84 Use span<FloatBufferLine> for MixSamples 2019-05-29 21:58:37 -07:00
Chris Robinson 8af7b4c6e0 Use FloatBufferLine and span<> for MixRowSamples 2019-05-29 20:45:33 -07:00
Chris Robinson 7e6b6d7ad9 Use FloatBufferLine and span<> in the reverb effect 2019-05-29 19:36:06 -07:00
Chris Robinson ba449d2b08 Pass a span to MixDirectHrtf instead of a pointer+size 2019-05-29 17:32:16 -07:00
Chris Robinson d0f0a5fdca Separate two HRTF passes into two loops 2019-05-29 11:13:35 -07:00
Chris Robinson ec3a6f8cde Use FloatBufferLine for the effect process method 2019-05-29 09:37:25 -07:00
Chris Robinson b923eb1879 Use FloatBufferLine with the HRTF mixer functions 2019-05-28 22:44:50 -07:00
Chris Robinson 838e2bae80 Improve a couple algorithms 2019-05-28 17:18:22 -07:00
Chris Robinson c80ee5b701 Use std::array for most mixing buffer arrays 2019-05-28 16:22:36 -07:00
Chris Robinson 7ce2b632f5 Simplify template type requirement checking 2019-05-28 08:49:53 -07:00
Chris Robinson aa4b6afad6 Small cleanup for is_span and is_std_array 2019-05-27 22:01:59 -07:00
Chris Robinson bac52e9537 Fix default constructor for static-sized spans 2019-05-27 17:48:19 -07:00
Chris Robinson 28fbb5178a Remove unnecessary assignment operators 2019-05-27 15:56:50 -07:00
Chris Robinson aad49d666a Implement static-sized spans, and handle overload requirements
Note that span is specialized such that a static-sized span only has a single
data member, making it a suitable replacement for  Type (&arg)[Size]  style
variables/parameters.
2019-05-27 15:36:10 -07:00
Chris Robinson f57fedec7f Get rid of the COUNTOF macro 2019-05-26 21:28:51 -07:00
Chris Robinson f6f220025b Fix subspan 2019-05-26 21:00:17 -07:00
Chris Robinson 3007fbf5e5 Use a span for loading source samples 2019-05-26 20:05:12 -07:00
Chris Robinson b0e12ccf71 Ensure a couple calls are constexpr 2019-05-26 15:57:17 -07:00
Chris Robinson 01f717ae59 Use SL_ANDROID_DATAFORMAT_PCM_EX for extended PCM info 2019-05-26 14:45:53 -07:00
Chris Robinson 20e3c78aef Use al::byte instead of char for generic data storage 2019-05-26 12:54:54 -07:00
Chris Robinson 63a130204c Add a few more methods to the span class 2019-05-26 12:05:43 -07:00
Chris Robinson 7cbf82afe4 Avoid a few more NUM_LINES assumptions 2019-05-26 11:34:17 -07:00
Chris Robinson 8ca97a7d9a Move a couple functions into its related class 2019-05-25 11:54:51 -07:00
Chris Robinson 5b5dee07b4 Remove a couple unused functions 2019-05-25 11:36:13 -07:00
Chris Robinson ebf33b7c6b Avoid some uses of RESTRICT 2019-05-25 08:17:37 -07:00
Chris Robinson 219f818b16 Use al::byte for ADPCM decoders 2019-05-24 13:32:20 -07:00
Chris Robinson b4fbc271d2 Add byte ops that take an integer-based rhs parameter 2019-05-24 13:30:40 -07:00
Chris Robinson 674ca3cf24 Move a couple table definitions to where they're used 2019-05-24 12:06:52 -07:00
Chris Robinson 1f45cc051a Fix some truncation warnings with MSVC 2019-05-24 06:47:24 -07:00
Chris Robinson e90a6beaa2 Remove an unnecessary struct member 2019-05-24 06:25:18 -07:00
Chris Robinson 9c63bbd6ce Use raw bytes for the buffer data 2019-05-24 06:12:20 -07:00
Chris Robinson 1945b50834 Add a unique byte type for dealing with raw bytes 2019-05-24 06:11:21 -07:00
Chris Robinson 857473b6b0 Store the span extents as a pair of pointers 2019-05-24 04:55:38 -07:00
Chris Robinson 3a9caec72f Use a span for referencing the channel map to initialize 2019-05-23 13:30:16 -07:00
Chris Robinson 517b8158a7 Add a missing include 2019-05-23 08:30:02 -07:00
Chris Robinson b46eca97a6 Use a span for resource data 2019-05-23 08:17:05 -07:00
Chris Robinson a4af617532 Add a span class to act as a view to contiguous data 2019-05-23 08:15:02 -07:00
Chris Robinson e3d6f3d988 Use a function to mark a source for updating 2019-05-23 05:06:26 -07:00
Chris Robinson 6708504026 Properly reverse the HRTF field order when loading it
And combine a couple arrays into an array structure
2019-05-22 11:07:12 -07:00
Chris Robinson b48bab3394 Allow initializing splitter filters with constructors 2019-05-22 03:03:24 -07:00
Chris Robinson 657978c732 Don't change the format tag in MakeExtensible 2019-05-21 09:01:51 -07:00
Chris Robinson d502397b37 Simplify the echo feedback loop 2019-05-20 22:07:41 -07:00
Chris Robinson 55845f316d Don't use coverage spread for the echo spread 2019-05-20 21:30:56 -07:00
Chris Robinson 6dbd488d13 Avoid unnecessary use of CalcAngleCoeffs 2019-05-20 21:16:13 -07:00
Chris Robinson 136caf0cb0 Don't attenuate the repeated sample
The mixing gain stepping will handle it
2019-05-19 00:53:39 -07:00
Chris Robinson 515a201e30 Restructure some voice fields 2019-05-17 20:39:28 -07:00
Chris Robinson 5ac19673db Fix a couple type truncation warnings with MSVC 2019-05-12 19:41:34 -07:00
Chris Robinson 8d437a5135 Use available buffer samples when pausing a source 2019-05-05 01:51:46 -07:00
Chris Robinson 0759ed7213 Properly mark a likely branch 2019-05-05 00:05:41 -07:00
Chris Robinson 0c09c8378b Set the device channel order from PulseAudio's channel map 2019-05-04 23:45:27 -07:00
Chris Robinson 42dfefbb0e Fix when PulseAudio channel configuration changes 2019-05-04 22:11:29 -07:00
Chris Robinson 1b11ca48dc Avoid unnecessary parsing for channel maps 2019-05-04 20:54:49 -07:00
Chris Robinson 31c29f1ea8 Use exceptions when opening and reseting the PulseAudio backend 2019-05-04 20:35:24 -07:00
Chris Robinson 95d19be36b Catch exceptions from backend reset 2019-05-04 20:34:33 -07:00
Chris Robinson b9592bddbc Add the printf format attribute to backend_exception's constructor 2019-05-04 18:25:59 -07:00
Chris Robinson 5ff8d5ae32 Add an exception class to cover backend creation and opening 2019-05-04 18:03:25 -07:00
Chris Robinson 1607f9c525 Report the threshold limit for the output limiter 2019-05-04 13:53:11 -07:00
Chris Robinson 9eea2e4c73 Use BUFFERSIZE for the reverb loop limit
At 44/48khz, the main delay line comes out to 20k to 22k samples, which gets
rounded up to 32k as the next power of two. This leaves plenty of room for the
full 1k BUFFERSIZE without having to increase the delay line beyond what it
already is.
2019-05-03 12:59:04 -07:00
Chris Robinson 2f2ec2b6e3 Add some assumes and consts 2019-05-03 00:50:47 -07:00
Chris Robinson a72c47164c Avoid a few more array length assumptions 2019-05-01 11:15:17 -07:00
Chris Robinson 6281f6e85a Avoid masking offsets in the inner reverb loops 2019-04-30 23:28:57 -07:00
Chris Robinson 85439cbd87 Add some missing include directories 2019-04-29 20:03:51 -07:00
Chris Robinson fb52413a6e Make a number of settings tristate 2019-04-29 19:47:14 -07:00
Chris Robinson cef7eebed6 Disable NFC by default 2019-04-29 19:47:14 -07:00
Chris Robinson b42694ea31 Allow NFC filters without HQ decoding
It still requires a proper custom decoder configuration for speakers, so that
proper distances can be specified. Ambisonic output still relies on the
nfc-ref-delay option, and HRTF uses the dataset's distance.
2019-04-29 19:47:14 -07:00
kcat 1ab783c7ac Merge pull request #285 from ArthurSonzogni/master
Update CMakeLists.txt. Simplify examples build and export PUBLIC headers.
2019-04-29 19:46:39 -07:00
ArthurSonzogni 2eb657f2df CMAKE: export PUBLIC headers of OpenAL.
Remove the cmake function:
  INCLUDE_DIRECTORIES(..)
Replace it by:
  TARGET_INCLUDE_DIRECTORIES(...)

It gives us the opportunity to define whether or not OpenAL dependencies
should be exported or not (using PUBLIC or PRIVATE keywoard).

[user visible changes]
The OpenAL PUBLIC headers are exported. When a target depends on OpenAL,
it will have access to its public headers.

Some small refactor along the way.
2019-04-30 04:17:43 +02:00
Chris Robinson c7e388873c Increase the period size slider and don't use steps of 64 2019-04-28 22:21:23 -07:00
Chris Robinson 8ca849655f Properly replace sinc4 with cubic in alsoft-config 2019-04-28 21:51:47 -07:00
Chris Robinson ace7481b8e Only enable NFC with HRTF when hq-mode is enabled 2019-04-28 21:49:10 -07:00
Chris Robinson 9d6619efdc Combine some reverb processing loops
Specifically, the A2B and output mixing, as well as applying the band-pass with
B2A mixing (the latter of which hiding a bug that was overwriting the early
buffer storage).
2019-04-28 08:56:41 -07:00
Chris Robinson ec869234f7 Remove restrict from in+out parameters 2019-04-28 08:51:50 -07:00
Chris Robinson 014936ceff Be more robust with PulseAudio capture
Particularly, handle "holes" in the record stream and premature end-of-buffer.
Also don't bail out when capturing while disconnected (the extension says it
should provide anything it previously reported available, going to silence for
anything no longer readable).
2019-04-27 21:25:42 -07:00
Chris Robinson 0577028b65 Add missing function pointers 2019-04-27 18:01:26 -07:00
Chris Robinson c135629eae Remove some ancient PulseAudio KDE/Phonon/Qt hacks
Unfortuantely, the relevant KDE bug still seems to exist (streams are forced to
KDE's default device after opening, even when they're created with a specific
device at user request). I do not know why KDE thinks this is in any way good
behavior (the user doesn't get their desired device, nor does the stream get
the appropriate format for the device its ultimately put on), but making
streams non-movable as a workaround has been a thorn in the side of non-KDE
users for too long. C'mon KDE, it's been nearly (if not more than) 7 years now.
2019-04-27 14:13:07 -07:00
Chris Robinson 150dc92f68 Remove unused pa_threaded* functions 2019-04-27 13:06:39 -07:00
Chris Robinson f9f34def82 Use a custom PulseAudio mainloop
This allows using RT priority again with the mixer. It also consolidates all
mainloop instances into one.
2019-04-27 12:25:49 -07:00
Chris Robinson c724798c85 Don't round WASAPI updates to the update size 2019-04-27 08:48:14 -07:00
Chris Robinson dd2c43be42 Update the period size placeholder text 2019-04-26 19:02:18 -07:00
Chris Robinson b502bbaf5c Change the default period size to 20ms 2019-04-26 18:56:54 -07:00
Chris Robinson 9e4ee500b6 Scale the update size with sample rate changes 2019-04-26 18:04:22 -07:00
Chris Robinson b29be3b39e Set the ALSA period size first 2019-04-26 16:20:27 -07:00
Chris Robinson f23ff0394d Specify the buffer size as itself instead of the period count
Certain backends don't need a buffer size to be a strict multiple of the period
count, which allows a little more flexibility. The period/update size simply
acts as the minimum request, which helps control CPU load by determining how
often parameter and other pre-mixing updates are processed.
2019-04-26 15:58:25 -07:00
Chris Robinson 348e01dc4b Work around a 32-bit GCC compiler bug 2019-04-26 07:11:09 -07:00
Chris Robinson 725ceb128e Open the playback device earlier
So we actually have a device name to get the initial settings for. Be aware
that some backends set a format when opening instead of on reset, so such
devices will only set the default format (it can't get a desired format without
the device name, but the format will already be set once that's known). The
affected backends are WinMM, SDL2, and PortAudio (none of which are generally
used). This could be fixed by reopening the device during reset, but it would
need to be done carefully.
2019-04-20 19:29:33 -07:00
Chris Robinson 87e279082f Don't round the PulseAudio write size to the period multiple 2019-04-17 05:51:40 -07:00
Chris Robinson 09dbc4c2da Fix PulseAudio backend initialization 2019-04-16 17:23:28 -07:00
Chris Robinson 61f7e7716c Remove the backend factory deinit method
It was never actually called anywhere, and there's no safe place where it can
be called. It's probably better to let the individual backends worry about
cleaning themselves up anyway.
2019-04-14 04:05:07 -07:00
Chris Robinson 7f52678099 Use a custom message queue for the WASAPI backend 2019-04-14 01:03:36 -07:00
Chris Robinson 629cfa04a3 Fix some integer truncation warnings in MSVC 2019-04-12 19:19:24 -07:00
Chris Robinson 6761fe137f Load buffer data using the current and end pointer 2019-04-12 16:24:11 -07:00
Chris Robinson 4b95d310ae Remove the SZFMT macro
C++11 mandates the %zu/d formatter
2019-04-11 16:01:11 -07:00
Chris Robinson 8215251571 Simplify sorting and initializing the backends 2019-04-11 02:49:13 -07:00
Chris Robinson abcca1acd1 Add exception protection to the last of the API functions 2019-04-10 17:47:13 -07:00
Chris Robinson 1d214dba1e Add exception protection to effect, filter, and buffer functions 2019-04-10 17:33:21 -07:00
Chris Robinson b4a8615ea9 Add exception protection to the listener functions 2019-04-10 14:16:15 -07:00
Chris Robinson 5ad50e4f8e Add exception protection to context state functions 2019-04-10 13:44:46 -07:00
Chris Robinson f39d4598b7 Add exception protection to some AL functions 2019-04-10 13:05:21 -07:00
Chris Robinson d6f72b777a Fix usage of the voice's HRTF accumulation buffer when blending 2019-04-09 22:42:45 -07:00
Chris Robinson 55a3f38405 Don't use a fixed size buffer for printed messages 2019-04-09 22:13:21 -07:00
Chris Robinson 460a01443c Add macros to stop exceptions from leaving API functions
Effectively makes the functions act as noexcept, since there's no meaningful
reason to propogate exceptions from "C" functions. Currently only applied to
ALC functions, but can incrementally be applied to AL functions too. In the
future, this could also handle ALC and AL errors with unique exception types
(functions that utilize this behavior would need to ensure proper cleanup).
2019-04-09 20:48:01 -07:00
Chris Robinson e6daab51e8 Make sure the all-pass filter is cleared before use 2019-04-04 22:57:29 -07:00
Chris Robinson dc5759f80c Fix MSVC error about parenthesized explicit type conversion 2019-04-03 10:32:48 -07:00
Chris Robinson e66632b301 Avoid directly using CalcAmbiCoeffs 2019-04-02 20:23:19 -07:00
Chris Robinson d04f9326c9 Change RealMixParams::ChannelName to better reflect its use 2019-04-02 16:06:45 -07:00
Chris Robinson 9ac3524521 Spread out the ALC extensions like the AL extensions 2019-04-01 21:41:11 -07:00
Chris Robinson 86683264c9 Avoid multiple using statements for the same things 2019-04-01 21:35:52 -07:00
Chris Robinson 8781a32df5 Copy the voice's ambisonic upsampler state when reallocating 2019-04-01 17:27:31 -07:00
Chris Robinson da8494dd8a Use the existing function to get the ambisonic channel count 2019-03-31 22:50:30 -07:00
Chris Robinson 0e0e85af72 Don't bother with a separate reset method for BFormatDec 2019-03-31 22:27:56 -07:00
Chris Robinson 423333c594 Fix use of reverse_copy 2019-03-31 19:09:21 -07:00
Chris Robinson c88b09170c Use phase correction for the front stablizer 2019-03-31 18:54:43 -07:00
Chris Robinson d8c76ba0c0 Remove a couple unused member variables 2019-03-30 23:41:47 -07:00
Chris Robinson 89210cddb7 Don't stop the backend if it's not running 2019-03-30 23:30:15 -07:00
Chris Robinson cc91490b61 Use a temporary buffer for HRTF filter accumulation
Similar to the history buffer, to avoid using the state buffer as a ring
buffer.
2019-03-29 11:33:04 -07:00
Chris Robinson fe7918465e Use a function reference for a template parameter 2019-03-28 10:00:35 -07:00
Chris Robinson dfb81ff42d Avoid using the HRTF history buffer as a ring buffer
The HRTF mixers now get a full input buffer with the history prepended, so the
delay offsets just need to account for the start point and read forward for
each sample.
2019-03-28 09:34:31 -07:00
Chris Robinson e7bfe1ebd0 Simplify applying the field magnitude factor 2019-03-27 11:52:35 -07:00
Chris Robinson 095a0b9beb Set the EARLY_REQUESTS flag for PulseAudio when possible 2019-03-26 17:03:31 -07:00
Chris Robinson 608e4e916e Avoid recreating temporary buffers all the time 2019-03-26 17:01:45 -07:00
Chris Robinson 2960d729ef Fix a return type for accumulation 2019-03-26 13:47:24 -07:00
Chris Robinson 8b0c9fcf30 Require IR dimensions from a SOFA file 2019-03-26 13:45:25 -07:00
Chris Robinson 2446ee0cca Avoid some explicit loops 2019-03-26 12:37:52 -07:00
Chris Robinson a5b442b1a4 Add a missing include for varargs 2019-03-26 11:37:28 -07:00
Chris Robinson bcbe7c493c Remove another unnecessary distance check 2019-03-26 11:04:52 -07:00
Chris Robinson 525b5e65b2 Remove a couple unnecessary duration_casts 2019-03-26 10:20:32 -07:00
Chris Robinson 2c37d4fae1 Move an enum to a more appropriate header 2019-03-26 10:08:26 -07:00
Chris Robinson 4c32e55ac2 Don't mention which rendering methods aren't used 2019-03-25 23:17:34 -07:00
Chris Robinson a57116f788 Scale floats directly to 32-bit integer values
Rather than scaling to a 25-bit integer and shifting for the extra 7 bits. This
should improve precision for values closer to 0.
2019-03-25 22:52:15 -07:00
Chris Robinson 0a4d1c858e Support loading SOFA files directly with makemhr
This extracts the definition info it can from the SOFA, and uses the same logic
as sofa-info to automatically detect an appropriate layout. There is a bit of
code duplication from loaddef.cpp and sofa-info.cpp, though there are slight
modifications.
2019-03-25 20:16:02 -07:00
Chris Robinson cb02bb00be Detect SOFA files for makemhr input 2019-03-25 13:01:44 -07:00
Chris Robinson a3687db015 Use only a single function to load the HrirDataT 2019-03-25 00:21:45 -07:00
Chris Robinson 78aeca10c9 Add a command-line option to force mono input with makemhr 2019-03-24 22:43:43 -07:00
Chris Robinson 6edfa6e6e0 Cleanup some unnecessary parameters 2019-03-24 22:06:01 -07:00
Chris Robinson f392d9c138 Move makemhr's .def loading code to a separate source 2019-03-24 19:00:58 -07:00
Chris Robinson 5e6e738681 Change references of makehrtf to makemhr 2019-03-24 17:35:32 -07:00
Chris Robinson 7c16b1e02f Rename makehrtf to makemhr and move it to a subdirectory 2019-03-24 17:31:10 -07:00
Chris Robinson c8e8ac42ab Use false instead of 0 for a boolean 2019-03-24 14:02:06 -07:00
Chris Robinson f7ab7b45f7 Mark the device and context deletes as unlikely 2019-03-24 13:54:49 -07:00
Chris Robinson 12d4953a5f Remove a couple redundant distance checks 2019-03-23 13:55:07 -07:00
Chris Robinson 6df673e01c Don't redundantly set the voice's direct output buffer 2019-03-23 13:30:15 -07:00
Chris Robinson 819ec8a653 Rename and move ALeffectProps 2019-03-22 22:48:12 -07:00
Chris Robinson 3a6e741e90 Use MixParams for the wet buffers 2019-03-22 19:25:55 -07:00
Chris Robinson 92adfaebce Fix a comment typo 2019-03-22 18:37:47 -07:00
Chris Robinson edc32b40a5 Slightly improve casting in the modulator effect 2019-03-22 17:47:19 -07:00
Chris Robinson ea8b02dead Pass ALeffectProps directly to the get/setParam* methods 2019-03-22 16:04:13 -07:00
Chris Robinson e7e585f65c Use the effect state factory to set the default effect props 2019-03-22 15:00:37 -07:00
Chris Robinson f951f4a66b Implement getDefaultProps for effect state factories 2019-03-22 12:58:24 -07:00
Chris Robinson 935f386982 Use a separate EffectStateFactory for standard reverb 2019-03-22 11:57:32 -07:00
Chris Robinson 9790135127 Add a new EffectStateFactory method to get the default properties 2019-03-22 11:52:55 -07:00
Chris Robinson 8a1b1f4204 Make sure the file is rewound before writing the wave header 2019-03-19 22:10:49 -07:00
Chris Robinson 7880540407 Use a sorted vector for looking up contexts
Note that the device still holds and uses a linked list of its contexts. The
sorted vector is used to verify handles given by callers.
2019-03-19 18:53:32 -07:00
Chris Robinson 90465e9124 Use a sorted vector for devices instead of a linked list 2019-03-19 15:56:30 -07:00
Chris Robinson cf6545ebb2 Avoid AL types in the common alnumeric.h header 2019-03-19 14:34:44 -07:00
Chris Robinson 813976e58d Add some enum casts for the WASAPI backend 2019-03-19 01:33:36 -07:00
Chris Robinson 9695952c8d Rename DevProbe enum names 2019-03-19 00:24:54 -07:00
Chris Robinson d31514f8be Move some inline functions from alMain.h to alnumeric.h 2019-03-18 22:06:01 -07:00
Chris Robinson 3e816de4fb Use SSE intrinsics in a few more places 2019-03-18 20:58:48 -07:00
Chris Robinson 73a43fb19c Don't bother trying _controlfp or __control87_2 2019-03-18 20:27:25 -07:00
Chris Robinson 2de1d17bc6 Use SSE intrinsics to set FTZ and DAZ 2019-03-18 20:14:40 -07:00
Chris Robinson 6a0b2ed0ca Check compile-time support for SSE intrinsics 2019-03-18 20:05:15 -07:00
Chris Robinson 821c7565cf Don't bother checking for SSE1 alone
SSE2 support is now the minimum required for SSE. Run-time can still disable
SSE2-specific functions separately from SSE1, but build-time support can't be
separated.
2019-03-18 17:38:02 -07:00
Chris Robinson e61cec8f17 Don't bother checking for C99 inline semantics
Should be unneeded with C++
2019-03-18 14:31:12 -07:00
Chris Robinson d802a5785e Fix indexing for basic B-Format decoding 2019-03-17 14:03:21 -07:00
Chris Robinson f96c37120b Always reset all voices on disconnect 2019-03-16 17:01:04 -07:00
Chris Robinson 339a37b034 Include std::placeholders once in an anonymouse namespace 2019-03-16 15:15:59 -07:00
Chris Robinson 12999f9efe Simplify calculating azimuth and elevation indices for HRTF 2019-03-14 22:26:19 -07:00
Chris Robinson 9f5a7a7a50 Fix wrapping for the upper HRIR index 2019-03-14 13:17:07 -07:00
Chris Robinson 3a19b94503 Mirror a couple HRIR elevations from the top for the bottom
Because the ears are offset from center, linear interpolation from the lowest
defined elevation to the -90 degree bottom misses this slight deviation.
Mirroring one or two more elevations from the top helps catch it, and bilinear
interpolation is used to transition back to the lowest known measurements.
2019-03-13 12:27:44 -07:00
Chris Robinson 3a39a2c790 Set the correct target gain after a fade-out HRTF mix 2019-03-12 16:45:34 -07:00
Chris Robinson 24ae12f58e Check a value where its used 2019-03-12 09:56:02 -07:00
Chris Robinson 098c7ca6ea Fix not looping when the source offset is beyond the loop end 2019-03-11 22:09:30 -07:00
Chris Robinson 24b43fe852 Remove a couple redundant local variables 2019-03-11 22:00:28 -07:00
Chris Robinson 30a7c6d86f Pass the voice state as a parameter instead of reloading it 2019-03-11 20:00:14 -07:00
Chris Robinson e0daad6a16 Update comment for SynthesizeOnsets 2019-03-11 14:28:13 -07:00
Chris Robinson d2b4099024 Process minimum phase reconstruction in parallel 2019-03-11 13:41:26 -07:00
Chris Robinson 0689333da8 Mirror the +90 degree elevation delays for -90 degrees 2019-03-10 21:47:05 -07:00
Chris Robinson 2c67ab0d2c Rename ALvoice fields for consistency 2019-03-10 16:29:06 -07:00
Chris Robinson 030b24a40d Improve handling of voice's AmbiScales for upsampling 2019-03-10 15:46:46 -07:00
Chris Robinson 663a6ce4e7 Use the correct value for MAX_AMBI2D_CHANNELS 2019-03-10 15:39:31 -07:00
Chris Robinson d99d2a400a Don't copy old coeffs in MixHrtfBlendBase 2019-03-10 14:31:46 -07:00
Chris Robinson 80e55b87ed Avoid an extra level of indentation 2019-03-10 12:50:38 -07:00
Chris Robinson 0d295cf811 Don't directly use a buffer for updating source parameters 2019-03-10 11:52:39 -07:00
Chris Robinson e5651c15dd Avoid excessive transformations of the source position 2019-03-10 11:33:08 -07:00
Chris Robinson 12721f94a7 Add a method to apply an HF scale without band-splitting 2019-03-10 10:30:33 -07:00
Chris Robinson 6a95189ef6 Fix for MSVC decaying arrays to pointers with ?: 2019-03-09 21:54:15 -08:00
Chris Robinson 106671d451 Fade out voices that end normally
Sometimes a sound may end with non-0 amplitude, particularly if a buffer queue
underruns. This helps avoid clicks and pops for sources that don't already end
in silence.
2019-03-09 21:32:14 -08:00
Chris Robinson 87479b4b32 Play dummy samples and force a fade out on stopping voices 2019-03-09 18:34:00 -08:00
Chris Robinson ef0f335132 Add a Stopping state for voices
This currently doesn't do much, except have the mixer progress it to Stopped.
It's valid to have without a source or buffers, and in the future will allow
fading out when a source is paused or stopped.
2019-03-09 16:48:07 -08:00
Chris Robinson cde4500e24 Increment the active voice count ahead of playing the sources 2019-03-09 15:04:51 -08:00
Chris Robinson d39cfcc7bb Clear the voice's buffer when detaching from source 2019-03-09 13:20:10 -08:00
Chris Robinson 972a5c0f1d Exclude far-ear IRs for the synthesized -90 degree elevation 2019-03-09 13:00:08 -08:00
Chris Robinson 0e4402b2e1 Avoid some explicit loops 2019-03-05 06:21:09 -08:00
Chris Robinson da9ec374d8 Normalize HRIRs using their maximum RMS 2019-03-04 08:00:04 -08:00
Chris Robinson 90d1e1212b Pass a reference to function for a template parameter 2019-03-03 00:24:33 -08:00
Chris Robinson 2d14de3fb0 Use more specific names for temp buffer storage 2019-03-02 23:41:26 -08:00
Chris Robinson 0aa0f24dd7 Use a proper flag to indicate audio is prepared in alffplay 2019-03-01 22:46:56 -08:00
Chris Robinson e4b76d2627 Reverse the HRTF field array
Most often a sound's distance will be beyond the farthest field measurement, so
It's more efficient to have the farthest field first and avoid looping through
the whole field array for them.
2019-02-27 23:13:40 -08:00
Chris Robinson 86926f0998 Combine the reverb output mixes into a single call 2019-02-25 05:52:37 -08:00
Chris Robinson 8a34bd59b0 Unlock the audio decoder mutex before disabling events in alffplay
The callback may be waiting on the mutex, but disabling the callback needs any
current invocation to finish first.
2019-02-25 02:03:43 -08:00
Chris Robinson 81aa5af3b9 Make sure the reverb fading completes
The processing loop doesn't depend on being aligned anymore, so it won't get
stuck when only less than 4 samples can be done in a non-final update.
2019-02-25 01:08:11 -08:00
Chris Robinson 4d6b66163c Convert the device frequency to float just once 2019-02-24 19:07:21 -08:00
Chris Robinson d6fb4d5f52 Make sure the voice's direct buffer is always set 2019-02-24 19:02:49 -08:00
Chris Robinson d2664e2192 Remove a duplicate index array 2019-02-24 18:00:49 -08:00
Chris Robinson cadff0f6c1 Reduce BUFFERSIZE to match the default period size
Also adds a bit more space to the temp source data buffer, to avoid needing to
loop on matching sample rates.
2019-02-24 16:13:51 -08:00
Chris Robinson a2ba550ebf Rework reverb A/B-Format conversion mixing
This should help improve performance using the optimized mixers, and fewer
passes on the transforms, though at the cost of more memory.
2019-02-24 15:43:56 -08:00
Chris Robinson 4ec0aed286 Change some functions to proper methods 2019-02-23 03:18:12 -08:00
Chris Robinson 42934b09e5 Avoid multiple int-to-float conversions 2019-02-23 01:43:55 -08:00
Chris Robinson 3683e31662 Constify some parameters and remove an explicit loop 2019-02-23 00:01:38 -08:00
Chris Robinson 317206e8f3 Remove the FOAOut mixing buffer and associated post-processes 2019-02-22 22:35:37 -08:00
Chris Robinson 45378fe687 Remove the unused FOAOut EffectTarget 2019-02-22 22:03:04 -08:00
Chris Robinson 5b35e60a98 Apply ambisonic upsampling on reverb output as needed
This isn't the greatest thing since it splits the A-to-B-Format transform from
the panning transform. The A-to-B and HF scale mixes are also not as optimal as
they could be, since they can't use the main mixer functions (wrong buffer line
length).

It does, however, get rid of the final use of the FOAOut buffer, so the
upsampling post-process is no longer needed.
2019-02-22 21:53:45 -08:00
Chris Robinson 6a3c10d850 Ensure reverb fading doesn't end with less than 4 samples 2019-02-22 02:15:22 -08:00
Chris Robinson 6a2cf96448 Avoid some unnecessary local variables 2019-02-22 01:00:57 -08:00
Chris Robinson 01ac6e2e56 Combine reverb transform matrices one column at a time 2019-02-21 19:05:20 -08:00
Chris Robinson 7a9d67934f Mix B-Format sources directly to the dry buffer
Now the only thing that utilizes FOAOut is reverb output.
2019-02-21 17:48:08 -08:00
Chris Robinson 1eea3cb2d1 Remove RESTRICT from the bandsplitter process method
The compiler can see there's no aliasing with the local variables, and the
input/output buffers are handled sequentially one element at a time anyway.
2019-02-21 17:37:02 -08:00
Chris Robinson 169a11dffc Add some preliminary fields for mix-time ambisonic upsampling 2019-02-21 04:57:56 -08:00
Chris Robinson 8ac2d34706 Allow processing some effects in higher order ambisonics
Reverb notably is still only first-order (any higher order channels are
dropped, and it writes to FOAOut). But others, like the equalizer, work on all
available channels.
2019-02-21 04:23:01 -08:00
Chris Robinson a964890537 Add helpers to get the channel count from an ambisonic order 2019-02-21 04:05:49 -08:00
Chris Robinson a35255291f Fix unused parameter warning 2019-02-21 03:54:12 -08:00
Chris Robinson 8d2d7c63da Get rid of the MAX_EFFECT_CHANNELS macro 2019-02-21 03:52:54 -08:00
Chris Robinson 462e320847 Make sure the B2A matrix has enough values for the input count 2019-02-21 03:31:24 -08:00
Chris Robinson d95e144c48 Remove some now-unnecessary ReverbState fields 2019-02-21 03:07:36 -08:00
Chris Robinson 4b4041319d Pass the number of input channels to EffectState::process 2019-02-21 02:57:39 -08:00
Chris Robinson 7e00f646d9 Add a method to get the row of an ambisonic identity transform 2019-02-21 02:13:30 -08:00
Chris Robinson 615446d6d5 Ensure the device's mAmbiOrder is always set appropriately
The Dry target is always ambisonic, so set its order correctly.
2019-02-21 01:23:11 -08:00
Chris Robinson 3966665ca3 Store effect slots in groups of 64
Now that their wet buffers are allocated dynamically, the ALeffectslot object
itself is rather small.
2019-02-20 22:00:26 -08:00
Chris Robinson c43381d811 Allocate the effect slot wet buffer dynamically 2019-02-20 21:01:08 -08:00
Chris Robinson 77ba61ecb3 Partially handle non-periphonic reverb input 2019-02-19 22:55:43 -08:00
Chris Robinson 87902fa3b2 Use the right macro for the number of reverb panning gains 2019-02-19 20:51:10 -08:00
Chris Robinson 194fcb6bb7 Rename MAX_AMBI_COEFFS and MAX_AMBI2D_COEFFS 2019-02-19 15:39:33 -08:00
Chris Robinson a75bc26173 Reduce some indenting 2019-02-19 02:30:01 -08:00
Chris Robinson 2c04095d22 Apply the upsampler's all-pass when mixing the band-split samples 2019-02-19 02:05:35 -08:00
Chris Robinson d713f5f828 Clean up some AmbiUpsampler initialization 2019-02-19 01:59:30 -08:00
Chris Robinson 69ab36cc14 Avoid a temp buffer for the distance compensation delay 2019-02-17 00:43:10 -08:00
Chris Robinson c95d6a5d8c Apply phase correction to the ambisonic decoder HRIRs
This preserves the original phase of the HRIR frequencies for decoding the
ambisonic signal. This should help prevent extra coloration from the band-
splitter used to scale the HF response.
2019-02-16 22:57:38 -08:00
Chris Robinson 69f6f56160 Avoid using internal AL[u]int64 types 2019-02-11 12:16:58 -08:00
Chris Robinson 2fc8461c14 Don't check for __int64 2019-02-11 11:44:35 -08:00
Chris Robinson dac609b29a Move some more functions to alnumeric.h 2019-02-11 11:31:31 -08:00
Chris Robinson 21aaa18c50 Get rid of the FAM_SIZE macro 2019-02-11 11:14:34 -08:00
Chris Robinson 995c9649cb Move some number-related stuff to a separate header 2019-02-11 11:07:06 -08:00
Chris Robinson 69d8c6546d Use std::arrays for HRIR coeffs and values 2019-02-07 08:38:49 -08:00
Chris Robinson b371862fb2 Simply setting a couple indices 2019-02-04 22:17:28 -08:00
Chris Robinson 05cdc2cb30 Use relaxed memory ordering for initializing atomic_flags 2019-02-04 21:28:37 -08:00
Chris Robinson e104f580b3 Use a unique_ptr to store the loaded HrtfEntry 2019-02-04 21:22:43 -08:00
Chris Robinson b7b4cfae2b Fixed alffplay underrun recovery timing 2019-02-04 21:22:43 -08:00
kcat 367191cbfb Merge pull request #269 from gongminmin/FixForVS2019
Fix compiling problems on VS2019 with vc142 toolset
2019-02-04 21:22:16 -08:00
Minmin Gong 41b9d473a2 Fix compiling problems on VS2019 with vc142 toolset
Msvc142 in VS2019 preview 2 doesn't allow std::atomic_flag to be initialized by a bool. Call test_and_set in the constructors instead.
2019-02-04 20:03:18 -08:00
Chris Robinson c25433986a Add missing include 2019-01-29 14:14:52 -08:00
Chris Robinson 613145e900 Calculate the correct evidx for the HRTF B-Format decoder 2019-01-29 03:52:28 -08:00
Chris Robinson 44d72942fb Load and use all available HRTF fields 2019-01-29 00:42:18 -08:00
Chris Robinson 4c740636c2 Pass the distance to GetHrtfCoeffs 2019-01-28 22:22:34 -08:00
Chris Robinson 2d93a9cdb1 Collect HRTF field info into a struct 2019-01-28 20:43:34 -08:00
Chris Robinson df5d238e3d Clean up calculating HRIR indices to blend 2019-01-28 19:39:45 -08:00
Chris Robinson b3fe540c49 Don't normalize HRIRs per-field 2019-01-27 15:50:02 -08:00
Chris Robinson e55f9b42e9 Partially handle ambisonics in alffplay
This is currently really only applicable to Opus-encoded files. It assumes
AmbiX (SN3D normalization, ACN ordering) and only comes into play when the
channel layout is blank. FFmpeg/libavcodec doesn't have a way to detect
B-Format input or what normalization and ordering it uses. Note in particular
.amb files do not play correctly (libavcodec seems to apply a default channel
layout for 4-channel wav-type files, regardless of its channel mask value).
2019-01-27 14:53:46 -08:00
Chris Robinson d7af17ab87 Round the calculated field distance in makehrtf 2019-01-27 14:35:58 -08:00
Chris Robinson f98a24fb8a Don't try to clean up the async ringbuffer if it isn't allocated 2019-01-26 19:29:26 -08:00
Chris Robinson f5e17bafea Fix applied scalars for B-Format source mixing 2019-01-26 17:33:57 -08:00
Chris Robinson 8eab75f312 Update a function comment 2019-01-24 17:05:13 -08:00
Chris Robinson b5569d8455 Slight cleanup for NfcFilterCreate4 2019-01-24 10:22:38 -08:00
Chris Robinson 98be1d1bf5 Make IncRef and DecRef member functions 2019-01-24 10:05:37 -08:00
Chris Robinson 7757789590 Clean up the NFC filters a bit 2019-01-23 16:32:53 -08:00
Chris Robinson ebb46cf4cf Rename a header
To workaround an apparent MSVC error
2019-01-23 15:09:11 -08:00
Chris Robinson 1ec8686407 Add missing includes 2019-01-23 13:46:43 -08:00
Chris Robinson ce3acf4d1b Make hrtf_inc.cpp a proper header 2019-01-23 12:33:18 -08:00
Chris Robinson 8429770c3f Use template declarations for the HRTF mixers 2019-01-23 12:23:05 -08:00
Chris Robinson d3842a632d Use template declarations for the normal mixing functions 2019-01-23 11:21:03 -08:00
Chris Robinson 23179d0bcd Use a template declaration for the resampler functions 2019-01-23 11:11:41 -08:00
Chris Robinson e332ac8528 Use the farthest of multi-field HRTFs 2019-01-22 21:36:40 -08:00
Chris Robinson 3ab4bc5186 Don't install utilities that aren't built 2019-01-22 17:34:45 -08:00
Chris Robinson 5c0dcd1f24 Fix building makehrtf and sofa-info on Windows 2019-01-22 13:17:21 -08:00
Chris Robinson ecafa19bcf Support loading sofa files in makehrtf
The makehrtf utility now requires libmysofa to build. This isn't necessay for
the OpenAL Soft library itself.
2019-01-22 11:24:57 -08:00
Chris Robinson b4a2532ffc Partially allow loading of multi-field HRTFs
Only the first field actually gets used, but all the data is processed when
loading.
2019-01-22 10:27:04 -08:00
Chris Robinson 1565d7e276 Add a sofa-info utility to check sofa files 2019-01-21 10:23:17 -08:00
Chris Robinson c239ae44f8 Add a FindMySOFA cmake module 2019-01-21 10:21:13 -08:00
Chris Robinson d1a8607515 Handle a missing default WASAPI device ID 2019-01-18 17:23:46 -08:00
Chris Robinson 40c6f74bb7 Add front and back methods to FlexArray 2019-01-17 02:23:57 -08:00
Chris Robinson 101be788e1 Replace a couple more C math calls 2019-01-14 22:30:18 -08:00
Chris Robinson 867161d55f Constify some parameters 2019-01-12 21:08:34 -08:00
Chris Robinson 5b382a69b6 Avoid implicit conversions from signed to unsigned 2019-01-12 18:08:43 -08:00
Chris Robinson d64eaba322 Use a flexible array for DirectHrtfState and ALvoice 2019-01-12 01:25:33 -08:00
Chris Robinson ab16671466 Use a flexible array for HrtfHandle and SampleConverter 2019-01-11 22:09:57 -08:00
Chris Robinson f8c2e54b47 Make the min/max/clamp functions constexpr 2019-01-11 21:04:50 -08:00
Chris Robinson 77447fcd54 Add placement operator delete 2019-01-11 20:06:23 -08:00
Chris Robinson 852c5005ab Add a missing include 2019-01-11 08:07:25 -08:00
Chris Robinson 81e7222633 Use a flexible array for the active effect slots 2019-01-11 07:28:44 -08:00
Chris Robinson 8aedaea5fb Add a flexible array template container 2019-01-11 06:00:41 -08:00
Chris Robinson e7d77f5caa Use a vector for ALeffectslotArray 2019-01-09 18:29:22 -08:00
kcat 30184613a5 Merge pull request #264 from ShFil119/impr/cleanup
Cleanup continuation
2019-01-09 17:16:28 -08:00
Filip Gawin f7fe15e1ce Use = default to define trivial dtor/ctor 2019-01-09 19:57:35 +01:00
Filip Gawin 6ddb2c36fc Remove redundant void argument list in function def 2019-01-09 19:43:54 +01:00
Filip Gawin 4169c6f37d Use c++ headers 2019-01-09 19:42:40 +01:00
Chris Robinson 8f35f464a1 Change a true/false ALenum atomic into a bool 2019-01-09 01:06:19 -08:00
Chris Robinson b49c45d3a4 Add prefixes for ALCdevice and ALCcontext 2019-01-09 00:31:57 -08:00
Chris Robinson 673983dc5d Make Create methods for structs with flexible array members ...
... that are used with unique_ptr.
2019-01-08 23:44:08 -08:00
Chris Robinson bc1eeb5df0 Avoid calling unique_ptr::reset with a raw pointer 2019-01-08 23:15:58 -08:00
Chris Robinson 173c97c2db Remove an unused macro 2019-01-08 21:51:56 -08:00
Chris Robinson a46f1f810d Rename ALC_SOFT_loopback2 and renumber its enums 2019-01-08 20:02:36 -08:00
Chris Robinson d7eee03272 Replace a couple more C-style casts 2019-01-08 19:08:03 -08:00
Chris Robinson 0763dfa954 Apply the all-pass separately from the upsampling mix 2019-01-08 18:34:45 -08:00
Chris Robinson edba7da8ab Avoid macros for indexing HF and LF fequency bands 2019-01-08 17:21:22 -08:00
Chris Robinson 0c3662d8b2 Fix some comments and reduce indentation 2019-01-08 17:21:22 -08:00
kcat 009d738c1e Merge pull request #261 from ShFil119/avoid_old_style_casts
Avoid using old style casts
2019-01-08 17:19:15 -08:00
Filip Gawin 0d3a0635d9 Avoid using old style casts
To think about:
examples/alffplay.cpp:600
OpenAL32/Include/alMain.h:295
2019-01-08 19:42:44 +01:00
kcat 2a7f27ca58 Merge pull request #260 from ShFil119/impr/nullptr
Use nullptr in cpp files
2019-01-07 04:36:50 -08:00
Chris Robinson 4d047e2bc1 Use user-defined literals for 64-bit literals 2019-01-07 04:06:40 -08:00
Filip Gawin 0537414baf Use nullptr in cpp files 2019-01-07 12:37:13 +01:00
Chris Robinson 67c9cf8174 Define logging calls to be unlikely 2019-01-07 01:13:06 -08:00
Chris Robinson 648b76ed65 Move some macros to a common header 2019-01-07 01:12:09 -08:00
Chris Robinson 20a5306bdf Clean up some initializers and use of C methods 2019-01-06 21:16:08 -08:00
Chris Robinson fababe76c4 Don't use the dual-band upsampler for basic ambisonic decoding 2019-01-06 17:45:44 -08:00
Chris Robinson f5b6d761ff Apply an all-pass on the existing output when upsampling ambisonics 2019-01-06 06:50:58 -08:00
Chris Robinson 13056b45b0 Revert back to using a band-splitter to increase the HF response
Unfortunately the shelf filter causes issues due to the shelf gain magnitude
creating a varying phase offset. The splitter also creates phase offsets, but
it's consistent regardless of gain.
2019-01-06 05:15:11 -08:00
Chris Robinson d2e34e509b Make the band-splitter and splitter-allpass filters templated
With float and double explicit instantiations
2019-01-06 04:45:35 -08:00
Chris Robinson da3a916042 Replace macros with constexpr inline functions 2019-01-06 04:16:51 -08:00
Chris Robinson 98f3e6a162 Remove an unnecessary undef 2019-01-06 02:49:24 -08:00
Chris Robinson e645d25b62 Use double-precision biquads for the HRTF shelf filters 2019-01-06 00:54:39 -08:00
Chris Robinson 607e778344 Make BiquadFilter a templated class
With explicit instantiations for float and double
2019-01-06 00:53:02 -08:00
Chris Robinson 03aeb7edf6 Use a shelf filter for the HRTF B-Format decoder HF scale 2019-01-06 00:23:15 -08:00
Chris Robinson 3b91010e21 Pass the normalized crossover frequency to the reset method 2019-01-05 22:31:13 -08:00
Chris Robinson aa29bf5933 Remove unnecessary structs and buffers 2019-01-05 22:24:38 -08:00
Chris Robinson 3f35fcc4b5 Simplify MixParams and AmbiUpsampler
Since the dry buffer is always an ambisonic target now
2019-01-05 21:59:04 -08:00
Chris Robinson cff20c2fe8 Use BFormatDec for custom and built-in ambisonic decoding 2019-01-05 19:21:25 -08:00
Chris Robinson 641bbf075d Enable simplified upsampling for AmbiUpsampler 2019-01-05 00:23:06 -08:00
Chris Robinson 6d7e8cf8b0 Simplify the BFormatDec upsampler
Since the only difference applied is a scalar on the high-frequency response, a
B-Format-to-B-Format transcode can be accomplished with a high-shelf filter. A
similar thing can be done with AmbiUpsampler, but only when outputing to
periphonic B-Format.
2019-01-04 22:17:30 -08:00
Chris Robinson 0a6748156d Make sure the OpenSL capture buffer queue has the correct size
The ring buffer size may round up and have more queueable elements than OpenSL
was allocated with, leading to errors when queueing those extra elements. Now
OpenSL allocates the same number of elements that can be written to the ring
buffer.
2019-01-03 18:16:46 -08:00
Chris Robinson 21a17620e3 Fix the ringbuffer write limit
Previously it just limited the returned write space, irrespective of how much
had already been written. The buffer could still be filled up by doing multiple
writes. Now the size is limited by adjusting the read pointer by the real vs
limited difference when calculating the writable space.
2019-01-03 15:54:18 -08:00
Chris Robinson 88c2f11dcf Use DeviceRef and ContextRef when creating a new device and context 2019-01-01 22:44:46 -08:00
Chris Robinson 399dfca1e3 Rename ALCdevice_struct and ALCcontext_struct
A (possibly contentious?) change in the public headers. Those names were never
part of any specification, and I don't know why the struct names differed from
the actual type name. But with C++, which takes the original struct declaration
as the original name, it was affecting the type's internal symbols.

This shouldn't affect user code since ALCdevice_struct and ALCcontext_struct
were never part of the spec. If issues arise from this change, it should be
reported.
2019-01-01 18:13:33 -08:00
Chris Robinson 1630a33567 Use standard unique_lock and lock_guard for the backend lock 2019-01-01 16:42:54 -08:00
Chris Robinson 2f1566e0b4 Add and use a make_unique function 2019-01-01 14:33:01 -08:00
Chris Robinson c36798fd07 Avoid unnecessary extra buffers for filter chains 2019-01-01 02:41:27 -08:00
Chris Robinson e930c70eaa Don't make BiquadFilter's src and dst restrict
There's no technical reason they can't be the same since IIR filters can't be
auto-vectorized anyway.
2019-01-01 00:04:46 -08:00
Chris Robinson a9e71c2668 Handle all input channels in MixDirectHrtf 2018-12-31 23:33:04 -08:00
Chris Robinson 0452c9201d Add a missing include 2018-12-31 20:51:51 -08:00
Chris Robinson 87143b6645 Remove some C-isms from makehrtf
Remove unnecessary typedefs, use C++11 stamndard types and values, avoid
explicit allocations.
2018-12-31 19:54:34 -08:00
Chris Robinson 4c4572ae8a Compile makehrtf as C++ 2018-12-31 18:23:30 -08:00
Chris Robinson d8eecc89e0 Pass a reference to an array instead of a pointer 2018-12-31 18:07:00 -08:00
Chris Robinson 5a9a1c8d7d Further improve HRTF methods to avoid masking in the inner loops 2018-12-31 04:12:20 -08:00
Chris Robinson 5e03941701 Use an atomic bool on things that only take true or false 2018-12-30 21:58:14 -08:00
Chris Robinson 9f5c9a2260 Rename BackendLock to StateLock 2018-12-30 21:38:42 -08:00
Chris Robinson 3df1d185f8 Don't make the backend's lock/unlock methods noexcept 2018-12-29 17:29:52 -08:00
Chris Robinson 24e763f2a1 Get rid of ALCdevice_Lock/Unlock 2018-12-29 14:00:34 -08:00
Chris Robinson bdf7c16cfb Clean up a member name 2018-12-29 13:21:47 -08:00
Chris Robinson 8a84e7b662 Use member functions for the sample and channel converters 2018-12-29 12:26:45 -08:00
Chris Robinson 9b0b722d72 Add missing includes for mem_fn 2018-12-29 03:11:06 -08:00
Chris Robinson 63fc74beaa Don't bother with an explicit Loopback backend type 2018-12-29 02:21:53 -08:00
Chris Robinson 71a4d6db6f Return a unique_ptr for the backend 2018-12-29 02:16:16 -08:00
Chris Robinson 3c637d5fd7 Make the backend type an enum class 2018-12-29 01:38:26 -08:00
Chris Robinson 8fd4477215 Get CoreAudio capture samples in one call 2018-12-28 23:19:49 -08:00
Chris Robinson 3d92e8c4df Convert the backends to use proper inheritence 2018-12-28 22:56:20 -08:00
Chris Robinson 67b874328d Use static_cast instead of reinterpret_cast where possible 2018-12-28 16:13:28 -08:00
Chris Robinson 01ed98c99b Finish turning ancillary backend methods into member functions 2018-12-28 15:09:51 -08:00
Chris Robinson b7f5166d59 Turn even more methods into member functions 2018-12-28 14:06:15 -08:00
Chris Robinson 200e267b81 Turn some more methods into member functions 2018-12-28 12:58:01 -08:00
Chris Robinson 983904bbdc Add a method to prefix logged function names 2018-12-28 12:29:05 -08:00
Chris Robinson aff58265cb Make more methods into member functions 2018-12-27 23:37:24 -08:00
Chris Robinson 28308226e7 Turn more methods into member functions 2018-12-27 21:50:54 -08:00
Chris Robinson 885f68268f Turn more methods into member functions 2018-12-27 20:24:35 -08:00
Chris Robinson 015a4b060b Make some ancillary methods into member functions 2018-12-27 19:38:02 -08:00
Chris Robinson 0f36459028 Use vectors instead of malloc'd buffers 2018-12-27 18:33:49 -08:00
Chris Robinson 465ab11748 Finish renaming backend struct fields 2018-12-27 17:48:02 -08:00
Chris Robinson e48b8c4cda Rename some more struct members for consistency 2018-12-27 17:09:14 -08:00
Chris Robinson 515edc3dee Fix ring buffer vector methods 2018-12-27 15:05:12 -08:00
Chris Robinson f2c2b7c538 Get rid of the unnecessary STATIC_(UP)CAST macros 2018-12-27 14:27:35 -08:00
Chris Robinson 4782d6107d Use a proper constructor/destructor for the ALCbackend base 2018-12-27 13:40:43 -08:00
Chris Robinson 7880f27054 Rename DSound class members for consistency 2018-12-27 13:07:14 -08:00
Chris Robinson 1a4387d137 Return unique_ptrs instead of raw pointers
For the ring buffer, channel converter, and sample converter.
2018-12-27 12:55:43 -08:00
Chris Robinson 323cf58f02 Simplify resampling with CoreAudio capture
The ringbuffer holds the samples from the device, and we use our own converter
for resampling, calling it on demand with data from the ring buffer.
2018-12-27 12:04:18 -08:00
Chris Robinson 4dca2f2ee5 Use a unique_ptr for the AsyncEvents ringbuffer 2018-12-27 10:44:02 -08:00
Chris Robinson be85ab6f82 Rename some ALSA class members for consistency 2018-12-27 10:34:22 -08:00
Chris Robinson c3f370fa41 Constify and reorder a couple device fields 2018-12-27 10:25:09 -08:00
Chris Robinson 7d821551ac Recognize ambix as an alias for acn+sn3d 2018-12-27 01:18:10 -08:00
Chris Robinson d367093c06 Use std::array for appropriate source and listener properties 2018-12-26 22:27:34 -08:00
Chris Robinson a4ac43b602 Avoid a lambda to find a not-null entry in an array 2018-12-26 21:55:39 -08:00
Chris Robinson 8a0295503d Clean up the ring buffer struct and use member functions 2018-12-26 21:22:17 -08:00
Chris Robinson 4f253a935a Handle HRTF coefficients and values by reference where possible 2018-12-26 15:35:05 -08:00
Chris Robinson c5be03b51e Avoid masking in ApplyCoeffs's inner loop
This unfortunately does not apply to NEON, which would need a bit more
reworking of its method.
2018-12-26 14:59:21 -08:00
Chris Robinson 5c449de73f Improve UHJ2 encoding 2018-12-26 13:20:59 -08:00
Chris Robinson 3b9defa4af Improve some post-process handlers 2018-12-26 12:46:01 -08:00
Chris Robinson 5cc545f157 More aggressively try to decrement an effect's refcount in-place 2018-12-26 12:25:34 -08:00
Chris Robinson 38537a35cc Avoid using a local for a temporary 2018-12-25 22:32:30 -08:00
Chris Robinson b2665a503f Do some pre-mixing fading checks once before preparing to mix 2018-12-25 19:54:14 -08:00
Chris Robinson a7c58decfb Add a couple more ASSUMEs for number of channels and sends 2018-12-25 18:49:12 -08:00
Chris Robinson 497226f11e Add an adjust-latency config option for PulseAudio 2018-12-25 17:04:54 -08:00
Chris Robinson 6a8c791e3c Rework the pulseaudio backend to avoid an explicit mixer thread 2018-12-25 16:31:31 -08:00
Chris Robinson 0314370eb5 Cache the process binary path and name 2018-12-25 11:27:22 -08:00
Chris Robinson 208ea76922 Cleanup some includes 2018-12-25 11:09:41 -08:00
Chris Robinson 8336de6653 Rename a couple filter files for consistency 2018-12-25 10:28:02 -08:00
Chris Robinson 9e19acd9e1 Avoid making static local copies of constexpr values 2018-12-25 10:06:17 -08:00
Chris Robinson 63df7cd537 Construct AsyncEvent objects directly in the ringbuffer 2018-12-25 09:32:38 -08:00
Chris Robinson 3b7f668b28 Avoid an intermediate mixing buffer 2018-12-24 20:44:55 -08:00
Chris Robinson fbae41020d Remove extraneous typedef, struct, and enum keywords 2018-12-24 19:29:01 -08:00
Chris Robinson 194e7ff815 Add an in-progress extension to set the effect slot target 2018-12-24 15:52:37 -08:00
Chris Robinson bbf9e6931c Propagate an effectslot target property 2018-12-24 15:17:38 -08:00
Chris Robinson ae86aef4db Provide effect target parameters through a common struct 2018-12-24 13:29:36 -08:00
Chris Robinson cd213fe6b7 Avoid using select() 2018-12-24 09:58:48 -08:00
Chris Robinson 95631aa358 Make the Compressor more class-like 2018-12-24 09:17:00 -08:00
Chris Robinson d49eeb576c Only check ambisonic attributes with B-Format output 2018-12-24 07:33:38 -08:00
Chris Robinson 68352d3188 Apply the limiter before distance compensation 2018-12-24 07:30:01 -08:00
Chris Robinson ef10152361 Assume alignment for some buffers 2018-12-23 20:56:27 -08:00
Chris Robinson 1f966c11ef Add some more ASSUMEs 2018-12-23 17:56:01 -08:00
Chris Robinson 11d815cfd3 Repack some AmbiUpsampler fields for better access patterns 2018-12-23 15:55:12 -08:00
Chris Robinson ba9aba699d Properly rebalance the HF scale with ambisonic upsampling 2018-12-23 10:37:07 -08:00
Chris Robinson 3fe38fed7c Mix effect slot output to the effect target if it's set 2018-12-23 08:51:28 -08:00
Chris Robinson e218999b4f Dynamically sort the effect slots when mixing
This is to be able to support effects that output to other effects. When an
effect outputs to another effect, the former needs to process first, so the
former mixes to the latter's buffer before the latter is processed.

This sorting needs to happen in the mixer because the effect slot's "Target"
property changes asynchronously.
2018-12-22 22:31:26 -08:00
Chris Robinson ebfe818d2e Fix narrowing conversion from double to float 2018-12-22 20:32:00 -08:00
Chris Robinson bfa98be48a Cleanup definitions and declarations in reverb.cpp 2018-12-22 19:31:12 -08:00
Chris Robinson 86caf2683e Constify a parameter 2018-12-22 18:43:34 -08:00
Chris Robinson 334b3a905a Clean up some math stuff 2018-12-22 16:01:14 -08:00
Chris Robinson d4d98e2fe9 Fix for C++11 compatibility
std::array::operator[] isn't constexpr until C++14.
2018-12-22 11:41:03 -08:00
Chris Robinson 10ce121dbd Use a normal delete instead of ll_ringbuffer_free
And use RingBufferPtr in more places
2018-12-22 11:38:38 -08:00
Chris Robinson b955c5cf5d A bit of cleanup for CalcPanningAndFilters 2018-12-22 10:00:06 -08:00
Chris Robinson 87724db6e3 Rename a couple HRTF structs 2018-12-22 09:20:50 -08:00
Chris Robinson 985d03d13d Try to help GetHrtfCoeffs vectorize 2018-12-21 21:07:42 -08:00
Chris Robinson 5e4378f30a Small cleanup for BuildBFormatHrtf 2018-12-21 18:17:59 -08:00
Chris Robinson 3553ce1f67 Don't convert the HRTF decoder virtual speaker positions to radians 2018-12-21 08:55:22 -08:00
Chris Robinson b785d80526 Use a dodecahedron for the ambisonic HRTF decode
Also uses full second-order for "basic" HRTF rendering. Note that the supplied
matrix is full third-order, but only the first- and second-order coefficients
are used. The base matrices are the identical, only differing by the high-
frequency scalars.
2018-12-21 06:32:18 -08:00
Chris Robinson 7744e4ff72 Pass RealMixParams by reference instead of pointer 2018-12-20 13:26:39 -08:00
Chris Robinson 9fde260df9 Fix the type used for another subtraction 2018-12-20 12:29:46 -08:00
Chris Robinson 768ed3cfbb Silence some MSVC warnings 2018-12-20 12:04:34 -08:00
Chris Robinson 837881eb79 Fix the type used for subtraction 2018-12-20 12:03:32 -08:00
Chris Robinson 7dc553350b Clean up most of the compressor loops 2018-12-20 11:46:57 -08:00
Chris Robinson 08b79b9bbf Add an assume_aligned helper 2018-12-20 11:46:40 -08:00
Chris Robinson 10f87c5d26 Use std::accumulate to get the max composited buffer length loaded 2018-12-20 07:10:09 -08:00
Chris Robinson 49ac268334 Add index maps from 2D and 3D 2018-12-20 04:19:35 -08:00
Chris Robinson 8d3f7651c9 Use std::array in place of some C-style arrays 2018-12-20 03:26:46 -08:00
Chris Robinson d18140391a Rename some conversion arrays 2018-12-20 02:46:59 -08:00
Chris Robinson 0214a11024 Use inline methods for the device format sizes 2018-12-19 05:57:36 -08:00
Chris Robinson b49e8985a4 Don't hardcode the channel count from the device ambisonic order 2018-12-19 03:13:15 -08:00
Chris Robinson fbd47961d5 Don't allow FuMa ordering or normalization above third-order 2018-12-19 02:55:21 -08:00
Chris Robinson e2896dc839 Combine handling of attribute processing 2018-12-18 09:27:00 -08:00
Chris Robinson 5243516149 Use the AmbiUpsampler with higher order basic and custom panning
Also allocate the BFormatDec and AmbiUpsampler where they're (re)set.
2018-12-17 07:29:55 -08:00
Chris Robinson 59013b5cb5 Avoid hard-coded scale factors in BFormatDec's upsampler 2018-12-17 07:12:37 -08:00
Chris Robinson a359cb85e6 Mix each frequency band individually for ambisonic upsampling 2018-12-16 22:37:29 -08:00
Chris Robinson 3b0fd20bee Always use the transcode method with the AmbiUpsampler 2018-12-16 21:03:24 -08:00
Chris Robinson 064f4f500a Avoid extraneous alignment requirements 2018-12-16 01:27:52 -08:00
Chris Robinson 741861eaa6 Put the ACN index map in a header
Also put it and the Ambisonic scales in a more appropriate header.
2018-12-15 23:28:49 -08:00
Chris Robinson a6a5634adb Reorder some math terms to help optimizations
Because floating-point math is not associative ((a*b)*c does not necessarily
give the same result as a*(b*c)), the ordering of terms can inhibit reuse of
temporary values. For example, both

coeffs[9]  =  2.091650066f * y * (3.0f*x*x - y*y);
and
coeffs[15] =  2.091650066f * x * (x*x - 3.0f*y*y);

contain x*x and y*y terms that could be calculated once, stored in temporary
registers, and reused to multiply with 3. But since 3.0f*(x*x) would produce
different results, the compiler is not allowed to make that optimization. If,
however, the multiply with 3 is moved to the right side:

coeffs[9]  =  2.091650066f * y * (x*x*3.0f - y*y);
and
coeffs[15] =  2.091650066f * x * (x*x - y*y*3.0f);

in both cases x*x and y*y are calculated first in their respective groups,
guaranteeing the same results for both instances prior to the multiply with 3
and allowing the compiler to reuse those intermediate values.
2018-12-15 20:28:52 -08:00
Chris Robinson dea077cbae Add encoding calculations for fourth-order ambisonics 2018-12-15 19:23:42 -08:00
Chris Robinson e0f635b20d Move some ambisonic-related macros to a separate header 2018-12-15 03:30:47 -08:00
Chris Robinson 0dd13a9dfe Make the AmbDec speaker and matrix arrays dynamic 2018-12-15 02:56:19 -08:00
Chris Robinson 640c06c292 Avoid some explicit loop counts 2018-12-15 01:48:54 -08:00
Chris Robinson 4d36730baa Clean up panning.cpp a bit 2018-12-15 00:38:38 -08:00
Chris Robinson 2d13e0af29 Add macros for the ambisonic order masks 2018-12-14 23:26:44 -08:00
Chris Robinson 0882728dec Cleanup bformatdec.cpp a bit 2018-12-13 22:48:02 -08:00
Chris Robinson d18ae81e9c Add POPCNT32 and CTZ32 macros 2018-12-13 22:05:47 -08:00
Chris Robinson 0d73b13f59 Add more casts for MSVC 2018-12-12 21:58:41 -08:00
Chris Robinson b779ebb512 Fix some MSVC conversion warnings 2018-12-12 21:18:31 -08:00
Chris Robinson b37bc9f8b7 Fix an MSVC warning 2018-12-12 19:24:06 -08:00
Chris Robinson 43f6a7c626 Remove an unused source 2018-12-12 04:34:09 -08:00
Chris Robinson 5a283c66ee Use proper classes for Vector and Matrix types 2018-12-12 04:22:11 -08:00
Chris Robinson 19c5c41c70 Cleanup alu.cpp some 2018-12-12 01:09:04 -08:00
Chris Robinson 0d36ba0fbb Use helpers to get the Ambisonic scales and layout maps 2018-12-10 22:35:32 -08:00
Chris Robinson e87eb07db4 A bit more cleanup 2018-12-10 21:30:22 -08:00
Chris Robinson ed18fd76c5 Clean up a few more loops 2018-12-10 14:49:57 -08:00
Chris Robinson d91ada2e02 Add missing header 2018-12-10 02:13:57 -08:00
Chris Robinson 9bb7ed0129 Put static methods into an anonymous namespace 2018-12-10 02:08:54 -08:00
Chris Robinson 0d56c59f14 Avoid some more explicit loops 2018-12-10 00:01:13 -08:00
Chris Robinson 0a805727db Use std::accumulate to find the max channel count 2018-12-09 18:16:00 -08:00
Chris Robinson b1beb7dfdc Avoid some more explicit loops 2018-12-09 17:24:00 -08:00
Chris Robinson efb8e076c7 Pass a reference to an array for a function parameter 2018-12-09 15:21:24 -08:00
Chris Robinson 6c8f45b5f6 Rename a couple global variables
Avoid clashing with an enum name
2018-12-09 15:07:44 -08:00
Chris Robinson d7d9870839 Add default construction to HrtfEntry 2018-12-09 13:36:13 -08:00
Chris Robinson 621f0dfe73 Use a vector for LoadedHrtfs 2018-12-09 03:04:18 -08:00
Chris Robinson e5db9b2378 Avoid static global initialization functions 2018-12-08 21:58:44 -08:00
Chris Robinson a4009c47e7 Add a cmake option to specify prebuilt native tools
This should only be used with automated build systems that guarantee the native
tools' binaries are up-to-date. Otherwise it's best to leave it alone so it can
automatically rebuild them as needed.
2018-12-08 17:22:10 -08:00
Chris Robinson 30a3a19574 Rename a member variable and inline a function 2018-12-08 16:30:19 -08:00
Chris Robinson c9f5617f06 Avoid several uses of memset 2018-12-08 14:22:20 -08:00
Chris Robinson fc8da0c16b Add missing include 2018-12-08 13:06:31 -08:00
Chris Robinson 5a9a2cb1ed Increase GAIN_MIX_MAX 2018-12-08 04:10:59 -08:00
Chris Robinson 9e9b771e8a A bit more cleanup 2018-12-08 04:10:45 -08:00
Chris Robinson 5ea3c8fb60 Use member functions for BFormatDec and AmbiUpsampler 2018-12-08 02:50:34 -08:00
Chris Robinson 7695afe0cb Clean up some more loops 2018-12-08 02:15:00 -08:00
Chris Robinson a603cc906f Inline a simple method 2018-12-08 00:08:39 -08:00
Chris Robinson ab5a11d5f3 Avoid using the deprecated mem_fun_ref method 2018-12-07 21:46:22 -08:00
Chris Robinson 6e4c856257 Add fourth-order methods to the NFC filter
Unused, but it finishes out the currently possible implementations.
2018-12-07 18:38:56 -08:00
Chris Robinson 0f24139b57 Use a constructor instead of a macro to initialize AsyncEvent 2018-12-06 22:24:20 -08:00
Chris Robinson 42d26472eb Remove some more explicit loops 2018-12-05 17:11:33 -08:00
Chris Robinson e60d0886d0 Use class methods for BandSplitter and SplitterAllpass filters 2018-12-05 15:38:05 -08:00
Chris Robinson 10b39d57d5 Use class methods for the NFC filters 2018-12-05 15:20:52 -08:00
Chris Robinson 164a86a381 Use class methods for the biquad filter 2018-12-04 22:31:08 -08:00
Chris Robinson 36a8b615c8 Avoid using AL types with the filters 2018-12-04 20:55:10 -08:00
Chris Robinson 2a30ae3807 Avoid some more explicit loops in the filters 2018-12-04 20:35:23 -08:00
Chris Robinson 3866c9f941 Avoid more explicit loops 2018-12-04 19:45:11 -08:00
Chris Robinson a94bfd3ec9 Increase the async event queue size 2018-12-04 16:38:22 -08:00
Chris Robinson 6409910543 Handle EventType_BufferCompleted uniquely 2018-12-04 16:33:26 -08:00
Chris Robinson f79b0e24d4 Avoid a few more explicit loops 2018-12-04 15:30:39 -08:00
Chris Robinson a0967967de Read atomic variables in the reverse order they're set 2018-12-04 14:48:08 -08:00
Chris Robinson 31b54eb86a Avoid a few more explicit loops 2018-12-03 01:41:52 -08:00
Chris Robinson 85e83bbdff Use a std::string for the source state change message 2018-12-02 19:34:09 -08:00
Chris Robinson 45b65366bb Improve some checks for compiler analysis 2018-12-02 15:29:26 -08:00
Chris Robinson 60254488e9 Add a couple missing source lock guards 2018-12-02 14:36:53 -08:00
Chris Robinson 4e17ad3ec0 Avoid a few more explicit loops 2018-12-02 13:35:07 -08:00
Chris Robinson 607fb3e632 Use a lambda to apply the NFC mixes 2018-12-01 23:25:34 -08:00
Chris Robinson e1af866a3d Rework source sample counting for mixing a bit 2018-12-01 13:47:56 -08:00
Chris Robinson 255c07def9 Try to pacify MSVC's missing a suitable default constructor 2018-12-01 11:28:11 -08:00
Chris Robinson cc161fe7c1 Add a missing include 2018-12-01 01:12:05 -08:00
Chris Robinson 7b1548af3c Handle source state changed events uniquely in the event loop
To avoid the need of constructing the string in the mixer thread, which is
commonly formatted anyway.
2018-11-30 21:39:59 -08:00
Chris Robinson 1e6e84374b Use std::array for the voice's PrevSamples 2018-11-30 21:23:43 -08:00
Chris Robinson c7569c31ad Improve construction and destruction of ALvoices 2018-11-30 19:04:38 -08:00
Chris Robinson 4b7ac4a6ed Don't bother making ALvoiceProps dynamically sized 2018-11-30 16:56:23 -08:00
Chris Robinson 364850e8f9 Move a variable declaration to a more appropriate place 2018-11-30 10:08:18 -08:00
Chris Robinson b582f1346a Avoid hard-coding a couple sizes 2018-11-29 22:56:33 -08:00
Chris Robinson 8ca8da30bd Store the source ID with the voice instead of the source pointer 2018-11-29 22:49:01 -08:00
Chris Robinson 0d2bbe17f2 Rename a function for consistency 2018-11-29 13:34:06 -08:00
Chris Robinson c5d5d574e6 Reorganize some device members 2018-11-29 13:10:46 -08:00
Chris Robinson 57bb467072 Pass the desired resampler to CreateSampleConverter 2018-11-28 23:19:25 -08:00
Chris Robinson e017d9e40f Clean up the converter a bit 2018-11-28 22:42:46 -08:00
Chris Robinson 5df89c504e Remove an improper Connected check 2018-11-28 19:29:20 -08:00
Chris Robinson 38f4a0cf2c Avoid the update size going to 0 with a relatively large device period 2018-11-28 18:45:35 -08:00
Chris Robinson c3c0a5022a Don't bother with (really) old pulseaudio headers 2018-11-28 15:09:19 -08:00
Chris Robinson 14d746e7c2 Don't sever a paused source from its voice on disconnect 2018-11-28 14:38:26 -08:00
Chris Robinson 598851fed7 Attempt to reconnect lost devices with alcResetDeviceSOFT
Be aware there's currently possible race conditions with PulseAudio's callbacks
(the state callbacks need to be cleared while not playing).

Also paused sources will assert/crash if attempted to play again without being
explicitly stopped or rewound first.

Both of these will eventually be fixed (though a paused source's offset will be
lost regardless).
2018-11-28 12:24:12 -08:00
Chris Robinson 16aa3ab196 Reduce some indentation 2018-11-28 11:55:43 -08:00
Chris Robinson ed2e456dfb Avoid some explicit loops 2018-11-28 11:30:44 -08:00
Chris Robinson 8d95b6a0f2 Avoid an unnecessary lambda 2018-11-28 08:42:30 -08:00
Chris Robinson ec2927cd32 Small cleanup for ~ALCcontext_struct 2018-11-27 21:19:30 -08:00
Chris Robinson ff8c694957 Clean up the JACK backend 2018-11-27 20:29:54 -08:00
Chris Robinson 0116e763ea Add a unique_ptr alias for the ringbuffer 2018-11-27 19:49:45 -08:00
Chris Robinson 2179388430 Remove unneeded mutex checks 2018-11-27 15:23:18 -08:00
Chris Robinson c2da83dea8 Avoid alsem* wrappers for al::semaphore 2018-11-27 14:52:04 -08:00
Chris Robinson 89abbe8d94 Replace last uses of alsem_t with al::semaphore 2018-11-27 14:05:56 -08:00
Chris Robinson f26083e9ed Make and use a semaphore class 2018-11-27 13:41:30 -08:00
Chris Robinson 3f745be1dc Return a signed integer from altime_get 2018-11-27 12:11:11 -08:00
Chris Robinson ff6dda06ad Add the appropriate include for the _POSIX_TIMERS macro 2018-11-27 10:35:00 -08:00
Chris Robinson 07670ed36a Swap context references in the move assignment 2018-11-27 10:08:25 -08:00
Chris Robinson ee2d756d94 Disable MSVC warning C4065
"switch statement contains 'default' but no 'case' labels"
2018-11-27 00:04:55 -08:00
Chris Robinson 9b2b83f99c Fix use of clock_gettime 2018-11-27 00:02:31 -08:00
Chris Robinson 2530370ff2 Avoid relying on struct timespec 2018-11-26 23:45:04 -08:00
Chris Robinson d7d99adc91 Avoid including threads.h in the example helpers 2018-11-26 23:18:51 -08:00
Chris Robinson d06f76957c Remove althrd_yield 2018-11-26 23:06:49 -08:00
Chris Robinson 4c1fc3ae00 Remove unused almtx stuff 2018-11-26 22:36:55 -08:00
Chris Robinson 2d45ec8dc3 Use a standard mutex for the remaining locks 2018-11-26 22:06:53 -08:00
Chris Robinson 68eef6abb4 Use a standard mutex for the source and effect slot locks 2018-11-26 21:50:48 -08:00
Chris Robinson 2e73c2ca92 Use a standard mutex for the proplock 2018-11-26 21:39:31 -08:00
Chris Robinson d8b9230ee4 Use a standard mutex for the backend lock 2018-11-26 21:39:31 -08:00
Chris Robinson b108d0acfd Remove the last remaining uses of althrd_t 2018-11-26 21:39:31 -08:00
kcat 7860a11ae2 Merge pull request #249 from wangwenx190/patch-win10-sdk
Support build with latest Win10 SDK
2018-11-26 21:39:05 -08:00
wangwenx190 00ad4a2f5d Update FindWindowsSDK.cmake 2018-11-27 13:20:04 +08:00
wangwenx190 9070426ece Update FindWindowsSDK.cmake
Support latest Win10 SDK.
2018-11-27 13:18:30 +08:00
Chris Robinson 9ac76c0a7f Simplify some binary search lookups 2018-11-26 20:55:00 -08:00
Chris Robinson bf9db1fe3d Remove althrd_t from Windows 2018-11-26 20:34:16 -08:00
Chris Robinson ecab90802a Replace some uses of althrd_t with std::thread 2018-11-26 20:06:22 -08:00
Chris Robinson 05390fa827 Add a missing std::begin 2018-11-26 19:01:14 -08:00
Chris Robinson 75b39cafc8 Get rid of some unnecessary functions 2018-11-26 18:25:29 -08:00
Chris Robinson df3dcc879f Get rid of the last ATOMIC macro uses 2018-11-26 18:19:58 -08:00
Chris Robinson 461ef4196e Avoid using ATOMIC_LOAD on ALCdevice::Connected 2018-11-26 18:07:52 -08:00
Chris Robinson 1a9edd4e35 Use a unique_ptr for the QSA backend data 2018-11-26 17:53:56 -08:00
Chris Robinson 04c2802a82 Rename some struct members 2018-11-26 17:31:04 -08:00
Chris Robinson 04cbdbd569 Remove some unused macros 2018-11-26 14:53:20 -08:00
Chris Robinson 053599b243 Avoid using the ATOMIC() macro 2018-11-26 14:48:26 -08:00
Chris Robinson a6923790fa Avoid using ATOMIC_INIT 2018-11-26 14:31:54 -08:00
Chris Robinson 5b2b96b245 Don't explicitly clear vector objects in the destructor 2018-11-25 17:51:39 -08:00
Chris Robinson 127ec026e7 Automatically clean up filters and effects with their sublists 2018-11-25 16:16:40 -08:00
Chris Robinson 8ae07ad1ae Automatically clean up buffers with ther sublist 2018-11-25 15:30:32 -08:00
Chris Robinson 05845b53e8 Clean up MixSource 2018-11-25 13:20:26 -08:00
Chris Robinson ad2639248a Avoid another DECL_TEMPLATE macro 2018-11-25 10:52:17 -08:00
Chris Robinson adcdb8ce64 Fix an incorrect function call 2018-11-25 09:27:50 -08:00
Chris Robinson ec97360352 Avoid a separate function to clean up effect slots 2018-11-25 09:23:01 -08:00
Chris Robinson bf4518fe5c Atuomatically clean up sources with its sublist's destruction 2018-11-25 08:42:43 -08:00
Chris Robinson 7c0605f09e Properly initialize the sublists' freemask 2018-11-25 07:40:15 -08:00
Chris Robinson f5f2cdaaf3 Add a POPCNT64 macro
To count the number of 1/on bits in a 64-bit value
2018-11-24 21:16:53 -08:00
Chris Robinson 71660df5e5 Move bs2b.h to a more appropriate place 2018-11-24 19:54:30 -08:00
Chris Robinson 377325e794 Use C++ methods a bit more 2018-11-24 19:16:21 -08:00
Chris Robinson 9e10f632c7 Replace remaining uses of std::vector with al::vector
Which uses a custom allocator that uses our allocation functions.
2018-11-24 16:58:49 -08:00
Chris Robinson 16a60dc371 Avoid an extraneous boolean 2018-11-24 15:30:28 -08:00
Chris Robinson bd8db0d27b Make GetContextRef return a ContextRef 2018-11-24 14:07:32 -08:00
Chris Robinson 62d38b1187 Use RAII in alEffect.cpp 2018-11-24 13:23:27 -08:00
Chris Robinson 6a84a2ca61 Fix a couple comments 2018-11-24 12:09:37 -08:00
Chris Robinson d83cff02e5 Ensure an enum is the appropriate size/type 2018-11-24 11:41:50 -08:00
Chris Robinson b508a760c8 Use a normal vector to store buffer data 2018-11-24 10:07:48 -08:00
Chris Robinson 505e535655 Use RAII more with alFilter.cpp 2018-11-24 09:40:13 -08:00
Chris Robinson 48154c94d7 Use a unique_ptr while opening a device 2018-11-23 20:16:34 -08:00
Chris Robinson a7c556f814 Use RAII for device references 2018-11-23 17:54:12 -08:00
Chris Robinson df057d4118 Make the context VoiceCount atomic 2018-11-23 16:16:31 -08:00
Chris Robinson 73528c9f55 Fix an incorrect check 2018-11-23 14:42:02 -08:00
Chris Robinson fb94cdcfd3 Restructure and clean up alu.cpp a bit 2018-11-23 13:12:48 -08:00
Chris Robinson 438e626993 Avoid a couple explicit loops 2018-11-22 22:53:22 -08:00
Chris Robinson 976e49711b Add noexcept to a bunch of inline functions 2018-11-22 19:47:41 -08:00
Chris Robinson 30ee6e1b3f Make a DirectHrtfState constructor to try appeasing MSVC 2018-11-22 14:36:37 -08:00
Chris Robinson d26b5d9467 Use proper time types for the device clock time and latency 2018-11-22 14:32:48 -08:00
Chris Robinson 84f0f74d07 Use standard types for the device clock times 2018-11-22 12:53:16 -08:00
Chris Robinson bb9d8db73c Clean up alSource.cpp some 2018-11-22 12:02:02 -08:00
Chris Robinson cc938c34b2 Clean up the Chorus a little 2018-11-22 08:08:46 -08:00
Chris Robinson 9c155a57fb Use unique_ptr for DirectHrtfState 2018-11-22 07:54:29 -08:00
Chris Robinson ba8c865513 Add and use a macro to define placement-new-only allocators
This is for structs that utilize over-allocation, either flexible array
members, or which store optional additional objects in the same allocation
block.
2018-11-22 07:06:42 -08:00
Chris Robinson ab6db9a589 Clean up some unnecessary specifiers 2018-11-22 06:59:32 -08:00
Chris Robinson 9d73e03aaa Use unique_ptr for bs2b 2018-11-22 06:49:37 -08:00
Chris Robinson b3b4220182 Use unique_ptr for BFormatDec and AmbiUpsampler 2018-11-22 05:37:35 -08:00
Chris Robinson 671ed1abf8 Use a unique_ptr for the FrontStablizer 2018-11-22 04:53:29 -08:00
Chris Robinson cc3e2a838f Use a unique_ptr for the Compressor 2018-11-21 16:46:52 -08:00
Chris Robinson eefc379a23 Use a unique_ptr for Uhj2Encoder 2018-11-21 15:31:32 -08:00
Chris Robinson dfcc98afbf Fix deleting the same buffer ID multiple times in one call 2018-11-21 11:51:34 -08:00
Chris Robinson d20522166e Use RAII more in alSource.cpp 2018-11-21 11:11:25 -08:00
Chris Robinson 9f2a77f788 Use RAII when handling the mixer's FPU state 2018-11-21 09:07:02 -08:00
Chris Robinson 8f43f737ba Avoid using ATOMIC macros 2018-11-21 07:52:17 -08:00
Chris Robinson de4bb7aca1 Replace a couple more almtx_lock/unlock calls 2018-11-21 06:41:49 -08:00
Chris Robinson 75213ee6f9 Always use RAII with EffectSlotLock 2018-11-21 05:35:47 -08:00
Chris Robinson 757c42c74b Use a normal vector for the distance buffer storage 2018-11-21 05:06:31 -08:00
Chris Robinson 4c64fa1e4e Make VerifyContext return a context reference 2018-11-21 02:02:53 -08:00
Chris Robinson 8ec25da12e Rename a method to be clearer about its behavior 2018-11-21 00:40:41 -08:00
Chris Robinson ed06169569 Use RAII locks with the BackendLock 2018-11-20 23:42:21 -08:00
Chris Robinson d64d64d4a5 Use an enum class for AmbiLayout/Norm settings 2018-11-20 22:47:24 -08:00
Chris Robinson dc8ef8264a Try to improve alffplay underrun device time adjustment 2018-11-20 12:32:42 -08:00
Chris Robinson f3e01ae9d4 Use a normal vector for the source send properties 2018-11-20 12:25:15 -08:00
Chris Robinson a111254c26 Use C++ more in alListener.cpp 2018-11-20 11:17:54 -08:00
Chris Robinson f9e969a339 Use a standard mutex for the event callback lock 2018-11-20 10:55:57 -08:00
Chris Robinson 191ea90de3 Use atomic_flags and atomic<bools>s where appropriate 2018-11-20 10:45:01 -08:00
Chris Robinson 1e31ac469e Store effect slots as unique_ptrs 2018-11-20 10:01:20 -08:00
Chris Robinson 29558c091b Use C++ more in alAuxEffectSlot.cpp 2018-11-20 09:47:49 -08:00
Chris Robinson 1df42c4a0f Use a unique_ptr for the default effect slot 2018-11-20 05:01:08 -08:00
Chris Robinson 66fdd027d0 Use more C++ templates instead of DECL_TEMPLATE tricks 2018-11-20 03:35:41 -08:00
Chris Robinson c7472f8642 Use C++ templates instead of macro definitions more 2018-11-20 02:59:02 -08:00
Chris Robinson 16d0e79db7 Add compile-time traits for FmtType enum values 2018-11-20 02:42:49 -08:00
Chris Robinson 457d484872 Use C++ templates instead of macro-defined variations 2018-11-20 02:12:04 -08:00
Chris Robinson 5881cd2183 Add compile-time traits for DevFmtType enum values 2018-11-20 02:01:03 -08:00
Chris Robinson eb2937de84 Avoid another case of a variable named the same as a type 2018-11-19 23:48:45 -08:00
Chris Robinson 288dbbe886 Use default initialization for the reverb effect state 2018-11-19 22:48:56 -08:00
Chris Robinson 8995306203 Use proper templating for the modulator's Modulate function 2018-11-19 22:43:28 -08:00
Chris Robinson 8472a9d916 Use proper inheritence for the effect state objects 2018-11-19 22:34:26 -08:00
Chris Robinson 6ac84c7a5f Clean up the remaining effect struct member names 2018-11-19 21:04:50 -08:00
Chris Robinson 67da3850cb Use a normal vector for the echo buffer 2018-11-19 20:15:40 -08:00
Chris Robinson a346380e2b Clean up more effects' struct members 2018-11-19 19:57:30 -08:00
Chris Robinson 7f3584ec4c Fix the reverb buffer size calculation 2018-11-19 19:13:44 -08:00
Chris Robinson 4810def613 Use a normal vector for the chorus sample buffer 2018-11-19 19:08:30 -08:00
Chris Robinson ba33f6a7a4 Clean up the chorus/flanger struct members 2018-11-19 18:51:01 -08:00
Chris Robinson 3ae1c78d1a Use std::isfinite instead of isfinite 2018-11-19 09:51:29 -08:00
Chris Robinson ddfed7187f Use a regular vector for the reverb sample buffer 2018-11-19 09:36:59 -08:00
Chris Robinson 55637ff1fc Rename reverb struct members 2018-11-19 09:29:22 -08:00
Chris Robinson 387a34ca00 Clean up the biquad filter a bit 2018-11-19 09:10:36 -08:00
Chris Robinson 07386e79de Fix up the struct member names in the autowah effect 2018-11-19 09:06:17 -08:00
Chris Robinson f51f6703d8 Add a missing include 2018-11-19 06:50:37 -08:00
Chris Robinson c5c537cc5f Use proper inheritence for EffectStateFactory 2018-11-19 06:43:37 -08:00
Chris Robinson f0cc34a60e Use a vector to handle mixing buffer storage 2018-11-19 06:22:09 -08:00
Chris Robinson f766437569 Fix a couple incorrect uses of ringbuffer pointer data 2018-11-19 05:07:09 -08:00
Chris Robinson ac2a420351 Remove the ATOMIC_THREAD_FENCE macro 2018-11-19 05:04:17 -08:00
Chris Robinson ad5f9d9b22 Return the ringbuffer data pointers as a pair 2018-11-19 04:46:49 -08:00
Chris Robinson a14f39ea06 Make ll_ringbuffer_write/read take void*/const void* 2018-11-19 04:11:21 -08:00
Chris Robinson c01743fe5d Remove the CONST_CAST hack 2018-11-19 03:53:31 -08:00
Chris Robinson e6c2c1f3b6 Remove unnecessary using statements 2018-11-19 03:32:16 -08:00
Chris Robinson e24435ef58 Remove the atomic exchange macros 2018-11-19 03:21:58 -08:00
Chris Robinson c5142530d6 Simplify the RefCount type 2018-11-19 02:17:06 -08:00
Chris Robinson 6e114a7a70 Replace ATOMIC_REPLACE_HEAD with an inline function 2018-11-19 01:20:03 -08:00
Chris Robinson 190de1452e Remove the unused vector macros 2018-11-18 23:37:56 -08:00
Chris Robinson dbf2381a3b Use a normal vector in the QSA backend 2018-11-18 23:23:12 -08:00
Chris Robinson e0d0faaa63 Use a normal vector for filter sublists 2018-11-18 23:02:31 -08:00
Chris Robinson 362979cefb Use a normal vector for effect sublists 2018-11-18 22:50:04 -08:00
Chris Robinson 81aed2ea01 Use a normal vector for buffer sublists 2018-11-18 22:31:51 -08:00
Chris Robinson becbaab2dc Remove some unnecessary static specifiers 2018-11-18 22:14:44 -08:00
Chris Robinson e23796aabe Use a standard string for the enumerated HRTF names 2018-11-18 21:18:19 -08:00
Chris Robinson 140c139852 Use a standard string for the device's HRTF name 2018-11-18 19:28:01 -08:00
Chris Robinson 2c06ec7093 Use a regular vector for the enumerated HRTF list 2018-11-18 19:19:35 -08:00
Chris Robinson bafcba7194 Use a std::string for the device name 2018-11-18 18:45:45 -08:00
Chris Robinson b10e7d08c3 Use a std::thread for the event thread 2018-11-18 18:04:27 -08:00
Chris Robinson ef7995cfd0 Fix the initial device refcount 2018-11-18 08:14:23 -08:00
Chris Robinson aef52be432 Fix a float constant type 2018-11-18 08:03:22 -08:00
Chris Robinson 8b8f01e25d Avoid more cases of an enum variable and type name clash 2018-11-18 08:01:50 -08:00
Chris Robinson 336b7b7799 Use constructors/destructors with ALCdevice
As with ALCcontext, this is really ALCdevice_struct because of the way it's
declared in the public header.
2018-11-18 07:33:42 -08:00
Chris Robinson 1bc88ed751 Avoid a fixed-size string buffer 2018-11-18 07:00:43 -08:00
Chris Robinson 5bbddff2f3 Separate class and variable definitions 2018-11-18 06:46:50 -08:00
Chris Robinson e194d896de Use constructors/destructors for ALeffectslot 2018-11-18 06:35:45 -08:00
Chris Robinson ad82a70a65 Use cleaner constructor/destructor calls for ALCcontext
Note that the actual type name is ALCcontext_struct, because of how it's
defined in AL/alc.h (ALCcontext is just an alias to struct ALCcontext_struct).
2018-11-18 05:40:00 -08:00
Chris Robinson de13f30e28 Improve audio underrun recordery in alffplay
Now it has a better idea to skip samples during refill instead of after
restarting.
2018-11-18 05:38:03 -08:00
Chris Robinson 0851dc12b4 Remove an unused typedef 2018-11-18 04:26:28 -08:00
Chris Robinson f48be9d73b Remove the pointer-specific atomic exchange macros 2018-11-18 03:59:39 -08:00
Chris Robinson 7433cb5f4c Avoid naming a struct member the same as an enum type 2018-11-18 03:39:32 -08:00
Chris Robinson bf30eb0391 Use std::isfinite instead of the global isfinite 2018-11-18 03:25:32 -08:00
Chris Robinson 310770c531 Add and use new/delete operators to ALeffectslot 2018-11-18 02:52:46 -08:00
Chris Robinson 3eededf5d6 Use a normal vector for auxiliary effect slots 2018-11-18 02:39:27 -08:00
Chris Robinson 7b95712f38 Use a proper vector for the source sublists 2018-11-18 02:15:31 -08:00
Chris Robinson 4dc8f44d00 Move the alignment-aware allocator and vector to headers 2018-11-18 01:33:26 -08:00
Chris Robinson d7cc9b912b Use new/delete for ALCcontext objects 2018-11-18 00:38:31 -08:00
Chris Robinson 38d6df9c1d Store the listener directly in the context 2018-11-17 23:41:11 -08:00
Chris Robinson e79d9bdd1a Move the vector and matrix declarations to a separate header 2018-11-17 23:21:37 -08:00
Chris Robinson fa3c34268d Move the ALCcontext definition to its own header 2018-11-17 23:02:27 -08:00
Chris Robinson 8f6e0f97ec Try to clear up an MSVC warning 2018-11-17 21:48:54 -08:00
Chris Robinson 20e62aa959 Avoid an explicit static_cast to bool 2018-11-17 20:39:45 -08:00
Chris Robinson d10301c209 Remove unused headers and checks 2018-11-17 19:52:54 -08:00
Chris Robinson 1ac41d3ea0 Convert almalloc.c to C++ 2018-11-17 19:01:10 -08:00
Chris Robinson 8c69fb046c Always use C++11 atomics 2018-11-17 18:56:00 -08:00
Chris Robinson 1468dee831 Convert bs2b.c to C++ 2018-11-17 18:54:10 -08:00
Chris Robinson 9992cef915 Remove now unneeded inldefs.c
All code using inline functions is now C++, so will generate callable functions
as-needed.
2018-11-17 17:52:23 -08:00
Chris Robinson 7f69dbb517 Convert the mixers to C++ 2018-11-17 17:49:55 -08:00
Chris Robinson 2d4ff77410 Remove ASSUME_ALIGNED
It's become a liability with C++ since it returns void* instead of the input
pointer type, and it doesn't seem to help optimizations anyway (auto-
vectorization still produces unaligned loads and stores).
2018-11-17 17:35:52 -08:00
Chris Robinson e5442db803 Convert the filters to C++ 2018-11-17 17:22:32 -08:00
Chris Robinson ed5d222eed Remove the old unused bsincgen.c 2018-11-17 07:50:46 -08:00
Chris Robinson 057b1411f9 Convert ringbuffer.c to C++ 2018-11-17 07:45:10 -08:00
Chris Robinson ff4219e54e Convert mastering.c to C++ 2018-11-17 07:40:10 -08:00
Chris Robinson 6e6a024058 Convert converter.c to C++ 2018-11-17 07:35:11 -08:00
Chris Robinson 13478126cb Convert the remaining effects to C++ 2018-11-17 07:08:41 -08:00
Chris Robinson 93d96ced9c Convert the dedicated, distortion, echo, and equalizer to C++ 2018-11-17 06:53:20 -08:00
Chris Robinson ad34855a2b Add a couple missing includes 2018-11-17 06:11:55 -08:00
Chris Robinson 7b537c795b Don't pass the current thread to althrd_setname 2018-11-17 06:07:04 -08:00
Chris Robinson 1fae8c16a8 Convert threads.c to C++
Also vastly simplify and remove related code.
2018-11-17 05:31:29 -08:00
Chris Robinson ccdaca80c9 Use standard complex types instead of custom 2018-11-17 04:14:57 -08:00
Chris Robinson 09943683b5 Remove some more unused stuff 2018-11-17 02:41:21 -08:00
Chris Robinson b485cbe53a Make the Hann windows const 2018-11-17 02:30:41 -08:00
Chris Robinson b69b3bd89f Convert fshifter.c to C++ 2018-11-17 01:58:38 -08:00
Chris Robinson aa66ed0ea5 Convert modulator.c to C++ 2018-11-17 01:53:39 -08:00
Chris Robinson a7d3c24b51 Convert null.c to C++ 2018-11-17 01:49:26 -08:00
Chris Robinson 557048afa2 Convert pshifter.c to C++ 2018-11-17 01:36:47 -08:00
Chris Robinson f1731af282 Remove unneeded declarations and definitions 2018-11-17 01:29:35 -08:00
Chris Robinson 3bbfd0c099 Fix compilation with MSVC 2018-11-16 23:01:40 -08:00
Chris Robinson 1bd40d9434 Put a lambda in the call_once call
This optimizes better and avoids a visible symbol (templating to an anonymous
lambda type will generate a unique anonymous instantiation, making inlining
more likely, whereas passing a general function pointer/reference type will
likely use or generate a publically available instance).
2018-11-16 22:41:04 -08:00
Chris Robinson de8d8b5216 Remove unused altss types and methods 2018-11-16 22:12:35 -08:00
Chris Robinson ee8e6733e1 Avoid using C-style TLS 2018-11-16 21:55:11 -08:00
Chris Robinson 8410e71a34 Convert the reverb effect to C++ 2018-11-16 21:31:52 -08:00
Chris Robinson a68d0b68d7 Convert mixvoice.c to C++ 2018-11-16 20:46:50 -08:00
Chris Robinson 53373a43b8 Convert ALu.c to C++
Required changes to bsincgen to generate C++-friendly structures.
2018-11-16 20:32:19 -08:00
Chris Robinson 317acd6ae2 Convert alEffect.c and alFilter.c to C++ 2018-11-16 18:44:43 -08:00
Chris Robinson a15a678da6 Convert alListener.c to C++ 2018-11-16 18:37:55 -08:00
Chris Robinson 0e0fe15b98 Convert alAuxEffectSlot.c to C++ 2018-11-16 18:28:39 -08:00
Chris Robinson ce370be52b Remove some unneeded includes 2018-11-16 08:09:56 -08:00
Chris Robinson 50f36d39f7 Use lock_guard instead of manual lock/unlock calls 2018-11-16 07:11:27 -08:00
Chris Robinson 165c162d01 Convert alState.c to C++ 2018-11-16 06:48:33 -08:00
Chris Robinson 436db28a3f Convert alError.c to C++ 2018-11-16 06:24:24 -08:00
Chris Robinson 8be45fe8a5 Convert alExtension.c to C++ 2018-11-16 06:08:25 -08:00
Chris Robinson fedd5ebbea Convert event.c to C++ 2018-11-16 06:00:28 -08:00
Chris Robinson 5cdd28c736 Convert sample_cvt.c to C++ 2018-11-16 05:24:15 -08:00
Chris Robinson 02eae1123f Use iterators instead of indexed loops 2018-11-16 05:23:42 -08:00
Chris Robinson fc8191012a Cleanup some syntax in alc.cpp 2018-11-16 00:06:54 -08:00
Chris Robinson 7a06a54af7 Simplify a couple checks 2018-11-15 23:58:34 -08:00
Chris Robinson a727978eb9 Remove unused declarations and definitions 2018-11-15 23:55:23 -08:00
Chris Robinson 7ef7477a13 Convert the PortAudio backend factory 2018-11-15 23:50:15 -08:00
Chris Robinson 78eb68a89f Convert the WinMM backend factory 2018-11-15 23:41:09 -08:00
Chris Robinson 2ca6119e0e Convert the DSound backend factory 2018-11-15 23:32:28 -08:00
Chris Robinson 5dc1956e18 Convert the QSA backend factory 2018-11-15 23:10:06 -08:00
Chris Robinson c78b42fb4e Convert the OSS backend factory 2018-11-15 23:02:26 -08:00
Chris Robinson 7884cec02b Convert the SndIO backend factory 2018-11-15 22:36:49 -08:00
Chris Robinson 271cfcf8e3 Convert the Solaris backend factory 2018-11-15 22:23:29 -08:00
Chris Robinson d73d01548d Convert the SDL2 backend factory 2018-11-15 22:15:10 -08:00
Chris Robinson 2a839e5762 Convert the JACK backend factory 2018-11-15 22:03:20 -08:00
Chris Robinson d4928d4e7d Convert the Wave Writer backend factory 2018-11-15 21:53:14 -08:00
Chris Robinson cc113ce6ab Convert the OpenSL backend factory 2018-11-15 21:42:17 -08:00
Chris Robinson 9d43b548cc Convert the ALSA backend factory 2018-11-15 21:33:44 -08:00
Chris Robinson ead830814b Convert the CoreAudio backend factory 2018-11-15 21:24:09 -08:00
Chris Robinson b1fb2e9e14 Convert the WASAPI backend factory 2018-11-15 21:14:20 -08:00
Chris Robinson e716c7b988 Convert the PulseAudio backend factory 2018-11-15 19:57:21 -08:00
Chris Robinson 4311c609e4 Update the loopback backend 2018-11-15 19:42:13 -08:00
Chris Robinson 49d8ac2537 Start a new backend factory API
Using proper class inheritance. Be aware this breaks all backends except null
(and loopback). They will be restored individually in due time.
2018-11-15 19:15:14 -08:00
Chris Robinson 7c93308771 Remove checks for functions that always exist
They're part of C++11 and available on the testing systems. If some system has
trouble, switching to proper C++ calls should fix it.
2018-11-15 17:53:05 -08:00
Chris Robinson dc622b3182 Use std::isfinite from the cmath header 2018-11-15 06:48:52 -08:00
Chris Robinson a3b644374b Add a missing cast for MSVC 2018-11-15 06:48:00 -08:00
Chris Robinson 245b7ff0b4 Remove the unused al_string API 2018-11-15 06:32:01 -08:00
Chris Robinson 7b3a2085aa Use a regular char* for the device's name 2018-11-15 06:23:01 -08:00
Chris Robinson ab9f8162b8 Pass a normal const char* to EnumerateHrtf 2018-11-15 05:38:27 -08:00
Chris Robinson 9d9d626d99 Avoid some more uses of al_string 2018-11-15 05:33:02 -08:00
Chris Robinson 08bee7cb58 Search the right list for WASAPI name duplicates 2018-11-15 04:35:35 -08:00
Chris Robinson 17161131e5 Remove an unused extern inline 2018-11-15 04:28:47 -08:00
Chris Robinson 1971d0f5c6 Use std::string instead of al_string for enumerating 2018-11-15 04:24:33 -08:00
Chris Robinson d4f64b9e29 Use a C++ mutex with the device backend base 2018-11-15 03:49:59 -08:00
Chris Robinson b7daddb564 Try to clean up compat.h's macro block spaghetti a bit 2018-11-15 03:29:56 -08:00
Chris Robinson 8f771a0387 Convert alSource.c to C++ 2018-11-15 03:13:54 -08:00
Chris Robinson 27e7168ad4 Only declare the device backend stuff with C++ 2018-11-15 02:36:05 -08:00
Chris Robinson 1d2cc90175 Use C++ more in the OSS backend 2018-11-14 20:49:08 -08:00
Chris Robinson 83dec6b975 Use a C++ vector for enumerated OSS devices 2018-11-14 19:16:50 -08:00
Chris Robinson 3b1b029a75 Remove some now-unused function checks 2018-11-14 17:08:14 -08:00
Chris Robinson c4d3444a6d Use std::pow and std::log2 2018-11-14 17:03:04 -08:00
Chris Robinson 46301a087c Use C++ a bit more with alc.cpp 2018-11-14 06:17:47 -08:00
Chris Robinson 3021a426c0 Convert ALc.c to C++ 2018-11-14 04:15:44 -08:00
Chris Robinson dfcb6d3e6d More clearly check if the buffer is not empty 2018-11-14 02:41:21 -08:00
Chris Robinson b15dcea4bb Move extern inline declarations to their own C source 2018-11-14 00:07:50 -08:00
Chris Robinson 6ae217d005 Fix some comment indentation 2018-11-13 23:26:42 -08:00
Chris Robinson c3ee206129 Use an anonymous namespace in the OSS backend 2018-11-13 23:23:46 -08:00
Chris Robinson 6510a44dba Use C++ more with the ALSA backend 2018-11-13 23:07:23 -08:00
Chris Robinson a5f68c2121 Avoid using ATOMIC_FLAG
Although it cant potentially be better than a regular atomic, it presents
compatibility issues when non-C11 atomics are mixed with C++
2018-11-13 20:26:32 -08:00
Chris Robinson 5867c7b8c2 Don't bother inlining some functions 2018-11-13 19:45:26 -08:00
Chris Robinson 7088f4e34a Avoid calling through the vtable in the backends 2018-11-13 18:41:24 -08:00
Chris Robinson 0dfb805fa2 Use utf8_to_wstr to convert UTF-8 to wstring 2018-11-13 06:29:02 -08:00
Chris Robinson 6f635d3b1a Make ReadALConfig noexcept in C++ 2018-11-13 02:15:10 -08:00
Chris Robinson 2445bfd578 Convert the CoreAudio backend to C++ 2018-11-13 02:09:21 -08:00
Chris Robinson 0ff349a714 Convert the QSA backend to C++
This may very well not work, since there's no testing and my IDE is not able to
show real problems over the incompatibilities with ALSA headers.
2018-11-13 01:56:34 -08:00
Chris Robinson 09ea1d58f6 Convert the backend base to C++ 2018-11-13 01:39:42 -08:00
Chris Robinson 74d1337cc7 Convert the ALSA backend to C++ 2018-11-13 01:36:17 -08:00
Chris Robinson 1f34af4718 Convert the JACK backend to C++ 2018-11-13 01:20:15 -08:00
Chris Robinson 51a1310902 Convert the OpenSL backend to C++ 2018-11-13 00:33:25 -08:00
Chris Robinson c93b7ca0da Convert the OSS backend to C++ 2018-11-12 23:49:11 -08:00
Chris Robinson d9a47ab63c Convert the PortAudio backend to C++ 2018-11-12 23:32:11 -08:00
Chris Robinson dd9ccde055 Convert the SoundIO backend to C++ 2018-11-12 23:17:27 -08:00
Chris Robinson efae7bfb72 Convert the Solaris backend to C++ 2018-11-12 23:06:31 -08:00
Chris Robinson e7ab5053e4 Convert the SDL2 backend to C++ 2018-11-12 22:57:39 -08:00
Chris Robinson a6d780574d Make the enumerated HRTF entry name a char*
Would ideally be a std::string with the HRTF name itself, but they're still
seen in C code.
2018-11-12 22:26:12 -08:00
Chris Robinson d4d0b1fdd4 Use a regular char* for the HRTF string name 2018-11-12 19:02:38 -08:00
Chris Robinson c0f2858f3d Split Windows-specific SetRTPriority 2018-11-12 18:05:16 -08:00
Chris Robinson 4def2a60f2 Clean up helpers.cpp some 2018-11-11 23:13:11 -08:00
Chris Robinson c66db3cdf6 Use the correct type for __control87_2 2018-11-11 22:27:37 -08:00
Chris Robinson c23ea494ea Fix getting the process binary for FreeBSD or macOS 2018-11-11 22:19:32 -08:00
Chris Robinson 5848dab92d Make AmbDecConf::load noexcept
To ease the ovewrhead of destructors that call C (non-noexcept) functions.
2018-11-11 20:40:37 -08:00
Chris Robinson f8bda31c72 Remove unused wstr functions 2018-11-11 20:40:02 -08:00
Chris Robinson 51ed335833 Use C++ more with helpers.cpp 2018-11-11 19:17:40 -08:00
Chris Robinson 4793e5c4ae Use C++ for GetProcBinary 2018-11-11 16:09:24 -08:00
Chris Robinson 58a71a1a00 Convert helpers.c to C++ 2018-11-11 14:56:25 -08:00
Chris Robinson d3c4bab7bb Use a vector in the wave backend 2018-11-11 03:29:43 -08:00
Chris Robinson 2db82bea6f Define the examples' common library earlier 2018-11-11 00:47:57 -08:00
Chris Robinson f99b16daa9 Use C++ threads in the null and wave backends 2018-11-11 00:33:04 -08:00
Chris Robinson 5bc2918e16 Remove the unused condition variable APIs 2018-11-10 22:53:03 -08:00
Chris Robinson f3ce7bc7dc Move altimespec_get and al_nssleep to examples' common code 2018-11-10 21:09:54 -08:00
Chris Robinson 2f42f74418 Ensure ambdec parsing stops at unexpected EOF 2018-11-10 19:38:52 -08:00
Chris Robinson 3939878cc0 Use standard timing methods for the null and wave backends 2018-11-10 19:31:23 -08:00
Chris Robinson dc31969b04 Get rid of the last few al_fopen calls 2018-11-10 04:27:10 -08:00
Chris Robinson 58eb0e754d Load config files using C++
Specifically, avoid al_fopen
2018-11-10 03:42:18 -08:00
Chris Robinson 5ec644f859 Convert alconfig to C++ 2018-11-10 00:13:35 -08:00
Chris Robinson 981205de36 Remove unused strdupW 2018-11-10 00:01:34 -08:00
Chris Robinson 68bcf81756 Remove the unused file mapping calls 2018-11-09 23:51:29 -08:00
Chris Robinson d8163a416a Use C++ more in hrtf.cpp 2018-11-09 23:47:42 -08:00
Chris Robinson add776ddbb Handle the open mode in al::ifstream 2018-11-09 21:56:05 -08:00
Chris Robinson db56fd5e3d Convert hrtf.c to C++ 2018-11-09 18:08:42 -08:00
Chris Robinson e8679e7214 Convert the null backend to C++ 2018-11-09 03:46:30 -08:00
Chris Robinson 781ca7c58b Convert the loopback backend to C++ 2018-11-09 03:15:57 -08:00
Chris Robinson b327a50a15 Add missing consttructor and destructor calls for the wave backend 2018-11-09 02:55:28 -08:00
Chris Robinson b2cdfe58eb Convert the wave backend to C++ 2018-11-09 01:55:54 -08:00
Chris Robinson 55c860deec Use C++ with the winmm backend 2018-11-09 01:34:59 -08:00
Chris Robinson 75eeb6ba4b Convert the WinMM backend to C++ 2018-11-08 20:32:31 -08:00
Chris Robinson 34c836c490 Disambiguate operation order 2018-11-08 14:25:05 -08:00
Chris Robinson d71e50f7f7 Avoid using out parameters 2018-11-08 14:09:13 -08:00
Chris Robinson b27ccb8aa6 Fix a macro typo 2018-11-08 12:21:25 -08:00
Chris Robinson 29435ad966 Prevent MSVC's dumb min/max macros 2018-11-08 04:15:16 -08:00
kcat 22c8330200 Merge pull request #246 from john-preston/fix_macro
Fix Resample_bsinc_SSE pointer casts.
2018-11-08 03:50:37 -08:00
Chris Robinson 4eed3e9236 Move static functions to an anonymous namespace 2018-11-08 03:31:42 -08:00
John Preston c354ba2d2e Fix Resample_bsinc_SSE pointer casts.
Regression was introduced in 5ec11a017c.
2018-11-08 14:18:34 +04:00
Chris Robinson 41aa9845c4 Hide function pointer wrapper macros for IDE parsing 2018-11-07 18:26:33 -08:00
Chris Robinson 5e5e2f654e Disable MSVC warning C4200
"nonstandard extension used: zero-sized array in struct/union"
2018-11-06 21:45:40 -08:00
Chris Robinson 3cb0999a23 Use the proper enum values for atomic ops 2018-11-06 21:33:08 -08:00
Chris Robinson aeb3849904 Also extract the Android support includes 2018-11-06 20:04:28 -08:00
Chris Robinson 620ae6491b Handle CMake policy CMP0075 2018-11-06 19:43:14 -08:00
Chris Robinson 8954d3a438 Fix Android extraction paths 2018-11-06 19:34:19 -08:00
Chris Robinson 9d4af941b9 Update Travis build images
GCC 4.x is too problematic with C++11. Ubuntu Xenial has GCC 5 which has more
complete C++11 conformance. For Android, NDK r16 includes libc++ as an
alternative to the deprecated GCC 4.9's libstdc++.
2018-11-06 19:23:00 -08:00
Chris Robinson bf63c3d3d8 Don't set _FILE_OFFSET_BITS on Android 2018-11-06 19:15:09 -08:00
Chris Robinson 27fbccfb23 Don't directly declare standard function names 2018-11-06 19:14:17 -08:00
Chris Robinson d76e9800da Use a more appropriate type for the result of tellg 2018-11-06 18:55:00 -08:00
Chris Robinson 3b664041ba Convert the DSound backend to C++ 2018-11-06 02:17:20 -08:00
Chris Robinson 9fa31fcd07 Avoid moving istringstreams
Doesn't work with GCC 4.x. Hopefully swapping does.
2018-11-04 19:11:07 -08:00
Chris Robinson 4dafb7dab1 Use C++ to read and parse ambdec files 2018-11-04 15:24:24 -08:00
Chris Robinson 26f7007507 Allocate the appropriate amount in the aligned allocator 2018-11-04 15:21:03 -08:00
Chris Robinson 087fdd3ca9 Properly mark arrays constexpr and/or static 2018-11-04 15:19:48 -08:00
Chris Robinson 96819237d6 Convert ambdec.c to C++ 2018-11-03 19:51:23 -07:00
Chris Robinson 4bfaa173c4 Convert panning.c to C++ 2018-11-03 19:05:23 -07:00
Chris Robinson ba5ec8b074 Be more C++-friendly with the B-Format decoder 2018-11-03 18:00:05 -07:00
Chris Robinson 67f9efdad4 Convert the BFormat decoder to C++ 2018-11-03 15:32:09 -07:00
Chris Robinson 12d5b638f2 Convert the band-split filter to C++ 2018-11-03 15:23:15 -07:00
Chris Robinson 18e1d10338 Convert the UHJ encoder to C++ 2018-11-03 14:40:40 -07:00
Chris Robinson e9d17c5191 Move wstr_to_utf8 to compat.h 2018-11-03 12:55:02 -07:00
Chris Robinson 5482efc921 Make the polymorphic allocators allocate cleared memory 2018-11-02 18:48:08 -07:00
Chris Robinson a7dcc1c6d1 Expand the anonymous namespaces 2018-11-02 14:56:10 -07:00
Chris Robinson b54c4b02c7 Add a wrapper to manage PROPVARIANT objects 2018-11-02 13:32:42 -07:00
Chris Robinson 9e8e3c146f Workaround lack of roundf with early MSVC 2018-11-02 09:39:57 -07:00
Chris Robinson e20d2cdbce Use HUGE_VALF instead of INFINITY
Older MSVC lacks INFINITY, and we define a HUGE_VALF fallback when needed.
2018-11-02 09:25:19 -07:00
Chris Robinson 4ec757c1de Specify the correct array size for casting 2018-11-02 09:22:12 -07:00
Chris Robinson aaa31d987f Check the correct propvariant object 2018-11-02 00:42:50 -07:00
Chris Robinson d66664122a Try another fix to declare GUIDs in C++ 2018-11-02 00:13:07 -07:00
Chris Robinson 66df771d96 Make the polymorphism macros less hacky in C++
In particular, it relies on derived structs using C++-style inheritence. Any
implementation's source that's converted to C++ will consequently need to make
that change.
2018-11-01 23:52:53 -07:00
Chris Robinson 7307c2d5aa Workaround some issue with DEFINE_GUID in C++ 2018-11-01 21:42:26 -07:00
Chris Robinson a90b17113a Use more C++ types where possible 2018-11-01 21:39:34 -07:00
Chris Robinson 434582b8e3 Use an anonymous namespace instead of static for some things 2018-11-01 20:18:51 -07:00
Chris Robinson 1ca4e268f6 Preliminary conversion of the WASAPI backend to C++
A very sparse conversion. Will clean up more later after seeing what MSVC does.
2018-11-01 16:26:42 -07:00
Chris Robinson dee2905f4a Remove unused CXX_FLAGS variable
The Visual Studio generators apparently don't like the $<COMPILE_LANGUAGE:...>
expression. Since it's not actually used for anything at the moment, remove it.
2018-11-01 16:10:46 -07:00
Chris Robinson c0ce71a175 Fix a macro check 2018-11-01 16:00:38 -07:00
Chris Robinson 11967dc2da Use a wrapper function to simplify a check 2018-11-01 16:00:16 -07:00
Chris Robinson 69162cf9c6 Use perfect forwarding to initialize DevMap entries 2018-11-01 13:12:42 -07:00
Chris Robinson cd68530ab4 Simplify a couple loops 2018-11-01 11:44:11 -07:00
Chris Robinson 5d092a1c58 Use the appropriate enums for standard atomics 2018-11-01 08:43:31 -07:00
Chris Robinson d28c0eb556 Avoid uniform initialization with references
Also doesn't work with GCC 4.x
2018-11-01 08:41:23 -07:00
Chris Robinson 48f877e859 Remove unused header 2018-11-01 08:19:51 -07:00
Chris Robinson ccf356a03d Include a missing header for atomic 2018-11-01 08:17:23 -07:00
Chris Robinson 95966c4d92 Fix another use of auto uniform initialization 2018-11-01 08:15:47 -07:00
Chris Robinson 759c3a996c Avoid all uniform initialization with auto
Because of early C++11 (GCC 4.x) deficiencies, it's not interpreted correctly.
Either declare the type name explicitly with uniform initization, or use auto
with = initialization. It'll be fine when updating to GCC 5 or Clang 3.6.
2018-11-01 08:04:21 -07:00
Chris Robinson 353bb1ed17 Avoid uniform initialization on auto for integer types
To work around a deficiency with early C++11 compilers (GCC 4.x).
2018-10-31 22:10:26 -07:00
Chris Robinson d41fbd5c2d Convert the PulseAudio backend to C++ 2018-10-31 20:09:14 -07:00
Chris Robinson de275408fa Make a const array constexpr 2018-10-31 13:31:14 -07:00
Chris Robinson 0e7986eaa8 Move some extern inline declarations to C 2018-10-31 13:21:05 -07:00
Chris Robinson 9dba90fa20 Avoid assigning in an if statement 2018-10-31 13:00:08 -07:00
Chris Robinson 77bac8eeaa Add a missing <array> include 2018-10-31 12:01:07 -07:00
Chris Robinson 4d422dfb24 Fix some backup atomic macros 2018-10-31 12:00:02 -07:00
Chris Robinson 624bc1c839 Convert alBuffer.c to C++
A test to ensure everything works.
2018-10-31 10:37:16 -07:00
Chris Robinson 4b7af24ed5 Add specializations for lock_guard and unique_lock to take almtx_t 2018-10-31 10:35:12 -07:00
Chris Robinson 1e8c6df7b9 Add a C++ ContextRef helper to manage a ALCcontext reference 2018-10-31 10:05:15 -07:00
Chris Robinson da150572f9 Clean up the DriverIface in its destructor 2018-10-30 17:03:21 -07:00
Chris Robinson 615c025b67 Add a missing include for array 2018-10-30 16:35:14 -07:00
Chris Robinson 08aa9d898b Remove an unnecessary include 2018-10-30 16:32:25 -07:00
Chris Robinson 6a8f5e5950 Build the router with AppVeyor 2018-10-30 15:38:11 -07:00
Chris Robinson f747ac8882 Clean up the router's PtrIntMap 2018-10-30 14:28:19 -07:00
Chris Robinson 44f91760b5 Use std::array instead of raw arrays 2018-10-30 10:01:49 -07:00
Chris Robinson e2a1dd4503 Use std::wstring in place of some fixed WCHAR arrays 2018-10-30 09:31:52 -07:00
Chris Robinson ce212c911c Add a cmake option to static-link winpthread 2018-10-30 09:24:39 -07:00
Chris Robinson c17e59f63a Use std::vector instead of custom dynamic arrays 2018-10-30 08:33:40 -07:00
Chris Robinson e75e0a342e Use C++ atomics and mutexes in the router 2018-10-30 07:30:46 -07:00
Chris Robinson a0d03e50e8 Convert the router to C++ 2018-10-30 07:06:03 -07:00
Chris Robinson f17b930638 Add extern "C" for router.h 2018-10-30 06:45:44 -07:00
Chris Robinson 71303b73a4 Add a cmake option to static-link libstdc++ 2018-10-29 20:55:45 -07:00
Chris Robinson 3deb7c6ed5 Workaround C++ compatiility issues for atomic.h
This isn't wholly correct since neither C11 or C++11 guarantee compatibility
between atomic implementations. It's desired behavior and mostly works, see:

http://www.open-std.org/jtc1/sc22/wg21/docs/papers/2018/p0943r1.html

Alignment issues can be fixed with manual alignas() specifications, should the
need arise.
2018-10-29 20:50:57 -07:00
Chris Robinson 5fea511cce Fix some more headers for C++ 2018-10-29 11:51:41 -07:00
Chris Robinson 662e1c5cc2 Fix some uses of RESTRICT 2018-10-29 11:50:41 -07:00
Chris Robinson ea3789d985 Update version for AppVeyor 2018-10-29 11:35:43 -07:00
Chris Robinson 903d878460 Replace restrict with RESTRICT 2018-10-29 11:32:50 -07:00
Chris Robinson 184241f2ef Simplify a couple more checks 2018-10-29 10:17:30 -07:00
Chris Robinson e876b50113 Remove unnecessary uses of IN_IDE_PARSER 2018-10-29 10:10:37 -07:00
Chris Robinson 7d5a288e83 Add a couple casts for compiling with C++
Also avoid using __builtin_types_compatible_p, which seems broken with C++?
2018-10-29 10:10:37 -07:00
Chris Robinson 1d7b0f54be Add another missing extern "C" 2018-10-29 10:10:37 -07:00
Chris Robinson 8a8ab68f1c Add a more C++-friendly VECTOR_RESIZE 2018-10-29 10:10:37 -07:00
Chris Robinson fd1361c198 Add a RESTRICT macro to help with C++ compatibility 2018-10-29 10:10:37 -07:00
Chris Robinson 5148a73a7b Add missing extern "C" 2018-10-29 10:10:37 -07:00
Chris Robinson f1058c635f Handle C++-only compile flags in cmake 2018-10-29 10:10:37 -07:00
Chris Robinson 7f8ef092aa Fix a couple internal headers to compile with C++ 2018-10-29 10:10:37 -07:00
kcat 44a4602508 Merge pull request #240 from ShFil119/impr/simplify_statements
Simplify some statements
2018-10-29 10:10:06 -07:00
Filip Gawin 08226bc6b0 Simplify some statements 2018-10-29 13:38:58 +01:00
kcat 56b8b97642 Merge pull request #236 from alexey-lysiuk/macos_semaphore
Use GCD semaphore on macOS
2018-10-15 14:16:49 -07:00
alexey.lysiuk c8d866a25a Use GCD semaphore on macOS
Unnamed POSIX semaphore doesn't work on macOS
2018-10-15 19:41:22 +03:00
301 changed files with 53589 additions and 56887 deletions
+15 -13
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@@ -1,8 +1,8 @@
language: c
language: cpp
matrix:
include:
- os: linux
dist: trusty
dist: xenial
- os: linux
dist: trusty
env:
@@ -24,17 +24,18 @@ install:
fi
- >
if [[ "${TRAVIS_OS_NAME}" == "linux" && "${BUILD_ANDROID}" == "true" ]]; then
curl -o ~/android-ndk.zip https://dl.google.com/android/repository/android-ndk-r15-linux-x86_64.zip
curl -o ~/android-ndk.zip https://dl.google.com/android/repository/android-ndk-r16b-linux-x86_64.zip
unzip -q ~/android-ndk.zip -d ~ \
'android-ndk-r15/build/cmake/*' \
'android-ndk-r15/build/core/toolchains/arm-linux-androideabi-*/*' \
'android-ndk-r15/platforms/android-14/arch-arm/*' \
'android-ndk-r15/source.properties' \
'android-ndk-r15/sources/cxx-stl/gnu-libstdc++/4.9/libs/armeabi-v7a/*' \
'android-ndk-r15/sources/cxx-stl/gnu-libstdc++/4.9/include/*' \
'android-ndk-r15/sysroot/*' \
'android-ndk-r15/toolchains/arm-linux-androideabi-4.9/prebuilt/linux-x86_64/*' \
'android-ndk-r15/toolchains/llvm/prebuilt/linux-x86_64/*'
'android-ndk-r16b/build/cmake/*' \
'android-ndk-r16b/build/core/toolchains/arm-linux-androideabi-*/*' \
'android-ndk-r16b/platforms/android-14/arch-arm/*' \
'android-ndk-r16b/source.properties' \
'android-ndk-r16b/sources/android/support/include/*' \
'android-ndk-r16b/sources/cxx-stl/llvm-libc++/libs/armeabi-v7a/*' \
'android-ndk-r16b/sources/cxx-stl/llvm-libc++/include/*' \
'android-ndk-r16b/sysroot/*' \
'android-ndk-r16b/toolchains/arm-linux-androideabi-4.9/prebuilt/linux-x86_64/*' \
'android-ndk-r16b/toolchains/llvm/prebuilt/linux-x86_64/*'
fi
script:
- >
@@ -51,7 +52,8 @@ script:
- >
if [[ "${TRAVIS_OS_NAME}" == "linux" && "${BUILD_ANDROID}" == "true" ]]; then
cmake \
-DCMAKE_TOOLCHAIN_FILE=~/android-ndk-r15/build/cmake/android.toolchain.cmake \
-DANDROID_STL=c++_shared \
-DCMAKE_TOOLCHAIN_FILE=~/android-ndk-r16b/build/cmake/android.toolchain.cmake \
-DALSOFT_REQUIRE_OPENSL=ON \
-DALSOFT_EMBED_HRTF_DATA=YES \
.
-4703
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-1923
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-698
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@@ -1,698 +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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* 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>
#include <string.h>
#ifdef _WIN32_IE
#include <windows.h>
#include <shlobj.h>
#endif
#ifdef __APPLE__
#include <CoreFoundation/CoreFoundation.h>
#endif
#include "alMain.h"
#include "alconfig.h"
#include "compat.h"
#include "bool.h"
typedef struct ConfigEntry {
char *key;
char *value;
} ConfigEntry;
typedef struct ConfigBlock {
ConfigEntry *entries;
unsigned int entryCount;
} ConfigBlock;
static ConfigBlock cfgBlock;
static char *lstrip(char *line)
{
while(isspace(line[0]))
line++;
return line;
}
static char *rstrip(char *line)
{
size_t len = strlen(line);
while(len > 0 && isspace(line[len-1]))
len--;
line[len] = 0;
return line;
}
static int readline(FILE *f, char **output, size_t *maxlen)
{
size_t len = 0;
int c;
while((c=fgetc(f)) != EOF && (c == '\r' || c == '\n'))
;
if(c == EOF)
return 0;
do {
if(len+1 >= *maxlen)
{
void *temp = NULL;
size_t newmax;
newmax = (*maxlen ? (*maxlen)<<1 : 32);
if(newmax > *maxlen)
temp = realloc(*output, newmax);
if(!temp)
{
ERR("Failed to realloc "SZFMT" bytes from "SZFMT"!\n", newmax, *maxlen);
return 0;
}
*output = temp;
*maxlen = newmax;
}
(*output)[len++] = c;
(*output)[len] = '\0';
} while((c=fgetc(f)) != EOF && c != '\r' && c != '\n');
return 1;
}
static char *expdup(const char *str)
{
char *output = NULL;
size_t maxlen = 0;
size_t len = 0;
while(*str != '\0')
{
const char *addstr;
size_t addstrlen;
size_t i;
if(str[0] != '$')
{
const char *next = strchr(str, '$');
addstr = str;
addstrlen = next ? (size_t)(next-str) : strlen(str);
str += addstrlen;
}
else
{
str++;
if(*str == '$')
{
const char *next = strchr(str+1, '$');
addstr = str;
addstrlen = next ? (size_t)(next-str) : strlen(str);
str += addstrlen;
}
else
{
bool hasbraces;
char envname[1024];
size_t k = 0;
hasbraces = (*str == '{');
if(hasbraces) str++;
while((isalnum(*str) || *str == '_') && k < sizeof(envname)-1)
envname[k++] = *(str++);
envname[k++] = '\0';
if(hasbraces && *str != '}')
continue;
if(hasbraces) str++;
if((addstr=getenv(envname)) == NULL)
continue;
addstrlen = strlen(addstr);
}
}
if(addstrlen == 0)
continue;
if(addstrlen >= maxlen-len)
{
void *temp = NULL;
size_t newmax;
newmax = len+addstrlen+1;
if(newmax > maxlen)
temp = realloc(output, newmax);
if(!temp)
{
ERR("Failed to realloc "SZFMT" bytes from "SZFMT"!\n", newmax, maxlen);
return output;
}
output = temp;
maxlen = newmax;
}
for(i = 0;i < addstrlen;i++)
output[len++] = addstr[i];
output[len] = '\0';
}
return output ? output : calloc(1, 1);
}
static void LoadConfigFromFile(FILE *f)
{
char curSection[128] = "";
char *buffer = NULL;
size_t maxlen = 0;
ConfigEntry *ent;
while(readline(f, &buffer, &maxlen))
{
char *line, *comment;
char key[256] = "";
char value[256] = "";
line = rstrip(lstrip(buffer));
if(!line[0]) continue;
if(line[0] == '[')
{
char *section = line+1;
char *endsection;
endsection = strchr(section, ']');
if(!endsection || section == endsection)
{
ERR("config parse error: bad line \"%s\"\n", line);
continue;
}
if(endsection[1] != 0)
{
char *end = endsection+1;
while(isspace(*end))
++end;
if(*end != 0 && *end != '#')
{
ERR("config parse error: bad line \"%s\"\n", line);
continue;
}
}
*endsection = 0;
if(strcasecmp(section, "general") == 0)
curSection[0] = 0;
else
{
size_t len, p = 0;
do {
char *nextp = strchr(section, '%');
if(!nextp)
{
strncpy(curSection+p, section, sizeof(curSection)-1-p);
break;
}
len = nextp - section;
if(len > sizeof(curSection)-1-p)
len = sizeof(curSection)-1-p;
strncpy(curSection+p, section, len);
p += len;
section = nextp;
if(((section[1] >= '0' && section[1] <= '9') ||
(section[1] >= 'a' && section[1] <= 'f') ||
(section[1] >= 'A' && section[1] <= 'F')) &&
((section[2] >= '0' && section[2] <= '9') ||
(section[2] >= 'a' && section[2] <= 'f') ||
(section[2] >= 'A' && section[2] <= 'F')))
{
unsigned char b = 0;
if(section[1] >= '0' && section[1] <= '9')
b = (section[1]-'0') << 4;
else if(section[1] >= 'a' && section[1] <= 'f')
b = (section[1]-'a'+0xa) << 4;
else if(section[1] >= 'A' && section[1] <= 'F')
b = (section[1]-'A'+0x0a) << 4;
if(section[2] >= '0' && section[2] <= '9')
b |= (section[2]-'0');
else if(section[2] >= 'a' && section[2] <= 'f')
b |= (section[2]-'a'+0xa);
else if(section[2] >= 'A' && section[2] <= 'F')
b |= (section[2]-'A'+0x0a);
if(p < sizeof(curSection)-1)
curSection[p++] = b;
section += 3;
}
else if(section[1] == '%')
{
if(p < sizeof(curSection)-1)
curSection[p++] = '%';
section += 2;
}
else
{
if(p < sizeof(curSection)-1)
curSection[p++] = '%';
section += 1;
}
if(p < sizeof(curSection)-1)
curSection[p] = 0;
} while(p < sizeof(curSection)-1 && *section != 0);
curSection[sizeof(curSection)-1] = 0;
}
continue;
}
comment = strchr(line, '#');
if(comment) *(comment++) = 0;
if(!line[0]) continue;
if(sscanf(line, "%255[^=] = \"%255[^\"]\"", key, value) == 2 ||
sscanf(line, "%255[^=] = '%255[^\']'", key, value) == 2 ||
sscanf(line, "%255[^=] = %255[^\n]", key, value) == 2)
{
/* sscanf doesn't handle '' or "" as empty values, so clip it
* manually. */
if(strcmp(value, "\"\"") == 0 || strcmp(value, "''") == 0)
value[0] = 0;
}
else if(sscanf(line, "%255[^=] %255[=]", key, value) == 2)
{
/* Special case for 'key =' */
value[0] = 0;
}
else
{
ERR("config parse error: malformed option line: \"%s\"\n\n", line);
continue;
}
rstrip(key);
if(curSection[0] != 0)
{
size_t len = strlen(curSection);
memmove(&key[len+1], key, sizeof(key)-1-len);
key[len] = '/';
memcpy(key, curSection, len);
}
/* Check if we already have this option set */
ent = cfgBlock.entries;
while((unsigned int)(ent-cfgBlock.entries) < cfgBlock.entryCount)
{
if(strcasecmp(ent->key, key) == 0)
break;
ent++;
}
if((unsigned int)(ent-cfgBlock.entries) >= cfgBlock.entryCount)
{
/* Allocate a new option entry */
ent = realloc(cfgBlock.entries, (cfgBlock.entryCount+1)*sizeof(ConfigEntry));
if(!ent)
{
ERR("config parse error: error reallocating config entries\n");
continue;
}
cfgBlock.entries = ent;
ent = cfgBlock.entries + cfgBlock.entryCount;
cfgBlock.entryCount++;
ent->key = strdup(key);
ent->value = NULL;
}
free(ent->value);
ent->value = expdup(value);
TRACE("found '%s' = '%s'\n", ent->key, ent->value);
}
free(buffer);
}
#ifdef _WIN32
void ReadALConfig(void)
{
al_string ppath = AL_STRING_INIT_STATIC();
WCHAR buffer[MAX_PATH];
const WCHAR *str;
FILE *f;
if(SHGetSpecialFolderPathW(NULL, buffer, CSIDL_APPDATA, FALSE) != FALSE)
{
al_string filepath = AL_STRING_INIT_STATIC();
alstr_copy_wcstr(&filepath, buffer);
alstr_append_cstr(&filepath, "\\alsoft.ini");
TRACE("Loading config %s...\n", alstr_get_cstr(filepath));
f = al_fopen(alstr_get_cstr(filepath), "rt");
if(f)
{
LoadConfigFromFile(f);
fclose(f);
}
alstr_reset(&filepath);
}
GetProcBinary(&ppath, NULL);
if(!alstr_empty(ppath))
{
alstr_append_cstr(&ppath, "\\alsoft.ini");
TRACE("Loading config %s...\n", alstr_get_cstr(ppath));
f = al_fopen(alstr_get_cstr(ppath), "r");
if(f)
{
LoadConfigFromFile(f);
fclose(f);
}
}
if((str=_wgetenv(L"ALSOFT_CONF")) != NULL && *str)
{
al_string filepath = AL_STRING_INIT_STATIC();
alstr_copy_wcstr(&filepath, str);
TRACE("Loading config %s...\n", alstr_get_cstr(filepath));
f = al_fopen(alstr_get_cstr(filepath), "rt");
if(f)
{
LoadConfigFromFile(f);
fclose(f);
}
alstr_reset(&filepath);
}
alstr_reset(&ppath);
}
#else
void ReadALConfig(void)
{
al_string confpaths = AL_STRING_INIT_STATIC();
al_string fname = AL_STRING_INIT_STATIC();
const char *str;
FILE *f;
str = "/etc/openal/alsoft.conf";
TRACE("Loading config %s...\n", str);
f = al_fopen(str, "r");
if(f)
{
LoadConfigFromFile(f);
fclose(f);
}
if(!(str=getenv("XDG_CONFIG_DIRS")) || str[0] == 0)
str = "/etc/xdg";
alstr_copy_cstr(&confpaths, str);
/* Go through the list in reverse, since "the order of base directories
* denotes their importance; the first directory listed is the most
* important". Ergo, we need to load the settings from the later dirs
* first so that the settings in the earlier dirs override them.
*/
while(!alstr_empty(confpaths))
{
char *next = strrchr(alstr_get_cstr(confpaths), ':');
if(next)
{
size_t len = next - alstr_get_cstr(confpaths);
alstr_copy_cstr(&fname, next+1);
VECTOR_RESIZE(confpaths, len, len+1);
VECTOR_ELEM(confpaths, len) = 0;
}
else
{
alstr_reset(&fname);
fname = confpaths;
AL_STRING_INIT(confpaths);
}
if(alstr_empty(fname) || VECTOR_FRONT(fname) != '/')
WARN("Ignoring XDG config dir: %s\n", alstr_get_cstr(fname));
else
{
if(VECTOR_BACK(fname) != '/') alstr_append_cstr(&fname, "/alsoft.conf");
else alstr_append_cstr(&fname, "alsoft.conf");
TRACE("Loading config %s...\n", alstr_get_cstr(fname));
f = al_fopen(alstr_get_cstr(fname), "r");
if(f)
{
LoadConfigFromFile(f);
fclose(f);
}
}
alstr_clear(&fname);
}
#ifdef __APPLE__
CFBundleRef mainBundle = CFBundleGetMainBundle();
if(mainBundle)
{
unsigned char fileName[PATH_MAX];
CFURLRef configURL;
if((configURL=CFBundleCopyResourceURL(mainBundle, CFSTR(".alsoftrc"), CFSTR(""), NULL)) &&
CFURLGetFileSystemRepresentation(configURL, true, fileName, sizeof(fileName)))
{
f = al_fopen((const char*)fileName, "r");
if(f)
{
LoadConfigFromFile(f);
fclose(f);
}
}
}
#endif
if((str=getenv("HOME")) != NULL && *str)
{
alstr_copy_cstr(&fname, str);
if(VECTOR_BACK(fname) != '/') alstr_append_cstr(&fname, "/.alsoftrc");
else alstr_append_cstr(&fname, ".alsoftrc");
TRACE("Loading config %s...\n", alstr_get_cstr(fname));
f = al_fopen(alstr_get_cstr(fname), "r");
if(f)
{
LoadConfigFromFile(f);
fclose(f);
}
}
if((str=getenv("XDG_CONFIG_HOME")) != NULL && str[0] != 0)
{
alstr_copy_cstr(&fname, str);
if(VECTOR_BACK(fname) != '/') alstr_append_cstr(&fname, "/alsoft.conf");
else alstr_append_cstr(&fname, "alsoft.conf");
}
else
{
alstr_clear(&fname);
if((str=getenv("HOME")) != NULL && str[0] != 0)
{
alstr_copy_cstr(&fname, str);
if(VECTOR_BACK(fname) != '/') alstr_append_cstr(&fname, "/.config/alsoft.conf");
else alstr_append_cstr(&fname, ".config/alsoft.conf");
}
}
if(!alstr_empty(fname))
{
TRACE("Loading config %s...\n", alstr_get_cstr(fname));
f = al_fopen(alstr_get_cstr(fname), "r");
if(f)
{
LoadConfigFromFile(f);
fclose(f);
}
}
alstr_clear(&fname);
GetProcBinary(&fname, NULL);
if(!alstr_empty(fname))
{
if(VECTOR_BACK(fname) != '/') alstr_append_cstr(&fname, "/alsoft.conf");
else alstr_append_cstr(&fname, "alsoft.conf");
TRACE("Loading config %s...\n", alstr_get_cstr(fname));
f = al_fopen(alstr_get_cstr(fname), "r");
if(f)
{
LoadConfigFromFile(f);
fclose(f);
}
}
if((str=getenv("ALSOFT_CONF")) != NULL && *str)
{
TRACE("Loading config %s...\n", str);
f = al_fopen(str, "r");
if(f)
{
LoadConfigFromFile(f);
fclose(f);
}
}
alstr_reset(&fname);
alstr_reset(&confpaths);
}
#endif
void FreeALConfig(void)
{
unsigned int i;
for(i = 0;i < cfgBlock.entryCount;i++)
{
free(cfgBlock.entries[i].key);
free(cfgBlock.entries[i].value);
}
free(cfgBlock.entries);
}
const char *GetConfigValue(const char *devName, const char *blockName, const char *keyName, const char *def)
{
unsigned int i;
char key[256];
if(!keyName)
return def;
if(blockName && strcasecmp(blockName, "general") != 0)
{
if(devName)
snprintf(key, sizeof(key), "%s/%s/%s", blockName, devName, keyName);
else
snprintf(key, sizeof(key), "%s/%s", blockName, keyName);
}
else
{
if(devName)
snprintf(key, sizeof(key), "%s/%s", devName, keyName);
else
{
strncpy(key, keyName, sizeof(key)-1);
key[sizeof(key)-1] = 0;
}
}
for(i = 0;i < cfgBlock.entryCount;i++)
{
if(strcmp(cfgBlock.entries[i].key, key) == 0)
{
TRACE("Found %s = \"%s\"\n", key, cfgBlock.entries[i].value);
if(cfgBlock.entries[i].value[0])
return cfgBlock.entries[i].value;
return def;
}
}
if(!devName)
{
TRACE("Key %s not found\n", key);
return def;
}
return GetConfigValue(NULL, blockName, keyName, def);
}
int ConfigValueExists(const char *devName, const char *blockName, const char *keyName)
{
const char *val = GetConfigValue(devName, blockName, keyName, "");
return !!val[0];
}
int ConfigValueStr(const char *devName, const char *blockName, const char *keyName, const char **ret)
{
const char *val = GetConfigValue(devName, blockName, keyName, "");
if(!val[0]) return 0;
*ret = val;
return 1;
}
int ConfigValueInt(const char *devName, const char *blockName, const char *keyName, int *ret)
{
const char *val = GetConfigValue(devName, blockName, keyName, "");
if(!val[0]) return 0;
*ret = strtol(val, NULL, 0);
return 1;
}
int ConfigValueUInt(const char *devName, const char *blockName, const char *keyName, unsigned int *ret)
{
const char *val = GetConfigValue(devName, blockName, keyName, "");
if(!val[0]) return 0;
*ret = strtoul(val, NULL, 0);
return 1;
}
int ConfigValueFloat(const char *devName, const char *blockName, const char *keyName, float *ret)
{
const char *val = GetConfigValue(devName, blockName, keyName, "");
if(!val[0]) return 0;
#ifdef HAVE_STRTOF
*ret = strtof(val, NULL);
#else
*ret = (float)strtod(val, NULL);
#endif
return 1;
}
int ConfigValueBool(const char *devName, const char *blockName, const char *keyName, int *ret)
{
const char *val = GetConfigValue(devName, blockName, keyName, "");
if(!val[0]) return 0;
*ret = (strcasecmp(val, "true") == 0 || strcasecmp(val, "yes") == 0 ||
strcasecmp(val, "on") == 0 || atoi(val) != 0);
return 1;
}
int GetConfigValueBool(const char *devName, const char *blockName, const char *keyName, int def)
{
const char *val = GetConfigValue(devName, blockName, keyName, "");
if(!val[0]) return !!def;
return (strcasecmp(val, "true") == 0 || strcasecmp(val, "yes") == 0 ||
strcasecmp(val, "on") == 0 || atoi(val) != 0);
}
-17
View File
@@ -1,17 +0,0 @@
#ifndef ALCONFIG_H
#define ALCONFIG_H
void ReadALConfig(void);
void FreeALConfig(void);
int ConfigValueExists(const char *devName, const char *blockName, const char *keyName);
const char *GetConfigValue(const char *devName, const char *blockName, const char *keyName, const char *def);
int GetConfigValueBool(const char *devName, const char *blockName, const char *keyName, int def);
int ConfigValueStr(const char *devName, const char *blockName, const char *keyName, const char **ret);
int ConfigValueInt(const char *devName, const char *blockName, const char *keyName, int *ret);
int ConfigValueUInt(const char *devName, const char *blockName, const char *keyName, unsigned int *ret);
int ConfigValueFloat(const char *devName, const char *blockName, const char *keyName, float *ret);
int ConfigValueBool(const char *devName, const char *blockName, const char *keyName, int *ret);
#endif /* ALCONFIG_H */
-58
View File
@@ -1,58 +0,0 @@
#ifndef ALSTRING_H
#define ALSTRING_H
#include <string.h>
#include "vector.h"
#ifdef __cplusplus
extern "C" {
#endif
typedef char al_string_char_type;
TYPEDEF_VECTOR(al_string_char_type, al_string)
TYPEDEF_VECTOR(al_string, vector_al_string)
inline void alstr_reset(al_string *str)
{ VECTOR_DEINIT(*str); }
#define AL_STRING_INIT(_x) do { (_x) = (al_string)NULL; } while(0)
#define AL_STRING_INIT_STATIC() ((al_string)NULL)
#define AL_STRING_DEINIT(_x) alstr_reset(&(_x))
inline size_t alstr_length(const_al_string str)
{ return VECTOR_SIZE(str); }
inline ALboolean alstr_empty(const_al_string str)
{ return alstr_length(str) == 0; }
inline const al_string_char_type *alstr_get_cstr(const_al_string str)
{ return str ? &VECTOR_FRONT(str) : ""; }
void alstr_clear(al_string *str);
int alstr_cmp(const_al_string str1, const_al_string str2);
int alstr_cmp_cstr(const_al_string str1, const al_string_char_type *str2);
void alstr_copy(al_string *str, const_al_string from);
void alstr_copy_cstr(al_string *str, const al_string_char_type *from);
void alstr_copy_range(al_string *str, const al_string_char_type *from, const al_string_char_type *to);
void alstr_append_char(al_string *str, const al_string_char_type c);
void alstr_append_cstr(al_string *str, const al_string_char_type *from);
void alstr_append_range(al_string *str, const al_string_char_type *from, const al_string_char_type *to);
#ifdef _WIN32
#include <wchar.h>
/* Windows-only methods to deal with WideChar strings. */
void alstr_copy_wcstr(al_string *str, const wchar_t *from);
void alstr_append_wcstr(al_string *str, const wchar_t *from);
void alstr_copy_wrange(al_string *str, const wchar_t *from, const wchar_t *to);
void alstr_append_wrange(al_string *str, const wchar_t *from, const wchar_t *to);
#endif
#ifdef __cplusplus
} /* extern "C" */
#endif
#endif /* ALSTRING_H */
-566
View File
@@ -1,566 +0,0 @@
#include "config.h"
#include "ambdec.h"
#include <stdio.h>
#include <string.h>
#include <ctype.h>
#include "compat.h"
static char *lstrip(char *line)
{
while(isspace(line[0]))
line++;
return line;
}
static char *rstrip(char *line)
{
size_t len = strlen(line);
while(len > 0 && isspace(line[len-1]))
len--;
line[len] = 0;
return line;
}
static int readline(FILE *f, char **output, size_t *maxlen)
{
size_t len = 0;
int c;
while((c=fgetc(f)) != EOF && (c == '\r' || c == '\n'))
;
if(c == EOF)
return 0;
do {
if(len+1 >= *maxlen)
{
void *temp = NULL;
size_t newmax;
newmax = (*maxlen ? (*maxlen)<<1 : 32);
if(newmax > *maxlen)
temp = realloc(*output, newmax);
if(!temp)
{
ERR("Failed to realloc "SZFMT" bytes from "SZFMT"!\n", newmax, *maxlen);
return 0;
}
*output = temp;
*maxlen = newmax;
}
(*output)[len++] = c;
(*output)[len] = '\0';
} while((c=fgetc(f)) != EOF && c != '\r' && c != '\n');
return 1;
}
/* Custom strtok_r, since we can't rely on it existing. */
static char *my_strtok_r(char *str, const char *delim, char **saveptr)
{
/* Sanity check and update internal pointer. */
if(!saveptr || !delim) return NULL;
if(str) *saveptr = str;
str = *saveptr;
/* Nothing more to do with this string. */
if(!str) return NULL;
/* Find the first non-delimiter character. */
while(*str != '\0' && strchr(delim, *str) != NULL)
str++;
if(*str == '\0')
{
/* End of string. */
*saveptr = NULL;
return NULL;
}
/* Find the next delimiter character. */
*saveptr = strpbrk(str, delim);
if(*saveptr) *((*saveptr)++) = '\0';
return str;
}
static char *read_int(ALint *num, const char *line, int base)
{
char *end;
*num = strtol(line, &end, base);
if(end && *end != '\0')
end = lstrip(end);
return end;
}
static char *read_uint(ALuint *num, const char *line, int base)
{
char *end;
*num = strtoul(line, &end, base);
if(end && *end != '\0')
end = lstrip(end);
return end;
}
static char *read_float(ALfloat *num, const char *line)
{
char *end;
#ifdef HAVE_STRTOF
*num = strtof(line, &end);
#else
*num = (ALfloat)strtod(line, &end);
#endif
if(end && *end != '\0')
end = lstrip(end);
return end;
}
char *read_clipped_line(FILE *f, char **buffer, size_t *maxlen)
{
while(readline(f, buffer, maxlen))
{
char *line, *comment;
line = lstrip(*buffer);
comment = strchr(line, '#');
if(comment) *(comment++) = 0;
line = rstrip(line);
if(line[0]) return line;
}
return NULL;
}
static int load_ambdec_speakers(AmbDecConf *conf, FILE *f, char **buffer, size_t *maxlen, char **saveptr)
{
ALsizei cur = 0;
while(cur < conf->NumSpeakers)
{
const char *cmd = my_strtok_r(NULL, " \t", saveptr);
if(!cmd)
{
char *line = read_clipped_line(f, buffer, maxlen);
if(!line)
{
ERR("Unexpected end of file\n");
return 0;
}
cmd = my_strtok_r(line, " \t", saveptr);
}
if(strcmp(cmd, "add_spkr") == 0)
{
const char *name = my_strtok_r(NULL, " \t", saveptr);
const char *dist = my_strtok_r(NULL, " \t", saveptr);
const char *az = my_strtok_r(NULL, " \t", saveptr);
const char *elev = my_strtok_r(NULL, " \t", saveptr);
const char *conn = my_strtok_r(NULL, " \t", saveptr);
if(!name) WARN("Name not specified for speaker %u\n", cur+1);
else alstr_copy_cstr(&conf->Speakers[cur].Name, name);
if(!dist) WARN("Distance not specified for speaker %u\n", cur+1);
else read_float(&conf->Speakers[cur].Distance, dist);
if(!az) WARN("Azimuth not specified for speaker %u\n", cur+1);
else read_float(&conf->Speakers[cur].Azimuth, az);
if(!elev) WARN("Elevation not specified for speaker %u\n", cur+1);
else read_float(&conf->Speakers[cur].Elevation, elev);
if(!conn) TRACE("Connection not specified for speaker %u\n", cur+1);
else alstr_copy_cstr(&conf->Speakers[cur].Connection, conn);
cur++;
}
else
{
ERR("Unexpected speakers command: %s\n", cmd);
return 0;
}
cmd = my_strtok_r(NULL, " \t", saveptr);
if(cmd)
{
ERR("Unexpected junk on line: %s\n", cmd);
return 0;
}
}
return 1;
}
static int load_ambdec_matrix(ALfloat *gains, ALfloat (*matrix)[MAX_AMBI_COEFFS], ALsizei maxrow, FILE *f, char **buffer, size_t *maxlen, char **saveptr)
{
int gotgains = 0;
ALsizei cur = 0;
while(cur < maxrow)
{
const char *cmd = my_strtok_r(NULL, " \t", saveptr);
if(!cmd)
{
char *line = read_clipped_line(f, buffer, maxlen);
if(!line)
{
ERR("Unexpected end of file\n");
return 0;
}
cmd = my_strtok_r(line, " \t", saveptr);
}
if(strcmp(cmd, "order_gain") == 0)
{
ALuint curgain = 0;
char *line;
while((line=my_strtok_r(NULL, " \t", saveptr)) != NULL)
{
ALfloat value;
line = read_float(&value, line);
if(line && *line != '\0')
{
ERR("Extra junk on gain %u: %s\n", curgain+1, line);
return 0;
}
if(curgain < MAX_AMBI_ORDER+1)
gains[curgain] = value;
curgain++;
}
while(curgain < MAX_AMBI_ORDER+1)
gains[curgain++] = 0.0f;
gotgains = 1;
}
else if(strcmp(cmd, "add_row") == 0)
{
ALuint curidx = 0;
char *line;
while((line=my_strtok_r(NULL, " \t", saveptr)) != NULL)
{
ALfloat value;
line = read_float(&value, line);
if(line && *line != '\0')
{
ERR("Extra junk on matrix element %ux%u: %s\n", cur, curidx, line);
return 0;
}
if(curidx < MAX_AMBI_COEFFS)
matrix[cur][curidx] = value;
curidx++;
}
while(curidx < MAX_AMBI_COEFFS)
matrix[cur][curidx++] = 0.0f;
cur++;
}
else
{
ERR("Unexpected speakers command: %s\n", cmd);
return 0;
}
cmd = my_strtok_r(NULL, " \t", saveptr);
if(cmd)
{
ERR("Unexpected junk on line: %s\n", cmd);
return 0;
}
}
if(!gotgains)
{
ERR("Matrix order_gain not specified\n");
return 0;
}
return 1;
}
void ambdec_init(AmbDecConf *conf)
{
ALsizei i;
memset(conf, 0, sizeof(*conf));
AL_STRING_INIT(conf->Description);
for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
{
AL_STRING_INIT(conf->Speakers[i].Name);
AL_STRING_INIT(conf->Speakers[i].Connection);
}
}
void ambdec_deinit(AmbDecConf *conf)
{
ALsizei i;
alstr_reset(&conf->Description);
for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
{
alstr_reset(&conf->Speakers[i].Name);
alstr_reset(&conf->Speakers[i].Connection);
}
memset(conf, 0, sizeof(*conf));
}
int ambdec_load(AmbDecConf *conf, const char *fname)
{
char *buffer = NULL;
size_t maxlen = 0;
char *line;
FILE *f;
f = al_fopen(fname, "r");
if(!f)
{
ERR("Failed to open: %s\n", fname);
return 0;
}
while((line=read_clipped_line(f, &buffer, &maxlen)) != NULL)
{
char *saveptr;
char *command;
command = my_strtok_r(line, "/ \t", &saveptr);
if(!command)
{
ERR("Malformed line: %s\n", line);
goto fail;
}
if(strcmp(command, "description") == 0)
{
char *value = my_strtok_r(NULL, "", &saveptr);
alstr_copy_cstr(&conf->Description, lstrip(value));
}
else if(strcmp(command, "version") == 0)
{
line = my_strtok_r(NULL, "", &saveptr);
line = read_uint(&conf->Version, line, 10);
if(line && *line != '\0')
{
ERR("Extra junk after version: %s\n", line);
goto fail;
}
if(conf->Version != 3)
{
ERR("Unsupported version: %u\n", conf->Version);
goto fail;
}
}
else if(strcmp(command, "dec") == 0)
{
const char *dec = my_strtok_r(NULL, "/ \t", &saveptr);
if(strcmp(dec, "chan_mask") == 0)
{
line = my_strtok_r(NULL, "", &saveptr);
line = read_uint(&conf->ChanMask, line, 16);
if(line && *line != '\0')
{
ERR("Extra junk after mask: %s\n", line);
goto fail;
}
}
else if(strcmp(dec, "freq_bands") == 0)
{
line = my_strtok_r(NULL, "", &saveptr);
line = read_uint(&conf->FreqBands, line, 10);
if(line && *line != '\0')
{
ERR("Extra junk after freq_bands: %s\n", line);
goto fail;
}
if(conf->FreqBands != 1 && conf->FreqBands != 2)
{
ERR("Invalid freq_bands value: %u\n", conf->FreqBands);
goto fail;
}
}
else if(strcmp(dec, "speakers") == 0)
{
line = my_strtok_r(NULL, "", &saveptr);
line = read_int(&conf->NumSpeakers, line, 10);
if(line && *line != '\0')
{
ERR("Extra junk after speakers: %s\n", line);
goto fail;
}
if(conf->NumSpeakers > MAX_OUTPUT_CHANNELS)
{
ERR("Unsupported speaker count: %u\n", conf->NumSpeakers);
goto fail;
}
}
else if(strcmp(dec, "coeff_scale") == 0)
{
line = my_strtok_r(NULL, " \t", &saveptr);
if(strcmp(line, "n3d") == 0)
conf->CoeffScale = ADS_N3D;
else if(strcmp(line, "sn3d") == 0)
conf->CoeffScale = ADS_SN3D;
else if(strcmp(line, "fuma") == 0)
conf->CoeffScale = ADS_FuMa;
else
{
ERR("Unsupported coeff scale: %s\n", line);
goto fail;
}
}
else
{
ERR("Unexpected /dec option: %s\n", dec);
goto fail;
}
}
else if(strcmp(command, "opt") == 0)
{
const char *opt = my_strtok_r(NULL, "/ \t", &saveptr);
if(strcmp(opt, "xover_freq") == 0)
{
line = my_strtok_r(NULL, "", &saveptr);
line = read_float(&conf->XOverFreq, line);
if(line && *line != '\0')
{
ERR("Extra junk after xover_freq: %s\n", line);
goto fail;
}
}
else if(strcmp(opt, "xover_ratio") == 0)
{
line = my_strtok_r(NULL, "", &saveptr);
line = read_float(&conf->XOverRatio, line);
if(line && *line != '\0')
{
ERR("Extra junk after xover_ratio: %s\n", line);
goto fail;
}
}
else if(strcmp(opt, "input_scale") == 0 || strcmp(opt, "nfeff_comp") == 0 ||
strcmp(opt, "delay_comp") == 0 || strcmp(opt, "level_comp") == 0)
{
/* Unused */
my_strtok_r(NULL, " \t", &saveptr);
}
else
{
ERR("Unexpected /opt option: %s\n", opt);
goto fail;
}
}
else if(strcmp(command, "speakers") == 0)
{
const char *value = my_strtok_r(NULL, "/ \t", &saveptr);
if(strcmp(value, "{") != 0)
{
ERR("Expected { after %s command, got %s\n", command, value);
goto fail;
}
if(!load_ambdec_speakers(conf, f, &buffer, &maxlen, &saveptr))
goto fail;
value = my_strtok_r(NULL, "/ \t", &saveptr);
if(!value)
{
line = read_clipped_line(f, &buffer, &maxlen);
if(!line)
{
ERR("Unexpected end of file\n");
goto fail;
}
value = my_strtok_r(line, "/ \t", &saveptr);
}
if(strcmp(value, "}") != 0)
{
ERR("Expected } after speaker definitions, got %s\n", value);
goto fail;
}
}
else if(strcmp(command, "lfmatrix") == 0 || strcmp(command, "hfmatrix") == 0 ||
strcmp(command, "matrix") == 0)
{
const char *value = my_strtok_r(NULL, "/ \t", &saveptr);
if(strcmp(value, "{") != 0)
{
ERR("Expected { after %s command, got %s\n", command, value);
goto fail;
}
if(conf->FreqBands == 1)
{
if(strcmp(command, "matrix") != 0)
{
ERR("Unexpected \"%s\" type for a single-band decoder\n", command);
goto fail;
}
if(!load_ambdec_matrix(conf->HFOrderGain, conf->HFMatrix, conf->NumSpeakers,
f, &buffer, &maxlen, &saveptr))
goto fail;
}
else
{
if(strcmp(command, "lfmatrix") == 0)
{
if(!load_ambdec_matrix(conf->LFOrderGain, conf->LFMatrix, conf->NumSpeakers,
f, &buffer, &maxlen, &saveptr))
goto fail;
}
else if(strcmp(command, "hfmatrix") == 0)
{
if(!load_ambdec_matrix(conf->HFOrderGain, conf->HFMatrix, conf->NumSpeakers,
f, &buffer, &maxlen, &saveptr))
goto fail;
}
else
{
ERR("Unexpected \"%s\" type for a dual-band decoder\n", command);
goto fail;
}
}
value = my_strtok_r(NULL, "/ \t", &saveptr);
if(!value)
{
line = read_clipped_line(f, &buffer, &maxlen);
if(!line)
{
ERR("Unexpected end of file\n");
goto fail;
}
value = my_strtok_r(line, "/ \t", &saveptr);
}
if(strcmp(value, "}") != 0)
{
ERR("Expected } after matrix definitions, got %s\n", value);
goto fail;
}
}
else if(strcmp(command, "end") == 0)
{
line = my_strtok_r(NULL, "/ \t", &saveptr);
if(line)
{
ERR("Unexpected junk on end: %s\n", line);
goto fail;
}
fclose(f);
free(buffer);
return 1;
}
else
{
ERR("Unexpected command: %s\n", command);
goto fail;
}
line = my_strtok_r(NULL, "/ \t", &saveptr);
if(line)
{
ERR("Unexpected junk on line: %s\n", line);
goto fail;
}
}
ERR("Unexpected end of file\n");
fail:
fclose(f);
free(buffer);
return 0;
}
-46
View File
@@ -1,46 +0,0 @@
#ifndef AMBDEC_H
#define AMBDEC_H
#include "alstring.h"
#include "alMain.h"
/* Helpers to read .ambdec configuration files. */
enum AmbDecScaleType {
ADS_N3D,
ADS_SN3D,
ADS_FuMa,
};
typedef struct AmbDecConf {
al_string Description;
ALuint Version; /* Must be 3 */
ALuint ChanMask;
ALuint FreqBands; /* Must be 1 or 2 */
ALsizei NumSpeakers;
enum AmbDecScaleType CoeffScale;
ALfloat XOverFreq;
ALfloat XOverRatio;
struct {
al_string Name;
ALfloat Distance;
ALfloat Azimuth;
ALfloat Elevation;
al_string Connection;
} Speakers[MAX_OUTPUT_CHANNELS];
/* Unused when FreqBands == 1 */
ALfloat LFOrderGain[MAX_AMBI_ORDER+1];
ALfloat LFMatrix[MAX_OUTPUT_CHANNELS][MAX_AMBI_COEFFS];
ALfloat HFOrderGain[MAX_AMBI_ORDER+1];
ALfloat HFMatrix[MAX_OUTPUT_CHANNELS][MAX_AMBI_COEFFS];
} AmbDecConf;
void ambdec_init(AmbDecConf *conf);
void ambdec_deinit(AmbDecConf *conf);
int ambdec_load(AmbDecConf *conf, const char *fname);
#endif /* AMBDEC_H */
-1464
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File diff suppressed because it is too large Load Diff
-84
View File
@@ -1,84 +0,0 @@
#include "config.h"
#include <stdlib.h>
#include "alMain.h"
#include "alu.h"
#include "backends/base.h"
extern inline ALuint64 GetDeviceClockTime(ALCdevice *device);
extern inline void ALCdevice_Lock(ALCdevice *device);
extern inline void ALCdevice_Unlock(ALCdevice *device);
extern inline ClockLatency GetClockLatency(ALCdevice *device);
/* Base ALCbackend method implementations. */
void ALCbackend_Construct(ALCbackend *self, ALCdevice *device)
{
int ret = almtx_init(&self->mMutex, almtx_recursive);
assert(ret == althrd_success);
self->mDevice = device;
}
void ALCbackend_Destruct(ALCbackend *self)
{
almtx_destroy(&self->mMutex);
}
ALCboolean ALCbackend_reset(ALCbackend* UNUSED(self))
{
return ALC_FALSE;
}
ALCenum ALCbackend_captureSamples(ALCbackend* UNUSED(self), void* UNUSED(buffer), ALCuint UNUSED(samples))
{
return ALC_INVALID_DEVICE;
}
ALCuint ALCbackend_availableSamples(ALCbackend* UNUSED(self))
{
return 0;
}
ClockLatency ALCbackend_getClockLatency(ALCbackend *self)
{
ALCdevice *device = self->mDevice;
ALuint refcount;
ClockLatency ret;
do {
while(((refcount=ATOMIC_LOAD(&device->MixCount, almemory_order_acquire))&1))
althrd_yield();
ret.ClockTime = GetDeviceClockTime(device);
ATOMIC_THREAD_FENCE(almemory_order_acquire);
} while(refcount != ATOMIC_LOAD(&device->MixCount, almemory_order_relaxed));
/* NOTE: The device will generally have about all but one periods filled at
* any given time during playback. Without a more accurate measurement from
* the output, this is an okay approximation.
*/
ret.Latency = device->UpdateSize * DEVICE_CLOCK_RES / device->Frequency *
maxu(device->NumUpdates-1, 1);
return ret;
}
void ALCbackend_lock(ALCbackend *self)
{
int ret = almtx_lock(&self->mMutex);
assert(ret == althrd_success);
}
void ALCbackend_unlock(ALCbackend *self)
{
int ret = almtx_unlock(&self->mMutex);
assert(ret == althrd_success);
}
/* Base ALCbackendFactory method implementations. */
void ALCbackendFactory_deinit(ALCbackendFactory* UNUSED(self))
{
}
-178
View File
@@ -1,178 +0,0 @@
#ifndef AL_BACKENDS_BASE_H
#define AL_BACKENDS_BASE_H
#include "alMain.h"
#include "threads.h"
#include "alstring.h"
#ifdef __cplusplus
extern "C" {
#endif
typedef struct ClockLatency {
ALint64 ClockTime;
ALint64 Latency;
} ClockLatency;
/* Helper to get the current clock time from the device's ClockBase, and
* SamplesDone converted from the sample rate.
*/
inline ALuint64 GetDeviceClockTime(ALCdevice *device)
{
return device->ClockBase + (device->SamplesDone * DEVICE_CLOCK_RES /
device->Frequency);
}
struct ALCbackendVtable;
typedef struct ALCbackend {
const struct ALCbackendVtable *vtbl;
ALCdevice *mDevice;
almtx_t mMutex;
} ALCbackend;
void ALCbackend_Construct(ALCbackend *self, ALCdevice *device);
void ALCbackend_Destruct(ALCbackend *self);
ALCboolean ALCbackend_reset(ALCbackend *self);
ALCenum ALCbackend_captureSamples(ALCbackend *self, void *buffer, ALCuint samples);
ALCuint ALCbackend_availableSamples(ALCbackend *self);
ClockLatency ALCbackend_getClockLatency(ALCbackend *self);
void ALCbackend_lock(ALCbackend *self);
void ALCbackend_unlock(ALCbackend *self);
struct ALCbackendVtable {
void (*const Destruct)(ALCbackend*);
ALCenum (*const open)(ALCbackend*, const ALCchar*);
ALCboolean (*const reset)(ALCbackend*);
ALCboolean (*const start)(ALCbackend*);
void (*const stop)(ALCbackend*);
ALCenum (*const captureSamples)(ALCbackend*, void*, ALCuint);
ALCuint (*const availableSamples)(ALCbackend*);
ClockLatency (*const getClockLatency)(ALCbackend*);
void (*const lock)(ALCbackend*);
void (*const unlock)(ALCbackend*);
void (*const Delete)(void*);
};
#define DEFINE_ALCBACKEND_VTABLE(T) \
DECLARE_THUNK(T, ALCbackend, void, Destruct) \
DECLARE_THUNK1(T, ALCbackend, ALCenum, open, const ALCchar*) \
DECLARE_THUNK(T, ALCbackend, ALCboolean, reset) \
DECLARE_THUNK(T, ALCbackend, ALCboolean, start) \
DECLARE_THUNK(T, ALCbackend, void, stop) \
DECLARE_THUNK2(T, ALCbackend, ALCenum, captureSamples, void*, ALCuint) \
DECLARE_THUNK(T, ALCbackend, ALCuint, availableSamples) \
DECLARE_THUNK(T, ALCbackend, ClockLatency, getClockLatency) \
DECLARE_THUNK(T, ALCbackend, void, lock) \
DECLARE_THUNK(T, ALCbackend, void, unlock) \
static void T##_ALCbackend_Delete(void *ptr) \
{ T##_Delete(STATIC_UPCAST(T, ALCbackend, (ALCbackend*)ptr)); } \
\
static const struct ALCbackendVtable T##_ALCbackend_vtable = { \
T##_ALCbackend_Destruct, \
\
T##_ALCbackend_open, \
T##_ALCbackend_reset, \
T##_ALCbackend_start, \
T##_ALCbackend_stop, \
T##_ALCbackend_captureSamples, \
T##_ALCbackend_availableSamples, \
T##_ALCbackend_getClockLatency, \
T##_ALCbackend_lock, \
T##_ALCbackend_unlock, \
\
T##_ALCbackend_Delete, \
}
typedef enum ALCbackend_Type {
ALCbackend_Playback,
ALCbackend_Capture,
ALCbackend_Loopback
} ALCbackend_Type;
struct ALCbackendFactoryVtable;
typedef struct ALCbackendFactory {
const struct ALCbackendFactoryVtable *vtbl;
} ALCbackendFactory;
void ALCbackendFactory_deinit(ALCbackendFactory *self);
struct ALCbackendFactoryVtable {
ALCboolean (*const init)(ALCbackendFactory *self);
void (*const deinit)(ALCbackendFactory *self);
ALCboolean (*const querySupport)(ALCbackendFactory *self, ALCbackend_Type type);
void (*const probe)(ALCbackendFactory *self, enum DevProbe type, al_string *outnames);
ALCbackend* (*const createBackend)(ALCbackendFactory *self, ALCdevice *device, ALCbackend_Type type);
};
#define DEFINE_ALCBACKENDFACTORY_VTABLE(T) \
DECLARE_THUNK(T, ALCbackendFactory, ALCboolean, init) \
DECLARE_THUNK(T, ALCbackendFactory, void, deinit) \
DECLARE_THUNK1(T, ALCbackendFactory, ALCboolean, querySupport, ALCbackend_Type) \
DECLARE_THUNK2(T, ALCbackendFactory, void, probe, enum DevProbe, al_string*) \
DECLARE_THUNK2(T, ALCbackendFactory, ALCbackend*, createBackend, ALCdevice*, ALCbackend_Type) \
\
static const struct ALCbackendFactoryVtable T##_ALCbackendFactory_vtable = { \
T##_ALCbackendFactory_init, \
T##_ALCbackendFactory_deinit, \
T##_ALCbackendFactory_querySupport, \
T##_ALCbackendFactory_probe, \
T##_ALCbackendFactory_createBackend, \
}
ALCbackendFactory *ALCpulseBackendFactory_getFactory(void);
ALCbackendFactory *ALCalsaBackendFactory_getFactory(void);
ALCbackendFactory *ALCcoreAudioBackendFactory_getFactory(void);
ALCbackendFactory *ALCossBackendFactory_getFactory(void);
ALCbackendFactory *ALCjackBackendFactory_getFactory(void);
ALCbackendFactory *ALCsolarisBackendFactory_getFactory(void);
ALCbackendFactory *SndioBackendFactory_getFactory(void);
ALCbackendFactory *ALCqsaBackendFactory_getFactory(void);
ALCbackendFactory *ALCwasapiBackendFactory_getFactory(void);
ALCbackendFactory *ALCdsoundBackendFactory_getFactory(void);
ALCbackendFactory *ALCwinmmBackendFactory_getFactory(void);
ALCbackendFactory *ALCportBackendFactory_getFactory(void);
ALCbackendFactory *ALCopenslBackendFactory_getFactory(void);
ALCbackendFactory *ALCnullBackendFactory_getFactory(void);
ALCbackendFactory *ALCwaveBackendFactory_getFactory(void);
ALCbackendFactory *ALCsdl2BackendFactory_getFactory(void);
ALCbackendFactory *ALCloopbackFactory_getFactory(void);
inline void ALCdevice_Lock(ALCdevice *device)
{ V0(device->Backend,lock)(); }
inline void ALCdevice_Unlock(ALCdevice *device)
{ V0(device->Backend,unlock)(); }
inline ClockLatency GetClockLatency(ALCdevice *device)
{
ClockLatency ret = V0(device->Backend,getClockLatency)();
ret.Latency += device->FixedLatency;
return ret;
}
#ifdef __cplusplus
} /* extern "C" */
#endif
#endif /* AL_BACKENDS_BASE_H */
-816
View File
@@ -1,816 +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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 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 "ringbuffer.h"
#include <unistd.h>
#include <AudioUnit/AudioUnit.h>
#include <AudioToolbox/AudioToolbox.h>
#include "backends/base.h"
static const ALCchar ca_device[] = "CoreAudio Default";
typedef struct ALCcoreAudioPlayback {
DERIVE_FROM_TYPE(ALCbackend);
AudioUnit audioUnit;
ALuint frameSize;
AudioStreamBasicDescription format; // This is the OpenAL format as a CoreAudio ASBD
} ALCcoreAudioPlayback;
static void ALCcoreAudioPlayback_Construct(ALCcoreAudioPlayback *self, ALCdevice *device);
static void ALCcoreAudioPlayback_Destruct(ALCcoreAudioPlayback *self);
static ALCenum ALCcoreAudioPlayback_open(ALCcoreAudioPlayback *self, const ALCchar *name);
static ALCboolean ALCcoreAudioPlayback_reset(ALCcoreAudioPlayback *self);
static ALCboolean ALCcoreAudioPlayback_start(ALCcoreAudioPlayback *self);
static void ALCcoreAudioPlayback_stop(ALCcoreAudioPlayback *self);
static DECLARE_FORWARD2(ALCcoreAudioPlayback, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, ALCuint, availableSamples)
static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, ClockLatency, getClockLatency)
static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, void, lock)
static DECLARE_FORWARD(ALCcoreAudioPlayback, ALCbackend, void, unlock)
DECLARE_DEFAULT_ALLOCATORS(ALCcoreAudioPlayback)
DEFINE_ALCBACKEND_VTABLE(ALCcoreAudioPlayback);
static void ALCcoreAudioPlayback_Construct(ALCcoreAudioPlayback *self, ALCdevice *device)
{
ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
SET_VTABLE2(ALCcoreAudioPlayback, ALCbackend, self);
self->frameSize = 0;
memset(&self->format, 0, sizeof(self->format));
}
static void ALCcoreAudioPlayback_Destruct(ALCcoreAudioPlayback *self)
{
AudioUnitUninitialize(self->audioUnit);
AudioComponentInstanceDispose(self->audioUnit);
ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
}
static OSStatus ALCcoreAudioPlayback_MixerProc(void *inRefCon,
AudioUnitRenderActionFlags* UNUSED(ioActionFlags), const AudioTimeStamp* UNUSED(inTimeStamp),
UInt32 UNUSED(inBusNumber), UInt32 UNUSED(inNumberFrames), AudioBufferList *ioData)
{
ALCcoreAudioPlayback *self = inRefCon;
ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
ALCcoreAudioPlayback_lock(self);
aluMixData(device, ioData->mBuffers[0].mData,
ioData->mBuffers[0].mDataByteSize / self->frameSize);
ALCcoreAudioPlayback_unlock(self);
return noErr;
}
static ALCenum ALCcoreAudioPlayback_open(ALCcoreAudioPlayback *self, const ALCchar *name)
{
ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
AudioComponentDescription desc;
AudioComponent comp;
OSStatus err;
if(!name)
name = ca_device;
else if(strcmp(name, ca_device) != 0)
return ALC_INVALID_VALUE;
/* open the default output unit */
desc.componentType = kAudioUnitType_Output;
#if TARGET_OS_IOS
desc.componentSubType = kAudioUnitSubType_RemoteIO;
#else
desc.componentSubType = kAudioUnitSubType_DefaultOutput;
#endif
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
comp = AudioComponentFindNext(NULL, &desc);
if(comp == NULL)
{
ERR("AudioComponentFindNext failed\n");
return ALC_INVALID_VALUE;
}
err = AudioComponentInstanceNew(comp, &self->audioUnit);
if(err != noErr)
{
ERR("AudioComponentInstanceNew failed\n");
return ALC_INVALID_VALUE;
}
/* init and start the default audio unit... */
err = AudioUnitInitialize(self->audioUnit);
if(err != noErr)
{
ERR("AudioUnitInitialize failed\n");
AudioComponentInstanceDispose(self->audioUnit);
return ALC_INVALID_VALUE;
}
alstr_copy_cstr(&device->DeviceName, name);
return ALC_NO_ERROR;
}
static ALCboolean ALCcoreAudioPlayback_reset(ALCcoreAudioPlayback *self)
{
ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
AudioStreamBasicDescription streamFormat;
AURenderCallbackStruct input;
OSStatus err;
UInt32 size;
err = AudioUnitUninitialize(self->audioUnit);
if(err != noErr)
ERR("-- AudioUnitUninitialize failed.\n");
/* retrieve default output unit's properties (output side) */
size = sizeof(AudioStreamBasicDescription);
err = AudioUnitGetProperty(self->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(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, size);
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
return ALC_FALSE;
}
if(device->Frequency != streamFormat.mSampleRate)
{
device->NumUpdates = (ALuint)((ALuint64)device->NumUpdates *
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(self->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, sizeof(AudioStreamBasicDescription));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
return ALC_FALSE;
}
/* setup callback */
self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
input.inputProc = ALCcoreAudioPlayback_MixerProc;
input.inputProcRefCon = self;
err = AudioUnitSetProperty(self->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(self->audioUnit);
if(err != noErr)
{
ERR("AudioUnitInitialize failed\n");
return ALC_FALSE;
}
return ALC_TRUE;
}
static ALCboolean ALCcoreAudioPlayback_start(ALCcoreAudioPlayback *self)
{
OSStatus err = AudioOutputUnitStart(self->audioUnit);
if(err != noErr)
{
ERR("AudioOutputUnitStart failed\n");
return ALC_FALSE;
}
return ALC_TRUE;
}
static void ALCcoreAudioPlayback_stop(ALCcoreAudioPlayback *self)
{
OSStatus err = AudioOutputUnitStop(self->audioUnit);
if(err != noErr)
ERR("AudioOutputUnitStop failed\n");
}
typedef struct ALCcoreAudioCapture {
DERIVE_FROM_TYPE(ALCbackend);
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
ll_ringbuffer_t *ring;
} ALCcoreAudioCapture;
static void ALCcoreAudioCapture_Construct(ALCcoreAudioCapture *self, ALCdevice *device);
static void ALCcoreAudioCapture_Destruct(ALCcoreAudioCapture *self);
static ALCenum ALCcoreAudioCapture_open(ALCcoreAudioCapture *self, const ALCchar *name);
static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, ALCboolean, reset)
static ALCboolean ALCcoreAudioCapture_start(ALCcoreAudioCapture *self);
static void ALCcoreAudioCapture_stop(ALCcoreAudioCapture *self);
static ALCenum ALCcoreAudioCapture_captureSamples(ALCcoreAudioCapture *self, ALCvoid *buffer, ALCuint samples);
static ALCuint ALCcoreAudioCapture_availableSamples(ALCcoreAudioCapture *self);
static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, ClockLatency, getClockLatency)
static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, void, lock)
static DECLARE_FORWARD(ALCcoreAudioCapture, ALCbackend, void, unlock)
DECLARE_DEFAULT_ALLOCATORS(ALCcoreAudioCapture)
DEFINE_ALCBACKEND_VTABLE(ALCcoreAudioCapture);
static AudioBufferList *allocate_buffer_list(UInt32 channelCount, UInt32 byteSize)
{
AudioBufferList *list;
list = calloc(1, FAM_SIZE(AudioBufferList, mBuffers, 1) + byteSize);
if(list)
{
list->mNumberBuffers = 1;
list->mBuffers[0].mNumberChannels = channelCount;
list->mBuffers[0].mDataByteSize = byteSize;
list->mBuffers[0].mData = &list->mBuffers[1];
}
return list;
}
static void destroy_buffer_list(AudioBufferList *list)
{
free(list);
}
static void ALCcoreAudioCapture_Construct(ALCcoreAudioCapture *self, ALCdevice *device)
{
ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
SET_VTABLE2(ALCcoreAudioCapture, ALCbackend, self);
self->audioUnit = 0;
self->audioConverter = NULL;
self->bufferList = NULL;
self->resampleBuffer = NULL;
self->ring = NULL;
}
static void ALCcoreAudioCapture_Destruct(ALCcoreAudioCapture *self)
{
ll_ringbuffer_free(self->ring);
self->ring = NULL;
free(self->resampleBuffer);
self->resampleBuffer = NULL;
destroy_buffer_list(self->bufferList);
self->bufferList = NULL;
if(self->audioConverter)
AudioConverterDispose(self->audioConverter);
self->audioConverter = NULL;
if(self->audioUnit)
AudioComponentInstanceDispose(self->audioUnit);
self->audioUnit = 0;
ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
}
static OSStatus ALCcoreAudioCapture_RecordProc(void *inRefCon,
AudioUnitRenderActionFlags* UNUSED(ioActionFlags),
const AudioTimeStamp *inTimeStamp, UInt32 UNUSED(inBusNumber),
UInt32 inNumberFrames, AudioBufferList* UNUSED(ioData))
{
ALCcoreAudioCapture *self = inRefCon;
AudioUnitRenderActionFlags flags = 0;
OSStatus err;
// fill the bufferList with data from the input device
err = AudioUnitRender(self->audioUnit, &flags, inTimeStamp, 1, inNumberFrames, self->bufferList);
if(err != noErr)
{
ERR("AudioUnitRender error: %d\n", err);
return err;
}
ll_ringbuffer_write(self->ring, self->bufferList->mBuffers[0].mData, inNumberFrames);
return noErr;
}
static OSStatus ALCcoreAudioCapture_ConvertCallback(AudioConverterRef UNUSED(inAudioConverter),
UInt32 *ioNumberDataPackets, AudioBufferList *ioData,
AudioStreamPacketDescription** UNUSED(outDataPacketDescription),
void *inUserData)
{
ALCcoreAudioCapture *self = inUserData;
// Read from the ring buffer and store temporarily in a large buffer
ll_ringbuffer_read(self->ring, self->resampleBuffer, *ioNumberDataPackets);
// Set the input data
ioData->mNumberBuffers = 1;
ioData->mBuffers[0].mNumberChannels = self->format.mChannelsPerFrame;
ioData->mBuffers[0].mData = self->resampleBuffer;
ioData->mBuffers[0].mDataByteSize = (*ioNumberDataPackets) * self->format.mBytesPerFrame;
return noErr;
}
static ALCenum ALCcoreAudioCapture_open(ALCcoreAudioCapture *self, const ALCchar *name)
{
ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
AudioStreamBasicDescription requestedFormat; // The application requested format
AudioStreamBasicDescription hardwareFormat; // The hardware format
AudioStreamBasicDescription outputFormat; // The AudioUnit output format
AURenderCallbackStruct input;
AudioComponentDescription desc;
UInt32 outputFrameCount;
UInt32 propertySize;
AudioObjectPropertyAddress propertyAddress;
UInt32 enableIO;
AudioComponent comp;
OSStatus err;
if(!name)
name = ca_device;
else if(strcmp(name, ca_device) != 0)
return ALC_INVALID_VALUE;
desc.componentType = kAudioUnitType_Output;
#if TARGET_OS_IOS
desc.componentSubType = kAudioUnitSubType_RemoteIO;
#else
desc.componentSubType = kAudioUnitSubType_HALOutput;
#endif
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
// Search for component with given description
comp = AudioComponentFindNext(NULL, &desc);
if(comp == NULL)
{
ERR("AudioComponentFindNext failed\n");
return ALC_INVALID_VALUE;
}
// Open the component
err = AudioComponentInstanceNew(comp, &self->audioUnit);
if(err != noErr)
{
ERR("AudioComponentInstanceNew failed\n");
goto error;
}
// Turn off AudioUnit output
enableIO = 0;
err = AudioUnitSetProperty(self->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(self->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &enableIO, sizeof(ALuint));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
goto error;
}
#if !TARGET_OS_IOS
// Get the default input device
AudioDeviceID inputDevice = kAudioDeviceUnknown;
propertySize = sizeof(AudioDeviceID);
propertyAddress.mSelector = kAudioHardwarePropertyDefaultInputDevice;
propertyAddress.mScope = kAudioObjectPropertyScopeGlobal;
propertyAddress.mElement = kAudioObjectPropertyElementMaster;
err = AudioObjectGetPropertyData(kAudioObjectSystemObject, &propertyAddress, 0, NULL, &propertySize, &inputDevice);
if(err != noErr)
{
ERR("AudioObjectGetPropertyData failed\n");
goto error;
}
if(inputDevice == kAudioDeviceUnknown)
{
ERR("No input device found\n");
goto error;
}
// Track the input device
err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDevice, sizeof(AudioDeviceID));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
goto error;
}
#endif
// set capture callback
input.inputProc = ALCcoreAudioCapture_RecordProc;
input.inputProcRefCon = self;
err = AudioUnitSetProperty(self->audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &input, sizeof(AURenderCallbackStruct));
if(err != noErr)
{
ERR("AudioUnitSetProperty failed\n");
goto error;
}
// Initialize the device
err = AudioUnitInitialize(self->audioUnit);
if(err != noErr)
{
ERR("AudioUnitInitialize failed\n");
goto error;
}
// Get the hardware format
propertySize = sizeof(AudioStreamBasicDescription);
err = AudioUnitGetProperty(self->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 DevFmtX51Rear:
case DevFmtX61:
case DevFmtX71:
case DevFmtAmbi3D:
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
self->format = requestedFormat;
self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
// 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
self->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(self->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 * self->sampleRateRatio;
err = AudioUnitSetProperty(self->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, &self->audioConverter);
if(err != noErr)
{
ERR("AudioConverterNew failed: %d\n", err);
goto error;
}
// Create a buffer for use in the resample callback
self->resampleBuffer = malloc(device->UpdateSize * self->frameSize * self->sampleRateRatio);
// Allocate buffer for the AudioUnit output
self->bufferList = allocate_buffer_list(outputFormat.mChannelsPerFrame, device->UpdateSize * self->frameSize * self->sampleRateRatio);
if(self->bufferList == NULL)
goto error;
self->ring = ll_ringbuffer_create(
(size_t)ceil(device->UpdateSize*self->sampleRateRatio*device->NumUpdates),
self->frameSize, false
);
if(!self->ring) goto error;
alstr_copy_cstr(&device->DeviceName, name);
return ALC_NO_ERROR;
error:
ll_ringbuffer_free(self->ring);
self->ring = NULL;
free(self->resampleBuffer);
self->resampleBuffer = NULL;
destroy_buffer_list(self->bufferList);
self->bufferList = NULL;
if(self->audioConverter)
AudioConverterDispose(self->audioConverter);
self->audioConverter = NULL;
if(self->audioUnit)
AudioComponentInstanceDispose(self->audioUnit);
self->audioUnit = 0;
return ALC_INVALID_VALUE;
}
static ALCboolean ALCcoreAudioCapture_start(ALCcoreAudioCapture *self)
{
OSStatus err = AudioOutputUnitStart(self->audioUnit);
if(err != noErr)
{
ERR("AudioOutputUnitStart failed\n");
return ALC_FALSE;
}
return ALC_TRUE;
}
static void ALCcoreAudioCapture_stop(ALCcoreAudioCapture *self)
{
OSStatus err = AudioOutputUnitStop(self->audioUnit);
if(err != noErr)
ERR("AudioOutputUnitStop failed\n");
}
static ALCenum ALCcoreAudioCapture_captureSamples(ALCcoreAudioCapture *self, ALCvoid *buffer, ALCuint samples)
{
union {
ALbyte _[sizeof(AudioBufferList) + sizeof(AudioBuffer)];
AudioBufferList list;
} audiobuf = { { 0 } };
UInt32 frameCount;
OSStatus err;
// If no samples are requested, just return
if(samples == 0) return ALC_NO_ERROR;
// Point the resampling buffer to the capture buffer
audiobuf.list.mNumberBuffers = 1;
audiobuf.list.mBuffers[0].mNumberChannels = self->format.mChannelsPerFrame;
audiobuf.list.mBuffers[0].mDataByteSize = samples * self->frameSize;
audiobuf.list.mBuffers[0].mData = buffer;
// Resample into another AudioBufferList
frameCount = samples;
err = AudioConverterFillComplexBuffer(self->audioConverter,
ALCcoreAudioCapture_ConvertCallback, self, &frameCount, &audiobuf.list, NULL
);
if(err != noErr)
{
ERR("AudioConverterFillComplexBuffer error: %d\n", err);
return ALC_INVALID_VALUE;
}
return ALC_NO_ERROR;
}
static ALCuint ALCcoreAudioCapture_availableSamples(ALCcoreAudioCapture *self)
{
return ll_ringbuffer_read_space(self->ring) / self->sampleRateRatio;
}
typedef struct ALCcoreAudioBackendFactory {
DERIVE_FROM_TYPE(ALCbackendFactory);
} ALCcoreAudioBackendFactory;
#define ALCCOREAUDIOBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCcoreAudioBackendFactory, ALCbackendFactory) } }
ALCbackendFactory *ALCcoreAudioBackendFactory_getFactory(void);
static ALCboolean ALCcoreAudioBackendFactory_init(ALCcoreAudioBackendFactory *self);
static DECLARE_FORWARD(ALCcoreAudioBackendFactory, ALCbackendFactory, void, deinit)
static ALCboolean ALCcoreAudioBackendFactory_querySupport(ALCcoreAudioBackendFactory *self, ALCbackend_Type type);
static void ALCcoreAudioBackendFactory_probe(ALCcoreAudioBackendFactory *self, enum DevProbe type, al_string *outnames);
static ALCbackend* ALCcoreAudioBackendFactory_createBackend(ALCcoreAudioBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
DEFINE_ALCBACKENDFACTORY_VTABLE(ALCcoreAudioBackendFactory);
ALCbackendFactory *ALCcoreAudioBackendFactory_getFactory(void)
{
static ALCcoreAudioBackendFactory factory = ALCCOREAUDIOBACKENDFACTORY_INITIALIZER;
return STATIC_CAST(ALCbackendFactory, &factory);
}
static ALCboolean ALCcoreAudioBackendFactory_init(ALCcoreAudioBackendFactory* UNUSED(self))
{
return ALC_TRUE;
}
static ALCboolean ALCcoreAudioBackendFactory_querySupport(ALCcoreAudioBackendFactory* UNUSED(self), ALCbackend_Type type)
{
if(type == ALCbackend_Playback || ALCbackend_Capture)
return ALC_TRUE;
return ALC_FALSE;
}
static void ALCcoreAudioBackendFactory_probe(ALCcoreAudioBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames)
{
switch(type)
{
case ALL_DEVICE_PROBE:
case CAPTURE_DEVICE_PROBE:
alstr_append_range(outnames, ca_device, ca_device+sizeof(ca_device));
break;
}
}
static ALCbackend* ALCcoreAudioBackendFactory_createBackend(ALCcoreAudioBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
{
if(type == ALCbackend_Playback)
{
ALCcoreAudioPlayback *backend;
NEW_OBJ(backend, ALCcoreAudioPlayback)(device);
if(!backend) return NULL;
return STATIC_CAST(ALCbackend, backend);
}
if(type == ALCbackend_Capture)
{
ALCcoreAudioCapture *backend;
NEW_OBJ(backend, ALCcoreAudioCapture)(device);
if(!backend) return NULL;
return STATIC_CAST(ALCbackend, backend);
}
return NULL;
}
File diff suppressed because it is too large Load Diff
-607
View File
@@ -1,607 +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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdlib.h>
#include <stdio.h>
#include <memory.h>
#include "alMain.h"
#include "alu.h"
#include "alconfig.h"
#include "ringbuffer.h"
#include "threads.h"
#include "compat.h"
#include "backends/base.h"
#include <jack/jack.h>
#include <jack/ringbuffer.h>
static const ALCchar jackDevice[] = "JACK Default";
#ifdef HAVE_DYNLOAD
#define JACK_FUNCS(MAGIC) \
MAGIC(jack_client_open); \
MAGIC(jack_client_close); \
MAGIC(jack_client_name_size); \
MAGIC(jack_get_client_name); \
MAGIC(jack_connect); \
MAGIC(jack_activate); \
MAGIC(jack_deactivate); \
MAGIC(jack_port_register); \
MAGIC(jack_port_unregister); \
MAGIC(jack_port_get_buffer); \
MAGIC(jack_port_name); \
MAGIC(jack_get_ports); \
MAGIC(jack_free); \
MAGIC(jack_get_sample_rate); \
MAGIC(jack_set_error_function); \
MAGIC(jack_set_process_callback); \
MAGIC(jack_set_buffer_size_callback); \
MAGIC(jack_set_buffer_size); \
MAGIC(jack_get_buffer_size);
static void *jack_handle;
#define MAKE_FUNC(f) static __typeof(f) * p##f
JACK_FUNCS(MAKE_FUNC);
static __typeof(jack_error_callback) * pjack_error_callback;
#undef MAKE_FUNC
#define jack_client_open pjack_client_open
#define jack_client_close pjack_client_close
#define jack_client_name_size pjack_client_name_size
#define jack_get_client_name pjack_get_client_name
#define jack_connect pjack_connect
#define jack_activate pjack_activate
#define jack_deactivate pjack_deactivate
#define jack_port_register pjack_port_register
#define jack_port_unregister pjack_port_unregister
#define jack_port_get_buffer pjack_port_get_buffer
#define jack_port_name pjack_port_name
#define jack_get_ports pjack_get_ports
#define jack_free pjack_free
#define jack_get_sample_rate pjack_get_sample_rate
#define jack_set_error_function pjack_set_error_function
#define jack_set_process_callback pjack_set_process_callback
#define jack_set_buffer_size_callback pjack_set_buffer_size_callback
#define jack_set_buffer_size pjack_set_buffer_size
#define jack_get_buffer_size pjack_get_buffer_size
#define jack_error_callback (*pjack_error_callback)
#endif
static jack_options_t ClientOptions = JackNullOption;
static ALCboolean jack_load(void)
{
ALCboolean error = ALC_FALSE;
#ifdef HAVE_DYNLOAD
if(!jack_handle)
{
al_string missing_funcs = AL_STRING_INIT_STATIC();
#ifdef _WIN32
#define JACKLIB "libjack.dll"
#else
#define JACKLIB "libjack.so.0"
#endif
jack_handle = LoadLib(JACKLIB);
if(!jack_handle)
{
WARN("Failed to load %s\n", JACKLIB);
return ALC_FALSE;
}
error = ALC_FALSE;
#define LOAD_FUNC(f) do { \
p##f = GetSymbol(jack_handle, #f); \
if(p##f == NULL) { \
error = ALC_TRUE; \
alstr_append_cstr(&missing_funcs, "\n" #f); \
} \
} while(0)
JACK_FUNCS(LOAD_FUNC);
#undef LOAD_FUNC
/* Optional symbols. These don't exist in all versions of JACK. */
#define LOAD_SYM(f) p##f = GetSymbol(jack_handle, #f)
LOAD_SYM(jack_error_callback);
#undef LOAD_SYM
if(error)
{
WARN("Missing expected functions:%s\n", alstr_get_cstr(missing_funcs));
CloseLib(jack_handle);
jack_handle = NULL;
}
alstr_reset(&missing_funcs);
}
#endif
return !error;
}
typedef struct ALCjackPlayback {
DERIVE_FROM_TYPE(ALCbackend);
jack_client_t *Client;
jack_port_t *Port[MAX_OUTPUT_CHANNELS];
ll_ringbuffer_t *Ring;
alsem_t Sem;
ATOMIC(ALenum) killNow;
althrd_t thread;
} ALCjackPlayback;
static int ALCjackPlayback_bufferSizeNotify(jack_nframes_t numframes, void *arg);
static int ALCjackPlayback_process(jack_nframes_t numframes, void *arg);
static int ALCjackPlayback_mixerProc(void *arg);
static void ALCjackPlayback_Construct(ALCjackPlayback *self, ALCdevice *device);
static void ALCjackPlayback_Destruct(ALCjackPlayback *self);
static ALCenum ALCjackPlayback_open(ALCjackPlayback *self, const ALCchar *name);
static ALCboolean ALCjackPlayback_reset(ALCjackPlayback *self);
static ALCboolean ALCjackPlayback_start(ALCjackPlayback *self);
static void ALCjackPlayback_stop(ALCjackPlayback *self);
static DECLARE_FORWARD2(ALCjackPlayback, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
static DECLARE_FORWARD(ALCjackPlayback, ALCbackend, ALCuint, availableSamples)
static ClockLatency ALCjackPlayback_getClockLatency(ALCjackPlayback *self);
static DECLARE_FORWARD(ALCjackPlayback, ALCbackend, void, lock)
static DECLARE_FORWARD(ALCjackPlayback, ALCbackend, void, unlock)
DECLARE_DEFAULT_ALLOCATORS(ALCjackPlayback)
DEFINE_ALCBACKEND_VTABLE(ALCjackPlayback);
static void ALCjackPlayback_Construct(ALCjackPlayback *self, ALCdevice *device)
{
ALuint i;
ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
SET_VTABLE2(ALCjackPlayback, ALCbackend, self);
alsem_init(&self->Sem, 0);
self->Client = NULL;
for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
self->Port[i] = NULL;
self->Ring = NULL;
ATOMIC_INIT(&self->killNow, AL_TRUE);
}
static void ALCjackPlayback_Destruct(ALCjackPlayback *self)
{
ALuint i;
if(self->Client)
{
for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
{
if(self->Port[i])
jack_port_unregister(self->Client, self->Port[i]);
self->Port[i] = NULL;
}
jack_client_close(self->Client);
self->Client = NULL;
}
alsem_destroy(&self->Sem);
ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
}
static int ALCjackPlayback_bufferSizeNotify(jack_nframes_t numframes, void *arg)
{
ALCjackPlayback *self = arg;
ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
ALuint bufsize;
ALCjackPlayback_lock(self);
device->UpdateSize = numframes;
device->NumUpdates = 2;
bufsize = device->UpdateSize;
if(ConfigValueUInt(alstr_get_cstr(device->DeviceName), "jack", "buffer-size", &bufsize))
bufsize = maxu(NextPowerOf2(bufsize), device->UpdateSize);
device->NumUpdates = (bufsize+device->UpdateSize) / device->UpdateSize;
TRACE("%u update size x%u\n", device->UpdateSize, device->NumUpdates);
ll_ringbuffer_free(self->Ring);
self->Ring = ll_ringbuffer_create(bufsize,
FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder),
true
);
if(!self->Ring)
{
ERR("Failed to reallocate ringbuffer\n");
aluHandleDisconnect(device, "Failed to reallocate %u-sample buffer", bufsize);
}
ALCjackPlayback_unlock(self);
return 0;
}
static int ALCjackPlayback_process(jack_nframes_t numframes, void *arg)
{
ALCjackPlayback *self = arg;
jack_default_audio_sample_t *out[MAX_OUTPUT_CHANNELS];
ll_ringbuffer_data_t data[2];
jack_nframes_t total = 0;
jack_nframes_t todo;
ALsizei i, c, numchans;
ll_ringbuffer_get_read_vector(self->Ring, data);
for(c = 0;c < MAX_OUTPUT_CHANNELS && self->Port[c];c++)
out[c] = jack_port_get_buffer(self->Port[c], numframes);
numchans = c;
todo = minu(numframes, data[0].len);
for(c = 0;c < numchans;c++)
{
const ALfloat *restrict in = ((ALfloat*)data[0].buf) + c;
for(i = 0;(jack_nframes_t)i < todo;i++)
out[c][i] = in[i*numchans];
out[c] += todo;
}
total += todo;
todo = minu(numframes-total, data[1].len);
if(todo > 0)
{
for(c = 0;c < numchans;c++)
{
const ALfloat *restrict in = ((ALfloat*)data[1].buf) + c;
for(i = 0;(jack_nframes_t)i < todo;i++)
out[c][i] = in[i*numchans];
out[c] += todo;
}
total += todo;
}
ll_ringbuffer_read_advance(self->Ring, total);
alsem_post(&self->Sem);
if(numframes > total)
{
todo = numframes-total;
for(c = 0;c < numchans;c++)
{
for(i = 0;(jack_nframes_t)i < todo;i++)
out[c][i] = 0.0f;
}
}
return 0;
}
static int ALCjackPlayback_mixerProc(void *arg)
{
ALCjackPlayback *self = arg;
ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
ll_ringbuffer_data_t data[2];
SetRTPriority();
althrd_setname(althrd_current(), MIXER_THREAD_NAME);
ALCjackPlayback_lock(self);
while(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) &&
ATOMIC_LOAD(&device->Connected, almemory_order_acquire))
{
ALuint todo, len1, len2;
if(ll_ringbuffer_write_space(self->Ring) < device->UpdateSize)
{
ALCjackPlayback_unlock(self);
alsem_wait(&self->Sem);
ALCjackPlayback_lock(self);
continue;
}
ll_ringbuffer_get_write_vector(self->Ring, data);
todo = data[0].len + data[1].len;
todo -= todo%device->UpdateSize;
len1 = minu(data[0].len, todo);
len2 = minu(data[1].len, todo-len1);
aluMixData(device, data[0].buf, len1);
if(len2 > 0)
aluMixData(device, data[1].buf, len2);
ll_ringbuffer_write_advance(self->Ring, todo);
}
ALCjackPlayback_unlock(self);
return 0;
}
static ALCenum ALCjackPlayback_open(ALCjackPlayback *self, const ALCchar *name)
{
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
const char *client_name = "alsoft";
jack_status_t status;
if(!name)
name = jackDevice;
else if(strcmp(name, jackDevice) != 0)
return ALC_INVALID_VALUE;
self->Client = jack_client_open(client_name, ClientOptions, &status, NULL);
if(self->Client == NULL)
{
ERR("jack_client_open() failed, status = 0x%02x\n", status);
return ALC_INVALID_VALUE;
}
if((status&JackServerStarted))
TRACE("JACK server started\n");
if((status&JackNameNotUnique))
{
client_name = jack_get_client_name(self->Client);
TRACE("Client name not unique, got `%s' instead\n", client_name);
}
jack_set_process_callback(self->Client, ALCjackPlayback_process, self);
jack_set_buffer_size_callback(self->Client, ALCjackPlayback_bufferSizeNotify, self);
alstr_copy_cstr(&device->DeviceName, name);
return ALC_NO_ERROR;
}
static ALCboolean ALCjackPlayback_reset(ALCjackPlayback *self)
{
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
ALsizei numchans, i;
ALuint bufsize;
for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
{
if(self->Port[i])
jack_port_unregister(self->Client, self->Port[i]);
self->Port[i] = NULL;
}
/* Ignore the requested buffer metrics and just keep one JACK-sized buffer
* ready for when requested.
*/
device->Frequency = jack_get_sample_rate(self->Client);
device->UpdateSize = jack_get_buffer_size(self->Client);
device->NumUpdates = 2;
bufsize = device->UpdateSize;
if(ConfigValueUInt(alstr_get_cstr(device->DeviceName), "jack", "buffer-size", &bufsize))
bufsize = maxu(NextPowerOf2(bufsize), device->UpdateSize);
device->NumUpdates = (bufsize+device->UpdateSize) / device->UpdateSize;
/* Force 32-bit float output. */
device->FmtType = DevFmtFloat;
numchans = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
for(i = 0;i < numchans;i++)
{
char name[64];
snprintf(name, sizeof(name), "channel_%d", i+1);
self->Port[i] = jack_port_register(self->Client, name, JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
if(self->Port[i] == NULL)
{
ERR("Not enough JACK ports available for %s output\n", DevFmtChannelsString(device->FmtChans));
if(i == 0) return ALC_FALSE;
break;
}
}
if(i < numchans)
{
if(i == 1)
device->FmtChans = DevFmtMono;
else
{
for(--i;i >= 2;i--)
{
jack_port_unregister(self->Client, self->Port[i]);
self->Port[i] = NULL;
}
device->FmtChans = DevFmtStereo;
}
}
ll_ringbuffer_free(self->Ring);
self->Ring = ll_ringbuffer_create(bufsize,
FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder),
true
);
if(!self->Ring)
{
ERR("Failed to allocate ringbuffer\n");
return ALC_FALSE;
}
SetDefaultChannelOrder(device);
return ALC_TRUE;
}
static ALCboolean ALCjackPlayback_start(ALCjackPlayback *self)
{
const char **ports;
ALsizei i;
if(jack_activate(self->Client))
{
ERR("Failed to activate client\n");
return ALC_FALSE;
}
ports = jack_get_ports(self->Client, NULL, NULL, JackPortIsPhysical|JackPortIsInput);
if(ports == NULL)
{
ERR("No physical playback ports found\n");
jack_deactivate(self->Client);
return ALC_FALSE;
}
for(i = 0;i < MAX_OUTPUT_CHANNELS && self->Port[i];i++)
{
if(!ports[i])
{
ERR("No physical playback port for \"%s\"\n", jack_port_name(self->Port[i]));
break;
}
if(jack_connect(self->Client, jack_port_name(self->Port[i]), ports[i]))
ERR("Failed to connect output port \"%s\" to \"%s\"\n", jack_port_name(self->Port[i]), ports[i]);
}
jack_free(ports);
ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release);
if(althrd_create(&self->thread, ALCjackPlayback_mixerProc, self) != althrd_success)
{
jack_deactivate(self->Client);
return ALC_FALSE;
}
return ALC_TRUE;
}
static void ALCjackPlayback_stop(ALCjackPlayback *self)
{
int res;
if(ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel))
return;
alsem_post(&self->Sem);
althrd_join(self->thread, &res);
jack_deactivate(self->Client);
}
static ClockLatency ALCjackPlayback_getClockLatency(ALCjackPlayback *self)
{
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
ClockLatency ret;
ALCjackPlayback_lock(self);
ret.ClockTime = GetDeviceClockTime(device);
ret.Latency = ll_ringbuffer_read_space(self->Ring) * DEVICE_CLOCK_RES /
device->Frequency;
ALCjackPlayback_unlock(self);
return ret;
}
static void jack_msg_handler(const char *message)
{
WARN("%s\n", message);
}
typedef struct ALCjackBackendFactory {
DERIVE_FROM_TYPE(ALCbackendFactory);
} ALCjackBackendFactory;
#define ALCJACKBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCjackBackendFactory, ALCbackendFactory) } }
static ALCboolean ALCjackBackendFactory_init(ALCjackBackendFactory* UNUSED(self))
{
void (*old_error_cb)(const char*);
jack_client_t *client;
jack_status_t status;
if(!jack_load())
return ALC_FALSE;
if(!GetConfigValueBool(NULL, "jack", "spawn-server", 0))
ClientOptions |= JackNoStartServer;
old_error_cb = (&jack_error_callback ? jack_error_callback : NULL);
jack_set_error_function(jack_msg_handler);
client = jack_client_open("alsoft", ClientOptions, &status, NULL);
jack_set_error_function(old_error_cb);
if(client == NULL)
{
WARN("jack_client_open() failed, 0x%02x\n", status);
if((status&JackServerFailed) && !(ClientOptions&JackNoStartServer))
ERR("Unable to connect to JACK server\n");
return ALC_FALSE;
}
jack_client_close(client);
return ALC_TRUE;
}
static void ALCjackBackendFactory_deinit(ALCjackBackendFactory* UNUSED(self))
{
#ifdef HAVE_DYNLOAD
if(jack_handle)
CloseLib(jack_handle);
jack_handle = NULL;
#endif
}
static ALCboolean ALCjackBackendFactory_querySupport(ALCjackBackendFactory* UNUSED(self), ALCbackend_Type type)
{
if(type == ALCbackend_Playback)
return ALC_TRUE;
return ALC_FALSE;
}
static void ALCjackBackendFactory_probe(ALCjackBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames)
{
switch(type)
{
case ALL_DEVICE_PROBE:
alstr_append_range(outnames, jackDevice, jackDevice+sizeof(jackDevice));
break;
case CAPTURE_DEVICE_PROBE:
break;
}
}
static ALCbackend* ALCjackBackendFactory_createBackend(ALCjackBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
{
if(type == ALCbackend_Playback)
{
ALCjackPlayback *backend;
NEW_OBJ(backend, ALCjackPlayback)(device);
if(!backend) return NULL;
return STATIC_CAST(ALCbackend, backend);
}
return NULL;
}
DEFINE_ALCBACKENDFACTORY_VTABLE(ALCjackBackendFactory);
ALCbackendFactory *ALCjackBackendFactory_getFactory(void)
{
static ALCjackBackendFactory factory = ALCJACKBACKENDFACTORY_INITIALIZER;
return STATIC_CAST(ALCbackendFactory, &factory);
}
-128
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@@ -1,128 +0,0 @@
/**
* 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdlib.h>
#include "alMain.h"
#include "alu.h"
#include "backends/base.h"
typedef struct ALCloopback {
DERIVE_FROM_TYPE(ALCbackend);
} ALCloopback;
static void ALCloopback_Construct(ALCloopback *self, ALCdevice *device);
static DECLARE_FORWARD(ALCloopback, ALCbackend, void, Destruct)
static ALCenum ALCloopback_open(ALCloopback *self, const ALCchar *name);
static ALCboolean ALCloopback_reset(ALCloopback *self);
static ALCboolean ALCloopback_start(ALCloopback *self);
static void ALCloopback_stop(ALCloopback *self);
static DECLARE_FORWARD2(ALCloopback, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
static DECLARE_FORWARD(ALCloopback, ALCbackend, ALCuint, availableSamples)
static DECLARE_FORWARD(ALCloopback, ALCbackend, ClockLatency, getClockLatency)
static DECLARE_FORWARD(ALCloopback, ALCbackend, void, lock)
static DECLARE_FORWARD(ALCloopback, ALCbackend, void, unlock)
DECLARE_DEFAULT_ALLOCATORS(ALCloopback)
DEFINE_ALCBACKEND_VTABLE(ALCloopback);
static void ALCloopback_Construct(ALCloopback *self, ALCdevice *device)
{
ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
SET_VTABLE2(ALCloopback, ALCbackend, self);
}
static ALCenum ALCloopback_open(ALCloopback *self, const ALCchar *name)
{
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
alstr_copy_cstr(&device->DeviceName, name);
return ALC_NO_ERROR;
}
static ALCboolean ALCloopback_reset(ALCloopback *self)
{
SetDefaultWFXChannelOrder(STATIC_CAST(ALCbackend, self)->mDevice);
return ALC_TRUE;
}
static ALCboolean ALCloopback_start(ALCloopback* UNUSED(self))
{
return ALC_TRUE;
}
static void ALCloopback_stop(ALCloopback* UNUSED(self))
{
}
typedef struct ALCloopbackFactory {
DERIVE_FROM_TYPE(ALCbackendFactory);
} ALCloopbackFactory;
#define ALCNULLBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCloopbackFactory, ALCbackendFactory) } }
ALCbackendFactory *ALCloopbackFactory_getFactory(void);
static ALCboolean ALCloopbackFactory_init(ALCloopbackFactory *self);
static DECLARE_FORWARD(ALCloopbackFactory, ALCbackendFactory, void, deinit)
static ALCboolean ALCloopbackFactory_querySupport(ALCloopbackFactory *self, ALCbackend_Type type);
static void ALCloopbackFactory_probe(ALCloopbackFactory *self, enum DevProbe type, al_string *outnames);
static ALCbackend* ALCloopbackFactory_createBackend(ALCloopbackFactory *self, ALCdevice *device, ALCbackend_Type type);
DEFINE_ALCBACKENDFACTORY_VTABLE(ALCloopbackFactory);
ALCbackendFactory *ALCloopbackFactory_getFactory(void)
{
static ALCloopbackFactory factory = ALCNULLBACKENDFACTORY_INITIALIZER;
return STATIC_CAST(ALCbackendFactory, &factory);
}
static ALCboolean ALCloopbackFactory_init(ALCloopbackFactory* UNUSED(self))
{
return ALC_TRUE;
}
static ALCboolean ALCloopbackFactory_querySupport(ALCloopbackFactory* UNUSED(self), ALCbackend_Type type)
{
if(type == ALCbackend_Loopback)
return ALC_TRUE;
return ALC_FALSE;
}
static void ALCloopbackFactory_probe(ALCloopbackFactory* UNUSED(self), enum DevProbe UNUSED(type), al_string* UNUSED(outnames))
{
}
static ALCbackend* ALCloopbackFactory_createBackend(ALCloopbackFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
{
if(type == ALCbackend_Loopback)
{
ALCloopback *backend;
NEW_OBJ(backend, ALCloopback)(device);
if(!backend) return NULL;
return STATIC_CAST(ALCbackend, backend);
}
return NULL;
}
-220
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@@ -1,220 +0,0 @@
/**
* 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 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"
#include "threads.h"
#include "compat.h"
#include "backends/base.h"
typedef struct ALCnullBackend {
DERIVE_FROM_TYPE(ALCbackend);
ATOMIC(int) killNow;
althrd_t thread;
} ALCnullBackend;
static int ALCnullBackend_mixerProc(void *ptr);
static void ALCnullBackend_Construct(ALCnullBackend *self, ALCdevice *device);
static DECLARE_FORWARD(ALCnullBackend, ALCbackend, void, Destruct)
static ALCenum ALCnullBackend_open(ALCnullBackend *self, const ALCchar *name);
static ALCboolean ALCnullBackend_reset(ALCnullBackend *self);
static ALCboolean ALCnullBackend_start(ALCnullBackend *self);
static void ALCnullBackend_stop(ALCnullBackend *self);
static DECLARE_FORWARD2(ALCnullBackend, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
static DECLARE_FORWARD(ALCnullBackend, ALCbackend, ALCuint, availableSamples)
static DECLARE_FORWARD(ALCnullBackend, ALCbackend, ClockLatency, getClockLatency)
static DECLARE_FORWARD(ALCnullBackend, ALCbackend, void, lock)
static DECLARE_FORWARD(ALCnullBackend, ALCbackend, void, unlock)
DECLARE_DEFAULT_ALLOCATORS(ALCnullBackend)
DEFINE_ALCBACKEND_VTABLE(ALCnullBackend);
static const ALCchar nullDevice[] = "No Output";
static void ALCnullBackend_Construct(ALCnullBackend *self, ALCdevice *device)
{
ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
SET_VTABLE2(ALCnullBackend, ALCbackend, self);
ATOMIC_INIT(&self->killNow, AL_TRUE);
}
static int ALCnullBackend_mixerProc(void *ptr)
{
ALCnullBackend *self = (ALCnullBackend*)ptr;
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
struct timespec now, start;
ALuint64 avail, done;
const long restTime = (long)((ALuint64)device->UpdateSize * 1000000000 /
device->Frequency / 2);
SetRTPriority();
althrd_setname(althrd_current(), MIXER_THREAD_NAME);
done = 0;
if(altimespec_get(&start, AL_TIME_UTC) != AL_TIME_UTC)
{
ERR("Failed to get starting time\n");
return 1;
}
while(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) &&
ATOMIC_LOAD(&device->Connected, almemory_order_acquire))
{
if(altimespec_get(&now, AL_TIME_UTC) != AL_TIME_UTC)
{
ERR("Failed to get current time\n");
return 1;
}
avail = (now.tv_sec - start.tv_sec) * device->Frequency;
avail += (ALint64)(now.tv_nsec - start.tv_nsec) * device->Frequency / 1000000000;
if(avail < done)
{
/* Oops, time skipped backwards. Reset the number of samples done
* with one update available since we (likely) just came back from
* sleeping. */
done = avail - device->UpdateSize;
}
if(avail-done < device->UpdateSize)
al_nssleep(restTime);
else while(avail-done >= device->UpdateSize)
{
ALCnullBackend_lock(self);
aluMixData(device, NULL, device->UpdateSize);
ALCnullBackend_unlock(self);
done += device->UpdateSize;
}
}
return 0;
}
static ALCenum ALCnullBackend_open(ALCnullBackend *self, const ALCchar *name)
{
ALCdevice *device;
if(!name)
name = nullDevice;
else if(strcmp(name, nullDevice) != 0)
return ALC_INVALID_VALUE;
device = STATIC_CAST(ALCbackend, self)->mDevice;
alstr_copy_cstr(&device->DeviceName, name);
return ALC_NO_ERROR;
}
static ALCboolean ALCnullBackend_reset(ALCnullBackend *self)
{
SetDefaultWFXChannelOrder(STATIC_CAST(ALCbackend, self)->mDevice);
return ALC_TRUE;
}
static ALCboolean ALCnullBackend_start(ALCnullBackend *self)
{
ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release);
if(althrd_create(&self->thread, ALCnullBackend_mixerProc, self) != althrd_success)
return ALC_FALSE;
return ALC_TRUE;
}
static void ALCnullBackend_stop(ALCnullBackend *self)
{
int res;
if(ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel))
return;
althrd_join(self->thread, &res);
}
typedef struct ALCnullBackendFactory {
DERIVE_FROM_TYPE(ALCbackendFactory);
} ALCnullBackendFactory;
#define ALCNULLBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCnullBackendFactory, ALCbackendFactory) } }
ALCbackendFactory *ALCnullBackendFactory_getFactory(void);
static ALCboolean ALCnullBackendFactory_init(ALCnullBackendFactory *self);
static DECLARE_FORWARD(ALCnullBackendFactory, ALCbackendFactory, void, deinit)
static ALCboolean ALCnullBackendFactory_querySupport(ALCnullBackendFactory *self, ALCbackend_Type type);
static void ALCnullBackendFactory_probe(ALCnullBackendFactory *self, enum DevProbe type, al_string *outnames);
static ALCbackend* ALCnullBackendFactory_createBackend(ALCnullBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
DEFINE_ALCBACKENDFACTORY_VTABLE(ALCnullBackendFactory);
ALCbackendFactory *ALCnullBackendFactory_getFactory(void)
{
static ALCnullBackendFactory factory = ALCNULLBACKENDFACTORY_INITIALIZER;
return STATIC_CAST(ALCbackendFactory, &factory);
}
static ALCboolean ALCnullBackendFactory_init(ALCnullBackendFactory* UNUSED(self))
{
return ALC_TRUE;
}
static ALCboolean ALCnullBackendFactory_querySupport(ALCnullBackendFactory* UNUSED(self), ALCbackend_Type type)
{
if(type == ALCbackend_Playback)
return ALC_TRUE;
return ALC_FALSE;
}
static void ALCnullBackendFactory_probe(ALCnullBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames)
{
switch(type)
{
case ALL_DEVICE_PROBE:
case CAPTURE_DEVICE_PROBE:
alstr_append_range(outnames, nullDevice, nullDevice+sizeof(nullDevice));
break;
}
}
static ALCbackend* ALCnullBackendFactory_createBackend(ALCnullBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
{
if(type == ALCbackend_Playback)
{
ALCnullBackend *backend;
NEW_OBJ(backend, ALCnullBackend)(device);
if(!backend) return NULL;
return STATIC_CAST(ALCbackend, backend);
}
return NULL;
}
File diff suppressed because it is too large Load Diff
-869
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@@ -1,869 +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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <sys/ioctl.h>
#include <sys/types.h>
#include <sys/time.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <memory.h>
#include <unistd.h>
#include <errno.h>
#include <math.h>
#include "alMain.h"
#include "alu.h"
#include "alconfig.h"
#include "ringbuffer.h"
#include "threads.h"
#include "compat.h"
#include "backends/base.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
#if defined(SOUND_VERSION) && (SOUND_VERSION < 0x040000)
#define ALC_OSS_COMPAT
#endif
#ifndef SNDCTL_AUDIOINFO
#define ALC_OSS_COMPAT
#endif
/*
* FreeBSD strongly discourages the use of specific devices,
* such as those returned in oss_audioinfo.devnode
*/
#ifdef __FreeBSD__
#define ALC_OSS_DEVNODE_TRUC
#endif
struct oss_device {
const ALCchar *handle;
const char *path;
struct oss_device *next;
};
static struct oss_device oss_playback = {
"OSS Default",
"/dev/dsp",
NULL
};
static struct oss_device oss_capture = {
"OSS Default",
"/dev/dsp",
NULL
};
#ifdef ALC_OSS_COMPAT
#define DSP_CAP_OUTPUT 0x00020000
#define DSP_CAP_INPUT 0x00010000
static void ALCossListPopulate(struct oss_device *UNUSED(devlist), int UNUSED(type_flag))
{
}
#else
#ifndef HAVE_STRNLEN
static size_t strnlen(const char *str, size_t maxlen)
{
const char *end = memchr(str, 0, maxlen);
if(!end) return maxlen;
return end - str;
}
#endif
static void ALCossListAppend(struct oss_device *list, const char *handle, size_t hlen, const char *path, size_t plen)
{
struct oss_device *next;
struct oss_device *last;
size_t i;
/* skip the first item "OSS Default" */
last = list;
next = list->next;
#ifdef ALC_OSS_DEVNODE_TRUC
for(i = 0;i < plen;i++)
{
if(path[i] == '.')
{
if(strncmp(path + i, handle + hlen + i - plen, plen - i) == 0)
hlen = hlen + i - plen;
plen = i;
}
}
#else
(void)i;
#endif
if(handle[0] == '\0')
{
handle = path;
hlen = plen;
}
while(next != NULL)
{
if(strncmp(next->path, path, plen) == 0)
return;
last = next;
next = next->next;
}
next = (struct oss_device*)malloc(sizeof(struct oss_device) + hlen + plen + 2);
next->handle = (char*)(next + 1);
next->path = next->handle + hlen + 1;
next->next = NULL;
last->next = next;
strncpy((char*)next->handle, handle, hlen);
((char*)next->handle)[hlen] = '\0';
strncpy((char*)next->path, path, plen);
((char*)next->path)[plen] = '\0';
TRACE("Got device \"%s\", \"%s\"\n", next->handle, next->path);
}
static void ALCossListPopulate(struct oss_device *devlist, int type_flag)
{
struct oss_sysinfo si;
struct oss_audioinfo ai;
int fd, i;
if((fd=open("/dev/mixer", O_RDONLY)) < 0)
{
TRACE("Could not open /dev/mixer: %s\n", strerror(errno));
return;
}
if(ioctl(fd, SNDCTL_SYSINFO, &si) == -1)
{
TRACE("SNDCTL_SYSINFO failed: %s\n", strerror(errno));
goto done;
}
for(i = 0;i < si.numaudios;i++)
{
const char *handle;
size_t len;
ai.dev = i;
if(ioctl(fd, SNDCTL_AUDIOINFO, &ai) == -1)
{
ERR("SNDCTL_AUDIOINFO (%d) failed: %s\n", i, strerror(errno));
continue;
}
if(ai.devnode[0] == '\0')
continue;
if(ai.handle[0] != '\0')
{
len = strnlen(ai.handle, sizeof(ai.handle));
handle = ai.handle;
}
else
{
len = strnlen(ai.name, sizeof(ai.name));
handle = ai.name;
}
if((ai.caps&type_flag))
ALCossListAppend(devlist, handle, len, ai.devnode,
strnlen(ai.devnode, sizeof(ai.devnode)));
}
done:
close(fd);
}
#endif
static void ALCossListFree(struct oss_device *list)
{
struct oss_device *cur;
if(list == NULL)
return;
/* skip the first item "OSS Default" */
cur = list->next;
list->next = NULL;
while(cur != NULL)
{
struct oss_device *next = cur->next;
free(cur);
cur = next;
}
}
static int log2i(ALCuint x)
{
int y = 0;
while (x > 1)
{
x >>= 1;
y++;
}
return y;
}
typedef struct ALCplaybackOSS {
DERIVE_FROM_TYPE(ALCbackend);
int fd;
ALubyte *mix_data;
int data_size;
ATOMIC(ALenum) killNow;
althrd_t thread;
} ALCplaybackOSS;
static int ALCplaybackOSS_mixerProc(void *ptr);
static void ALCplaybackOSS_Construct(ALCplaybackOSS *self, ALCdevice *device);
static void ALCplaybackOSS_Destruct(ALCplaybackOSS *self);
static ALCenum ALCplaybackOSS_open(ALCplaybackOSS *self, const ALCchar *name);
static ALCboolean ALCplaybackOSS_reset(ALCplaybackOSS *self);
static ALCboolean ALCplaybackOSS_start(ALCplaybackOSS *self);
static void ALCplaybackOSS_stop(ALCplaybackOSS *self);
static DECLARE_FORWARD2(ALCplaybackOSS, ALCbackend, ALCenum, captureSamples, ALCvoid*, ALCuint)
static DECLARE_FORWARD(ALCplaybackOSS, ALCbackend, ALCuint, availableSamples)
static DECLARE_FORWARD(ALCplaybackOSS, ALCbackend, ClockLatency, getClockLatency)
static DECLARE_FORWARD(ALCplaybackOSS, ALCbackend, void, lock)
static DECLARE_FORWARD(ALCplaybackOSS, ALCbackend, void, unlock)
DECLARE_DEFAULT_ALLOCATORS(ALCplaybackOSS)
DEFINE_ALCBACKEND_VTABLE(ALCplaybackOSS);
static int ALCplaybackOSS_mixerProc(void *ptr)
{
ALCplaybackOSS *self = (ALCplaybackOSS*)ptr;
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
struct timeval timeout;
ALubyte *write_ptr;
ALint frame_size;
ALint to_write;
ssize_t wrote;
fd_set wfds;
int sret;
SetRTPriority();
althrd_setname(althrd_current(), MIXER_THREAD_NAME);
frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
ALCplaybackOSS_lock(self);
while(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) &&
ATOMIC_LOAD(&device->Connected, almemory_order_acquire))
{
FD_ZERO(&wfds);
FD_SET(self->fd, &wfds);
timeout.tv_sec = 1;
timeout.tv_usec = 0;
ALCplaybackOSS_unlock(self);
sret = select(self->fd+1, NULL, &wfds, NULL, &timeout);
ALCplaybackOSS_lock(self);
if(sret < 0)
{
if(errno == EINTR)
continue;
ERR("select failed: %s\n", strerror(errno));
aluHandleDisconnect(device, "Failed waiting for playback buffer: %s", strerror(errno));
break;
}
else if(sret == 0)
{
WARN("select timeout\n");
continue;
}
write_ptr = self->mix_data;
to_write = self->data_size;
aluMixData(device, write_ptr, to_write/frame_size);
while(to_write > 0 && !ATOMIC_LOAD_SEQ(&self->killNow))
{
wrote = write(self->fd, write_ptr, to_write);
if(wrote < 0)
{
if(errno == EAGAIN || errno == EWOULDBLOCK || errno == EINTR)
continue;
ERR("write failed: %s\n", strerror(errno));
aluHandleDisconnect(device, "Failed writing playback samples: %s",
strerror(errno));
break;
}
to_write -= wrote;
write_ptr += wrote;
}
}
ALCplaybackOSS_unlock(self);
return 0;
}
static void ALCplaybackOSS_Construct(ALCplaybackOSS *self, ALCdevice *device)
{
ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
SET_VTABLE2(ALCplaybackOSS, ALCbackend, self);
self->fd = -1;
ATOMIC_INIT(&self->killNow, AL_FALSE);
}
static void ALCplaybackOSS_Destruct(ALCplaybackOSS *self)
{
if(self->fd != -1)
close(self->fd);
self->fd = -1;
ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
}
static ALCenum ALCplaybackOSS_open(ALCplaybackOSS *self, const ALCchar *name)
{
struct oss_device *dev = &oss_playback;
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
if(!name || strcmp(name, dev->handle) == 0)
name = dev->handle;
else
{
if(!dev->next)
{
ALCossListPopulate(&oss_playback, DSP_CAP_OUTPUT);
dev = &oss_playback;
}
while(dev != NULL)
{
if (strcmp(dev->handle, name) == 0)
break;
dev = dev->next;
}
if(dev == NULL)
{
WARN("Could not find \"%s\" in device list\n", name);
return ALC_INVALID_VALUE;
}
}
self->fd = open(dev->path, O_WRONLY);
if(self->fd == -1)
{
ERR("Could not open %s: %s\n", dev->path, strerror(errno));
return ALC_INVALID_VALUE;
}
alstr_copy_cstr(&device->DeviceName, name);
return ALC_NO_ERROR;
}
static ALCboolean ALCplaybackOSS_reset(ALCplaybackOSS *self)
{
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
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, device->AmbiOrder);
ossSpeed = device->Frequency;
frameSize = numChannels * BytesFromDevFmt(device->FmtType);
/* According to the OSS spec, 16 bytes (log2(16)) is the minimum. */
log2FragmentSize = maxi(log2i(device->UpdateSize*frameSize), 4);
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(self->fd, SNDCTL_DSP_SETFRAGMENT, &numFragmentsLogSize);
CHECKERR(ioctl(self->fd, SNDCTL_DSP_SETFMT, &ossFormat));
CHECKERR(ioctl(self->fd, SNDCTL_DSP_CHANNELS, &numChannels));
CHECKERR(ioctl(self->fd, SNDCTL_DSP_SPEED, &ossSpeed));
CHECKERR(ioctl(self->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, device->AmbiOrder) != 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;
SetDefaultChannelOrder(device);
return ALC_TRUE;
}
static ALCboolean ALCplaybackOSS_start(ALCplaybackOSS *self)
{
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
self->data_size = device->UpdateSize * FrameSizeFromDevFmt(
device->FmtChans, device->FmtType, device->AmbiOrder
);
self->mix_data = calloc(1, self->data_size);
ATOMIC_STORE_SEQ(&self->killNow, AL_FALSE);
if(althrd_create(&self->thread, ALCplaybackOSS_mixerProc, self) != althrd_success)
{
free(self->mix_data);
self->mix_data = NULL;
return ALC_FALSE;
}
return ALC_TRUE;
}
static void ALCplaybackOSS_stop(ALCplaybackOSS *self)
{
int res;
if(ATOMIC_EXCHANGE_SEQ(&self->killNow, AL_TRUE))
return;
althrd_join(self->thread, &res);
if(ioctl(self->fd, SNDCTL_DSP_RESET) != 0)
ERR("Error resetting device: %s\n", strerror(errno));
free(self->mix_data);
self->mix_data = NULL;
}
typedef struct ALCcaptureOSS {
DERIVE_FROM_TYPE(ALCbackend);
int fd;
ll_ringbuffer_t *ring;
ATOMIC(ALenum) killNow;
althrd_t thread;
} ALCcaptureOSS;
static int ALCcaptureOSS_recordProc(void *ptr);
static void ALCcaptureOSS_Construct(ALCcaptureOSS *self, ALCdevice *device);
static void ALCcaptureOSS_Destruct(ALCcaptureOSS *self);
static ALCenum ALCcaptureOSS_open(ALCcaptureOSS *self, const ALCchar *name);
static DECLARE_FORWARD(ALCcaptureOSS, ALCbackend, ALCboolean, reset)
static ALCboolean ALCcaptureOSS_start(ALCcaptureOSS *self);
static void ALCcaptureOSS_stop(ALCcaptureOSS *self);
static ALCenum ALCcaptureOSS_captureSamples(ALCcaptureOSS *self, ALCvoid *buffer, ALCuint samples);
static ALCuint ALCcaptureOSS_availableSamples(ALCcaptureOSS *self);
static DECLARE_FORWARD(ALCcaptureOSS, ALCbackend, ClockLatency, getClockLatency)
static DECLARE_FORWARD(ALCcaptureOSS, ALCbackend, void, lock)
static DECLARE_FORWARD(ALCcaptureOSS, ALCbackend, void, unlock)
DECLARE_DEFAULT_ALLOCATORS(ALCcaptureOSS)
DEFINE_ALCBACKEND_VTABLE(ALCcaptureOSS);
static int ALCcaptureOSS_recordProc(void *ptr)
{
ALCcaptureOSS *self = (ALCcaptureOSS*)ptr;
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
struct timeval timeout;
int frame_size;
fd_set rfds;
ssize_t amt;
int sret;
SetRTPriority();
althrd_setname(althrd_current(), RECORD_THREAD_NAME);
frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
while(!ATOMIC_LOAD_SEQ(&self->killNow))
{
ll_ringbuffer_data_t vec[2];
FD_ZERO(&rfds);
FD_SET(self->fd, &rfds);
timeout.tv_sec = 1;
timeout.tv_usec = 0;
sret = select(self->fd+1, &rfds, NULL, NULL, &timeout);
if(sret < 0)
{
if(errno == EINTR)
continue;
ERR("select failed: %s\n", strerror(errno));
aluHandleDisconnect(device, "Failed to check capture samples: %s", strerror(errno));
break;
}
else if(sret == 0)
{
WARN("select timeout\n");
continue;
}
ll_ringbuffer_get_write_vector(self->ring, vec);
if(vec[0].len > 0)
{
amt = read(self->fd, vec[0].buf, vec[0].len*frame_size);
if(amt < 0)
{
ERR("read failed: %s\n", strerror(errno));
ALCcaptureOSS_lock(self);
aluHandleDisconnect(device, "Failed reading capture samples: %s", strerror(errno));
ALCcaptureOSS_unlock(self);
break;
}
ll_ringbuffer_write_advance(self->ring, amt/frame_size);
}
}
return 0;
}
static void ALCcaptureOSS_Construct(ALCcaptureOSS *self, ALCdevice *device)
{
ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
SET_VTABLE2(ALCcaptureOSS, ALCbackend, self);
self->fd = -1;
self->ring = NULL;
ATOMIC_INIT(&self->killNow, AL_FALSE);
}
static void ALCcaptureOSS_Destruct(ALCcaptureOSS *self)
{
if(self->fd != -1)
close(self->fd);
self->fd = -1;
ll_ringbuffer_free(self->ring);
self->ring = NULL;
ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
}
static ALCenum ALCcaptureOSS_open(ALCcaptureOSS *self, const ALCchar *name)
{
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
struct oss_device *dev = &oss_capture;
int numFragmentsLogSize;
int log2FragmentSize;
unsigned int periods;
audio_buf_info info;
ALuint frameSize;
int numChannels;
int ossFormat;
int ossSpeed;
char *err;
if(!name || strcmp(name, dev->handle) == 0)
name = dev->handle;
else
{
if(!dev->next)
{
ALCossListPopulate(&oss_capture, DSP_CAP_INPUT);
dev = &oss_capture;
}
while(dev != NULL)
{
if (strcmp(dev->handle, name) == 0)
break;
dev = dev->next;
}
if(dev == NULL)
{
WARN("Could not find \"%s\" in device list\n", name);
return ALC_INVALID_VALUE;
}
}
self->fd = open(dev->path, O_RDONLY);
if(self->fd == -1)
{
ERR("Could not open %s: %s\n", dev->path, 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:
ERR("%s capture samples not supported\n", DevFmtTypeString(device->FmtType));
return ALC_INVALID_VALUE;
}
periods = 4;
numChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
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(self->fd, SNDCTL_DSP_SETFRAGMENT, &numFragmentsLogSize));
CHECKERR(ioctl(self->fd, SNDCTL_DSP_SETFMT, &ossFormat));
CHECKERR(ioctl(self->fd, SNDCTL_DSP_CHANNELS, &numChannels));
CHECKERR(ioctl(self->fd, SNDCTL_DSP_SPEED, &ossSpeed));
CHECKERR(ioctl(self->fd, SNDCTL_DSP_GETISPACE, &info));
if(0)
{
err:
ERR("%s failed: %s\n", err, strerror(errno));
close(self->fd);
self->fd = -1;
return ALC_INVALID_VALUE;
}
#undef CHECKERR
if((int)ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder) != numChannels)
{
ERR("Failed to set %s, got %d channels instead\n", DevFmtChannelsString(device->FmtChans), numChannels);
close(self->fd);
self->fd = -1;
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(self->fd);
self->fd = -1;
return ALC_INVALID_VALUE;
}
self->ring = ll_ringbuffer_create(device->UpdateSize*device->NumUpdates, frameSize, false);
if(!self->ring)
{
ERR("Ring buffer create failed\n");
close(self->fd);
self->fd = -1;
return ALC_OUT_OF_MEMORY;
}
alstr_copy_cstr(&device->DeviceName, name);
return ALC_NO_ERROR;
}
static ALCboolean ALCcaptureOSS_start(ALCcaptureOSS *self)
{
ATOMIC_STORE_SEQ(&self->killNow, AL_FALSE);
if(althrd_create(&self->thread, ALCcaptureOSS_recordProc, self) != althrd_success)
return ALC_FALSE;
return ALC_TRUE;
}
static void ALCcaptureOSS_stop(ALCcaptureOSS *self)
{
int res;
if(ATOMIC_EXCHANGE_SEQ(&self->killNow, AL_TRUE))
return;
althrd_join(self->thread, &res);
if(ioctl(self->fd, SNDCTL_DSP_RESET) != 0)
ERR("Error resetting device: %s\n", strerror(errno));
}
static ALCenum ALCcaptureOSS_captureSamples(ALCcaptureOSS *self, ALCvoid *buffer, ALCuint samples)
{
ll_ringbuffer_read(self->ring, buffer, samples);
return ALC_NO_ERROR;
}
static ALCuint ALCcaptureOSS_availableSamples(ALCcaptureOSS *self)
{
return ll_ringbuffer_read_space(self->ring);
}
typedef struct ALCossBackendFactory {
DERIVE_FROM_TYPE(ALCbackendFactory);
} ALCossBackendFactory;
#define ALCOSSBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCossBackendFactory, ALCbackendFactory) } }
ALCbackendFactory *ALCossBackendFactory_getFactory(void);
static ALCboolean ALCossBackendFactory_init(ALCossBackendFactory *self);
static void ALCossBackendFactory_deinit(ALCossBackendFactory *self);
static ALCboolean ALCossBackendFactory_querySupport(ALCossBackendFactory *self, ALCbackend_Type type);
static void ALCossBackendFactory_probe(ALCossBackendFactory *self, enum DevProbe type, al_string *outnames);
static ALCbackend* ALCossBackendFactory_createBackend(ALCossBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
DEFINE_ALCBACKENDFACTORY_VTABLE(ALCossBackendFactory);
ALCbackendFactory *ALCossBackendFactory_getFactory(void)
{
static ALCossBackendFactory factory = ALCOSSBACKENDFACTORY_INITIALIZER;
return STATIC_CAST(ALCbackendFactory, &factory);
}
ALCboolean ALCossBackendFactory_init(ALCossBackendFactory* UNUSED(self))
{
ConfigValueStr(NULL, "oss", "device", &oss_playback.path);
ConfigValueStr(NULL, "oss", "capture", &oss_capture.path);
return ALC_TRUE;
}
void ALCossBackendFactory_deinit(ALCossBackendFactory* UNUSED(self))
{
ALCossListFree(&oss_playback);
ALCossListFree(&oss_capture);
}
ALCboolean ALCossBackendFactory_querySupport(ALCossBackendFactory* UNUSED(self), ALCbackend_Type type)
{
if(type == ALCbackend_Playback || type == ALCbackend_Capture)
return ALC_TRUE;
return ALC_FALSE;
}
void ALCossBackendFactory_probe(ALCossBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames)
{
struct oss_device *cur = NULL;
switch(type)
{
case ALL_DEVICE_PROBE:
ALCossListFree(&oss_playback);
ALCossListPopulate(&oss_playback, DSP_CAP_OUTPUT);
cur = &oss_playback;
break;
case CAPTURE_DEVICE_PROBE:
ALCossListFree(&oss_capture);
ALCossListPopulate(&oss_capture, DSP_CAP_INPUT);
cur = &oss_capture;
break;
}
while(cur != NULL)
{
#ifdef HAVE_STAT
struct stat buf;
if(stat(cur->path, &buf) == 0)
#endif
alstr_append_range(outnames, cur->handle, cur->handle+strlen(cur->handle)+1);
cur = cur->next;
}
}
ALCbackend* ALCossBackendFactory_createBackend(ALCossBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
{
if(type == ALCbackend_Playback)
{
ALCplaybackOSS *backend;
NEW_OBJ(backend, ALCplaybackOSS)(device);
if(!backend) return NULL;
return STATIC_CAST(ALCbackend, backend);
}
if(type == ALCbackend_Capture)
{
ALCcaptureOSS *backend;
NEW_OBJ(backend, ALCcaptureOSS)(device);
if(!backend) return NULL;
return STATIC_CAST(ALCbackend, backend);
}
return NULL;
}
-555
View File
@@ -1,555 +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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 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 "alconfig.h"
#include "ringbuffer.h"
#include "compat.h"
#include "backends/base.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_GetDefaultInputDevice);
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_GetDefaultInputDevice pPa_GetDefaultInputDevice
#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_GetDefaultInputDevice);
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 ALCportPlayback {
DERIVE_FROM_TYPE(ALCbackend);
PaStream *stream;
PaStreamParameters params;
ALuint update_size;
} ALCportPlayback;
static int ALCportPlayback_WriteCallback(const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo *timeInfo,
const PaStreamCallbackFlags statusFlags, void *userData);
static void ALCportPlayback_Construct(ALCportPlayback *self, ALCdevice *device);
static void ALCportPlayback_Destruct(ALCportPlayback *self);
static ALCenum ALCportPlayback_open(ALCportPlayback *self, const ALCchar *name);
static ALCboolean ALCportPlayback_reset(ALCportPlayback *self);
static ALCboolean ALCportPlayback_start(ALCportPlayback *self);
static void ALCportPlayback_stop(ALCportPlayback *self);
static DECLARE_FORWARD2(ALCportPlayback, ALCbackend, ALCenum, captureSamples, ALCvoid*, ALCuint)
static DECLARE_FORWARD(ALCportPlayback, ALCbackend, ALCuint, availableSamples)
static DECLARE_FORWARD(ALCportPlayback, ALCbackend, ClockLatency, getClockLatency)
static DECLARE_FORWARD(ALCportPlayback, ALCbackend, void, lock)
static DECLARE_FORWARD(ALCportPlayback, ALCbackend, void, unlock)
DECLARE_DEFAULT_ALLOCATORS(ALCportPlayback)
DEFINE_ALCBACKEND_VTABLE(ALCportPlayback);
static void ALCportPlayback_Construct(ALCportPlayback *self, ALCdevice *device)
{
ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
SET_VTABLE2(ALCportPlayback, ALCbackend, self);
self->stream = NULL;
}
static void ALCportPlayback_Destruct(ALCportPlayback *self)
{
PaError err = self->stream ? Pa_CloseStream(self->stream) : paNoError;
if(err != paNoError)
ERR("Error closing stream: %s\n", Pa_GetErrorText(err));
self->stream = NULL;
ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
}
static int ALCportPlayback_WriteCallback(const void *UNUSED(inputBuffer), void *outputBuffer,
unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo *UNUSED(timeInfo),
const PaStreamCallbackFlags UNUSED(statusFlags), void *userData)
{
ALCportPlayback *self = userData;
ALCportPlayback_lock(self);
aluMixData(STATIC_CAST(ALCbackend, self)->mDevice, outputBuffer, framesPerBuffer);
ALCportPlayback_unlock(self);
return 0;
}
static ALCenum ALCportPlayback_open(ALCportPlayback *self, const ALCchar *name)
{
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
PaError err;
if(!name)
name = pa_device;
else if(strcmp(name, pa_device) != 0)
return ALC_INVALID_VALUE;
self->update_size = device->UpdateSize;
self->params.device = -1;
if(!ConfigValueInt(NULL, "port", "device", &self->params.device) ||
self->params.device < 0)
self->params.device = Pa_GetDefaultOutputDevice();
self->params.suggestedLatency = (device->UpdateSize*device->NumUpdates) /
(float)device->Frequency;
self->params.hostApiSpecificStreamInfo = NULL;
self->params.channelCount = ((device->FmtChans == DevFmtMono) ? 1 : 2);
switch(device->FmtType)
{
case DevFmtByte:
self->params.sampleFormat = paInt8;
break;
case DevFmtUByte:
self->params.sampleFormat = paUInt8;
break;
case DevFmtUShort:
/* fall-through */
case DevFmtShort:
self->params.sampleFormat = paInt16;
break;
case DevFmtUInt:
/* fall-through */
case DevFmtInt:
self->params.sampleFormat = paInt32;
break;
case DevFmtFloat:
self->params.sampleFormat = paFloat32;
break;
}
retry_open:
err = Pa_OpenStream(&self->stream, NULL, &self->params,
device->Frequency, device->UpdateSize, paNoFlag,
ALCportPlayback_WriteCallback, self
);
if(err != paNoError)
{
if(self->params.sampleFormat == paFloat32)
{
self->params.sampleFormat = paInt16;
goto retry_open;
}
ERR("Pa_OpenStream() returned an error: %s\n", Pa_GetErrorText(err));
return ALC_INVALID_VALUE;
}
alstr_copy_cstr(&device->DeviceName, name);
return ALC_NO_ERROR;
}
static ALCboolean ALCportPlayback_reset(ALCportPlayback *self)
{
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
const PaStreamInfo *streamInfo;
streamInfo = Pa_GetStreamInfo(self->stream);
device->Frequency = streamInfo->sampleRate;
device->UpdateSize = self->update_size;
if(self->params.sampleFormat == paInt8)
device->FmtType = DevFmtByte;
else if(self->params.sampleFormat == paUInt8)
device->FmtType = DevFmtUByte;
else if(self->params.sampleFormat == paInt16)
device->FmtType = DevFmtShort;
else if(self->params.sampleFormat == paInt32)
device->FmtType = DevFmtInt;
else if(self->params.sampleFormat == paFloat32)
device->FmtType = DevFmtFloat;
else
{
ERR("Unexpected sample format: 0x%lx\n", self->params.sampleFormat);
return ALC_FALSE;
}
if(self->params.channelCount == 2)
device->FmtChans = DevFmtStereo;
else if(self->params.channelCount == 1)
device->FmtChans = DevFmtMono;
else
{
ERR("Unexpected channel count: %u\n", self->params.channelCount);
return ALC_FALSE;
}
SetDefaultChannelOrder(device);
return ALC_TRUE;
}
static ALCboolean ALCportPlayback_start(ALCportPlayback *self)
{
PaError err;
err = Pa_StartStream(self->stream);
if(err != paNoError)
{
ERR("Pa_StartStream() returned an error: %s\n", Pa_GetErrorText(err));
return ALC_FALSE;
}
return ALC_TRUE;
}
static void ALCportPlayback_stop(ALCportPlayback *self)
{
PaError err = Pa_StopStream(self->stream);
if(err != paNoError)
ERR("Error stopping stream: %s\n", Pa_GetErrorText(err));
}
typedef struct ALCportCapture {
DERIVE_FROM_TYPE(ALCbackend);
PaStream *stream;
PaStreamParameters params;
ll_ringbuffer_t *ring;
} ALCportCapture;
static int ALCportCapture_ReadCallback(const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo *timeInfo,
const PaStreamCallbackFlags statusFlags, void *userData);
static void ALCportCapture_Construct(ALCportCapture *self, ALCdevice *device);
static void ALCportCapture_Destruct(ALCportCapture *self);
static ALCenum ALCportCapture_open(ALCportCapture *self, const ALCchar *name);
static DECLARE_FORWARD(ALCportCapture, ALCbackend, ALCboolean, reset)
static ALCboolean ALCportCapture_start(ALCportCapture *self);
static void ALCportCapture_stop(ALCportCapture *self);
static ALCenum ALCportCapture_captureSamples(ALCportCapture *self, ALCvoid *buffer, ALCuint samples);
static ALCuint ALCportCapture_availableSamples(ALCportCapture *self);
static DECLARE_FORWARD(ALCportCapture, ALCbackend, ClockLatency, getClockLatency)
static DECLARE_FORWARD(ALCportCapture, ALCbackend, void, lock)
static DECLARE_FORWARD(ALCportCapture, ALCbackend, void, unlock)
DECLARE_DEFAULT_ALLOCATORS(ALCportCapture)
DEFINE_ALCBACKEND_VTABLE(ALCportCapture);
static void ALCportCapture_Construct(ALCportCapture *self, ALCdevice *device)
{
ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
SET_VTABLE2(ALCportCapture, ALCbackend, self);
self->stream = NULL;
self->ring = NULL;
}
static void ALCportCapture_Destruct(ALCportCapture *self)
{
PaError err = self->stream ? Pa_CloseStream(self->stream) : paNoError;
if(err != paNoError)
ERR("Error closing stream: %s\n", Pa_GetErrorText(err));
self->stream = NULL;
ll_ringbuffer_free(self->ring);
self->ring = NULL;
ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
}
static int ALCportCapture_ReadCallback(const void *inputBuffer, void *UNUSED(outputBuffer),
unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo *UNUSED(timeInfo),
const PaStreamCallbackFlags UNUSED(statusFlags), void *userData)
{
ALCportCapture *self = userData;
size_t writable = ll_ringbuffer_write_space(self->ring);
if(framesPerBuffer > writable)
framesPerBuffer = writable;
ll_ringbuffer_write(self->ring, inputBuffer, framesPerBuffer);
return 0;
}
static ALCenum ALCportCapture_open(ALCportCapture *self, const ALCchar *name)
{
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
ALuint samples, frame_size;
PaError err;
if(!name)
name = pa_device;
else if(strcmp(name, pa_device) != 0)
return ALC_INVALID_VALUE;
samples = device->UpdateSize * device->NumUpdates;
samples = maxu(samples, 100 * device->Frequency / 1000);
frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
self->ring = ll_ringbuffer_create(samples, frame_size, false);
if(self->ring == NULL) return ALC_INVALID_VALUE;
self->params.device = -1;
if(!ConfigValueInt(NULL, "port", "capture", &self->params.device) ||
self->params.device < 0)
self->params.device = Pa_GetDefaultInputDevice();
self->params.suggestedLatency = 0.0f;
self->params.hostApiSpecificStreamInfo = NULL;
switch(device->FmtType)
{
case DevFmtByte:
self->params.sampleFormat = paInt8;
break;
case DevFmtUByte:
self->params.sampleFormat = paUInt8;
break;
case DevFmtShort:
self->params.sampleFormat = paInt16;
break;
case DevFmtInt:
self->params.sampleFormat = paInt32;
break;
case DevFmtFloat:
self->params.sampleFormat = paFloat32;
break;
case DevFmtUInt:
case DevFmtUShort:
ERR("%s samples not supported\n", DevFmtTypeString(device->FmtType));
return ALC_INVALID_VALUE;
}
self->params.channelCount = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
err = Pa_OpenStream(&self->stream, &self->params, NULL,
device->Frequency, paFramesPerBufferUnspecified, paNoFlag,
ALCportCapture_ReadCallback, self
);
if(err != paNoError)
{
ERR("Pa_OpenStream() returned an error: %s\n", Pa_GetErrorText(err));
return ALC_INVALID_VALUE;
}
alstr_copy_cstr(&device->DeviceName, name);
return ALC_NO_ERROR;
}
static ALCboolean ALCportCapture_start(ALCportCapture *self)
{
PaError err = Pa_StartStream(self->stream);
if(err != paNoError)
{
ERR("Error starting stream: %s\n", Pa_GetErrorText(err));
return ALC_FALSE;
}
return ALC_TRUE;
}
static void ALCportCapture_stop(ALCportCapture *self)
{
PaError err = Pa_StopStream(self->stream);
if(err != paNoError)
ERR("Error stopping stream: %s\n", Pa_GetErrorText(err));
}
static ALCuint ALCportCapture_availableSamples(ALCportCapture *self)
{
return ll_ringbuffer_read_space(self->ring);
}
static ALCenum ALCportCapture_captureSamples(ALCportCapture *self, ALCvoid *buffer, ALCuint samples)
{
ll_ringbuffer_read(self->ring, buffer, samples);
return ALC_NO_ERROR;
}
typedef struct ALCportBackendFactory {
DERIVE_FROM_TYPE(ALCbackendFactory);
} ALCportBackendFactory;
#define ALCPORTBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCportBackendFactory, ALCbackendFactory) } }
static ALCboolean ALCportBackendFactory_init(ALCportBackendFactory *self);
static void ALCportBackendFactory_deinit(ALCportBackendFactory *self);
static ALCboolean ALCportBackendFactory_querySupport(ALCportBackendFactory *self, ALCbackend_Type type);
static void ALCportBackendFactory_probe(ALCportBackendFactory *self, enum DevProbe type, al_string *outnames);
static ALCbackend* ALCportBackendFactory_createBackend(ALCportBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
DEFINE_ALCBACKENDFACTORY_VTABLE(ALCportBackendFactory);
static ALCboolean ALCportBackendFactory_init(ALCportBackendFactory* UNUSED(self))
{
if(!pa_load())
return ALC_FALSE;
return ALC_TRUE;
}
static void ALCportBackendFactory_deinit(ALCportBackendFactory* UNUSED(self))
{
#ifdef HAVE_DYNLOAD
if(pa_handle)
{
Pa_Terminate();
CloseLib(pa_handle);
pa_handle = NULL;
}
#else
Pa_Terminate();
#endif
}
static ALCboolean ALCportBackendFactory_querySupport(ALCportBackendFactory* UNUSED(self), ALCbackend_Type type)
{
if(type == ALCbackend_Playback || type == ALCbackend_Capture)
return ALC_TRUE;
return ALC_FALSE;
}
static void ALCportBackendFactory_probe(ALCportBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames)
{
switch(type)
{
case ALL_DEVICE_PROBE:
case CAPTURE_DEVICE_PROBE:
alstr_append_range(outnames, pa_device, pa_device+sizeof(pa_device));
break;
}
}
static ALCbackend* ALCportBackendFactory_createBackend(ALCportBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
{
if(type == ALCbackend_Playback)
{
ALCportPlayback *backend;
NEW_OBJ(backend, ALCportPlayback)(device);
if(!backend) return NULL;
return STATIC_CAST(ALCbackend, backend);
}
if(type == ALCbackend_Capture)
{
ALCportCapture *backend;
NEW_OBJ(backend, ALCportCapture)(device);
if(!backend) return NULL;
return STATIC_CAST(ALCbackend, backend);
}
return NULL;
}
ALCbackendFactory *ALCportBackendFactory_getFactory(void)
{
static ALCportBackendFactory factory = ALCPORTBACKENDFACTORY_INITIALIZER;
return STATIC_CAST(ALCbackendFactory, &factory);
}
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@@ -1,288 +0,0 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 2018 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdlib.h>
#include <SDL2/SDL.h>
#include "alMain.h"
#include "alu.h"
#include "threads.h"
#include "compat.h"
#include "backends/base.h"
#ifdef _WIN32
#define DEVNAME_PREFIX "OpenAL Soft on "
#else
#define DEVNAME_PREFIX ""
#endif
typedef struct ALCsdl2Backend {
DERIVE_FROM_TYPE(ALCbackend);
SDL_AudioDeviceID deviceID;
ALsizei frameSize;
ALuint Frequency;
enum DevFmtChannels FmtChans;
enum DevFmtType FmtType;
ALuint UpdateSize;
} ALCsdl2Backend;
static void ALCsdl2Backend_Construct(ALCsdl2Backend *self, ALCdevice *device);
static void ALCsdl2Backend_Destruct(ALCsdl2Backend *self);
static ALCenum ALCsdl2Backend_open(ALCsdl2Backend *self, const ALCchar *name);
static ALCboolean ALCsdl2Backend_reset(ALCsdl2Backend *self);
static ALCboolean ALCsdl2Backend_start(ALCsdl2Backend *self);
static void ALCsdl2Backend_stop(ALCsdl2Backend *self);
static DECLARE_FORWARD2(ALCsdl2Backend, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
static DECLARE_FORWARD(ALCsdl2Backend, ALCbackend, ALCuint, availableSamples)
static DECLARE_FORWARD(ALCsdl2Backend, ALCbackend, ClockLatency, getClockLatency)
static void ALCsdl2Backend_lock(ALCsdl2Backend *self);
static void ALCsdl2Backend_unlock(ALCsdl2Backend *self);
DECLARE_DEFAULT_ALLOCATORS(ALCsdl2Backend)
DEFINE_ALCBACKEND_VTABLE(ALCsdl2Backend);
static const ALCchar defaultDeviceName[] = DEVNAME_PREFIX "Default Device";
static void ALCsdl2Backend_Construct(ALCsdl2Backend *self, ALCdevice *device)
{
ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
SET_VTABLE2(ALCsdl2Backend, ALCbackend, self);
self->deviceID = 0;
self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
self->Frequency = device->Frequency;
self->FmtChans = device->FmtChans;
self->FmtType = device->FmtType;
self->UpdateSize = device->UpdateSize;
}
static void ALCsdl2Backend_Destruct(ALCsdl2Backend *self)
{
if(self->deviceID)
SDL_CloseAudioDevice(self->deviceID);
self->deviceID = 0;
ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
}
static void ALCsdl2Backend_audioCallback(void *ptr, Uint8 *stream, int len)
{
ALCsdl2Backend *self = (ALCsdl2Backend*)ptr;
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
assert((len % self->frameSize) == 0);
aluMixData(device, stream, len / self->frameSize);
}
static ALCenum ALCsdl2Backend_open(ALCsdl2Backend *self, const ALCchar *name)
{
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
SDL_AudioSpec want, have;
SDL_zero(want);
SDL_zero(have);
want.freq = device->Frequency;
switch(device->FmtType)
{
case DevFmtUByte: want.format = AUDIO_U8; break;
case DevFmtByte: want.format = AUDIO_S8; break;
case DevFmtUShort: want.format = AUDIO_U16SYS; break;
case DevFmtShort: want.format = AUDIO_S16SYS; break;
case DevFmtUInt: /* fall-through */
case DevFmtInt: want.format = AUDIO_S32SYS; break;
case DevFmtFloat: want.format = AUDIO_F32; break;
}
want.channels = (device->FmtChans == DevFmtMono) ? 1 : 2;
want.samples = device->UpdateSize;
want.callback = ALCsdl2Backend_audioCallback;
want.userdata = self;
/* Passing NULL to SDL_OpenAudioDevice opens a default, which isn't
* necessarily the first in the list.
*/
if(!name || strcmp(name, defaultDeviceName) == 0)
self->deviceID = SDL_OpenAudioDevice(NULL, SDL_FALSE, &want, &have,
SDL_AUDIO_ALLOW_ANY_CHANGE);
else
{
const size_t prefix_len = strlen(DEVNAME_PREFIX);
if(strncmp(name, DEVNAME_PREFIX, prefix_len) == 0)
self->deviceID = SDL_OpenAudioDevice(name+prefix_len, SDL_FALSE, &want, &have,
SDL_AUDIO_ALLOW_ANY_CHANGE);
else
self->deviceID = SDL_OpenAudioDevice(name, SDL_FALSE, &want, &have,
SDL_AUDIO_ALLOW_ANY_CHANGE);
}
if(self->deviceID == 0)
return ALC_INVALID_VALUE;
device->Frequency = have.freq;
if(have.channels == 1)
device->FmtChans = DevFmtMono;
else if(have.channels == 2)
device->FmtChans = DevFmtStereo;
else
{
ERR("Got unhandled SDL channel count: %d\n", (int)have.channels);
return ALC_INVALID_VALUE;
}
switch(have.format)
{
case AUDIO_U8: device->FmtType = DevFmtUByte; break;
case AUDIO_S8: device->FmtType = DevFmtByte; break;
case AUDIO_U16SYS: device->FmtType = DevFmtUShort; break;
case AUDIO_S16SYS: device->FmtType = DevFmtShort; break;
case AUDIO_S32SYS: device->FmtType = DevFmtInt; break;
case AUDIO_F32SYS: device->FmtType = DevFmtFloat; break;
default:
ERR("Got unsupported SDL format: 0x%04x\n", have.format);
return ALC_INVALID_VALUE;
}
device->UpdateSize = have.samples;
device->NumUpdates = 2; /* SDL always (tries to) use two periods. */
self->frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
self->Frequency = device->Frequency;
self->FmtChans = device->FmtChans;
self->FmtType = device->FmtType;
self->UpdateSize = device->UpdateSize;
alstr_copy_cstr(&device->DeviceName, name ? name : defaultDeviceName);
return ALC_NO_ERROR;
}
static ALCboolean ALCsdl2Backend_reset(ALCsdl2Backend *self)
{
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
device->Frequency = self->Frequency;
device->FmtChans = self->FmtChans;
device->FmtType = self->FmtType;
device->UpdateSize = self->UpdateSize;
device->NumUpdates = 2;
SetDefaultWFXChannelOrder(device);
return ALC_TRUE;
}
static ALCboolean ALCsdl2Backend_start(ALCsdl2Backend *self)
{
SDL_PauseAudioDevice(self->deviceID, 0);
return ALC_TRUE;
}
static void ALCsdl2Backend_stop(ALCsdl2Backend *self)
{
SDL_PauseAudioDevice(self->deviceID, 1);
}
static void ALCsdl2Backend_lock(ALCsdl2Backend *self)
{
SDL_LockAudioDevice(self->deviceID);
}
static void ALCsdl2Backend_unlock(ALCsdl2Backend *self)
{
SDL_UnlockAudioDevice(self->deviceID);
}
typedef struct ALCsdl2BackendFactory {
DERIVE_FROM_TYPE(ALCbackendFactory);
} ALCsdl2BackendFactory;
#define ALCsdl2BACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCsdl2BackendFactory, ALCbackendFactory) } }
ALCbackendFactory *ALCsdl2BackendFactory_getFactory(void);
static ALCboolean ALCsdl2BackendFactory_init(ALCsdl2BackendFactory *self);
static void ALCsdl2BackendFactory_deinit(ALCsdl2BackendFactory *self);
static ALCboolean ALCsdl2BackendFactory_querySupport(ALCsdl2BackendFactory *self, ALCbackend_Type type);
static void ALCsdl2BackendFactory_probe(ALCsdl2BackendFactory *self, enum DevProbe type, al_string *outnames);
static ALCbackend* ALCsdl2BackendFactory_createBackend(ALCsdl2BackendFactory *self, ALCdevice *device, ALCbackend_Type type);
DEFINE_ALCBACKENDFACTORY_VTABLE(ALCsdl2BackendFactory);
ALCbackendFactory *ALCsdl2BackendFactory_getFactory(void)
{
static ALCsdl2BackendFactory factory = ALCsdl2BACKENDFACTORY_INITIALIZER;
return STATIC_CAST(ALCbackendFactory, &factory);
}
static ALCboolean ALCsdl2BackendFactory_init(ALCsdl2BackendFactory* UNUSED(self))
{
if(SDL_InitSubSystem(SDL_INIT_AUDIO) == 0)
return AL_TRUE;
return ALC_FALSE;
}
static void ALCsdl2BackendFactory_deinit(ALCsdl2BackendFactory* UNUSED(self))
{
SDL_QuitSubSystem(SDL_INIT_AUDIO);
}
static ALCboolean ALCsdl2BackendFactory_querySupport(ALCsdl2BackendFactory* UNUSED(self), ALCbackend_Type type)
{
if(type == ALCbackend_Playback)
return ALC_TRUE;
return ALC_FALSE;
}
static void ALCsdl2BackendFactory_probe(ALCsdl2BackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames)
{
int num_devices, i;
al_string name;
if(type != ALL_DEVICE_PROBE)
return;
AL_STRING_INIT(name);
num_devices = SDL_GetNumAudioDevices(SDL_FALSE);
alstr_append_range(outnames, defaultDeviceName, defaultDeviceName+sizeof(defaultDeviceName));
for(i = 0;i < num_devices;++i)
{
alstr_copy_cstr(&name, DEVNAME_PREFIX);
alstr_append_cstr(&name, SDL_GetAudioDeviceName(i, SDL_FALSE));
if(!alstr_empty(name))
alstr_append_range(outnames, VECTOR_BEGIN(name), VECTOR_END(name)+1);
}
alstr_reset(&name);
}
static ALCbackend* ALCsdl2BackendFactory_createBackend(ALCsdl2BackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
{
if(type == ALCbackend_Playback)
{
ALCsdl2Backend *backend;
NEW_OBJ(backend, ALCsdl2Backend)(device);
if(!backend) return NULL;
return STATIC_CAST(ALCbackend, backend);
}
return NULL;
}
-600
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@@ -1,600 +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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 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 "threads.h"
#include "ringbuffer.h"
#include "backends/base.h"
#include <sndio.h>
static const ALCchar sndio_device[] = "SndIO Default";
typedef struct SndioPlayback {
DERIVE_FROM_TYPE(ALCbackend);
struct sio_hdl *sndHandle;
ALvoid *mix_data;
ALsizei data_size;
ATOMIC(int) killNow;
althrd_t thread;
} SndioPlayback;
static int SndioPlayback_mixerProc(void *ptr);
static void SndioPlayback_Construct(SndioPlayback *self, ALCdevice *device);
static void SndioPlayback_Destruct(SndioPlayback *self);
static ALCenum SndioPlayback_open(SndioPlayback *self, const ALCchar *name);
static ALCboolean SndioPlayback_reset(SndioPlayback *self);
static ALCboolean SndioPlayback_start(SndioPlayback *self);
static void SndioPlayback_stop(SndioPlayback *self);
static DECLARE_FORWARD2(SndioPlayback, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
static DECLARE_FORWARD(SndioPlayback, ALCbackend, ALCuint, availableSamples)
static DECLARE_FORWARD(SndioPlayback, ALCbackend, ClockLatency, getClockLatency)
static DECLARE_FORWARD(SndioPlayback, ALCbackend, void, lock)
static DECLARE_FORWARD(SndioPlayback, ALCbackend, void, unlock)
DECLARE_DEFAULT_ALLOCATORS(SndioPlayback)
DEFINE_ALCBACKEND_VTABLE(SndioPlayback);
static void SndioPlayback_Construct(SndioPlayback *self, ALCdevice *device)
{
ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
SET_VTABLE2(SndioPlayback, ALCbackend, self);
self->sndHandle = NULL;
self->mix_data = NULL;
ATOMIC_INIT(&self->killNow, AL_TRUE);
}
static void SndioPlayback_Destruct(SndioPlayback *self)
{
if(self->sndHandle)
sio_close(self->sndHandle);
self->sndHandle = NULL;
al_free(self->mix_data);
self->mix_data = NULL;
ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
}
static int SndioPlayback_mixerProc(void *ptr)
{
SndioPlayback *self = (SndioPlayback*)ptr;
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
ALsizei frameSize;
size_t wrote;
SetRTPriority();
althrd_setname(althrd_current(), MIXER_THREAD_NAME);
frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
while(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) &&
ATOMIC_LOAD(&device->Connected, almemory_order_acquire))
{
ALsizei len = self->data_size;
ALubyte *WritePtr = self->mix_data;
SndioPlayback_lock(self);
aluMixData(device, WritePtr, len/frameSize);
SndioPlayback_unlock(self);
while(len > 0 && !ATOMIC_LOAD(&self->killNow, almemory_order_acquire))
{
wrote = sio_write(self->sndHandle, WritePtr, len);
if(wrote == 0)
{
ERR("sio_write failed\n");
ALCdevice_Lock(device);
aluHandleDisconnect(device, "Failed to write playback samples");
ALCdevice_Unlock(device);
break;
}
len -= wrote;
WritePtr += wrote;
}
}
return 0;
}
static ALCenum SndioPlayback_open(SndioPlayback *self, const ALCchar *name)
{
ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
if(!name)
name = sndio_device;
else if(strcmp(name, sndio_device) != 0)
return ALC_INVALID_VALUE;
self->sndHandle = sio_open(NULL, SIO_PLAY, 0);
if(self->sndHandle == NULL)
{
ERR("Could not open device\n");
return ALC_INVALID_VALUE;
}
alstr_copy_cstr(&device->DeviceName, name);
return ALC_NO_ERROR;
}
static ALCboolean SndioPlayback_reset(SndioPlayback *self)
{
ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
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(self->sndHandle, &par) || !sio_getpar(self->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 SndioPlayback_start(SndioPlayback *self)
{
ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
self->data_size = device->UpdateSize * FrameSizeFromDevFmt(
device->FmtChans, device->FmtType, device->AmbiOrder
);
al_free(self->mix_data);
self->mix_data = al_calloc(16, self->data_size);
if(!sio_start(self->sndHandle))
{
ERR("Error starting playback\n");
return ALC_FALSE;
}
ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release);
if(althrd_create(&self->thread, SndioPlayback_mixerProc, self) != althrd_success)
{
sio_stop(self->sndHandle);
return ALC_FALSE;
}
return ALC_TRUE;
}
static void SndioPlayback_stop(SndioPlayback *self)
{
int res;
if(ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel))
return;
althrd_join(self->thread, &res);
if(!sio_stop(self->sndHandle))
ERR("Error stopping device\n");
al_free(self->mix_data);
self->mix_data = NULL;
}
typedef struct SndioCapture {
DERIVE_FROM_TYPE(ALCbackend);
struct sio_hdl *sndHandle;
ll_ringbuffer_t *ring;
ATOMIC(int) killNow;
althrd_t thread;
} SndioCapture;
static int SndioCapture_recordProc(void *ptr);
static void SndioCapture_Construct(SndioCapture *self, ALCdevice *device);
static void SndioCapture_Destruct(SndioCapture *self);
static ALCenum SndioCapture_open(SndioCapture *self, const ALCchar *name);
static DECLARE_FORWARD(SndioCapture, ALCbackend, ALCboolean, reset)
static ALCboolean SndioCapture_start(SndioCapture *self);
static void SndioCapture_stop(SndioCapture *self);
static ALCenum SndioCapture_captureSamples(SndioCapture *self, void *buffer, ALCuint samples);
static ALCuint SndioCapture_availableSamples(SndioCapture *self);
static DECLARE_FORWARD(SndioCapture, ALCbackend, ClockLatency, getClockLatency)
static DECLARE_FORWARD(SndioCapture, ALCbackend, void, lock)
static DECLARE_FORWARD(SndioCapture, ALCbackend, void, unlock)
DECLARE_DEFAULT_ALLOCATORS(SndioCapture)
DEFINE_ALCBACKEND_VTABLE(SndioCapture);
static void SndioCapture_Construct(SndioCapture *self, ALCdevice *device)
{
ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
SET_VTABLE2(SndioCapture, ALCbackend, self);
self->sndHandle = NULL;
self->ring = NULL;
ATOMIC_INIT(&self->killNow, AL_TRUE);
}
static void SndioCapture_Destruct(SndioCapture *self)
{
if(self->sndHandle)
sio_close(self->sndHandle);
self->sndHandle = NULL;
ll_ringbuffer_free(self->ring);
self->ring = NULL;
ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
}
static int SndioCapture_recordProc(void* ptr)
{
SndioCapture *self = (SndioCapture*)ptr;
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
ALsizei frameSize;
SetRTPriority();
althrd_setname(althrd_current(), RECORD_THREAD_NAME);
frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
while(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) &&
ATOMIC_LOAD(&device->Connected, almemory_order_acquire))
{
ll_ringbuffer_data_t data[2];
size_t total, todo;
ll_ringbuffer_get_write_vector(self->ring, data);
todo = data[0].len + data[1].len;
if(todo == 0)
{
static char junk[4096];
sio_read(self->sndHandle, junk, minz(sizeof(junk)/frameSize, device->UpdateSize)*frameSize);
continue;
}
total = 0;
data[0].len *= frameSize;
data[1].len *= frameSize;
todo = minz(todo, device->UpdateSize) * frameSize;
while(total < todo)
{
size_t got;
if(!data[0].len)
data[0] = data[1];
got = sio_read(self->sndHandle, data[0].buf, minz(todo-total, data[0].len));
if(!got)
{
SndioCapture_lock(self);
aluHandleDisconnect(device, "Failed to read capture samples");
SndioCapture_unlock(self);
break;
}
data[0].buf += got;
data[0].len -= got;
total += got;
}
ll_ringbuffer_write_advance(self->ring, total / frameSize);
}
return 0;
}
static ALCenum SndioCapture_open(SndioCapture *self, const ALCchar *name)
{
ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
struct sio_par par;
if(!name)
name = sndio_device;
else if(strcmp(name, sndio_device) != 0)
return ALC_INVALID_VALUE;
self->sndHandle = sio_open(NULL, SIO_REC, 0);
if(self->sndHandle == NULL)
{
ERR("Could not open device\n");
return ALC_INVALID_VALUE;
}
sio_initpar(&par);
switch(device->FmtType)
{
case DevFmtByte:
par.bps = 1;
par.sig = 1;
break;
case DevFmtUByte:
par.bps = 1;
par.sig = 0;
break;
case DevFmtShort:
par.bps = 2;
par.sig = 1;
break;
case DevFmtUShort:
par.bps = 2;
par.sig = 0;
break;
case DevFmtInt:
par.bps = 4;
par.sig = 1;
break;
case DevFmtUInt:
par.bps = 4;
par.sig = 0;
break;
case DevFmtFloat:
ERR("%s capture samples not supported\n", DevFmtTypeString(device->FmtType));
return ALC_INVALID_VALUE;
}
par.bits = par.bps * 8;
par.le = SIO_LE_NATIVE;
par.msb = SIO_LE_NATIVE ? 0 : 1;
par.rchan = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
par.rate = device->Frequency;
par.appbufsz = maxu(device->UpdateSize*device->NumUpdates, (device->Frequency+9)/10);
par.round = clampu(par.appbufsz/device->NumUpdates, (device->Frequency+99)/100,
(device->Frequency+19)/20);
device->UpdateSize = par.round;
device->NumUpdates = maxu(par.appbufsz/par.round, 1);
if(!sio_setpar(self->sndHandle, &par) || !sio_getpar(self->sndHandle, &par))
{
ERR("Failed to set device parameters\n");
return ALC_INVALID_VALUE;
}
if(par.bits != par.bps*8)
{
ERR("Padded samples not supported (%u of %u bits)\n", par.bits, par.bps*8);
return ALC_INVALID_VALUE;
}
if(!((device->FmtType == DevFmtByte && par.bits == 8 && par.sig != 0) ||
(device->FmtType == DevFmtUByte && par.bits == 8 && par.sig == 0) ||
(device->FmtType == DevFmtShort && par.bits == 16 && par.sig != 0) ||
(device->FmtType == DevFmtUShort && par.bits == 16 && par.sig == 0) ||
(device->FmtType == DevFmtInt && par.bits == 32 && par.sig != 0) ||
(device->FmtType == DevFmtUInt && par.bits == 32 && par.sig == 0)) ||
ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder) != (ALsizei)par.rchan ||
device->Frequency != par.rate)
{
ERR("Failed to set format %s %s %uhz, got %c%u %u-channel %uhz instead\n",
DevFmtTypeString(device->FmtType), DevFmtChannelsString(device->FmtChans),
device->Frequency, par.sig?'s':'u', par.bits, par.rchan, par.rate);
return ALC_INVALID_VALUE;
}
self->ring = ll_ringbuffer_create(device->UpdateSize*device->NumUpdates, par.bps*par.rchan, 0);
if(!self->ring)
{
ERR("Failed to allocate %u-byte ringbuffer\n",
device->UpdateSize*device->NumUpdates*par.bps*par.rchan);
return ALC_OUT_OF_MEMORY;
}
SetDefaultChannelOrder(device);
alstr_copy_cstr(&device->DeviceName, name);
return ALC_NO_ERROR;
}
static ALCboolean SndioCapture_start(SndioCapture *self)
{
if(!sio_start(self->sndHandle))
{
ERR("Error starting playback\n");
return ALC_FALSE;
}
ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release);
if(althrd_create(&self->thread, SndioCapture_recordProc, self) != althrd_success)
{
sio_stop(self->sndHandle);
return ALC_FALSE;
}
return ALC_TRUE;
}
static void SndioCapture_stop(SndioCapture *self)
{
int res;
if(ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel))
return;
althrd_join(self->thread, &res);
if(!sio_stop(self->sndHandle))
ERR("Error stopping device\n");
}
static ALCenum SndioCapture_captureSamples(SndioCapture *self, void *buffer, ALCuint samples)
{
ll_ringbuffer_read(self->ring, buffer, samples);
return ALC_NO_ERROR;
}
static ALCuint SndioCapture_availableSamples(SndioCapture *self)
{
return ll_ringbuffer_read_space(self->ring);
}
typedef struct SndioBackendFactory {
DERIVE_FROM_TYPE(ALCbackendFactory);
} SndioBackendFactory;
#define SNDIOBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(SndioBackendFactory, ALCbackendFactory) } }
ALCbackendFactory *SndioBackendFactory_getFactory(void);
static ALCboolean SndioBackendFactory_init(SndioBackendFactory *self);
static DECLARE_FORWARD(SndioBackendFactory, ALCbackendFactory, void, deinit)
static ALCboolean SndioBackendFactory_querySupport(SndioBackendFactory *self, ALCbackend_Type type);
static void SndioBackendFactory_probe(SndioBackendFactory *self, enum DevProbe type, al_string *outnames);
static ALCbackend* SndioBackendFactory_createBackend(SndioBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
DEFINE_ALCBACKENDFACTORY_VTABLE(SndioBackendFactory);
ALCbackendFactory *SndioBackendFactory_getFactory(void)
{
static SndioBackendFactory factory = SNDIOBACKENDFACTORY_INITIALIZER;
return STATIC_CAST(ALCbackendFactory, &factory);
}
static ALCboolean SndioBackendFactory_init(SndioBackendFactory* UNUSED(self))
{
/* No dynamic loading */
return ALC_TRUE;
}
static ALCboolean SndioBackendFactory_querySupport(SndioBackendFactory* UNUSED(self), ALCbackend_Type type)
{
if(type == ALCbackend_Playback || type == ALCbackend_Capture)
return ALC_TRUE;
return ALC_FALSE;
}
static void SndioBackendFactory_probe(SndioBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames)
{
switch(type)
{
case ALL_DEVICE_PROBE:
case CAPTURE_DEVICE_PROBE:
alstr_append_range(outnames, sndio_device, sndio_device+sizeof(sndio_device));
break;
}
}
static ALCbackend* SndioBackendFactory_createBackend(SndioBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
{
if(type == ALCbackend_Playback)
{
SndioPlayback *backend;
NEW_OBJ(backend, SndioPlayback)(device);
if(!backend) return NULL;
return STATIC_CAST(ALCbackend, backend);
}
if(type == ALCbackend_Capture)
{
SndioCapture *backend;
NEW_OBJ(backend, SndioCapture)(device);
if(!backend) return NULL;
return STATIC_CAST(ALCbackend, backend);
}
return NULL;
}
-360
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@@ -1,360 +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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <sys/ioctl.h>
#include <sys/types.h>
#include <sys/time.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 "alconfig.h"
#include "threads.h"
#include "compat.h"
#include "backends/base.h"
#include <sys/audioio.h>
typedef struct ALCsolarisBackend {
DERIVE_FROM_TYPE(ALCbackend);
int fd;
ALubyte *mix_data;
int data_size;
ATOMIC(ALenum) killNow;
althrd_t thread;
} ALCsolarisBackend;
static int ALCsolarisBackend_mixerProc(void *ptr);
static void ALCsolarisBackend_Construct(ALCsolarisBackend *self, ALCdevice *device);
static void ALCsolarisBackend_Destruct(ALCsolarisBackend *self);
static ALCenum ALCsolarisBackend_open(ALCsolarisBackend *self, const ALCchar *name);
static ALCboolean ALCsolarisBackend_reset(ALCsolarisBackend *self);
static ALCboolean ALCsolarisBackend_start(ALCsolarisBackend *self);
static void ALCsolarisBackend_stop(ALCsolarisBackend *self);
static DECLARE_FORWARD2(ALCsolarisBackend, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
static DECLARE_FORWARD(ALCsolarisBackend, ALCbackend, ALCuint, availableSamples)
static DECLARE_FORWARD(ALCsolarisBackend, ALCbackend, ClockLatency, getClockLatency)
static DECLARE_FORWARD(ALCsolarisBackend, ALCbackend, void, lock)
static DECLARE_FORWARD(ALCsolarisBackend, ALCbackend, void, unlock)
DECLARE_DEFAULT_ALLOCATORS(ALCsolarisBackend)
DEFINE_ALCBACKEND_VTABLE(ALCsolarisBackend);
static const ALCchar solaris_device[] = "Solaris Default";
static const char *solaris_driver = "/dev/audio";
static void ALCsolarisBackend_Construct(ALCsolarisBackend *self, ALCdevice *device)
{
ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
SET_VTABLE2(ALCsolarisBackend, ALCbackend, self);
self->fd = -1;
self->mix_data = NULL;
ATOMIC_INIT(&self->killNow, AL_FALSE);
}
static void ALCsolarisBackend_Destruct(ALCsolarisBackend *self)
{
if(self->fd != -1)
close(self->fd);
self->fd = -1;
free(self->mix_data);
self->mix_data = NULL;
self->data_size = 0;
ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
}
static int ALCsolarisBackend_mixerProc(void *ptr)
{
ALCsolarisBackend *self = ptr;
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
struct timeval timeout;
ALubyte *write_ptr;
ALint frame_size;
ALint to_write;
ssize_t wrote;
fd_set wfds;
int sret;
SetRTPriority();
althrd_setname(althrd_current(), MIXER_THREAD_NAME);
frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
ALCsolarisBackend_lock(self);
while(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) &&
ATOMIC_LOAD(&device->Connected, almemory_order_acquire))
{
FD_ZERO(&wfds);
FD_SET(self->fd, &wfds);
timeout.tv_sec = 1;
timeout.tv_usec = 0;
ALCsolarisBackend_unlock(self);
sret = select(self->fd+1, NULL, &wfds, NULL, &timeout);
ALCsolarisBackend_lock(self);
if(sret < 0)
{
if(errno == EINTR)
continue;
ERR("select failed: %s\n", strerror(errno));
aluHandleDisconnect(device, "Failed to wait for playback buffer: %s", strerror(errno));
break;
}
else if(sret == 0)
{
WARN("select timeout\n");
continue;
}
write_ptr = self->mix_data;
to_write = self->data_size;
aluMixData(device, write_ptr, to_write/frame_size);
while(to_write > 0 && !ATOMIC_LOAD_SEQ(&self->killNow))
{
wrote = write(self->fd, write_ptr, to_write);
if(wrote < 0)
{
if(errno == EAGAIN || errno == EWOULDBLOCK || errno == EINTR)
continue;
ERR("write failed: %s\n", strerror(errno));
aluHandleDisconnect(device, "Failed to write playback samples: %s",
strerror(errno));
break;
}
to_write -= wrote;
write_ptr += wrote;
}
}
ALCsolarisBackend_unlock(self);
return 0;
}
static ALCenum ALCsolarisBackend_open(ALCsolarisBackend *self, const ALCchar *name)
{
ALCdevice *device;
if(!name)
name = solaris_device;
else if(strcmp(name, solaris_device) != 0)
return ALC_INVALID_VALUE;
self->fd = open(solaris_driver, O_WRONLY);
if(self->fd == -1)
{
ERR("Could not open %s: %s\n", solaris_driver, strerror(errno));
return ALC_INVALID_VALUE;
}
device = STATIC_CAST(ALCbackend,self)->mDevice;
alstr_copy_cstr(&device->DeviceName, name);
return ALC_NO_ERROR;
}
static ALCboolean ALCsolarisBackend_reset(ALCsolarisBackend *self)
{
ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice;
audio_info_t info;
ALsizei frameSize;
ALsizei numChannels;
AUDIO_INITINFO(&info);
info.play.sample_rate = device->Frequency;
if(device->FmtChans != DevFmtMono)
device->FmtChans = DevFmtStereo;
numChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
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(self->fd, AUDIO_SETINFO, &info) < 0)
{
ERR("ioctl failed: %s\n", strerror(errno));
return ALC_FALSE;
}
if(ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder) != (ALsizei)info.play.channels)
{
ERR("Failed to set %s, got %u channels instead\n", DevFmtChannelsString(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);
free(self->mix_data);
self->data_size = device->UpdateSize * FrameSizeFromDevFmt(
device->FmtChans, device->FmtType, device->AmbiOrder
);
self->mix_data = calloc(1, self->data_size);
return ALC_TRUE;
}
static ALCboolean ALCsolarisBackend_start(ALCsolarisBackend *self)
{
ATOMIC_STORE_SEQ(&self->killNow, AL_FALSE);
if(althrd_create(&self->thread, ALCsolarisBackend_mixerProc, self) != althrd_success)
return ALC_FALSE;
return ALC_TRUE;
}
static void ALCsolarisBackend_stop(ALCsolarisBackend *self)
{
int res;
if(ATOMIC_EXCHANGE_SEQ(&self->killNow, AL_TRUE))
return;
althrd_join(self->thread, &res);
if(ioctl(self->fd, AUDIO_DRAIN) < 0)
ERR("Error draining device: %s\n", strerror(errno));
}
typedef struct ALCsolarisBackendFactory {
DERIVE_FROM_TYPE(ALCbackendFactory);
} ALCsolarisBackendFactory;
#define ALCSOLARISBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCsolarisBackendFactory, ALCbackendFactory) } }
ALCbackendFactory *ALCsolarisBackendFactory_getFactory(void);
static ALCboolean ALCsolarisBackendFactory_init(ALCsolarisBackendFactory *self);
static DECLARE_FORWARD(ALCsolarisBackendFactory, ALCbackendFactory, void, deinit)
static ALCboolean ALCsolarisBackendFactory_querySupport(ALCsolarisBackendFactory *self, ALCbackend_Type type);
static void ALCsolarisBackendFactory_probe(ALCsolarisBackendFactory *self, enum DevProbe type, al_string *outnames);
static ALCbackend* ALCsolarisBackendFactory_createBackend(ALCsolarisBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
DEFINE_ALCBACKENDFACTORY_VTABLE(ALCsolarisBackendFactory);
ALCbackendFactory *ALCsolarisBackendFactory_getFactory(void)
{
static ALCsolarisBackendFactory factory = ALCSOLARISBACKENDFACTORY_INITIALIZER;
return STATIC_CAST(ALCbackendFactory, &factory);
}
static ALCboolean ALCsolarisBackendFactory_init(ALCsolarisBackendFactory* UNUSED(self))
{
ConfigValueStr(NULL, "solaris", "device", &solaris_driver);
return ALC_TRUE;
}
static ALCboolean ALCsolarisBackendFactory_querySupport(ALCsolarisBackendFactory* UNUSED(self), ALCbackend_Type type)
{
if(type == ALCbackend_Playback)
return ALC_TRUE;
return ALC_FALSE;
}
static void ALCsolarisBackendFactory_probe(ALCsolarisBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames)
{
switch(type)
{
case ALL_DEVICE_PROBE:
{
#ifdef HAVE_STAT
struct stat buf;
if(stat(solaris_driver, &buf) == 0)
#endif
alstr_append_range(outnames, solaris_device, solaris_device+sizeof(solaris_device));
}
break;
case CAPTURE_DEVICE_PROBE:
break;
}
}
ALCbackend* ALCsolarisBackendFactory_createBackend(ALCsolarisBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
{
if(type == ALCbackend_Playback)
{
ALCsolarisBackend *backend;
NEW_OBJ(backend, ALCsolarisBackend)(device);
if(!backend) return NULL;
return STATIC_CAST(ALCbackend, backend);
}
return NULL;
}
File diff suppressed because it is too large Load Diff
-453
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@@ -1,453 +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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdlib.h>
#include <stdio.h>
#include <memory.h>
#include <errno.h>
#include "alMain.h"
#include "alu.h"
#include "alconfig.h"
#include "threads.h"
#include "compat.h"
#include "backends/base.h"
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 ALubyte SUBTYPE_BFORMAT_PCM[] = {
0x01, 0x00, 0x00, 0x00, 0x21, 0x07, 0xd3, 0x11, 0x86, 0x44, 0xc8, 0xc1,
0xca, 0x00, 0x00, 0x00
};
static const ALubyte SUBTYPE_BFORMAT_FLOAT[] = {
0x03, 0x00, 0x00, 0x00, 0x21, 0x07, 0xd3, 0x11, 0x86, 0x44, 0xc8, 0xc1,
0xca, 0x00, 0x00, 0x00
};
static void fwrite16le(ALushort val, FILE *f)
{
ALubyte data[2] = { val&0xff, (val>>8)&0xff };
fwrite(data, 1, 2, f);
}
static void fwrite32le(ALuint val, FILE *f)
{
ALubyte data[4] = { val&0xff, (val>>8)&0xff, (val>>16)&0xff, (val>>24)&0xff };
fwrite(data, 1, 4, f);
}
typedef struct ALCwaveBackend {
DERIVE_FROM_TYPE(ALCbackend);
FILE *mFile;
long mDataStart;
ALvoid *mBuffer;
ALuint mSize;
ATOMIC(ALenum) killNow;
althrd_t thread;
} ALCwaveBackend;
static int ALCwaveBackend_mixerProc(void *ptr);
static void ALCwaveBackend_Construct(ALCwaveBackend *self, ALCdevice *device);
static void ALCwaveBackend_Destruct(ALCwaveBackend *self);
static ALCenum ALCwaveBackend_open(ALCwaveBackend *self, const ALCchar *name);
static ALCboolean ALCwaveBackend_reset(ALCwaveBackend *self);
static ALCboolean ALCwaveBackend_start(ALCwaveBackend *self);
static void ALCwaveBackend_stop(ALCwaveBackend *self);
static DECLARE_FORWARD2(ALCwaveBackend, ALCbackend, ALCenum, captureSamples, void*, ALCuint)
static DECLARE_FORWARD(ALCwaveBackend, ALCbackend, ALCuint, availableSamples)
static DECLARE_FORWARD(ALCwaveBackend, ALCbackend, ClockLatency, getClockLatency)
static DECLARE_FORWARD(ALCwaveBackend, ALCbackend, void, lock)
static DECLARE_FORWARD(ALCwaveBackend, ALCbackend, void, unlock)
DECLARE_DEFAULT_ALLOCATORS(ALCwaveBackend)
DEFINE_ALCBACKEND_VTABLE(ALCwaveBackend);
static void ALCwaveBackend_Construct(ALCwaveBackend *self, ALCdevice *device)
{
ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
SET_VTABLE2(ALCwaveBackend, ALCbackend, self);
self->mFile = NULL;
self->mDataStart = -1;
self->mBuffer = NULL;
self->mSize = 0;
ATOMIC_INIT(&self->killNow, AL_TRUE);
}
static void ALCwaveBackend_Destruct(ALCwaveBackend *self)
{
if(self->mFile)
fclose(self->mFile);
self->mFile = NULL;
ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
}
static int ALCwaveBackend_mixerProc(void *ptr)
{
ALCwaveBackend *self = (ALCwaveBackend*)ptr;
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
struct timespec now, start;
ALint64 avail, done;
ALuint frameSize;
size_t fs;
const long restTime = (long)((ALuint64)device->UpdateSize * 1000000000 /
device->Frequency / 2);
althrd_setname(althrd_current(), MIXER_THREAD_NAME);
frameSize = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
done = 0;
if(altimespec_get(&start, AL_TIME_UTC) != AL_TIME_UTC)
{
ERR("Failed to get starting time\n");
return 1;
}
while(!ATOMIC_LOAD(&self->killNow, almemory_order_acquire) &&
ATOMIC_LOAD(&device->Connected, almemory_order_acquire))
{
if(altimespec_get(&now, AL_TIME_UTC) != AL_TIME_UTC)
{
ERR("Failed to get current time\n");
return 1;
}
avail = (now.tv_sec - start.tv_sec) * device->Frequency;
avail += (ALint64)(now.tv_nsec - start.tv_nsec) * device->Frequency / 1000000000;
if(avail < done)
{
/* Oops, time skipped backwards. Reset the number of samples done
* with one update available since we (likely) just came back from
* sleeping. */
done = avail - device->UpdateSize;
}
if(avail-done < device->UpdateSize)
al_nssleep(restTime);
else while(avail-done >= device->UpdateSize)
{
ALCwaveBackend_lock(self);
aluMixData(device, self->mBuffer, device->UpdateSize);
ALCwaveBackend_unlock(self);
done += device->UpdateSize;
if(!IS_LITTLE_ENDIAN)
{
ALuint bytesize = BytesFromDevFmt(device->FmtType);
ALuint i;
if(bytesize == 2)
{
ALushort *samples = self->mBuffer;
ALuint len = self->mSize / 2;
for(i = 0;i < len;i++)
{
ALushort samp = samples[i];
samples[i] = (samp>>8) | (samp<<8);
}
}
else if(bytesize == 4)
{
ALuint *samples = self->mBuffer;
ALuint len = self->mSize / 4;
for(i = 0;i < len;i++)
{
ALuint samp = samples[i];
samples[i] = (samp>>24) | ((samp>>8)&0x0000ff00) |
((samp<<8)&0x00ff0000) | (samp<<24);
}
}
}
fs = fwrite(self->mBuffer, frameSize, device->UpdateSize, self->mFile);
(void)fs;
if(ferror(self->mFile))
{
ERR("Error writing to file\n");
ALCdevice_Lock(device);
aluHandleDisconnect(device, "Failed to write playback samples");
ALCdevice_Unlock(device);
break;
}
}
}
return 0;
}
static ALCenum ALCwaveBackend_open(ALCwaveBackend *self, const ALCchar *name)
{
ALCdevice *device;
const char *fname;
fname = GetConfigValue(NULL, "wave", "file", "");
if(!fname[0]) return ALC_INVALID_VALUE;
if(!name)
name = waveDevice;
else if(strcmp(name, waveDevice) != 0)
return ALC_INVALID_VALUE;
self->mFile = al_fopen(fname, "wb");
if(!self->mFile)
{
ERR("Could not open file '%s': %s\n", fname, strerror(errno));
return ALC_INVALID_VALUE;
}
device = STATIC_CAST(ALCbackend, self)->mDevice;
alstr_copy_cstr(&device->DeviceName, name);
return ALC_NO_ERROR;
}
static ALCboolean ALCwaveBackend_reset(ALCwaveBackend *self)
{
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
ALuint channels=0, bits=0, chanmask=0;
int isbformat = 0;
size_t val;
fseek(self->mFile, 0, SEEK_SET);
clearerr(self->mFile);
if(GetConfigValueBool(NULL, "wave", "bformat", 0))
{
device->FmtChans = DevFmtAmbi3D;
device->AmbiOrder = 1;
}
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;
}
switch(device->FmtChans)
{
case DevFmtMono: chanmask = 0x04; break;
case DevFmtStereo: chanmask = 0x01 | 0x02; break;
case DevFmtQuad: chanmask = 0x01 | 0x02 | 0x10 | 0x20; break;
case DevFmtX51: chanmask = 0x01 | 0x02 | 0x04 | 0x08 | 0x200 | 0x400; break;
case DevFmtX51Rear: chanmask = 0x01 | 0x02 | 0x04 | 0x08 | 0x010 | 0x020; break;
case DevFmtX61: chanmask = 0x01 | 0x02 | 0x04 | 0x08 | 0x100 | 0x200 | 0x400; break;
case DevFmtX71: chanmask = 0x01 | 0x02 | 0x04 | 0x08 | 0x010 | 0x020 | 0x200 | 0x400; break;
case DevFmtAmbi3D:
/* .amb output requires FuMa */
device->AmbiLayout = AmbiLayout_FuMa;
device->AmbiScale = AmbiNorm_FuMa;
isbformat = 1;
chanmask = 0;
break;
}
bits = BytesFromDevFmt(device->FmtType) * 8;
channels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
fputs("RIFF", self->mFile);
fwrite32le(0xFFFFFFFF, self->mFile); // 'RIFF' header len; filled in at close
fputs("WAVE", self->mFile);
fputs("fmt ", self->mFile);
fwrite32le(40, self->mFile); // 'fmt ' header len; 40 bytes for EXTENSIBLE
// 16-bit val, format type id (extensible: 0xFFFE)
fwrite16le(0xFFFE, self->mFile);
// 16-bit val, channel count
fwrite16le(channels, self->mFile);
// 32-bit val, frequency
fwrite32le(device->Frequency, self->mFile);
// 32-bit val, bytes per second
fwrite32le(device->Frequency * channels * bits / 8, self->mFile);
// 16-bit val, frame size
fwrite16le(channels * bits / 8, self->mFile);
// 16-bit val, bits per sample
fwrite16le(bits, self->mFile);
// 16-bit val, extra byte count
fwrite16le(22, self->mFile);
// 16-bit val, valid bits per sample
fwrite16le(bits, self->mFile);
// 32-bit val, channel mask
fwrite32le(chanmask, self->mFile);
// 16 byte GUID, sub-type format
val = fwrite((device->FmtType == DevFmtFloat) ?
(isbformat ? SUBTYPE_BFORMAT_FLOAT : SUBTYPE_FLOAT) :
(isbformat ? SUBTYPE_BFORMAT_PCM : SUBTYPE_PCM), 1, 16, self->mFile);
(void)val;
fputs("data", self->mFile);
fwrite32le(0xFFFFFFFF, self->mFile); // 'data' header len; filled in at close
if(ferror(self->mFile))
{
ERR("Error writing header: %s\n", strerror(errno));
return ALC_FALSE;
}
self->mDataStart = ftell(self->mFile);
SetDefaultWFXChannelOrder(device);
return ALC_TRUE;
}
static ALCboolean ALCwaveBackend_start(ALCwaveBackend *self)
{
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
self->mSize = device->UpdateSize * FrameSizeFromDevFmt(
device->FmtChans, device->FmtType, device->AmbiOrder
);
self->mBuffer = malloc(self->mSize);
if(!self->mBuffer)
{
ERR("Buffer malloc failed\n");
return ALC_FALSE;
}
ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release);
if(althrd_create(&self->thread, ALCwaveBackend_mixerProc, self) != althrd_success)
{
free(self->mBuffer);
self->mBuffer = NULL;
self->mSize = 0;
return ALC_FALSE;
}
return ALC_TRUE;
}
static void ALCwaveBackend_stop(ALCwaveBackend *self)
{
ALuint dataLen;
long size;
int res;
if(ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel))
return;
althrd_join(self->thread, &res);
free(self->mBuffer);
self->mBuffer = NULL;
size = ftell(self->mFile);
if(size > 0)
{
dataLen = size - self->mDataStart;
if(fseek(self->mFile, self->mDataStart-4, SEEK_SET) == 0)
fwrite32le(dataLen, self->mFile); // 'data' header len
if(fseek(self->mFile, 4, SEEK_SET) == 0)
fwrite32le(size-8, self->mFile); // 'WAVE' header len
}
}
typedef struct ALCwaveBackendFactory {
DERIVE_FROM_TYPE(ALCbackendFactory);
} ALCwaveBackendFactory;
#define ALCWAVEBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCwaveBackendFactory, ALCbackendFactory) } }
ALCbackendFactory *ALCwaveBackendFactory_getFactory(void);
static ALCboolean ALCwaveBackendFactory_init(ALCwaveBackendFactory *self);
static DECLARE_FORWARD(ALCwaveBackendFactory, ALCbackendFactory, void, deinit)
static ALCboolean ALCwaveBackendFactory_querySupport(ALCwaveBackendFactory *self, ALCbackend_Type type);
static void ALCwaveBackendFactory_probe(ALCwaveBackendFactory *self, enum DevProbe type, al_string *outnames);
static ALCbackend* ALCwaveBackendFactory_createBackend(ALCwaveBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
DEFINE_ALCBACKENDFACTORY_VTABLE(ALCwaveBackendFactory);
ALCbackendFactory *ALCwaveBackendFactory_getFactory(void)
{
static ALCwaveBackendFactory factory = ALCWAVEBACKENDFACTORY_INITIALIZER;
return STATIC_CAST(ALCbackendFactory, &factory);
}
static ALCboolean ALCwaveBackendFactory_init(ALCwaveBackendFactory* UNUSED(self))
{
return ALC_TRUE;
}
static ALCboolean ALCwaveBackendFactory_querySupport(ALCwaveBackendFactory* UNUSED(self), ALCbackend_Type type)
{
if(type == ALCbackend_Playback)
return !!ConfigValueExists(NULL, "wave", "file");
return ALC_FALSE;
}
static void ALCwaveBackendFactory_probe(ALCwaveBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames)
{
switch(type)
{
case ALL_DEVICE_PROBE:
alstr_append_range(outnames, waveDevice, waveDevice+sizeof(waveDevice));
break;
case CAPTURE_DEVICE_PROBE:
break;
}
}
static ALCbackend* ALCwaveBackendFactory_createBackend(ALCwaveBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
{
if(type == ALCbackend_Playback)
{
ALCwaveBackend *backend;
NEW_OBJ(backend, ALCwaveBackend)(device);
if(!backend) return NULL;
return STATIC_CAST(ALCbackend, backend);
}
return NULL;
}
-786
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@@ -1,786 +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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 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"
#include "ringbuffer.h"
#include "threads.h"
#include "backends/base.h"
#ifndef WAVE_FORMAT_IEEE_FLOAT
#define WAVE_FORMAT_IEEE_FLOAT 0x0003
#endif
#define DEVNAME_HEAD "OpenAL Soft on "
static vector_al_string PlaybackDevices;
static vector_al_string CaptureDevices;
static void clear_devlist(vector_al_string *list)
{
VECTOR_FOR_EACH(al_string, *list, alstr_reset);
VECTOR_RESIZE(*list, 0, 0);
}
static void ProbePlaybackDevices(void)
{
ALuint numdevs;
ALuint i;
clear_devlist(&PlaybackDevices);
numdevs = waveOutGetNumDevs();
VECTOR_RESIZE(PlaybackDevices, 0, numdevs);
for(i = 0;i < numdevs;i++)
{
WAVEOUTCAPSW WaveCaps;
const al_string *iter;
al_string dname;
AL_STRING_INIT(dname);
if(waveOutGetDevCapsW(i, &WaveCaps, sizeof(WaveCaps)) == MMSYSERR_NOERROR)
{
ALuint count = 0;
while(1)
{
alstr_copy_cstr(&dname, DEVNAME_HEAD);
alstr_append_wcstr(&dname, WaveCaps.szPname);
if(count != 0)
{
char str[64];
snprintf(str, sizeof(str), " #%d", count+1);
alstr_append_cstr(&dname, str);
}
count++;
#define MATCH_ENTRY(i) (alstr_cmp(dname, *(i)) == 0)
VECTOR_FIND_IF(iter, const al_string, PlaybackDevices, MATCH_ENTRY);
if(iter == VECTOR_END(PlaybackDevices)) break;
#undef MATCH_ENTRY
}
TRACE("Got device \"%s\", ID %u\n", alstr_get_cstr(dname), i);
}
VECTOR_PUSH_BACK(PlaybackDevices, dname);
}
}
static void ProbeCaptureDevices(void)
{
ALuint numdevs;
ALuint i;
clear_devlist(&CaptureDevices);
numdevs = waveInGetNumDevs();
VECTOR_RESIZE(CaptureDevices, 0, numdevs);
for(i = 0;i < numdevs;i++)
{
WAVEINCAPSW WaveCaps;
const al_string *iter;
al_string dname;
AL_STRING_INIT(dname);
if(waveInGetDevCapsW(i, &WaveCaps, sizeof(WaveCaps)) == MMSYSERR_NOERROR)
{
ALuint count = 0;
while(1)
{
alstr_copy_cstr(&dname, DEVNAME_HEAD);
alstr_append_wcstr(&dname, WaveCaps.szPname);
if(count != 0)
{
char str[64];
snprintf(str, sizeof(str), " #%d", count+1);
alstr_append_cstr(&dname, str);
}
count++;
#define MATCH_ENTRY(i) (alstr_cmp(dname, *(i)) == 0)
VECTOR_FIND_IF(iter, const al_string, CaptureDevices, MATCH_ENTRY);
if(iter == VECTOR_END(CaptureDevices)) break;
#undef MATCH_ENTRY
}
TRACE("Got device \"%s\", ID %u\n", alstr_get_cstr(dname), i);
}
VECTOR_PUSH_BACK(CaptureDevices, dname);
}
}
typedef struct ALCwinmmPlayback {
DERIVE_FROM_TYPE(ALCbackend);
RefCount WaveBuffersCommitted;
WAVEHDR WaveBuffer[4];
HWAVEOUT OutHdl;
WAVEFORMATEX Format;
ATOMIC(ALenum) killNow;
althrd_t thread;
} ALCwinmmPlayback;
static void ALCwinmmPlayback_Construct(ALCwinmmPlayback *self, ALCdevice *device);
static void ALCwinmmPlayback_Destruct(ALCwinmmPlayback *self);
static void CALLBACK ALCwinmmPlayback_waveOutProc(HWAVEOUT device, UINT msg, DWORD_PTR instance, DWORD_PTR param1, DWORD_PTR param2);
static int ALCwinmmPlayback_mixerProc(void *arg);
static ALCenum ALCwinmmPlayback_open(ALCwinmmPlayback *self, const ALCchar *name);
static ALCboolean ALCwinmmPlayback_reset(ALCwinmmPlayback *self);
static ALCboolean ALCwinmmPlayback_start(ALCwinmmPlayback *self);
static void ALCwinmmPlayback_stop(ALCwinmmPlayback *self);
static DECLARE_FORWARD2(ALCwinmmPlayback, ALCbackend, ALCenum, captureSamples, ALCvoid*, ALCuint)
static DECLARE_FORWARD(ALCwinmmPlayback, ALCbackend, ALCuint, availableSamples)
static DECLARE_FORWARD(ALCwinmmPlayback, ALCbackend, ClockLatency, getClockLatency)
static DECLARE_FORWARD(ALCwinmmPlayback, ALCbackend, void, lock)
static DECLARE_FORWARD(ALCwinmmPlayback, ALCbackend, void, unlock)
DECLARE_DEFAULT_ALLOCATORS(ALCwinmmPlayback)
DEFINE_ALCBACKEND_VTABLE(ALCwinmmPlayback);
static void ALCwinmmPlayback_Construct(ALCwinmmPlayback *self, ALCdevice *device)
{
ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
SET_VTABLE2(ALCwinmmPlayback, ALCbackend, self);
InitRef(&self->WaveBuffersCommitted, 0);
self->OutHdl = NULL;
ATOMIC_INIT(&self->killNow, AL_TRUE);
}
static void ALCwinmmPlayback_Destruct(ALCwinmmPlayback *self)
{
if(self->OutHdl)
waveOutClose(self->OutHdl);
self->OutHdl = 0;
ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
}
/* ALCwinmmPlayback_waveOutProc
*
* Posts a message to 'ALCwinmmPlayback_mixerProc' everytime a WaveOut Buffer
* is completed and returns to the application (for more data)
*/
static void CALLBACK ALCwinmmPlayback_waveOutProc(HWAVEOUT UNUSED(device), UINT msg, DWORD_PTR instance, DWORD_PTR param1, DWORD_PTR UNUSED(param2))
{
ALCwinmmPlayback *self = (ALCwinmmPlayback*)instance;
if(msg != WOM_DONE)
return;
DecrementRef(&self->WaveBuffersCommitted);
PostThreadMessage(self->thread, msg, 0, param1);
}
FORCE_ALIGN static int ALCwinmmPlayback_mixerProc(void *arg)
{
ALCwinmmPlayback *self = arg;
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
WAVEHDR *WaveHdr;
MSG msg;
SetRTPriority();
althrd_setname(althrd_current(), MIXER_THREAD_NAME);
while(GetMessage(&msg, NULL, 0, 0))
{
if(msg.message != WOM_DONE)
continue;
if(ATOMIC_LOAD(&self->killNow, almemory_order_acquire))
{
if(ReadRef(&self->WaveBuffersCommitted) == 0)
break;
continue;
}
WaveHdr = ((WAVEHDR*)msg.lParam);
ALCwinmmPlayback_lock(self);
aluMixData(device, WaveHdr->lpData, WaveHdr->dwBufferLength /
self->Format.nBlockAlign);
ALCwinmmPlayback_unlock(self);
// Send buffer back to play more data
waveOutWrite(self->OutHdl, WaveHdr, sizeof(WAVEHDR));
IncrementRef(&self->WaveBuffersCommitted);
}
return 0;
}
static ALCenum ALCwinmmPlayback_open(ALCwinmmPlayback *self, const ALCchar *deviceName)
{
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
const al_string *iter;
UINT DeviceID;
MMRESULT res;
if(VECTOR_SIZE(PlaybackDevices) == 0)
ProbePlaybackDevices();
// Find the Device ID matching the deviceName if valid
#define MATCH_DEVNAME(iter) (!alstr_empty(*(iter)) && \
(!deviceName || alstr_cmp_cstr(*(iter), deviceName) == 0))
VECTOR_FIND_IF(iter, const al_string, PlaybackDevices, MATCH_DEVNAME);
if(iter == VECTOR_END(PlaybackDevices))
return ALC_INVALID_VALUE;
#undef MATCH_DEVNAME
DeviceID = (UINT)(iter - VECTOR_BEGIN(PlaybackDevices));
retry_open:
memset(&self->Format, 0, sizeof(WAVEFORMATEX));
if(device->FmtType == DevFmtFloat)
{
self->Format.wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
self->Format.wBitsPerSample = 32;
}
else
{
self->Format.wFormatTag = WAVE_FORMAT_PCM;
if(device->FmtType == DevFmtUByte || device->FmtType == DevFmtByte)
self->Format.wBitsPerSample = 8;
else
self->Format.wBitsPerSample = 16;
}
self->Format.nChannels = ((device->FmtChans == DevFmtMono) ? 1 : 2);
self->Format.nBlockAlign = self->Format.wBitsPerSample *
self->Format.nChannels / 8;
self->Format.nSamplesPerSec = device->Frequency;
self->Format.nAvgBytesPerSec = self->Format.nSamplesPerSec *
self->Format.nBlockAlign;
self->Format.cbSize = 0;
if((res=waveOutOpen(&self->OutHdl, DeviceID, &self->Format, (DWORD_PTR)&ALCwinmmPlayback_waveOutProc, (DWORD_PTR)self, CALLBACK_FUNCTION)) != MMSYSERR_NOERROR)
{
if(device->FmtType == DevFmtFloat)
{
device->FmtType = DevFmtShort;
goto retry_open;
}
ERR("waveOutOpen failed: %u\n", res);
goto failure;
}
alstr_copy(&device->DeviceName, VECTOR_ELEM(PlaybackDevices, DeviceID));
return ALC_NO_ERROR;
failure:
if(self->OutHdl)
waveOutClose(self->OutHdl);
self->OutHdl = NULL;
return ALC_INVALID_VALUE;
}
static ALCboolean ALCwinmmPlayback_reset(ALCwinmmPlayback *self)
{
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
device->UpdateSize = (ALuint)((ALuint64)device->UpdateSize *
self->Format.nSamplesPerSec /
device->Frequency);
device->UpdateSize = (device->UpdateSize*device->NumUpdates + 3) / 4;
device->NumUpdates = 4;
device->Frequency = self->Format.nSamplesPerSec;
if(self->Format.wFormatTag == WAVE_FORMAT_IEEE_FLOAT)
{
if(self->Format.wBitsPerSample == 32)
device->FmtType = DevFmtFloat;
else
{
ERR("Unhandled IEEE float sample depth: %d\n", self->Format.wBitsPerSample);
return ALC_FALSE;
}
}
else if(self->Format.wFormatTag == WAVE_FORMAT_PCM)
{
if(self->Format.wBitsPerSample == 16)
device->FmtType = DevFmtShort;
else if(self->Format.wBitsPerSample == 8)
device->FmtType = DevFmtUByte;
else
{
ERR("Unhandled PCM sample depth: %d\n", self->Format.wBitsPerSample);
return ALC_FALSE;
}
}
else
{
ERR("Unhandled format tag: 0x%04x\n", self->Format.wFormatTag);
return ALC_FALSE;
}
if(self->Format.nChannels == 2)
device->FmtChans = DevFmtStereo;
else if(self->Format.nChannels == 1)
device->FmtChans = DevFmtMono;
else
{
ERR("Unhandled channel count: %d\n", self->Format.nChannels);
return ALC_FALSE;
}
SetDefaultWFXChannelOrder(device);
return ALC_TRUE;
}
static ALCboolean ALCwinmmPlayback_start(ALCwinmmPlayback *self)
{
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
ALbyte *BufferData;
ALint BufferSize;
ALuint i;
ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release);
if(althrd_create(&self->thread, ALCwinmmPlayback_mixerProc, self) != althrd_success)
return ALC_FALSE;
InitRef(&self->WaveBuffersCommitted, 0);
// Create 4 Buffers
BufferSize = device->UpdateSize*device->NumUpdates / 4;
BufferSize *= FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder);
BufferData = calloc(4, BufferSize);
for(i = 0;i < 4;i++)
{
memset(&self->WaveBuffer[i], 0, sizeof(WAVEHDR));
self->WaveBuffer[i].dwBufferLength = BufferSize;
self->WaveBuffer[i].lpData = ((i==0) ? (CHAR*)BufferData :
(self->WaveBuffer[i-1].lpData +
self->WaveBuffer[i-1].dwBufferLength));
waveOutPrepareHeader(self->OutHdl, &self->WaveBuffer[i], sizeof(WAVEHDR));
waveOutWrite(self->OutHdl, &self->WaveBuffer[i], sizeof(WAVEHDR));
IncrementRef(&self->WaveBuffersCommitted);
}
return ALC_TRUE;
}
static void ALCwinmmPlayback_stop(ALCwinmmPlayback *self)
{
void *buffer = NULL;
int i;
if(ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel))
return;
althrd_join(self->thread, &i);
// Release the wave buffers
for(i = 0;i < 4;i++)
{
waveOutUnprepareHeader(self->OutHdl, &self->WaveBuffer[i], sizeof(WAVEHDR));
if(i == 0) buffer = self->WaveBuffer[i].lpData;
self->WaveBuffer[i].lpData = NULL;
}
free(buffer);
}
typedef struct ALCwinmmCapture {
DERIVE_FROM_TYPE(ALCbackend);
RefCount WaveBuffersCommitted;
WAVEHDR WaveBuffer[4];
HWAVEIN InHdl;
ll_ringbuffer_t *Ring;
WAVEFORMATEX Format;
ATOMIC(ALenum) killNow;
althrd_t thread;
} ALCwinmmCapture;
static void ALCwinmmCapture_Construct(ALCwinmmCapture *self, ALCdevice *device);
static void ALCwinmmCapture_Destruct(ALCwinmmCapture *self);
static void CALLBACK ALCwinmmCapture_waveInProc(HWAVEIN device, UINT msg, DWORD_PTR instance, DWORD_PTR param1, DWORD_PTR param2);
static int ALCwinmmCapture_captureProc(void *arg);
static ALCenum ALCwinmmCapture_open(ALCwinmmCapture *self, const ALCchar *name);
static DECLARE_FORWARD(ALCwinmmCapture, ALCbackend, ALCboolean, reset)
static ALCboolean ALCwinmmCapture_start(ALCwinmmCapture *self);
static void ALCwinmmCapture_stop(ALCwinmmCapture *self);
static ALCenum ALCwinmmCapture_captureSamples(ALCwinmmCapture *self, ALCvoid *buffer, ALCuint samples);
static ALCuint ALCwinmmCapture_availableSamples(ALCwinmmCapture *self);
static DECLARE_FORWARD(ALCwinmmCapture, ALCbackend, ClockLatency, getClockLatency)
static DECLARE_FORWARD(ALCwinmmCapture, ALCbackend, void, lock)
static DECLARE_FORWARD(ALCwinmmCapture, ALCbackend, void, unlock)
DECLARE_DEFAULT_ALLOCATORS(ALCwinmmCapture)
DEFINE_ALCBACKEND_VTABLE(ALCwinmmCapture);
static void ALCwinmmCapture_Construct(ALCwinmmCapture *self, ALCdevice *device)
{
ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device);
SET_VTABLE2(ALCwinmmCapture, ALCbackend, self);
InitRef(&self->WaveBuffersCommitted, 0);
self->InHdl = NULL;
ATOMIC_INIT(&self->killNow, AL_TRUE);
}
static void ALCwinmmCapture_Destruct(ALCwinmmCapture *self)
{
void *buffer = NULL;
int i;
/* Tell the processing thread to quit and wait for it to do so. */
if(!ATOMIC_EXCHANGE(&self->killNow, AL_TRUE, almemory_order_acq_rel))
{
PostThreadMessage(self->thread, WM_QUIT, 0, 0);
althrd_join(self->thread, &i);
/* Make sure capture is stopped and all pending buffers are flushed. */
waveInReset(self->InHdl);
// Release the wave buffers
for(i = 0;i < 4;i++)
{
waveInUnprepareHeader(self->InHdl, &self->WaveBuffer[i], sizeof(WAVEHDR));
if(i == 0) buffer = self->WaveBuffer[i].lpData;
self->WaveBuffer[i].lpData = NULL;
}
free(buffer);
}
ll_ringbuffer_free(self->Ring);
self->Ring = NULL;
// Close the Wave device
if(self->InHdl)
waveInClose(self->InHdl);
self->InHdl = 0;
ALCbackend_Destruct(STATIC_CAST(ALCbackend, self));
}
/* ALCwinmmCapture_waveInProc
*
* Posts a message to 'ALCwinmmCapture_captureProc' everytime a WaveIn Buffer
* is completed and returns to the application (with more data).
*/
static void CALLBACK ALCwinmmCapture_waveInProc(HWAVEIN UNUSED(device), UINT msg, DWORD_PTR instance, DWORD_PTR param1, DWORD_PTR UNUSED(param2))
{
ALCwinmmCapture *self = (ALCwinmmCapture*)instance;
if(msg != WIM_DATA)
return;
DecrementRef(&self->WaveBuffersCommitted);
PostThreadMessage(self->thread, msg, 0, param1);
}
static int ALCwinmmCapture_captureProc(void *arg)
{
ALCwinmmCapture *self = arg;
WAVEHDR *WaveHdr;
MSG msg;
althrd_setname(althrd_current(), RECORD_THREAD_NAME);
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(ATOMIC_LOAD(&self->killNow, almemory_order_acquire))
break;
WaveHdr = ((WAVEHDR*)msg.lParam);
ll_ringbuffer_write(self->Ring, WaveHdr->lpData,
WaveHdr->dwBytesRecorded / self->Format.nBlockAlign
);
// Send buffer back to capture more data
waveInAddBuffer(self->InHdl, WaveHdr, sizeof(WAVEHDR));
IncrementRef(&self->WaveBuffersCommitted);
}
return 0;
}
static ALCenum ALCwinmmCapture_open(ALCwinmmCapture *self, const ALCchar *name)
{
ALCdevice *device = STATIC_CAST(ALCbackend, self)->mDevice;
const al_string *iter;
ALbyte *BufferData = NULL;
DWORD CapturedDataSize;
ALint BufferSize;
UINT DeviceID;
MMRESULT res;
ALuint i;
if(VECTOR_SIZE(CaptureDevices) == 0)
ProbeCaptureDevices();
// Find the Device ID matching the deviceName if valid
#define MATCH_DEVNAME(iter) (!alstr_empty(*(iter)) && (!name || alstr_cmp_cstr(*iter, name) == 0))
VECTOR_FIND_IF(iter, const al_string, CaptureDevices, MATCH_DEVNAME);
if(iter == VECTOR_END(CaptureDevices))
return ALC_INVALID_VALUE;
#undef MATCH_DEVNAME
DeviceID = (UINT)(iter - VECTOR_BEGIN(CaptureDevices));
switch(device->FmtChans)
{
case DevFmtMono:
case DevFmtStereo:
break;
case DevFmtQuad:
case DevFmtX51:
case DevFmtX51Rear:
case DevFmtX61:
case DevFmtX71:
case DevFmtAmbi3D:
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;
}
memset(&self->Format, 0, sizeof(WAVEFORMATEX));
self->Format.wFormatTag = ((device->FmtType == DevFmtFloat) ?
WAVE_FORMAT_IEEE_FLOAT : WAVE_FORMAT_PCM);
self->Format.nChannels = ChannelsFromDevFmt(device->FmtChans, device->AmbiOrder);
self->Format.wBitsPerSample = BytesFromDevFmt(device->FmtType) * 8;
self->Format.nBlockAlign = self->Format.wBitsPerSample *
self->Format.nChannels / 8;
self->Format.nSamplesPerSec = device->Frequency;
self->Format.nAvgBytesPerSec = self->Format.nSamplesPerSec *
self->Format.nBlockAlign;
self->Format.cbSize = 0;
if((res=waveInOpen(&self->InHdl, DeviceID, &self->Format, (DWORD_PTR)&ALCwinmmCapture_waveInProc, (DWORD_PTR)self, CALLBACK_FUNCTION)) != MMSYSERR_NOERROR)
{
ERR("waveInOpen failed: %u\n", res);
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 < (self->Format.nSamplesPerSec / 10))
CapturedDataSize = self->Format.nSamplesPerSec / 10;
self->Ring = ll_ringbuffer_create(CapturedDataSize, self->Format.nBlockAlign, false);
if(!self->Ring) goto failure;
InitRef(&self->WaveBuffersCommitted, 0);
// Create 4 Buffers of 50ms each
BufferSize = self->Format.nAvgBytesPerSec / 20;
BufferSize -= (BufferSize % self->Format.nBlockAlign);
BufferData = calloc(4, BufferSize);
if(!BufferData) goto failure;
for(i = 0;i < 4;i++)
{
memset(&self->WaveBuffer[i], 0, sizeof(WAVEHDR));
self->WaveBuffer[i].dwBufferLength = BufferSize;
self->WaveBuffer[i].lpData = ((i==0) ? (CHAR*)BufferData :
(self->WaveBuffer[i-1].lpData +
self->WaveBuffer[i-1].dwBufferLength));
self->WaveBuffer[i].dwFlags = 0;
self->WaveBuffer[i].dwLoops = 0;
waveInPrepareHeader(self->InHdl, &self->WaveBuffer[i], sizeof(WAVEHDR));
waveInAddBuffer(self->InHdl, &self->WaveBuffer[i], sizeof(WAVEHDR));
IncrementRef(&self->WaveBuffersCommitted);
}
ATOMIC_STORE(&self->killNow, AL_FALSE, almemory_order_release);
if(althrd_create(&self->thread, ALCwinmmCapture_captureProc, self) != althrd_success)
goto failure;
alstr_copy(&device->DeviceName, VECTOR_ELEM(CaptureDevices, DeviceID));
return ALC_NO_ERROR;
failure:
if(BufferData)
{
for(i = 0;i < 4;i++)
waveInUnprepareHeader(self->InHdl, &self->WaveBuffer[i], sizeof(WAVEHDR));
free(BufferData);
}
ll_ringbuffer_free(self->Ring);
self->Ring = NULL;
if(self->InHdl)
waveInClose(self->InHdl);
self->InHdl = NULL;
return ALC_INVALID_VALUE;
}
static ALCboolean ALCwinmmCapture_start(ALCwinmmCapture *self)
{
waveInStart(self->InHdl);
return ALC_TRUE;
}
static void ALCwinmmCapture_stop(ALCwinmmCapture *self)
{
waveInStop(self->InHdl);
}
static ALCenum ALCwinmmCapture_captureSamples(ALCwinmmCapture *self, ALCvoid *buffer, ALCuint samples)
{
ll_ringbuffer_read(self->Ring, buffer, samples);
return ALC_NO_ERROR;
}
static ALCuint ALCwinmmCapture_availableSamples(ALCwinmmCapture *self)
{
return (ALCuint)ll_ringbuffer_read_space(self->Ring);
}
typedef struct ALCwinmmBackendFactory {
DERIVE_FROM_TYPE(ALCbackendFactory);
} ALCwinmmBackendFactory;
#define ALCWINMMBACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCwinmmBackendFactory, ALCbackendFactory) } }
static ALCboolean ALCwinmmBackendFactory_init(ALCwinmmBackendFactory *self);
static void ALCwinmmBackendFactory_deinit(ALCwinmmBackendFactory *self);
static ALCboolean ALCwinmmBackendFactory_querySupport(ALCwinmmBackendFactory *self, ALCbackend_Type type);
static void ALCwinmmBackendFactory_probe(ALCwinmmBackendFactory *self, enum DevProbe type, al_string *outnames);
static ALCbackend* ALCwinmmBackendFactory_createBackend(ALCwinmmBackendFactory *self, ALCdevice *device, ALCbackend_Type type);
DEFINE_ALCBACKENDFACTORY_VTABLE(ALCwinmmBackendFactory);
static ALCboolean ALCwinmmBackendFactory_init(ALCwinmmBackendFactory* UNUSED(self))
{
VECTOR_INIT(PlaybackDevices);
VECTOR_INIT(CaptureDevices);
return ALC_TRUE;
}
static void ALCwinmmBackendFactory_deinit(ALCwinmmBackendFactory* UNUSED(self))
{
clear_devlist(&PlaybackDevices);
VECTOR_DEINIT(PlaybackDevices);
clear_devlist(&CaptureDevices);
VECTOR_DEINIT(CaptureDevices);
}
static ALCboolean ALCwinmmBackendFactory_querySupport(ALCwinmmBackendFactory* UNUSED(self), ALCbackend_Type type)
{
if(type == ALCbackend_Playback || type == ALCbackend_Capture)
return ALC_TRUE;
return ALC_FALSE;
}
static void ALCwinmmBackendFactory_probe(ALCwinmmBackendFactory* UNUSED(self), enum DevProbe type, al_string *outnames)
{
switch(type)
{
#define APPEND_OUTNAME(n) do { \
if(!alstr_empty(*(n))) \
alstr_append_range(outnames, VECTOR_BEGIN(*(n)), VECTOR_END(*(n))+1); \
} while(0)
case ALL_DEVICE_PROBE:
ProbePlaybackDevices();
VECTOR_FOR_EACH(const al_string, PlaybackDevices, APPEND_OUTNAME);
break;
case CAPTURE_DEVICE_PROBE:
ProbeCaptureDevices();
VECTOR_FOR_EACH(const al_string, CaptureDevices, APPEND_OUTNAME);
break;
#undef APPEND_OUTNAME
}
}
static ALCbackend* ALCwinmmBackendFactory_createBackend(ALCwinmmBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type)
{
if(type == ALCbackend_Playback)
{
ALCwinmmPlayback *backend;
NEW_OBJ(backend, ALCwinmmPlayback)(device);
if(!backend) return NULL;
return STATIC_CAST(ALCbackend, backend);
}
if(type == ALCbackend_Capture)
{
ALCwinmmCapture *backend;
NEW_OBJ(backend, ALCwinmmCapture)(device);
if(!backend) return NULL;
return STATIC_CAST(ALCbackend, backend);
}
return NULL;
}
ALCbackendFactory *ALCwinmmBackendFactory_getFactory(void)
{
static ALCwinmmBackendFactory factory = ALCWINMMBACKENDFACTORY_INITIALIZER;
return STATIC_CAST(ALCbackendFactory, &factory);
}
-492
View File
@@ -1,492 +0,0 @@
#include "config.h"
#include "bformatdec.h"
#include "ambdec.h"
#include "filters/splitter.h"
#include "alu.h"
#include "bool.h"
#include "threads.h"
#include "almalloc.h"
/* NOTE: These are scale factors as applied to Ambisonics content. Decoder
* coefficients should be divided by these values to get proper N3D scalings.
*/
const ALfloat N3D2N3DScale[MAX_AMBI_COEFFS] = {
1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f,
1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f
};
const ALfloat SN3D2N3DScale[MAX_AMBI_COEFFS] = {
1.000000000f, /* ACN 0 (W), sqrt(1) */
1.732050808f, /* ACN 1 (Y), sqrt(3) */
1.732050808f, /* ACN 2 (Z), sqrt(3) */
1.732050808f, /* ACN 3 (X), sqrt(3) */
2.236067978f, /* ACN 4 (V), sqrt(5) */
2.236067978f, /* ACN 5 (T), sqrt(5) */
2.236067978f, /* ACN 6 (R), sqrt(5) */
2.236067978f, /* ACN 7 (S), sqrt(5) */
2.236067978f, /* ACN 8 (U), sqrt(5) */
2.645751311f, /* ACN 9 (Q), sqrt(7) */
2.645751311f, /* ACN 10 (O), sqrt(7) */
2.645751311f, /* ACN 11 (M), sqrt(7) */
2.645751311f, /* ACN 12 (K), sqrt(7) */
2.645751311f, /* ACN 13 (L), sqrt(7) */
2.645751311f, /* ACN 14 (N), sqrt(7) */
2.645751311f, /* ACN 15 (P), sqrt(7) */
};
const ALfloat FuMa2N3DScale[MAX_AMBI_COEFFS] = {
1.414213562f, /* ACN 0 (W), sqrt(2) */
1.732050808f, /* ACN 1 (Y), sqrt(3) */
1.732050808f, /* ACN 2 (Z), sqrt(3) */
1.732050808f, /* ACN 3 (X), sqrt(3) */
1.936491673f, /* ACN 4 (V), sqrt(15)/2 */
1.936491673f, /* ACN 5 (T), sqrt(15)/2 */
2.236067978f, /* ACN 6 (R), sqrt(5) */
1.936491673f, /* ACN 7 (S), sqrt(15)/2 */
1.936491673f, /* ACN 8 (U), sqrt(15)/2 */
2.091650066f, /* ACN 9 (Q), sqrt(35/8) */
1.972026594f, /* ACN 10 (O), sqrt(35)/3 */
2.231093404f, /* ACN 11 (M), sqrt(224/45) */
2.645751311f, /* ACN 12 (K), sqrt(7) */
2.231093404f, /* ACN 13 (L), sqrt(224/45) */
1.972026594f, /* ACN 14 (N), sqrt(35)/3 */
2.091650066f, /* ACN 15 (P), sqrt(35/8) */
};
#define HF_BAND 0
#define LF_BAND 1
#define NUM_BANDS 2
/* These points are in AL coordinates! */
static const ALfloat Ambi3DPoints[8][3] = {
{ -0.577350269f, 0.577350269f, -0.577350269f },
{ 0.577350269f, 0.577350269f, -0.577350269f },
{ -0.577350269f, 0.577350269f, 0.577350269f },
{ 0.577350269f, 0.577350269f, 0.577350269f },
{ -0.577350269f, -0.577350269f, -0.577350269f },
{ 0.577350269f, -0.577350269f, -0.577350269f },
{ -0.577350269f, -0.577350269f, 0.577350269f },
{ 0.577350269f, -0.577350269f, 0.577350269f },
};
static const ALfloat Ambi3DDecoder[8][MAX_AMBI_COEFFS] = {
{ 0.125f, 0.125f, 0.125f, 0.125f },
{ 0.125f, -0.125f, 0.125f, 0.125f },
{ 0.125f, 0.125f, 0.125f, -0.125f },
{ 0.125f, -0.125f, 0.125f, -0.125f },
{ 0.125f, 0.125f, -0.125f, 0.125f },
{ 0.125f, -0.125f, -0.125f, 0.125f },
{ 0.125f, 0.125f, -0.125f, -0.125f },
{ 0.125f, -0.125f, -0.125f, -0.125f },
};
static const ALfloat Ambi3DDecoderHFScale[MAX_AMBI_COEFFS] = {
2.0f,
1.15470054f, 1.15470054f, 1.15470054f
};
/* NOTE: BandSplitter filters are unused with single-band decoding */
typedef struct BFormatDec {
ALuint Enabled; /* Bitfield of enabled channels. */
union {
alignas(16) ALfloat Dual[MAX_OUTPUT_CHANNELS][NUM_BANDS][MAX_AMBI_COEFFS];
alignas(16) ALfloat Single[MAX_OUTPUT_CHANNELS][MAX_AMBI_COEFFS];
} Matrix;
BandSplitter XOver[MAX_AMBI_COEFFS];
ALfloat (*Samples)[BUFFERSIZE];
/* These two alias into Samples */
ALfloat (*SamplesHF)[BUFFERSIZE];
ALfloat (*SamplesLF)[BUFFERSIZE];
alignas(16) ALfloat ChannelMix[BUFFERSIZE];
struct {
BandSplitter XOver;
ALfloat Gains[NUM_BANDS];
} UpSampler[4];
ALsizei NumChannels;
ALboolean DualBand;
} BFormatDec;
BFormatDec *bformatdec_alloc()
{
return al_calloc(16, sizeof(BFormatDec));
}
void bformatdec_free(BFormatDec **dec)
{
if(dec && *dec)
{
al_free((*dec)->Samples);
(*dec)->Samples = NULL;
(*dec)->SamplesHF = NULL;
(*dec)->SamplesLF = NULL;
al_free(*dec);
*dec = NULL;
}
}
void bformatdec_reset(BFormatDec *dec, const AmbDecConf *conf, ALsizei chancount, ALuint srate, const ALsizei chanmap[MAX_OUTPUT_CHANNELS])
{
static const ALsizei map2DTo3D[MAX_AMBI2D_COEFFS] = {
0, 1, 3, 4, 8, 9, 15
};
const ALfloat *coeff_scale = N3D2N3DScale;
bool periphonic;
ALfloat ratio;
ALsizei i;
al_free(dec->Samples);
dec->Samples = NULL;
dec->SamplesHF = NULL;
dec->SamplesLF = NULL;
dec->NumChannels = chancount;
dec->Samples = al_calloc(16, dec->NumChannels*2 * sizeof(dec->Samples[0]));
dec->SamplesHF = dec->Samples;
dec->SamplesLF = dec->SamplesHF + dec->NumChannels;
dec->Enabled = 0;
for(i = 0;i < conf->NumSpeakers;i++)
dec->Enabled |= 1 << chanmap[i];
if(conf->CoeffScale == ADS_SN3D)
coeff_scale = SN3D2N3DScale;
else if(conf->CoeffScale == ADS_FuMa)
coeff_scale = FuMa2N3DScale;
memset(dec->UpSampler, 0, sizeof(dec->UpSampler));
ratio = 400.0f / (ALfloat)srate;
for(i = 0;i < 4;i++)
bandsplit_init(&dec->UpSampler[i].XOver, ratio);
if((conf->ChanMask&AMBI_PERIPHONIC_MASK))
{
periphonic = true;
dec->UpSampler[0].Gains[HF_BAND] = (conf->ChanMask > 0x1ff) ? W_SCALE_3H3P :
(conf->ChanMask > 0xf) ? W_SCALE_2H2P : 1.0f;
dec->UpSampler[0].Gains[LF_BAND] = 1.0f;
for(i = 1;i < 4;i++)
{
dec->UpSampler[i].Gains[HF_BAND] = (conf->ChanMask > 0x1ff) ? XYZ_SCALE_3H3P :
(conf->ChanMask > 0xf) ? XYZ_SCALE_2H2P : 1.0f;
dec->UpSampler[i].Gains[LF_BAND] = 1.0f;
}
}
else
{
periphonic = false;
dec->UpSampler[0].Gains[HF_BAND] = (conf->ChanMask > 0x1ff) ? W_SCALE_3H0P :
(conf->ChanMask > 0xf) ? W_SCALE_2H0P : 1.0f;
dec->UpSampler[0].Gains[LF_BAND] = 1.0f;
for(i = 1;i < 3;i++)
{
dec->UpSampler[i].Gains[HF_BAND] = (conf->ChanMask > 0x1ff) ? XYZ_SCALE_3H0P :
(conf->ChanMask > 0xf) ? XYZ_SCALE_2H0P : 1.0f;
dec->UpSampler[i].Gains[LF_BAND] = 1.0f;
}
dec->UpSampler[3].Gains[HF_BAND] = 0.0f;
dec->UpSampler[3].Gains[LF_BAND] = 0.0f;
}
memset(&dec->Matrix, 0, sizeof(dec->Matrix));
if(conf->FreqBands == 1)
{
dec->DualBand = AL_FALSE;
for(i = 0;i < conf->NumSpeakers;i++)
{
ALsizei chan = chanmap[i];
ALfloat gain;
ALsizei j, k;
if(!periphonic)
{
for(j = 0,k = 0;j < MAX_AMBI2D_COEFFS;j++)
{
ALsizei l = map2DTo3D[j];
if(j == 0) gain = conf->HFOrderGain[0];
else if(j == 1) gain = conf->HFOrderGain[1];
else if(j == 3) gain = conf->HFOrderGain[2];
else if(j == 5) gain = conf->HFOrderGain[3];
if((conf->ChanMask&(1<<l)))
dec->Matrix.Single[chan][j] = conf->HFMatrix[i][k++] / coeff_scale[l] *
gain;
}
}
else
{
for(j = 0,k = 0;j < MAX_AMBI_COEFFS;j++)
{
if(j == 0) gain = conf->HFOrderGain[0];
else if(j == 1) gain = conf->HFOrderGain[1];
else if(j == 4) gain = conf->HFOrderGain[2];
else if(j == 9) gain = conf->HFOrderGain[3];
if((conf->ChanMask&(1<<j)))
dec->Matrix.Single[chan][j] = conf->HFMatrix[i][k++] / coeff_scale[j] *
gain;
}
}
}
}
else
{
dec->DualBand = AL_TRUE;
ratio = conf->XOverFreq / (ALfloat)srate;
for(i = 0;i < MAX_AMBI_COEFFS;i++)
bandsplit_init(&dec->XOver[i], ratio);
ratio = powf(10.0f, conf->XOverRatio / 40.0f);
for(i = 0;i < conf->NumSpeakers;i++)
{
ALsizei chan = chanmap[i];
ALfloat gain;
ALsizei j, k;
if(!periphonic)
{
for(j = 0,k = 0;j < MAX_AMBI2D_COEFFS;j++)
{
ALsizei l = map2DTo3D[j];
if(j == 0) gain = conf->HFOrderGain[0] * ratio;
else if(j == 1) gain = conf->HFOrderGain[1] * ratio;
else if(j == 3) gain = conf->HFOrderGain[2] * ratio;
else if(j == 5) gain = conf->HFOrderGain[3] * ratio;
if((conf->ChanMask&(1<<l)))
dec->Matrix.Dual[chan][HF_BAND][j] = conf->HFMatrix[i][k++] /
coeff_scale[l] * gain;
}
for(j = 0,k = 0;j < MAX_AMBI2D_COEFFS;j++)
{
ALsizei l = map2DTo3D[j];
if(j == 0) gain = conf->LFOrderGain[0] / ratio;
else if(j == 1) gain = conf->LFOrderGain[1] / ratio;
else if(j == 3) gain = conf->LFOrderGain[2] / ratio;
else if(j == 5) gain = conf->LFOrderGain[3] / ratio;
if((conf->ChanMask&(1<<l)))
dec->Matrix.Dual[chan][LF_BAND][j] = conf->LFMatrix[i][k++] /
coeff_scale[l] * gain;
}
}
else
{
for(j = 0,k = 0;j < MAX_AMBI_COEFFS;j++)
{
if(j == 0) gain = conf->HFOrderGain[0] * ratio;
else if(j == 1) gain = conf->HFOrderGain[1] * ratio;
else if(j == 4) gain = conf->HFOrderGain[2] * ratio;
else if(j == 9) gain = conf->HFOrderGain[3] * ratio;
if((conf->ChanMask&(1<<j)))
dec->Matrix.Dual[chan][HF_BAND][j] = conf->HFMatrix[i][k++] /
coeff_scale[j] * gain;
}
for(j = 0,k = 0;j < MAX_AMBI_COEFFS;j++)
{
if(j == 0) gain = conf->LFOrderGain[0] / ratio;
else if(j == 1) gain = conf->LFOrderGain[1] / ratio;
else if(j == 4) gain = conf->LFOrderGain[2] / ratio;
else if(j == 9) gain = conf->LFOrderGain[3] / ratio;
if((conf->ChanMask&(1<<j)))
dec->Matrix.Dual[chan][LF_BAND][j] = conf->LFMatrix[i][k++] /
coeff_scale[j] * gain;
}
}
}
}
}
void bformatdec_process(struct BFormatDec *dec, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALsizei OutChannels, const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo)
{
ALsizei chan, i;
OutBuffer = ASSUME_ALIGNED(OutBuffer, 16);
if(dec->DualBand)
{
for(i = 0;i < dec->NumChannels;i++)
bandsplit_process(&dec->XOver[i], dec->SamplesHF[i], dec->SamplesLF[i],
InSamples[i], SamplesToDo);
for(chan = 0;chan < OutChannels;chan++)
{
if(!(dec->Enabled&(1<<chan)))
continue;
memset(dec->ChannelMix, 0, SamplesToDo*sizeof(ALfloat));
MixRowSamples(dec->ChannelMix, dec->Matrix.Dual[chan][HF_BAND],
dec->SamplesHF, dec->NumChannels, 0, SamplesToDo
);
MixRowSamples(dec->ChannelMix, dec->Matrix.Dual[chan][LF_BAND],
dec->SamplesLF, dec->NumChannels, 0, SamplesToDo
);
for(i = 0;i < SamplesToDo;i++)
OutBuffer[chan][i] += dec->ChannelMix[i];
}
}
else
{
for(chan = 0;chan < OutChannels;chan++)
{
if(!(dec->Enabled&(1<<chan)))
continue;
memset(dec->ChannelMix, 0, SamplesToDo*sizeof(ALfloat));
MixRowSamples(dec->ChannelMix, dec->Matrix.Single[chan], InSamples,
dec->NumChannels, 0, SamplesToDo);
for(i = 0;i < SamplesToDo;i++)
OutBuffer[chan][i] += dec->ChannelMix[i];
}
}
}
void bformatdec_upSample(struct BFormatDec *dec, ALfloat (*restrict OutBuffer)[BUFFERSIZE], const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei InChannels, ALsizei SamplesToDo)
{
ALsizei i;
/* This up-sampler leverages the differences observed in dual-band second-
* and third-order decoder matrices compared to first-order. For the same
* output channel configuration, the low-frequency matrix has identical
* coefficients in the shared input channels, while the high-frequency
* matrix has extra scalars applied to the W channel and X/Y/Z channels.
* Mixing the first-order content into the higher-order stream with the
* appropriate counter-scales applied to the HF response results in the
* subsequent higher-order decode generating the same response as a first-
* order decode.
*/
for(i = 0;i < InChannels;i++)
{
/* First, split the first-order components into low and high frequency
* bands.
*/
bandsplit_process(&dec->UpSampler[i].XOver,
dec->Samples[HF_BAND], dec->Samples[LF_BAND],
InSamples[i], SamplesToDo
);
/* Now write each band to the output. */
MixRowSamples(OutBuffer[i], dec->UpSampler[i].Gains,
dec->Samples, NUM_BANDS, 0, SamplesToDo
);
}
}
#define INVALID_UPSAMPLE_INDEX INT_MAX
static ALsizei GetACNIndex(const BFChannelConfig *chans, ALsizei numchans, ALsizei acn)
{
ALsizei i;
for(i = 0;i < numchans;i++)
{
if(chans[i].Index == acn)
return i;
}
return INVALID_UPSAMPLE_INDEX;
}
#define GetChannelForACN(b, a) GetACNIndex((b).Ambi.Map, (b).NumChannels, (a))
typedef struct AmbiUpsampler {
alignas(16) ALfloat Samples[NUM_BANDS][BUFFERSIZE];
BandSplitter XOver[4];
ALfloat Gains[4][MAX_OUTPUT_CHANNELS][NUM_BANDS];
} AmbiUpsampler;
AmbiUpsampler *ambiup_alloc()
{
return al_calloc(16, sizeof(AmbiUpsampler));
}
void ambiup_free(struct AmbiUpsampler **ambiup)
{
if(ambiup)
{
al_free(*ambiup);
*ambiup = NULL;
}
}
void ambiup_reset(struct AmbiUpsampler *ambiup, const ALCdevice *device, ALfloat w_scale, ALfloat xyz_scale)
{
ALfloat ratio;
ALsizei i;
ratio = 400.0f / (ALfloat)device->Frequency;
for(i = 0;i < 4;i++)
bandsplit_init(&ambiup->XOver[i], ratio);
memset(ambiup->Gains, 0, sizeof(ambiup->Gains));
if(device->Dry.CoeffCount > 0)
{
ALfloat encgains[8][MAX_OUTPUT_CHANNELS];
ALsizei j;
size_t k;
for(k = 0;k < COUNTOF(Ambi3DPoints);k++)
{
ALfloat coeffs[MAX_AMBI_COEFFS] = { 0.0f };
CalcDirectionCoeffs(Ambi3DPoints[k], 0.0f, coeffs);
ComputePanGains(&device->Dry, coeffs, 1.0f, encgains[k]);
}
/* Combine the matrices that do the in->virt and virt->out conversions
* so we get a single in->out conversion. NOTE: the Encoder matrix
* (encgains) and output are transposed, so the input channels line up
* with the rows and the output channels line up with the columns.
*/
for(i = 0;i < 4;i++)
{
for(j = 0;j < device->Dry.NumChannels;j++)
{
ALdouble gain = 0.0;
for(k = 0;k < COUNTOF(Ambi3DDecoder);k++)
gain += (ALdouble)Ambi3DDecoder[k][i] * encgains[k][j];
ambiup->Gains[i][j][HF_BAND] = (ALfloat)(gain * Ambi3DDecoderHFScale[i]);
ambiup->Gains[i][j][LF_BAND] = (ALfloat)gain;
}
}
}
else
{
for(i = 0;i < 4;i++)
{
ALsizei index = GetChannelForACN(device->Dry, i);
if(index != INVALID_UPSAMPLE_INDEX)
{
ALfloat scale = device->Dry.Ambi.Map[index].Scale;
ambiup->Gains[i][index][HF_BAND] = scale * ((i==0) ? w_scale : xyz_scale);
ambiup->Gains[i][index][LF_BAND] = scale;
}
}
}
}
void ambiup_process(struct AmbiUpsampler *ambiup, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALsizei OutChannels, const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo)
{
ALsizei i, j;
for(i = 0;i < 4;i++)
{
bandsplit_process(&ambiup->XOver[i],
ambiup->Samples[HF_BAND], ambiup->Samples[LF_BAND],
InSamples[i], SamplesToDo
);
for(j = 0;j < OutChannels;j++)
MixRowSamples(OutBuffer[j], ambiup->Gains[i][j],
ambiup->Samples, NUM_BANDS, 0, SamplesToDo
);
}
}
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@@ -1,57 +0,0 @@
#ifndef BFORMATDEC_H
#define BFORMATDEC_H
#include "alMain.h"
/* These are the necessary scales for first-order HF responses to play over
* higher-order 2D (non-periphonic) decoders.
*/
#define W_SCALE_2H0P 1.224744871f /* sqrt(1.5) */
#define XYZ_SCALE_2H0P 1.0f
#define W_SCALE_3H0P 1.414213562f /* sqrt(2) */
#define XYZ_SCALE_3H0P 1.082392196f
/* These are the necessary scales for first-order HF responses to play over
* higher-order 3D (periphonic) decoders.
*/
#define W_SCALE_2H2P 1.341640787f /* sqrt(1.8) */
#define XYZ_SCALE_2H2P 1.0f
#define W_SCALE_3H3P 1.695486018f
#define XYZ_SCALE_3H3P 1.136697713f
/* NOTE: These are scale factors as applied to Ambisonics content. Decoder
* coefficients should be divided by these values to get proper N3D scalings.
*/
const ALfloat N3D2N3DScale[MAX_AMBI_COEFFS];
const ALfloat SN3D2N3DScale[MAX_AMBI_COEFFS];
const ALfloat FuMa2N3DScale[MAX_AMBI_COEFFS];
struct AmbDecConf;
struct BFormatDec;
struct AmbiUpsampler;
struct BFormatDec *bformatdec_alloc();
void bformatdec_free(struct BFormatDec **dec);
void bformatdec_reset(struct BFormatDec *dec, const struct AmbDecConf *conf, ALsizei chancount, ALuint srate, const ALsizei chanmap[MAX_OUTPUT_CHANNELS]);
/* Decodes the ambisonic input to the given output channels. */
void bformatdec_process(struct BFormatDec *dec, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALsizei OutChannels, const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo);
/* Up-samples a first-order input to the decoder's configuration. */
void bformatdec_upSample(struct BFormatDec *dec, ALfloat (*restrict OutBuffer)[BUFFERSIZE], const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei InChannels, ALsizei SamplesToDo);
/* Stand-alone first-order upsampler. Kept here because it shares some stuff
* with bformatdec. Assumes a periphonic (4-channel) input mix!
*/
struct AmbiUpsampler *ambiup_alloc();
void ambiup_free(struct AmbiUpsampler **ambiup);
void ambiup_reset(struct AmbiUpsampler *ambiup, const ALCdevice *device, ALfloat w_scale, ALfloat xyz_scale);
void ambiup_process(struct AmbiUpsampler *ambiup, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALsizei OutChannels, const ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo);
#endif /* BFORMATDEC_H */
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@@ -1,57 +0,0 @@
#ifndef AL_COMPAT_H
#define AL_COMPAT_H
#include "alstring.h"
#ifdef __cplusplus
extern "C" {
#endif
#ifdef _WIN32
#define WIN32_LEAN_AND_MEAN
#include <windows.h>
WCHAR *strdupW(const WCHAR *str);
/* Opens a file with standard I/O. The filename is expected to be UTF-8. */
FILE *al_fopen(const char *fname, const char *mode);
#define HAVE_DYNLOAD 1
#else
#define al_fopen fopen
#if defined(HAVE_DLFCN_H) && !defined(IN_IDE_PARSER)
#define HAVE_DYNLOAD 1
#endif
#endif
struct FileMapping {
#ifdef _WIN32
HANDLE file;
HANDLE fmap;
#else
int fd;
#endif
void *ptr;
size_t len;
};
struct FileMapping MapFileToMem(const char *fname);
void UnmapFileMem(const struct FileMapping *mapping);
void GetProcBinary(al_string *path, al_string *fname);
#ifdef HAVE_DYNLOAD
void *LoadLib(const char *name);
void CloseLib(void *handle);
void *GetSymbol(void *handle, const char *name);
#endif
#ifdef __cplusplus
} /* extern "C" */
#endif
#endif /* AL_COMPAT_H */
-468
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@@ -1,468 +0,0 @@
#include "config.h"
#include "converter.h"
#include "fpu_modes.h"
#include "mixer/defs.h"
SampleConverter *CreateSampleConverter(enum DevFmtType srcType, enum DevFmtType dstType, ALsizei numchans, ALsizei srcRate, ALsizei dstRate)
{
SampleConverter *converter;
ALsizei step;
if(numchans <= 0 || srcRate <= 0 || dstRate <= 0)
return NULL;
converter = al_calloc(16, FAM_SIZE(SampleConverter, Chan, numchans));
converter->mSrcType = srcType;
converter->mDstType = dstType;
converter->mNumChannels = numchans;
converter->mSrcTypeSize = BytesFromDevFmt(srcType);
converter->mDstTypeSize = BytesFromDevFmt(dstType);
converter->mSrcPrepCount = 0;
converter->mFracOffset = 0;
/* Have to set the mixer FPU mode since that's what the resampler code expects. */
START_MIXER_MODE();
step = (ALsizei)mind(((ALdouble)srcRate/dstRate*FRACTIONONE) + 0.5,
MAX_PITCH * FRACTIONONE);
converter->mIncrement = maxi(step, 1);
if(converter->mIncrement == FRACTIONONE)
converter->mResample = Resample_copy_C;
else
{
/* TODO: Allow other resamplers. */
BsincPrepare(converter->mIncrement, &converter->mState.bsinc, &bsinc12);
converter->mResample = SelectResampler(BSinc12Resampler);
}
END_MIXER_MODE();
return converter;
}
void DestroySampleConverter(SampleConverter **converter)
{
if(converter)
{
al_free(*converter);
*converter = NULL;
}
}
static inline ALfloat Sample_ALbyte(ALbyte val)
{ return val * (1.0f/128.0f); }
static inline ALfloat Sample_ALubyte(ALubyte val)
{ return Sample_ALbyte((ALint)val - 128); }
static inline ALfloat Sample_ALshort(ALshort val)
{ return val * (1.0f/32768.0f); }
static inline ALfloat Sample_ALushort(ALushort val)
{ return Sample_ALshort((ALint)val - 32768); }
static inline ALfloat Sample_ALint(ALint val)
{ return (val>>7) * (1.0f/16777216.0f); }
static inline ALfloat Sample_ALuint(ALuint val)
{ return Sample_ALint(val - INT_MAX - 1); }
static inline ALfloat Sample_ALfloat(ALfloat val)
{ return val; }
#define DECL_TEMPLATE(T) \
static inline void Load_##T(ALfloat *restrict dst, const T *restrict src, \
ALint srcstep, ALsizei samples) \
{ \
ALsizei i; \
for(i = 0;i < samples;i++) \
dst[i] = Sample_##T(src[i*srcstep]); \
}
DECL_TEMPLATE(ALbyte)
DECL_TEMPLATE(ALubyte)
DECL_TEMPLATE(ALshort)
DECL_TEMPLATE(ALushort)
DECL_TEMPLATE(ALint)
DECL_TEMPLATE(ALuint)
DECL_TEMPLATE(ALfloat)
#undef DECL_TEMPLATE
static void LoadSamples(ALfloat *dst, const ALvoid *src, ALint srcstep, enum DevFmtType srctype, ALsizei samples)
{
switch(srctype)
{
case DevFmtByte:
Load_ALbyte(dst, src, srcstep, samples);
break;
case DevFmtUByte:
Load_ALubyte(dst, src, srcstep, samples);
break;
case DevFmtShort:
Load_ALshort(dst, src, srcstep, samples);
break;
case DevFmtUShort:
Load_ALushort(dst, src, srcstep, samples);
break;
case DevFmtInt:
Load_ALint(dst, src, srcstep, samples);
break;
case DevFmtUInt:
Load_ALuint(dst, src, srcstep, samples);
break;
case DevFmtFloat:
Load_ALfloat(dst, src, srcstep, samples);
break;
}
}
static inline ALbyte ALbyte_Sample(ALfloat val)
{ return fastf2i(clampf(val*128.0f, -128.0f, 127.0f)); }
static inline ALubyte ALubyte_Sample(ALfloat val)
{ return ALbyte_Sample(val)+128; }
static inline ALshort ALshort_Sample(ALfloat val)
{ return fastf2i(clampf(val*32768.0f, -32768.0f, 32767.0f)); }
static inline ALushort ALushort_Sample(ALfloat val)
{ return ALshort_Sample(val)+32768; }
static inline ALint ALint_Sample(ALfloat val)
{ return fastf2i(clampf(val*16777216.0f, -16777216.0f, 16777215.0f)) << 7; }
static inline ALuint ALuint_Sample(ALfloat val)
{ return ALint_Sample(val)+INT_MAX+1; }
static inline ALfloat ALfloat_Sample(ALfloat val)
{ return val; }
#define DECL_TEMPLATE(T) \
static inline void Store_##T(T *restrict dst, const ALfloat *restrict src, \
ALint dststep, ALsizei samples) \
{ \
ALsizei i; \
for(i = 0;i < samples;i++) \
dst[i*dststep] = T##_Sample(src[i]); \
}
DECL_TEMPLATE(ALbyte)
DECL_TEMPLATE(ALubyte)
DECL_TEMPLATE(ALshort)
DECL_TEMPLATE(ALushort)
DECL_TEMPLATE(ALint)
DECL_TEMPLATE(ALuint)
DECL_TEMPLATE(ALfloat)
#undef DECL_TEMPLATE
static void StoreSamples(ALvoid *dst, const ALfloat *src, ALint dststep, enum DevFmtType dsttype, ALsizei samples)
{
switch(dsttype)
{
case DevFmtByte:
Store_ALbyte(dst, src, dststep, samples);
break;
case DevFmtUByte:
Store_ALubyte(dst, src, dststep, samples);
break;
case DevFmtShort:
Store_ALshort(dst, src, dststep, samples);
break;
case DevFmtUShort:
Store_ALushort(dst, src, dststep, samples);
break;
case DevFmtInt:
Store_ALint(dst, src, dststep, samples);
break;
case DevFmtUInt:
Store_ALuint(dst, src, dststep, samples);
break;
case DevFmtFloat:
Store_ALfloat(dst, src, dststep, samples);
break;
}
}
ALsizei SampleConverterAvailableOut(SampleConverter *converter, ALsizei srcframes)
{
ALint prepcount = converter->mSrcPrepCount;
ALsizei increment = converter->mIncrement;
ALsizei DataPosFrac = converter->mFracOffset;
ALuint64 DataSize64;
if(prepcount < 0)
{
/* Negative prepcount means we need to skip that many input samples. */
if(-prepcount >= srcframes)
return 0;
srcframes += prepcount;
prepcount = 0;
}
if(srcframes < 1)
{
/* No output samples if there's no input samples. */
return 0;
}
if(prepcount < MAX_RESAMPLE_PADDING*2 &&
MAX_RESAMPLE_PADDING*2 - prepcount >= srcframes)
{
/* Not enough input samples to generate an output sample. */
return 0;
}
DataSize64 = prepcount;
DataSize64 += srcframes;
DataSize64 -= MAX_RESAMPLE_PADDING*2;
DataSize64 <<= FRACTIONBITS;
DataSize64 -= DataPosFrac;
/* If we have a full prep, we can generate at least one sample. */
return (ALsizei)clampu64((DataSize64 + increment-1)/increment, 1, BUFFERSIZE);
}
ALsizei SampleConverterInput(SampleConverter *converter, const ALvoid **src, ALsizei *srcframes, ALvoid *dst, ALsizei dstframes)
{
const ALsizei SrcFrameSize = converter->mNumChannels * converter->mSrcTypeSize;
const ALsizei DstFrameSize = converter->mNumChannels * converter->mDstTypeSize;
const ALsizei increment = converter->mIncrement;
ALsizei pos = 0;
START_MIXER_MODE();
while(pos < dstframes && *srcframes > 0)
{
ALfloat *restrict SrcData = ASSUME_ALIGNED(converter->mSrcSamples, 16);
ALfloat *restrict DstData = ASSUME_ALIGNED(converter->mDstSamples, 16);
ALint prepcount = converter->mSrcPrepCount;
ALsizei DataPosFrac = converter->mFracOffset;
ALuint64 DataSize64;
ALsizei DstSize;
ALint toread;
ALsizei chan;
if(prepcount < 0)
{
/* Negative prepcount means we need to skip that many input samples. */
if(-prepcount >= *srcframes)
{
converter->mSrcPrepCount = prepcount + *srcframes;
*srcframes = 0;
break;
}
*src = (const ALbyte*)*src + SrcFrameSize*-prepcount;
*srcframes += prepcount;
converter->mSrcPrepCount = 0;
continue;
}
toread = mini(*srcframes, BUFFERSIZE - MAX_RESAMPLE_PADDING*2);
if(prepcount < MAX_RESAMPLE_PADDING*2 &&
MAX_RESAMPLE_PADDING*2 - prepcount >= toread)
{
/* Not enough input samples to generate an output sample. Store
* what we're given for later.
*/
for(chan = 0;chan < converter->mNumChannels;chan++)
LoadSamples(&converter->Chan[chan].mPrevSamples[prepcount],
(const ALbyte*)*src + converter->mSrcTypeSize*chan,
converter->mNumChannels, converter->mSrcType, toread
);
converter->mSrcPrepCount = prepcount + toread;
*srcframes = 0;
break;
}
DataSize64 = prepcount;
DataSize64 += toread;
DataSize64 -= MAX_RESAMPLE_PADDING*2;
DataSize64 <<= FRACTIONBITS;
DataSize64 -= DataPosFrac;
/* If we have a full prep, we can generate at least one sample. */
DstSize = (ALsizei)clampu64((DataSize64 + increment-1)/increment, 1, BUFFERSIZE);
DstSize = mini(DstSize, dstframes-pos);
for(chan = 0;chan < converter->mNumChannels;chan++)
{
const ALbyte *SrcSamples = (const ALbyte*)*src + converter->mSrcTypeSize*chan;
ALbyte *DstSamples = (ALbyte*)dst + converter->mDstTypeSize*chan;
const ALfloat *ResampledData;
ALsizei SrcDataEnd;
/* Load the previous samples into the source data first, then the
* new samples from the input buffer.
*/
memcpy(SrcData, converter->Chan[chan].mPrevSamples,
prepcount*sizeof(ALfloat));
LoadSamples(SrcData + prepcount, SrcSamples,
converter->mNumChannels, converter->mSrcType, toread
);
/* Store as many prep samples for next time as possible, given the
* number of output samples being generated.
*/
SrcDataEnd = (DataPosFrac + increment*DstSize)>>FRACTIONBITS;
if(SrcDataEnd >= prepcount+toread)
memset(converter->Chan[chan].mPrevSamples, 0,
sizeof(converter->Chan[chan].mPrevSamples));
else
{
size_t len = mini(MAX_RESAMPLE_PADDING*2, prepcount+toread-SrcDataEnd);
memcpy(converter->Chan[chan].mPrevSamples, &SrcData[SrcDataEnd],
len*sizeof(ALfloat));
memset(converter->Chan[chan].mPrevSamples+len, 0,
sizeof(converter->Chan[chan].mPrevSamples) - len*sizeof(ALfloat));
}
/* Now resample, and store the result in the output buffer. */
ResampledData = converter->mResample(&converter->mState,
SrcData+MAX_RESAMPLE_PADDING, DataPosFrac, increment,
DstData, DstSize
);
StoreSamples(DstSamples, ResampledData, converter->mNumChannels,
converter->mDstType, DstSize);
}
/* Update the number of prep samples still available, as well as the
* fractional offset.
*/
DataPosFrac += increment*DstSize;
converter->mSrcPrepCount = mini(prepcount + toread - (DataPosFrac>>FRACTIONBITS),
MAX_RESAMPLE_PADDING*2);
converter->mFracOffset = DataPosFrac & FRACTIONMASK;
/* Update the src and dst pointers in case there's still more to do. */
*src = (const ALbyte*)*src + SrcFrameSize*(DataPosFrac>>FRACTIONBITS);
*srcframes -= mini(*srcframes, (DataPosFrac>>FRACTIONBITS));
dst = (ALbyte*)dst + DstFrameSize*DstSize;
pos += DstSize;
}
END_MIXER_MODE();
return pos;
}
ChannelConverter *CreateChannelConverter(enum DevFmtType srcType, enum DevFmtChannels srcChans, enum DevFmtChannels dstChans)
{
ChannelConverter *converter;
if(srcChans != dstChans && !((srcChans == DevFmtMono && dstChans == DevFmtStereo) ||
(srcChans == DevFmtStereo && dstChans == DevFmtMono)))
return NULL;
converter = al_calloc(DEF_ALIGN, sizeof(*converter));
converter->mSrcType = srcType;
converter->mSrcChans = srcChans;
converter->mDstChans = dstChans;
return converter;
}
void DestroyChannelConverter(ChannelConverter **converter)
{
if(converter)
{
al_free(*converter);
*converter = NULL;
}
}
#define DECL_TEMPLATE(T) \
static void Mono2Stereo##T(ALfloat *restrict dst, const T *src, ALsizei frames)\
{ \
ALsizei i; \
for(i = 0;i < frames;i++) \
dst[i*2 + 1] = dst[i*2 + 0] = Sample_##T(src[i]) * 0.707106781187f; \
} \
\
static void Stereo2Mono##T(ALfloat *restrict dst, const T *src, ALsizei frames)\
{ \
ALsizei i; \
for(i = 0;i < frames;i++) \
dst[i] = (Sample_##T(src[i*2 + 0])+Sample_##T(src[i*2 + 1])) * \
0.707106781187f; \
}
DECL_TEMPLATE(ALbyte)
DECL_TEMPLATE(ALubyte)
DECL_TEMPLATE(ALshort)
DECL_TEMPLATE(ALushort)
DECL_TEMPLATE(ALint)
DECL_TEMPLATE(ALuint)
DECL_TEMPLATE(ALfloat)
#undef DECL_TEMPLATE
void ChannelConverterInput(ChannelConverter *converter, const ALvoid *src, ALfloat *dst, ALsizei frames)
{
if(converter->mSrcChans == converter->mDstChans)
{
LoadSamples(dst, src, 1, converter->mSrcType,
frames*ChannelsFromDevFmt(converter->mSrcChans, 0));
return;
}
if(converter->mSrcChans == DevFmtStereo && converter->mDstChans == DevFmtMono)
{
switch(converter->mSrcType)
{
case DevFmtByte:
Stereo2MonoALbyte(dst, src, frames);
break;
case DevFmtUByte:
Stereo2MonoALubyte(dst, src, frames);
break;
case DevFmtShort:
Stereo2MonoALshort(dst, src, frames);
break;
case DevFmtUShort:
Stereo2MonoALushort(dst, src, frames);
break;
case DevFmtInt:
Stereo2MonoALint(dst, src, frames);
break;
case DevFmtUInt:
Stereo2MonoALuint(dst, src, frames);
break;
case DevFmtFloat:
Stereo2MonoALfloat(dst, src, frames);
break;
}
}
else /*if(converter->mSrcChans == DevFmtMono && converter->mDstChans == DevFmtStereo)*/
{
switch(converter->mSrcType)
{
case DevFmtByte:
Mono2StereoALbyte(dst, src, frames);
break;
case DevFmtUByte:
Mono2StereoALubyte(dst, src, frames);
break;
case DevFmtShort:
Mono2StereoALshort(dst, src, frames);
break;
case DevFmtUShort:
Mono2StereoALushort(dst, src, frames);
break;
case DevFmtInt:
Mono2StereoALint(dst, src, frames);
break;
case DevFmtUInt:
Mono2StereoALuint(dst, src, frames);
break;
case DevFmtFloat:
Mono2StereoALfloat(dst, src, frames);
break;
}
}
}
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#ifndef CONVERTER_H
#define CONVERTER_H
#include "alMain.h"
#include "alu.h"
#ifdef __cpluspluc
extern "C" {
#endif
typedef struct SampleConverter {
enum DevFmtType mSrcType;
enum DevFmtType mDstType;
ALsizei mNumChannels;
ALsizei mSrcTypeSize;
ALsizei mDstTypeSize;
ALint mSrcPrepCount;
ALsizei mFracOffset;
ALsizei mIncrement;
InterpState mState;
ResamplerFunc mResample;
alignas(16) ALfloat mSrcSamples[BUFFERSIZE];
alignas(16) ALfloat mDstSamples[BUFFERSIZE];
struct {
alignas(16) ALfloat mPrevSamples[MAX_RESAMPLE_PADDING*2];
} Chan[];
} SampleConverter;
SampleConverter *CreateSampleConverter(enum DevFmtType srcType, enum DevFmtType dstType, ALsizei numchans, ALsizei srcRate, ALsizei dstRate);
void DestroySampleConverter(SampleConverter **converter);
ALsizei SampleConverterInput(SampleConverter *converter, const ALvoid **src, ALsizei *srcframes, ALvoid *dst, ALsizei dstframes);
ALsizei SampleConverterAvailableOut(SampleConverter *converter, ALsizei srcframes);
typedef struct ChannelConverter {
enum DevFmtType mSrcType;
enum DevFmtChannels mSrcChans;
enum DevFmtChannels mDstChans;
} ChannelConverter;
ChannelConverter *CreateChannelConverter(enum DevFmtType srcType, enum DevFmtChannels srcChans, enum DevFmtChannels dstChans);
void DestroyChannelConverter(ChannelConverter **converter);
void ChannelConverterInput(ChannelConverter *converter, const ALvoid *src, ALfloat *dst, ALsizei frames);
#ifdef __cpluspluc
}
#endif
#endif /* CONVERTER_H */
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/**
* OpenAL cross platform audio library
* Copyright (C) 2018 by Raul Herraiz.
* 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <math.h>
#include <stdlib.h>
#include "alMain.h"
#include "alAuxEffectSlot.h"
#include "alError.h"
#include "alu.h"
#include "filters/defs.h"
#define MIN_FREQ 20.0f
#define MAX_FREQ 2500.0f
#define Q_FACTOR 5.0f
typedef struct ALautowahState {
DERIVE_FROM_TYPE(ALeffectState);
/* Effect parameters */
ALfloat AttackRate;
ALfloat ReleaseRate;
ALfloat ResonanceGain;
ALfloat PeakGain;
ALfloat FreqMinNorm;
ALfloat BandwidthNorm;
ALfloat env_delay;
/* Filter components derived from the envelope. */
struct {
ALfloat cos_w0;
ALfloat alpha;
} Env[BUFFERSIZE];
struct {
/* Effect filters' history. */
struct {
ALfloat z1, z2;
} Filter;
/* Effect gains for each output channel */
ALfloat CurrentGains[MAX_OUTPUT_CHANNELS];
ALfloat TargetGains[MAX_OUTPUT_CHANNELS];
} Chans[MAX_EFFECT_CHANNELS];
/* Effects buffers */
alignas(16) ALfloat BufferOut[BUFFERSIZE];
} ALautowahState;
static ALvoid ALautowahState_Destruct(ALautowahState *state);
static ALboolean ALautowahState_deviceUpdate(ALautowahState *state, ALCdevice *device);
static ALvoid ALautowahState_update(ALautowahState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props);
static ALvoid ALautowahState_process(ALautowahState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
DECLARE_DEFAULT_ALLOCATORS(ALautowahState)
DEFINE_ALEFFECTSTATE_VTABLE(ALautowahState);
static void ALautowahState_Construct(ALautowahState *state)
{
ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
SET_VTABLE2(ALautowahState, ALeffectState, state);
}
static ALvoid ALautowahState_Destruct(ALautowahState *state)
{
ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
}
static ALboolean ALautowahState_deviceUpdate(ALautowahState *state, ALCdevice *UNUSED(device))
{
/* (Re-)initializing parameters and clear the buffers. */
ALsizei i, j;
state->AttackRate = 1.0f;
state->ReleaseRate = 1.0f;
state->ResonanceGain = 10.0f;
state->PeakGain = 4.5f;
state->FreqMinNorm = 4.5e-4f;
state->BandwidthNorm = 0.05f;
state->env_delay = 0.0f;
memset(state->Env, 0, sizeof(state->Env));
for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
{
for(j = 0;j < MAX_OUTPUT_CHANNELS;j++)
state->Chans[i].CurrentGains[j] = 0.0f;
state->Chans[i].Filter.z1 = 0.0f;
state->Chans[i].Filter.z2 = 0.0f;
}
return AL_TRUE;
}
static ALvoid ALautowahState_update(ALautowahState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props)
{
const ALCdevice *device = context->Device;
ALfloat ReleaseTime;
ALsizei i;
ReleaseTime = clampf(props->Autowah.ReleaseTime, 0.001f, 1.0f);
state->AttackRate = expf(-1.0f / (props->Autowah.AttackTime*device->Frequency));
state->ReleaseRate = expf(-1.0f / (ReleaseTime*device->Frequency));
/* 0-20dB Resonance Peak gain */
state->ResonanceGain = sqrtf(log10f(props->Autowah.Resonance)*10.0f / 3.0f);
state->PeakGain = 1.0f - log10f(props->Autowah.PeakGain/AL_AUTOWAH_MAX_PEAK_GAIN);
state->FreqMinNorm = MIN_FREQ / device->Frequency;
state->BandwidthNorm = (MAX_FREQ-MIN_FREQ) / device->Frequency;
STATIC_CAST(ALeffectState,state)->OutBuffer = device->FOAOut.Buffer;
STATIC_CAST(ALeffectState,state)->OutChannels = device->FOAOut.NumChannels;
for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
ComputePanGains(&device->FOAOut, IdentityMatrixf.m[i], slot->Params.Gain,
state->Chans[i].TargetGains);
}
static ALvoid ALautowahState_process(ALautowahState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
{
const ALfloat attack_rate = state->AttackRate;
const ALfloat release_rate = state->ReleaseRate;
const ALfloat res_gain = state->ResonanceGain;
const ALfloat peak_gain = state->PeakGain;
const ALfloat freq_min = state->FreqMinNorm;
const ALfloat bandwidth = state->BandwidthNorm;
ALfloat env_delay;
ALsizei c, i;
env_delay = state->env_delay;
for(i = 0;i < SamplesToDo;i++)
{
ALfloat w0, sample, a;
/* Envelope follower described on the book: Audio Effects, Theory,
* Implementation and Application.
*/
sample = peak_gain * fabsf(SamplesIn[0][i]);
a = (sample > env_delay) ? attack_rate : release_rate;
env_delay = lerp(sample, env_delay, a);
/* Calculate the cos and alpha components for this sample's filter. */
w0 = minf((bandwidth*env_delay + freq_min), 0.46f) * F_TAU;
state->Env[i].cos_w0 = cosf(w0);
state->Env[i].alpha = sinf(w0)/(2.0f * Q_FACTOR);
}
state->env_delay = env_delay;
for(c = 0;c < MAX_EFFECT_CHANNELS; c++)
{
/* This effectively inlines BiquadFilter_setParams for a peaking
* filter and BiquadFilter_processC. The alpha and cosine components
* for the filter coefficients were previously calculated with the
* envelope. Because the filter changes for each sample, the
* coefficients are transient and don't need to be held.
*/
ALfloat z1 = state->Chans[c].Filter.z1;
ALfloat z2 = state->Chans[c].Filter.z2;
for(i = 0;i < SamplesToDo;i++)
{
const ALfloat alpha = state->Env[i].alpha;
const ALfloat cos_w0 = state->Env[i].cos_w0;
ALfloat input, output;
ALfloat a[3], b[3];
b[0] = 1.0f + alpha*res_gain;
b[1] = -2.0f * cos_w0;
b[2] = 1.0f - alpha*res_gain;
a[0] = 1.0f + alpha/res_gain;
a[1] = -2.0f * cos_w0;
a[2] = 1.0f - alpha/res_gain;
input = SamplesIn[c][i];
output = input*(b[0]/a[0]) + z1;
z1 = input*(b[1]/a[0]) - output*(a[1]/a[0]) + z2;
z2 = input*(b[2]/a[0]) - output*(a[2]/a[0]);
state->BufferOut[i] = output;
}
state->Chans[c].Filter.z1 = z1;
state->Chans[c].Filter.z2 = z2;
/* Now, mix the processed sound data to the output. */
MixSamples(state->BufferOut, NumChannels, SamplesOut, state->Chans[c].CurrentGains,
state->Chans[c].TargetGains, SamplesToDo, 0, SamplesToDo);
}
}
typedef struct AutowahStateFactory {
DERIVE_FROM_TYPE(EffectStateFactory);
} AutowahStateFactory;
static ALeffectState *AutowahStateFactory_create(AutowahStateFactory *UNUSED(factory))
{
ALautowahState *state;
NEW_OBJ0(state, ALautowahState)();
if(!state) return NULL;
return STATIC_CAST(ALeffectState, state);
}
DEFINE_EFFECTSTATEFACTORY_VTABLE(AutowahStateFactory);
EffectStateFactory *AutowahStateFactory_getFactory(void)
{
static AutowahStateFactory AutowahFactory = { { GET_VTABLE2(AutowahStateFactory, EffectStateFactory) } };
return STATIC_CAST(EffectStateFactory, &AutowahFactory);
}
void ALautowah_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
{
ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_AUTOWAH_ATTACK_TIME:
if(!(val >= AL_AUTOWAH_MIN_ATTACK_TIME && val <= AL_AUTOWAH_MAX_ATTACK_TIME))
SETERR_RETURN(context, AL_INVALID_VALUE,,"Autowah attack time out of range");
props->Autowah.AttackTime = val;
break;
case AL_AUTOWAH_RELEASE_TIME:
if(!(val >= AL_AUTOWAH_MIN_RELEASE_TIME && val <= AL_AUTOWAH_MAX_RELEASE_TIME))
SETERR_RETURN(context, AL_INVALID_VALUE,,"Autowah release time out of range");
props->Autowah.ReleaseTime = val;
break;
case AL_AUTOWAH_RESONANCE:
if(!(val >= AL_AUTOWAH_MIN_RESONANCE && val <= AL_AUTOWAH_MAX_RESONANCE))
SETERR_RETURN(context, AL_INVALID_VALUE,,"Autowah resonance out of range");
props->Autowah.Resonance = val;
break;
case AL_AUTOWAH_PEAK_GAIN:
if(!(val >= AL_AUTOWAH_MIN_PEAK_GAIN && val <= AL_AUTOWAH_MAX_PEAK_GAIN))
SETERR_RETURN(context, AL_INVALID_VALUE,,"Autowah peak gain out of range");
props->Autowah.PeakGain = val;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid autowah float property 0x%04x", param);
}
}
void ALautowah_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
{
ALautowah_setParamf(effect, context, param, vals[0]);
}
void ALautowah_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint UNUSED(val))
{
alSetError(context, AL_INVALID_ENUM, "Invalid autowah integer property 0x%04x", param);
}
void ALautowah_setParamiv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALint *UNUSED(vals))
{
alSetError(context, AL_INVALID_ENUM, "Invalid autowah integer vector property 0x%04x", param);
}
void ALautowah_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(val))
{
alSetError(context, AL_INVALID_ENUM, "Invalid autowah integer property 0x%04x", param);
}
void ALautowah_getParamiv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(vals))
{
alSetError(context, AL_INVALID_ENUM, "Invalid autowah integer vector property 0x%04x", param);
}
void ALautowah_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
{
const ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_AUTOWAH_ATTACK_TIME:
*val = props->Autowah.AttackTime;
break;
case AL_AUTOWAH_RELEASE_TIME:
*val = props->Autowah.ReleaseTime;
break;
case AL_AUTOWAH_RESONANCE:
*val = props->Autowah.Resonance;
break;
case AL_AUTOWAH_PEAK_GAIN:
*val = props->Autowah.PeakGain;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid autowah float property 0x%04x", param);
}
}
void ALautowah_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
{
ALautowah_getParamf(effect, context, param, vals);
}
DEFINE_ALEFFECT_VTABLE(ALautowah);
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/**
* OpenAL cross platform audio library
* Copyright (C) 2013 by Mike Gorchak
* 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <math.h>
#include <stdlib.h>
#include "alMain.h"
#include "alAuxEffectSlot.h"
#include "alError.h"
#include "alu.h"
#include "filters/defs.h"
static_assert(AL_CHORUS_WAVEFORM_SINUSOID == AL_FLANGER_WAVEFORM_SINUSOID, "Chorus/Flanger waveform value mismatch");
static_assert(AL_CHORUS_WAVEFORM_TRIANGLE == AL_FLANGER_WAVEFORM_TRIANGLE, "Chorus/Flanger waveform value mismatch");
enum WaveForm {
WF_Sinusoid,
WF_Triangle
};
typedef struct ALchorusState {
DERIVE_FROM_TYPE(ALeffectState);
ALfloat *SampleBuffer;
ALsizei BufferLength;
ALsizei offset;
ALsizei lfo_offset;
ALsizei lfo_range;
ALfloat lfo_scale;
ALint lfo_disp;
/* Gains for left and right sides */
struct {
ALfloat Current[MAX_OUTPUT_CHANNELS];
ALfloat Target[MAX_OUTPUT_CHANNELS];
} Gains[2];
/* effect parameters */
enum WaveForm waveform;
ALint delay;
ALfloat depth;
ALfloat feedback;
} ALchorusState;
static ALvoid ALchorusState_Destruct(ALchorusState *state);
static ALboolean ALchorusState_deviceUpdate(ALchorusState *state, ALCdevice *Device);
static ALvoid ALchorusState_update(ALchorusState *state, const ALCcontext *Context, const ALeffectslot *Slot, const ALeffectProps *props);
static ALvoid ALchorusState_process(ALchorusState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
DECLARE_DEFAULT_ALLOCATORS(ALchorusState)
DEFINE_ALEFFECTSTATE_VTABLE(ALchorusState);
static void ALchorusState_Construct(ALchorusState *state)
{
ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
SET_VTABLE2(ALchorusState, ALeffectState, state);
state->BufferLength = 0;
state->SampleBuffer = NULL;
state->offset = 0;
state->lfo_offset = 0;
state->lfo_range = 1;
state->waveform = WF_Triangle;
}
static ALvoid ALchorusState_Destruct(ALchorusState *state)
{
al_free(state->SampleBuffer);
state->SampleBuffer = NULL;
ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
}
static ALboolean ALchorusState_deviceUpdate(ALchorusState *state, ALCdevice *Device)
{
const ALfloat max_delay = maxf(AL_CHORUS_MAX_DELAY, AL_FLANGER_MAX_DELAY);
ALsizei maxlen;
maxlen = NextPowerOf2(float2int(max_delay*2.0f*Device->Frequency) + 1u);
if(maxlen <= 0) return AL_FALSE;
if(maxlen != state->BufferLength)
{
void *temp = al_calloc(16, maxlen * sizeof(ALfloat));
if(!temp) return AL_FALSE;
al_free(state->SampleBuffer);
state->SampleBuffer = temp;
state->BufferLength = maxlen;
}
memset(state->SampleBuffer, 0, state->BufferLength*sizeof(ALfloat));
memset(state->Gains, 0, sizeof(state->Gains));
return AL_TRUE;
}
static ALvoid ALchorusState_update(ALchorusState *state, const ALCcontext *Context, const ALeffectslot *Slot, const ALeffectProps *props)
{
const ALsizei mindelay = MAX_RESAMPLE_PADDING << FRACTIONBITS;
const ALCdevice *device = Context->Device;
ALfloat frequency = (ALfloat)device->Frequency;
ALfloat coeffs[MAX_AMBI_COEFFS];
ALfloat rate;
ALint phase;
switch(props->Chorus.Waveform)
{
case AL_CHORUS_WAVEFORM_TRIANGLE:
state->waveform = WF_Triangle;
break;
case AL_CHORUS_WAVEFORM_SINUSOID:
state->waveform = WF_Sinusoid;
break;
}
/* The LFO depth is scaled to be relative to the sample delay. Clamp the
* delay and depth to allow enough padding for resampling.
*/
state->delay = maxi(float2int(props->Chorus.Delay*frequency*FRACTIONONE + 0.5f),
mindelay);
state->depth = minf(props->Chorus.Depth * state->delay,
(ALfloat)(state->delay - mindelay));
state->feedback = props->Chorus.Feedback;
/* Gains for left and right sides */
CalcAngleCoeffs(-F_PI_2, 0.0f, 0.0f, coeffs);
ComputePanGains(&device->Dry, coeffs, Slot->Params.Gain, state->Gains[0].Target);
CalcAngleCoeffs( F_PI_2, 0.0f, 0.0f, coeffs);
ComputePanGains(&device->Dry, coeffs, Slot->Params.Gain, state->Gains[1].Target);
phase = props->Chorus.Phase;
rate = props->Chorus.Rate;
if(!(rate > 0.0f))
{
state->lfo_offset = 0;
state->lfo_range = 1;
state->lfo_scale = 0.0f;
state->lfo_disp = 0;
}
else
{
/* Calculate LFO coefficient (number of samples per cycle). Limit the
* max range to avoid overflow when calculating the displacement.
*/
ALsizei lfo_range = float2int(minf(frequency/rate + 0.5f, (ALfloat)(INT_MAX/360 - 180)));
state->lfo_offset = float2int((ALfloat)state->lfo_offset/state->lfo_range*
lfo_range + 0.5f) % lfo_range;
state->lfo_range = lfo_range;
switch(state->waveform)
{
case WF_Triangle:
state->lfo_scale = 4.0f / state->lfo_range;
break;
case WF_Sinusoid:
state->lfo_scale = F_TAU / state->lfo_range;
break;
}
/* Calculate lfo phase displacement */
if(phase < 0) phase = 360 + phase;
state->lfo_disp = (state->lfo_range*phase + 180) / 360;
}
}
static void GetTriangleDelays(ALint *restrict delays, ALsizei offset, const ALsizei lfo_range,
const ALfloat lfo_scale, const ALfloat depth, const ALsizei delay,
const ALsizei todo)
{
ALsizei i;
for(i = 0;i < todo;i++)
{
delays[i] = fastf2i((1.0f - fabsf(2.0f - lfo_scale*offset)) * depth) + delay;
offset = (offset+1)%lfo_range;
}
}
static void GetSinusoidDelays(ALint *restrict delays, ALsizei offset, const ALsizei lfo_range,
const ALfloat lfo_scale, const ALfloat depth, const ALsizei delay,
const ALsizei todo)
{
ALsizei i;
for(i = 0;i < todo;i++)
{
delays[i] = fastf2i(sinf(lfo_scale*offset) * depth) + delay;
offset = (offset+1)%lfo_range;
}
}
static ALvoid ALchorusState_process(ALchorusState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
{
const ALsizei bufmask = state->BufferLength-1;
const ALfloat feedback = state->feedback;
const ALsizei avgdelay = (state->delay + (FRACTIONONE>>1)) >> FRACTIONBITS;
ALfloat *restrict delaybuf = state->SampleBuffer;
ALsizei offset = state->offset;
ALsizei i, c;
ALsizei base;
for(base = 0;base < SamplesToDo;)
{
const ALsizei todo = mini(256, SamplesToDo-base);
ALint moddelays[2][256];
alignas(16) ALfloat temps[2][256];
if(state->waveform == WF_Sinusoid)
{
GetSinusoidDelays(moddelays[0], state->lfo_offset, state->lfo_range, state->lfo_scale,
state->depth, state->delay, todo);
GetSinusoidDelays(moddelays[1], (state->lfo_offset+state->lfo_disp)%state->lfo_range,
state->lfo_range, state->lfo_scale, state->depth, state->delay,
todo);
}
else /*if(state->waveform == WF_Triangle)*/
{
GetTriangleDelays(moddelays[0], state->lfo_offset, state->lfo_range, state->lfo_scale,
state->depth, state->delay, todo);
GetTriangleDelays(moddelays[1], (state->lfo_offset+state->lfo_disp)%state->lfo_range,
state->lfo_range, state->lfo_scale, state->depth, state->delay,
todo);
}
state->lfo_offset = (state->lfo_offset+todo) % state->lfo_range;
for(i = 0;i < todo;i++)
{
ALint delay;
ALfloat mu;
// Feed the buffer's input first (necessary for delays < 1).
delaybuf[offset&bufmask] = SamplesIn[0][base+i];
// Tap for the left output.
delay = offset - (moddelays[0][i]>>FRACTIONBITS);
mu = (moddelays[0][i]&FRACTIONMASK) * (1.0f/FRACTIONONE);
temps[0][i] = cubic(delaybuf[(delay+1) & bufmask], delaybuf[(delay ) & bufmask],
delaybuf[(delay-1) & bufmask], delaybuf[(delay-2) & bufmask],
mu);
// Tap for the right output.
delay = offset - (moddelays[1][i]>>FRACTIONBITS);
mu = (moddelays[1][i]&FRACTIONMASK) * (1.0f/FRACTIONONE);
temps[1][i] = cubic(delaybuf[(delay+1) & bufmask], delaybuf[(delay ) & bufmask],
delaybuf[(delay-1) & bufmask], delaybuf[(delay-2) & bufmask],
mu);
// Accumulate feedback from the average delay of the taps.
delaybuf[offset&bufmask] += delaybuf[(offset-avgdelay) & bufmask] * feedback;
offset++;
}
for(c = 0;c < 2;c++)
MixSamples(temps[c], NumChannels, SamplesOut, state->Gains[c].Current,
state->Gains[c].Target, SamplesToDo-base, base, todo);
base += todo;
}
state->offset = offset;
}
typedef struct ChorusStateFactory {
DERIVE_FROM_TYPE(EffectStateFactory);
} ChorusStateFactory;
static ALeffectState *ChorusStateFactory_create(ChorusStateFactory *UNUSED(factory))
{
ALchorusState *state;
NEW_OBJ0(state, ALchorusState)();
if(!state) return NULL;
return STATIC_CAST(ALeffectState, state);
}
DEFINE_EFFECTSTATEFACTORY_VTABLE(ChorusStateFactory);
EffectStateFactory *ChorusStateFactory_getFactory(void)
{
static ChorusStateFactory ChorusFactory = { { GET_VTABLE2(ChorusStateFactory, EffectStateFactory) } };
return STATIC_CAST(EffectStateFactory, &ChorusFactory);
}
void ALchorus_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
{
ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_CHORUS_WAVEFORM:
if(!(val >= AL_CHORUS_MIN_WAVEFORM && val <= AL_CHORUS_MAX_WAVEFORM))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Invalid chorus waveform");
props->Chorus.Waveform = val;
break;
case AL_CHORUS_PHASE:
if(!(val >= AL_CHORUS_MIN_PHASE && val <= AL_CHORUS_MAX_PHASE))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Chorus phase out of range");
props->Chorus.Phase = val;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid chorus integer property 0x%04x", param);
}
}
void ALchorus_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
{ ALchorus_setParami(effect, context, param, vals[0]); }
void ALchorus_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
{
ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_CHORUS_RATE:
if(!(val >= AL_CHORUS_MIN_RATE && val <= AL_CHORUS_MAX_RATE))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Chorus rate out of range");
props->Chorus.Rate = val;
break;
case AL_CHORUS_DEPTH:
if(!(val >= AL_CHORUS_MIN_DEPTH && val <= AL_CHORUS_MAX_DEPTH))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Chorus depth out of range");
props->Chorus.Depth = val;
break;
case AL_CHORUS_FEEDBACK:
if(!(val >= AL_CHORUS_MIN_FEEDBACK && val <= AL_CHORUS_MAX_FEEDBACK))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Chorus feedback out of range");
props->Chorus.Feedback = val;
break;
case AL_CHORUS_DELAY:
if(!(val >= AL_CHORUS_MIN_DELAY && val <= AL_CHORUS_MAX_DELAY))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Chorus delay out of range");
props->Chorus.Delay = val;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid chorus float property 0x%04x", param);
}
}
void ALchorus_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
{ ALchorus_setParamf(effect, context, param, vals[0]); }
void ALchorus_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
{
const ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_CHORUS_WAVEFORM:
*val = props->Chorus.Waveform;
break;
case AL_CHORUS_PHASE:
*val = props->Chorus.Phase;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid chorus integer property 0x%04x", param);
}
}
void ALchorus_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
{ ALchorus_getParami(effect, context, param, vals); }
void ALchorus_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
{
const ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_CHORUS_RATE:
*val = props->Chorus.Rate;
break;
case AL_CHORUS_DEPTH:
*val = props->Chorus.Depth;
break;
case AL_CHORUS_FEEDBACK:
*val = props->Chorus.Feedback;
break;
case AL_CHORUS_DELAY:
*val = props->Chorus.Delay;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid chorus float property 0x%04x", param);
}
}
void ALchorus_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
{ ALchorus_getParamf(effect, context, param, vals); }
DEFINE_ALEFFECT_VTABLE(ALchorus);
/* Flanger is basically a chorus with a really short delay. They can both use
* the same processing functions, so piggyback flanger on the chorus functions.
*/
typedef struct FlangerStateFactory {
DERIVE_FROM_TYPE(EffectStateFactory);
} FlangerStateFactory;
ALeffectState *FlangerStateFactory_create(FlangerStateFactory *UNUSED(factory))
{
ALchorusState *state;
NEW_OBJ0(state, ALchorusState)();
if(!state) return NULL;
return STATIC_CAST(ALeffectState, state);
}
DEFINE_EFFECTSTATEFACTORY_VTABLE(FlangerStateFactory);
EffectStateFactory *FlangerStateFactory_getFactory(void)
{
static FlangerStateFactory FlangerFactory = { { GET_VTABLE2(FlangerStateFactory, EffectStateFactory) } };
return STATIC_CAST(EffectStateFactory, &FlangerFactory);
}
void ALflanger_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
{
ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_FLANGER_WAVEFORM:
if(!(val >= AL_FLANGER_MIN_WAVEFORM && val <= AL_FLANGER_MAX_WAVEFORM))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Invalid flanger waveform");
props->Chorus.Waveform = val;
break;
case AL_FLANGER_PHASE:
if(!(val >= AL_FLANGER_MIN_PHASE && val <= AL_FLANGER_MAX_PHASE))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Flanger phase out of range");
props->Chorus.Phase = val;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid flanger integer property 0x%04x", param);
}
}
void ALflanger_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
{ ALflanger_setParami(effect, context, param, vals[0]); }
void ALflanger_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
{
ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_FLANGER_RATE:
if(!(val >= AL_FLANGER_MIN_RATE && val <= AL_FLANGER_MAX_RATE))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Flanger rate out of range");
props->Chorus.Rate = val;
break;
case AL_FLANGER_DEPTH:
if(!(val >= AL_FLANGER_MIN_DEPTH && val <= AL_FLANGER_MAX_DEPTH))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Flanger depth out of range");
props->Chorus.Depth = val;
break;
case AL_FLANGER_FEEDBACK:
if(!(val >= AL_FLANGER_MIN_FEEDBACK && val <= AL_FLANGER_MAX_FEEDBACK))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Flanger feedback out of range");
props->Chorus.Feedback = val;
break;
case AL_FLANGER_DELAY:
if(!(val >= AL_FLANGER_MIN_DELAY && val <= AL_FLANGER_MAX_DELAY))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Flanger delay out of range");
props->Chorus.Delay = val;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid flanger float property 0x%04x", param);
}
}
void ALflanger_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
{ ALflanger_setParamf(effect, context, param, vals[0]); }
void ALflanger_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
{
const ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_FLANGER_WAVEFORM:
*val = props->Chorus.Waveform;
break;
case AL_FLANGER_PHASE:
*val = props->Chorus.Phase;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid flanger integer property 0x%04x", param);
}
}
void ALflanger_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
{ ALflanger_getParami(effect, context, param, vals); }
void ALflanger_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
{
const ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_FLANGER_RATE:
*val = props->Chorus.Rate;
break;
case AL_FLANGER_DEPTH:
*val = props->Chorus.Depth;
break;
case AL_FLANGER_FEEDBACK:
*val = props->Chorus.Feedback;
break;
case AL_FLANGER_DELAY:
*val = props->Chorus.Delay;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid flanger float property 0x%04x", param);
}
}
void ALflanger_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
{ ALflanger_getParamf(effect, context, param, vals); }
DEFINE_ALEFFECT_VTABLE(ALflanger);
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/**
* OpenAL cross platform audio library
* Copyright (C) 2013 by Anis A. Hireche
* 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include <stdlib.h>
#include "config.h"
#include "alError.h"
#include "alMain.h"
#include "alAuxEffectSlot.h"
#include "alu.h"
#define AMP_ENVELOPE_MIN 0.5f
#define AMP_ENVELOPE_MAX 2.0f
#define ATTACK_TIME 0.1f /* 100ms to rise from min to max */
#define RELEASE_TIME 0.2f /* 200ms to drop from max to min */
typedef struct ALcompressorState {
DERIVE_FROM_TYPE(ALeffectState);
/* Effect gains for each channel */
ALfloat Gain[MAX_EFFECT_CHANNELS][MAX_OUTPUT_CHANNELS];
/* Effect parameters */
ALboolean Enabled;
ALfloat AttackMult;
ALfloat ReleaseMult;
ALfloat EnvFollower;
} ALcompressorState;
static ALvoid ALcompressorState_Destruct(ALcompressorState *state);
static ALboolean ALcompressorState_deviceUpdate(ALcompressorState *state, ALCdevice *device);
static ALvoid ALcompressorState_update(ALcompressorState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props);
static ALvoid ALcompressorState_process(ALcompressorState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
DECLARE_DEFAULT_ALLOCATORS(ALcompressorState)
DEFINE_ALEFFECTSTATE_VTABLE(ALcompressorState);
static void ALcompressorState_Construct(ALcompressorState *state)
{
ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
SET_VTABLE2(ALcompressorState, ALeffectState, state);
state->Enabled = AL_TRUE;
state->AttackMult = 1.0f;
state->ReleaseMult = 1.0f;
state->EnvFollower = 1.0f;
}
static ALvoid ALcompressorState_Destruct(ALcompressorState *state)
{
ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
}
static ALboolean ALcompressorState_deviceUpdate(ALcompressorState *state, ALCdevice *device)
{
/* Number of samples to do a full attack and release (non-integer sample
* counts are okay).
*/
const ALfloat attackCount = (ALfloat)device->Frequency * ATTACK_TIME;
const ALfloat releaseCount = (ALfloat)device->Frequency * RELEASE_TIME;
/* Calculate per-sample multipliers to attack and release at the desired
* rates.
*/
state->AttackMult = powf(AMP_ENVELOPE_MAX/AMP_ENVELOPE_MIN, 1.0f/attackCount);
state->ReleaseMult = powf(AMP_ENVELOPE_MIN/AMP_ENVELOPE_MAX, 1.0f/releaseCount);
return AL_TRUE;
}
static ALvoid ALcompressorState_update(ALcompressorState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props)
{
const ALCdevice *device = context->Device;
ALuint i;
state->Enabled = props->Compressor.OnOff;
STATIC_CAST(ALeffectState,state)->OutBuffer = device->FOAOut.Buffer;
STATIC_CAST(ALeffectState,state)->OutChannels = device->FOAOut.NumChannels;
for(i = 0;i < 4;i++)
ComputePanGains(&device->FOAOut, IdentityMatrixf.m[i], slot->Params.Gain, state->Gain[i]);
}
static ALvoid ALcompressorState_process(ALcompressorState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
{
ALsizei i, j, k;
ALsizei base;
for(base = 0;base < SamplesToDo;)
{
ALfloat gains[256];
ALsizei td = mini(256, SamplesToDo-base);
ALfloat env = state->EnvFollower;
/* Generate the per-sample gains from the signal envelope. */
if(state->Enabled)
{
for(i = 0;i < td;++i)
{
/* Clamp the absolute amplitude to the defined envelope limits,
* then attack or release the envelope to reach it.
*/
ALfloat amplitude = clampf(fabsf(SamplesIn[0][base+i]),
AMP_ENVELOPE_MIN, AMP_ENVELOPE_MAX);
if(amplitude > env)
env = minf(env*state->AttackMult, amplitude);
else if(amplitude < env)
env = maxf(env*state->ReleaseMult, amplitude);
/* Apply the reciprocal of the envelope to normalize the volume
* (compress the dynamic range).
*/
gains[i] = 1.0f / env;
}
}
else
{
/* Same as above, except the amplitude is forced to 1. This helps
* ensure smooth gain changes when the compressor is turned on and
* off.
*/
for(i = 0;i < td;++i)
{
ALfloat amplitude = 1.0f;
if(amplitude > env)
env = minf(env*state->AttackMult, amplitude);
else if(amplitude < env)
env = maxf(env*state->ReleaseMult, amplitude);
gains[i] = 1.0f / env;
}
}
state->EnvFollower = env;
/* Now compress the signal amplitude to output. */
for(j = 0;j < MAX_EFFECT_CHANNELS;j++)
{
for(k = 0;k < NumChannels;k++)
{
ALfloat gain = state->Gain[j][k];
if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
continue;
for(i = 0;i < td;i++)
SamplesOut[k][base+i] += SamplesIn[j][base+i] * gains[i] * gain;
}
}
base += td;
}
}
typedef struct CompressorStateFactory {
DERIVE_FROM_TYPE(EffectStateFactory);
} CompressorStateFactory;
static ALeffectState *CompressorStateFactory_create(CompressorStateFactory *UNUSED(factory))
{
ALcompressorState *state;
NEW_OBJ0(state, ALcompressorState)();
if(!state) return NULL;
return STATIC_CAST(ALeffectState, state);
}
DEFINE_EFFECTSTATEFACTORY_VTABLE(CompressorStateFactory);
EffectStateFactory *CompressorStateFactory_getFactory(void)
{
static CompressorStateFactory CompressorFactory = { { GET_VTABLE2(CompressorStateFactory, EffectStateFactory) } };
return STATIC_CAST(EffectStateFactory, &CompressorFactory);
}
void ALcompressor_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
{
ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_COMPRESSOR_ONOFF:
if(!(val >= AL_COMPRESSOR_MIN_ONOFF && val <= AL_COMPRESSOR_MAX_ONOFF))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Compressor state out of range");
props->Compressor.OnOff = val;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid compressor integer property 0x%04x",
param);
}
}
void ALcompressor_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
{ ALcompressor_setParami(effect, context, param, vals[0]); }
void ALcompressor_setParamf(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat UNUSED(val))
{ alSetError(context, AL_INVALID_ENUM, "Invalid compressor float property 0x%04x", param); }
void ALcompressor_setParamfv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALfloat *UNUSED(vals))
{ alSetError(context, AL_INVALID_ENUM, "Invalid compressor float-vector property 0x%04x", param); }
void ALcompressor_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
{
const ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_COMPRESSOR_ONOFF:
*val = props->Compressor.OnOff;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid compressor integer property 0x%04x",
param);
}
}
void ALcompressor_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
{ ALcompressor_getParami(effect, context, param, vals); }
void ALcompressor_getParamf(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat *UNUSED(val))
{ alSetError(context, AL_INVALID_ENUM, "Invalid compressor float property 0x%04x", param); }
void ALcompressor_getParamfv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat *UNUSED(vals))
{ alSetError(context, AL_INVALID_ENUM, "Invalid compressor float-vector property 0x%04x", param); }
DEFINE_ALEFFECT_VTABLE(ALcompressor);
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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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdlib.h>
#include "alMain.h"
#include "alAuxEffectSlot.h"
#include "alError.h"
#include "alu.h"
#include "filters/defs.h"
typedef struct ALdedicatedState {
DERIVE_FROM_TYPE(ALeffectState);
ALfloat CurrentGains[MAX_OUTPUT_CHANNELS];
ALfloat TargetGains[MAX_OUTPUT_CHANNELS];
} ALdedicatedState;
static ALvoid ALdedicatedState_Destruct(ALdedicatedState *state);
static ALboolean ALdedicatedState_deviceUpdate(ALdedicatedState *state, ALCdevice *device);
static ALvoid ALdedicatedState_update(ALdedicatedState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props);
static ALvoid ALdedicatedState_process(ALdedicatedState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
DECLARE_DEFAULT_ALLOCATORS(ALdedicatedState)
DEFINE_ALEFFECTSTATE_VTABLE(ALdedicatedState);
static void ALdedicatedState_Construct(ALdedicatedState *state)
{
ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
SET_VTABLE2(ALdedicatedState, ALeffectState, state);
}
static ALvoid ALdedicatedState_Destruct(ALdedicatedState *state)
{
ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
}
static ALboolean ALdedicatedState_deviceUpdate(ALdedicatedState *state, ALCdevice *UNUSED(device))
{
ALsizei i;
for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
state->CurrentGains[i] = 0.0f;
return AL_TRUE;
}
static ALvoid ALdedicatedState_update(ALdedicatedState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props)
{
const ALCdevice *device = context->Device;
ALfloat Gain;
ALsizei i;
for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
state->TargetGains[i] = 0.0f;
Gain = slot->Params.Gain * props->Dedicated.Gain;
if(slot->Params.EffectType == AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT)
{
int idx;
if((idx=GetChannelIdxByName(&device->RealOut, LFE)) != -1)
{
STATIC_CAST(ALeffectState,state)->OutBuffer = device->RealOut.Buffer;
STATIC_CAST(ALeffectState,state)->OutChannels = device->RealOut.NumChannels;
state->TargetGains[idx] = Gain;
}
}
else if(slot->Params.EffectType == AL_EFFECT_DEDICATED_DIALOGUE)
{
int idx;
/* Dialog goes to the front-center speaker if it exists, otherwise it
* plays from the front-center location. */
if((idx=GetChannelIdxByName(&device->RealOut, FrontCenter)) != -1)
{
STATIC_CAST(ALeffectState,state)->OutBuffer = device->RealOut.Buffer;
STATIC_CAST(ALeffectState,state)->OutChannels = device->RealOut.NumChannels;
state->TargetGains[idx] = Gain;
}
else
{
ALfloat coeffs[MAX_AMBI_COEFFS];
CalcAngleCoeffs(0.0f, 0.0f, 0.0f, coeffs);
STATIC_CAST(ALeffectState,state)->OutBuffer = device->Dry.Buffer;
STATIC_CAST(ALeffectState,state)->OutChannels = device->Dry.NumChannels;
ComputePanGains(&device->Dry, coeffs, Gain, state->TargetGains);
}
}
}
static ALvoid ALdedicatedState_process(ALdedicatedState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
{
MixSamples(SamplesIn[0], NumChannels, SamplesOut, state->CurrentGains,
state->TargetGains, SamplesToDo, 0, SamplesToDo);
}
typedef struct DedicatedStateFactory {
DERIVE_FROM_TYPE(EffectStateFactory);
} DedicatedStateFactory;
ALeffectState *DedicatedStateFactory_create(DedicatedStateFactory *UNUSED(factory))
{
ALdedicatedState *state;
NEW_OBJ0(state, ALdedicatedState)();
if(!state) return NULL;
return STATIC_CAST(ALeffectState, state);
}
DEFINE_EFFECTSTATEFACTORY_VTABLE(DedicatedStateFactory);
EffectStateFactory *DedicatedStateFactory_getFactory(void)
{
static DedicatedStateFactory DedicatedFactory = { { GET_VTABLE2(DedicatedStateFactory, EffectStateFactory) } };
return STATIC_CAST(EffectStateFactory, &DedicatedFactory);
}
void ALdedicated_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint UNUSED(val))
{ alSetError(context, AL_INVALID_ENUM, "Invalid dedicated integer property 0x%04x", param); }
void ALdedicated_setParamiv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALint *UNUSED(vals))
{ alSetError(context, AL_INVALID_ENUM, "Invalid dedicated integer-vector property 0x%04x", param); }
void ALdedicated_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
{
ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_DEDICATED_GAIN:
if(!(val >= 0.0f && isfinite(val)))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Dedicated gain out of range");
props->Dedicated.Gain = val;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid dedicated float property 0x%04x", param);
}
}
void ALdedicated_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
{ ALdedicated_setParamf(effect, context, param, vals[0]); }
void ALdedicated_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(val))
{ alSetError(context, AL_INVALID_ENUM, "Invalid dedicated integer property 0x%04x", param); }
void ALdedicated_getParamiv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(vals))
{ alSetError(context, AL_INVALID_ENUM, "Invalid dedicated integer-vector property 0x%04x", param); }
void ALdedicated_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
{
const ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_DEDICATED_GAIN:
*val = props->Dedicated.Gain;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid dedicated float property 0x%04x", param);
}
}
void ALdedicated_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
{ ALdedicated_getParamf(effect, context, param, vals); }
DEFINE_ALEFFECT_VTABLE(ALdedicated);
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/**
* OpenAL cross platform audio library
* Copyright (C) 2013 by Mike Gorchak
* 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <math.h>
#include <stdlib.h>
#include "alMain.h"
#include "alAuxEffectSlot.h"
#include "alError.h"
#include "alu.h"
#include "filters/defs.h"
typedef struct ALdistortionState {
DERIVE_FROM_TYPE(ALeffectState);
/* Effect gains for each channel */
ALfloat Gain[MAX_OUTPUT_CHANNELS];
/* Effect parameters */
BiquadFilter lowpass;
BiquadFilter bandpass;
ALfloat attenuation;
ALfloat edge_coeff;
ALfloat Buffer[2][BUFFERSIZE];
} ALdistortionState;
static ALvoid ALdistortionState_Destruct(ALdistortionState *state);
static ALboolean ALdistortionState_deviceUpdate(ALdistortionState *state, ALCdevice *device);
static ALvoid ALdistortionState_update(ALdistortionState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props);
static ALvoid ALdistortionState_process(ALdistortionState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
DECLARE_DEFAULT_ALLOCATORS(ALdistortionState)
DEFINE_ALEFFECTSTATE_VTABLE(ALdistortionState);
static void ALdistortionState_Construct(ALdistortionState *state)
{
ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
SET_VTABLE2(ALdistortionState, ALeffectState, state);
}
static ALvoid ALdistortionState_Destruct(ALdistortionState *state)
{
ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
}
static ALboolean ALdistortionState_deviceUpdate(ALdistortionState *state, ALCdevice *UNUSED(device))
{
BiquadFilter_clear(&state->lowpass);
BiquadFilter_clear(&state->bandpass);
return AL_TRUE;
}
static ALvoid ALdistortionState_update(ALdistortionState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props)
{
const ALCdevice *device = context->Device;
ALfloat frequency = (ALfloat)device->Frequency;
ALfloat coeffs[MAX_AMBI_COEFFS];
ALfloat bandwidth;
ALfloat cutoff;
ALfloat edge;
/* Store waveshaper edge settings. */
edge = sinf(props->Distortion.Edge * (F_PI_2));
edge = minf(edge, 0.99f);
state->edge_coeff = 2.0f * edge / (1.0f-edge);
cutoff = props->Distortion.LowpassCutoff;
/* Bandwidth value is constant in octaves. */
bandwidth = (cutoff / 2.0f) / (cutoff * 0.67f);
/* Multiply sampling frequency by the amount of oversampling done during
* processing.
*/
BiquadFilter_setParams(&state->lowpass, BiquadType_LowPass, 1.0f,
cutoff / (frequency*4.0f), calc_rcpQ_from_bandwidth(cutoff / (frequency*4.0f), bandwidth)
);
cutoff = props->Distortion.EQCenter;
/* Convert bandwidth in Hz to octaves. */
bandwidth = props->Distortion.EQBandwidth / (cutoff * 0.67f);
BiquadFilter_setParams(&state->bandpass, BiquadType_BandPass, 1.0f,
cutoff / (frequency*4.0f), calc_rcpQ_from_bandwidth(cutoff / (frequency*4.0f), bandwidth)
);
CalcAngleCoeffs(0.0f, 0.0f, 0.0f, coeffs);
ComputePanGains(&device->Dry, coeffs, slot->Params.Gain*props->Distortion.Gain, state->Gain);
}
static ALvoid ALdistortionState_process(ALdistortionState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
{
ALfloat (*restrict buffer)[BUFFERSIZE] = state->Buffer;
const ALfloat fc = state->edge_coeff;
ALsizei base;
ALsizei i, k;
for(base = 0;base < SamplesToDo;)
{
/* Perform 4x oversampling to avoid aliasing. Oversampling greatly
* improves distortion quality and allows to implement lowpass and
* bandpass filters using high frequencies, at which classic IIR
* filters became unstable.
*/
ALsizei todo = mini(BUFFERSIZE, (SamplesToDo-base) * 4);
/* Fill oversample buffer using zero stuffing. Multiply the sample by
* the amount of oversampling to maintain the signal's power.
*/
for(i = 0;i < todo;i++)
buffer[0][i] = !(i&3) ? SamplesIn[0][(i>>2)+base] * 4.0f : 0.0f;
/* First step, do lowpass filtering of original signal. Additionally
* perform buffer interpolation and lowpass cutoff for oversampling
* (which is fortunately first step of distortion). So combine three
* operations into the one.
*/
BiquadFilter_process(&state->lowpass, buffer[1], buffer[0], todo);
/* Second step, do distortion using waveshaper function to emulate
* signal processing during tube overdriving. Three steps of
* waveshaping are intended to modify waveform without boost/clipping/
* attenuation process.
*/
for(i = 0;i < todo;i++)
{
ALfloat smp = buffer[1][i];
smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp));
smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp)) * -1.0f;
smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp));
buffer[0][i] = smp;
}
/* Third step, do bandpass filtering of distorted signal. */
BiquadFilter_process(&state->bandpass, buffer[1], buffer[0], todo);
todo >>= 2;
for(k = 0;k < NumChannels;k++)
{
/* Fourth step, final, do attenuation and perform decimation,
* storing only one sample out of four.
*/
ALfloat gain = state->Gain[k];
if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
continue;
for(i = 0;i < todo;i++)
SamplesOut[k][base+i] += gain * buffer[1][i*4];
}
base += todo;
}
}
typedef struct DistortionStateFactory {
DERIVE_FROM_TYPE(EffectStateFactory);
} DistortionStateFactory;
static ALeffectState *DistortionStateFactory_create(DistortionStateFactory *UNUSED(factory))
{
ALdistortionState *state;
NEW_OBJ0(state, ALdistortionState)();
if(!state) return NULL;
return STATIC_CAST(ALeffectState, state);
}
DEFINE_EFFECTSTATEFACTORY_VTABLE(DistortionStateFactory);
EffectStateFactory *DistortionStateFactory_getFactory(void)
{
static DistortionStateFactory DistortionFactory = { { GET_VTABLE2(DistortionStateFactory, EffectStateFactory) } };
return STATIC_CAST(EffectStateFactory, &DistortionFactory);
}
void ALdistortion_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint UNUSED(val))
{ alSetError(context, AL_INVALID_ENUM, "Invalid distortion integer property 0x%04x", param); }
void ALdistortion_setParamiv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALint *UNUSED(vals))
{ alSetError(context, AL_INVALID_ENUM, "Invalid distortion integer-vector property 0x%04x", param); }
void ALdistortion_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
{
ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_DISTORTION_EDGE:
if(!(val >= AL_DISTORTION_MIN_EDGE && val <= AL_DISTORTION_MAX_EDGE))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Distortion edge out of range");
props->Distortion.Edge = val;
break;
case AL_DISTORTION_GAIN:
if(!(val >= AL_DISTORTION_MIN_GAIN && val <= AL_DISTORTION_MAX_GAIN))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Distortion gain out of range");
props->Distortion.Gain = val;
break;
case AL_DISTORTION_LOWPASS_CUTOFF:
if(!(val >= AL_DISTORTION_MIN_LOWPASS_CUTOFF && val <= AL_DISTORTION_MAX_LOWPASS_CUTOFF))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Distortion low-pass cutoff out of range");
props->Distortion.LowpassCutoff = val;
break;
case AL_DISTORTION_EQCENTER:
if(!(val >= AL_DISTORTION_MIN_EQCENTER && val <= AL_DISTORTION_MAX_EQCENTER))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Distortion EQ center out of range");
props->Distortion.EQCenter = val;
break;
case AL_DISTORTION_EQBANDWIDTH:
if(!(val >= AL_DISTORTION_MIN_EQBANDWIDTH && val <= AL_DISTORTION_MAX_EQBANDWIDTH))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Distortion EQ bandwidth out of range");
props->Distortion.EQBandwidth = val;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid distortion float property 0x%04x",
param);
}
}
void ALdistortion_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
{ ALdistortion_setParamf(effect, context, param, vals[0]); }
void ALdistortion_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(val))
{ alSetError(context, AL_INVALID_ENUM, "Invalid distortion integer property 0x%04x", param); }
void ALdistortion_getParamiv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(vals))
{ alSetError(context, AL_INVALID_ENUM, "Invalid distortion integer-vector property 0x%04x", param); }
void ALdistortion_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
{
const ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_DISTORTION_EDGE:
*val = props->Distortion.Edge;
break;
case AL_DISTORTION_GAIN:
*val = props->Distortion.Gain;
break;
case AL_DISTORTION_LOWPASS_CUTOFF:
*val = props->Distortion.LowpassCutoff;
break;
case AL_DISTORTION_EQCENTER:
*val = props->Distortion.EQCenter;
break;
case AL_DISTORTION_EQBANDWIDTH:
*val = props->Distortion.EQBandwidth;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid distortion float property 0x%04x",
param);
}
}
void ALdistortion_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
{ ALdistortion_getParamf(effect, context, param, vals); }
DEFINE_ALEFFECT_VTABLE(ALdistortion);
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/**
* 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 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"
#include "filters/defs.h"
typedef struct ALechoState {
DERIVE_FROM_TYPE(ALeffectState);
ALfloat *SampleBuffer;
ALsizei BufferLength;
// The echo is two tap. The delay is the number of samples from before the
// current offset
struct {
ALsizei delay;
} Tap[2];
ALsizei Offset;
/* The panning gains for the two taps */
struct {
ALfloat Current[MAX_OUTPUT_CHANNELS];
ALfloat Target[MAX_OUTPUT_CHANNELS];
} Gains[2];
ALfloat FeedGain;
BiquadFilter Filter;
} ALechoState;
static ALvoid ALechoState_Destruct(ALechoState *state);
static ALboolean ALechoState_deviceUpdate(ALechoState *state, ALCdevice *Device);
static ALvoid ALechoState_update(ALechoState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props);
static ALvoid ALechoState_process(ALechoState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
DECLARE_DEFAULT_ALLOCATORS(ALechoState)
DEFINE_ALEFFECTSTATE_VTABLE(ALechoState);
static void ALechoState_Construct(ALechoState *state)
{
ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
SET_VTABLE2(ALechoState, ALeffectState, state);
state->BufferLength = 0;
state->SampleBuffer = NULL;
state->Tap[0].delay = 0;
state->Tap[1].delay = 0;
state->Offset = 0;
BiquadFilter_clear(&state->Filter);
}
static ALvoid ALechoState_Destruct(ALechoState *state)
{
al_free(state->SampleBuffer);
state->SampleBuffer = NULL;
ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
}
static ALboolean ALechoState_deviceUpdate(ALechoState *state, ALCdevice *Device)
{
ALsizei maxlen;
// 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 = float2int(AL_ECHO_MAX_DELAY*Device->Frequency + 0.5f) +
float2int(AL_ECHO_MAX_LRDELAY*Device->Frequency + 0.5f);
maxlen = NextPowerOf2(maxlen);
if(maxlen <= 0) return AL_FALSE;
if(maxlen != state->BufferLength)
{
void *temp = al_calloc(16, maxlen * sizeof(ALfloat));
if(!temp) return AL_FALSE;
al_free(state->SampleBuffer);
state->SampleBuffer = temp;
state->BufferLength = maxlen;
}
memset(state->SampleBuffer, 0, state->BufferLength*sizeof(ALfloat));
memset(state->Gains, 0, sizeof(state->Gains));
return AL_TRUE;
}
static ALvoid ALechoState_update(ALechoState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props)
{
const ALCdevice *device = context->Device;
ALuint frequency = device->Frequency;
ALfloat coeffs[MAX_AMBI_COEFFS];
ALfloat gainhf, lrpan, spread;
state->Tap[0].delay = maxi(float2int(props->Echo.Delay*frequency + 0.5f), 1);
state->Tap[1].delay = float2int(props->Echo.LRDelay*frequency + 0.5f);
state->Tap[1].delay += state->Tap[0].delay;
spread = props->Echo.Spread;
if(spread < 0.0f) lrpan = -1.0f;
else lrpan = 1.0f;
/* Convert echo spread (where 0 = omni, +/-1 = directional) to coverage
* spread (where 0 = point, tau = omni).
*/
spread = asinf(1.0f - fabsf(spread))*4.0f;
state->FeedGain = props->Echo.Feedback;
gainhf = maxf(1.0f - props->Echo.Damping, 0.0625f); /* Limit -24dB */
BiquadFilter_setParams(&state->Filter, BiquadType_HighShelf,
gainhf, LOWPASSFREQREF/frequency, calc_rcpQ_from_slope(gainhf, 1.0f)
);
/* First tap panning */
CalcAngleCoeffs(-F_PI_2*lrpan, 0.0f, spread, coeffs);
ComputePanGains(&device->Dry, coeffs, slot->Params.Gain, state->Gains[0].Target);
/* Second tap panning */
CalcAngleCoeffs( F_PI_2*lrpan, 0.0f, spread, coeffs);
ComputePanGains(&device->Dry, coeffs, slot->Params.Gain, state->Gains[1].Target);
}
static ALvoid ALechoState_process(ALechoState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
{
const ALsizei mask = state->BufferLength-1;
const ALsizei tap1 = state->Tap[0].delay;
const ALsizei tap2 = state->Tap[1].delay;
ALfloat *restrict delaybuf = state->SampleBuffer;
ALsizei offset = state->Offset;
ALfloat z1, z2, in, out;
ALsizei base;
ALsizei c, i;
z1 = state->Filter.z1;
z2 = state->Filter.z2;
for(base = 0;base < SamplesToDo;)
{
alignas(16) ALfloat temps[2][128];
ALsizei td = mini(128, SamplesToDo-base);
for(i = 0;i < td;i++)
{
/* Feed the delay buffer's input first. */
delaybuf[offset&mask] = SamplesIn[0][i+base];
/* First tap */
temps[0][i] = delaybuf[(offset-tap1) & mask];
/* Second tap */
temps[1][i] = delaybuf[(offset-tap2) & mask];
/* Apply damping to the second tap, then add it to the buffer with
* feedback attenuation.
*/
in = temps[1][i];
out = in*state->Filter.b0 + z1;
z1 = in*state->Filter.b1 - out*state->Filter.a1 + z2;
z2 = in*state->Filter.b2 - out*state->Filter.a2;
delaybuf[offset&mask] += out * state->FeedGain;
offset++;
}
for(c = 0;c < 2;c++)
MixSamples(temps[c], NumChannels, SamplesOut, state->Gains[c].Current,
state->Gains[c].Target, SamplesToDo-base, base, td);
base += td;
}
state->Filter.z1 = z1;
state->Filter.z2 = z2;
state->Offset = offset;
}
typedef struct EchoStateFactory {
DERIVE_FROM_TYPE(EffectStateFactory);
} EchoStateFactory;
ALeffectState *EchoStateFactory_create(EchoStateFactory *UNUSED(factory))
{
ALechoState *state;
NEW_OBJ0(state, ALechoState)();
if(!state) return NULL;
return STATIC_CAST(ALeffectState, state);
}
DEFINE_EFFECTSTATEFACTORY_VTABLE(EchoStateFactory);
EffectStateFactory *EchoStateFactory_getFactory(void)
{
static EchoStateFactory EchoFactory = { { GET_VTABLE2(EchoStateFactory, EffectStateFactory) } };
return STATIC_CAST(EffectStateFactory, &EchoFactory);
}
void ALecho_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint UNUSED(val))
{ alSetError(context, AL_INVALID_ENUM, "Invalid echo integer property 0x%04x", param); }
void ALecho_setParamiv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALint *UNUSED(vals))
{ alSetError(context, AL_INVALID_ENUM, "Invalid echo integer-vector property 0x%04x", param); }
void ALecho_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
{
ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_ECHO_DELAY:
if(!(val >= AL_ECHO_MIN_DELAY && val <= AL_ECHO_MAX_DELAY))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Echo delay out of range");
props->Echo.Delay = val;
break;
case AL_ECHO_LRDELAY:
if(!(val >= AL_ECHO_MIN_LRDELAY && val <= AL_ECHO_MAX_LRDELAY))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Echo LR delay out of range");
props->Echo.LRDelay = val;
break;
case AL_ECHO_DAMPING:
if(!(val >= AL_ECHO_MIN_DAMPING && val <= AL_ECHO_MAX_DAMPING))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Echo damping out of range");
props->Echo.Damping = val;
break;
case AL_ECHO_FEEDBACK:
if(!(val >= AL_ECHO_MIN_FEEDBACK && val <= AL_ECHO_MAX_FEEDBACK))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Echo feedback out of range");
props->Echo.Feedback = val;
break;
case AL_ECHO_SPREAD:
if(!(val >= AL_ECHO_MIN_SPREAD && val <= AL_ECHO_MAX_SPREAD))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Echo spread out of range");
props->Echo.Spread = val;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid echo float property 0x%04x", param);
}
}
void ALecho_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
{ ALecho_setParamf(effect, context, param, vals[0]); }
void ALecho_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(val))
{ alSetError(context, AL_INVALID_ENUM, "Invalid echo integer property 0x%04x", param); }
void ALecho_getParamiv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(vals))
{ alSetError(context, AL_INVALID_ENUM, "Invalid echo integer-vector property 0x%04x", param); }
void ALecho_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
{
const ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_ECHO_DELAY:
*val = props->Echo.Delay;
break;
case AL_ECHO_LRDELAY:
*val = props->Echo.LRDelay;
break;
case AL_ECHO_DAMPING:
*val = props->Echo.Damping;
break;
case AL_ECHO_FEEDBACK:
*val = props->Echo.Feedback;
break;
case AL_ECHO_SPREAD:
*val = props->Echo.Spread;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid echo float property 0x%04x", param);
}
}
void ALecho_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
{ ALecho_getParamf(effect, context, param, vals); }
DEFINE_ALEFFECT_VTABLE(ALecho);
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@@ -1,329 +0,0 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 2018 by Raul Herraiz.
* 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <math.h>
#include <stdlib.h>
#include "alMain.h"
#include "alAuxEffectSlot.h"
#include "alError.h"
#include "alu.h"
#include "filters/defs.h"
#include "alcomplex.h"
#define HIL_SIZE 1024
#define OVERSAMP (1<<2)
#define HIL_STEP (HIL_SIZE / OVERSAMP)
#define FIFO_LATENCY (HIL_STEP * (OVERSAMP-1))
typedef struct ALfshifterState {
DERIVE_FROM_TYPE(ALeffectState);
/* Effect parameters */
ALsizei count;
ALsizei PhaseStep;
ALsizei Phase;
ALdouble ld_sign;
/*Effects buffers*/
ALfloat InFIFO[HIL_SIZE];
ALcomplex OutFIFO[HIL_SIZE];
ALcomplex OutputAccum[HIL_SIZE];
ALcomplex Analytic[HIL_SIZE];
ALcomplex Outdata[BUFFERSIZE];
alignas(16) ALfloat BufferOut[BUFFERSIZE];
/* Effect gains for each output channel */
ALfloat CurrentGains[MAX_OUTPUT_CHANNELS];
ALfloat TargetGains[MAX_OUTPUT_CHANNELS];
} ALfshifterState;
static ALvoid ALfshifterState_Destruct(ALfshifterState *state);
static ALboolean ALfshifterState_deviceUpdate(ALfshifterState *state, ALCdevice *device);
static ALvoid ALfshifterState_update(ALfshifterState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props);
static ALvoid ALfshifterState_process(ALfshifterState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
DECLARE_DEFAULT_ALLOCATORS(ALfshifterState)
DEFINE_ALEFFECTSTATE_VTABLE(ALfshifterState);
/* Define a Hann window, used to filter the HIL input and output. */
alignas(16) static ALdouble HannWindow[HIL_SIZE];
static void InitHannWindow(void)
{
ALsizei i;
/* Create lookup table of the Hann window for the desired size, i.e. HIL_SIZE */
for(i = 0;i < HIL_SIZE>>1;i++)
{
ALdouble val = sin(M_PI * (ALdouble)i / (ALdouble)(HIL_SIZE-1));
HannWindow[i] = HannWindow[HIL_SIZE-1-i] = val * val;
}
}
static alonce_flag HannInitOnce = AL_ONCE_FLAG_INIT;
static void ALfshifterState_Construct(ALfshifterState *state)
{
ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
SET_VTABLE2(ALfshifterState, ALeffectState, state);
alcall_once(&HannInitOnce, InitHannWindow);
}
static ALvoid ALfshifterState_Destruct(ALfshifterState *state)
{
ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
}
static ALboolean ALfshifterState_deviceUpdate(ALfshifterState *state, ALCdevice *UNUSED(device))
{
/* (Re-)initializing parameters and clear the buffers. */
state->count = FIFO_LATENCY;
state->PhaseStep = 0;
state->Phase = 0;
state->ld_sign = 1.0;
memset(state->InFIFO, 0, sizeof(state->InFIFO));
memset(state->OutFIFO, 0, sizeof(state->OutFIFO));
memset(state->OutputAccum, 0, sizeof(state->OutputAccum));
memset(state->Analytic, 0, sizeof(state->Analytic));
memset(state->CurrentGains, 0, sizeof(state->CurrentGains));
memset(state->TargetGains, 0, sizeof(state->TargetGains));
return AL_TRUE;
}
static ALvoid ALfshifterState_update(ALfshifterState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props)
{
const ALCdevice *device = context->Device;
ALfloat coeffs[MAX_AMBI_COEFFS];
ALfloat step;
step = props->Fshifter.Frequency / (ALfloat)device->Frequency;
state->PhaseStep = fastf2i(minf(step, 0.5f) * FRACTIONONE);
switch(props->Fshifter.LeftDirection)
{
case AL_FREQUENCY_SHIFTER_DIRECTION_DOWN:
state->ld_sign = -1.0;
break;
case AL_FREQUENCY_SHIFTER_DIRECTION_UP:
state->ld_sign = 1.0;
break;
case AL_FREQUENCY_SHIFTER_DIRECTION_OFF:
state->Phase = 0;
state->PhaseStep = 0;
break;
}
CalcAngleCoeffs(0.0f, 0.0f, 0.0f, coeffs);
ComputePanGains(&device->Dry, coeffs, slot->Params.Gain, state->TargetGains);
}
static ALvoid ALfshifterState_process(ALfshifterState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
{
static const ALcomplex complex_zero = { 0.0, 0.0 };
ALfloat *restrict BufferOut = state->BufferOut;
ALsizei j, k, base;
for(base = 0;base < SamplesToDo;)
{
ALsizei todo = mini(HIL_SIZE-state->count, SamplesToDo-base);
ASSUME(todo > 0);
/* Fill FIFO buffer with samples data */
k = state->count;
for(j = 0;j < todo;j++,k++)
{
state->InFIFO[k] = SamplesIn[0][base+j];
state->Outdata[base+j] = state->OutFIFO[k-FIFO_LATENCY];
}
state->count += todo;
base += todo;
/* Check whether FIFO buffer is filled */
if(state->count < HIL_SIZE) continue;
state->count = FIFO_LATENCY;
/* Real signal windowing and store in Analytic buffer */
for(k = 0;k < HIL_SIZE;k++)
{
state->Analytic[k].Real = state->InFIFO[k] * HannWindow[k];
state->Analytic[k].Imag = 0.0;
}
/* Processing signal by Discrete Hilbert Transform (analytical signal). */
complex_hilbert(state->Analytic, HIL_SIZE);
/* Windowing and add to output accumulator */
for(k = 0;k < HIL_SIZE;k++)
{
state->OutputAccum[k].Real += 2.0/OVERSAMP*HannWindow[k]*state->Analytic[k].Real;
state->OutputAccum[k].Imag += 2.0/OVERSAMP*HannWindow[k]*state->Analytic[k].Imag;
}
/* Shift accumulator, input & output FIFO */
for(k = 0;k < HIL_STEP;k++) state->OutFIFO[k] = state->OutputAccum[k];
for(j = 0;k < HIL_SIZE;k++,j++) state->OutputAccum[j] = state->OutputAccum[k];
for(;j < HIL_SIZE;j++) state->OutputAccum[j] = complex_zero;
for(k = 0;k < FIFO_LATENCY;k++)
state->InFIFO[k] = state->InFIFO[k+HIL_STEP];
}
/* Process frequency shifter using the analytic signal obtained. */
for(k = 0;k < SamplesToDo;k++)
{
ALdouble phase = state->Phase * ((1.0/FRACTIONONE) * 2.0*M_PI);
BufferOut[k] = (ALfloat)(state->Outdata[k].Real*cos(phase) +
state->Outdata[k].Imag*sin(phase)*state->ld_sign);
state->Phase += state->PhaseStep;
state->Phase &= FRACTIONMASK;
}
/* Now, mix the processed sound data to the output. */
MixSamples(BufferOut, NumChannels, SamplesOut, state->CurrentGains, state->TargetGains,
maxi(SamplesToDo, 512), 0, SamplesToDo);
}
typedef struct FshifterStateFactory {
DERIVE_FROM_TYPE(EffectStateFactory);
} FshifterStateFactory;
static ALeffectState *FshifterStateFactory_create(FshifterStateFactory *UNUSED(factory))
{
ALfshifterState *state;
NEW_OBJ0(state, ALfshifterState)();
if(!state) return NULL;
return STATIC_CAST(ALeffectState, state);
}
DEFINE_EFFECTSTATEFACTORY_VTABLE(FshifterStateFactory);
EffectStateFactory *FshifterStateFactory_getFactory(void)
{
static FshifterStateFactory FshifterFactory = { { GET_VTABLE2(FshifterStateFactory, EffectStateFactory) } };
return STATIC_CAST(EffectStateFactory, &FshifterFactory);
}
void ALfshifter_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
{
ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_FREQUENCY_SHIFTER_FREQUENCY:
if(!(val >= AL_FREQUENCY_SHIFTER_MIN_FREQUENCY && val <= AL_FREQUENCY_SHIFTER_MAX_FREQUENCY))
SETERR_RETURN(context, AL_INVALID_VALUE,,"Frequency shifter frequency out of range");
props->Fshifter.Frequency = val;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid frequency shifter float property 0x%04x", param);
}
}
void ALfshifter_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
{
ALfshifter_setParamf(effect, context, param, vals[0]);
}
void ALfshifter_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
{
ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_FREQUENCY_SHIFTER_LEFT_DIRECTION:
if(!(val >= AL_FREQUENCY_SHIFTER_MIN_LEFT_DIRECTION && val <= AL_FREQUENCY_SHIFTER_MAX_LEFT_DIRECTION))
SETERR_RETURN(context, AL_INVALID_VALUE,,"Frequency shifter left direction out of range");
props->Fshifter.LeftDirection = val;
break;
case AL_FREQUENCY_SHIFTER_RIGHT_DIRECTION:
if(!(val >= AL_FREQUENCY_SHIFTER_MIN_RIGHT_DIRECTION && val <= AL_FREQUENCY_SHIFTER_MAX_RIGHT_DIRECTION))
SETERR_RETURN(context, AL_INVALID_VALUE,,"Frequency shifter right direction out of range");
props->Fshifter.RightDirection = val;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid frequency shifter integer property 0x%04x", param);
}
}
void ALfshifter_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
{
ALfshifter_setParami(effect, context, param, vals[0]);
}
void ALfshifter_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
{
const ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_FREQUENCY_SHIFTER_LEFT_DIRECTION:
*val = props->Fshifter.LeftDirection;
break;
case AL_FREQUENCY_SHIFTER_RIGHT_DIRECTION:
*val = props->Fshifter.RightDirection;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid frequency shifter integer property 0x%04x", param);
}
}
void ALfshifter_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
{
ALfshifter_getParami(effect, context, param, vals);
}
void ALfshifter_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
{
const ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_FREQUENCY_SHIFTER_FREQUENCY:
*val = props->Fshifter.Frequency;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid frequency shifter float property 0x%04x", param);
}
}
void ALfshifter_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
{
ALfshifter_getParamf(effect, context, param, vals);
}
DEFINE_ALEFFECT_VTABLE(ALfshifter);
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/**
* 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <math.h>
#include <stdlib.h>
#include "alMain.h"
#include "alAuxEffectSlot.h"
#include "alError.h"
#include "alu.h"
#include "filters/defs.h"
#define MAX_UPDATE_SAMPLES 128
typedef struct ALmodulatorState {
DERIVE_FROM_TYPE(ALeffectState);
void (*GetSamples)(ALfloat*, ALsizei, const ALsizei, ALsizei);
ALsizei index;
ALsizei step;
struct {
BiquadFilter Filter;
ALfloat CurrentGains[MAX_OUTPUT_CHANNELS];
ALfloat TargetGains[MAX_OUTPUT_CHANNELS];
} Chans[MAX_EFFECT_CHANNELS];
} ALmodulatorState;
static ALvoid ALmodulatorState_Destruct(ALmodulatorState *state);
static ALboolean ALmodulatorState_deviceUpdate(ALmodulatorState *state, ALCdevice *device);
static ALvoid ALmodulatorState_update(ALmodulatorState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props);
static ALvoid ALmodulatorState_process(ALmodulatorState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
DECLARE_DEFAULT_ALLOCATORS(ALmodulatorState)
DEFINE_ALEFFECTSTATE_VTABLE(ALmodulatorState);
#define WAVEFORM_FRACBITS 24
#define WAVEFORM_FRACONE (1<<WAVEFORM_FRACBITS)
#define WAVEFORM_FRACMASK (WAVEFORM_FRACONE-1)
static inline ALfloat Sin(ALsizei index)
{
return sinf((ALfloat)index * (F_TAU / WAVEFORM_FRACONE));
}
static inline ALfloat Saw(ALsizei index)
{
return (ALfloat)index*(2.0f/WAVEFORM_FRACONE) - 1.0f;
}
static inline ALfloat Square(ALsizei index)
{
return (ALfloat)(((index>>(WAVEFORM_FRACBITS-2))&2) - 1);
}
static inline ALfloat One(ALsizei UNUSED(index))
{
return 1.0f;
}
#define DECL_TEMPLATE(func) \
static void Modulate##func(ALfloat *restrict dst, ALsizei index, \
const ALsizei step, ALsizei todo) \
{ \
ALsizei i; \
for(i = 0;i < todo;i++) \
{ \
index += step; \
index &= WAVEFORM_FRACMASK; \
dst[i] = func(index); \
} \
}
DECL_TEMPLATE(Sin)
DECL_TEMPLATE(Saw)
DECL_TEMPLATE(Square)
DECL_TEMPLATE(One)
#undef DECL_TEMPLATE
static void ALmodulatorState_Construct(ALmodulatorState *state)
{
ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
SET_VTABLE2(ALmodulatorState, ALeffectState, state);
state->index = 0;
state->step = 1;
}
static ALvoid ALmodulatorState_Destruct(ALmodulatorState *state)
{
ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
}
static ALboolean ALmodulatorState_deviceUpdate(ALmodulatorState *state, ALCdevice *UNUSED(device))
{
ALsizei i, j;
for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
{
BiquadFilter_clear(&state->Chans[i].Filter);
for(j = 0;j < MAX_OUTPUT_CHANNELS;j++)
state->Chans[i].CurrentGains[j] = 0.0f;
}
return AL_TRUE;
}
static ALvoid ALmodulatorState_update(ALmodulatorState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props)
{
const ALCdevice *device = context->Device;
ALfloat f0norm;
ALsizei i;
state->step = fastf2i(props->Modulator.Frequency / (ALfloat)device->Frequency *
WAVEFORM_FRACONE);
state->step = clampi(state->step, 0, WAVEFORM_FRACONE-1);
if(state->step == 0)
state->GetSamples = ModulateOne;
else if(props->Modulator.Waveform == AL_RING_MODULATOR_SINUSOID)
state->GetSamples = ModulateSin;
else if(props->Modulator.Waveform == AL_RING_MODULATOR_SAWTOOTH)
state->GetSamples = ModulateSaw;
else /*if(Slot->Params.EffectProps.Modulator.Waveform == AL_RING_MODULATOR_SQUARE)*/
state->GetSamples = ModulateSquare;
f0norm = props->Modulator.HighPassCutoff / (ALfloat)device->Frequency;
f0norm = clampf(f0norm, 1.0f/512.0f, 0.49f);
/* Bandwidth value is constant in octaves. */
BiquadFilter_setParams(&state->Chans[0].Filter, BiquadType_HighPass, 1.0f,
f0norm, calc_rcpQ_from_bandwidth(f0norm, 0.75f));
for(i = 1;i < MAX_EFFECT_CHANNELS;i++)
BiquadFilter_copyParams(&state->Chans[i].Filter, &state->Chans[0].Filter);
STATIC_CAST(ALeffectState,state)->OutBuffer = device->FOAOut.Buffer;
STATIC_CAST(ALeffectState,state)->OutChannels = device->FOAOut.NumChannels;
for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
ComputePanGains(&device->FOAOut, IdentityMatrixf.m[i], slot->Params.Gain,
state->Chans[i].TargetGains);
}
static ALvoid ALmodulatorState_process(ALmodulatorState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
{
const ALsizei step = state->step;
ALsizei base;
for(base = 0;base < SamplesToDo;)
{
alignas(16) ALfloat modsamples[MAX_UPDATE_SAMPLES];
ALsizei td = mini(MAX_UPDATE_SAMPLES, SamplesToDo-base);
ALsizei c, i;
state->GetSamples(modsamples, state->index, step, td);
state->index += (step*td) & WAVEFORM_FRACMASK;
state->index &= WAVEFORM_FRACMASK;
for(c = 0;c < MAX_EFFECT_CHANNELS;c++)
{
alignas(16) ALfloat temps[MAX_UPDATE_SAMPLES];
BiquadFilter_process(&state->Chans[c].Filter, temps, &SamplesIn[c][base], td);
for(i = 0;i < td;i++)
temps[i] *= modsamples[i];
MixSamples(temps, NumChannels, SamplesOut, state->Chans[c].CurrentGains,
state->Chans[c].TargetGains, SamplesToDo-base, base, td);
}
base += td;
}
}
typedef struct ModulatorStateFactory {
DERIVE_FROM_TYPE(EffectStateFactory);
} ModulatorStateFactory;
static ALeffectState *ModulatorStateFactory_create(ModulatorStateFactory *UNUSED(factory))
{
ALmodulatorState *state;
NEW_OBJ0(state, ALmodulatorState)();
if(!state) return NULL;
return STATIC_CAST(ALeffectState, state);
}
DEFINE_EFFECTSTATEFACTORY_VTABLE(ModulatorStateFactory);
EffectStateFactory *ModulatorStateFactory_getFactory(void)
{
static ModulatorStateFactory ModulatorFactory = { { GET_VTABLE2(ModulatorStateFactory, EffectStateFactory) } };
return STATIC_CAST(EffectStateFactory, &ModulatorFactory);
}
void ALmodulator_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
{
ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_RING_MODULATOR_FREQUENCY:
if(!(val >= AL_RING_MODULATOR_MIN_FREQUENCY && val <= AL_RING_MODULATOR_MAX_FREQUENCY))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Modulator frequency out of range");
props->Modulator.Frequency = val;
break;
case AL_RING_MODULATOR_HIGHPASS_CUTOFF:
if(!(val >= AL_RING_MODULATOR_MIN_HIGHPASS_CUTOFF && val <= AL_RING_MODULATOR_MAX_HIGHPASS_CUTOFF))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Modulator high-pass cutoff out of range");
props->Modulator.HighPassCutoff = val;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid modulator float property 0x%04x", param);
}
}
void ALmodulator_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
{ ALmodulator_setParamf(effect, context, param, vals[0]); }
void ALmodulator_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
{
ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_RING_MODULATOR_FREQUENCY:
case AL_RING_MODULATOR_HIGHPASS_CUTOFF:
ALmodulator_setParamf(effect, context, param, (ALfloat)val);
break;
case AL_RING_MODULATOR_WAVEFORM:
if(!(val >= AL_RING_MODULATOR_MIN_WAVEFORM && val <= AL_RING_MODULATOR_MAX_WAVEFORM))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Invalid modulator waveform");
props->Modulator.Waveform = val;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid modulator integer property 0x%04x", param);
}
}
void ALmodulator_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
{ ALmodulator_setParami(effect, context, param, vals[0]); }
void ALmodulator_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
{
const ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_RING_MODULATOR_FREQUENCY:
*val = (ALint)props->Modulator.Frequency;
break;
case AL_RING_MODULATOR_HIGHPASS_CUTOFF:
*val = (ALint)props->Modulator.HighPassCutoff;
break;
case AL_RING_MODULATOR_WAVEFORM:
*val = props->Modulator.Waveform;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid modulator integer property 0x%04x", param);
}
}
void ALmodulator_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
{ ALmodulator_getParami(effect, context, param, vals); }
void ALmodulator_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
{
const ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_RING_MODULATOR_FREQUENCY:
*val = props->Modulator.Frequency;
break;
case AL_RING_MODULATOR_HIGHPASS_CUTOFF:
*val = props->Modulator.HighPassCutoff;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid modulator float property 0x%04x", param);
}
}
void ALmodulator_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
{ ALmodulator_getParamf(effect, context, param, vals); }
DEFINE_ALEFFECT_VTABLE(ALmodulator);
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#include "config.h"
#include <stdlib.h>
#include "AL/al.h"
#include "AL/alc.h"
#include "alMain.h"
#include "alAuxEffectSlot.h"
#include "alError.h"
typedef struct ALnullState {
DERIVE_FROM_TYPE(ALeffectState);
} ALnullState;
/* Forward-declare "virtual" functions to define the vtable with. */
static ALvoid ALnullState_Destruct(ALnullState *state);
static ALboolean ALnullState_deviceUpdate(ALnullState *state, ALCdevice *device);
static ALvoid ALnullState_update(ALnullState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props);
static ALvoid ALnullState_process(ALnullState *state, ALsizei samplesToDo, const ALfloat (*restrict samplesIn)[BUFFERSIZE], ALfloat (*restrict samplesOut)[BUFFERSIZE], ALsizei mumChannels);
static void *ALnullState_New(size_t size);
static void ALnullState_Delete(void *ptr);
/* Define the ALeffectState vtable for this type. */
DEFINE_ALEFFECTSTATE_VTABLE(ALnullState);
/* This constructs the effect state. It's called when the object is first
* created. Make sure to call the parent Construct function first, and set the
* vtable!
*/
static void ALnullState_Construct(ALnullState *state)
{
ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
SET_VTABLE2(ALnullState, ALeffectState, state);
}
/* This destructs (not free!) the effect state. It's called only when the
* effect slot is no longer used. Make sure to call the parent Destruct
* function before returning!
*/
static ALvoid ALnullState_Destruct(ALnullState *state)
{
ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
}
/* This updates the device-dependant effect state. This is called on
* initialization and any time the device parameters (eg. playback frequency,
* format) have been changed.
*/
static ALboolean ALnullState_deviceUpdate(ALnullState* UNUSED(state), ALCdevice* UNUSED(device))
{
return AL_TRUE;
}
/* This updates the effect state. This is called any time the effect is
* (re)loaded into a slot.
*/
static ALvoid ALnullState_update(ALnullState* UNUSED(state), const ALCcontext* UNUSED(context), const ALeffectslot* UNUSED(slot), const ALeffectProps* UNUSED(props))
{
}
/* This processes the effect state, for the given number of samples from the
* input to the output buffer. The result should be added to the output buffer,
* not replace it.
*/
static ALvoid ALnullState_process(ALnullState* UNUSED(state), ALsizei UNUSED(samplesToDo), const ALfloatBUFFERSIZE*restrict UNUSED(samplesIn), ALfloatBUFFERSIZE*restrict UNUSED(samplesOut), ALsizei UNUSED(numChannels))
{
}
/* This allocates memory to store the object, before it gets constructed.
* DECLARE_DEFAULT_ALLOCATORS can be used to declare a default method.
*/
static void *ALnullState_New(size_t size)
{
return al_malloc(16, size);
}
/* This frees the memory used by the object, after it has been destructed.
* DECLARE_DEFAULT_ALLOCATORS can be used to declare a default method.
*/
static void ALnullState_Delete(void *ptr)
{
al_free(ptr);
}
typedef struct NullStateFactory {
DERIVE_FROM_TYPE(EffectStateFactory);
} NullStateFactory;
/* Creates ALeffectState objects of the appropriate type. */
ALeffectState *NullStateFactory_create(NullStateFactory *UNUSED(factory))
{
ALnullState *state;
NEW_OBJ0(state, ALnullState)();
if(!state) return NULL;
return STATIC_CAST(ALeffectState, state);
}
/* Define the EffectStateFactory vtable for this type. */
DEFINE_EFFECTSTATEFACTORY_VTABLE(NullStateFactory);
EffectStateFactory *NullStateFactory_getFactory(void)
{
static NullStateFactory NullFactory = { { GET_VTABLE2(NullStateFactory, EffectStateFactory) } };
return STATIC_CAST(EffectStateFactory, &NullFactory);
}
void ALnull_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint UNUSED(val))
{
switch(param)
{
default:
alSetError(context, AL_INVALID_ENUM, "Invalid null effect integer property 0x%04x", param);
}
}
void ALnull_setParamiv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALint* UNUSED(vals))
{
switch(param)
{
default:
alSetError(context, AL_INVALID_ENUM, "Invalid null effect integer-vector property 0x%04x", param);
}
}
void ALnull_setParamf(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat UNUSED(val))
{
switch(param)
{
default:
alSetError(context, AL_INVALID_ENUM, "Invalid null effect float property 0x%04x", param);
}
}
void ALnull_setParamfv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALfloat* UNUSED(vals))
{
switch(param)
{
default:
alSetError(context, AL_INVALID_ENUM, "Invalid null effect float-vector property 0x%04x", param);
}
}
void ALnull_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint* UNUSED(val))
{
switch(param)
{
default:
alSetError(context, AL_INVALID_ENUM, "Invalid null effect integer property 0x%04x", param);
}
}
void ALnull_getParamiv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint* UNUSED(vals))
{
switch(param)
{
default:
alSetError(context, AL_INVALID_ENUM, "Invalid null effect integer-vector property 0x%04x", param);
}
}
void ALnull_getParamf(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat* UNUSED(val))
{
switch(param)
{
default:
alSetError(context, AL_INVALID_ENUM, "Invalid null effect float property 0x%04x", param);
}
}
void ALnull_getParamfv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat* UNUSED(vals))
{
switch(param)
{
default:
alSetError(context, AL_INVALID_ENUM, "Invalid null effect float-vector property 0x%04x", param);
}
}
DEFINE_ALEFFECT_VTABLE(ALnull);
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@@ -1,441 +0,0 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 2018 by Raul Herraiz.
* 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <math.h>
#include <stdlib.h>
#include "alMain.h"
#include "alAuxEffectSlot.h"
#include "alError.h"
#include "alu.h"
#include "filters/defs.h"
#include "alcomplex.h"
#define STFT_SIZE 1024
#define STFT_HALF_SIZE (STFT_SIZE>>1)
#define OVERSAMP (1<<2)
#define STFT_STEP (STFT_SIZE / OVERSAMP)
#define FIFO_LATENCY (STFT_STEP * (OVERSAMP-1))
typedef struct ALphasor {
ALdouble Amplitude;
ALdouble Phase;
} ALphasor;
typedef struct ALFrequencyDomain {
ALdouble Amplitude;
ALdouble Frequency;
} ALfrequencyDomain;
typedef struct ALpshifterState {
DERIVE_FROM_TYPE(ALeffectState);
/* Effect parameters */
ALsizei count;
ALsizei PitchShiftI;
ALfloat PitchShift;
ALfloat FreqPerBin;
/*Effects buffers*/
ALfloat InFIFO[STFT_SIZE];
ALfloat OutFIFO[STFT_STEP];
ALdouble LastPhase[STFT_HALF_SIZE+1];
ALdouble SumPhase[STFT_HALF_SIZE+1];
ALdouble OutputAccum[STFT_SIZE];
ALcomplex FFTbuffer[STFT_SIZE];
ALfrequencyDomain Analysis_buffer[STFT_HALF_SIZE+1];
ALfrequencyDomain Syntesis_buffer[STFT_HALF_SIZE+1];
alignas(16) ALfloat BufferOut[BUFFERSIZE];
/* Effect gains for each output channel */
ALfloat CurrentGains[MAX_OUTPUT_CHANNELS];
ALfloat TargetGains[MAX_OUTPUT_CHANNELS];
} ALpshifterState;
static ALvoid ALpshifterState_Destruct(ALpshifterState *state);
static ALboolean ALpshifterState_deviceUpdate(ALpshifterState *state, ALCdevice *device);
static ALvoid ALpshifterState_update(ALpshifterState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props);
static ALvoid ALpshifterState_process(ALpshifterState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
DECLARE_DEFAULT_ALLOCATORS(ALpshifterState)
DEFINE_ALEFFECTSTATE_VTABLE(ALpshifterState);
/* Define a Hann window, used to filter the STFT input and output. */
alignas(16) static ALdouble HannWindow[STFT_SIZE];
static void InitHannWindow(void)
{
ALsizei i;
/* Create lookup table of the Hann window for the desired size, i.e. STFT_SIZE */
for(i = 0;i < STFT_SIZE>>1;i++)
{
ALdouble val = sin(M_PI * (ALdouble)i / (ALdouble)(STFT_SIZE-1));
HannWindow[i] = HannWindow[STFT_SIZE-1-i] = val * val;
}
}
static alonce_flag HannInitOnce = AL_ONCE_FLAG_INIT;
static inline ALint double2int(ALdouble d)
{
#if ((defined(__GNUC__) || defined(__clang__)) && (defined(__i386__) || defined(__x86_64__)) && \
!defined(__SSE2_MATH__)) || (defined(_MSC_VER) && defined(_M_IX86_FP) && _M_IX86_FP < 2)
ALint sign, shift;
ALint64 mant;
union {
ALdouble d;
ALint64 i64;
} conv;
conv.d = d;
sign = (conv.i64>>63) | 1;
shift = ((conv.i64>>52)&0x7ff) - (1023+52);
/* Over/underflow */
if(UNLIKELY(shift >= 63 || shift < -52))
return 0;
mant = (conv.i64&I64(0xfffffffffffff)) | I64(0x10000000000000);
if(LIKELY(shift < 0))
return (ALint)(mant >> -shift) * sign;
return (ALint)(mant << shift) * sign;
#else
return (ALint)d;
#endif
}
/* Converts ALcomplex to ALphasor */
static inline ALphasor rect2polar(ALcomplex number)
{
ALphasor polar;
polar.Amplitude = sqrt(number.Real*number.Real + number.Imag*number.Imag);
polar.Phase = atan2(number.Imag, number.Real);
return polar;
}
/* Converts ALphasor to ALcomplex */
static inline ALcomplex polar2rect(ALphasor number)
{
ALcomplex cartesian;
cartesian.Real = number.Amplitude * cos(number.Phase);
cartesian.Imag = number.Amplitude * sin(number.Phase);
return cartesian;
}
static void ALpshifterState_Construct(ALpshifterState *state)
{
ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
SET_VTABLE2(ALpshifterState, ALeffectState, state);
alcall_once(&HannInitOnce, InitHannWindow);
}
static ALvoid ALpshifterState_Destruct(ALpshifterState *state)
{
ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
}
static ALboolean ALpshifterState_deviceUpdate(ALpshifterState *state, ALCdevice *device)
{
/* (Re-)initializing parameters and clear the buffers. */
state->count = FIFO_LATENCY;
state->PitchShiftI = FRACTIONONE;
state->PitchShift = 1.0f;
state->FreqPerBin = device->Frequency / (ALfloat)STFT_SIZE;
memset(state->InFIFO, 0, sizeof(state->InFIFO));
memset(state->OutFIFO, 0, sizeof(state->OutFIFO));
memset(state->FFTbuffer, 0, sizeof(state->FFTbuffer));
memset(state->LastPhase, 0, sizeof(state->LastPhase));
memset(state->SumPhase, 0, sizeof(state->SumPhase));
memset(state->OutputAccum, 0, sizeof(state->OutputAccum));
memset(state->Analysis_buffer, 0, sizeof(state->Analysis_buffer));
memset(state->Syntesis_buffer, 0, sizeof(state->Syntesis_buffer));
memset(state->CurrentGains, 0, sizeof(state->CurrentGains));
memset(state->TargetGains, 0, sizeof(state->TargetGains));
return AL_TRUE;
}
static ALvoid ALpshifterState_update(ALpshifterState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props)
{
const ALCdevice *device = context->Device;
ALfloat coeffs[MAX_AMBI_COEFFS];
float pitch;
pitch = powf(2.0f,
(ALfloat)(props->Pshifter.CoarseTune*100 + props->Pshifter.FineTune) / 1200.0f
);
state->PitchShiftI = fastf2i(pitch*FRACTIONONE);
state->PitchShift = state->PitchShiftI * (1.0f/FRACTIONONE);
CalcAngleCoeffs(0.0f, 0.0f, 0.0f, coeffs);
ComputePanGains(&device->Dry, coeffs, slot->Params.Gain, state->TargetGains);
}
static ALvoid ALpshifterState_process(ALpshifterState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
{
/* Pitch shifter engine based on the work of Stephan Bernsee.
* http://blogs.zynaptiq.com/bernsee/pitch-shifting-using-the-ft/
*/
static const ALdouble expected = M_PI*2.0 / OVERSAMP;
const ALdouble freq_per_bin = state->FreqPerBin;
ALfloat *restrict bufferOut = state->BufferOut;
ALsizei count = state->count;
ALsizei i, j, k;
for(i = 0;i < SamplesToDo;)
{
do {
/* Fill FIFO buffer with samples data */
state->InFIFO[count] = SamplesIn[0][i];
bufferOut[i] = state->OutFIFO[count - FIFO_LATENCY];
count++;
} while(++i < SamplesToDo && count < STFT_SIZE);
/* Check whether FIFO buffer is filled */
if(count < STFT_SIZE) break;
count = FIFO_LATENCY;
/* Real signal windowing and store in FFTbuffer */
for(k = 0;k < STFT_SIZE;k++)
{
state->FFTbuffer[k].Real = state->InFIFO[k] * HannWindow[k];
state->FFTbuffer[k].Imag = 0.0;
}
/* ANALYSIS */
/* Apply FFT to FFTbuffer data */
complex_fft(state->FFTbuffer, STFT_SIZE, -1.0);
/* Analyze the obtained data. Since the real FFT is symmetric, only
* STFT_HALF_SIZE+1 samples are needed.
*/
for(k = 0;k < STFT_HALF_SIZE+1;k++)
{
ALphasor component;
ALdouble tmp;
ALint qpd;
/* Compute amplitude and phase */
component = rect2polar(state->FFTbuffer[k]);
/* Compute phase difference and subtract expected phase difference */
tmp = (component.Phase - state->LastPhase[k]) - k*expected;
/* Map delta phase into +/- Pi interval */
qpd = double2int(tmp / M_PI);
tmp -= M_PI * (qpd + (qpd%2));
/* Get deviation from bin frequency from the +/- Pi interval */
tmp /= expected;
/* Compute the k-th partials' true frequency, twice the amplitude
* for maintain the gain (because half of bins are used) and store
* amplitude and true frequency in analysis buffer.
*/
state->Analysis_buffer[k].Amplitude = 2.0 * component.Amplitude;
state->Analysis_buffer[k].Frequency = (k + tmp) * freq_per_bin;
/* Store actual phase[k] for the calculations in the next frame*/
state->LastPhase[k] = component.Phase;
}
/* PROCESSING */
/* pitch shifting */
for(k = 0;k < STFT_HALF_SIZE+1;k++)
{
state->Syntesis_buffer[k].Amplitude = 0.0;
state->Syntesis_buffer[k].Frequency = 0.0;
}
for(k = 0;k < STFT_HALF_SIZE+1;k++)
{
j = (k*state->PitchShiftI) >> FRACTIONBITS;
if(j >= STFT_HALF_SIZE+1) break;
state->Syntesis_buffer[j].Amplitude += state->Analysis_buffer[k].Amplitude;
state->Syntesis_buffer[j].Frequency = state->Analysis_buffer[k].Frequency *
state->PitchShift;
}
/* SYNTHESIS */
/* Synthesis the processing data */
for(k = 0;k < STFT_HALF_SIZE+1;k++)
{
ALphasor component;
ALdouble tmp;
/* Compute bin deviation from scaled freq */
tmp = state->Syntesis_buffer[k].Frequency/freq_per_bin - k;
/* Calculate actual delta phase and accumulate it to get bin phase */
state->SumPhase[k] += (k + tmp) * expected;
component.Amplitude = state->Syntesis_buffer[k].Amplitude;
component.Phase = state->SumPhase[k];
/* Compute phasor component to cartesian complex number and storage it into FFTbuffer*/
state->FFTbuffer[k] = polar2rect(component);
}
/* zero negative frequencies for recontruct a real signal */
for(k = STFT_HALF_SIZE+1;k < STFT_SIZE;k++)
{
state->FFTbuffer[k].Real = 0.0;
state->FFTbuffer[k].Imag = 0.0;
}
/* Apply iFFT to buffer data */
complex_fft(state->FFTbuffer, STFT_SIZE, 1.0);
/* Windowing and add to output */
for(k = 0;k < STFT_SIZE;k++)
state->OutputAccum[k] += HannWindow[k] * state->FFTbuffer[k].Real /
(0.5 * STFT_HALF_SIZE * OVERSAMP);
/* Shift accumulator, input & output FIFO */
for(k = 0;k < STFT_STEP;k++) state->OutFIFO[k] = (ALfloat)state->OutputAccum[k];
for(j = 0;k < STFT_SIZE;k++,j++) state->OutputAccum[j] = state->OutputAccum[k];
for(;j < STFT_SIZE;j++) state->OutputAccum[j] = 0.0;
for(k = 0;k < FIFO_LATENCY;k++)
state->InFIFO[k] = state->InFIFO[k+STFT_STEP];
}
state->count = count;
/* Now, mix the processed sound data to the output. */
MixSamples(bufferOut, NumChannels, SamplesOut, state->CurrentGains, state->TargetGains,
maxi(SamplesToDo, 512), 0, SamplesToDo);
}
typedef struct PshifterStateFactory {
DERIVE_FROM_TYPE(EffectStateFactory);
} PshifterStateFactory;
static ALeffectState *PshifterStateFactory_create(PshifterStateFactory *UNUSED(factory))
{
ALpshifterState *state;
NEW_OBJ0(state, ALpshifterState)();
if(!state) return NULL;
return STATIC_CAST(ALeffectState, state);
}
DEFINE_EFFECTSTATEFACTORY_VTABLE(PshifterStateFactory);
EffectStateFactory *PshifterStateFactory_getFactory(void)
{
static PshifterStateFactory PshifterFactory = { { GET_VTABLE2(PshifterStateFactory, EffectStateFactory) } };
return STATIC_CAST(EffectStateFactory, &PshifterFactory);
}
void ALpshifter_setParamf(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat UNUSED(val))
{
alSetError( context, AL_INVALID_ENUM, "Invalid pitch shifter float property 0x%04x", param );
}
void ALpshifter_setParamfv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALfloat *UNUSED(vals))
{
alSetError( context, AL_INVALID_ENUM, "Invalid pitch shifter float-vector property 0x%04x", param );
}
void ALpshifter_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
{
ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_PITCH_SHIFTER_COARSE_TUNE:
if(!(val >= AL_PITCH_SHIFTER_MIN_COARSE_TUNE && val <= AL_PITCH_SHIFTER_MAX_COARSE_TUNE))
SETERR_RETURN(context, AL_INVALID_VALUE,,"Pitch shifter coarse tune out of range");
props->Pshifter.CoarseTune = val;
break;
case AL_PITCH_SHIFTER_FINE_TUNE:
if(!(val >= AL_PITCH_SHIFTER_MIN_FINE_TUNE && val <= AL_PITCH_SHIFTER_MAX_FINE_TUNE))
SETERR_RETURN(context, AL_INVALID_VALUE,,"Pitch shifter fine tune out of range");
props->Pshifter.FineTune = val;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter integer property 0x%04x", param);
}
}
void ALpshifter_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
{
ALpshifter_setParami(effect, context, param, vals[0]);
}
void ALpshifter_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
{
const ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_PITCH_SHIFTER_COARSE_TUNE:
*val = (ALint)props->Pshifter.CoarseTune;
break;
case AL_PITCH_SHIFTER_FINE_TUNE:
*val = (ALint)props->Pshifter.FineTune;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter integer property 0x%04x", param);
}
}
void ALpshifter_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
{
ALpshifter_getParami(effect, context, param, vals);
}
void ALpshifter_getParamf(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat *UNUSED(val))
{
alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter float property 0x%04x", param);
}
void ALpshifter_getParamfv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat *UNUSED(vals))
{
alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter float vector-property 0x%04x", param);
}
DEFINE_ALEFFECT_VTABLE(ALpshifter);
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#ifndef ALC_FILTER_H
#define ALC_FILTER_H
#include "AL/al.h"
#include "math_defs.h"
/* Filters implementation is based on the "Cookbook formulae for audio
* EQ biquad filter coefficients" by Robert Bristow-Johnson
* http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt
*/
/* Implementation note: For the shelf filters, the specified gain is for the
* reference frequency, which is the centerpoint of the transition band. This
* better matches EFX filter design. To set the gain for the shelf itself, use
* the square root of the desired linear gain (or halve the dB gain).
*/
typedef enum BiquadType {
/** EFX-style low-pass filter, specifying a gain and reference frequency. */
BiquadType_HighShelf,
/** EFX-style high-pass filter, specifying a gain and reference frequency. */
BiquadType_LowShelf,
/** Peaking filter, specifying a gain and reference frequency. */
BiquadType_Peaking,
/** Low-pass cut-off filter, specifying a cut-off frequency. */
BiquadType_LowPass,
/** High-pass cut-off filter, specifying a cut-off frequency. */
BiquadType_HighPass,
/** Band-pass filter, specifying a center frequency. */
BiquadType_BandPass,
} BiquadType;
typedef struct BiquadFilter {
ALfloat z1, z2; /* Last two delayed components for direct form II. */
ALfloat b0, b1, b2; /* Transfer function coefficients "b" (numerator) */
ALfloat a1, a2; /* Transfer function coefficients "a" (denominator; a0 is
* pre-applied). */
} BiquadFilter;
/* Currently only a C-based filter process method is implemented. */
#define BiquadFilter_process BiquadFilter_processC
/**
* Calculates the rcpQ (i.e. 1/Q) coefficient for shelving filters, using the
* reference gain and shelf slope parameter.
* \param gain 0 < gain
* \param slope 0 < slope <= 1
*/
inline ALfloat calc_rcpQ_from_slope(ALfloat gain, ALfloat slope)
{
return sqrtf((gain + 1.0f/gain)*(1.0f/slope - 1.0f) + 2.0f);
}
/**
* Calculates the rcpQ (i.e. 1/Q) coefficient for filters, using the normalized
* reference frequency and bandwidth.
* \param f0norm 0 < f0norm < 0.5.
* \param bandwidth 0 < bandwidth
*/
inline ALfloat calc_rcpQ_from_bandwidth(ALfloat f0norm, ALfloat bandwidth)
{
ALfloat w0 = F_TAU * f0norm;
return 2.0f*sinhf(logf(2.0f)/2.0f*bandwidth*w0/sinf(w0));
}
inline void BiquadFilter_clear(BiquadFilter *filter)
{
filter->z1 = 0.0f;
filter->z2 = 0.0f;
}
/**
* Sets up the filter state for the specified filter type and its parameters.
*
* \param filter The filter object to prepare.
* \param type The type of filter for the object to apply.
* \param gain The gain for the reference frequency response. Only used by the
* Shelf and Peaking filter types.
* \param f0norm The normalized reference frequency (ref_freq / sample_rate).
* This is the center point for the Shelf, Peaking, and BandPass
* filter types, or the cutoff frequency for the LowPass and
* HighPass filter types.
* \param rcpQ The reciprocal of the Q coefficient for the filter's transition
* band. Can be generated from calc_rcpQ_from_slope or
* calc_rcpQ_from_bandwidth depending on the available data.
*/
void BiquadFilter_setParams(BiquadFilter *filter, BiquadType type, ALfloat gain, ALfloat f0norm, ALfloat rcpQ);
inline void BiquadFilter_copyParams(BiquadFilter *restrict dst, const BiquadFilter *restrict src)
{
dst->b0 = src->b0;
dst->b1 = src->b1;
dst->b2 = src->b2;
dst->a1 = src->a1;
dst->a2 = src->a2;
}
void BiquadFilter_processC(BiquadFilter *filter, ALfloat *restrict dst, const ALfloat *restrict src, ALsizei numsamples);
inline void BiquadFilter_passthru(BiquadFilter *filter, ALsizei numsamples)
{
if(LIKELY(numsamples >= 2))
{
filter->z1 = 0.0f;
filter->z2 = 0.0f;
}
else if(numsamples == 1)
{
filter->z1 = filter->z2;
filter->z2 = 0.0f;
}
}
#endif /* ALC_FILTER_H */
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#include "config.h"
#include "nfc.h"
#include "alMain.h"
#include <string.h>
/* Near-field control filters are the basis for handling the near-field effect.
* The near-field effect is a bass-boost present in the directional components
* of a recorded signal, created as a result of the wavefront curvature (itself
* a function of sound distance). Proper reproduction dictates this be
* compensated for using a bass-cut given the playback speaker distance, to
* avoid excessive bass in the playback.
*
* For real-time rendered audio, emulating the near-field effect based on the
* sound source's distance, and subsequently compensating for it at output
* based on the speaker distances, can create a more realistic perception of
* sound distance beyond a simple 1/r attenuation.
*
* These filters do just that. Each one applies a low-shelf filter, created as
* the combination of a bass-boost for a given sound source distance (near-
* field emulation) along with a bass-cut for a given control/speaker distance
* (near-field compensation).
*
* Note that it is necessary to apply a cut along with the boost, since the
* boost alone is unstable in higher-order ambisonics as it causes an infinite
* DC gain (even first-order ambisonics requires there to be no DC offset for
* the boost to work). Consequently, ambisonics requires a control parameter to
* be used to avoid an unstable boost-only filter. NFC-HOA defines this control
* as a reference delay, calculated with:
*
* reference_delay = control_distance / speed_of_sound
*
* This means w0 (for input) or w1 (for output) should be set to:
*
* wN = 1 / (reference_delay * sample_rate)
*
* when dealing with NFC-HOA content. For FOA input content, which does not
* specify a reference_delay variable, w0 should be set to 0 to apply only
* near-field compensation for output. It's important that w1 be a finite,
* positive, non-0 value or else the bass-boost will become unstable again.
* Also, w0 should not be too large compared to w1, to avoid excessively loud
* low frequencies.
*/
static const float B[4][3] = {
{ 0.0f },
{ 1.0f },
{ 3.0f, 3.0f },
{ 3.6778f, 6.4595f, 2.3222f },
/*{ 4.2076f, 11.4877f, 5.7924f, 9.1401f }*/
};
static void NfcFilterCreate1(struct NfcFilter1 *nfc, const float w0, const float w1)
{
float b_00, g_0;
float r;
nfc->base_gain = 1.0f;
nfc->gain = 1.0f;
/* Calculate bass-boost coefficients. */
r = 0.5f * w0;
b_00 = B[1][0] * r;
g_0 = 1.0f + b_00;
nfc->gain *= g_0;
nfc->b1 = 2.0f * b_00 / g_0;
/* Calculate bass-cut coefficients. */
r = 0.5f * w1;
b_00 = B[1][0] * r;
g_0 = 1.0f + b_00;
nfc->base_gain /= g_0;
nfc->gain /= g_0;
nfc->a1 = 2.0f * b_00 / g_0;
}
static void NfcFilterAdjust1(struct NfcFilter1 *nfc, const float w0)
{
float b_00, g_0;
float r;
r = 0.5f * w0;
b_00 = B[1][0] * r;
g_0 = 1.0f + b_00;
nfc->gain = nfc->base_gain * g_0;
nfc->b1 = 2.0f * b_00 / g_0;
}
static void NfcFilterCreate2(struct NfcFilter2 *nfc, const float w0, const float w1)
{
float b_10, b_11, g_1;
float r;
nfc->base_gain = 1.0f;
nfc->gain = 1.0f;
/* Calculate bass-boost coefficients. */
r = 0.5f * w0;
b_10 = B[2][0] * r;
b_11 = B[2][1] * r * r;
g_1 = 1.0f + b_10 + b_11;
nfc->gain *= g_1;
nfc->b1 = (2.0f*b_10 + 4.0f*b_11) / g_1;
nfc->b2 = 4.0f * b_11 / g_1;
/* Calculate bass-cut coefficients. */
r = 0.5f * w1;
b_10 = B[2][0] * r;
b_11 = B[2][1] * r * r;
g_1 = 1.0f + b_10 + b_11;
nfc->base_gain /= g_1;
nfc->gain /= g_1;
nfc->a1 = (2.0f*b_10 + 4.0f*b_11) / g_1;
nfc->a2 = 4.0f * b_11 / g_1;
}
static void NfcFilterAdjust2(struct NfcFilter2 *nfc, const float w0)
{
float b_10, b_11, g_1;
float r;
r = 0.5f * w0;
b_10 = B[2][0] * r;
b_11 = B[2][1] * r * r;
g_1 = 1.0f + b_10 + b_11;
nfc->gain = nfc->base_gain * g_1;
nfc->b1 = (2.0f*b_10 + 4.0f*b_11) / g_1;
nfc->b2 = 4.0f * b_11 / g_1;
}
static void NfcFilterCreate3(struct NfcFilter3 *nfc, const float w0, const float w1)
{
float b_10, b_11, g_1;
float b_00, g_0;
float r;
nfc->base_gain = 1.0f;
nfc->gain = 1.0f;
/* Calculate bass-boost coefficients. */
r = 0.5f * w0;
b_10 = B[3][0] * r;
b_11 = B[3][1] * r * r;
g_1 = 1.0f + b_10 + b_11;
nfc->gain *= g_1;
nfc->b1 = (2.0f*b_10 + 4.0f*b_11) / g_1;
nfc->b2 = 4.0f * b_11 / g_1;
b_00 = B[3][2] * r;
g_0 = 1.0f + b_00;
nfc->gain *= g_0;
nfc->b3 = 2.0f * b_00 / g_0;
/* Calculate bass-cut coefficients. */
r = 0.5f * w1;
b_10 = B[3][0] * r;
b_11 = B[3][1] * r * r;
g_1 = 1.0f + b_10 + b_11;
nfc->base_gain /= g_1;
nfc->gain /= g_1;
nfc->a1 = (2.0f*b_10 + 4.0f*b_11) / g_1;
nfc->a2 = 4.0f * b_11 / g_1;
b_00 = B[3][2] * r;
g_0 = 1.0f + b_00;
nfc->base_gain /= g_0;
nfc->gain /= g_0;
nfc->a3 = 2.0f * b_00 / g_0;
}
static void NfcFilterAdjust3(struct NfcFilter3 *nfc, const float w0)
{
float b_10, b_11, g_1;
float b_00, g_0;
float r;
r = 0.5f * w0;
b_10 = B[3][0] * r;
b_11 = B[3][1] * r * r;
g_1 = 1.0f + b_10 + b_11;
nfc->gain = nfc->base_gain * g_1;
nfc->b1 = (2.0f*b_10 + 4.0f*b_11) / g_1;
nfc->b2 = 4.0f * b_11 / g_1;
b_00 = B[3][2] * r;
g_0 = 1.0f + b_00;
nfc->gain *= g_0;
nfc->b3 = 2.0f * b_00 / g_0;
}
void NfcFilterCreate(NfcFilter *nfc, const float w0, const float w1)
{
memset(nfc, 0, sizeof(*nfc));
NfcFilterCreate1(&nfc->first, w0, w1);
NfcFilterCreate2(&nfc->second, w0, w1);
NfcFilterCreate3(&nfc->third, w0, w1);
}
void NfcFilterAdjust(NfcFilter *nfc, const float w0)
{
NfcFilterAdjust1(&nfc->first, w0);
NfcFilterAdjust2(&nfc->second, w0);
NfcFilterAdjust3(&nfc->third, w0);
}
void NfcFilterProcess1(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count)
{
const float gain = nfc->first.gain;
const float b1 = nfc->first.b1;
const float a1 = nfc->first.a1;
float z1 = nfc->first.z[0];
int i;
ASSUME(count > 0);
for(i = 0;i < count;i++)
{
float y = src[i]*gain - a1*z1;
float out = y + b1*z1;
z1 += y;
dst[i] = out;
}
nfc->first.z[0] = z1;
}
void NfcFilterProcess2(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count)
{
const float gain = nfc->second.gain;
const float b1 = nfc->second.b1;
const float b2 = nfc->second.b2;
const float a1 = nfc->second.a1;
const float a2 = nfc->second.a2;
float z1 = nfc->second.z[0];
float z2 = nfc->second.z[1];
int i;
ASSUME(count > 0);
for(i = 0;i < count;i++)
{
float y = src[i]*gain - a1*z1 - a2*z2;
float out = y + b1*z1 + b2*z2;
z2 += z1;
z1 += y;
dst[i] = out;
}
nfc->second.z[0] = z1;
nfc->second.z[1] = z2;
}
void NfcFilterProcess3(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count)
{
const float gain = nfc->third.gain;
const float b1 = nfc->third.b1;
const float b2 = nfc->third.b2;
const float b3 = nfc->third.b3;
const float a1 = nfc->third.a1;
const float a2 = nfc->third.a2;
const float a3 = nfc->third.a3;
float z1 = nfc->third.z[0];
float z2 = nfc->third.z[1];
float z3 = nfc->third.z[2];
int i;
ASSUME(count > 0);
for(i = 0;i < count;i++)
{
float y = src[i]*gain - a1*z1 - a2*z2;
float out = y + b1*z1 + b2*z2;
z2 += z1;
z1 += y;
y = out - a3*z3;
out = y + b3*z3;
z3 += y;
dst[i] = out;
}
nfc->third.z[0] = z1;
nfc->third.z[1] = z2;
nfc->third.z[2] = z3;
}
#if 0 /* Original methods the above are derived from. */
static void NfcFilterCreate(NfcFilter *nfc, const ALsizei order, const float src_dist, const float ctl_dist, const float rate)
{
static const float B[4][5] = {
{ },
{ 1.0f },
{ 3.0f, 3.0f },
{ 3.6778f, 6.4595f, 2.3222f },
{ 4.2076f, 11.4877f, 5.7924f, 9.1401f }
};
float w0 = SPEEDOFSOUNDMETRESPERSEC / (src_dist * rate);
float w1 = SPEEDOFSOUNDMETRESPERSEC / (ctl_dist * rate);
ALsizei i;
float r;
nfc->g = 1.0f;
nfc->coeffs[0] = 1.0f;
/* NOTE: Slight adjustment from the literature to raise the center
* frequency a bit (0.5 -> 1.0).
*/
r = 1.0f * w0;
for(i = 0; i < (order-1);i += 2)
{
float b_10 = B[order][i ] * r;
float b_11 = B[order][i+1] * r * r;
float g_1 = 1.0f + b_10 + b_11;
nfc->b[i] = b_10;
nfc->b[i + 1] = b_11;
nfc->coeffs[0] *= g_1;
nfc->coeffs[i+1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1;
nfc->coeffs[i+2] = (4.0f * b_11) / g_1;
}
if(i < order)
{
float b_00 = B[order][i] * r;
float g_0 = 1.0f + b_00;
nfc->b[i] = b_00;
nfc->coeffs[0] *= g_0;
nfc->coeffs[i+1] = (2.0f * b_00) / g_0;
}
r = 1.0f * w1;
for(i = 0;i < (order-1);i += 2)
{
float b_10 = B[order][i ] * r;
float b_11 = B[order][i+1] * r * r;
float g_1 = 1.0f + b_10 + b_11;
nfc->g /= g_1;
nfc->coeffs[0] /= g_1;
nfc->coeffs[order+i+1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1;
nfc->coeffs[order+i+2] = (4.0f * b_11) / g_1;
}
if(i < order)
{
float b_00 = B[order][i] * r;
float g_0 = 1.0f + b_00;
nfc->g /= g_0;
nfc->coeffs[0] /= g_0;
nfc->coeffs[order+i+1] = (2.0f * b_00) / g_0;
}
for(i = 0; i < MAX_AMBI_ORDER; i++)
nfc->history[i] = 0.0f;
}
static void NfcFilterAdjust(NfcFilter *nfc, const float distance)
{
int i;
nfc->coeffs[0] = nfc->g;
for(i = 0;i < (nfc->order-1);i += 2)
{
float b_10 = nfc->b[i] / distance;
float b_11 = nfc->b[i+1] / (distance * distance);
float g_1 = 1.0f + b_10 + b_11;
nfc->coeffs[0] *= g_1;
nfc->coeffs[i+1] = ((2.0f * b_10) + (4.0f * b_11)) / g_1;
nfc->coeffs[i+2] = (4.0f * b_11) / g_1;
}
if(i < nfc->order)
{
float b_00 = nfc->b[i] / distance;
float g_0 = 1.0f + b_00;
nfc->coeffs[0] *= g_0;
nfc->coeffs[i+1] = (2.0f * b_00) / g_0;
}
}
static float NfcFilterProcess(const float in, NfcFilter *nfc)
{
int i;
float out = in * nfc->coeffs[0];
for(i = 0;i < (nfc->order-1);i += 2)
{
float y = out - (nfc->coeffs[nfc->order+i+1] * nfc->history[i]) -
(nfc->coeffs[nfc->order+i+2] * nfc->history[i+1]) + 1.0e-30f;
out = y + (nfc->coeffs[i+1]*nfc->history[i]) + (nfc->coeffs[i+2]*nfc->history[i+1]);
nfc->history[i+1] += nfc->history[i];
nfc->history[i] += y;
}
if(i < nfc->order)
{
float y = out - (nfc->coeffs[nfc->order+i+1] * nfc->history[i]) + 1.0e-30f;
out = y + (nfc->coeffs[i+1] * nfc->history[i]);
nfc->history[i] += y;
}
return out;
}
#endif
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#ifndef FILTER_NFC_H
#define FILTER_NFC_H
struct NfcFilter1 {
float base_gain, gain;
float b1, a1;
float z[1];
};
struct NfcFilter2 {
float base_gain, gain;
float b1, b2, a1, a2;
float z[2];
};
struct NfcFilter3 {
float base_gain, gain;
float b1, b2, b3, a1, a2, a3;
float z[3];
};
typedef struct NfcFilter {
struct NfcFilter1 first;
struct NfcFilter2 second;
struct NfcFilter3 third;
} NfcFilter;
/* NOTE:
* w0 = speed_of_sound / (source_distance * sample_rate);
* w1 = speed_of_sound / (control_distance * sample_rate);
*
* Generally speaking, the control distance should be approximately the average
* speaker distance, or based on the reference delay if outputing NFC-HOA. It
* must not be negative, 0, or infinite. The source distance should not be too
* small relative to the control distance.
*/
void NfcFilterCreate(NfcFilter *nfc, const float w0, const float w1);
void NfcFilterAdjust(NfcFilter *nfc, const float w0);
/* Near-field control filter for first-order ambisonic channels (1-3). */
void NfcFilterProcess1(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count);
/* Near-field control filter for second-order ambisonic channels (4-8). */
void NfcFilterProcess2(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count);
/* Near-field control filter for third-order ambisonic channels (9-15). */
void NfcFilterProcess3(NfcFilter *nfc, float *restrict dst, const float *restrict src, const int count);
#endif /* FILTER_NFC_H */
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#include "config.h"
#include "splitter.h"
#include "math_defs.h"
void bandsplit_init(BandSplitter *splitter, ALfloat f0norm)
{
ALfloat w = f0norm * F_TAU;
ALfloat cw = cosf(w);
if(cw > FLT_EPSILON)
splitter->coeff = (sinf(w) - 1.0f) / cw;
else
splitter->coeff = cw * -0.5f;
splitter->lp_z1 = 0.0f;
splitter->lp_z2 = 0.0f;
splitter->hp_z1 = 0.0f;
}
void bandsplit_clear(BandSplitter *splitter)
{
splitter->lp_z1 = 0.0f;
splitter->lp_z2 = 0.0f;
splitter->hp_z1 = 0.0f;
}
void bandsplit_process(BandSplitter *splitter, ALfloat *restrict hpout, ALfloat *restrict lpout,
const ALfloat *input, ALsizei count)
{
ALfloat lp_coeff, hp_coeff, lp_y, hp_y, d;
ALfloat lp_z1, lp_z2, hp_z1;
ALsizei i;
ASSUME(count > 0);
hp_coeff = splitter->coeff;
lp_coeff = splitter->coeff*0.5f + 0.5f;
lp_z1 = splitter->lp_z1;
lp_z2 = splitter->lp_z2;
hp_z1 = splitter->hp_z1;
for(i = 0;i < count;i++)
{
ALfloat in = input[i];
/* Low-pass sample processing. */
d = (in - lp_z1) * lp_coeff;
lp_y = lp_z1 + d;
lp_z1 = lp_y + d;
d = (lp_y - lp_z2) * lp_coeff;
lp_y = lp_z2 + d;
lp_z2 = lp_y + d;
lpout[i] = lp_y;
/* All-pass sample processing. */
hp_y = in*hp_coeff + hp_z1;
hp_z1 = in - hp_y*hp_coeff;
/* High-pass generated from removing low-passed output. */
hpout[i] = hp_y - lp_y;
}
splitter->lp_z1 = lp_z1;
splitter->lp_z2 = lp_z2;
splitter->hp_z1 = hp_z1;
}
void splitterap_init(SplitterAllpass *splitter, ALfloat f0norm)
{
ALfloat w = f0norm * F_TAU;
ALfloat cw = cosf(w);
if(cw > FLT_EPSILON)
splitter->coeff = (sinf(w) - 1.0f) / cw;
else
splitter->coeff = cw * -0.5f;
splitter->z1 = 0.0f;
}
void splitterap_clear(SplitterAllpass *splitter)
{
splitter->z1 = 0.0f;
}
void splitterap_process(SplitterAllpass *splitter, ALfloat *restrict samples, ALsizei count)
{
ALfloat coeff, in, out;
ALfloat z1;
ALsizei i;
ASSUME(count > 0);
coeff = splitter->coeff;
z1 = splitter->z1;
for(i = 0;i < count;i++)
{
in = samples[i];
out = in*coeff + z1;
z1 = in - out*coeff;
samples[i] = out;
}
splitter->z1 = z1;
}
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#ifndef FILTER_SPLITTER_H
#define FILTER_SPLITTER_H
#include "alMain.h"
/* Band splitter. Splits a signal into two phase-matching frequency bands. */
typedef struct BandSplitter {
ALfloat coeff;
ALfloat lp_z1;
ALfloat lp_z2;
ALfloat hp_z1;
} BandSplitter;
void bandsplit_init(BandSplitter *splitter, ALfloat f0norm);
void bandsplit_clear(BandSplitter *splitter);
void bandsplit_process(BandSplitter *splitter, ALfloat *restrict hpout, ALfloat *restrict lpout,
const ALfloat *input, ALsizei count);
/* The all-pass portion of the band splitter. Applies the same phase shift
* without splitting the signal.
*/
typedef struct SplitterAllpass {
ALfloat coeff;
ALfloat z1;
} SplitterAllpass;
void splitterap_init(SplitterAllpass *splitter, ALfloat f0norm);
void splitterap_clear(SplitterAllpass *splitter);
void splitterap_process(SplitterAllpass *splitter, ALfloat *restrict samples, ALsizei count);
typedef struct FrontStablizer {
SplitterAllpass APFilter[MAX_OUTPUT_CHANNELS];
BandSplitter LFilter, RFilter;
alignas(16) ALfloat LSplit[2][BUFFERSIZE];
alignas(16) ALfloat RSplit[2][BUFFERSIZE];
} FrontStablizer;
#endif /* FILTER_SPLITTER_H */
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#ifndef FPU_MODES_H
#define FPU_MODES_H
#ifdef HAVE_FENV_H
#include <fenv.h>
#endif
typedef struct FPUCtl {
#if defined(__GNUC__) && defined(HAVE_SSE)
unsigned int sse_state;
#elif defined(HAVE___CONTROL87_2)
unsigned int state;
unsigned int sse_state;
#elif defined(HAVE__CONTROLFP)
unsigned int state;
#endif
} FPUCtl;
void SetMixerFPUMode(FPUCtl *ctl);
void RestoreFPUMode(const FPUCtl *ctl);
#ifdef __GNUC__
/* Use an alternate macro set with GCC to avoid accidental continue or break
* statements within the mixer mode.
*/
#define START_MIXER_MODE() __extension__({ FPUCtl _oldMode; SetMixerFPUMode(&_oldMode)
#define END_MIXER_MODE() RestoreFPUMode(&_oldMode); })
#else
#define START_MIXER_MODE() do { FPUCtl _oldMode; SetMixerFPUMode(&_oldMode)
#define END_MIXER_MODE() RestoreFPUMode(&_oldMode); } while(0)
#endif
#define LEAVE_MIXER_MODE() RestoreFPUMode(&_oldMode)
#endif /* FPU_MODES_H */
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#ifndef ALC_HRTF_H
#define ALC_HRTF_H
#include "AL/al.h"
#include "AL/alc.h"
#include "alMain.h"
#include "alstring.h"
#include "atomic.h"
#define HRTF_HISTORY_BITS (6)
#define HRTF_HISTORY_LENGTH (1<<HRTF_HISTORY_BITS)
#define HRTF_HISTORY_MASK (HRTF_HISTORY_LENGTH-1)
#define HRIR_BITS (7)
#define HRIR_LENGTH (1<<HRIR_BITS)
#define HRIR_MASK (HRIR_LENGTH-1)
struct HrtfEntry;
struct Hrtf {
RefCount ref;
ALuint sampleRate;
ALsizei irSize;
ALfloat distance;
ALubyte evCount;
const ALubyte *azCount;
const ALushort *evOffset;
const ALfloat (*coeffs)[2];
const ALubyte (*delays)[2];
};
typedef struct HrtfState {
alignas(16) ALfloat History[HRTF_HISTORY_LENGTH];
alignas(16) ALfloat Values[HRIR_LENGTH][2];
} HrtfState;
typedef struct HrtfParams {
alignas(16) ALfloat Coeffs[HRIR_LENGTH][2];
ALsizei Delay[2];
ALfloat Gain;
} HrtfParams;
typedef struct DirectHrtfState {
/* HRTF filter state for dry buffer content */
ALsizei Offset;
ALsizei IrSize;
struct {
alignas(16) ALfloat Values[HRIR_LENGTH][2];
alignas(16) ALfloat Coeffs[HRIR_LENGTH][2];
} Chan[];
} DirectHrtfState;
struct AngularPoint {
ALfloat Elev;
ALfloat Azim;
};
void FreeHrtfs(void);
vector_EnumeratedHrtf EnumerateHrtf(const_al_string devname);
void FreeHrtfList(vector_EnumeratedHrtf *list);
struct Hrtf *GetLoadedHrtf(struct HrtfEntry *entry);
void Hrtf_IncRef(struct Hrtf *hrtf);
void Hrtf_DecRef(struct Hrtf *hrtf);
void GetHrtfCoeffs(const struct Hrtf *Hrtf, ALfloat elevation, ALfloat azimuth, ALfloat spread, ALfloat (*coeffs)[2], ALsizei *delays);
/**
* Produces HRTF filter coefficients for decoding B-Format, given a set of
* virtual speaker positions, a matching decoding matrix, and per-order high-
* frequency gains for the decoder. The calculated impulse responses are
* ordered and scaled according to the matrix input.
*/
void BuildBFormatHrtf(const struct Hrtf *Hrtf, DirectHrtfState *state, ALsizei NumChannels, const struct AngularPoint *AmbiPoints, const ALfloat (*restrict AmbiMatrix)[MAX_AMBI_COEFFS], ALsizei AmbiCount, const ALfloat *restrict AmbiOrderHFGain);
#endif /* ALC_HRTF_H */
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#ifndef LOGGING_H
#define LOGGING_H
#include <stdio.h>
#ifdef __GNUC__
#define DECL_FORMAT(x, y, z) __attribute__((format(x, (y), (z))))
#else
#define DECL_FORMAT(x, y, z)
#endif
#ifdef __cplusplus
extern "C" {
#endif
extern FILE *LogFile;
#if defined(__GNUC__) && !defined(_WIN32)
#define AL_PRINT(T, MSG, ...) fprintf(LogFile, "AL lib: %s %s: "MSG, T, __FUNCTION__ , ## __VA_ARGS__)
#else
void al_print(const char *type, const char *func, const char *fmt, ...) DECL_FORMAT(printf, 3,4);
#define AL_PRINT(T, ...) al_print((T), __FUNCTION__, __VA_ARGS__)
#endif
#ifdef __ANDROID__
#include <android/log.h>
#define LOG_ANDROID(T, MSG, ...) __android_log_print(T, "openal", "AL lib: %s: "MSG, __FUNCTION__ , ## __VA_ARGS__)
#else
#define LOG_ANDROID(T, MSG, ...) ((void)0)
#endif
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__); \
LOG_ANDROID(ANDROID_LOG_DEBUG, __VA_ARGS__); \
} while(0)
#define WARN(...) do { \
if(LogLevel >= LogWarning) \
AL_PRINT("(WW)", __VA_ARGS__); \
LOG_ANDROID(ANDROID_LOG_WARN, __VA_ARGS__); \
} while(0)
#define ERR(...) do { \
if(LogLevel >= LogError) \
AL_PRINT("(EE)", __VA_ARGS__); \
LOG_ANDROID(ANDROID_LOG_ERROR, __VA_ARGS__); \
} while(0)
#ifdef __cplusplus
} /* extern "C" */
#endif
#endif /* LOGGING_H */
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#include "config.h"
#include <math.h>
#include "mastering.h"
#include "alu.h"
#include "almalloc.h"
#include "static_assert.h"
/* These structures assume BUFFERSIZE is a power of 2. */
static_assert((BUFFERSIZE & (BUFFERSIZE-1)) == 0, "BUFFERSIZE is not a power of 2");
typedef struct SlidingHold {
ALfloat Values[BUFFERSIZE];
ALsizei Expiries[BUFFERSIZE];
ALsizei LowerIndex;
ALsizei UpperIndex;
ALsizei Length;
} SlidingHold;
/* General topology and basic automation was based on the following paper:
*
* D. Giannoulis, M. Massberg and J. D. Reiss,
* "Parameter Automation in a Dynamic Range Compressor,"
* Journal of the Audio Engineering Society, v61 (10), Oct. 2013
*
* Available (along with supplemental reading) at:
*
* http://c4dm.eecs.qmul.ac.uk/audioengineering/compressors/
*/
typedef struct Compressor {
ALsizei NumChans;
ALuint SampleRate;
struct {
ALuint Knee : 1;
ALuint Attack : 1;
ALuint Release : 1;
ALuint PostGain : 1;
ALuint Declip : 1;
} Auto;
ALsizei LookAhead;
ALfloat PreGain;
ALfloat PostGain;
ALfloat Threshold;
ALfloat Slope;
ALfloat Knee;
ALfloat Attack;
ALfloat Release;
alignas(16) ALfloat SideChain[2*BUFFERSIZE];
alignas(16) ALfloat CrestFactor[BUFFERSIZE];
SlidingHold *Hold;
ALfloat (*Delay)[BUFFERSIZE];
ALsizei DelayIndex;
ALfloat CrestCoeff;
ALfloat GainEstimate;
ALfloat AdaptCoeff;
ALfloat LastPeakSq;
ALfloat LastRmsSq;
ALfloat LastRelease;
ALfloat LastAttack;
ALfloat LastGainDev;
} Compressor;
/* This sliding hold follows the input level with an instant attack and a
* fixed duration hold before an instant release to the next highest level.
* It is a sliding window maximum (descending maxima) implementation based on
* Richard Harter's ascending minima algorithm available at:
*
* http://www.richardhartersworld.com/cri/2001/slidingmin.html
*/
static ALfloat UpdateSlidingHold(SlidingHold *Hold, const ALsizei i, const ALfloat in)
{
const ALsizei mask = BUFFERSIZE - 1;
const ALsizei length = Hold->Length;
ALfloat *restrict values = Hold->Values;
ALsizei *restrict expiries = Hold->Expiries;
ALsizei lowerIndex = Hold->LowerIndex;
ALsizei upperIndex = Hold->UpperIndex;
if(i >= expiries[upperIndex])
upperIndex = (upperIndex + 1) & mask;
if(in >= values[upperIndex])
{
values[upperIndex] = in;
expiries[upperIndex] = i + length;
lowerIndex = upperIndex;
}
else
{
do {
do {
if(!(in >= values[lowerIndex]))
goto found_place;
} while(lowerIndex--);
lowerIndex = mask;
} while(1);
found_place:
lowerIndex = (lowerIndex + 1) & mask;
values[lowerIndex] = in;
expiries[lowerIndex] = i + length;
}
Hold->LowerIndex = lowerIndex;
Hold->UpperIndex = upperIndex;
return values[upperIndex];
}
static void ShiftSlidingHold(SlidingHold *Hold, const ALsizei n)
{
const ALsizei lowerIndex = Hold->LowerIndex;
ALsizei *restrict expiries = Hold->Expiries;
ALsizei i = Hold->UpperIndex;
if(lowerIndex < i)
{
for(;i < BUFFERSIZE;i++)
expiries[i] -= n;
i = 0;
}
for(;i < lowerIndex;i++)
expiries[i] -= n;
expiries[i] -= n;
}
/* Multichannel compression is linked via the absolute maximum of all
* channels.
*/
static void LinkChannels(Compressor *Comp, const ALsizei SamplesToDo, ALfloat (*restrict OutBuffer)[BUFFERSIZE])
{
const ALsizei index = Comp->LookAhead;
const ALsizei numChans = Comp->NumChans;
ALfloat *restrict sideChain = Comp->SideChain;
ALsizei c, i;
ASSUME(SamplesToDo > 0);
ASSUME(numChans > 0);
for(i = 0;i < SamplesToDo;i++)
sideChain[index + i] = 0.0f;
for(c = 0;c < numChans;c++)
{
ALsizei offset = index;
for(i = 0;i < SamplesToDo;i++)
{
sideChain[offset] = maxf(sideChain[offset], fabsf(OutBuffer[c][i]));
++offset;
}
}
}
/* This calculates the squared crest factor of the control signal for the
* basic automation of the attack/release times. As suggested by the paper,
* it uses an instantaneous squared peak detector and a squared RMS detector
* both with 200ms release times.
*/
static void CrestDetector(Compressor *Comp, const ALsizei SamplesToDo)
{
const ALfloat a_crest = Comp->CrestCoeff;
const ALsizei index = Comp->LookAhead;
const ALfloat *restrict sideChain = Comp->SideChain;
ALfloat *restrict crestFactor = Comp->CrestFactor;
ALfloat y2_peak = Comp->LastPeakSq;
ALfloat y2_rms = Comp->LastRmsSq;
ALsizei i;
ASSUME(SamplesToDo > 0);
for(i = 0;i < SamplesToDo;i++)
{
ALfloat x_abs = sideChain[index + i];
ALfloat x2 = maxf(0.000001f, x_abs * x_abs);
y2_peak = maxf(x2, lerp(x2, y2_peak, a_crest));
y2_rms = lerp(x2, y2_rms, a_crest);
crestFactor[i] = y2_peak / y2_rms;
}
Comp->LastPeakSq = y2_peak;
Comp->LastRmsSq = y2_rms;
}
/* The side-chain starts with a simple peak detector (based on the absolute
* value of the incoming signal) and performs most of its operations in the
* log domain.
*/
static void PeakDetector(Compressor *Comp, const ALsizei SamplesToDo)
{
const ALsizei index = Comp->LookAhead;
ALfloat *restrict sideChain = Comp->SideChain;
ALsizei i;
ASSUME(SamplesToDo > 0);
for(i = 0;i < SamplesToDo;i++)
{
const ALuint offset = index + i;
const ALfloat x_abs = sideChain[offset];
sideChain[offset] = logf(maxf(0.000001f, x_abs));
}
}
/* An optional hold can be used to extend the peak detector so it can more
* solidly detect fast transients. This is best used when operating as a
* limiter.
*/
static void PeakHoldDetector(Compressor *Comp, const ALsizei SamplesToDo)
{
const ALsizei index = Comp->LookAhead;
ALfloat *restrict sideChain = Comp->SideChain;
SlidingHold *hold = Comp->Hold;
ALsizei i;
ASSUME(SamplesToDo > 0);
for(i = 0;i < SamplesToDo;i++)
{
const ALsizei offset = index + i;
const ALfloat x_abs = sideChain[offset];
const ALfloat x_G = logf(maxf(0.000001f, x_abs));
sideChain[offset] = UpdateSlidingHold(hold, i, x_G);
}
ShiftSlidingHold(hold, SamplesToDo);
}
/* This is the heart of the feed-forward compressor. It operates in the log
* domain (to better match human hearing) and can apply some basic automation
* to knee width, attack/release times, make-up/post gain, and clipping
* reduction.
*/
static void GainCompressor(Compressor *Comp, const ALsizei SamplesToDo)
{
const bool autoKnee = Comp->Auto.Knee;
const bool autoAttack = Comp->Auto.Attack;
const bool autoRelease = Comp->Auto.Release;
const bool autoPostGain = Comp->Auto.PostGain;
const bool autoDeclip = Comp->Auto.Declip;
const ALsizei lookAhead = Comp->LookAhead;
const ALfloat threshold = Comp->Threshold;
const ALfloat slope = Comp->Slope;
const ALfloat attack = Comp->Attack;
const ALfloat release = Comp->Release;
const ALfloat c_est = Comp->GainEstimate;
const ALfloat a_adp = Comp->AdaptCoeff;
const ALfloat *restrict crestFactor = Comp->CrestFactor;
ALfloat *restrict sideChain = Comp->SideChain;
ALfloat postGain = Comp->PostGain;
ALfloat knee = Comp->Knee;
ALfloat t_att = attack;
ALfloat t_rel = release - attack;
ALfloat a_att = expf(-1.0f / t_att);
ALfloat a_rel = expf(-1.0f / t_rel);
ALfloat y_1 = Comp->LastRelease;
ALfloat y_L = Comp->LastAttack;
ALfloat c_dev = Comp->LastGainDev;
ALsizei i;
ASSUME(SamplesToDo > 0);
for(i = 0;i < SamplesToDo;i++)
{
const ALfloat y2_crest = crestFactor[i];
const ALfloat x_G = sideChain[lookAhead + i];
const ALfloat x_over = x_G - threshold;
ALfloat knee_h;
ALfloat y_G;
ALfloat x_L;
if(autoKnee)
knee = maxf(0.0f, 2.5f * (c_dev + c_est));
knee_h = 0.5f * knee;
/* This is the gain computer. It applies a static compression curve
* to the control signal.
*/
if(x_over <= -knee_h)
y_G = 0.0f;
else if(fabsf(x_over) < knee_h)
y_G = (x_over + knee_h) * (x_over + knee_h) / (2.0f * knee);
else
y_G = x_over;
x_L = -slope * y_G;
if(autoAttack)
{
t_att = 2.0f * attack / y2_crest;
a_att = expf(-1.0f / t_att);
}
if(autoRelease)
{
t_rel = 2.0f * release / y2_crest - t_att;
a_rel = expf(-1.0f / t_rel);
}
/* Gain smoothing (ballistics) is done via a smooth decoupled peak
* detector. The attack time is subtracted from the release time
* above to compensate for the chained operating mode.
*/
y_1 = maxf(x_L, lerp(x_L, y_1, a_rel));
y_L = lerp(y_1, y_L, a_att);
/* Knee width and make-up gain automation make use of a smoothed
* measurement of deviation between the control signal and estimate.
* The estimate is also used to bias the measurement to hot-start its
* average.
*/
c_dev = lerp(-y_L - c_est, c_dev, a_adp);
if(autoPostGain)
{
/* Clipping reduction is only viable when make-up gain is being
* automated. It modifies the deviation to further attenuate the
* control signal when clipping is detected. The adaptation
* time is sufficiently long enough to suppress further clipping
* at the same output level.
*/
if(autoDeclip)
c_dev = maxf(c_dev, sideChain[i] - y_L - threshold - c_est);
postGain = -(c_dev + c_est);
}
sideChain[i] = expf(postGain - y_L);
}
Comp->LastRelease = y_1;
Comp->LastAttack = y_L;
Comp->LastGainDev = c_dev;
}
/* Combined with the hold time, a look-ahead delay can improve handling of
* fast transients by allowing the envelope time to converge prior to
* reaching the offending impulse. This is best used when operating as a
* limiter.
*/
static void SignalDelay(Compressor *Comp, const ALsizei SamplesToDo, ALfloat (*restrict OutBuffer)[BUFFERSIZE])
{
const ALsizei mask = BUFFERSIZE - 1;
const ALsizei numChans = Comp->NumChans;
const ALsizei indexIn = Comp->DelayIndex;
const ALsizei indexOut = Comp->DelayIndex - Comp->LookAhead;
ALfloat (*restrict delay)[BUFFERSIZE] = Comp->Delay;
ALsizei c, i;
ASSUME(SamplesToDo > 0);
ASSUME(numChans > 0);
for(c = 0;c < numChans;c++)
{
for(i = 0;i < SamplesToDo;i++)
{
ALfloat sig = OutBuffer[c][i];
OutBuffer[c][i] = delay[c][(indexOut + i) & mask];
delay[c][(indexIn + i) & mask] = sig;
}
}
Comp->DelayIndex = (indexIn + SamplesToDo) & mask;
}
/* The compressor is initialized with the following settings:
*
* NumChans - Number of channels to process.
* SampleRate - Sample rate to process.
* AutoKnee - Whether to automate the knee width parameter.
* AutoAttack - Whether to automate the attack time parameter.
* AutoRelease - Whether to automate the release time parameter.
* AutoPostGain - Whether to automate the make-up (post) gain parameter.
* AutoDeclip - Whether to automate clipping reduction. Ignored when
* not automating make-up gain.
* LookAheadTime - Look-ahead time (in seconds).
* HoldTime - Peak hold-time (in seconds).
* PreGainDb - Gain applied before detection (in dB).
* PostGainDb - Make-up gain applied after compression (in dB).
* ThresholdDb - Triggering threshold (in dB).
* Ratio - Compression ratio (x:1). Set to INFINITY for true
* limiting. Ignored when automating knee width.
* KneeDb - Knee width (in dB). Ignored when automating knee
* width.
* AttackTimeMin - Attack time (in seconds). Acts as a maximum when
* automating attack time.
* ReleaseTimeMin - Release time (in seconds). Acts as a maximum when
* automating release time.
*/
Compressor* CompressorInit(const ALsizei NumChans, const ALuint SampleRate,
const ALboolean AutoKnee, const ALboolean AutoAttack,
const ALboolean AutoRelease, const ALboolean AutoPostGain,
const ALboolean AutoDeclip, const ALfloat LookAheadTime,
const ALfloat HoldTime, const ALfloat PreGainDb,
const ALfloat PostGainDb, const ALfloat ThresholdDb,
const ALfloat Ratio, const ALfloat KneeDb,
const ALfloat AttackTime, const ALfloat ReleaseTime)
{
Compressor *Comp;
ALsizei lookAhead;
ALsizei hold;
size_t size;
lookAhead = (ALsizei)clampf(roundf(LookAheadTime*SampleRate), 0.0f, BUFFERSIZE-1);
hold = (ALsizei)clampf(roundf(HoldTime*SampleRate), 0.0f, BUFFERSIZE-1);
/* The sliding hold implementation doesn't handle a length of 1. A 1-sample
* hold is useless anyway, it would only ever give back what was just given
* to it.
*/
if(hold == 1)
hold = 0;
size = sizeof(*Comp);
if(lookAhead > 0)
{
size += sizeof(*Comp->Delay) * NumChans;
if(hold > 0)
size += sizeof(*Comp->Hold);
}
Comp = al_calloc(16, size);
Comp->NumChans = NumChans;
Comp->SampleRate = SampleRate;
Comp->Auto.Knee = AutoKnee;
Comp->Auto.Attack = AutoAttack;
Comp->Auto.Release = AutoRelease;
Comp->Auto.PostGain = AutoPostGain;
Comp->Auto.Declip = AutoPostGain && AutoDeclip;
Comp->LookAhead = lookAhead;
Comp->PreGain = powf(10.0f, PreGainDb / 20.0f);
Comp->PostGain = PostGainDb * logf(10.0f) / 20.0f;
Comp->Threshold = ThresholdDb * logf(10.0f) / 20.0f;
Comp->Slope = 1.0f / maxf(1.0f, Ratio) - 1.0f;
Comp->Knee = maxf(0.0f, KneeDb * logf(10.0f) / 20.0f);
Comp->Attack = maxf(1.0f, AttackTime * SampleRate);
Comp->Release = maxf(1.0f, ReleaseTime * SampleRate);
/* Knee width automation actually treats the compressor as a limiter. By
* varying the knee width, it can effectively be seen as applying
* compression over a wide range of ratios.
*/
if(AutoKnee)
Comp->Slope = -1.0f;
if(lookAhead > 0)
{
if(hold > 0)
{
Comp->Hold = (SlidingHold*)(Comp + 1);
Comp->Hold->Values[0] = -INFINITY;
Comp->Hold->Expiries[0] = hold;
Comp->Hold->Length = hold;
Comp->Delay = (ALfloat(*)[])(Comp->Hold + 1);
}
else
{
Comp->Delay = (ALfloat(*)[])(Comp + 1);
}
}
Comp->CrestCoeff = expf(-1.0f / (0.200f * SampleRate)); // 200ms
Comp->GainEstimate = Comp->Threshold * -0.5f * Comp->Slope;
Comp->AdaptCoeff = expf(-1.0f / (2.0f * SampleRate)); // 2s
return Comp;
}
void ApplyCompression(Compressor *Comp, const ALsizei SamplesToDo, ALfloat (*restrict OutBuffer)[BUFFERSIZE])
{
const ALsizei numChans = Comp->NumChans;
const ALfloat preGain = Comp->PreGain;
ALfloat *restrict sideChain;
ALsizei c, i;
ASSUME(SamplesToDo > 0);
ASSUME(numChans > 0);
if(preGain != 1.0f)
{
for(c = 0;c < numChans;c++)
{
for(i = 0;i < SamplesToDo;i++)
OutBuffer[c][i] *= preGain;
}
}
LinkChannels(Comp, SamplesToDo, OutBuffer);
if(Comp->Auto.Attack || Comp->Auto.Release)
CrestDetector(Comp, SamplesToDo);
if(Comp->Hold)
PeakHoldDetector(Comp, SamplesToDo);
else
PeakDetector(Comp, SamplesToDo);
GainCompressor(Comp, SamplesToDo);
if(Comp->Delay)
SignalDelay(Comp, SamplesToDo, OutBuffer);
sideChain = Comp->SideChain;
for(c = 0;c < numChans;c++)
{
for(i = 0;i < SamplesToDo;i++)
OutBuffer[c][i] *= sideChain[i];
}
memmove(sideChain, sideChain+SamplesToDo, Comp->LookAhead*sizeof(ALfloat));
}
ALsizei GetCompressorLookAhead(const Compressor *Comp)
{ return Comp->LookAhead; }
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#ifndef MASTERING_H
#define MASTERING_H
#include "AL/al.h"
/* For BUFFERSIZE. */
#include "alMain.h"
struct Compressor;
/* The compressor is initialized with the following settings:
*
* NumChans - Number of channels to process.
* SampleRate - Sample rate to process.
* AutoKnee - Whether to automate the knee width parameter.
* AutoAttack - Whether to automate the attack time parameter.
* AutoRelease - Whether to automate the release time parameter.
* AutoPostGain - Whether to automate the make-up (post) gain parameter.
* AutoDeclip - Whether to automate clipping reduction. Ignored when
* not automating make-up gain.
* LookAheadTime - Look-ahead time (in seconds).
* HoldTime - Peak hold-time (in seconds).
* PreGainDb - Gain applied before detection (in dB).
* PostGainDb - Make-up gain applied after compression (in dB).
* ThresholdDb - Triggering threshold (in dB).
* Ratio - Compression ratio (x:1). Set to INFINIFTY for true
* limiting. Ignored when automating knee width.
* KneeDb - Knee width (in dB). Ignored when automating knee
* width.
* AttackTimeMin - Attack time (in seconds). Acts as a maximum when
* automating attack time.
* ReleaseTimeMin - Release time (in seconds). Acts as a maximum when
* automating release time.
*/
struct Compressor* CompressorInit(const ALsizei NumChans, const ALuint SampleRate,
const ALboolean AutoKnee, const ALboolean AutoAttack,
const ALboolean AutoRelease, const ALboolean AutoPostGain,
const ALboolean AutoDeclip, const ALfloat LookAheadTime,
const ALfloat HoldTime, const ALfloat PreGainDb,
const ALfloat PostGainDb, const ALfloat ThresholdDb,
const ALfloat Ratio, const ALfloat KneeDb,
const ALfloat AttackTime, const ALfloat ReleaseTime);
void ApplyCompression(struct Compressor *Comp, const ALsizei SamplesToDo,
ALfloat (*restrict OutBuffer)[BUFFERSIZE]);
ALsizei GetCompressorLookAhead(const struct Compressor *Comp);
#endif /* MASTERING_H */
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#ifndef MIXER_DEFS_H
#define MIXER_DEFS_H
#include "AL/alc.h"
#include "AL/al.h"
#include "alMain.h"
#include "alu.h"
struct MixGains;
struct MixHrtfParams;
struct HrtfState;
/* C resamplers */
const ALfloat *Resample_copy_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen);
const ALfloat *Resample_point_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen);
const ALfloat *Resample_lerp_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen);
const ALfloat *Resample_cubic_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen);
const ALfloat *Resample_bsinc_C(const InterpState *state, const ALfloat *restrict src, ALsizei frac, ALint increment, ALfloat *restrict dst, ALsizei dstlen);
/* C mixers */
void MixHrtf_C(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
const ALfloat *data, ALsizei Offset, ALsizei OutPos,
const ALsizei IrSize, struct MixHrtfParams *hrtfparams,
struct HrtfState *hrtfstate, ALsizei BufferSize);
void MixHrtfBlend_C(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
const ALfloat *data, ALsizei Offset, ALsizei OutPos,
const ALsizei IrSize, const HrtfParams *oldparams,
MixHrtfParams *newparams, HrtfState *hrtfstate,
ALsizei BufferSize);
void MixDirectHrtf_C(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
const ALfloat *data, ALsizei Offset, const ALsizei IrSize,
const ALfloat (*restrict Coeffs)[2], ALfloat (*restrict Values)[2],
ALsizei BufferSize);
void Mix_C(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE],
ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
ALsizei BufferSize);
void MixRow_C(ALfloat *OutBuffer, const ALfloat *Gains,
const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans,
ALsizei InPos, ALsizei BufferSize);
/* SSE mixers */
void MixHrtf_SSE(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
const ALfloat *data, ALsizei Offset, ALsizei OutPos,
const ALsizei IrSize, struct MixHrtfParams *hrtfparams,
struct HrtfState *hrtfstate, ALsizei BufferSize);
void MixHrtfBlend_SSE(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
const ALfloat *data, ALsizei Offset, ALsizei OutPos,
const ALsizei IrSize, const HrtfParams *oldparams,
MixHrtfParams *newparams, HrtfState *hrtfstate,
ALsizei BufferSize);
void MixDirectHrtf_SSE(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
const ALfloat *data, ALsizei Offset, const ALsizei IrSize,
const ALfloat (*restrict Coeffs)[2], ALfloat (*restrict Values)[2],
ALsizei BufferSize);
void Mix_SSE(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE],
ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
ALsizei BufferSize);
void MixRow_SSE(ALfloat *OutBuffer, const ALfloat *Gains,
const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans,
ALsizei InPos, ALsizei BufferSize);
/* SSE resamplers */
inline void InitiatePositionArrays(ALsizei frac, ALint increment, ALsizei *restrict frac_arr, ALsizei *restrict pos_arr, ALsizei size)
{
ALsizei i;
pos_arr[0] = 0;
frac_arr[0] = frac;
for(i = 1;i < size;i++)
{
ALint frac_tmp = frac_arr[i-1] + increment;
pos_arr[i] = pos_arr[i-1] + (frac_tmp>>FRACTIONBITS);
frac_arr[i] = frac_tmp&FRACTIONMASK;
}
}
const ALfloat *Resample_lerp_SSE2(const InterpState *state, const ALfloat *restrict src,
ALsizei frac, ALint increment, ALfloat *restrict dst,
ALsizei numsamples);
const ALfloat *Resample_lerp_SSE41(const InterpState *state, const ALfloat *restrict src,
ALsizei frac, ALint increment, ALfloat *restrict dst,
ALsizei numsamples);
const ALfloat *Resample_bsinc_SSE(const InterpState *state, const ALfloat *restrict src,
ALsizei frac, ALint increment, ALfloat *restrict dst,
ALsizei dstlen);
/* Neon mixers */
void MixHrtf_Neon(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
const ALfloat *data, ALsizei Offset, ALsizei OutPos,
const ALsizei IrSize, struct MixHrtfParams *hrtfparams,
struct HrtfState *hrtfstate, ALsizei BufferSize);
void MixHrtfBlend_Neon(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
const ALfloat *data, ALsizei Offset, ALsizei OutPos,
const ALsizei IrSize, const HrtfParams *oldparams,
MixHrtfParams *newparams, HrtfState *hrtfstate,
ALsizei BufferSize);
void MixDirectHrtf_Neon(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
const ALfloat *data, ALsizei Offset, const ALsizei IrSize,
const ALfloat (*restrict Coeffs)[2], ALfloat (*restrict Values)[2],
ALsizei BufferSize);
void Mix_Neon(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE],
ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
ALsizei BufferSize);
void MixRow_Neon(ALfloat *OutBuffer, const ALfloat *Gains,
const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans,
ALsizei InPos, ALsizei BufferSize);
/* Neon resamplers */
const ALfloat *Resample_lerp_Neon(const InterpState *state, const ALfloat *restrict src,
ALsizei frac, ALint increment, ALfloat *restrict dst,
ALsizei numsamples);
const ALfloat *Resample_bsinc_Neon(const InterpState *state, const ALfloat *restrict src,
ALsizei frac, ALint increment, ALfloat *restrict dst,
ALsizei dstlen);
#endif /* MIXER_DEFS_H */
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#include "config.h"
#include "alMain.h"
#include "alSource.h"
#include "hrtf.h"
#include "align.h"
#include "alu.h"
#include "defs.h"
static inline void ApplyCoeffs(ALsizei Offset, ALfloat (*restrict Values)[2],
const ALsizei irSize,
const ALfloat (*restrict Coeffs)[2],
ALfloat left, ALfloat right);
void MixHrtf(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
const ALfloat *data, ALsizei Offset, ALsizei OutPos,
const ALsizei IrSize, MixHrtfParams *hrtfparams, HrtfState *hrtfstate,
ALsizei BufferSize)
{
const ALfloat (*Coeffs)[2] = ASSUME_ALIGNED(hrtfparams->Coeffs, 16);
const ALsizei Delay[2] = { hrtfparams->Delay[0], hrtfparams->Delay[1] };
const ALfloat gainstep = hrtfparams->GainStep;
const ALfloat gain = hrtfparams->Gain;
ALfloat g, stepcount = 0.0f;
ALfloat left, right;
ALsizei i;
ASSUME(IrSize >= 4);
ASSUME(BufferSize > 0);
LeftOut += OutPos;
RightOut += OutPos;
for(i = 0;i < BufferSize;i++)
{
hrtfstate->History[Offset&HRTF_HISTORY_MASK] = *(data++);
g = gain + gainstep*stepcount;
left = hrtfstate->History[(Offset-Delay[0])&HRTF_HISTORY_MASK]*g;
right = hrtfstate->History[(Offset-Delay[1])&HRTF_HISTORY_MASK]*g;
hrtfstate->Values[(Offset+IrSize-1)&HRIR_MASK][0] = 0.0f;
hrtfstate->Values[(Offset+IrSize-1)&HRIR_MASK][1] = 0.0f;
ApplyCoeffs(Offset, hrtfstate->Values, IrSize, Coeffs, left, right);
*(LeftOut++) += hrtfstate->Values[Offset&HRIR_MASK][0];
*(RightOut++) += hrtfstate->Values[Offset&HRIR_MASK][1];
stepcount += 1.0f;
Offset++;
}
hrtfparams->Gain = gain + gainstep*stepcount;
}
void MixHrtfBlend(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
const ALfloat *data, ALsizei Offset, ALsizei OutPos,
const ALsizei IrSize, const HrtfParams *oldparams,
MixHrtfParams *newparams, HrtfState *hrtfstate,
ALsizei BufferSize)
{
const ALfloat (*OldCoeffs)[2] = ASSUME_ALIGNED(oldparams->Coeffs, 16);
const ALsizei OldDelay[2] = { oldparams->Delay[0], oldparams->Delay[1] };
const ALfloat oldGain = oldparams->Gain;
const ALfloat oldGainStep = -oldGain / (ALfloat)BufferSize;
const ALfloat (*NewCoeffs)[2] = ASSUME_ALIGNED(newparams->Coeffs, 16);
const ALsizei NewDelay[2] = { newparams->Delay[0], newparams->Delay[1] };
const ALfloat newGain = newparams->Gain;
const ALfloat newGainStep = newparams->GainStep;
ALfloat g, stepcount = 0.0f;
ALfloat left, right;
ALsizei i;
ASSUME(IrSize >= 4);
ASSUME(BufferSize > 0);
LeftOut += OutPos;
RightOut += OutPos;
for(i = 0;i < BufferSize;i++)
{
hrtfstate->Values[(Offset+IrSize-1)&HRIR_MASK][0] = 0.0f;
hrtfstate->Values[(Offset+IrSize-1)&HRIR_MASK][1] = 0.0f;
hrtfstate->History[Offset&HRTF_HISTORY_MASK] = *(data++);
g = oldGain + oldGainStep*stepcount;
left = hrtfstate->History[(Offset-OldDelay[0])&HRTF_HISTORY_MASK]*g;
right = hrtfstate->History[(Offset-OldDelay[1])&HRTF_HISTORY_MASK]*g;
ApplyCoeffs(Offset, hrtfstate->Values, IrSize, OldCoeffs, left, right);
g = newGain + newGainStep*stepcount;
left = hrtfstate->History[(Offset-NewDelay[0])&HRTF_HISTORY_MASK]*g;
right = hrtfstate->History[(Offset-NewDelay[1])&HRTF_HISTORY_MASK]*g;
ApplyCoeffs(Offset, hrtfstate->Values, IrSize, NewCoeffs, left, right);
*(LeftOut++) += hrtfstate->Values[Offset&HRIR_MASK][0];
*(RightOut++) += hrtfstate->Values[Offset&HRIR_MASK][1];
stepcount += 1.0f;
Offset++;
}
newparams->Gain = newGain + newGainStep*stepcount;
}
void MixDirectHrtf(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
const ALfloat *data, ALsizei Offset, const ALsizei IrSize,
const ALfloat (*restrict Coeffs)[2], ALfloat (*restrict Values)[2],
ALsizei BufferSize)
{
ALfloat insample;
ALsizei i;
ASSUME(IrSize >= 4);
ASSUME(BufferSize > 0);
for(i = 0;i < BufferSize;i++)
{
Values[(Offset+IrSize)&HRIR_MASK][0] = 0.0f;
Values[(Offset+IrSize)&HRIR_MASK][1] = 0.0f;
Offset++;
insample = *(data++);
ApplyCoeffs(Offset, Values, IrSize, Coeffs, insample, insample);
*(LeftOut++) += Values[Offset&HRIR_MASK][0];
*(RightOut++) += Values[Offset&HRIR_MASK][1];
}
}
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#include "config.h"
#include <assert.h>
#include "alMain.h"
#include "alu.h"
#include "alSource.h"
#include "alAuxEffectSlot.h"
#include "defs.h"
static inline ALfloat do_point(const InterpState* UNUSED(state), const ALfloat *restrict vals, ALsizei UNUSED(frac))
{ return vals[0]; }
static inline ALfloat do_lerp(const InterpState* UNUSED(state), const ALfloat *restrict vals, ALsizei frac)
{ return lerp(vals[0], vals[1], frac * (1.0f/FRACTIONONE)); }
static inline ALfloat do_cubic(const InterpState* UNUSED(state), const ALfloat *restrict vals, ALsizei frac)
{ return cubic(vals[0], vals[1], vals[2], vals[3], frac * (1.0f/FRACTIONONE)); }
static inline ALfloat do_bsinc(const InterpState *state, const ALfloat *restrict vals, ALsizei frac)
{
const ALfloat *fil, *scd, *phd, *spd;
ALsizei j_f, pi;
ALfloat pf, r;
ASSUME(state->bsinc.m > 0);
// Calculate the phase index and factor.
#define FRAC_PHASE_BITDIFF (FRACTIONBITS-BSINC_PHASE_BITS)
pi = frac >> FRAC_PHASE_BITDIFF;
pf = (frac & ((1<<FRAC_PHASE_BITDIFF)-1)) * (1.0f/(1<<FRAC_PHASE_BITDIFF));
#undef FRAC_PHASE_BITDIFF
fil = ASSUME_ALIGNED(state->bsinc.filter + state->bsinc.m*pi*4, 16);
scd = ASSUME_ALIGNED(fil + state->bsinc.m, 16);
phd = ASSUME_ALIGNED(scd + state->bsinc.m, 16);
spd = ASSUME_ALIGNED(phd + state->bsinc.m, 16);
// Apply the scale and phase interpolated filter.
r = 0.0f;
for(j_f = 0;j_f < state->bsinc.m;j_f++)
r += (fil[j_f] + state->bsinc.sf*scd[j_f] + pf*(phd[j_f] + state->bsinc.sf*spd[j_f])) * vals[j_f];
return r;
}
const ALfloat *Resample_copy_C(const InterpState* UNUSED(state),
const ALfloat *restrict src, ALsizei UNUSED(frac), ALint UNUSED(increment),
ALfloat *restrict dst, ALsizei numsamples)
{
#if defined(HAVE_SSE) || defined(HAVE_NEON)
/* Avoid copying the source data if it's aligned like the destination. */
if((((intptr_t)src)&15) == (((intptr_t)dst)&15))
return src;
#endif
memcpy(dst, src, numsamples*sizeof(ALfloat));
return dst;
}
#define DECL_TEMPLATE(Tag, Sampler, O) \
const ALfloat *Resample_##Tag##_C(const InterpState *state, \
const ALfloat *restrict src, ALsizei frac, ALint increment, \
ALfloat *restrict dst, ALsizei numsamples) \
{ \
const InterpState istate = *state; \
ALsizei i; \
\
ASSUME(numsamples > 0); \
\
src -= O; \
for(i = 0;i < numsamples;i++) \
{ \
dst[i] = Sampler(&istate, src, frac); \
\
frac += increment; \
src += frac>>FRACTIONBITS; \
frac &= FRACTIONMASK; \
} \
return dst; \
}
DECL_TEMPLATE(point, do_point, 0)
DECL_TEMPLATE(lerp, do_lerp, 0)
DECL_TEMPLATE(cubic, do_cubic, 1)
DECL_TEMPLATE(bsinc, do_bsinc, istate.bsinc.l)
#undef DECL_TEMPLATE
static inline void ApplyCoeffs(ALsizei Offset, ALfloat (*restrict Values)[2],
const ALsizei IrSize,
const ALfloat (*restrict Coeffs)[2],
ALfloat left, ALfloat right)
{
ALsizei c;
for(c = 0;c < IrSize;c++)
{
const ALsizei off = (Offset+c)&HRIR_MASK;
Values[off][0] += Coeffs[c][0] * left;
Values[off][1] += Coeffs[c][1] * right;
}
}
#define MixHrtf MixHrtf_C
#define MixHrtfBlend MixHrtfBlend_C
#define MixDirectHrtf MixDirectHrtf_C
#include "hrtf_inc.c"
void Mix_C(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE],
ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
ALsizei BufferSize)
{
const ALfloat delta = (Counter > 0) ? 1.0f/(ALfloat)Counter : 0.0f;
ALsizei c;
ASSUME(OutChans > 0);
ASSUME(BufferSize > 0);
for(c = 0;c < OutChans;c++)
{
ALsizei pos = 0;
ALfloat gain = CurrentGains[c];
const ALfloat diff = TargetGains[c] - gain;
if(fabsf(diff) > FLT_EPSILON)
{
ALsizei minsize = mini(BufferSize, Counter);
const ALfloat step = diff * delta;
ALfloat step_count = 0.0f;
for(;pos < minsize;pos++)
{
OutBuffer[c][OutPos+pos] += data[pos] * (gain + step*step_count);
step_count += 1.0f;
}
if(pos == Counter)
gain = TargetGains[c];
else
gain += step*step_count;
CurrentGains[c] = gain;
}
if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
continue;
for(;pos < BufferSize;pos++)
OutBuffer[c][OutPos+pos] += data[pos]*gain;
}
}
/* Basically the inverse of the above. Rather than one input going to multiple
* outputs (each with its own gain), it's multiple inputs (each with its own
* gain) going to one output. This applies one row (vs one column) of a matrix
* transform. And as the matrices are more or less static once set up, no
* stepping is necessary.
*/
void MixRow_C(ALfloat *OutBuffer, const ALfloat *Gains, const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans, ALsizei InPos, ALsizei BufferSize)
{
ALsizei c, i;
ASSUME(InChans > 0);
ASSUME(BufferSize > 0);
for(c = 0;c < InChans;c++)
{
const ALfloat gain = Gains[c];
if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
continue;
for(i = 0;i < BufferSize;i++)
OutBuffer[i] += data[c][InPos+i] * gain;
}
}
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#include "config.h"
#include <arm_neon.h>
#include "AL/al.h"
#include "AL/alc.h"
#include "alMain.h"
#include "alu.h"
#include "hrtf.h"
#include "defs.h"
const ALfloat *Resample_lerp_Neon(const InterpState* UNUSED(state),
const ALfloat *restrict src, ALsizei frac, ALint increment,
ALfloat *restrict dst, ALsizei numsamples)
{
const int32x4_t increment4 = vdupq_n_s32(increment*4);
const float32x4_t fracOne4 = vdupq_n_f32(1.0f/FRACTIONONE);
const int32x4_t fracMask4 = vdupq_n_s32(FRACTIONMASK);
alignas(16) ALsizei pos_[4], frac_[4];
int32x4_t pos4, frac4;
ALsizei todo, pos, i;
ASSUME(numsamples > 0);
InitiatePositionArrays(frac, increment, frac_, pos_, 4);
frac4 = vld1q_s32(frac_);
pos4 = vld1q_s32(pos_);
todo = numsamples & ~3;
for(i = 0;i < todo;i += 4)
{
const int pos0 = vgetq_lane_s32(pos4, 0);
const int pos1 = vgetq_lane_s32(pos4, 1);
const int pos2 = vgetq_lane_s32(pos4, 2);
const int pos3 = vgetq_lane_s32(pos4, 3);
const float32x4_t val1 = (float32x4_t){src[pos0], src[pos1], src[pos2], src[pos3]};
const float32x4_t val2 = (float32x4_t){src[pos0+1], src[pos1+1], src[pos2+1], src[pos3+1]};
/* val1 + (val2-val1)*mu */
const float32x4_t r0 = vsubq_f32(val2, val1);
const float32x4_t mu = vmulq_f32(vcvtq_f32_s32(frac4), fracOne4);
const float32x4_t out = vmlaq_f32(val1, mu, r0);
vst1q_f32(&dst[i], out);
frac4 = vaddq_s32(frac4, increment4);
pos4 = vaddq_s32(pos4, vshrq_n_s32(frac4, FRACTIONBITS));
frac4 = vandq_s32(frac4, fracMask4);
}
/* NOTE: These four elements represent the position *after* the last four
* samples, so the lowest element is the next position to resample.
*/
pos = vgetq_lane_s32(pos4, 0);
frac = vgetq_lane_s32(frac4, 0);
for(;i < numsamples;++i)
{
dst[i] = lerp(src[pos], src[pos+1], frac * (1.0f/FRACTIONONE));
frac += increment;
pos += frac>>FRACTIONBITS;
frac &= FRACTIONMASK;
}
return dst;
}
const ALfloat *Resample_bsinc_Neon(const InterpState *state,
const ALfloat *restrict src, ALsizei frac, ALint increment,
ALfloat *restrict dst, ALsizei dstlen)
{
const ALfloat *const filter = state->bsinc.filter;
const float32x4_t sf4 = vdupq_n_f32(state->bsinc.sf);
const ALsizei m = state->bsinc.m;
const float32x4_t *fil, *scd, *phd, *spd;
ALsizei pi, i, j, offset;
float32x4_t r4;
ALfloat pf;
ASSUME(m > 0);
ASSUME(dstlen > 0);
src -= state->bsinc.l;
for(i = 0;i < dstlen;i++)
{
// Calculate the phase index and factor.
#define FRAC_PHASE_BITDIFF (FRACTIONBITS-BSINC_PHASE_BITS)
pi = frac >> FRAC_PHASE_BITDIFF;
pf = (frac & ((1<<FRAC_PHASE_BITDIFF)-1)) * (1.0f/(1<<FRAC_PHASE_BITDIFF));
#undef FRAC_PHASE_BITDIFF
offset = m*pi*4;
fil = ASSUME_ALIGNED(filter + offset, 16); offset += m;
scd = ASSUME_ALIGNED(filter + offset, 16); offset += m;
phd = ASSUME_ALIGNED(filter + offset, 16); offset += m;
spd = ASSUME_ALIGNED(filter + offset, 16);
// Apply the scale and phase interpolated filter.
r4 = vdupq_n_f32(0.0f);
{
const ALsizei count = m >> 2;
const float32x4_t pf4 = vdupq_n_f32(pf);
ASSUME(count > 0);
for(j = 0;j < count;j++)
{
/* f = ((fil + sf*scd) + pf*(phd + sf*spd)) */
const float32x4_t f4 = vmlaq_f32(
vmlaq_f32(fil[j], sf4, scd[j]),
pf4, vmlaq_f32(phd[j], sf4, spd[j])
);
/* r += f*src */
r4 = vmlaq_f32(r4, f4, vld1q_f32(&src[j*4]));
}
}
r4 = vaddq_f32(r4, vcombine_f32(vrev64_f32(vget_high_f32(r4)),
vrev64_f32(vget_low_f32(r4))));
dst[i] = vget_lane_f32(vadd_f32(vget_low_f32(r4), vget_high_f32(r4)), 0);
frac += increment;
src += frac>>FRACTIONBITS;
frac &= FRACTIONMASK;
}
return dst;
}
static inline void ApplyCoeffs(ALsizei Offset, ALfloat (*restrict Values)[2],
const ALsizei IrSize,
const ALfloat (*restrict Coeffs)[2],
ALfloat left, ALfloat right)
{
ALsizei 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);
}
Values = ASSUME_ALIGNED(Values, 16);
Coeffs = ASSUME_ALIGNED(Coeffs, 16);
for(c = 0;c < IrSize;c += 2)
{
const ALsizei o0 = (Offset+c)&HRIR_MASK;
const ALsizei 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 MixHrtf MixHrtf_Neon
#define MixHrtfBlend MixHrtfBlend_Neon
#define MixDirectHrtf MixDirectHrtf_Neon
#include "hrtf_inc.c"
void Mix_Neon(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE],
ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
ALsizei BufferSize)
{
const ALfloat delta = (Counter > 0) ? 1.0f/(ALfloat)Counter : 0.0f;
ALsizei c;
ASSUME(OutChans > 0);
ASSUME(BufferSize > 0);
data = ASSUME_ALIGNED(data, 16);
OutBuffer = ASSUME_ALIGNED(OutBuffer, 16);
for(c = 0;c < OutChans;c++)
{
ALsizei pos = 0;
ALfloat gain = CurrentGains[c];
const ALfloat diff = TargetGains[c] - gain;
if(fabsf(diff) > FLT_EPSILON)
{
ALsizei minsize = mini(BufferSize, Counter);
const ALfloat step = diff * delta;
ALfloat step_count = 0.0f;
/* Mix with applying gain steps in aligned multiples of 4. */
if(LIKELY(minsize > 3))
{
const float32x4_t four4 = vdupq_n_f32(4.0f);
const float32x4_t step4 = vdupq_n_f32(step);
const float32x4_t gain4 = vdupq_n_f32(gain);
float32x4_t step_count4 = vsetq_lane_f32(0.0f,
vsetq_lane_f32(1.0f,
vsetq_lane_f32(2.0f,
vsetq_lane_f32(3.0f, vdupq_n_f32(0.0f), 3),
2), 1), 0
);
ALsizei todo = minsize >> 2;
do {
const float32x4_t val4 = vld1q_f32(&data[pos]);
float32x4_t dry4 = vld1q_f32(&OutBuffer[c][OutPos+pos]);
dry4 = vmlaq_f32(dry4, val4, vmlaq_f32(gain4, step4, step_count4));
step_count4 = vaddq_f32(step_count4, four4);
vst1q_f32(&OutBuffer[c][OutPos+pos], dry4);
pos += 4;
} while(--todo);
/* NOTE: step_count4 now represents the next four counts after
* the last four mixed samples, so the lowest element
* represents the next step count to apply.
*/
step_count = vgetq_lane_f32(step_count4, 0);
}
/* Mix with applying left over gain steps that aren't aligned multiples of 4. */
for(;pos < minsize;pos++)
{
OutBuffer[c][OutPos+pos] += data[pos]*(gain + step*step_count);
step_count += 1.0f;
}
if(pos == Counter)
gain = TargetGains[c];
else
gain += step*step_count;
CurrentGains[c] = gain;
/* Mix until pos is aligned with 4 or the mix is done. */
minsize = mini(BufferSize, (pos+3)&~3);
for(;pos < minsize;pos++)
OutBuffer[c][OutPos+pos] += data[pos]*gain;
}
if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
continue;
if(LIKELY(BufferSize-pos > 3))
{
ALsizei todo = (BufferSize-pos) >> 2;
const float32x4_t gain4 = vdupq_n_f32(gain);
do {
const float32x4_t val4 = vld1q_f32(&data[pos]);
float32x4_t dry4 = vld1q_f32(&OutBuffer[c][OutPos+pos]);
dry4 = vmlaq_f32(dry4, val4, gain4);
vst1q_f32(&OutBuffer[c][OutPos+pos], dry4);
pos += 4;
} while(--todo);
}
for(;pos < BufferSize;pos++)
OutBuffer[c][OutPos+pos] += data[pos]*gain;
}
}
void MixRow_Neon(ALfloat *OutBuffer, const ALfloat *Gains, const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans, ALsizei InPos, ALsizei BufferSize)
{
ALsizei c;
ASSUME(InChans > 0);
ASSUME(BufferSize > 0);
for(c = 0;c < InChans;c++)
{
ALsizei pos = 0;
const ALfloat gain = Gains[c];
if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
continue;
if(LIKELY(BufferSize > 3))
{
ALsizei todo = BufferSize >> 2;
float32x4_t gain4 = vdupq_n_f32(gain);
do {
const float32x4_t val4 = vld1q_f32(&data[c][InPos+pos]);
float32x4_t dry4 = vld1q_f32(&OutBuffer[pos]);
dry4 = vmlaq_f32(dry4, val4, gain4);
vst1q_f32(&OutBuffer[pos], dry4);
pos += 4;
} while(--todo);
}
for(;pos < BufferSize;pos++)
OutBuffer[pos] += data[c][InPos+pos]*gain;
}
}
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#include "config.h"
#include <xmmintrin.h>
#include "AL/al.h"
#include "AL/alc.h"
#include "alMain.h"
#include "alu.h"
#include "alSource.h"
#include "alAuxEffectSlot.h"
#include "defs.h"
const ALfloat *Resample_bsinc_SSE(const InterpState *state, const ALfloat *restrict src,
ALsizei frac, ALint increment, ALfloat *restrict dst,
ALsizei dstlen)
{
const ALfloat *const filter = state->bsinc.filter;
const __m128 sf4 = _mm_set1_ps(state->bsinc.sf);
const ALsizei m = state->bsinc.m;
const __m128 *fil, *scd, *phd, *spd;
ALsizei pi, i, j, offset;
ALfloat pf;
__m128 r4;
ASSUME(m > 0);
ASSUME(dstlen > 0);
src -= state->bsinc.l;
for(i = 0;i < dstlen;i++)
{
// Calculate the phase index and factor.
#define FRAC_PHASE_BITDIFF (FRACTIONBITS-BSINC_PHASE_BITS)
pi = frac >> FRAC_PHASE_BITDIFF;
pf = (frac & ((1<<FRAC_PHASE_BITDIFF)-1)) * (1.0f/(1<<FRAC_PHASE_BITDIFF));
#undef FRAC_PHASE_BITDIFF
offset = m*pi*4;
fil = (const __m128*)ASSUME_ALIGNED(filter + offset, 16); offset += m;
scd = (const __m128*)ASSUME_ALIGNED(filter + offset, 16); offset += m;
phd = (const __m128*)ASSUME_ALIGNED(filter + offset, 16); offset += m;
spd = (const __m128*)ASSUME_ALIGNED(filter + offset, 16);
// Apply the scale and phase interpolated filter.
r4 = _mm_setzero_ps();
{
const ALsizei count = m >> 2;
const __m128 pf4 = _mm_set1_ps(pf);
ASSUME(count > 0);
#define MLA4(x, y, z) _mm_add_ps(x, _mm_mul_ps(y, z))
for(j = 0;j < count;j++)
{
/* f = ((fil + sf*scd) + pf*(phd + sf*spd)) */
const __m128 f4 = MLA4(
MLA4(fil[j], sf4, scd[j]),
pf4, MLA4(phd[j], sf4, spd[j])
);
/* r += f*src */
r4 = MLA4(r4, f4, _mm_loadu_ps(&src[j*4]));
}
#undef MLA4
}
r4 = _mm_add_ps(r4, _mm_shuffle_ps(r4, r4, _MM_SHUFFLE(0, 1, 2, 3)));
r4 = _mm_add_ps(r4, _mm_movehl_ps(r4, r4));
dst[i] = _mm_cvtss_f32(r4);
frac += increment;
src += frac>>FRACTIONBITS;
frac &= FRACTIONMASK;
}
return dst;
}
static inline void ApplyCoeffs(ALsizei Offset, ALfloat (*restrict Values)[2],
const ALsizei IrSize,
const ALfloat (*restrict Coeffs)[2],
ALfloat left, ALfloat right)
{
const __m128 lrlr = _mm_setr_ps(left, right, left, right);
__m128 vals = _mm_setzero_ps();
__m128 coeffs;
ALsizei i;
Values = ASSUME_ALIGNED(Values, 16);
Coeffs = ASSUME_ALIGNED(Coeffs, 16);
if((Offset&1))
{
const ALsizei o0 = Offset&HRIR_MASK;
const ALsizei 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 ALsizei 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 ALsizei 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 MixHrtf MixHrtf_SSE
#define MixHrtfBlend MixHrtfBlend_SSE
#define MixDirectHrtf MixDirectHrtf_SSE
#include "hrtf_inc.c"
void Mix_SSE(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE],
ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
ALsizei BufferSize)
{
const ALfloat delta = (Counter > 0) ? 1.0f/(ALfloat)Counter : 0.0f;
ALsizei c;
ASSUME(OutChans > 0);
ASSUME(BufferSize > 0);
for(c = 0;c < OutChans;c++)
{
ALsizei pos = 0;
ALfloat gain = CurrentGains[c];
const ALfloat diff = TargetGains[c] - gain;
if(fabsf(diff) > FLT_EPSILON)
{
ALsizei minsize = mini(BufferSize, Counter);
const ALfloat step = diff * delta;
ALfloat step_count = 0.0f;
/* Mix with applying gain steps in aligned multiples of 4. */
if(LIKELY(minsize > 3))
{
const __m128 four4 = _mm_set1_ps(4.0f);
const __m128 step4 = _mm_set1_ps(step);
const __m128 gain4 = _mm_set1_ps(gain);
__m128 step_count4 = _mm_setr_ps(0.0f, 1.0f, 2.0f, 3.0f);
ALsizei todo = minsize >> 2;
do {
const __m128 val4 = _mm_load_ps(&data[pos]);
__m128 dry4 = _mm_load_ps(&OutBuffer[c][OutPos+pos]);
#define MLA4(x, y, z) _mm_add_ps(x, _mm_mul_ps(y, z))
/* dry += val * (gain + step*step_count) */
dry4 = MLA4(dry4, val4, MLA4(gain4, step4, step_count4));
#undef MLA4
_mm_store_ps(&OutBuffer[c][OutPos+pos], dry4);
step_count4 = _mm_add_ps(step_count4, four4);
pos += 4;
} while(--todo);
/* NOTE: step_count4 now represents the next four counts after
* the last four mixed samples, so the lowest element
* represents the next step count to apply.
*/
step_count = _mm_cvtss_f32(step_count4);
}
/* Mix with applying left over gain steps that aren't aligned multiples of 4. */
for(;pos < minsize;pos++)
{
OutBuffer[c][OutPos+pos] += data[pos]*(gain + step*step_count);
step_count += 1.0f;
}
if(pos == Counter)
gain = TargetGains[c];
else
gain += step*step_count;
CurrentGains[c] = gain;
/* Mix until pos is aligned with 4 or the mix is done. */
minsize = mini(BufferSize, (pos+3)&~3);
for(;pos < minsize;pos++)
OutBuffer[c][OutPos+pos] += data[pos]*gain;
}
if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
continue;
if(LIKELY(BufferSize-pos > 3))
{
ALsizei todo = (BufferSize-pos) >> 2;
const __m128 gain4 = _mm_set1_ps(gain);
do {
const __m128 val4 = _mm_load_ps(&data[pos]);
__m128 dry4 = _mm_load_ps(&OutBuffer[c][OutPos+pos]);
dry4 = _mm_add_ps(dry4, _mm_mul_ps(val4, gain4));
_mm_store_ps(&OutBuffer[c][OutPos+pos], dry4);
pos += 4;
} while(--todo);
}
for(;pos < BufferSize;pos++)
OutBuffer[c][OutPos+pos] += data[pos]*gain;
}
}
void MixRow_SSE(ALfloat *OutBuffer, const ALfloat *Gains, const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans, ALsizei InPos, ALsizei BufferSize)
{
ALsizei c;
ASSUME(InChans > 0);
ASSUME(BufferSize > 0);
for(c = 0;c < InChans;c++)
{
ALsizei pos = 0;
const ALfloat gain = Gains[c];
if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
continue;
if(LIKELY(BufferSize > 3))
{
ALsizei todo = BufferSize >> 2;
const __m128 gain4 = _mm_set1_ps(gain);
do {
const __m128 val4 = _mm_load_ps(&data[c][InPos+pos]);
__m128 dry4 = _mm_load_ps(&OutBuffer[pos]);
dry4 = _mm_add_ps(dry4, _mm_mul_ps(val4, gain4));
_mm_store_ps(&OutBuffer[pos], dry4);
pos += 4;
} while(--todo);
}
for(;pos < BufferSize;pos++)
OutBuffer[pos] += data[c][InPos+pos]*gain;
}
}
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/**
* OpenAL cross platform audio library
* Copyright (C) 2014 by Timothy Arceri <t_arceri@yahoo.com.au>.
* 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <xmmintrin.h>
#include <emmintrin.h>
#include "alu.h"
#include "defs.h"
const ALfloat *Resample_lerp_SSE2(const InterpState* UNUSED(state),
const ALfloat *restrict src, ALsizei frac, ALint increment,
ALfloat *restrict dst, ALsizei numsamples)
{
const __m128i increment4 = _mm_set1_epi32(increment*4);
const __m128 fracOne4 = _mm_set1_ps(1.0f/FRACTIONONE);
const __m128i fracMask4 = _mm_set1_epi32(FRACTIONMASK);
alignas(16) ALsizei pos_[4], frac_[4];
__m128i frac4, pos4;
ALsizei todo, pos, i;
ASSUME(numsamples > 0);
InitiatePositionArrays(frac, increment, frac_, pos_, 4);
frac4 = _mm_setr_epi32(frac_[0], frac_[1], frac_[2], frac_[3]);
pos4 = _mm_setr_epi32(pos_[0], pos_[1], pos_[2], pos_[3]);
todo = numsamples & ~3;
for(i = 0;i < todo;i += 4)
{
const int pos0 = _mm_cvtsi128_si32(_mm_shuffle_epi32(pos4, _MM_SHUFFLE(0, 0, 0, 0)));
const int pos1 = _mm_cvtsi128_si32(_mm_shuffle_epi32(pos4, _MM_SHUFFLE(1, 1, 1, 1)));
const int pos2 = _mm_cvtsi128_si32(_mm_shuffle_epi32(pos4, _MM_SHUFFLE(2, 2, 2, 2)));
const int pos3 = _mm_cvtsi128_si32(_mm_shuffle_epi32(pos4, _MM_SHUFFLE(3, 3, 3, 3)));
const __m128 val1 = _mm_setr_ps(src[pos0 ], src[pos1 ], src[pos2 ], src[pos3 ]);
const __m128 val2 = _mm_setr_ps(src[pos0+1], src[pos1+1], src[pos2+1], src[pos3+1]);
/* val1 + (val2-val1)*mu */
const __m128 r0 = _mm_sub_ps(val2, val1);
const __m128 mu = _mm_mul_ps(_mm_cvtepi32_ps(frac4), fracOne4);
const __m128 out = _mm_add_ps(val1, _mm_mul_ps(mu, r0));
_mm_store_ps(&dst[i], out);
frac4 = _mm_add_epi32(frac4, increment4);
pos4 = _mm_add_epi32(pos4, _mm_srli_epi32(frac4, FRACTIONBITS));
frac4 = _mm_and_si128(frac4, fracMask4);
}
/* NOTE: These four elements represent the position *after* the last four
* samples, so the lowest element is the next position to resample.
*/
pos = _mm_cvtsi128_si32(pos4);
frac = _mm_cvtsi128_si32(frac4);
for(;i < numsamples;++i)
{
dst[i] = lerp(src[pos], src[pos+1], frac * (1.0f/FRACTIONONE));
frac += increment;
pos += frac>>FRACTIONBITS;
frac &= FRACTIONMASK;
}
return dst;
}
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/**
* OpenAL cross platform audio library
* Copyright (C) 2014 by Timothy Arceri <t_arceri@yahoo.com.au>.
* 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <xmmintrin.h>
#include <emmintrin.h>
#include <smmintrin.h>
#include "alu.h"
#include "defs.h"
const ALfloat *Resample_lerp_SSE41(const InterpState* UNUSED(state),
const ALfloat *restrict src, ALsizei frac, ALint increment,
ALfloat *restrict dst, ALsizei numsamples)
{
const __m128i increment4 = _mm_set1_epi32(increment*4);
const __m128 fracOne4 = _mm_set1_ps(1.0f/FRACTIONONE);
const __m128i fracMask4 = _mm_set1_epi32(FRACTIONMASK);
alignas(16) ALsizei pos_[4], frac_[4];
__m128i frac4, pos4;
ALsizei todo, pos, i;
ASSUME(numsamples > 0);
InitiatePositionArrays(frac, increment, frac_, pos_, 4);
frac4 = _mm_setr_epi32(frac_[0], frac_[1], frac_[2], frac_[3]);
pos4 = _mm_setr_epi32(pos_[0], pos_[1], pos_[2], pos_[3]);
todo = numsamples & ~3;
for(i = 0;i < todo;i += 4)
{
const int pos0 = _mm_extract_epi32(pos4, 0);
const int pos1 = _mm_extract_epi32(pos4, 1);
const int pos2 = _mm_extract_epi32(pos4, 2);
const int pos3 = _mm_extract_epi32(pos4, 3);
const __m128 val1 = _mm_setr_ps(src[pos0 ], src[pos1 ], src[pos2 ], src[pos3 ]);
const __m128 val2 = _mm_setr_ps(src[pos0+1], src[pos1+1], src[pos2+1], src[pos3+1]);
/* val1 + (val2-val1)*mu */
const __m128 r0 = _mm_sub_ps(val2, val1);
const __m128 mu = _mm_mul_ps(_mm_cvtepi32_ps(frac4), fracOne4);
const __m128 out = _mm_add_ps(val1, _mm_mul_ps(mu, r0));
_mm_store_ps(&dst[i], out);
frac4 = _mm_add_epi32(frac4, increment4);
pos4 = _mm_add_epi32(pos4, _mm_srli_epi32(frac4, FRACTIONBITS));
frac4 = _mm_and_si128(frac4, fracMask4);
}
/* NOTE: These four elements represent the position *after* the last four
* samples, so the lowest element is the next position to resample.
*/
pos = _mm_cvtsi128_si32(pos4);
frac = _mm_cvtsi128_si32(frac4);
for(;i < numsamples;++i)
{
dst[i] = lerp(src[pos], src[pos+1], frac * (1.0f/FRACTIONONE));
frac += increment;
pos += frac>>FRACTIONBITS;
frac &= FRACTIONMASK;
}
return dst;
}
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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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 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 "sample_cvt.h"
#include "alu.h"
#include "alconfig.h"
#include "ringbuffer.h"
#include "cpu_caps.h"
#include "mixer/defs.h"
static_assert((INT_MAX>>FRACTIONBITS)/MAX_PITCH > BUFFERSIZE,
"MAX_PITCH and/or BUFFERSIZE are too large for FRACTIONBITS!");
extern inline void InitiatePositionArrays(ALsizei frac, ALint increment, ALsizei *restrict frac_arr, ALsizei *restrict pos_arr, ALsizei size);
/* BSinc24 requires up to 23 extra samples before the current position, and 24 after. */
static_assert(MAX_RESAMPLE_PADDING >= 24, "MAX_RESAMPLE_PADDING must be at least 24!");
enum Resampler ResamplerDefault = LinearResampler;
MixerFunc MixSamples = Mix_C;
RowMixerFunc MixRowSamples = MixRow_C;
static HrtfMixerFunc MixHrtfSamples = MixHrtf_C;
static HrtfMixerBlendFunc MixHrtfBlendSamples = MixHrtfBlend_C;
static MixerFunc SelectMixer(void)
{
#ifdef HAVE_NEON
if((CPUCapFlags&CPU_CAP_NEON))
return Mix_Neon;
#endif
#ifdef HAVE_SSE
if((CPUCapFlags&CPU_CAP_SSE))
return Mix_SSE;
#endif
return Mix_C;
}
static RowMixerFunc SelectRowMixer(void)
{
#ifdef HAVE_NEON
if((CPUCapFlags&CPU_CAP_NEON))
return MixRow_Neon;
#endif
#ifdef HAVE_SSE
if((CPUCapFlags&CPU_CAP_SSE))
return MixRow_SSE;
#endif
return MixRow_C;
}
static inline HrtfMixerFunc SelectHrtfMixer(void)
{
#ifdef HAVE_NEON
if((CPUCapFlags&CPU_CAP_NEON))
return MixHrtf_Neon;
#endif
#ifdef HAVE_SSE
if((CPUCapFlags&CPU_CAP_SSE))
return MixHrtf_SSE;
#endif
return MixHrtf_C;
}
static inline HrtfMixerBlendFunc SelectHrtfBlendMixer(void)
{
#ifdef HAVE_NEON
if((CPUCapFlags&CPU_CAP_NEON))
return MixHrtfBlend_Neon;
#endif
#ifdef HAVE_SSE
if((CPUCapFlags&CPU_CAP_SSE))
return MixHrtfBlend_SSE;
#endif
return MixHrtfBlend_C;
}
ResamplerFunc SelectResampler(enum Resampler resampler)
{
switch(resampler)
{
case PointResampler:
return Resample_point_C;
case LinearResampler:
#ifdef HAVE_NEON
if((CPUCapFlags&CPU_CAP_NEON))
return Resample_lerp_Neon;
#endif
#ifdef HAVE_SSE4_1
if((CPUCapFlags&CPU_CAP_SSE4_1))
return Resample_lerp_SSE41;
#endif
#ifdef HAVE_SSE2
if((CPUCapFlags&CPU_CAP_SSE2))
return Resample_lerp_SSE2;
#endif
return Resample_lerp_C;
case FIR4Resampler:
return Resample_cubic_C;
case BSinc12Resampler:
case BSinc24Resampler:
#ifdef HAVE_NEON
if((CPUCapFlags&CPU_CAP_NEON))
return Resample_bsinc_Neon;
#endif
#ifdef HAVE_SSE
if((CPUCapFlags&CPU_CAP_SSE))
return Resample_bsinc_SSE;
#endif
return Resample_bsinc_C;
}
return Resample_point_C;
}
void aluInitMixer(void)
{
const char *str;
if(ConfigValueStr(NULL, NULL, "resampler", &str))
{
if(strcasecmp(str, "point") == 0 || strcasecmp(str, "none") == 0)
ResamplerDefault = PointResampler;
else if(strcasecmp(str, "linear") == 0)
ResamplerDefault = LinearResampler;
else if(strcasecmp(str, "cubic") == 0)
ResamplerDefault = FIR4Resampler;
else if(strcasecmp(str, "bsinc12") == 0)
ResamplerDefault = BSinc12Resampler;
else if(strcasecmp(str, "bsinc24") == 0)
ResamplerDefault = BSinc24Resampler;
else if(strcasecmp(str, "bsinc") == 0)
{
WARN("Resampler option \"%s\" is deprecated, using bsinc12\n", str);
ResamplerDefault = BSinc12Resampler;
}
else if(strcasecmp(str, "sinc4") == 0 || strcasecmp(str, "sinc8") == 0)
{
WARN("Resampler option \"%s\" is deprecated, using cubic\n", str);
ResamplerDefault = FIR4Resampler;
}
else
{
char *end;
long n = strtol(str, &end, 0);
if(*end == '\0' && (n == PointResampler || n == LinearResampler || n == FIR4Resampler))
ResamplerDefault = n;
else
WARN("Invalid resampler: %s\n", str);
}
}
MixHrtfBlendSamples = SelectHrtfBlendMixer();
MixHrtfSamples = SelectHrtfMixer();
MixSamples = SelectMixer();
MixRowSamples = SelectRowMixer();
}
static void SendAsyncEvent(ALCcontext *context, ALuint enumtype, ALenum type,
ALuint objid, ALuint param, const char *msg)
{
AsyncEvent evt = ASYNC_EVENT(enumtype);
evt.u.user.type = type;
evt.u.user.id = objid;
evt.u.user.param = param;
strcpy(evt.u.user.msg, msg);
if(ll_ringbuffer_write(context->AsyncEvents, (const char*)&evt, 1) == 1)
alsem_post(&context->EventSem);
}
static inline ALfloat Sample_ALubyte(ALubyte val)
{ return (val-128) * (1.0f/128.0f); }
static inline ALfloat Sample_ALshort(ALshort val)
{ return val * (1.0f/32768.0f); }
static inline ALfloat Sample_ALfloat(ALfloat val)
{ return val; }
static inline ALfloat Sample_ALdouble(ALdouble val)
{ return (ALfloat)val; }
typedef ALubyte ALmulaw;
static inline ALfloat Sample_ALmulaw(ALmulaw val)
{ return muLawDecompressionTable[val] * (1.0f/32768.0f); }
typedef ALubyte ALalaw;
static inline ALfloat Sample_ALalaw(ALalaw val)
{ return aLawDecompressionTable[val] * (1.0f/32768.0f); }
#define DECL_TEMPLATE(T) \
static inline void Load_##T(ALfloat *restrict dst, const T *restrict src, \
ALint srcstep, ALsizei samples) \
{ \
ALsizei i; \
for(i = 0;i < samples;i++) \
dst[i] += Sample_##T(src[i*srcstep]); \
}
DECL_TEMPLATE(ALubyte)
DECL_TEMPLATE(ALshort)
DECL_TEMPLATE(ALfloat)
DECL_TEMPLATE(ALdouble)
DECL_TEMPLATE(ALmulaw)
DECL_TEMPLATE(ALalaw)
#undef DECL_TEMPLATE
static void LoadSamples(ALfloat *restrict dst, const ALvoid *restrict src, ALint srcstep,
enum FmtType srctype, ALsizei samples)
{
#define HANDLE_FMT(ET, ST) case ET: Load_##ST(dst, src, srcstep, samples); break
switch(srctype)
{
HANDLE_FMT(FmtUByte, ALubyte);
HANDLE_FMT(FmtShort, ALshort);
HANDLE_FMT(FmtFloat, ALfloat);
HANDLE_FMT(FmtDouble, ALdouble);
HANDLE_FMT(FmtMulaw, ALmulaw);
HANDLE_FMT(FmtAlaw, ALalaw);
}
#undef HANDLE_FMT
}
static const ALfloat *DoFilters(BiquadFilter *lpfilter, BiquadFilter *hpfilter,
ALfloat *restrict dst, const ALfloat *restrict src,
ALsizei numsamples, enum ActiveFilters type)
{
ALsizei i;
switch(type)
{
case AF_None:
BiquadFilter_passthru(lpfilter, numsamples);
BiquadFilter_passthru(hpfilter, numsamples);
break;
case AF_LowPass:
BiquadFilter_process(lpfilter, dst, src, numsamples);
BiquadFilter_passthru(hpfilter, numsamples);
return dst;
case AF_HighPass:
BiquadFilter_passthru(lpfilter, numsamples);
BiquadFilter_process(hpfilter, dst, src, numsamples);
return dst;
case AF_BandPass:
for(i = 0;i < numsamples;)
{
ALfloat temp[256];
ALsizei todo = mini(256, numsamples-i);
BiquadFilter_process(lpfilter, temp, src+i, todo);
BiquadFilter_process(hpfilter, dst+i, temp, todo);
i += todo;
}
return dst;
}
return src;
}
/* This function uses these device temp buffers. */
#define SOURCE_DATA_BUF 0
#define RESAMPLED_BUF 1
#define FILTERED_BUF 2
#define NFC_DATA_BUF 3
ALboolean MixSource(ALvoice *voice, ALuint SourceID, ALCcontext *Context, ALsizei SamplesToDo)
{
ALCdevice *Device = Context->Device;
ALbufferlistitem *BufferListItem;
ALbufferlistitem *BufferLoopItem;
ALsizei NumChannels, SampleSize;
ALbitfieldSOFT enabledevt;
ALsizei buffers_done = 0;
ResamplerFunc Resample;
ALsizei DataPosInt;
ALsizei DataPosFrac;
ALint64 DataSize64;
ALint increment;
ALsizei Counter;
ALsizei OutPos;
ALsizei IrSize;
bool isplaying;
bool firstpass;
bool isstatic;
ALsizei chan;
ALsizei send;
/* Get source info */
isplaying = true; /* Will only be called while playing. */
isstatic = !!(voice->Flags&VOICE_IS_STATIC);
DataPosInt = ATOMIC_LOAD(&voice->position, almemory_order_acquire);
DataPosFrac = ATOMIC_LOAD(&voice->position_fraction, almemory_order_relaxed);
BufferListItem = ATOMIC_LOAD(&voice->current_buffer, almemory_order_relaxed);
BufferLoopItem = ATOMIC_LOAD(&voice->loop_buffer, almemory_order_relaxed);
NumChannels = voice->NumChannels;
SampleSize = voice->SampleSize;
increment = voice->Step;
IrSize = (Device->HrtfHandle ? Device->HrtfHandle->irSize : 0);
Resample = ((increment == FRACTIONONE && DataPosFrac == 0) ?
Resample_copy_C : voice->Resampler);
Counter = (voice->Flags&VOICE_IS_FADING) ? SamplesToDo : 0;
firstpass = true;
OutPos = 0;
do {
ALsizei SrcBufferSize, DstBufferSize;
/* Figure out how many buffer samples will be needed */
DataSize64 = SamplesToDo-OutPos;
DataSize64 *= increment;
DataSize64 += DataPosFrac+FRACTIONMASK;
DataSize64 >>= FRACTIONBITS;
DataSize64 += MAX_RESAMPLE_PADDING*2;
SrcBufferSize = (ALsizei)mini64(DataSize64, BUFFERSIZE);
/* Figure out how many samples we can actually mix from this. */
DataSize64 = SrcBufferSize;
DataSize64 -= MAX_RESAMPLE_PADDING*2;
DataSize64 <<= FRACTIONBITS;
DataSize64 -= DataPosFrac;
DstBufferSize = (ALsizei)mini64((DataSize64+(increment-1)) / increment,
SamplesToDo - OutPos);
/* Some mixers like having a multiple of 4, so try to give that unless
* this is the last update. */
if(DstBufferSize < SamplesToDo-OutPos)
DstBufferSize &= ~3;
/* It's impossible to have a buffer list item with no entries. */
assert(BufferListItem->num_buffers > 0);
for(chan = 0;chan < NumChannels;chan++)
{
const ALfloat *ResampledData;
ALfloat *SrcData = Device->TempBuffer[SOURCE_DATA_BUF];
ALsizei FilledAmt;
/* Load the previous samples into the source data first, and clear the rest. */
memcpy(SrcData, voice->PrevSamples[chan], MAX_RESAMPLE_PADDING*sizeof(ALfloat));
memset(SrcData+MAX_RESAMPLE_PADDING, 0, (BUFFERSIZE-MAX_RESAMPLE_PADDING)*
sizeof(ALfloat));
FilledAmt = MAX_RESAMPLE_PADDING;
if(isstatic)
{
/* TODO: For static sources, loop points are taken from the
* first buffer (should be adjusted by any buffer offset, to
* possibly be added later).
*/
const ALbuffer *Buffer0 = BufferListItem->buffers[0];
const ALsizei LoopStart = Buffer0->LoopStart;
const ALsizei LoopEnd = Buffer0->LoopEnd;
const ALsizei LoopSize = LoopEnd - LoopStart;
/* If current pos is beyond the loop range, do not loop */
if(!BufferLoopItem || DataPosInt >= LoopEnd)
{
ALsizei SizeToDo = SrcBufferSize - FilledAmt;
ALsizei CompLen = 0;
ALsizei i;
BufferLoopItem = NULL;
for(i = 0;i < BufferListItem->num_buffers;i++)
{
const ALbuffer *buffer = BufferListItem->buffers[i];
const ALubyte *Data = buffer->data;
ALsizei DataSize;
if(DataPosInt >= buffer->SampleLen)
continue;
/* Load what's left to play from the buffer */
DataSize = mini(SizeToDo, buffer->SampleLen - DataPosInt);
CompLen = maxi(CompLen, DataSize);
LoadSamples(&SrcData[FilledAmt],
&Data[(DataPosInt*NumChannels + chan)*SampleSize],
NumChannels, buffer->FmtType, DataSize
);
}
FilledAmt += CompLen;
}
else
{
ALsizei SizeToDo = mini(SrcBufferSize - FilledAmt, LoopEnd - DataPosInt);
ALsizei CompLen = 0;
ALsizei i;
for(i = 0;i < BufferListItem->num_buffers;i++)
{
const ALbuffer *buffer = BufferListItem->buffers[i];
const ALubyte *Data = buffer->data;
ALsizei DataSize;
if(DataPosInt >= buffer->SampleLen)
continue;
/* Load what's left of this loop iteration */
DataSize = mini(SizeToDo, buffer->SampleLen - DataPosInt);
CompLen = maxi(CompLen, DataSize);
LoadSamples(&SrcData[FilledAmt],
&Data[(DataPosInt*NumChannels + chan)*SampleSize],
NumChannels, buffer->FmtType, DataSize
);
}
FilledAmt += CompLen;
while(SrcBufferSize > FilledAmt)
{
const ALsizei SizeToDo = mini(SrcBufferSize - FilledAmt, LoopSize);
CompLen = 0;
for(i = 0;i < BufferListItem->num_buffers;i++)
{
const ALbuffer *buffer = BufferListItem->buffers[i];
const ALubyte *Data = buffer->data;
ALsizei DataSize;
if(LoopStart >= buffer->SampleLen)
continue;
DataSize = mini(SizeToDo, buffer->SampleLen - LoopStart);
CompLen = maxi(CompLen, DataSize);
LoadSamples(&SrcData[FilledAmt],
&Data[(LoopStart*NumChannels + chan)*SampleSize],
NumChannels, buffer->FmtType, DataSize
);
}
FilledAmt += CompLen;
}
}
}
else
{
/* Crawl the buffer queue to fill in the temp buffer */
ALbufferlistitem *tmpiter = BufferListItem;
ALsizei pos = DataPosInt;
while(tmpiter && SrcBufferSize > FilledAmt)
{
ALsizei SizeToDo = SrcBufferSize - FilledAmt;
ALsizei CompLen = 0;
ALsizei i;
for(i = 0;i < tmpiter->num_buffers;i++)
{
const ALbuffer *ALBuffer = tmpiter->buffers[i];
ALsizei DataSize = ALBuffer ? ALBuffer->SampleLen : 0;
if(DataSize > pos)
{
const ALubyte *Data = ALBuffer->data;
Data += (pos*NumChannels + chan)*SampleSize;
DataSize = mini(SizeToDo, DataSize - pos);
CompLen = maxi(CompLen, DataSize);
LoadSamples(&SrcData[FilledAmt], Data, NumChannels,
ALBuffer->FmtType, DataSize);
}
}
if(UNLIKELY(!CompLen))
pos -= tmpiter->max_samples;
else
{
FilledAmt += CompLen;
if(SrcBufferSize <= FilledAmt)
break;
pos = 0;
}
tmpiter = ATOMIC_LOAD(&tmpiter->next, almemory_order_acquire);
if(!tmpiter) tmpiter = BufferLoopItem;
}
}
/* Store the last source samples used for next time. */
memcpy(voice->PrevSamples[chan],
&SrcData[(increment*DstBufferSize + DataPosFrac)>>FRACTIONBITS],
MAX_RESAMPLE_PADDING*sizeof(ALfloat)
);
/* Now resample, then filter and mix to the appropriate outputs. */
ResampledData = Resample(&voice->ResampleState,
&SrcData[MAX_RESAMPLE_PADDING], DataPosFrac, increment,
Device->TempBuffer[RESAMPLED_BUF], DstBufferSize
);
{
DirectParams *parms = &voice->Direct.Params[chan];
const ALfloat *samples;
samples = DoFilters(
&parms->LowPass, &parms->HighPass, Device->TempBuffer[FILTERED_BUF],
ResampledData, DstBufferSize, voice->Direct.FilterType
);
if(!(voice->Flags&VOICE_HAS_HRTF))
{
if(!Counter)
memcpy(parms->Gains.Current, parms->Gains.Target,
sizeof(parms->Gains.Current));
if(!(voice->Flags&VOICE_HAS_NFC))
MixSamples(samples, voice->Direct.Channels, voice->Direct.Buffer,
parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
DstBufferSize
);
else
{
ALfloat *nfcsamples = Device->TempBuffer[NFC_DATA_BUF];
ALsizei chanoffset = 0;
MixSamples(samples,
voice->Direct.ChannelsPerOrder[0], voice->Direct.Buffer,
parms->Gains.Current, parms->Gains.Target, Counter, OutPos,
DstBufferSize
);
chanoffset += voice->Direct.ChannelsPerOrder[0];
#define APPLY_NFC_MIX(order) \
if(voice->Direct.ChannelsPerOrder[order] > 0) \
{ \
NfcFilterProcess##order(&parms->NFCtrlFilter, nfcsamples, samples, \
DstBufferSize); \
MixSamples(nfcsamples, voice->Direct.ChannelsPerOrder[order], \
voice->Direct.Buffer+chanoffset, parms->Gains.Current+chanoffset, \
parms->Gains.Target+chanoffset, Counter, OutPos, DstBufferSize \
); \
chanoffset += voice->Direct.ChannelsPerOrder[order]; \
}
APPLY_NFC_MIX(1)
APPLY_NFC_MIX(2)
APPLY_NFC_MIX(3)
#undef APPLY_NFC_MIX
}
}
else
{
MixHrtfParams hrtfparams;
ALsizei fademix = 0;
int lidx, ridx;
lidx = GetChannelIdxByName(&Device->RealOut, FrontLeft);
ridx = GetChannelIdxByName(&Device->RealOut, FrontRight);
assert(lidx != -1 && ridx != -1);
if(!Counter)
{
/* No fading, just overwrite the old HRTF params. */
parms->Hrtf.Old = parms->Hrtf.Target;
}
else if(!(parms->Hrtf.Old.Gain > GAIN_SILENCE_THRESHOLD))
{
/* The old HRTF params are silent, so overwrite the old
* coefficients with the new, and reset the old gain to
* 0. The future mix will then fade from silence.
*/
parms->Hrtf.Old = parms->Hrtf.Target;
parms->Hrtf.Old.Gain = 0.0f;
}
else if(firstpass)
{
ALfloat gain;
/* Fade between the coefficients over 128 samples. */
fademix = mini(DstBufferSize, 128);
/* The new coefficients need to fade in completely
* since they're replacing the old ones. To keep the
* gain fading consistent, interpolate between the old
* and new target gains given how much of the fade time
* this mix handles.
*/
gain = lerp(parms->Hrtf.Old.Gain, parms->Hrtf.Target.Gain,
minf(1.0f, (ALfloat)fademix/Counter));
hrtfparams.Coeffs = parms->Hrtf.Target.Coeffs;
hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0];
hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1];
hrtfparams.Gain = 0.0f;
hrtfparams.GainStep = gain / (ALfloat)fademix;
MixHrtfBlendSamples(
voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
samples, voice->Offset, OutPos, IrSize, &parms->Hrtf.Old,
&hrtfparams, &parms->Hrtf.State, fademix
);
/* Update the old parameters with the result. */
parms->Hrtf.Old = parms->Hrtf.Target;
if(fademix < Counter)
parms->Hrtf.Old.Gain = hrtfparams.Gain;
}
if(fademix < DstBufferSize)
{
ALsizei todo = DstBufferSize - fademix;
ALfloat gain = parms->Hrtf.Target.Gain;
/* Interpolate the target gain if the gain fading lasts
* longer than this mix.
*/
if(Counter > DstBufferSize)
gain = lerp(parms->Hrtf.Old.Gain, gain,
(ALfloat)todo/(Counter-fademix));
hrtfparams.Coeffs = parms->Hrtf.Target.Coeffs;
hrtfparams.Delay[0] = parms->Hrtf.Target.Delay[0];
hrtfparams.Delay[1] = parms->Hrtf.Target.Delay[1];
hrtfparams.Gain = parms->Hrtf.Old.Gain;
hrtfparams.GainStep = (gain - parms->Hrtf.Old.Gain) / (ALfloat)todo;
MixHrtfSamples(
voice->Direct.Buffer[lidx], voice->Direct.Buffer[ridx],
samples+fademix, voice->Offset+fademix, OutPos+fademix, IrSize,
&hrtfparams, &parms->Hrtf.State, todo
);
/* Store the interpolated gain or the final target gain
* depending if the fade is done.
*/
if(DstBufferSize < Counter)
parms->Hrtf.Old.Gain = gain;
else
parms->Hrtf.Old.Gain = parms->Hrtf.Target.Gain;
}
}
}
for(send = 0;send < Device->NumAuxSends;send++)
{
SendParams *parms = &voice->Send[send].Params[chan];
const ALfloat *samples;
if(!voice->Send[send].Buffer)
continue;
samples = DoFilters(
&parms->LowPass, &parms->HighPass, Device->TempBuffer[FILTERED_BUF],
ResampledData, DstBufferSize, voice->Send[send].FilterType
);
if(!Counter)
memcpy(parms->Gains.Current, parms->Gains.Target,
sizeof(parms->Gains.Current));
MixSamples(samples, voice->Send[send].Channels, voice->Send[send].Buffer,
parms->Gains.Current, parms->Gains.Target, Counter, OutPos, DstBufferSize
);
}
}
/* Update positions */
DataPosFrac += increment*DstBufferSize;
DataPosInt += DataPosFrac>>FRACTIONBITS;
DataPosFrac &= FRACTIONMASK;
OutPos += DstBufferSize;
voice->Offset += DstBufferSize;
Counter = maxi(DstBufferSize, Counter) - DstBufferSize;
firstpass = false;
if(isstatic)
{
if(BufferLoopItem)
{
/* Handle looping static source */
const ALbuffer *Buffer = BufferListItem->buffers[0];
ALsizei LoopStart = Buffer->LoopStart;
ALsizei LoopEnd = Buffer->LoopEnd;
if(DataPosInt >= LoopEnd)
{
assert(LoopEnd > LoopStart);
DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart;
}
}
else
{
/* Handle non-looping static source */
if(DataPosInt >= BufferListItem->max_samples)
{
isplaying = false;
BufferListItem = NULL;
DataPosInt = 0;
DataPosFrac = 0;
break;
}
}
}
else while(1)
{
/* Handle streaming source */
if(BufferListItem->max_samples > DataPosInt)
break;
DataPosInt -= BufferListItem->max_samples;
buffers_done += BufferListItem->num_buffers;
BufferListItem = ATOMIC_LOAD(&BufferListItem->next, almemory_order_relaxed);
if(!BufferListItem && !(BufferListItem=BufferLoopItem))
{
isplaying = false;
DataPosInt = 0;
DataPosFrac = 0;
break;
}
}
} while(isplaying && OutPos < SamplesToDo);
voice->Flags |= VOICE_IS_FADING;
/* Update source info */
ATOMIC_STORE(&voice->position, DataPosInt, almemory_order_relaxed);
ATOMIC_STORE(&voice->position_fraction, DataPosFrac, almemory_order_relaxed);
ATOMIC_STORE(&voice->current_buffer, BufferListItem, almemory_order_release);
/* Send any events now, after the position/buffer info was updated. */
enabledevt = ATOMIC_LOAD(&Context->EnabledEvts, almemory_order_acquire);
if(buffers_done > 0 && (enabledevt&EventType_BufferCompleted))
SendAsyncEvent(Context, EventType_BufferCompleted,
AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT, SourceID, buffers_done, "Buffer completed"
);
return isplaying;
}
-1239
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-105
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@@ -1,105 +0,0 @@
#ifndef POLYMORPHISM_H
#define POLYMORPHISM_H
/* Macros to declare inheriting types, and to (down-)cast and up-cast. */
#define DERIVE_FROM_TYPE(t) t t##_parent
#define STATIC_CAST(to, obj) (&(obj)->to##_parent)
#ifdef __GNUC__
#define STATIC_UPCAST(to, from, obj) __extension__({ \
static_assert(__builtin_types_compatible_p(from, __typeof(*(obj))), \
"Invalid upcast object from type"); \
(to*)((char*)(obj) - offsetof(to, from##_parent)); \
})
#else
#define STATIC_UPCAST(to, from, obj) ((to*)((char*)(obj) - offsetof(to, from##_parent)))
#endif
/* Defines method forwards, which call the given parent's (T2's) implementation. */
#define DECLARE_FORWARD(T1, T2, rettype, func) \
rettype T1##_##func(T1 *obj) \
{ return T2##_##func(STATIC_CAST(T2, obj)); }
#define DECLARE_FORWARD1(T1, T2, rettype, func, argtype1) \
rettype T1##_##func(T1 *obj, argtype1 a) \
{ return T2##_##func(STATIC_CAST(T2, obj), a); }
#define DECLARE_FORWARD2(T1, T2, rettype, func, argtype1, argtype2) \
rettype T1##_##func(T1 *obj, argtype1 a, argtype2 b) \
{ return T2##_##func(STATIC_CAST(T2, obj), a, b); }
#define DECLARE_FORWARD3(T1, T2, rettype, func, argtype1, argtype2, argtype3) \
rettype T1##_##func(T1 *obj, argtype1 a, argtype2 b, argtype3 c) \
{ return T2##_##func(STATIC_CAST(T2, obj), a, b, c); }
/* Defines method thunks, functions that call to the child's method. */
#define DECLARE_THUNK(T1, T2, rettype, func) \
static rettype T1##_##T2##_##func(T2 *obj) \
{ return T1##_##func(STATIC_UPCAST(T1, T2, obj)); }
#define DECLARE_THUNK1(T1, T2, rettype, func, argtype1) \
static rettype T1##_##T2##_##func(T2 *obj, argtype1 a) \
{ return T1##_##func(STATIC_UPCAST(T1, T2, obj), a); }
#define DECLARE_THUNK2(T1, T2, rettype, func, argtype1, argtype2) \
static rettype T1##_##T2##_##func(T2 *obj, argtype1 a, argtype2 b) \
{ return T1##_##func(STATIC_UPCAST(T1, T2, obj), a, b); }
#define DECLARE_THUNK3(T1, T2, rettype, func, argtype1, argtype2, argtype3) \
static rettype T1##_##T2##_##func(T2 *obj, argtype1 a, argtype2 b, argtype3 c) \
{ return T1##_##func(STATIC_UPCAST(T1, T2, obj), a, b, c); }
#define DECLARE_THUNK4(T1, T2, rettype, func, argtype1, argtype2, argtype3, argtype4) \
static rettype T1##_##T2##_##func(T2 *obj, argtype1 a, argtype2 b, argtype3 c, argtype4 d) \
{ return T1##_##func(STATIC_UPCAST(T1, T2, obj), a, b, c, d); }
/* Defines the default functions used to (de)allocate a polymorphic object. */
#define DECLARE_DEFAULT_ALLOCATORS(T) \
static void* T##_New(size_t size) { return al_malloc(16, size); } \
static void T##_Delete(void *ptr) { al_free(ptr); }
/* Helper to extract an argument list for virtual method calls. */
#define EXTRACT_VCALL_ARGS(...) __VA_ARGS__))
/* Call a "virtual" method on an object, with arguments. */
#define V(obj, func) ((obj)->vtbl->func((obj), EXTRACT_VCALL_ARGS
/* Call a "virtual" method on an object, with no arguments. */
#define V0(obj, func) ((obj)->vtbl->func((obj) EXTRACT_VCALL_ARGS
/* Helper to extract an argument list for NEW_OBJ calls. */
#define EXTRACT_NEW_ARGS(...) __VA_ARGS__); \
} \
} while(0)
/* Allocate and construct an object, with arguments. */
#define NEW_OBJ(_res, T) do { \
_res = T##_New(sizeof(T)); \
if(_res) \
{ \
memset(_res, 0, sizeof(T)); \
T##_Construct(_res, EXTRACT_NEW_ARGS
/* Allocate and construct an object, with no arguments. */
#define NEW_OBJ0(_res, T) do { \
_res = T##_New(sizeof(T)); \
if(_res) \
{ \
memset(_res, 0, sizeof(T)); \
T##_Construct(_res EXTRACT_NEW_ARGS
/* Destructs and deallocate an object. */
#define DELETE_OBJ(obj) do { \
if((obj) != NULL) \
{ \
V0((obj),Destruct)(); \
V0((obj),Delete)(); \
} \
} while(0)
/* Helper to get a type's vtable thunk for a child type. */
#define GET_VTABLE2(T1, T2) (&(T1##_##T2##_vtable))
/* Helper to set an object's vtable thunk for a child type. Used when constructing an object. */
#define SET_VTABLE2(T1, T2, obj) (STATIC_CAST(T2, obj)->vtbl = GET_VTABLE2(T1, T2))
#endif /* POLYMORPHISM_H */
-295
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@@ -1,295 +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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <string.h>
#include <stdlib.h>
#include <limits.h>
#include "ringbuffer.h"
#include "align.h"
#include "atomic.h"
#include "threads.h"
#include "almalloc.h"
#include "compat.h"
/* NOTE: This lockless ringbuffer implementation is copied from JACK, extended
* to include an element size. Consequently, parameters and return values for a
* size or count is in 'elements', not bytes. Additionally, it only supports
* single-consumer/single-provider operation. */
struct ll_ringbuffer {
ATOMIC(size_t) write_ptr;
ATOMIC(size_t) read_ptr;
size_t size;
size_t size_mask;
size_t elem_size;
alignas(16) char buf[];
};
ll_ringbuffer_t *ll_ringbuffer_create(size_t sz, size_t elem_sz, int limit_writes)
{
ll_ringbuffer_t *rb;
size_t power_of_two = 0;
if(sz > 0)
{
power_of_two = sz;
power_of_two |= power_of_two>>1;
power_of_two |= power_of_two>>2;
power_of_two |= power_of_two>>4;
power_of_two |= power_of_two>>8;
power_of_two |= power_of_two>>16;
#if SIZE_MAX > UINT_MAX
power_of_two |= power_of_two>>32;
#endif
}
power_of_two++;
if(power_of_two < sz) return NULL;
rb = al_malloc(16, sizeof(*rb) + power_of_two*elem_sz);
if(!rb) return NULL;
ATOMIC_INIT(&rb->write_ptr, 0);
ATOMIC_INIT(&rb->read_ptr, 0);
rb->size = limit_writes ? sz : power_of_two;
rb->size_mask = power_of_two - 1;
rb->elem_size = elem_sz;
return rb;
}
void ll_ringbuffer_free(ll_ringbuffer_t *rb)
{
al_free(rb);
}
void ll_ringbuffer_reset(ll_ringbuffer_t *rb)
{
ATOMIC_STORE(&rb->write_ptr, 0, almemory_order_release);
ATOMIC_STORE(&rb->read_ptr, 0, almemory_order_release);
memset(rb->buf, 0, (rb->size_mask+1)*rb->elem_size);
}
size_t ll_ringbuffer_read_space(const ll_ringbuffer_t *rb)
{
size_t w = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->write_ptr, almemory_order_acquire);
size_t r = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->read_ptr, almemory_order_acquire);
return (w-r) & rb->size_mask;
}
size_t ll_ringbuffer_write_space(const ll_ringbuffer_t *rb)
{
size_t w = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->write_ptr, almemory_order_acquire);
size_t r = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->read_ptr, almemory_order_acquire);
w = (r-w-1) & rb->size_mask;
return (w > rb->size) ? rb->size : w;
}
size_t ll_ringbuffer_read(ll_ringbuffer_t *rb, char *dest, size_t cnt)
{
size_t read_ptr;
size_t free_cnt;
size_t cnt2;
size_t to_read;
size_t n1, n2;
free_cnt = ll_ringbuffer_read_space(rb);
if(free_cnt == 0) return 0;
to_read = (cnt > free_cnt) ? free_cnt : cnt;
read_ptr = ATOMIC_LOAD(&rb->read_ptr, almemory_order_relaxed) & rb->size_mask;
cnt2 = read_ptr + to_read;
if(cnt2 > rb->size_mask+1)
{
n1 = rb->size_mask+1 - read_ptr;
n2 = cnt2 & rb->size_mask;
}
else
{
n1 = to_read;
n2 = 0;
}
memcpy(dest, &rb->buf[read_ptr*rb->elem_size], n1*rb->elem_size);
read_ptr += n1;
if(n2)
{
memcpy(dest + n1*rb->elem_size, &rb->buf[(read_ptr&rb->size_mask)*rb->elem_size],
n2*rb->elem_size);
read_ptr += n2;
}
ATOMIC_STORE(&rb->read_ptr, read_ptr, almemory_order_release);
return to_read;
}
size_t ll_ringbuffer_peek(ll_ringbuffer_t *rb, char *dest, size_t cnt)
{
size_t free_cnt;
size_t cnt2;
size_t to_read;
size_t n1, n2;
size_t read_ptr;
free_cnt = ll_ringbuffer_read_space(rb);
if(free_cnt == 0) return 0;
to_read = (cnt > free_cnt) ? free_cnt : cnt;
read_ptr = ATOMIC_LOAD(&rb->read_ptr, almemory_order_relaxed) & rb->size_mask;
cnt2 = read_ptr + to_read;
if(cnt2 > rb->size_mask+1)
{
n1 = rb->size_mask+1 - read_ptr;
n2 = cnt2 & rb->size_mask;
}
else
{
n1 = to_read;
n2 = 0;
}
memcpy(dest, &rb->buf[read_ptr*rb->elem_size], n1*rb->elem_size);
if(n2)
{
read_ptr += n1;
memcpy(dest + n1*rb->elem_size, &rb->buf[(read_ptr&rb->size_mask)*rb->elem_size],
n2*rb->elem_size);
}
return to_read;
}
size_t ll_ringbuffer_write(ll_ringbuffer_t *rb, const char *src, size_t cnt)
{
size_t write_ptr;
size_t free_cnt;
size_t cnt2;
size_t to_write;
size_t n1, n2;
free_cnt = ll_ringbuffer_write_space(rb);
if(free_cnt == 0) return 0;
to_write = (cnt > free_cnt) ? free_cnt : cnt;
write_ptr = ATOMIC_LOAD(&rb->write_ptr, almemory_order_relaxed) & rb->size_mask;
cnt2 = write_ptr + to_write;
if(cnt2 > rb->size_mask+1)
{
n1 = rb->size_mask+1 - write_ptr;
n2 = cnt2 & rb->size_mask;
}
else
{
n1 = to_write;
n2 = 0;
}
memcpy(&rb->buf[write_ptr*rb->elem_size], src, n1*rb->elem_size);
write_ptr += n1;
if(n2)
{
memcpy(&rb->buf[(write_ptr&rb->size_mask)*rb->elem_size], src + n1*rb->elem_size,
n2*rb->elem_size);
write_ptr += n2;
}
ATOMIC_STORE(&rb->write_ptr, write_ptr, almemory_order_release);
return to_write;
}
void ll_ringbuffer_read_advance(ll_ringbuffer_t *rb, size_t cnt)
{
ATOMIC_ADD(&rb->read_ptr, cnt, almemory_order_acq_rel);
}
void ll_ringbuffer_write_advance(ll_ringbuffer_t *rb, size_t cnt)
{
ATOMIC_ADD(&rb->write_ptr, cnt, almemory_order_acq_rel);
}
void ll_ringbuffer_get_read_vector(const ll_ringbuffer_t *rb, ll_ringbuffer_data_t vec[2])
{
size_t free_cnt;
size_t cnt2;
size_t w, r;
w = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->write_ptr, almemory_order_acquire);
r = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->read_ptr, almemory_order_acquire);
w &= rb->size_mask;
r &= rb->size_mask;
free_cnt = (w-r) & rb->size_mask;
cnt2 = r + free_cnt;
if(cnt2 > rb->size_mask+1)
{
/* Two part vector: the rest of the buffer after the current write ptr,
* plus some from the start of the buffer. */
vec[0].buf = (char*)&rb->buf[r*rb->elem_size];
vec[0].len = rb->size_mask+1 - r;
vec[1].buf = (char*)rb->buf;
vec[1].len = cnt2 & rb->size_mask;
}
else
{
/* Single part vector: just the rest of the buffer */
vec[0].buf = (char*)&rb->buf[r*rb->elem_size];
vec[0].len = free_cnt;
vec[1].buf = NULL;
vec[1].len = 0;
}
}
void ll_ringbuffer_get_write_vector(const ll_ringbuffer_t *rb, ll_ringbuffer_data_t vec[2])
{
size_t free_cnt;
size_t cnt2;
size_t w, r;
w = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->write_ptr, almemory_order_acquire);
r = ATOMIC_LOAD(&CONST_CAST(ll_ringbuffer_t*,rb)->read_ptr, almemory_order_acquire);
w &= rb->size_mask;
r &= rb->size_mask;
free_cnt = (r-w-1) & rb->size_mask;
if(free_cnt > rb->size) free_cnt = rb->size;
cnt2 = w + free_cnt;
if(cnt2 > rb->size_mask+1)
{
/* Two part vector: the rest of the buffer after the current write ptr,
* plus some from the start of the buffer. */
vec[0].buf = (char*)&rb->buf[w*rb->elem_size];
vec[0].len = rb->size_mask+1 - w;
vec[1].buf = (char*)rb->buf;
vec[1].len = cnt2 & rb->size_mask;
}
else
{
vec[0].buf = (char*)&rb->buf[w*rb->elem_size];
vec[0].len = free_cnt;
vec[1].buf = NULL;
vec[1].len = 0;
}
}
-77
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@@ -1,77 +0,0 @@
#ifndef RINGBUFFER_H
#define RINGBUFFER_H
#include <stddef.h>
#ifdef __cplusplus
extern "C" {
#endif
typedef struct ll_ringbuffer ll_ringbuffer_t;
typedef struct ll_ringbuffer_data {
char *buf;
size_t len;
} ll_ringbuffer_data_t;
/**
* Create a new ringbuffer to hold at least `sz' elements of `elem_sz' bytes.
* The number of elements is rounded up to the next power of two (even if it is
* already a power of two, to ensure the requested amount can be written).
*/
ll_ringbuffer_t *ll_ringbuffer_create(size_t sz, size_t elem_sz, int limit_writes);
/** Free all data associated with the ringbuffer `rb'. */
void ll_ringbuffer_free(ll_ringbuffer_t *rb);
/** Reset the read and write pointers to zero. This is not thread safe. */
void ll_ringbuffer_reset(ll_ringbuffer_t *rb);
/**
* The non-copying data reader. `vec' is an array of two places. Set the values
* at `vec' to hold the current readable data at `rb'. If the readable data is
* in one segment the second segment has zero length.
*/
void ll_ringbuffer_get_read_vector(const ll_ringbuffer_t *rb, ll_ringbuffer_data_t vec[2]);
/**
* The non-copying data writer. `vec' is an array of two places. Set the values
* at `vec' to hold the current writeable data at `rb'. If the writeable data
* is in one segment the second segment has zero length.
*/
void ll_ringbuffer_get_write_vector(const ll_ringbuffer_t *rb, ll_ringbuffer_data_t vec[2]);
/**
* Return the number of elements available for reading. This is the number of
* elements in front of the read pointer and behind the write pointer.
*/
size_t ll_ringbuffer_read_space(const ll_ringbuffer_t *rb);
/**
* The copying data reader. Copy at most `cnt' elements from `rb' to `dest'.
* Returns the actual number of elements copied.
*/
size_t ll_ringbuffer_read(ll_ringbuffer_t *rb, char *dest, size_t cnt);
/**
* The copying data reader w/o read pointer advance. Copy at most `cnt'
* elements from `rb' to `dest'. Returns the actual number of elements copied.
*/
size_t ll_ringbuffer_peek(ll_ringbuffer_t *rb, char *dest, size_t cnt);
/** Advance the read pointer `cnt' places. */
void ll_ringbuffer_read_advance(ll_ringbuffer_t *rb, size_t cnt);
/**
* Return the number of elements available for writing. This is the number of
* elements in front of the write pointer and behind the read pointer.
*/
size_t ll_ringbuffer_write_space(const ll_ringbuffer_t *rb);
/**
* The copying data writer. Copy at most `cnt' elements to `rb' from `src'.
* Returns the actual number of elements copied.
*/
size_t ll_ringbuffer_write(ll_ringbuffer_t *rb, const char *src, size_t cnt);
/** Advance the write pointer `cnt' places. */
void ll_ringbuffer_write_advance(ll_ringbuffer_t *rb, size_t cnt);
#ifdef __cplusplus
} /* extern "C" */
#endif
#endif /* RINGBUFFER_H */
-120
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@@ -1,120 +0,0 @@
#include "config.h"
#include "alu.h"
#include "uhjfilter.h"
/* This is the maximum number of samples processed for each inner loop
* iteration. */
#define MAX_UPDATE_SAMPLES 128
static const ALfloat Filter1CoeffSqr[4] = {
0.479400865589f, 0.876218493539f, 0.976597589508f, 0.997499255936f
};
static const ALfloat Filter2CoeffSqr[4] = {
0.161758498368f, 0.733028932341f, 0.945349700329f, 0.990599156685f
};
static void allpass_process(AllPassState *state, ALfloat *restrict dst, const ALfloat *restrict src, const ALfloat aa, ALsizei todo)
{
ALfloat z1 = state->z[0];
ALfloat z2 = state->z[1];
ALsizei i;
for(i = 0;i < todo;i++)
{
ALfloat input = src[i];
ALfloat output = input*aa + z1;
z1 = z2; z2 = output*aa - input;
dst[i] = output;
}
state->z[0] = z1;
state->z[1] = z2;
}
/* NOTE: There seems to be a bit of an inconsistency in how this encoding is
* supposed to work. Some references, such as
*
* http://members.tripod.com/martin_leese/Ambisonic/UHJ_file_format.html
*
* specify a pre-scaling of sqrt(2) on the W channel input, while other
* references, such as
*
* https://en.wikipedia.org/wiki/Ambisonic_UHJ_format#Encoding.5B1.5D
* and
* https://wiki.xiph.org/Ambisonics#UHJ_format
*
* do not. The sqrt(2) scaling is in line with B-Format decoder coefficients
* which include such a scaling for the W channel input, however the original
* source for this equation is a 1985 paper by Michael Gerzon, which does not
* apparently include the scaling. Applying the extra scaling creates a louder
* result with a narrower stereo image compared to not scaling, and I don't
* know which is the intended result.
*/
void EncodeUhj2(Uhj2Encoder *enc, ALfloat *restrict LeftOut, ALfloat *restrict RightOut, ALfloat (*restrict InSamples)[BUFFERSIZE], ALsizei SamplesToDo)
{
ALfloat D[MAX_UPDATE_SAMPLES], S[MAX_UPDATE_SAMPLES];
ALfloat temp[2][MAX_UPDATE_SAMPLES];
ALsizei base, i;
ASSUME(SamplesToDo > 0);
for(base = 0;base < SamplesToDo;)
{
ALsizei todo = mini(SamplesToDo - base, MAX_UPDATE_SAMPLES);
ASSUME(todo > 0);
/* D = 0.6554516*Y */
for(i = 0;i < todo;i++)
temp[0][i] = 0.6554516f*InSamples[2][base+i];
allpass_process(&enc->Filter1_Y[0], temp[1], temp[0], Filter1CoeffSqr[0], todo);
allpass_process(&enc->Filter1_Y[1], temp[0], temp[1], Filter1CoeffSqr[1], todo);
allpass_process(&enc->Filter1_Y[2], temp[1], temp[0], Filter1CoeffSqr[2], todo);
allpass_process(&enc->Filter1_Y[3], temp[0], temp[1], Filter1CoeffSqr[3], todo);
/* NOTE: Filter1 requires a 1 sample delay for the final output, so
* take the last processed sample from the previous run as the first
* output sample.
*/
D[0] = enc->LastY;
for(i = 1;i < todo;i++)
D[i] = temp[0][i-1];
enc->LastY = temp[0][i-1];
/* D += j(-0.3420201*W + 0.5098604*X) */
for(i = 0;i < todo;i++)
temp[0][i] = -0.3420201f*InSamples[0][base+i] +
0.5098604f*InSamples[1][base+i];
allpass_process(&enc->Filter2_WX[0], temp[1], temp[0], Filter2CoeffSqr[0], todo);
allpass_process(&enc->Filter2_WX[1], temp[0], temp[1], Filter2CoeffSqr[1], todo);
allpass_process(&enc->Filter2_WX[2], temp[1], temp[0], Filter2CoeffSqr[2], todo);
allpass_process(&enc->Filter2_WX[3], temp[0], temp[1], Filter2CoeffSqr[3], todo);
for(i = 0;i < todo;i++)
D[i] += temp[0][i];
/* S = 0.9396926*W + 0.1855740*X */
for(i = 0;i < todo;i++)
temp[0][i] = 0.9396926f*InSamples[0][base+i] +
0.1855740f*InSamples[1][base+i];
allpass_process(&enc->Filter1_WX[0], temp[1], temp[0], Filter1CoeffSqr[0], todo);
allpass_process(&enc->Filter1_WX[1], temp[0], temp[1], Filter1CoeffSqr[1], todo);
allpass_process(&enc->Filter1_WX[2], temp[1], temp[0], Filter1CoeffSqr[2], todo);
allpass_process(&enc->Filter1_WX[3], temp[0], temp[1], Filter1CoeffSqr[3], todo);
S[0] = enc->LastWX;
for(i = 1;i < todo;i++)
S[i] = temp[0][i-1];
enc->LastWX = temp[0][i-1];
/* Left = (S + D)/2.0 */
for(i = 0;i < todo;i++)
*(LeftOut++) += (S[i] + D[i]) * 0.5f;
/* Right = (S - D)/2.0 */
for(i = 0;i < todo;i++)
*(RightOut++) += (S[i] - D[i]) * 0.5f;
base += todo;
}
}
-94
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@@ -1,94 +0,0 @@
#ifndef AL_VECTOR_H
#define AL_VECTOR_H
#include <stdlib.h>
#include <AL/al.h>
#include "almalloc.h"
#define TYPEDEF_VECTOR(T, N) typedef struct { \
size_t Capacity; \
size_t Size; \
T Data[]; \
} _##N; \
typedef _##N* N; \
typedef const _##N* const_##N;
#define VECTOR(T) struct { \
size_t Capacity; \
size_t Size; \
T Data[]; \
}*
#define VECTOR_INIT(_x) do { (_x) = NULL; } while(0)
#define VECTOR_INIT_STATIC() NULL
#define VECTOR_DEINIT(_x) do { al_free((_x)); (_x) = NULL; } while(0)
#define VECTOR_RESIZE(_x, _s, _c) do { \
size_t _size = (_s); \
size_t _cap = (_c); \
if(_size > _cap) \
_cap = _size; \
\
if(!(_x) && _cap == 0) \
break; \
\
if(((_x) ? (_x)->Capacity : 0) < _cap) \
{ \
ptrdiff_t data_offset = (_x) ? (char*)((_x)->Data) - (char*)(_x) : \
sizeof(*(_x)); \
size_t old_size = ((_x) ? (_x)->Size : 0); \
void *temp; \
\
temp = al_calloc(16, data_offset + sizeof((_x)->Data[0])*_cap); \
assert(temp != NULL); \
if((_x)) \
memcpy(((char*)temp)+data_offset, (_x)->Data, \
sizeof((_x)->Data[0])*old_size); \
\
al_free((_x)); \
(_x) = temp; \
(_x)->Capacity = _cap; \
} \
(_x)->Size = _size; \
} while(0) \
#define VECTOR_CAPACITY(_x) ((_x) ? (_x)->Capacity : 0)
#define VECTOR_SIZE(_x) ((_x) ? (_x)->Size : 0)
#define VECTOR_BEGIN(_x) ((_x) ? (_x)->Data + 0 : NULL)
#define VECTOR_END(_x) ((_x) ? (_x)->Data + (_x)->Size : NULL)
#define VECTOR_PUSH_BACK(_x, _obj) do { \
size_t _pbsize = VECTOR_SIZE(_x)+1; \
VECTOR_RESIZE(_x, _pbsize, _pbsize); \
(_x)->Data[(_x)->Size-1] = (_obj); \
} while(0)
#define VECTOR_POP_BACK(_x) ((void)((_x)->Size--))
#define VECTOR_BACK(_x) ((_x)->Data[(_x)->Size-1])
#define VECTOR_FRONT(_x) ((_x)->Data[0])
#define VECTOR_ELEM(_x, _o) ((_x)->Data[(_o)])
#define VECTOR_FOR_EACH(_t, _x, _f) do { \
_t *_iter = VECTOR_BEGIN((_x)); \
_t *_end = VECTOR_END((_x)); \
for(;_iter != _end;++_iter) \
_f(_iter); \
} while(0)
#define VECTOR_FIND_IF(_i, _t, _x, _f) do { \
_t *_iter = VECTOR_BEGIN((_x)); \
_t *_end = VECTOR_END((_x)); \
for(;_iter != _end;++_iter) \
{ \
if(_f(_iter)) \
break; \
} \
(_i) = _iter; \
} while(0)
#endif /* AL_VECTOR_H */
+533 -724
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+62
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@@ -1,3 +1,65 @@
openal-soft-1.20.0:
Converted the library codebase to C++11. A lot of hacks and custom
structures have been replaced with standard or cleaner implementations.
Partially implemented the Vocal Morpher effect.
Fixed the bsinc SSE resamplers on non-GCC compilers.
Fixed OpenSL capture.
Fixed support for extended capture formats with OpenSL.
Fixed handling of WASAPI not reporting a default device.
Fixed performance problems relating to semaphores on macOS.
Modified the bsinc12 resampler's transition band to better avoid aliasing
noise.
Modified alcResetDeviceSOFT to attempt recovery of disconnected devices.
Modified the virtual speaker layout for HRTF B-Format decoding.
Modified the PulseAudio backend to use a custom processing loop.
Renamed the makehrtf utility to makemhr.
Improved the efficiency of the bsinc resamplers when up-sampling.
Improved the quality of the bsinc resamplers slightly.
Improved the efficiency of the HRTF filters.
Improved the HRTF B-Format decoder coefficient generation.
Improved reverb feedback fading to be more consistent with pan fading.
Improved handling of sources that end prematurely, avoiding loud clicks.
Improved the performance of some reverb processing loops.
Added fast_bsinc12 and 24 resamplers that improve efficiency at the cost of
some quality. Notably, down-sampling has less smooth pitch ramping.
Added support for SOFA input files with makemhr.
Added a build option to use pre-built native tools. For cross-compiling,
use with caution and ensure the native tools' binaries are kept up-to-date.
Added an adjust-latency config option for the PulseAudio backend.
Added basic support for multi-field HRTFs.
Added an option for mixing first- or second-order B-Format with HRTF
output. This can improve HRTF performance given a number of sources.
Added an RC file for proper DLL version information.
Disabled some old KDE workarounds by default. Specifically, PulseAudio
streams can now be moved (KDE may try to move them after opening).
openal-soft-1.19.1:
Implemented capture support for the SoundIO backend.
-185
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@@ -1,185 +0,0 @@
#ifndef _AL_AUXEFFECTSLOT_H_
#define _AL_AUXEFFECTSLOT_H_
#include "alMain.h"
#include "alEffect.h"
#include "atomic.h"
#include "align.h"
#ifdef __cplusplus
extern "C" {
#endif
struct ALeffectStateVtable;
struct ALeffectslot;
typedef struct ALeffectState {
RefCount Ref;
const struct ALeffectStateVtable *vtbl;
ALfloat (*OutBuffer)[BUFFERSIZE];
ALsizei OutChannels;
} ALeffectState;
void ALeffectState_Construct(ALeffectState *state);
void ALeffectState_Destruct(ALeffectState *state);
struct ALeffectStateVtable {
void (*const Destruct)(ALeffectState *state);
ALboolean (*const deviceUpdate)(ALeffectState *state, ALCdevice *device);
void (*const update)(ALeffectState *state, const ALCcontext *context, const struct ALeffectslot *slot, const union ALeffectProps *props);
void (*const process)(ALeffectState *state, ALsizei samplesToDo, const ALfloat (*restrict samplesIn)[BUFFERSIZE], ALfloat (*restrict samplesOut)[BUFFERSIZE], ALsizei numChannels);
void (*const Delete)(void *ptr);
};
/* Small hack to use a pointer-to-array types as a normal argument type.
* Shouldn't be used directly.
*/
typedef ALfloat ALfloatBUFFERSIZE[BUFFERSIZE];
#define DEFINE_ALEFFECTSTATE_VTABLE(T) \
DECLARE_THUNK(T, ALeffectState, void, Destruct) \
DECLARE_THUNK1(T, ALeffectState, ALboolean, deviceUpdate, ALCdevice*) \
DECLARE_THUNK3(T, ALeffectState, void, update, const ALCcontext*, const ALeffectslot*, const ALeffectProps*) \
DECLARE_THUNK4(T, ALeffectState, void, process, ALsizei, const ALfloatBUFFERSIZE*restrict, ALfloatBUFFERSIZE*restrict, ALsizei) \
static void T##_ALeffectState_Delete(void *ptr) \
{ return T##_Delete(STATIC_UPCAST(T, ALeffectState, (ALeffectState*)ptr)); } \
\
static const struct ALeffectStateVtable T##_ALeffectState_vtable = { \
T##_ALeffectState_Destruct, \
\
T##_ALeffectState_deviceUpdate, \
T##_ALeffectState_update, \
T##_ALeffectState_process, \
\
T##_ALeffectState_Delete, \
}
struct EffectStateFactoryVtable;
typedef struct EffectStateFactory {
const struct EffectStateFactoryVtable *vtab;
} EffectStateFactory;
struct EffectStateFactoryVtable {
ALeffectState *(*const create)(EffectStateFactory *factory);
};
#define EffectStateFactory_create(x) ((x)->vtab->create((x)))
#define DEFINE_EFFECTSTATEFACTORY_VTABLE(T) \
DECLARE_THUNK(T, EffectStateFactory, ALeffectState*, create) \
\
static const struct EffectStateFactoryVtable T##_EffectStateFactory_vtable = { \
T##_EffectStateFactory_create, \
}
#define MAX_EFFECT_CHANNELS (4)
struct ALeffectslotArray {
ALsizei count;
struct ALeffectslot *slot[];
};
struct ALeffectslotProps {
ALfloat Gain;
ALboolean AuxSendAuto;
ALenum Type;
ALeffectProps Props;
ALeffectState *State;
ATOMIC(struct ALeffectslotProps*) next;
};
typedef struct ALeffectslot {
ALfloat Gain;
ALboolean AuxSendAuto;
struct {
ALenum Type;
ALeffectProps Props;
ALeffectState *State;
} Effect;
ATOMIC_FLAG PropsClean;
RefCount ref;
ATOMIC(struct ALeffectslotProps*) Update;
struct {
ALfloat Gain;
ALboolean AuxSendAuto;
ALenum EffectType;
ALeffectProps EffectProps;
ALeffectState *EffectState;
ALfloat RoomRolloff; /* Added to the source's room rolloff, not multiplied. */
ALfloat DecayTime;
ALfloat DecayLFRatio;
ALfloat DecayHFRatio;
ALboolean DecayHFLimit;
ALfloat AirAbsorptionGainHF;
} Params;
/* Self ID */
ALuint id;
ALsizei NumChannels;
BFChannelConfig ChanMap[MAX_EFFECT_CHANNELS];
/* Wet buffer configuration is ACN channel order with N3D scaling:
* * Channel 0 is the unattenuated mono signal.
* * Channel 1 is OpenAL -X * sqrt(3)
* * Channel 2 is OpenAL Y * sqrt(3)
* * Channel 3 is OpenAL -Z * sqrt(3)
* Consequently, effects that only want to work with mono input can use
* channel 0 by itself. Effects that want multichannel can process the
* ambisonics signal and make a B-Format source pan for first-order device
* output (FOAOut).
*/
alignas(16) ALfloat WetBuffer[MAX_EFFECT_CHANNELS][BUFFERSIZE];
} ALeffectslot;
ALenum InitEffectSlot(ALeffectslot *slot);
void DeinitEffectSlot(ALeffectslot *slot);
void UpdateEffectSlotProps(ALeffectslot *slot, ALCcontext *context);
void UpdateAllEffectSlotProps(ALCcontext *context);
ALvoid ReleaseALAuxiliaryEffectSlots(ALCcontext *Context);
EffectStateFactory *NullStateFactory_getFactory(void);
EffectStateFactory *ReverbStateFactory_getFactory(void);
EffectStateFactory *AutowahStateFactory_getFactory(void);
EffectStateFactory *ChorusStateFactory_getFactory(void);
EffectStateFactory *CompressorStateFactory_getFactory(void);
EffectStateFactory *DistortionStateFactory_getFactory(void);
EffectStateFactory *EchoStateFactory_getFactory(void);
EffectStateFactory *EqualizerStateFactory_getFactory(void);
EffectStateFactory *FlangerStateFactory_getFactory(void);
EffectStateFactory *FshifterStateFactory_getFactory(void);
EffectStateFactory *ModulatorStateFactory_getFactory(void);
EffectStateFactory *PshifterStateFactory_getFactory(void);
EffectStateFactory *DedicatedStateFactory_getFactory(void);
ALenum InitializeEffect(ALCcontext *Context, ALeffectslot *EffectSlot, ALeffect *effect);
void ALeffectState_DecRef(ALeffectState *state);
#ifdef __cplusplus
}
#endif
#endif
-115
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@@ -1,115 +0,0 @@
#ifndef _AL_BUFFER_H_
#define _AL_BUFFER_H_
#include "AL/alc.h"
#include "AL/al.h"
#include "AL/alext.h"
#include "inprogext.h"
#include "atomic.h"
#include "rwlock.h"
#ifdef __cplusplus
extern "C" {
#endif
/* User formats */
enum UserFmtType {
UserFmtUByte,
UserFmtShort,
UserFmtFloat,
UserFmtDouble,
UserFmtMulaw,
UserFmtAlaw,
UserFmtIMA4,
UserFmtMSADPCM,
};
enum UserFmtChannels {
UserFmtMono,
UserFmtStereo,
UserFmtRear,
UserFmtQuad,
UserFmtX51, /* (WFX order) */
UserFmtX61, /* (WFX order) */
UserFmtX71, /* (WFX order) */
UserFmtBFormat2D, /* WXY */
UserFmtBFormat3D, /* WXYZ */
};
ALsizei BytesFromUserFmt(enum UserFmtType type);
ALsizei ChannelsFromUserFmt(enum UserFmtChannels chans);
inline ALsizei FrameSizeFromUserFmt(enum UserFmtChannels chans, enum UserFmtType type)
{
return ChannelsFromUserFmt(chans) * BytesFromUserFmt(type);
}
/* Storable formats */
enum FmtType {
FmtUByte = UserFmtUByte,
FmtShort = UserFmtShort,
FmtFloat = UserFmtFloat,
FmtDouble = UserFmtDouble,
FmtMulaw = UserFmtMulaw,
FmtAlaw = UserFmtAlaw,
};
enum FmtChannels {
FmtMono = UserFmtMono,
FmtStereo = UserFmtStereo,
FmtRear = UserFmtRear,
FmtQuad = UserFmtQuad,
FmtX51 = UserFmtX51,
FmtX61 = UserFmtX61,
FmtX71 = UserFmtX71,
FmtBFormat2D = UserFmtBFormat2D,
FmtBFormat3D = UserFmtBFormat3D,
};
#define MAX_INPUT_CHANNELS (8)
ALsizei BytesFromFmt(enum FmtType type);
ALsizei ChannelsFromFmt(enum FmtChannels chans);
inline ALsizei FrameSizeFromFmt(enum FmtChannels chans, enum FmtType type)
{
return ChannelsFromFmt(chans) * BytesFromFmt(type);
}
typedef struct ALbuffer {
ALvoid *data;
ALsizei Frequency;
ALbitfieldSOFT Access;
ALsizei SampleLen;
enum FmtChannels FmtChannels;
enum FmtType FmtType;
ALsizei BytesAlloc;
enum UserFmtType OriginalType;
ALsizei OriginalSize;
ALsizei OriginalAlign;
ALsizei LoopStart;
ALsizei LoopEnd;
ATOMIC(ALsizei) UnpackAlign;
ATOMIC(ALsizei) PackAlign;
ALbitfieldSOFT MappedAccess;
ALsizei MappedOffset;
ALsizei MappedSize;
/* Number of times buffer was attached to a source (deletion can only occur when 0) */
RefCount ref;
/* Self ID */
ALuint id;
} ALbuffer;
ALvoid ReleaseALBuffers(ALCdevice *device);
#ifdef __cplusplus
}
#endif
#endif
-213
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@@ -1,213 +0,0 @@
#ifndef _AL_EFFECT_H_
#define _AL_EFFECT_H_
#include "alMain.h"
#ifdef __cplusplus
extern "C" {
#endif
struct ALeffect;
enum {
EAXREVERB_EFFECT = 0,
REVERB_EFFECT,
AUTOWAH_EFFECT,
CHORUS_EFFECT,
COMPRESSOR_EFFECT,
DISTORTION_EFFECT,
ECHO_EFFECT,
EQUALIZER_EFFECT,
FLANGER_EFFECT,
FSHIFTER_EFFECT,
MODULATOR_EFFECT,
PSHIFTER_EFFECT,
DEDICATED_EFFECT,
MAX_EFFECTS
};
extern ALboolean DisabledEffects[MAX_EFFECTS];
extern ALfloat ReverbBoost;
struct EffectList {
const char name[16];
int type;
ALenum val;
};
#define EFFECTLIST_SIZE 14
extern const struct EffectList EffectList[EFFECTLIST_SIZE];
struct ALeffectVtable {
void (*const setParami)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALint val);
void (*const setParamiv)(struct ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals);
void (*const setParamf)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val);
void (*const setParamfv)(struct ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals);
void (*const getParami)(const struct ALeffect *effect, ALCcontext *context, ALenum param, ALint *val);
void (*const getParamiv)(const struct ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals);
void (*const getParamf)(const struct ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val);
void (*const getParamfv)(const struct ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals);
};
#define DEFINE_ALEFFECT_VTABLE(T) \
const struct ALeffectVtable T##_vtable = { \
T##_setParami, T##_setParamiv, \
T##_setParamf, T##_setParamfv, \
T##_getParami, T##_getParamiv, \
T##_getParamf, T##_getParamfv, \
}
extern const struct ALeffectVtable ALeaxreverb_vtable;
extern const struct ALeffectVtable ALreverb_vtable;
extern const struct ALeffectVtable ALautowah_vtable;
extern const struct ALeffectVtable ALchorus_vtable;
extern const struct ALeffectVtable ALcompressor_vtable;
extern const struct ALeffectVtable ALdistortion_vtable;
extern const struct ALeffectVtable ALecho_vtable;
extern const struct ALeffectVtable ALequalizer_vtable;
extern const struct ALeffectVtable ALflanger_vtable;
extern const struct ALeffectVtable ALfshifter_vtable;
extern const struct ALeffectVtable ALmodulator_vtable;
extern const struct ALeffectVtable ALnull_vtable;
extern const struct ALeffectVtable ALpshifter_vtable;
extern const struct ALeffectVtable ALdedicated_vtable;
typedef union ALeffectProps {
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 AttackTime;
ALfloat ReleaseTime;
ALfloat Resonance;
ALfloat PeakGain;
} Autowah;
struct {
ALint Waveform;
ALint Phase;
ALfloat Rate;
ALfloat Depth;
ALfloat Feedback;
ALfloat Delay;
} Chorus; /* Also Flanger */
struct {
ALboolean OnOff;
} Compressor;
struct {
ALfloat Edge;
ALfloat Gain;
ALfloat LowpassCutoff;
ALfloat EQCenter;
ALfloat EQBandwidth;
} Distortion;
struct {
ALfloat Delay;
ALfloat LRDelay;
ALfloat Damping;
ALfloat Feedback;
ALfloat Spread;
} Echo;
struct {
ALfloat LowCutoff;
ALfloat LowGain;
ALfloat Mid1Center;
ALfloat Mid1Gain;
ALfloat Mid1Width;
ALfloat Mid2Center;
ALfloat Mid2Gain;
ALfloat Mid2Width;
ALfloat HighCutoff;
ALfloat HighGain;
} Equalizer;
struct {
ALfloat Frequency;
ALint LeftDirection;
ALint RightDirection;
} Fshifter;
struct {
ALfloat Frequency;
ALfloat HighPassCutoff;
ALint Waveform;
} Modulator;
struct {
ALint CoarseTune;
ALint FineTune;
} Pshifter;
struct {
ALfloat Gain;
} Dedicated;
} ALeffectProps;
typedef struct ALeffect {
// Effect type (AL_EFFECT_NULL, ...)
ALenum type;
ALeffectProps Props;
const struct ALeffectVtable *vtab;
/* Self ID */
ALuint id;
} ALeffect;
#define ALeffect_setParami(o, c, p, v) ((o)->vtab->setParami(o, c, p, v))
#define ALeffect_setParamf(o, c, p, v) ((o)->vtab->setParamf(o, c, p, v))
#define ALeffect_setParamiv(o, c, p, v) ((o)->vtab->setParamiv(o, c, p, v))
#define ALeffect_setParamfv(o, c, p, v) ((o)->vtab->setParamfv(o, c, p, v))
#define ALeffect_getParami(o, c, p, v) ((o)->vtab->getParami(o, c, p, v))
#define ALeffect_getParamf(o, c, p, v) ((o)->vtab->getParamf(o, c, p, v))
#define ALeffect_getParamiv(o, c, p, v) ((o)->vtab->getParamiv(o, c, p, v))
#define ALeffect_getParamfv(o, c, p, v) ((o)->vtab->getParamfv(o, c, p, v))
inline ALboolean IsReverbEffect(ALenum type)
{ return type == AL_EFFECT_REVERB || type == AL_EFFECT_EAXREVERB; }
void InitEffect(ALeffect *effect);
void ReleaseALEffects(ALCdevice *device);
void LoadReverbPreset(const char *name, ALeffect *effect);
#ifdef __cplusplus
}
#endif
#endif
-29
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#ifndef _AL_ERROR_H_
#define _AL_ERROR_H_
#include "alMain.h"
#include "logging.h"
#ifdef __cplusplus
extern "C" {
#endif
extern ALboolean TrapALError;
void alSetError(ALCcontext *context, ALenum errorCode, const char *msg, ...) DECL_FORMAT(printf, 3, 4);
#define SETERR_GOTO(ctx, err, lbl, ...) do { \
alSetError((ctx), (err), __VA_ARGS__); \
goto lbl; \
} while(0)
#define SETERR_RETURN(ctx, err, retval, ...) do { \
alSetError((ctx), (err), __VA_ARGS__); \
return retval; \
} while(0)
#ifdef __cplusplus
}
#endif
#endif
-67
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@@ -1,67 +0,0 @@
#ifndef _AL_FILTER_H_
#define _AL_FILTER_H_
#include "AL/alc.h"
#include "AL/al.h"
#ifdef __cplusplus
extern "C" {
#endif
#define LOWPASSFREQREF (5000.0f)
#define HIGHPASSFREQREF (250.0f)
struct ALfilter;
typedef struct ALfilterVtable {
void (*const setParami)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALint val);
void (*const setParamiv)(struct ALfilter *filter, ALCcontext *context, ALenum param, const ALint *vals);
void (*const setParamf)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val);
void (*const setParamfv)(struct ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals);
void (*const getParami)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALint *val);
void (*const getParamiv)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALint *vals);
void (*const getParamf)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *val);
void (*const getParamfv)(struct ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals);
} ALfilterVtable;
#define DEFINE_ALFILTER_VTABLE(T) \
const struct ALfilterVtable T##_vtable = { \
T##_setParami, T##_setParamiv, \
T##_setParamf, T##_setParamfv, \
T##_getParami, T##_getParamiv, \
T##_getParamf, T##_getParamfv, \
}
typedef struct ALfilter {
// Filter type (AL_FILTER_NULL, ...)
ALenum type;
ALfloat Gain;
ALfloat GainHF;
ALfloat HFReference;
ALfloat GainLF;
ALfloat LFReference;
const struct ALfilterVtable *vtab;
/* Self ID */
ALuint id;
} ALfilter;
#define ALfilter_setParami(o, c, p, v) ((o)->vtab->setParami(o, c, p, v))
#define ALfilter_setParamf(o, c, p, v) ((o)->vtab->setParamf(o, c, p, v))
#define ALfilter_setParamiv(o, c, p, v) ((o)->vtab->setParamiv(o, c, p, v))
#define ALfilter_setParamfv(o, c, p, v) ((o)->vtab->setParamfv(o, c, p, v))
#define ALfilter_getParami(o, c, p, v) ((o)->vtab->getParami(o, c, p, v))
#define ALfilter_getParamf(o, c, p, v) ((o)->vtab->getParamf(o, c, p, v))
#define ALfilter_getParamiv(o, c, p, v) ((o)->vtab->getParamiv(o, c, p, v))
#define ALfilter_getParamfv(o, c, p, v) ((o)->vtab->getParamfv(o, c, p, v))
void ReleaseALFilters(ALCdevice *device);
#ifdef __cplusplus
}
#endif
#endif
-67
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@@ -1,67 +0,0 @@
#ifndef _AL_LISTENER_H_
#define _AL_LISTENER_H_
#include "alMain.h"
#include "alu.h"
#ifdef __cplusplus
extern "C" {
#endif
struct ALcontextProps {
ALfloat DopplerFactor;
ALfloat DopplerVelocity;
ALfloat SpeedOfSound;
ALboolean SourceDistanceModel;
enum DistanceModel DistanceModel;
ALfloat MetersPerUnit;
ATOMIC(struct ALcontextProps*) next;
};
struct ALlistenerProps {
ALfloat Position[3];
ALfloat Velocity[3];
ALfloat Forward[3];
ALfloat Up[3];
ALfloat Gain;
ATOMIC(struct ALlistenerProps*) next;
};
typedef struct ALlistener {
alignas(16) ALfloat Position[3];
ALfloat Velocity[3];
ALfloat Forward[3];
ALfloat Up[3];
ALfloat Gain;
ATOMIC_FLAG PropsClean;
/* Pointer to the most recent property values that are awaiting an update.
*/
ATOMIC(struct ALlistenerProps*) Update;
struct {
aluMatrixf Matrix;
aluVector Velocity;
ALfloat Gain;
ALfloat MetersPerUnit;
ALfloat DopplerFactor;
ALfloat SpeedOfSound; /* in units per sec! */
ALfloat ReverbSpeedOfSound; /* in meters per sec! */
ALboolean SourceDistanceModel;
enum DistanceModel DistanceModel;
} Params;
} ALlistener;
void UpdateListenerProps(ALCcontext *context);
#ifdef __cplusplus
}
#endif
#endif
-920
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@@ -1,920 +0,0 @@
#ifndef AL_MAIN_H
#define AL_MAIN_H
#include <string.h>
#include <stdio.h>
#include <stddef.h>
#include <stdarg.h>
#include <assert.h>
#include <math.h>
#include <limits.h>
#ifdef HAVE_STRINGS_H
#include <strings.h>
#endif
#ifdef HAVE_INTRIN_H
#include <intrin.h>
#endif
#include "AL/al.h"
#include "AL/alc.h"
#include "AL/alext.h"
#include "inprogext.h"
#include "logging.h"
#include "polymorphism.h"
#include "static_assert.h"
#include "align.h"
#include "atomic.h"
#include "vector.h"
#include "alstring.h"
#include "almalloc.h"
#include "threads.h"
#if defined(_WIN64)
#define SZFMT "%I64u"
#elif defined(_WIN32)
#define SZFMT "%u"
#else
#define SZFMT "%zu"
#endif
#ifdef __has_builtin
#define HAS_BUILTIN __has_builtin
#else
#define HAS_BUILTIN(x) (0)
#endif
#ifdef __GNUC__
/* LIKELY optimizes the case where the condition is true. The condition is not
* required to be true, but it can result in more optimal code for the true
* path at the expense of a less optimal false path.
*/
#define LIKELY(x) __builtin_expect(!!(x), !0)
/* The opposite of LIKELY, optimizing the case where the condition is false. */
#define UNLIKELY(x) __builtin_expect(!!(x), 0)
/* Unlike LIKELY, ASSUME requires the condition to be true or else it invokes
* undefined behavior. It's essentially an assert without actually checking the
* condition at run-time, allowing for stronger optimizations than LIKELY.
*/
#if HAS_BUILTIN(__builtin_assume)
#define ASSUME __builtin_assume
#else
#define ASSUME(x) do { if(!(x)) __builtin_unreachable(); } while(0)
#endif
#else
#define LIKELY(x) (!!(x))
#define UNLIKELY(x) (!!(x))
#ifdef _MSC_VER
#define ASSUME __assume
#else
#define ASSUME(x) ((void)0)
#endif
#endif
#ifndef UINT64_MAX
#define UINT64_MAX U64(18446744073709551615)
#endif
#ifndef UNUSED
#if defined(__cplusplus)
#define UNUSED(x)
#elif defined(__GNUC__)
#define UNUSED(x) UNUSED_##x __attribute__((unused))
#elif defined(__LCLINT__)
#define UNUSED(x) /*@unused@*/ x
#else
#define UNUSED(x) x
#endif
#endif
/* Calculates the size of a struct with N elements of a flexible array member.
* GCC and Clang allow offsetof(Type, fam[N]) for this, but MSVC seems to have
* trouble, so a bit more verbose workaround is needed.
*/
#define FAM_SIZE(T, M, N) (offsetof(T, M) + sizeof(((T*)NULL)->M[0])*(N))
#ifdef __cplusplus
extern "C" {
#endif
typedef ALint64SOFT ALint64;
typedef ALuint64SOFT ALuint64;
#ifndef U64
#if defined(_MSC_VER)
#define U64(x) ((ALuint64)(x##ui64))
#elif SIZEOF_LONG == 8
#define U64(x) ((ALuint64)(x##ul))
#elif SIZEOF_LONG_LONG == 8
#define U64(x) ((ALuint64)(x##ull))
#endif
#endif
#ifndef I64
#if defined(_MSC_VER)
#define I64(x) ((ALint64)(x##i64))
#elif SIZEOF_LONG == 8
#define I64(x) ((ALint64)(x##l))
#elif SIZEOF_LONG_LONG == 8
#define I64(x) ((ALint64)(x##ll))
#endif
#endif
/* Define a CTZ64 macro (count trailing zeros, for 64-bit integers). The result
* is *UNDEFINED* if the value is 0.
*/
#ifdef __GNUC__
#if SIZEOF_LONG == 8
#define CTZ64 __builtin_ctzl
#else
#define CTZ64 __builtin_ctzll
#endif
#elif defined(HAVE_BITSCANFORWARD64_INTRINSIC)
inline int msvc64_ctz64(ALuint64 v)
{
unsigned long idx = 64;
_BitScanForward64(&idx, v);
return (int)idx;
}
#define CTZ64 msvc64_ctz64
#elif defined(HAVE_BITSCANFORWARD_INTRINSIC)
inline int msvc_ctz64(ALuint64 v)
{
unsigned long idx = 64;
if(!_BitScanForward(&idx, v&0xffffffff))
{
if(_BitScanForward(&idx, v>>32))
idx += 32;
}
return (int)idx;
}
#define CTZ64 msvc_ctz64
#else
/* There be black magics here. The popcnt64 method is derived from
* https://graphics.stanford.edu/~seander/bithacks.html#CountBitsSetParallel
* while the ctz-utilizing-popcnt algorithm is shown here
* http://www.hackersdelight.org/hdcodetxt/ntz.c.txt
* as the ntz2 variant. These likely aren't the most efficient methods, but
* they're good enough if the GCC or MSVC intrinsics aren't available.
*/
inline int fallback_popcnt64(ALuint64 v)
{
v = v - ((v >> 1) & U64(0x5555555555555555));
v = (v & U64(0x3333333333333333)) + ((v >> 2) & U64(0x3333333333333333));
v = (v + (v >> 4)) & U64(0x0f0f0f0f0f0f0f0f);
return (int)((v * U64(0x0101010101010101)) >> 56);
}
inline int fallback_ctz64(ALuint64 value)
{
return fallback_popcnt64(~value & (value - 1));
}
#define CTZ64 fallback_ctz64
#endif
#if defined(__BYTE_ORDER__) && defined(__ORDER_LITTLE_ENDIAN__)
#define IS_LITTLE_ENDIAN (__BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__)
#else
static const union {
ALuint u;
ALubyte b[sizeof(ALuint)];
} EndianTest = { 1 };
#define IS_LITTLE_ENDIAN (EndianTest.b[0] == 1)
#endif
#define COUNTOF(x) (sizeof(x) / sizeof(0[x]))
struct ll_ringbuffer;
struct Hrtf;
struct HrtfEntry;
struct DirectHrtfState;
struct FrontStablizer;
struct Compressor;
struct ALCbackend;
struct ALbuffer;
struct ALeffect;
struct ALfilter;
struct ALsource;
struct ALcontextProps;
struct ALlistenerProps;
struct ALvoiceProps;
struct ALeffectslotProps;
#define DEFAULT_OUTPUT_RATE (44100)
#define MIN_OUTPUT_RATE (8000)
/* Find the next power-of-2 for non-power-of-2 numbers. */
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;
}
/** Round up a value to the next multiple. */
inline size_t RoundUp(size_t value, size_t r)
{
value += r-1;
return value - (value%r);
}
/* Fast float-to-int conversion. No particular rounding mode is assumed; the
* IEEE-754 default is round-to-nearest with ties-to-even, though an app could
* change it on its own threads. On some systems, a truncating conversion may
* always be the fastest method.
*/
inline ALint fastf2i(ALfloat f)
{
#if defined(HAVE_INTRIN_H) && ((defined(_M_IX86_FP) && (_M_IX86_FP > 0)) || defined(_M_X64))
return _mm_cvt_ss2si(_mm_set1_ps(f));
#elif defined(_MSC_VER) && defined(_M_IX86_FP)
ALint i;
__asm fld f
__asm fistp i
return i;
#elif (defined(__GNUC__) || defined(__clang__)) && (defined(__i386__) || defined(__x86_64__))
ALint i;
#ifdef __SSE_MATH__
__asm__("cvtss2si %1, %0" : "=r"(i) : "x"(f));
#else
__asm__ __volatile__("fistpl %0" : "=m"(i) : "t"(f) : "st");
#endif
return i;
/* On GCC when compiling with -fno-math-errno, lrintf can be inlined to
* some simple instructions. Clang does not inline it, always generating a
* libc call, while MSVC's implementation is horribly slow, so always fall
* back to a normal integer conversion for them.
*/
#elif defined(HAVE_LRINTF) && !defined(_MSC_VER) && !defined(__clang__)
return lrintf(f);
#else
return (ALint)f;
#endif
}
/* Converts float-to-int using standard behavior (truncation). */
inline int float2int(float f)
{
#if ((defined(__GNUC__) || defined(__clang__)) && (defined(__i386__) || defined(__x86_64__)) && \
!defined(__SSE_MATH__)) || (defined(_MSC_VER) && defined(_M_IX86_FP) && _M_IX86_FP == 0)
ALint sign, shift, mant;
union {
ALfloat f;
ALint i;
} conv;
conv.f = f;
sign = (conv.i>>31) | 1;
shift = ((conv.i>>23)&0xff) - (127+23);
/* Over/underflow */
if(UNLIKELY(shift >= 31 || shift < -23))
return 0;
mant = (conv.i&0x7fffff) | 0x800000;
if(LIKELY(shift < 0))
return (mant >> -shift) * sign;
return (mant << shift) * sign;
#else
return (ALint)f;
#endif
}
/* Rounds a float to the nearest integral value, according to the current
* rounding mode. This is essentially an inlined version of rintf, although
* makes fewer promises (e.g. -0 or -0.25 rounded to 0 may result in +0).
*/
inline float fast_roundf(float f)
{
#if (defined(__GNUC__) || defined(__clang__)) && (defined(__i386__) || defined(__x86_64__)) && \
!defined(__SSE_MATH__)
float out;
__asm__ __volatile__("frndint" : "=t"(out) : "0"(f));
return out;
#else
/* Integral limit, where sub-integral precision is not available for
* floats.
*/
static const float ilim[2] = {
8388608.0f /* 0x1.0p+23 */,
-8388608.0f /* -0x1.0p+23 */
};
ALuint sign, expo;
union {
ALfloat f;
ALuint i;
} conv;
conv.f = f;
sign = (conv.i>>31)&0x01;
expo = (conv.i>>23)&0xff;
if(UNLIKELY(expo >= 150/*+23*/))
{
/* An exponent (base-2) of 23 or higher is incapable of sub-integral
* precision, so it's already an integral value. We don't need to worry
* about infinity or NaN here.
*/
return f;
}
/* Adding the integral limit to the value (with a matching sign) forces a
* result that has no sub-integral precision, and is consequently forced to
* round to an integral value. Removing the integral limit then restores
* the initial value rounded to the integral. The compiler should not
* optimize this out because of non-associative rules on floating-point
* math (as long as you don't use -fassociative-math,
* -funsafe-math-optimizations, -ffast-math, or -Ofast, in which case this
* may break).
*/
f += ilim[sign];
return f - ilim[sign];
#endif
}
enum DevProbe {
ALL_DEVICE_PROBE,
CAPTURE_DEVICE_PROBE
};
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 Channel {
FrontLeft = 0,
FrontRight,
FrontCenter,
LFE,
BackLeft,
BackRight,
BackCenter,
SideLeft,
SideRight,
UpperFrontLeft,
UpperFrontRight,
UpperBackLeft,
UpperBackRight,
LowerFrontLeft,
LowerFrontRight,
LowerBackLeft,
LowerBackRight,
Aux0,
Aux1,
Aux2,
Aux3,
Aux4,
Aux5,
Aux6,
Aux7,
Aux8,
Aux9,
Aux10,
Aux11,
Aux12,
Aux13,
Aux14,
Aux15,
InvalidChannel
};
/* 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,
DevFmtAmbi3D = ALC_BFORMAT3D_SOFT,
/* Similar to 5.1, except using rear channels instead of sides */
DevFmtX51Rear = 0x80000000,
DevFmtChannelsDefault = DevFmtStereo
};
#define MAX_OUTPUT_CHANNELS (16)
ALsizei BytesFromDevFmt(enum DevFmtType type);
ALsizei ChannelsFromDevFmt(enum DevFmtChannels chans, ALsizei ambiorder);
inline ALsizei FrameSizeFromDevFmt(enum DevFmtChannels chans, enum DevFmtType type, ALsizei ambiorder)
{
return ChannelsFromDevFmt(chans, ambiorder) * BytesFromDevFmt(type);
}
enum AmbiLayout {
AmbiLayout_FuMa = ALC_FUMA_SOFT, /* FuMa channel order */
AmbiLayout_ACN = ALC_ACN_SOFT, /* ACN channel order */
AmbiLayout_Default = AmbiLayout_ACN
};
enum AmbiNorm {
AmbiNorm_FuMa = ALC_FUMA_SOFT, /* FuMa normalization */
AmbiNorm_SN3D = ALC_SN3D_SOFT, /* SN3D normalization */
AmbiNorm_N3D = ALC_N3D_SOFT, /* N3D normalization */
AmbiNorm_Default = AmbiNorm_SN3D
};
enum DeviceType {
Playback,
Capture,
Loopback
};
enum RenderMode {
NormalRender,
StereoPair,
HrtfRender
};
/* The maximum number of Ambisonics coefficients. For a given order (o), the
* size needed will be (o+1)**2, thus zero-order has 1, first-order has 4,
* second-order has 9, third-order has 16, and fourth-order has 25.
*/
#define MAX_AMBI_ORDER 3
#define MAX_AMBI_COEFFS ((MAX_AMBI_ORDER+1) * (MAX_AMBI_ORDER+1))
/* A bitmask of ambisonic channels with height information. If none of these
* channels are used/needed, there's no height (e.g. with most surround sound
* speaker setups). This only specifies up to 4th order, which is the highest
* order a 32-bit mask value can specify (a 64-bit mask could handle up to 7th
* order). This is ACN ordering, with bit 0 being ACN 0, etc.
*/
#define AMBI_PERIPHONIC_MASK (0xfe7ce4)
/* The maximum number of Ambisonic coefficients for 2D (non-periphonic)
* representation. This is 2 per each order above zero-order, plus 1 for zero-
* order. Or simply, o*2 + 1.
*/
#define MAX_AMBI2D_COEFFS (MAX_AMBI_ORDER*2 + 1)
typedef ALfloat ChannelConfig[MAX_AMBI_COEFFS];
typedef struct BFChannelConfig {
ALfloat Scale;
ALsizei Index;
} BFChannelConfig;
typedef union AmbiConfig {
/* Ambisonic coefficients for mixing to the dry buffer. */
ChannelConfig Coeffs[MAX_OUTPUT_CHANNELS];
/* Coefficient channel mapping for mixing to the dry buffer. */
BFChannelConfig Map[MAX_OUTPUT_CHANNELS];
} AmbiConfig;
typedef struct BufferSubList {
ALuint64 FreeMask;
struct ALbuffer *Buffers; /* 64 */
} BufferSubList;
TYPEDEF_VECTOR(BufferSubList, vector_BufferSubList)
typedef struct EffectSubList {
ALuint64 FreeMask;
struct ALeffect *Effects; /* 64 */
} EffectSubList;
TYPEDEF_VECTOR(EffectSubList, vector_EffectSubList)
typedef struct FilterSubList {
ALuint64 FreeMask;
struct ALfilter *Filters; /* 64 */
} FilterSubList;
TYPEDEF_VECTOR(FilterSubList, vector_FilterSubList)
typedef struct SourceSubList {
ALuint64 FreeMask;
struct ALsource *Sources; /* 64 */
} SourceSubList;
TYPEDEF_VECTOR(SourceSubList, vector_SourceSubList)
/* Effect slots are rather large, and apps aren't likely to have more than one
* or two (let alone 64), so hold them individually.
*/
typedef struct ALeffectslot *ALeffectslotPtr;
TYPEDEF_VECTOR(ALeffectslotPtr, vector_ALeffectslotPtr)
typedef struct EnumeratedHrtf {
al_string name;
struct HrtfEntry *hrtf;
} EnumeratedHrtf;
TYPEDEF_VECTOR(EnumeratedHrtf, vector_EnumeratedHrtf)
/* Maximum delay in samples for speaker distance compensation. */
#define MAX_DELAY_LENGTH 1024
typedef struct DistanceComp {
ALfloat Gain;
ALsizei Length; /* Valid range is [0...MAX_DELAY_LENGTH). */
ALfloat *Buffer;
} DistanceComp;
/* 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.
*/
#define BUFFERSIZE 2048
typedef struct MixParams {
AmbiConfig Ambi;
/* Number of coefficients in each Ambi.Coeffs to mix together (4 for first-
* order, 9 for second-order, etc). If the count is 0, Ambi.Map is used
* instead to map each output to a coefficient index.
*/
ALsizei CoeffCount;
ALfloat (*Buffer)[BUFFERSIZE];
ALsizei NumChannels;
} MixParams;
typedef struct RealMixParams {
enum Channel ChannelName[MAX_OUTPUT_CHANNELS];
ALfloat (*Buffer)[BUFFERSIZE];
ALsizei NumChannels;
} RealMixParams;
typedef void (*POSTPROCESS)(ALCdevice *device, ALsizei SamplesToDo);
struct ALCdevice_struct {
RefCount ref;
ATOMIC(ALenum) Connected;
enum DeviceType Type;
ALuint Frequency;
ALuint UpdateSize;
ALuint NumUpdates;
enum DevFmtChannels FmtChans;
enum DevFmtType FmtType;
ALboolean IsHeadphones;
ALsizei AmbiOrder;
/* For DevFmtAmbi* output only, specifies the channel order and
* normalization.
*/
enum AmbiLayout AmbiLayout;
enum AmbiNorm AmbiScale;
ALCenum LimiterState;
al_string DeviceName;
ATOMIC(ALCenum) LastError;
// Maximum number of sources that can be created
ALuint SourcesMax;
// Maximum number of slots that can be created
ALuint AuxiliaryEffectSlotMax;
ALCuint NumMonoSources;
ALCuint NumStereoSources;
ALsizei NumAuxSends;
// Map of Buffers for this device
vector_BufferSubList BufferList;
almtx_t BufferLock;
// Map of Effects for this device
vector_EffectSubList EffectList;
almtx_t EffectLock;
// Map of Filters for this device
vector_FilterSubList FilterList;
almtx_t FilterLock;
POSTPROCESS PostProcess;
/* HRTF state and info */
struct DirectHrtfState *Hrtf;
al_string HrtfName;
struct Hrtf *HrtfHandle;
vector_EnumeratedHrtf HrtfList;
ALCenum HrtfStatus;
/* UHJ encoder state */
struct Uhj2Encoder *Uhj_Encoder;
/* High quality Ambisonic decoder */
struct BFormatDec *AmbiDecoder;
/* Stereo-to-binaural filter */
struct bs2b *Bs2b;
/* First-order ambisonic upsampler for higher-order output */
struct AmbiUpsampler *AmbiUp;
/* Rendering mode. */
enum RenderMode Render_Mode;
// Device flags
ALuint Flags;
ALuint64 ClockBase;
ALuint SamplesDone;
ALuint FixedLatency;
/* Temp storage used for mixer processing. */
alignas(16) ALfloat TempBuffer[4][BUFFERSIZE];
/* The "dry" path corresponds to the main output. */
MixParams Dry;
ALsizei NumChannelsPerOrder[MAX_AMBI_ORDER+1];
/* First-order ambisonics output, to be upsampled to the dry buffer if different. */
MixParams FOAOut;
/* "Real" output, which will be written to the device buffer. May alias the
* dry buffer.
*/
RealMixParams RealOut;
struct FrontStablizer *Stablizer;
struct Compressor *Limiter;
/* The average speaker distance as determined by the ambdec configuration
* (or alternatively, by the NFC-HOA reference delay). Only used for NFC.
*/
ALfloat AvgSpeakerDist;
/* Delay buffers used to compensate for speaker distances. */
DistanceComp ChannelDelay[MAX_OUTPUT_CHANNELS];
/* Dithering control. */
ALfloat DitherDepth;
ALuint DitherSeed;
/* Running count of the mixer invocations, in 31.1 fixed point. This
* actually increments *twice* when mixing, first at the start and then at
* the end, so the bottom bit indicates if the device is currently mixing
* and the upper bits indicates how many mixes have been done.
*/
RefCount MixCount;
// Contexts created on this device
ATOMIC(ALCcontext*) ContextList;
almtx_t BackendLock;
struct ALCbackend *Backend;
ATOMIC(ALCdevice*) next;
};
// Frequency was requested by the app or config file
#define DEVICE_FREQUENCY_REQUEST (1u<<1)
// Channel configuration was requested by the config file
#define DEVICE_CHANNELS_REQUEST (1u<<2)
// Sample type was requested by the config file
#define DEVICE_SAMPLE_TYPE_REQUEST (1u<<3)
// Specifies if the DSP is paused at user request
#define DEVICE_PAUSED (1u<<30)
// Specifies if the device is currently running
#define DEVICE_RUNNING (1u<<31)
/* Nanosecond resolution for the device clock time. */
#define DEVICE_CLOCK_RES U64(1000000000)
/* Must be less than 15 characters (16 including terminating null) for
* compatibility with pthread_setname_np limitations. */
#define MIXER_THREAD_NAME "alsoft-mixer"
#define RECORD_THREAD_NAME "alsoft-record"
enum {
/* End event thread processing. */
EventType_KillThread = 0,
/* User event types. */
EventType_SourceStateChange = 1<<0,
EventType_BufferCompleted = 1<<1,
EventType_Error = 1<<2,
EventType_Performance = 1<<3,
EventType_Deprecated = 1<<4,
EventType_Disconnected = 1<<5,
/* Internal events. */
EventType_ReleaseEffectState = 65536,
};
typedef struct AsyncEvent {
unsigned int EnumType;
union {
char dummy;
struct {
ALenum type;
ALuint id;
ALuint param;
ALchar msg[1008];
} user;
struct ALeffectState *EffectState;
} u;
} AsyncEvent;
#define ASYNC_EVENT(t) { t, { 0 } }
struct ALCcontext_struct {
RefCount ref;
struct ALlistener *Listener;
vector_SourceSubList SourceList;
ALuint NumSources;
almtx_t SourceLock;
vector_ALeffectslotPtr EffectSlotList;
almtx_t EffectSlotLock;
ATOMIC(ALenum) LastError;
enum DistanceModel DistanceModel;
ALboolean SourceDistanceModel;
ALfloat DopplerFactor;
ALfloat DopplerVelocity;
ALfloat SpeedOfSound;
ALfloat MetersPerUnit;
ATOMIC_FLAG PropsClean;
ATOMIC(ALenum) DeferUpdates;
almtx_t PropLock;
/* Counter for the pre-mixing updates, in 31.1 fixed point (lowest bit
* indicates if updates are currently happening).
*/
RefCount UpdateCount;
ATOMIC(ALenum) HoldUpdates;
ALfloat GainBoost;
ATOMIC(struct ALcontextProps*) Update;
/* Linked lists of unused property containers, free to use for future
* updates.
*/
ATOMIC(struct ALcontextProps*) FreeContextProps;
ATOMIC(struct ALlistenerProps*) FreeListenerProps;
ATOMIC(struct ALvoiceProps*) FreeVoiceProps;
ATOMIC(struct ALeffectslotProps*) FreeEffectslotProps;
struct ALvoice **Voices;
ALsizei VoiceCount;
ALsizei MaxVoices;
ATOMIC(struct ALeffectslotArray*) ActiveAuxSlots;
althrd_t EventThread;
alsem_t EventSem;
struct ll_ringbuffer *AsyncEvents;
ATOMIC(ALbitfieldSOFT) EnabledEvts;
almtx_t EventCbLock;
ALEVENTPROCSOFT EventCb;
void *EventParam;
/* Default effect slot */
struct ALeffectslot *DefaultSlot;
ALCdevice *Device;
const ALCchar *ExtensionList;
ATOMIC(ALCcontext*) next;
/* Memory space used by the listener (and possibly default effect slot) */
alignas(16) ALCbyte _listener_mem[];
};
ALCcontext *GetContextRef(void);
void ALCcontext_DecRef(ALCcontext *context);
void ALCcontext_DeferUpdates(ALCcontext *context);
void ALCcontext_ProcessUpdates(ALCcontext *context);
void AllocateVoices(ALCcontext *context, ALsizei num_voices, ALsizei old_sends);
extern ALint RTPrioLevel;
void SetRTPriority(void);
void SetDefaultChannelOrder(ALCdevice *device);
void SetDefaultWFXChannelOrder(ALCdevice *device);
const ALCchar *DevFmtTypeString(enum DevFmtType type);
const ALCchar *DevFmtChannelsString(enum DevFmtChannels chans);
inline ALint GetChannelIndex(const enum Channel names[MAX_OUTPUT_CHANNELS], enum Channel chan)
{
ALint i;
for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
{
if(names[i] == chan)
return i;
}
return -1;
}
/**
* GetChannelIdxByName
*
* Returns the index for the given channel name (e.g. FrontCenter), or -1 if it
* doesn't exist.
*/
inline ALint GetChannelIdxByName(const RealMixParams *real, enum Channel chan)
{ return GetChannelIndex(real->ChannelName, chan); }
inline void LockBufferList(ALCdevice *device) { almtx_lock(&device->BufferLock); }
inline void UnlockBufferList(ALCdevice *device) { almtx_unlock(&device->BufferLock); }
inline void LockEffectList(ALCdevice *device) { almtx_lock(&device->EffectLock); }
inline void UnlockEffectList(ALCdevice *device) { almtx_unlock(&device->EffectLock); }
inline void LockFilterList(ALCdevice *device) { almtx_lock(&device->FilterLock); }
inline void UnlockFilterList(ALCdevice *device) { almtx_unlock(&device->FilterLock); }
inline void LockEffectSlotList(ALCcontext *context)
{ almtx_lock(&context->EffectSlotLock); }
inline void UnlockEffectSlotList(ALCcontext *context)
{ almtx_unlock(&context->EffectSlotLock); }
int EventThread(void *arg);
vector_al_string SearchDataFiles(const char *match, const char *subdir);
#ifdef __cplusplus
}
#endif
#endif
-120
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@@ -1,120 +0,0 @@
#ifndef _AL_SOURCE_H_
#define _AL_SOURCE_H_
#include "bool.h"
#include "alMain.h"
#include "alu.h"
#include "hrtf.h"
#include "atomic.h"
#define MAX_SENDS 16
#define DEFAULT_SENDS 2
#ifdef __cplusplus
extern "C" {
#endif
struct ALbuffer;
struct ALsource;
typedef struct ALbufferlistitem {
ATOMIC(struct ALbufferlistitem*) next;
ALsizei max_samples;
ALsizei num_buffers;
struct ALbuffer *buffers[];
} ALbufferlistitem;
typedef struct ALsource {
/** Source properties. */
ALfloat Pitch;
ALfloat Gain;
ALfloat OuterGain;
ALfloat MinGain;
ALfloat MaxGain;
ALfloat InnerAngle;
ALfloat OuterAngle;
ALfloat RefDistance;
ALfloat MaxDistance;
ALfloat RolloffFactor;
ALfloat Position[3];
ALfloat Velocity[3];
ALfloat Direction[3];
ALfloat Orientation[2][3];
ALboolean HeadRelative;
ALboolean Looping;
enum DistanceModel DistanceModel;
enum Resampler Resampler;
ALboolean DirectChannels;
enum SpatializeMode Spatialize;
ALboolean DryGainHFAuto;
ALboolean WetGainAuto;
ALboolean WetGainHFAuto;
ALfloat OuterGainHF;
ALfloat AirAbsorptionFactor;
ALfloat RoomRolloffFactor;
ALfloat DopplerFactor;
/* NOTE: Stereo pan angles are specified in radians, counter-clockwise
* rather than clockwise.
*/
ALfloat StereoPan[2];
ALfloat Radius;
/** Direct filter and auxiliary send info. */
struct {
ALfloat Gain;
ALfloat GainHF;
ALfloat HFReference;
ALfloat GainLF;
ALfloat LFReference;
} Direct;
struct {
struct ALeffectslot *Slot;
ALfloat Gain;
ALfloat GainHF;
ALfloat HFReference;
ALfloat GainLF;
ALfloat LFReference;
} *Send;
/**
* Last user-specified offset, and the offset type (bytes, samples, or
* seconds).
*/
ALdouble Offset;
ALenum OffsetType;
/** Source type (static, streaming, or undetermined) */
ALint SourceType;
/** Source state (initial, playing, paused, or stopped) */
ALenum state;
/** Source Buffer Queue head. */
ALbufferlistitem *queue;
ATOMIC_FLAG PropsClean;
/* Index into the context's Voices array. Lazily updated, only checked and
* reset when looking up the voice.
*/
ALint VoiceIdx;
/** Self ID */
ALuint id;
} ALsource;
void UpdateAllSourceProps(ALCcontext *context);
ALvoid ReleaseALSources(ALCcontext *Context);
#ifdef __cplusplus
}
#endif
#endif
-534
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@@ -1,534 +0,0 @@
#ifndef _ALU_H_
#define _ALU_H_
#include <limits.h>
#include <math.h>
#ifdef HAVE_FLOAT_H
#include <float.h>
#endif
#ifdef HAVE_IEEEFP_H
#include <ieeefp.h>
#endif
#include "alMain.h"
#include "alBuffer.h"
#include "hrtf.h"
#include "align.h"
#include "math_defs.h"
#include "filters/defs.h"
#include "filters/nfc.h"
#define MAX_PITCH (255)
/* Maximum number of samples to pad on either end of a buffer for resampling.
* Note that both the beginning and end need padding!
*/
#define MAX_RESAMPLE_PADDING 24
#ifdef __cplusplus
extern "C" {
#endif
struct BSincTable;
struct ALsource;
struct ALbufferlistitem;
struct ALvoice;
struct ALeffectslot;
#define DITHER_RNG_SEED 22222
enum SpatializeMode {
SpatializeOff = AL_FALSE,
SpatializeOn = AL_TRUE,
SpatializeAuto = AL_AUTO_SOFT
};
enum Resampler {
PointResampler,
LinearResampler,
FIR4Resampler,
BSinc12Resampler,
BSinc24Resampler,
ResamplerMax = BSinc24Resampler
};
extern enum Resampler ResamplerDefault;
/* The number of distinct scale and phase intervals within the bsinc filter
* table.
*/
#define BSINC_SCALE_BITS 4
#define BSINC_SCALE_COUNT (1<<BSINC_SCALE_BITS)
#define BSINC_PHASE_BITS 4
#define BSINC_PHASE_COUNT (1<<BSINC_PHASE_BITS)
/* Interpolator state. Kind of a misnomer since the interpolator itself is
* stateless. This just keeps it from having to recompute scale-related
* mappings for every sample.
*/
typedef struct BsincState {
ALfloat sf; /* Scale interpolation factor. */
ALsizei m; /* Coefficient count. */
ALsizei l; /* Left coefficient offset. */
/* Filter coefficients, followed by the scale, phase, and scale-phase
* delta coefficients. Starting at phase index 0, each subsequent phase
* index follows contiguously.
*/
const ALfloat *filter;
} BsincState;
typedef union InterpState {
BsincState bsinc;
} InterpState;
typedef const ALfloat* (*ResamplerFunc)(const InterpState *state,
const ALfloat *restrict src, ALsizei frac, ALint increment,
ALfloat *restrict dst, ALsizei dstlen
);
void BsincPrepare(const ALuint increment, BsincState *state, const struct BSincTable *table);
extern const struct BSincTable bsinc12;
extern const struct BSincTable bsinc24;
typedef union aluVector {
alignas(16) ALfloat v[4];
} aluVector;
inline void aluVectorSet(aluVector *vector, ALfloat x, ALfloat y, ALfloat z, ALfloat w)
{
vector->v[0] = x;
vector->v[1] = y;
vector->v[2] = z;
vector->v[3] = w;
}
typedef union aluMatrixf {
alignas(16) ALfloat m[4][4];
} aluMatrixf;
extern const aluMatrixf IdentityMatrixf;
inline void aluMatrixfSetRow(aluMatrixf *matrix, ALuint row,
ALfloat m0, ALfloat m1, ALfloat m2, ALfloat m3)
{
matrix->m[row][0] = m0;
matrix->m[row][1] = m1;
matrix->m[row][2] = m2;
matrix->m[row][3] = m3;
}
inline void aluMatrixfSet(aluMatrixf *matrix, ALfloat m00, ALfloat m01, ALfloat m02, ALfloat m03,
ALfloat m10, ALfloat m11, ALfloat m12, ALfloat m13,
ALfloat m20, ALfloat m21, ALfloat m22, ALfloat m23,
ALfloat m30, ALfloat m31, ALfloat m32, ALfloat m33)
{
aluMatrixfSetRow(matrix, 0, m00, m01, m02, m03);
aluMatrixfSetRow(matrix, 1, m10, m11, m12, m13);
aluMatrixfSetRow(matrix, 2, m20, m21, m22, m23);
aluMatrixfSetRow(matrix, 3, m30, m31, m32, m33);
}
enum ActiveFilters {
AF_None = 0,
AF_LowPass = 1,
AF_HighPass = 2,
AF_BandPass = AF_LowPass | AF_HighPass
};
typedef struct MixHrtfParams {
const ALfloat (*Coeffs)[2];
ALsizei Delay[2];
ALfloat Gain;
ALfloat GainStep;
} MixHrtfParams;
typedef struct DirectParams {
BiquadFilter LowPass;
BiquadFilter HighPass;
NfcFilter NFCtrlFilter;
struct {
HrtfParams Old;
HrtfParams Target;
HrtfState State;
} Hrtf;
struct {
ALfloat Current[MAX_OUTPUT_CHANNELS];
ALfloat Target[MAX_OUTPUT_CHANNELS];
} Gains;
} DirectParams;
typedef struct SendParams {
BiquadFilter LowPass;
BiquadFilter HighPass;
struct {
ALfloat Current[MAX_OUTPUT_CHANNELS];
ALfloat Target[MAX_OUTPUT_CHANNELS];
} Gains;
} SendParams;
struct ALvoiceProps {
ATOMIC(struct ALvoiceProps*) next;
ALfloat Pitch;
ALfloat Gain;
ALfloat OuterGain;
ALfloat MinGain;
ALfloat MaxGain;
ALfloat InnerAngle;
ALfloat OuterAngle;
ALfloat RefDistance;
ALfloat MaxDistance;
ALfloat RolloffFactor;
ALfloat Position[3];
ALfloat Velocity[3];
ALfloat Direction[3];
ALfloat Orientation[2][3];
ALboolean HeadRelative;
enum DistanceModel DistanceModel;
enum Resampler Resampler;
ALboolean DirectChannels;
enum SpatializeMode SpatializeMode;
ALboolean DryGainHFAuto;
ALboolean WetGainAuto;
ALboolean WetGainHFAuto;
ALfloat OuterGainHF;
ALfloat AirAbsorptionFactor;
ALfloat RoomRolloffFactor;
ALfloat DopplerFactor;
ALfloat StereoPan[2];
ALfloat Radius;
/** Direct filter and auxiliary send info. */
struct {
ALfloat Gain;
ALfloat GainHF;
ALfloat HFReference;
ALfloat GainLF;
ALfloat LFReference;
} Direct;
struct {
struct ALeffectslot *Slot;
ALfloat Gain;
ALfloat GainHF;
ALfloat HFReference;
ALfloat GainLF;
ALfloat LFReference;
} Send[];
};
#define VOICE_IS_STATIC (1<<0)
#define VOICE_IS_FADING (1<<1) /* Fading sources use gain stepping for smooth transitions. */
#define VOICE_HAS_HRTF (1<<2)
#define VOICE_HAS_NFC (1<<3)
typedef struct ALvoice {
struct ALvoiceProps *Props;
ATOMIC(struct ALvoiceProps*) Update;
ATOMIC(struct ALsource*) Source;
ATOMIC(bool) Playing;
/**
* Source offset in samples, relative to the currently playing buffer, NOT
* the whole queue, and the fractional (fixed-point) offset to the next
* sample.
*/
ATOMIC(ALuint) position;
ATOMIC(ALsizei) position_fraction;
/* Current buffer queue item being played. */
ATOMIC(struct ALbufferlistitem*) current_buffer;
/* Buffer queue item to loop to at end of queue (will be NULL for non-
* looping voices).
*/
ATOMIC(struct ALbufferlistitem*) loop_buffer;
/**
* Number of channels and bytes-per-sample for the attached source's
* buffer(s).
*/
ALsizei NumChannels;
ALsizei SampleSize;
/** Current target parameters used for mixing. */
ALint Step;
ResamplerFunc Resampler;
ALuint Flags;
ALuint Offset; /* Number of output samples mixed since starting. */
alignas(16) ALfloat PrevSamples[MAX_INPUT_CHANNELS][MAX_RESAMPLE_PADDING];
InterpState ResampleState;
struct {
enum ActiveFilters FilterType;
DirectParams Params[MAX_INPUT_CHANNELS];
ALfloat (*Buffer)[BUFFERSIZE];
ALsizei Channels;
ALsizei ChannelsPerOrder[MAX_AMBI_ORDER+1];
} Direct;
struct {
enum ActiveFilters FilterType;
SendParams Params[MAX_INPUT_CHANNELS];
ALfloat (*Buffer)[BUFFERSIZE];
ALsizei Channels;
} Send[];
} ALvoice;
void DeinitVoice(ALvoice *voice);
typedef void (*MixerFunc)(const ALfloat *data, ALsizei OutChans,
ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALfloat *CurrentGains,
const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
ALsizei BufferSize);
typedef void (*RowMixerFunc)(ALfloat *OutBuffer, const ALfloat *gains,
const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans,
ALsizei InPos, ALsizei BufferSize);
typedef void (*HrtfMixerFunc)(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
const ALfloat *data, ALsizei Offset, ALsizei OutPos,
const ALsizei IrSize, MixHrtfParams *hrtfparams,
HrtfState *hrtfstate, ALsizei BufferSize);
typedef void (*HrtfMixerBlendFunc)(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
const ALfloat *data, ALsizei Offset, ALsizei OutPos,
const ALsizei IrSize, const HrtfParams *oldparams,
MixHrtfParams *newparams, HrtfState *hrtfstate,
ALsizei BufferSize);
typedef void (*HrtfDirectMixerFunc)(ALfloat *restrict LeftOut, ALfloat *restrict RightOut,
const ALfloat *data, ALsizei Offset, const ALsizei IrSize,
const ALfloat (*restrict Coeffs)[2],
ALfloat (*restrict Values)[2], ALsizei BufferSize);
#define GAIN_MIX_MAX (16.0f) /* +24dB */
#define GAIN_SILENCE_THRESHOLD (0.00001f) /* -100dB */
#define SPEEDOFSOUNDMETRESPERSEC (343.3f)
#define AIRABSORBGAINHF (0.99426f) /* -0.05dB */
/* Target gain for the reverb decay feedback reaching the decay time. */
#define REVERB_DECAY_GAIN (0.001f) /* -60 dB */
#define FRACTIONBITS (12)
#define FRACTIONONE (1<<FRACTIONBITS)
#define FRACTIONMASK (FRACTIONONE-1)
inline ALfloat minf(ALfloat a, ALfloat b)
{ return ((a > b) ? b : a); }
inline ALfloat maxf(ALfloat a, ALfloat b)
{ return ((a > b) ? a : b); }
inline ALfloat clampf(ALfloat val, ALfloat min, ALfloat max)
{ return minf(max, maxf(min, val)); }
inline ALdouble mind(ALdouble a, ALdouble b)
{ return ((a > b) ? b : a); }
inline ALdouble maxd(ALdouble a, ALdouble b)
{ return ((a > b) ? a : b); }
inline ALdouble clampd(ALdouble val, ALdouble min, ALdouble max)
{ return mind(max, maxd(min, val)); }
inline ALuint minu(ALuint a, ALuint b)
{ return ((a > b) ? b : a); }
inline ALuint maxu(ALuint a, ALuint b)
{ return ((a > b) ? a : b); }
inline ALuint clampu(ALuint val, ALuint min, ALuint max)
{ return minu(max, maxu(min, val)); }
inline ALint mini(ALint a, ALint b)
{ return ((a > b) ? b : a); }
inline ALint maxi(ALint a, ALint b)
{ return ((a > b) ? a : b); }
inline ALint clampi(ALint val, ALint min, ALint max)
{ return mini(max, maxi(min, val)); }
inline ALint64 mini64(ALint64 a, ALint64 b)
{ return ((a > b) ? b : a); }
inline ALint64 maxi64(ALint64 a, ALint64 b)
{ return ((a > b) ? a : b); }
inline ALint64 clampi64(ALint64 val, ALint64 min, ALint64 max)
{ return mini64(max, maxi64(min, val)); }
inline ALuint64 minu64(ALuint64 a, ALuint64 b)
{ return ((a > b) ? b : a); }
inline ALuint64 maxu64(ALuint64 a, ALuint64 b)
{ return ((a > b) ? a : b); }
inline ALuint64 clampu64(ALuint64 val, ALuint64 min, ALuint64 max)
{ return minu64(max, maxu64(min, val)); }
inline size_t minz(size_t a, size_t b)
{ return ((a > b) ? b : a); }
inline size_t maxz(size_t a, size_t b)
{ return ((a > b) ? a : b); }
inline size_t clampz(size_t val, size_t min, size_t max)
{ return minz(max, maxz(min, val)); }
inline ALfloat lerp(ALfloat val1, ALfloat val2, ALfloat mu)
{
return val1 + (val2-val1)*mu;
}
inline ALfloat cubic(ALfloat val1, ALfloat val2, ALfloat val3, ALfloat val4, ALfloat mu)
{
ALfloat mu2 = mu*mu, mu3 = mu2*mu;
ALfloat a0 = -0.5f*mu3 + mu2 + -0.5f*mu;
ALfloat a1 = 1.5f*mu3 + -2.5f*mu2 + 1.0f;
ALfloat a2 = -1.5f*mu3 + 2.0f*mu2 + 0.5f*mu;
ALfloat a3 = 0.5f*mu3 + -0.5f*mu2;
return val1*a0 + val2*a1 + val3*a2 + val4*a3;
}
enum HrtfRequestMode {
Hrtf_Default = 0,
Hrtf_Enable = 1,
Hrtf_Disable = 2,
};
void aluInit(void);
void aluInitMixer(void);
ResamplerFunc SelectResampler(enum Resampler resampler);
/* aluInitRenderer
*
* Set up the appropriate panning method and mixing method given the device
* properties.
*/
void aluInitRenderer(ALCdevice *device, ALint hrtf_id, enum HrtfRequestMode hrtf_appreq, enum HrtfRequestMode hrtf_userreq);
void aluInitEffectPanning(struct ALeffectslot *slot);
void aluSelectPostProcess(ALCdevice *device);
/**
* Calculates ambisonic encoder coefficients using the X, Y, and Z direction
* components, which must represent a normalized (unit length) vector, and the
* spread is the angular width of the sound (0...tau).
*
* NOTE: The components use ambisonic coordinates. As a result:
*
* Ambisonic Y = OpenAL -X
* Ambisonic Z = OpenAL Y
* Ambisonic X = OpenAL -Z
*
* The components are ordered such that OpenAL's X, Y, and Z are the first,
* second, and third parameters respectively -- simply negate X and Z.
*/
void CalcAmbiCoeffs(const ALfloat y, const ALfloat z, const ALfloat x, const ALfloat spread,
ALfloat coeffs[MAX_AMBI_COEFFS]);
/**
* CalcDirectionCoeffs
*
* Calculates ambisonic coefficients based on an OpenAL direction vector. The
* vector must be normalized (unit length), and the spread is the angular width
* of the sound (0...tau).
*/
inline void CalcDirectionCoeffs(const ALfloat dir[3], ALfloat spread, ALfloat coeffs[MAX_AMBI_COEFFS])
{
/* Convert from OpenAL coords to Ambisonics. */
CalcAmbiCoeffs(-dir[0], dir[1], -dir[2], spread, coeffs);
}
/**
* CalcAngleCoeffs
*
* Calculates ambisonic coefficients based on azimuth and elevation. The
* azimuth and elevation parameters are in radians, going right and up
* respectively.
*/
inline void CalcAngleCoeffs(ALfloat azimuth, ALfloat elevation, ALfloat spread, ALfloat coeffs[MAX_AMBI_COEFFS])
{
ALfloat x = -sinf(azimuth) * cosf(elevation);
ALfloat y = sinf(elevation);
ALfloat z = cosf(azimuth) * cosf(elevation);
CalcAmbiCoeffs(x, y, z, spread, coeffs);
}
/**
* ScaleAzimuthFront
*
* Scales the given azimuth toward the side (+/- pi/2 radians) for positions in
* front.
*/
inline float ScaleAzimuthFront(float azimuth, float scale)
{
ALfloat sign = copysignf(1.0f, azimuth);
if(!(fabsf(azimuth) > F_PI_2))
return minf(fabsf(azimuth) * scale, F_PI_2) * sign;
return azimuth;
}
void ComputePanningGainsMC(const ChannelConfig *chancoeffs, ALsizei numchans, ALsizei numcoeffs, const ALfloat*restrict coeffs, ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]);
void ComputePanningGainsBF(const BFChannelConfig *chanmap, ALsizei numchans, const ALfloat*restrict coeffs, ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS]);
/**
* ComputePanGains
*
* Computes panning gains using the given channel decoder coefficients and the
* pre-calculated direction or angle coefficients. For B-Format sources, the
* coeffs are a 'slice' of a transform matrix for the input channel, used to
* scale and orient the sound samples.
*/
inline void ComputePanGains(const MixParams *dry, const ALfloat*restrict coeffs, ALfloat ingain, ALfloat gains[MAX_OUTPUT_CHANNELS])
{
if(dry->CoeffCount > 0)
ComputePanningGainsMC(dry->Ambi.Coeffs, dry->NumChannels, dry->CoeffCount,
coeffs, ingain, gains);
else
ComputePanningGainsBF(dry->Ambi.Map, dry->NumChannels, coeffs, ingain, gains);
}
ALboolean MixSource(struct ALvoice *voice, ALuint SourceID, ALCcontext *Context, ALsizei SamplesToDo);
void aluMixData(ALCdevice *device, ALvoid *OutBuffer, ALsizei NumSamples);
/* Caller must lock the device, and the mixer must not be running. */
void aluHandleDisconnect(ALCdevice *device, const char *msg, ...) DECL_FORMAT(printf, 2, 3);
void UpdateContextProps(ALCcontext *context);
extern MixerFunc MixSamples;
extern RowMixerFunc MixRowSamples;
extern ALfloat ConeScale;
extern ALfloat ZScale;
extern ALboolean OverrideReverbSpeedOfSound;
#ifdef __cplusplus
}
#endif
#endif
-15
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@@ -1,15 +0,0 @@
#ifndef SAMPLE_CVT_H
#define SAMPLE_CVT_H
#include "AL/al.h"
#include "alBuffer.h"
extern const ALshort muLawDecompressionTable[256];
extern const ALshort aLawDecompressionTable[256];
void Convert_ALshort_ALima4(ALshort *dst, const ALubyte *src, ALsizei numchans, ALsizei len,
ALsizei align);
void Convert_ALshort_ALmsadpcm(ALshort *dst, const ALubyte *src, ALsizei numchans, ALsizei len,
ALsizei align);
#endif /* SAMPLE_CVT_H */
-802
View File
@@ -1,802 +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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 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 "alError.h"
#include "alListener.h"
#include "alSource.h"
#include "fpu_modes.h"
#include "almalloc.h"
extern inline void LockEffectSlotList(ALCcontext *context);
extern inline void UnlockEffectSlotList(ALCcontext *context);
static void AddActiveEffectSlots(const ALuint *slotids, ALsizei count, ALCcontext *context);
static void RemoveActiveEffectSlots(const ALuint *slotids, ALsizei count, ALCcontext *context);
static const struct {
ALenum Type;
EffectStateFactory* (*GetFactory)(void);
} FactoryList[] = {
{ AL_EFFECT_NULL, NullStateFactory_getFactory },
{ AL_EFFECT_EAXREVERB, ReverbStateFactory_getFactory },
{ AL_EFFECT_REVERB, ReverbStateFactory_getFactory },
{ AL_EFFECT_AUTOWAH, AutowahStateFactory_getFactory },
{ AL_EFFECT_CHORUS, ChorusStateFactory_getFactory },
{ AL_EFFECT_COMPRESSOR, CompressorStateFactory_getFactory },
{ AL_EFFECT_DISTORTION, DistortionStateFactory_getFactory },
{ AL_EFFECT_ECHO, EchoStateFactory_getFactory },
{ AL_EFFECT_EQUALIZER, EqualizerStateFactory_getFactory },
{ AL_EFFECT_FLANGER, FlangerStateFactory_getFactory },
{ AL_EFFECT_FREQUENCY_SHIFTER, FshifterStateFactory_getFactory },
{ AL_EFFECT_RING_MODULATOR, ModulatorStateFactory_getFactory },
{ AL_EFFECT_PITCH_SHIFTER, PshifterStateFactory_getFactory},
{ AL_EFFECT_DEDICATED_DIALOGUE, DedicatedStateFactory_getFactory },
{ AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT, DedicatedStateFactory_getFactory }
};
static inline EffectStateFactory *getFactoryByType(ALenum type)
{
size_t i;
for(i = 0;i < COUNTOF(FactoryList);i++)
{
if(FactoryList[i].Type == type)
return FactoryList[i].GetFactory();
}
return NULL;
}
static void ALeffectState_IncRef(ALeffectState *state);
static inline ALeffectslot *LookupEffectSlot(ALCcontext *context, ALuint id)
{
id--;
if(UNLIKELY(id >= VECTOR_SIZE(context->EffectSlotList)))
return NULL;
return VECTOR_ELEM(context->EffectSlotList, id);
}
static inline ALeffect *LookupEffect(ALCdevice *device, ALuint id)
{
EffectSubList *sublist;
ALuint lidx = (id-1) >> 6;
ALsizei slidx = (id-1) & 0x3f;
if(UNLIKELY(lidx >= VECTOR_SIZE(device->EffectList)))
return NULL;
sublist = &VECTOR_ELEM(device->EffectList, lidx);
if(UNLIKELY(sublist->FreeMask & (U64(1)<<slidx)))
return NULL;
return sublist->Effects + slidx;
}
#define DO_UPDATEPROPS() do { \
if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire)) \
UpdateEffectSlotProps(slot, context); \
else \
ATOMIC_FLAG_CLEAR(&slot->PropsClean, almemory_order_release); \
} while(0)
AL_API ALvoid AL_APIENTRY alGenAuxiliaryEffectSlots(ALsizei n, ALuint *effectslots)
{
ALCdevice *device;
ALCcontext *context;
ALsizei cur;
context = GetContextRef();
if(!context) return;
if(!(n >= 0))
SETERR_GOTO(context, AL_INVALID_VALUE, done, "Generating %d effect slots", n);
if(n == 0) goto done;
LockEffectSlotList(context);
device = context->Device;
for(cur = 0;cur < n;cur++)
{
ALeffectslotPtr *iter = VECTOR_BEGIN(context->EffectSlotList);
ALeffectslotPtr *end = VECTOR_END(context->EffectSlotList);
ALeffectslot *slot = NULL;
ALenum err = AL_OUT_OF_MEMORY;
for(;iter != end;iter++)
{
if(!*iter)
break;
}
if(iter == end)
{
if(device->AuxiliaryEffectSlotMax == VECTOR_SIZE(context->EffectSlotList))
{
UnlockEffectSlotList(context);
alDeleteAuxiliaryEffectSlots(cur, effectslots);
SETERR_GOTO(context, AL_OUT_OF_MEMORY, done,
"Exceeding %u auxiliary effect slot limit", device->AuxiliaryEffectSlotMax);
}
VECTOR_PUSH_BACK(context->EffectSlotList, NULL);
iter = &VECTOR_BACK(context->EffectSlotList);
}
slot = al_calloc(16, sizeof(ALeffectslot));
if(!slot || (err=InitEffectSlot(slot)) != AL_NO_ERROR)
{
al_free(slot);
UnlockEffectSlotList(context);
alDeleteAuxiliaryEffectSlots(cur, effectslots);
SETERR_GOTO(context, err, done, "Effect slot object allocation failed");
}
aluInitEffectPanning(slot);
slot->id = (iter - VECTOR_BEGIN(context->EffectSlotList)) + 1;
*iter = slot;
effectslots[cur] = slot->id;
}
AddActiveEffectSlots(effectslots, n, context);
UnlockEffectSlotList(context);
done:
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;
LockEffectSlotList(context);
if(!(n >= 0))
SETERR_GOTO(context, AL_INVALID_VALUE, done, "Deleting %d effect slots", n);
if(n == 0) goto done;
for(i = 0;i < n;i++)
{
if((slot=LookupEffectSlot(context, effectslots[i])) == NULL)
SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u",
effectslots[i]);
if(ReadRef(&slot->ref) != 0)
SETERR_GOTO(context, AL_INVALID_NAME, done, "Deleting in-use effect slot %u",
effectslots[i]);
}
// All effectslots are valid
RemoveActiveEffectSlots(effectslots, n, context);
for(i = 0;i < n;i++)
{
if((slot=LookupEffectSlot(context, effectslots[i])) == NULL)
continue;
VECTOR_ELEM(context->EffectSlotList, effectslots[i]-1) = NULL;
DeinitEffectSlot(slot);
memset(slot, 0, sizeof(*slot));
al_free(slot);
}
done:
UnlockEffectSlotList(context);
ALCcontext_DecRef(context);
}
AL_API ALboolean AL_APIENTRY alIsAuxiliaryEffectSlot(ALuint effectslot)
{
ALCcontext *context;
ALboolean ret;
context = GetContextRef();
if(!context) return AL_FALSE;
LockEffectSlotList(context);
ret = (LookupEffectSlot(context, effectslot) ? AL_TRUE : AL_FALSE);
UnlockEffectSlotList(context);
ALCcontext_DecRef(context);
return ret;
}
AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSloti(ALuint effectslot, ALenum param, ALint value)
{
ALCdevice *device;
ALCcontext *context;
ALeffectslot *slot;
ALeffect *effect = NULL;
ALenum err;
context = GetContextRef();
if(!context) return;
almtx_lock(&context->PropLock);
LockEffectSlotList(context);
if((slot=LookupEffectSlot(context, effectslot)) == NULL)
SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot);
switch(param)
{
case AL_EFFECTSLOT_EFFECT:
device = context->Device;
LockEffectList(device);
effect = (value ? LookupEffect(device, value) : NULL);
if(!(value == 0 || effect != NULL))
{
UnlockEffectList(device);
SETERR_GOTO(context, AL_INVALID_VALUE, done, "Invalid effect ID %u", value);
}
err = InitializeEffect(context, slot, effect);
UnlockEffectList(device);
if(err != AL_NO_ERROR)
SETERR_GOTO(context, err, done, "Effect initialization failed");
break;
case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO:
if(!(value == AL_TRUE || value == AL_FALSE))
SETERR_GOTO(context, AL_INVALID_VALUE, done,
"Effect slot auxiliary send auto out of range");
slot->AuxSendAuto = value;
break;
default:
SETERR_GOTO(context, AL_INVALID_ENUM, done, "Invalid effect slot integer property 0x%04x",
param);
}
DO_UPDATEPROPS();
done:
UnlockEffectSlotList(context);
almtx_unlock(&context->PropLock);
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;
LockEffectSlotList(context);
if(LookupEffectSlot(context, effectslot) == NULL)
SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot);
switch(param)
{
default:
alSetError(context, AL_INVALID_ENUM, "Invalid effect slot integer-vector property 0x%04x",
param);
}
done:
UnlockEffectSlotList(context);
ALCcontext_DecRef(context);
}
AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotf(ALuint effectslot, ALenum param, ALfloat value)
{
ALCcontext *context;
ALeffectslot *slot;
context = GetContextRef();
if(!context) return;
almtx_lock(&context->PropLock);
LockEffectSlotList(context);
if((slot=LookupEffectSlot(context, effectslot)) == NULL)
SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot);
switch(param)
{
case AL_EFFECTSLOT_GAIN:
if(!(value >= 0.0f && value <= 1.0f))
SETERR_GOTO(context, AL_INVALID_VALUE, done, "Effect slot gain out of range");
slot->Gain = value;
break;
default:
SETERR_GOTO(context, AL_INVALID_ENUM, done, "Invalid effect slot float property 0x%04x",
param);
}
DO_UPDATEPROPS();
done:
UnlockEffectSlotList(context);
almtx_unlock(&context->PropLock);
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;
LockEffectSlotList(context);
if(LookupEffectSlot(context, effectslot) == NULL)
SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot);
switch(param)
{
default:
alSetError(context, AL_INVALID_ENUM, "Invalid effect slot float-vector property 0x%04x",
param);
}
done:
UnlockEffectSlotList(context);
ALCcontext_DecRef(context);
}
AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSloti(ALuint effectslot, ALenum param, ALint *value)
{
ALCcontext *context;
ALeffectslot *slot;
context = GetContextRef();
if(!context) return;
LockEffectSlotList(context);
if((slot=LookupEffectSlot(context, effectslot)) == NULL)
SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot);
switch(param)
{
case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO:
*value = slot->AuxSendAuto;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid effect slot integer property 0x%04x", param);
}
done:
UnlockEffectSlotList(context);
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;
LockEffectSlotList(context);
if(LookupEffectSlot(context, effectslot) == NULL)
SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot);
switch(param)
{
default:
alSetError(context, AL_INVALID_ENUM, "Invalid effect slot integer-vector property 0x%04x",
param);
}
done:
UnlockEffectSlotList(context);
ALCcontext_DecRef(context);
}
AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotf(ALuint effectslot, ALenum param, ALfloat *value)
{
ALCcontext *context;
ALeffectslot *slot;
context = GetContextRef();
if(!context) return;
LockEffectSlotList(context);
if((slot=LookupEffectSlot(context, effectslot)) == NULL)
SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot);
switch(param)
{
case AL_EFFECTSLOT_GAIN:
*value = slot->Gain;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid effect slot float property 0x%04x", param);
}
done:
UnlockEffectSlotList(context);
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;
LockEffectSlotList(context);
if(LookupEffectSlot(context, effectslot) == NULL)
SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect slot ID %u", effectslot);
switch(param)
{
default:
alSetError(context, AL_INVALID_ENUM, "Invalid effect slot float-vector property 0x%04x",
param);
}
done:
UnlockEffectSlotList(context);
ALCcontext_DecRef(context);
}
ALenum InitializeEffect(ALCcontext *Context, ALeffectslot *EffectSlot, ALeffect *effect)
{
ALCdevice *Device = Context->Device;
ALenum newtype = (effect ? effect->type : AL_EFFECT_NULL);
struct ALeffectslotProps *props;
ALeffectState *State;
if(newtype != EffectSlot->Effect.Type)
{
EffectStateFactory *factory;
factory = getFactoryByType(newtype);
if(!factory)
{
ERR("Failed to find factory for effect type 0x%04x\n", newtype);
return AL_INVALID_ENUM;
}
State = EffectStateFactory_create(factory);
if(!State) return AL_OUT_OF_MEMORY;
START_MIXER_MODE();
almtx_lock(&Device->BackendLock);
State->OutBuffer = Device->Dry.Buffer;
State->OutChannels = Device->Dry.NumChannels;
if(V(State,deviceUpdate)(Device) == AL_FALSE)
{
almtx_unlock(&Device->BackendLock);
LEAVE_MIXER_MODE();
ALeffectState_DecRef(State);
return AL_OUT_OF_MEMORY;
}
almtx_unlock(&Device->BackendLock);
END_MIXER_MODE();
if(!effect)
{
EffectSlot->Effect.Type = AL_EFFECT_NULL;
memset(&EffectSlot->Effect.Props, 0, sizeof(EffectSlot->Effect.Props));
}
else
{
EffectSlot->Effect.Type = effect->type;
EffectSlot->Effect.Props = effect->Props;
}
ALeffectState_DecRef(EffectSlot->Effect.State);
EffectSlot->Effect.State = State;
}
else if(effect)
EffectSlot->Effect.Props = effect->Props;
/* Remove state references from old effect slot property updates. */
props = ATOMIC_LOAD_SEQ(&Context->FreeEffectslotProps);
while(props)
{
if(props->State)
ALeffectState_DecRef(props->State);
props->State = NULL;
props = ATOMIC_LOAD(&props->next, almemory_order_relaxed);
}
return AL_NO_ERROR;
}
static void ALeffectState_IncRef(ALeffectState *state)
{
uint ref;
ref = IncrementRef(&state->Ref);
TRACEREF("%p increasing refcount to %u\n", state, ref);
}
void ALeffectState_DecRef(ALeffectState *state)
{
uint ref;
ref = DecrementRef(&state->Ref);
TRACEREF("%p decreasing refcount to %u\n", state, ref);
if(ref == 0) DELETE_OBJ(state);
}
void ALeffectState_Construct(ALeffectState *state)
{
InitRef(&state->Ref, 1);
state->OutBuffer = NULL;
state->OutChannels = 0;
}
void ALeffectState_Destruct(ALeffectState *UNUSED(state))
{
}
static void AddActiveEffectSlots(const ALuint *slotids, ALsizei count, ALCcontext *context)
{
struct ALeffectslotArray *curarray = ATOMIC_LOAD(&context->ActiveAuxSlots,
almemory_order_acquire);
struct ALeffectslotArray *newarray = NULL;
ALsizei newcount = curarray->count + count;
ALCdevice *device = context->Device;
ALsizei i, j;
/* Insert the new effect slots into the head of the array, followed by the
* existing ones.
*/
newarray = al_calloc(DEF_ALIGN, FAM_SIZE(struct ALeffectslotArray, slot, newcount));
newarray->count = newcount;
for(i = 0;i < count;i++)
newarray->slot[i] = LookupEffectSlot(context, slotids[i]);
for(j = 0;i < newcount;)
newarray->slot[i++] = curarray->slot[j++];
/* Remove any duplicates (first instance of each will be kept). */
for(i = 1;i < newcount;i++)
{
for(j = i;j != 0;)
{
if(UNLIKELY(newarray->slot[i] == newarray->slot[--j]))
{
newcount--;
for(j = i;j < newcount;j++)
newarray->slot[j] = newarray->slot[j+1];
i--;
break;
}
}
}
/* Reallocate newarray if the new size ended up smaller from duplicate
* removal.
*/
if(UNLIKELY(newcount < newarray->count))
{
struct ALeffectslotArray *tmpnewarray = al_calloc(DEF_ALIGN,
FAM_SIZE(struct ALeffectslotArray, slot, newcount));
memcpy(tmpnewarray, newarray, FAM_SIZE(struct ALeffectslotArray, slot, newcount));
al_free(newarray);
newarray = tmpnewarray;
newarray->count = newcount;
}
curarray = ATOMIC_EXCHANGE_PTR(&context->ActiveAuxSlots, newarray, almemory_order_acq_rel);
while((ATOMIC_LOAD(&device->MixCount, almemory_order_acquire)&1))
althrd_yield();
al_free(curarray);
}
static void RemoveActiveEffectSlots(const ALuint *slotids, ALsizei count, ALCcontext *context)
{
struct ALeffectslotArray *curarray = ATOMIC_LOAD(&context->ActiveAuxSlots,
almemory_order_acquire);
struct ALeffectslotArray *newarray = NULL;
ALCdevice *device = context->Device;
ALsizei i, j;
/* Don't shrink the allocated array size since we don't know how many (if
* any) of the effect slots to remove are in the array.
*/
newarray = al_calloc(DEF_ALIGN, FAM_SIZE(struct ALeffectslotArray, slot, curarray->count));
newarray->count = 0;
for(i = 0;i < curarray->count;i++)
{
/* Insert this slot into the new array only if it's not one to remove. */
ALeffectslot *slot = curarray->slot[i];
for(j = count;j != 0;)
{
if(slot->id == slotids[--j])
goto skip_ins;
}
newarray->slot[newarray->count++] = slot;
skip_ins: ;
}
/* TODO: Could reallocate newarray now that we know it's needed size. */
curarray = ATOMIC_EXCHANGE_PTR(&context->ActiveAuxSlots, newarray, almemory_order_acq_rel);
while((ATOMIC_LOAD(&device->MixCount, almemory_order_acquire)&1))
althrd_yield();
al_free(curarray);
}
ALenum InitEffectSlot(ALeffectslot *slot)
{
EffectStateFactory *factory;
slot->Effect.Type = AL_EFFECT_NULL;
factory = getFactoryByType(AL_EFFECT_NULL);
slot->Effect.State = EffectStateFactory_create(factory);
if(!slot->Effect.State) return AL_OUT_OF_MEMORY;
slot->Gain = 1.0;
slot->AuxSendAuto = AL_TRUE;
ATOMIC_FLAG_TEST_AND_SET(&slot->PropsClean, almemory_order_relaxed);
InitRef(&slot->ref, 0);
ATOMIC_INIT(&slot->Update, NULL);
slot->Params.Gain = 1.0f;
slot->Params.AuxSendAuto = AL_TRUE;
ALeffectState_IncRef(slot->Effect.State);
slot->Params.EffectState = slot->Effect.State;
slot->Params.RoomRolloff = 0.0f;
slot->Params.DecayTime = 0.0f;
slot->Params.DecayLFRatio = 0.0f;
slot->Params.DecayHFRatio = 0.0f;
slot->Params.DecayHFLimit = AL_FALSE;
slot->Params.AirAbsorptionGainHF = 1.0f;
return AL_NO_ERROR;
}
void DeinitEffectSlot(ALeffectslot *slot)
{
struct ALeffectslotProps *props;
props = ATOMIC_LOAD_SEQ(&slot->Update);
if(props)
{
if(props->State) ALeffectState_DecRef(props->State);
TRACE("Freed unapplied AuxiliaryEffectSlot update %p\n", props);
al_free(props);
}
ALeffectState_DecRef(slot->Effect.State);
if(slot->Params.EffectState)
ALeffectState_DecRef(slot->Params.EffectState);
}
void UpdateEffectSlotProps(ALeffectslot *slot, ALCcontext *context)
{
struct ALeffectslotProps *props;
ALeffectState *oldstate;
/* Get an unused property container, or allocate a new one as needed. */
props = ATOMIC_LOAD(&context->FreeEffectslotProps, almemory_order_relaxed);
if(!props)
props = al_calloc(16, sizeof(*props));
else
{
struct ALeffectslotProps *next;
do {
next = ATOMIC_LOAD(&props->next, almemory_order_relaxed);
} while(ATOMIC_COMPARE_EXCHANGE_PTR_WEAK(&context->FreeEffectslotProps, &props, next,
almemory_order_seq_cst, almemory_order_acquire) == 0);
}
/* Copy in current property values. */
props->Gain = slot->Gain;
props->AuxSendAuto = slot->AuxSendAuto;
props->Type = slot->Effect.Type;
props->Props = slot->Effect.Props;
/* Swap out any stale effect state object there may be in the container, to
* delete it.
*/
ALeffectState_IncRef(slot->Effect.State);
oldstate = props->State;
props->State = slot->Effect.State;
/* Set the new container for updating internal parameters. */
props = ATOMIC_EXCHANGE_PTR(&slot->Update, props, almemory_order_acq_rel);
if(props)
{
/* If there was an unused update container, put it back in the
* freelist.
*/
if(props->State)
ALeffectState_DecRef(props->State);
props->State = NULL;
ATOMIC_REPLACE_HEAD(struct ALeffectslotProps*, &context->FreeEffectslotProps, props);
}
if(oldstate)
ALeffectState_DecRef(oldstate);
}
void UpdateAllEffectSlotProps(ALCcontext *context)
{
struct ALeffectslotArray *auxslots;
ALsizei i;
LockEffectSlotList(context);
auxslots = ATOMIC_LOAD(&context->ActiveAuxSlots, almemory_order_acquire);
for(i = 0;i < auxslots->count;i++)
{
ALeffectslot *slot = auxslots->slot[i];
if(!ATOMIC_FLAG_TEST_AND_SET(&slot->PropsClean, almemory_order_acq_rel))
UpdateEffectSlotProps(slot, context);
}
UnlockEffectSlotList(context);
}
ALvoid ReleaseALAuxiliaryEffectSlots(ALCcontext *context)
{
ALeffectslotPtr *iter = VECTOR_BEGIN(context->EffectSlotList);
ALeffectslotPtr *end = VECTOR_END(context->EffectSlotList);
size_t leftover = 0;
for(;iter != end;iter++)
{
ALeffectslot *slot = *iter;
if(!slot) continue;
*iter = NULL;
DeinitEffectSlot(slot);
memset(slot, 0, sizeof(*slot));
al_free(slot);
++leftover;
}
if(leftover > 0)
WARN("(%p) Deleted "SZFMT" AuxiliaryEffectSlot%s\n", context, leftover, (leftover==1)?"":"s");
}
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-819
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@@ -1,819 +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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdlib.h>
#include <math.h>
#include <float.h>
#include "AL/al.h"
#include "AL/alc.h"
#include "alMain.h"
#include "alEffect.h"
#include "alError.h"
extern inline void LockEffectList(ALCdevice *device);
extern inline void UnlockEffectList(ALCdevice *device);
extern inline ALboolean IsReverbEffect(ALenum type);
const struct EffectList EffectList[EFFECTLIST_SIZE] = {
{ "eaxreverb", EAXREVERB_EFFECT, AL_EFFECT_EAXREVERB },
{ "reverb", REVERB_EFFECT, AL_EFFECT_REVERB },
{ "autowah", AUTOWAH_EFFECT, AL_EFFECT_AUTOWAH },
{ "chorus", CHORUS_EFFECT, AL_EFFECT_CHORUS },
{ "compressor", COMPRESSOR_EFFECT, AL_EFFECT_COMPRESSOR },
{ "distortion", DISTORTION_EFFECT, AL_EFFECT_DISTORTION },
{ "echo", ECHO_EFFECT, AL_EFFECT_ECHO },
{ "equalizer", EQUALIZER_EFFECT, AL_EFFECT_EQUALIZER },
{ "flanger", FLANGER_EFFECT, AL_EFFECT_FLANGER },
{ "fshifter", FSHIFTER_EFFECT, AL_EFFECT_FREQUENCY_SHIFTER },
{ "modulator", MODULATOR_EFFECT, AL_EFFECT_RING_MODULATOR },
{ "pshifter", PSHIFTER_EFFECT, AL_EFFECT_PITCH_SHIFTER },
{ "dedicated", DEDICATED_EFFECT, AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT },
{ "dedicated", DEDICATED_EFFECT, AL_EFFECT_DEDICATED_DIALOGUE },
};
ALboolean DisabledEffects[MAX_EFFECTS];
static ALeffect *AllocEffect(ALCcontext *context);
static void FreeEffect(ALCdevice *device, ALeffect *effect);
static void InitEffectParams(ALeffect *effect, ALenum type);
static inline ALeffect *LookupEffect(ALCdevice *device, ALuint id)
{
EffectSubList *sublist;
ALuint lidx = (id-1) >> 6;
ALsizei slidx = (id-1) & 0x3f;
if(UNLIKELY(lidx >= VECTOR_SIZE(device->EffectList)))
return NULL;
sublist = &VECTOR_ELEM(device->EffectList, lidx);
if(UNLIKELY(sublist->FreeMask & (U64(1)<<slidx)))
return NULL;
return sublist->Effects + slidx;
}
AL_API ALvoid AL_APIENTRY alGenEffects(ALsizei n, ALuint *effects)
{
ALCcontext *context;
ALsizei cur;
context = GetContextRef();
if(!context) return;
if(!(n >= 0))
alSetError(context, AL_INVALID_VALUE, "Generating %d effects", n);
else for(cur = 0;cur < n;cur++)
{
ALeffect *effect = AllocEffect(context);
if(!effect)
{
alDeleteEffects(cur, effects);
break;
}
effects[cur] = effect->id;
}
ALCcontext_DecRef(context);
}
AL_API ALvoid AL_APIENTRY alDeleteEffects(ALsizei n, const ALuint *effects)
{
ALCdevice *device;
ALCcontext *context;
ALeffect *effect;
ALsizei i;
context = GetContextRef();
if(!context) return;
device = context->Device;
LockEffectList(device);
if(!(n >= 0))
SETERR_GOTO(context, AL_INVALID_VALUE, done, "Deleting %d effects", n);
for(i = 0;i < n;i++)
{
if(effects[i] && LookupEffect(device, effects[i]) == NULL)
SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid effect ID %u", effects[i]);
}
for(i = 0;i < n;i++)
{
if((effect=LookupEffect(device, effects[i])) != NULL)
FreeEffect(device, effect);
}
done:
UnlockEffectList(device);
ALCcontext_DecRef(context);
}
AL_API ALboolean AL_APIENTRY alIsEffect(ALuint effect)
{
ALCcontext *Context;
ALboolean result;
Context = GetContextRef();
if(!Context) return AL_FALSE;
LockEffectList(Context->Device);
result = ((!effect || LookupEffect(Context->Device, effect)) ?
AL_TRUE : AL_FALSE);
UnlockEffectList(Context->Device);
ALCcontext_DecRef(Context);
return result;
}
AL_API ALvoid AL_APIENTRY alEffecti(ALuint effect, ALenum param, ALint value)
{
ALCcontext *Context;
ALCdevice *Device;
ALeffect *ALEffect;
Context = GetContextRef();
if(!Context) return;
Device = Context->Device;
LockEffectList(Device);
if((ALEffect=LookupEffect(Device, effect)) == NULL)
alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect);
else
{
if(param == AL_EFFECT_TYPE)
{
ALboolean isOk = (value == AL_EFFECT_NULL);
ALint i;
for(i = 0;!isOk && i < EFFECTLIST_SIZE;i++)
{
if(value == EffectList[i].val &&
!DisabledEffects[EffectList[i].type])
isOk = AL_TRUE;
}
if(isOk)
InitEffectParams(ALEffect, value);
else
alSetError(Context, AL_INVALID_VALUE, "Effect type 0x%04x not supported", value);
}
else
{
/* Call the appropriate handler */
ALeffect_setParami(ALEffect, Context, param, value);
}
}
UnlockEffectList(Device);
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alEffectiv(ALuint effect, ALenum param, const ALint *values)
{
ALCcontext *Context;
ALCdevice *Device;
ALeffect *ALEffect;
switch(param)
{
case AL_EFFECT_TYPE:
alEffecti(effect, param, values[0]);
return;
}
Context = GetContextRef();
if(!Context) return;
Device = Context->Device;
LockEffectList(Device);
if((ALEffect=LookupEffect(Device, effect)) == NULL)
alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect);
else
{
/* Call the appropriate handler */
ALeffect_setParamiv(ALEffect, Context, param, values);
}
UnlockEffectList(Device);
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alEffectf(ALuint effect, ALenum param, ALfloat value)
{
ALCcontext *Context;
ALCdevice *Device;
ALeffect *ALEffect;
Context = GetContextRef();
if(!Context) return;
Device = Context->Device;
LockEffectList(Device);
if((ALEffect=LookupEffect(Device, effect)) == NULL)
alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect);
else
{
/* Call the appropriate handler */
ALeffect_setParamf(ALEffect, Context, param, value);
}
UnlockEffectList(Device);
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alEffectfv(ALuint effect, ALenum param, const ALfloat *values)
{
ALCcontext *Context;
ALCdevice *Device;
ALeffect *ALEffect;
Context = GetContextRef();
if(!Context) return;
Device = Context->Device;
LockEffectList(Device);
if((ALEffect=LookupEffect(Device, effect)) == NULL)
alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect);
else
{
/* Call the appropriate handler */
ALeffect_setParamfv(ALEffect, Context, param, values);
}
UnlockEffectList(Device);
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alGetEffecti(ALuint effect, ALenum param, ALint *value)
{
ALCcontext *Context;
ALCdevice *Device;
ALeffect *ALEffect;
Context = GetContextRef();
if(!Context) return;
Device = Context->Device;
LockEffectList(Device);
if((ALEffect=LookupEffect(Device, effect)) == NULL)
alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect);
else
{
if(param == AL_EFFECT_TYPE)
*value = ALEffect->type;
else
{
/* Call the appropriate handler */
ALeffect_getParami(ALEffect, Context, param, value);
}
}
UnlockEffectList(Device);
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alGetEffectiv(ALuint effect, ALenum param, ALint *values)
{
ALCcontext *Context;
ALCdevice *Device;
ALeffect *ALEffect;
switch(param)
{
case AL_EFFECT_TYPE:
alGetEffecti(effect, param, values);
return;
}
Context = GetContextRef();
if(!Context) return;
Device = Context->Device;
LockEffectList(Device);
if((ALEffect=LookupEffect(Device, effect)) == NULL)
alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect);
else
{
/* Call the appropriate handler */
ALeffect_getParamiv(ALEffect, Context, param, values);
}
UnlockEffectList(Device);
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alGetEffectf(ALuint effect, ALenum param, ALfloat *value)
{
ALCcontext *Context;
ALCdevice *Device;
ALeffect *ALEffect;
Context = GetContextRef();
if(!Context) return;
Device = Context->Device;
LockEffectList(Device);
if((ALEffect=LookupEffect(Device, effect)) == NULL)
alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect);
else
{
/* Call the appropriate handler */
ALeffect_getParamf(ALEffect, Context, param, value);
}
UnlockEffectList(Device);
ALCcontext_DecRef(Context);
}
AL_API ALvoid AL_APIENTRY alGetEffectfv(ALuint effect, ALenum param, ALfloat *values)
{
ALCcontext *Context;
ALCdevice *Device;
ALeffect *ALEffect;
Context = GetContextRef();
if(!Context) return;
Device = Context->Device;
LockEffectList(Device);
if((ALEffect=LookupEffect(Device, effect)) == NULL)
alSetError(Context, AL_INVALID_NAME, "Invalid effect ID %u", effect);
else
{
/* Call the appropriate handler */
ALeffect_getParamfv(ALEffect, Context, param, values);
}
UnlockEffectList(Device);
ALCcontext_DecRef(Context);
}
void InitEffect(ALeffect *effect)
{
InitEffectParams(effect, AL_EFFECT_NULL);
}
static ALeffect *AllocEffect(ALCcontext *context)
{
ALCdevice *device = context->Device;
EffectSubList *sublist, *subend;
ALeffect *effect = NULL;
ALsizei lidx = 0;
ALsizei slidx;
almtx_lock(&device->EffectLock);
sublist = VECTOR_BEGIN(device->EffectList);
subend = VECTOR_END(device->EffectList);
for(;sublist != subend;++sublist)
{
if(sublist->FreeMask)
{
slidx = CTZ64(sublist->FreeMask);
effect = sublist->Effects + slidx;
break;
}
++lidx;
}
if(UNLIKELY(!effect))
{
const EffectSubList empty_sublist = { 0, NULL };
/* Don't allocate so many list entries that the 32-bit ID could
* overflow...
*/
if(UNLIKELY(VECTOR_SIZE(device->EffectList) >= 1<<25))
{
almtx_unlock(&device->EffectLock);
alSetError(context, AL_OUT_OF_MEMORY, "Too many effects allocated");
return NULL;
}
lidx = (ALsizei)VECTOR_SIZE(device->EffectList);
VECTOR_PUSH_BACK(device->EffectList, empty_sublist);
sublist = &VECTOR_BACK(device->EffectList);
sublist->FreeMask = ~U64(0);
sublist->Effects = al_calloc(16, sizeof(ALeffect)*64);
if(UNLIKELY(!sublist->Effects))
{
VECTOR_POP_BACK(device->EffectList);
almtx_unlock(&device->EffectLock);
alSetError(context, AL_OUT_OF_MEMORY, "Failed to allocate effect batch");
return NULL;
}
slidx = 0;
effect = sublist->Effects + slidx;
}
memset(effect, 0, sizeof(*effect));
InitEffectParams(effect, AL_EFFECT_NULL);
/* Add 1 to avoid effect ID 0. */
effect->id = ((lidx<<6) | slidx) + 1;
sublist->FreeMask &= ~(U64(1)<<slidx);
almtx_unlock(&device->EffectLock);
return effect;
}
static void FreeEffect(ALCdevice *device, ALeffect *effect)
{
ALuint id = effect->id - 1;
ALsizei lidx = id >> 6;
ALsizei slidx = id & 0x3f;
memset(effect, 0, sizeof(*effect));
VECTOR_ELEM(device->EffectList, lidx).FreeMask |= U64(1) << slidx;
}
void ReleaseALEffects(ALCdevice *device)
{
EffectSubList *sublist = VECTOR_BEGIN(device->EffectList);
EffectSubList *subend = VECTOR_END(device->EffectList);
size_t leftover = 0;
for(;sublist != subend;++sublist)
{
ALuint64 usemask = ~sublist->FreeMask;
while(usemask)
{
ALsizei idx = CTZ64(usemask);
ALeffect *effect = sublist->Effects + idx;
memset(effect, 0, sizeof(*effect));
++leftover;
usemask &= ~(U64(1) << idx);
}
sublist->FreeMask = ~usemask;
}
if(leftover > 0)
WARN("(%p) Deleted "SZFMT" Effect%s\n", device, leftover, (leftover==1)?"":"s");
}
static void InitEffectParams(ALeffect *effect, ALenum type)
{
switch(type)
{
case AL_EFFECT_EAXREVERB:
effect->Props.Reverb.Density = AL_EAXREVERB_DEFAULT_DENSITY;
effect->Props.Reverb.Diffusion = AL_EAXREVERB_DEFAULT_DIFFUSION;
effect->Props.Reverb.Gain = AL_EAXREVERB_DEFAULT_GAIN;
effect->Props.Reverb.GainHF = AL_EAXREVERB_DEFAULT_GAINHF;
effect->Props.Reverb.GainLF = AL_EAXREVERB_DEFAULT_GAINLF;
effect->Props.Reverb.DecayTime = AL_EAXREVERB_DEFAULT_DECAY_TIME;
effect->Props.Reverb.DecayHFRatio = AL_EAXREVERB_DEFAULT_DECAY_HFRATIO;
effect->Props.Reverb.DecayLFRatio = AL_EAXREVERB_DEFAULT_DECAY_LFRATIO;
effect->Props.Reverb.ReflectionsGain = AL_EAXREVERB_DEFAULT_REFLECTIONS_GAIN;
effect->Props.Reverb.ReflectionsDelay = AL_EAXREVERB_DEFAULT_REFLECTIONS_DELAY;
effect->Props.Reverb.ReflectionsPan[0] = AL_EAXREVERB_DEFAULT_REFLECTIONS_PAN_XYZ;
effect->Props.Reverb.ReflectionsPan[1] = AL_EAXREVERB_DEFAULT_REFLECTIONS_PAN_XYZ;
effect->Props.Reverb.ReflectionsPan[2] = AL_EAXREVERB_DEFAULT_REFLECTIONS_PAN_XYZ;
effect->Props.Reverb.LateReverbGain = AL_EAXREVERB_DEFAULT_LATE_REVERB_GAIN;
effect->Props.Reverb.LateReverbDelay = AL_EAXREVERB_DEFAULT_LATE_REVERB_DELAY;
effect->Props.Reverb.LateReverbPan[0] = AL_EAXREVERB_DEFAULT_LATE_REVERB_PAN_XYZ;
effect->Props.Reverb.LateReverbPan[1] = AL_EAXREVERB_DEFAULT_LATE_REVERB_PAN_XYZ;
effect->Props.Reverb.LateReverbPan[2] = AL_EAXREVERB_DEFAULT_LATE_REVERB_PAN_XYZ;
effect->Props.Reverb.EchoTime = AL_EAXREVERB_DEFAULT_ECHO_TIME;
effect->Props.Reverb.EchoDepth = AL_EAXREVERB_DEFAULT_ECHO_DEPTH;
effect->Props.Reverb.ModulationTime = AL_EAXREVERB_DEFAULT_MODULATION_TIME;
effect->Props.Reverb.ModulationDepth = AL_EAXREVERB_DEFAULT_MODULATION_DEPTH;
effect->Props.Reverb.AirAbsorptionGainHF = AL_EAXREVERB_DEFAULT_AIR_ABSORPTION_GAINHF;
effect->Props.Reverb.HFReference = AL_EAXREVERB_DEFAULT_HFREFERENCE;
effect->Props.Reverb.LFReference = AL_EAXREVERB_DEFAULT_LFREFERENCE;
effect->Props.Reverb.RoomRolloffFactor = AL_EAXREVERB_DEFAULT_ROOM_ROLLOFF_FACTOR;
effect->Props.Reverb.DecayHFLimit = AL_EAXREVERB_DEFAULT_DECAY_HFLIMIT;
effect->vtab = &ALeaxreverb_vtable;
break;
case AL_EFFECT_REVERB:
effect->Props.Reverb.Density = AL_REVERB_DEFAULT_DENSITY;
effect->Props.Reverb.Diffusion = AL_REVERB_DEFAULT_DIFFUSION;
effect->Props.Reverb.Gain = AL_REVERB_DEFAULT_GAIN;
effect->Props.Reverb.GainHF = AL_REVERB_DEFAULT_GAINHF;
effect->Props.Reverb.GainLF = 1.0f;
effect->Props.Reverb.DecayTime = AL_REVERB_DEFAULT_DECAY_TIME;
effect->Props.Reverb.DecayHFRatio = AL_REVERB_DEFAULT_DECAY_HFRATIO;
effect->Props.Reverb.DecayLFRatio = 1.0f;
effect->Props.Reverb.ReflectionsGain = AL_REVERB_DEFAULT_REFLECTIONS_GAIN;
effect->Props.Reverb.ReflectionsDelay = AL_REVERB_DEFAULT_REFLECTIONS_DELAY;
effect->Props.Reverb.ReflectionsPan[0] = 0.0f;
effect->Props.Reverb.ReflectionsPan[1] = 0.0f;
effect->Props.Reverb.ReflectionsPan[2] = 0.0f;
effect->Props.Reverb.LateReverbGain = AL_REVERB_DEFAULT_LATE_REVERB_GAIN;
effect->Props.Reverb.LateReverbDelay = AL_REVERB_DEFAULT_LATE_REVERB_DELAY;
effect->Props.Reverb.LateReverbPan[0] = 0.0f;
effect->Props.Reverb.LateReverbPan[1] = 0.0f;
effect->Props.Reverb.LateReverbPan[2] = 0.0f;
effect->Props.Reverb.EchoTime = 0.25f;
effect->Props.Reverb.EchoDepth = 0.0f;
effect->Props.Reverb.ModulationTime = 0.25f;
effect->Props.Reverb.ModulationDepth = 0.0f;
effect->Props.Reverb.AirAbsorptionGainHF = AL_REVERB_DEFAULT_AIR_ABSORPTION_GAINHF;
effect->Props.Reverb.HFReference = 5000.0f;
effect->Props.Reverb.LFReference = 250.0f;
effect->Props.Reverb.RoomRolloffFactor = AL_REVERB_DEFAULT_ROOM_ROLLOFF_FACTOR;
effect->Props.Reverb.DecayHFLimit = AL_REVERB_DEFAULT_DECAY_HFLIMIT;
effect->vtab = &ALreverb_vtable;
break;
case AL_EFFECT_AUTOWAH:
effect->Props.Autowah.AttackTime = AL_AUTOWAH_DEFAULT_ATTACK_TIME;
effect->Props.Autowah.ReleaseTime = AL_AUTOWAH_DEFAULT_RELEASE_TIME;
effect->Props.Autowah.Resonance = AL_AUTOWAH_DEFAULT_RESONANCE;
effect->Props.Autowah.PeakGain = AL_AUTOWAH_DEFAULT_PEAK_GAIN;
effect->vtab = &ALautowah_vtable;
break;
case AL_EFFECT_CHORUS:
effect->Props.Chorus.Waveform = AL_CHORUS_DEFAULT_WAVEFORM;
effect->Props.Chorus.Phase = AL_CHORUS_DEFAULT_PHASE;
effect->Props.Chorus.Rate = AL_CHORUS_DEFAULT_RATE;
effect->Props.Chorus.Depth = AL_CHORUS_DEFAULT_DEPTH;
effect->Props.Chorus.Feedback = AL_CHORUS_DEFAULT_FEEDBACK;
effect->Props.Chorus.Delay = AL_CHORUS_DEFAULT_DELAY;
effect->vtab = &ALchorus_vtable;
break;
case AL_EFFECT_COMPRESSOR:
effect->Props.Compressor.OnOff = AL_COMPRESSOR_DEFAULT_ONOFF;
effect->vtab = &ALcompressor_vtable;
break;
case AL_EFFECT_DISTORTION:
effect->Props.Distortion.Edge = AL_DISTORTION_DEFAULT_EDGE;
effect->Props.Distortion.Gain = AL_DISTORTION_DEFAULT_GAIN;
effect->Props.Distortion.LowpassCutoff = AL_DISTORTION_DEFAULT_LOWPASS_CUTOFF;
effect->Props.Distortion.EQCenter = AL_DISTORTION_DEFAULT_EQCENTER;
effect->Props.Distortion.EQBandwidth = AL_DISTORTION_DEFAULT_EQBANDWIDTH;
effect->vtab = &ALdistortion_vtable;
break;
case AL_EFFECT_ECHO:
effect->Props.Echo.Delay = AL_ECHO_DEFAULT_DELAY;
effect->Props.Echo.LRDelay = AL_ECHO_DEFAULT_LRDELAY;
effect->Props.Echo.Damping = AL_ECHO_DEFAULT_DAMPING;
effect->Props.Echo.Feedback = AL_ECHO_DEFAULT_FEEDBACK;
effect->Props.Echo.Spread = AL_ECHO_DEFAULT_SPREAD;
effect->vtab = &ALecho_vtable;
break;
case AL_EFFECT_EQUALIZER:
effect->Props.Equalizer.LowCutoff = AL_EQUALIZER_DEFAULT_LOW_CUTOFF;
effect->Props.Equalizer.LowGain = AL_EQUALIZER_DEFAULT_LOW_GAIN;
effect->Props.Equalizer.Mid1Center = AL_EQUALIZER_DEFAULT_MID1_CENTER;
effect->Props.Equalizer.Mid1Gain = AL_EQUALIZER_DEFAULT_MID1_GAIN;
effect->Props.Equalizer.Mid1Width = AL_EQUALIZER_DEFAULT_MID1_WIDTH;
effect->Props.Equalizer.Mid2Center = AL_EQUALIZER_DEFAULT_MID2_CENTER;
effect->Props.Equalizer.Mid2Gain = AL_EQUALIZER_DEFAULT_MID2_GAIN;
effect->Props.Equalizer.Mid2Width = AL_EQUALIZER_DEFAULT_MID2_WIDTH;
effect->Props.Equalizer.HighCutoff = AL_EQUALIZER_DEFAULT_HIGH_CUTOFF;
effect->Props.Equalizer.HighGain = AL_EQUALIZER_DEFAULT_HIGH_GAIN;
effect->vtab = &ALequalizer_vtable;
break;
case AL_EFFECT_FLANGER:
effect->Props.Chorus.Waveform = AL_FLANGER_DEFAULT_WAVEFORM;
effect->Props.Chorus.Phase = AL_FLANGER_DEFAULT_PHASE;
effect->Props.Chorus.Rate = AL_FLANGER_DEFAULT_RATE;
effect->Props.Chorus.Depth = AL_FLANGER_DEFAULT_DEPTH;
effect->Props.Chorus.Feedback = AL_FLANGER_DEFAULT_FEEDBACK;
effect->Props.Chorus.Delay = AL_FLANGER_DEFAULT_DELAY;
effect->vtab = &ALflanger_vtable;
break;
case AL_EFFECT_FREQUENCY_SHIFTER:
effect->Props.Fshifter.Frequency = AL_FREQUENCY_SHIFTER_DEFAULT_FREQUENCY;
effect->Props.Fshifter.LeftDirection = AL_FREQUENCY_SHIFTER_DEFAULT_LEFT_DIRECTION;
effect->Props.Fshifter.RightDirection = AL_FREQUENCY_SHIFTER_DEFAULT_RIGHT_DIRECTION;
effect->vtab = &ALfshifter_vtable;
break;
case AL_EFFECT_RING_MODULATOR:
effect->Props.Modulator.Frequency = AL_RING_MODULATOR_DEFAULT_FREQUENCY;
effect->Props.Modulator.HighPassCutoff = AL_RING_MODULATOR_DEFAULT_HIGHPASS_CUTOFF;
effect->Props.Modulator.Waveform = AL_RING_MODULATOR_DEFAULT_WAVEFORM;
effect->vtab = &ALmodulator_vtable;
break;
case AL_EFFECT_PITCH_SHIFTER:
effect->Props.Pshifter.CoarseTune = AL_PITCH_SHIFTER_DEFAULT_COARSE_TUNE;
effect->Props.Pshifter.FineTune = AL_PITCH_SHIFTER_DEFAULT_FINE_TUNE;
effect->vtab = &ALpshifter_vtable;
break;
case AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT:
case AL_EFFECT_DEDICATED_DIALOGUE:
effect->Props.Dedicated.Gain = 1.0f;
effect->vtab = &ALdedicated_vtable;
break;
default:
effect->vtab = &ALnull_vtable;
break;
}
effect->type = type;
}
#include "AL/efx-presets.h"
#define DECL(x) { #x, EFX_REVERB_PRESET_##x }
static const struct {
const char name[32];
EFXEAXREVERBPROPERTIES props;
} reverblist[] = {
DECL(GENERIC),
DECL(PADDEDCELL),
DECL(ROOM),
DECL(BATHROOM),
DECL(LIVINGROOM),
DECL(STONEROOM),
DECL(AUDITORIUM),
DECL(CONCERTHALL),
DECL(CAVE),
DECL(ARENA),
DECL(HANGAR),
DECL(CARPETEDHALLWAY),
DECL(HALLWAY),
DECL(STONECORRIDOR),
DECL(ALLEY),
DECL(FOREST),
DECL(CITY),
DECL(MOUNTAINS),
DECL(QUARRY),
DECL(PLAIN),
DECL(PARKINGLOT),
DECL(SEWERPIPE),
DECL(UNDERWATER),
DECL(DRUGGED),
DECL(DIZZY),
DECL(PSYCHOTIC),
DECL(CASTLE_SMALLROOM),
DECL(CASTLE_SHORTPASSAGE),
DECL(CASTLE_MEDIUMROOM),
DECL(CASTLE_LARGEROOM),
DECL(CASTLE_LONGPASSAGE),
DECL(CASTLE_HALL),
DECL(CASTLE_CUPBOARD),
DECL(CASTLE_COURTYARD),
DECL(CASTLE_ALCOVE),
DECL(FACTORY_SMALLROOM),
DECL(FACTORY_SHORTPASSAGE),
DECL(FACTORY_MEDIUMROOM),
DECL(FACTORY_LARGEROOM),
DECL(FACTORY_LONGPASSAGE),
DECL(FACTORY_HALL),
DECL(FACTORY_CUPBOARD),
DECL(FACTORY_COURTYARD),
DECL(FACTORY_ALCOVE),
DECL(ICEPALACE_SMALLROOM),
DECL(ICEPALACE_SHORTPASSAGE),
DECL(ICEPALACE_MEDIUMROOM),
DECL(ICEPALACE_LARGEROOM),
DECL(ICEPALACE_LONGPASSAGE),
DECL(ICEPALACE_HALL),
DECL(ICEPALACE_CUPBOARD),
DECL(ICEPALACE_COURTYARD),
DECL(ICEPALACE_ALCOVE),
DECL(SPACESTATION_SMALLROOM),
DECL(SPACESTATION_SHORTPASSAGE),
DECL(SPACESTATION_MEDIUMROOM),
DECL(SPACESTATION_LARGEROOM),
DECL(SPACESTATION_LONGPASSAGE),
DECL(SPACESTATION_HALL),
DECL(SPACESTATION_CUPBOARD),
DECL(SPACESTATION_ALCOVE),
DECL(WOODEN_SMALLROOM),
DECL(WOODEN_SHORTPASSAGE),
DECL(WOODEN_MEDIUMROOM),
DECL(WOODEN_LARGEROOM),
DECL(WOODEN_LONGPASSAGE),
DECL(WOODEN_HALL),
DECL(WOODEN_CUPBOARD),
DECL(WOODEN_COURTYARD),
DECL(WOODEN_ALCOVE),
DECL(SPORT_EMPTYSTADIUM),
DECL(SPORT_SQUASHCOURT),
DECL(SPORT_SMALLSWIMMINGPOOL),
DECL(SPORT_LARGESWIMMINGPOOL),
DECL(SPORT_GYMNASIUM),
DECL(SPORT_FULLSTADIUM),
DECL(SPORT_STADIUMTANNOY),
DECL(PREFAB_WORKSHOP),
DECL(PREFAB_SCHOOLROOM),
DECL(PREFAB_PRACTISEROOM),
DECL(PREFAB_OUTHOUSE),
DECL(PREFAB_CARAVAN),
DECL(DOME_TOMB),
DECL(PIPE_SMALL),
DECL(DOME_SAINTPAULS),
DECL(PIPE_LONGTHIN),
DECL(PIPE_LARGE),
DECL(PIPE_RESONANT),
DECL(OUTDOORS_BACKYARD),
DECL(OUTDOORS_ROLLINGPLAINS),
DECL(OUTDOORS_DEEPCANYON),
DECL(OUTDOORS_CREEK),
DECL(OUTDOORS_VALLEY),
DECL(MOOD_HEAVEN),
DECL(MOOD_HELL),
DECL(MOOD_MEMORY),
DECL(DRIVING_COMMENTATOR),
DECL(DRIVING_PITGARAGE),
DECL(DRIVING_INCAR_RACER),
DECL(DRIVING_INCAR_SPORTS),
DECL(DRIVING_INCAR_LUXURY),
DECL(DRIVING_FULLGRANDSTAND),
DECL(DRIVING_EMPTYGRANDSTAND),
DECL(DRIVING_TUNNEL),
DECL(CITY_STREETS),
DECL(CITY_SUBWAY),
DECL(CITY_MUSEUM),
DECL(CITY_LIBRARY),
DECL(CITY_UNDERPASS),
DECL(CITY_ABANDONED),
DECL(DUSTYROOM),
DECL(CHAPEL),
DECL(SMALLWATERROOM),
};
#undef DECL
void LoadReverbPreset(const char *name, ALeffect *effect)
{
size_t i;
if(strcasecmp(name, "NONE") == 0)
{
InitEffectParams(effect, AL_EFFECT_NULL);
TRACE("Loading reverb '%s'\n", "NONE");
return;
}
if(!DisabledEffects[EAXREVERB_EFFECT])
InitEffectParams(effect, AL_EFFECT_EAXREVERB);
else if(!DisabledEffects[REVERB_EFFECT])
InitEffectParams(effect, AL_EFFECT_REVERB);
else
InitEffectParams(effect, AL_EFFECT_NULL);
for(i = 0;i < COUNTOF(reverblist);i++)
{
const EFXEAXREVERBPROPERTIES *props;
if(strcasecmp(name, reverblist[i].name) != 0)
continue;
TRACE("Loading reverb '%s'\n", reverblist[i].name);
props = &reverblist[i].props;
effect->Props.Reverb.Density = props->flDensity;
effect->Props.Reverb.Diffusion = props->flDiffusion;
effect->Props.Reverb.Gain = props->flGain;
effect->Props.Reverb.GainHF = props->flGainHF;
effect->Props.Reverb.GainLF = props->flGainLF;
effect->Props.Reverb.DecayTime = props->flDecayTime;
effect->Props.Reverb.DecayHFRatio = props->flDecayHFRatio;
effect->Props.Reverb.DecayLFRatio = props->flDecayLFRatio;
effect->Props.Reverb.ReflectionsGain = props->flReflectionsGain;
effect->Props.Reverb.ReflectionsDelay = props->flReflectionsDelay;
effect->Props.Reverb.ReflectionsPan[0] = props->flReflectionsPan[0];
effect->Props.Reverb.ReflectionsPan[1] = props->flReflectionsPan[1];
effect->Props.Reverb.ReflectionsPan[2] = props->flReflectionsPan[2];
effect->Props.Reverb.LateReverbGain = props->flLateReverbGain;
effect->Props.Reverb.LateReverbDelay = props->flLateReverbDelay;
effect->Props.Reverb.LateReverbPan[0] = props->flLateReverbPan[0];
effect->Props.Reverb.LateReverbPan[1] = props->flLateReverbPan[1];
effect->Props.Reverb.LateReverbPan[2] = props->flLateReverbPan[2];
effect->Props.Reverb.EchoTime = props->flEchoTime;
effect->Props.Reverb.EchoDepth = props->flEchoDepth;
effect->Props.Reverb.ModulationTime = props->flModulationTime;
effect->Props.Reverb.ModulationDepth = props->flModulationDepth;
effect->Props.Reverb.AirAbsorptionGainHF = props->flAirAbsorptionGainHF;
effect->Props.Reverb.HFReference = props->flHFReference;
effect->Props.Reverb.LFReference = props->flLFReference;
effect->Props.Reverb.RoomRolloffFactor = props->flRoomRolloffFactor;
effect->Props.Reverb.DecayHFLimit = props->iDecayHFLimit;
return;
}
WARN("Reverb preset '%s' not found\n", name);
}
-671
View File
@@ -1,671 +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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 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 "alError.h"
#define FILTER_MIN_GAIN 0.0f
#define FILTER_MAX_GAIN 4.0f /* +12dB */
extern inline void LockFilterList(ALCdevice *device);
extern inline void UnlockFilterList(ALCdevice *device);
static ALfilter *AllocFilter(ALCcontext *context);
static void FreeFilter(ALCdevice *device, ALfilter *filter);
static void InitFilterParams(ALfilter *filter, ALenum type);
static inline ALfilter *LookupFilter(ALCdevice *device, ALuint id)
{
FilterSubList *sublist;
ALuint lidx = (id-1) >> 6;
ALsizei slidx = (id-1) & 0x3f;
if(UNLIKELY(lidx >= VECTOR_SIZE(device->FilterList)))
return NULL;
sublist = &VECTOR_ELEM(device->FilterList, lidx);
if(UNLIKELY(sublist->FreeMask & (U64(1)<<slidx)))
return NULL;
return sublist->Filters + slidx;
}
AL_API ALvoid AL_APIENTRY alGenFilters(ALsizei n, ALuint *filters)
{
ALCcontext *context;
ALsizei cur = 0;
context = GetContextRef();
if(!context) return;
if(!(n >= 0))
alSetError(context, AL_INVALID_VALUE, "Generating %d filters", n);
else for(cur = 0;cur < n;cur++)
{
ALfilter *filter = AllocFilter(context);
if(!filter)
{
alDeleteFilters(cur, filters);
break;
}
filters[cur] = filter->id;
}
ALCcontext_DecRef(context);
}
AL_API ALvoid AL_APIENTRY alDeleteFilters(ALsizei n, const ALuint *filters)
{
ALCdevice *device;
ALCcontext *context;
ALfilter *filter;
ALsizei i;
context = GetContextRef();
if(!context) return;
device = context->Device;
LockFilterList(device);
if(!(n >= 0))
SETERR_GOTO(context, AL_INVALID_VALUE, done, "Deleting %d filters", n);
for(i = 0;i < n;i++)
{
if(filters[i] && LookupFilter(device, filters[i]) == NULL)
SETERR_GOTO(context, AL_INVALID_NAME, done, "Invalid filter ID %u", filters[i]);
}
for(i = 0;i < n;i++)
{
if((filter=LookupFilter(device, filters[i])) != NULL)
FreeFilter(device, filter);
}
done:
UnlockFilterList(device);
ALCcontext_DecRef(context);
}
AL_API ALboolean AL_APIENTRY alIsFilter(ALuint filter)
{
ALCcontext *Context;
ALboolean result;
Context = GetContextRef();
if(!Context) return AL_FALSE;
LockFilterList(Context->Device);
result = ((!filter || LookupFilter(Context->Device, filter)) ?
AL_TRUE : AL_FALSE);
UnlockFilterList(Context->Device);
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;
LockFilterList(Device);
if((ALFilter=LookupFilter(Device, filter)) == NULL)
alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter);
else
{
if(param == AL_FILTER_TYPE)
{
if(value == AL_FILTER_NULL || value == AL_FILTER_LOWPASS ||
value == AL_FILTER_HIGHPASS || value == AL_FILTER_BANDPASS)
InitFilterParams(ALFilter, value);
else
alSetError(Context, AL_INVALID_VALUE, "Invalid filter type 0x%04x", value);
}
else
{
/* Call the appropriate handler */
ALfilter_setParami(ALFilter, Context, param, value);
}
}
UnlockFilterList(Device);
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;
LockFilterList(Device);
if((ALFilter=LookupFilter(Device, filter)) == NULL)
alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter);
else
{
/* Call the appropriate handler */
ALfilter_setParamiv(ALFilter, Context, param, values);
}
UnlockFilterList(Device);
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;
LockFilterList(Device);
if((ALFilter=LookupFilter(Device, filter)) == NULL)
alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter);
else
{
/* Call the appropriate handler */
ALfilter_setParamf(ALFilter, Context, param, value);
}
UnlockFilterList(Device);
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;
LockFilterList(Device);
if((ALFilter=LookupFilter(Device, filter)) == NULL)
alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter);
else
{
/* Call the appropriate handler */
ALfilter_setParamfv(ALFilter, Context, param, values);
}
UnlockFilterList(Device);
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;
LockFilterList(Device);
if((ALFilter=LookupFilter(Device, filter)) == NULL)
alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter);
else
{
if(param == AL_FILTER_TYPE)
*value = ALFilter->type;
else
{
/* Call the appropriate handler */
ALfilter_getParami(ALFilter, Context, param, value);
}
}
UnlockFilterList(Device);
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;
LockFilterList(Device);
if((ALFilter=LookupFilter(Device, filter)) == NULL)
alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter);
else
{
/* Call the appropriate handler */
ALfilter_getParamiv(ALFilter, Context, param, values);
}
UnlockFilterList(Device);
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;
LockFilterList(Device);
if((ALFilter=LookupFilter(Device, filter)) == NULL)
alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter);
else
{
/* Call the appropriate handler */
ALfilter_getParamf(ALFilter, Context, param, value);
}
UnlockFilterList(Device);
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;
LockFilterList(Device);
if((ALFilter=LookupFilter(Device, filter)) == NULL)
alSetError(Context, AL_INVALID_NAME, "Invalid filter ID %u", filter);
else
{
/* Call the appropriate handler */
ALfilter_getParamfv(ALFilter, Context, param, values);
}
UnlockFilterList(Device);
ALCcontext_DecRef(Context);
}
static void ALlowpass_setParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint UNUSED(val))
{ alSetError(context, AL_INVALID_ENUM, "Invalid low-pass integer property 0x%04x", param); }
static void ALlowpass_setParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, const ALint *UNUSED(vals))
{ alSetError(context, AL_INVALID_ENUM, "Invalid low-pass integer-vector property 0x%04x", param); }
static void ALlowpass_setParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val)
{
switch(param)
{
case AL_LOWPASS_GAIN:
if(!(val >= FILTER_MIN_GAIN && val <= FILTER_MAX_GAIN))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Low-pass gain %f out of range", val);
filter->Gain = val;
break;
case AL_LOWPASS_GAINHF:
if(!(val >= AL_LOWPASS_MIN_GAINHF && val <= AL_LOWPASS_MAX_GAINHF))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Low-pass gainhf %f out of range", val);
filter->GainHF = val;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid low-pass float property 0x%04x", param);
}
}
static void ALlowpass_setParamfv(ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals)
{ ALlowpass_setParamf(filter, context, param, vals[0]); }
static void ALlowpass_getParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(val))
{ alSetError(context, AL_INVALID_ENUM, "Invalid low-pass integer property 0x%04x", param); }
static void ALlowpass_getParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(vals))
{ alSetError(context, AL_INVALID_ENUM, "Invalid low-pass integer-vector property 0x%04x", param); }
static void ALlowpass_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, "Invalid low-pass float property 0x%04x", param);
}
}
static void ALlowpass_getParamfv(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals)
{ ALlowpass_getParamf(filter, context, param, vals); }
DEFINE_ALFILTER_VTABLE(ALlowpass);
static void ALhighpass_setParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint UNUSED(val))
{ alSetError(context, AL_INVALID_ENUM, "Invalid high-pass integer property 0x%04x", param); }
static void ALhighpass_setParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, const ALint *UNUSED(vals))
{ alSetError(context, AL_INVALID_ENUM, "Invalid high-pass integer-vector property 0x%04x", param); }
static void ALhighpass_setParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val)
{
switch(param)
{
case AL_HIGHPASS_GAIN:
if(!(val >= FILTER_MIN_GAIN && val <= FILTER_MAX_GAIN))
SETERR_RETURN(context, AL_INVALID_VALUE,, "High-pass gain out of range");
filter->Gain = val;
break;
case AL_HIGHPASS_GAINLF:
if(!(val >= AL_HIGHPASS_MIN_GAINLF && val <= AL_HIGHPASS_MAX_GAINLF))
SETERR_RETURN(context, AL_INVALID_VALUE,, "High-pass gainlf out of range");
filter->GainLF = val;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid high-pass float property 0x%04x", param);
}
}
static void ALhighpass_setParamfv(ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals)
{ ALhighpass_setParamf(filter, context, param, vals[0]); }
static void ALhighpass_getParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(val))
{ alSetError(context, AL_INVALID_ENUM, "Invalid high-pass integer property 0x%04x", param); }
static void ALhighpass_getParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(vals))
{ alSetError(context, AL_INVALID_ENUM, "Invalid high-pass integer-vector property 0x%04x", param); }
static void ALhighpass_getParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *val)
{
switch(param)
{
case AL_HIGHPASS_GAIN:
*val = filter->Gain;
break;
case AL_HIGHPASS_GAINLF:
*val = filter->GainLF;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid high-pass float property 0x%04x", param);
}
}
static void ALhighpass_getParamfv(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals)
{ ALhighpass_getParamf(filter, context, param, vals); }
DEFINE_ALFILTER_VTABLE(ALhighpass);
static void ALbandpass_setParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint UNUSED(val))
{ alSetError(context, AL_INVALID_ENUM, "Invalid band-pass integer property 0x%04x", param); }
static void ALbandpass_setParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, const ALint *UNUSED(vals))
{ alSetError(context, AL_INVALID_ENUM, "Invalid band-pass integer-vector property 0x%04x", param); }
static void ALbandpass_setParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val)
{
switch(param)
{
case AL_BANDPASS_GAIN:
if(!(val >= FILTER_MIN_GAIN && val <= FILTER_MAX_GAIN))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Band-pass gain out of range");
filter->Gain = val;
break;
case AL_BANDPASS_GAINHF:
if(!(val >= AL_BANDPASS_MIN_GAINHF && val <= AL_BANDPASS_MAX_GAINHF))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Band-pass gainhf out of range");
filter->GainHF = val;
break;
case AL_BANDPASS_GAINLF:
if(!(val >= AL_BANDPASS_MIN_GAINLF && val <= AL_BANDPASS_MAX_GAINLF))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Band-pass gainlf out of range");
filter->GainLF = val;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid band-pass float property 0x%04x", param);
}
}
static void ALbandpass_setParamfv(ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals)
{ ALbandpass_setParamf(filter, context, param, vals[0]); }
static void ALbandpass_getParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(val))
{ alSetError(context, AL_INVALID_ENUM, "Invalid band-pass integer property 0x%04x", param); }
static void ALbandpass_getParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(vals))
{ alSetError(context, AL_INVALID_ENUM, "Invalid band-pass integer-vector property 0x%04x", param); }
static void ALbandpass_getParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *val)
{
switch(param)
{
case AL_BANDPASS_GAIN:
*val = filter->Gain;
break;
case AL_BANDPASS_GAINHF:
*val = filter->GainHF;
break;
case AL_BANDPASS_GAINLF:
*val = filter->GainLF;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid band-pass float property 0x%04x", param);
}
}
static void ALbandpass_getParamfv(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals)
{ ALbandpass_getParamf(filter, context, param, vals); }
DEFINE_ALFILTER_VTABLE(ALbandpass);
static void ALnullfilter_setParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint UNUSED(val))
{ alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
static void ALnullfilter_setParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, const ALint *UNUSED(vals))
{ alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
static void ALnullfilter_setParamf(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALfloat UNUSED(val))
{ alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
static void ALnullfilter_setParamfv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, const ALfloat *UNUSED(vals))
{ alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
static void ALnullfilter_getParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(val))
{ alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
static void ALnullfilter_getParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint *UNUSED(vals))
{ alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
static void ALnullfilter_getParamf(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALfloat *UNUSED(val))
{ alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
static void ALnullfilter_getParamfv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALfloat *UNUSED(vals))
{ alSetError(context, AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
DEFINE_ALFILTER_VTABLE(ALnullfilter);
static ALfilter *AllocFilter(ALCcontext *context)
{
ALCdevice *device = context->Device;
FilterSubList *sublist, *subend;
ALfilter *filter = NULL;
ALsizei lidx = 0;
ALsizei slidx;
almtx_lock(&device->FilterLock);
sublist = VECTOR_BEGIN(device->FilterList);
subend = VECTOR_END(device->FilterList);
for(;sublist != subend;++sublist)
{
if(sublist->FreeMask)
{
slidx = CTZ64(sublist->FreeMask);
filter = sublist->Filters + slidx;
break;
}
++lidx;
}
if(UNLIKELY(!filter))
{
const FilterSubList empty_sublist = { 0, NULL };
/* Don't allocate so many list entries that the 32-bit ID could
* overflow...
*/
if(UNLIKELY(VECTOR_SIZE(device->FilterList) >= 1<<25))
{
almtx_unlock(&device->FilterLock);
alSetError(context, AL_OUT_OF_MEMORY, "Too many filters allocated");
return NULL;
}
lidx = (ALsizei)VECTOR_SIZE(device->FilterList);
VECTOR_PUSH_BACK(device->FilterList, empty_sublist);
sublist = &VECTOR_BACK(device->FilterList);
sublist->FreeMask = ~U64(0);
sublist->Filters = al_calloc(16, sizeof(ALfilter)*64);
if(UNLIKELY(!sublist->Filters))
{
VECTOR_POP_BACK(device->FilterList);
almtx_unlock(&device->FilterLock);
alSetError(context, AL_OUT_OF_MEMORY, "Failed to allocate filter batch");
return NULL;
}
slidx = 0;
filter = sublist->Filters + slidx;
}
memset(filter, 0, sizeof(*filter));
InitFilterParams(filter, AL_FILTER_NULL);
/* Add 1 to avoid filter ID 0. */
filter->id = ((lidx<<6) | slidx) + 1;
sublist->FreeMask &= ~(U64(1)<<slidx);
almtx_unlock(&device->FilterLock);
return filter;
}
static void FreeFilter(ALCdevice *device, ALfilter *filter)
{
ALuint id = filter->id - 1;
ALsizei lidx = id >> 6;
ALsizei slidx = id & 0x3f;
memset(filter, 0, sizeof(*filter));
VECTOR_ELEM(device->FilterList, lidx).FreeMask |= U64(1) << slidx;
}
void ReleaseALFilters(ALCdevice *device)
{
FilterSubList *sublist = VECTOR_BEGIN(device->FilterList);
FilterSubList *subend = VECTOR_END(device->FilterList);
size_t leftover = 0;
for(;sublist != subend;++sublist)
{
ALuint64 usemask = ~sublist->FreeMask;
while(usemask)
{
ALsizei idx = CTZ64(usemask);
ALfilter *filter = sublist->Filters + idx;
memset(filter, 0, sizeof(*filter));
++leftover;
usemask &= ~(U64(1) << idx);
}
sublist->FreeMask = ~usemask;
}
if(leftover > 0)
WARN("(%p) Deleted "SZFMT" Filter%s\n", device, leftover, (leftover==1)?"":"s");
}
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->HFReference = LOWPASSFREQREF;
filter->GainLF = 1.0f;
filter->LFReference = HIGHPASSFREQREF;
filter->vtab = &ALlowpass_vtable;
}
else if(type == AL_FILTER_HIGHPASS)
{
filter->Gain = AL_HIGHPASS_DEFAULT_GAIN;
filter->GainHF = 1.0f;
filter->HFReference = LOWPASSFREQREF;
filter->GainLF = AL_HIGHPASS_DEFAULT_GAINLF;
filter->LFReference = HIGHPASSFREQREF;
filter->vtab = &ALhighpass_vtable;
}
else if(type == AL_FILTER_BANDPASS)
{
filter->Gain = AL_BANDPASS_DEFAULT_GAIN;
filter->GainHF = AL_BANDPASS_DEFAULT_GAINHF;
filter->HFReference = LOWPASSFREQREF;
filter->GainLF = AL_BANDPASS_DEFAULT_GAINLF;
filter->LFReference = HIGHPASSFREQREF;
filter->vtab = &ALbandpass_vtable;
}
else
{
filter->Gain = 1.0f;
filter->GainHF = 1.0f;
filter->HFReference = LOWPASSFREQREF;
filter->GainLF = 1.0f;
filter->LFReference = HIGHPASSFREQREF;
filter->vtab = &ALnullfilter_vtable;
}
filter->type = type;
}
-502
View File
@@ -1,502 +0,0 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2000 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include "alMain.h"
#include "alu.h"
#include "alError.h"
#include "alListener.h"
#include "alSource.h"
#define DO_UPDATEPROPS() do { \
if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire)) \
UpdateListenerProps(context); \
else \
ATOMIC_FLAG_CLEAR(&listener->PropsClean, almemory_order_release); \
} while(0)
AL_API ALvoid AL_APIENTRY alListenerf(ALenum param, ALfloat value)
{
ALlistener *listener;
ALCcontext *context;
context = GetContextRef();
if(!context) return;
listener = context->Listener;
almtx_lock(&context->PropLock);
switch(param)
{
case AL_GAIN:
if(!(value >= 0.0f && isfinite(value)))
SETERR_GOTO(context, AL_INVALID_VALUE, done, "Listener gain out of range");
listener->Gain = value;
DO_UPDATEPROPS();
break;
case AL_METERS_PER_UNIT:
if(!(value >= AL_MIN_METERS_PER_UNIT && value <= AL_MAX_METERS_PER_UNIT))
SETERR_GOTO(context, AL_INVALID_VALUE, done, "Listener meters per unit out of range");
context->MetersPerUnit = value;
if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
UpdateContextProps(context);
else
ATOMIC_FLAG_CLEAR(&context->PropsClean, almemory_order_release);
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid listener float property");
}
done:
almtx_unlock(&context->PropLock);
ALCcontext_DecRef(context);
}
AL_API ALvoid AL_APIENTRY alListener3f(ALenum param, ALfloat value1, ALfloat value2, ALfloat value3)
{
ALlistener *listener;
ALCcontext *context;
context = GetContextRef();
if(!context) return;
listener = context->Listener;
almtx_lock(&context->PropLock);
switch(param)
{
case AL_POSITION:
if(!(isfinite(value1) && isfinite(value2) && isfinite(value3)))
SETERR_GOTO(context, AL_INVALID_VALUE, done, "Listener position out of range");
listener->Position[0] = value1;
listener->Position[1] = value2;
listener->Position[2] = value3;
DO_UPDATEPROPS();
break;
case AL_VELOCITY:
if(!(isfinite(value1) && isfinite(value2) && isfinite(value3)))
SETERR_GOTO(context, AL_INVALID_VALUE, done, "Listener velocity out of range");
listener->Velocity[0] = value1;
listener->Velocity[1] = value2;
listener->Velocity[2] = value3;
DO_UPDATEPROPS();
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid listener 3-float property");
}
done:
almtx_unlock(&context->PropLock);
ALCcontext_DecRef(context);
}
AL_API ALvoid AL_APIENTRY alListenerfv(ALenum param, const ALfloat *values)
{
ALlistener *listener;
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;
listener = context->Listener;
almtx_lock(&context->PropLock);
if(!values) SETERR_GOTO(context, AL_INVALID_VALUE, done, "NULL pointer");
switch(param)
{
case AL_ORIENTATION:
if(!(isfinite(values[0]) && isfinite(values[1]) && isfinite(values[2]) &&
isfinite(values[3]) && isfinite(values[4]) && isfinite(values[5])))
SETERR_GOTO(context, AL_INVALID_VALUE, done, "Listener orientation out of range");
/* AT then UP */
listener->Forward[0] = values[0];
listener->Forward[1] = values[1];
listener->Forward[2] = values[2];
listener->Up[0] = values[3];
listener->Up[1] = values[4];
listener->Up[2] = values[5];
DO_UPDATEPROPS();
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid listener float-vector property");
}
done:
almtx_unlock(&context->PropLock);
ALCcontext_DecRef(context);
}
AL_API ALvoid AL_APIENTRY alListeneri(ALenum param, ALint UNUSED(value))
{
ALCcontext *context;
context = GetContextRef();
if(!context) return;
almtx_lock(&context->PropLock);
switch(param)
{
default:
alSetError(context, AL_INVALID_ENUM, "Invalid listener integer property");
}
almtx_unlock(&context->PropLock);
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;
almtx_lock(&context->PropLock);
switch(param)
{
default:
alSetError(context, AL_INVALID_ENUM, "Invalid listener 3-integer property");
}
almtx_unlock(&context->PropLock);
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;
almtx_lock(&context->PropLock);
if(!values)
alSetError(context, AL_INVALID_VALUE, "NULL pointer");
else switch(param)
{
default:
alSetError(context, AL_INVALID_ENUM, "Invalid listener integer-vector property");
}
almtx_unlock(&context->PropLock);
ALCcontext_DecRef(context);
}
AL_API ALvoid AL_APIENTRY alGetListenerf(ALenum param, ALfloat *value)
{
ALCcontext *context;
context = GetContextRef();
if(!context) return;
almtx_lock(&context->PropLock);
if(!value)
alSetError(context, AL_INVALID_VALUE, "NULL pointer");
else switch(param)
{
case AL_GAIN:
*value = context->Listener->Gain;
break;
case AL_METERS_PER_UNIT:
*value = context->MetersPerUnit;
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid listener float property");
}
almtx_unlock(&context->PropLock);
ALCcontext_DecRef(context);
}
AL_API ALvoid AL_APIENTRY alGetListener3f(ALenum param, ALfloat *value1, ALfloat *value2, ALfloat *value3)
{
ALCcontext *context;
context = GetContextRef();
if(!context) return;
almtx_lock(&context->PropLock);
if(!value1 || !value2 || !value3)
alSetError(context, AL_INVALID_VALUE, "NULL pointer");
else switch(param)
{
case AL_POSITION:
*value1 = context->Listener->Position[0];
*value2 = context->Listener->Position[1];
*value3 = context->Listener->Position[2];
break;
case AL_VELOCITY:
*value1 = context->Listener->Velocity[0];
*value2 = context->Listener->Velocity[1];
*value3 = context->Listener->Velocity[2];
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid listener 3-float property");
}
almtx_unlock(&context->PropLock);
ALCcontext_DecRef(context);
}
AL_API ALvoid AL_APIENTRY alGetListenerfv(ALenum param, ALfloat *values)
{
ALCcontext *context;
switch(param)
{
case AL_GAIN:
case AL_METERS_PER_UNIT:
alGetListenerf(param, values);
return;
case AL_POSITION:
case AL_VELOCITY:
alGetListener3f(param, values+0, values+1, values+2);
return;
}
context = GetContextRef();
if(!context) return;
almtx_lock(&context->PropLock);
if(!values)
alSetError(context, AL_INVALID_VALUE, "NULL pointer");
else switch(param)
{
case AL_ORIENTATION:
// AT then UP
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];
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid listener float-vector property");
}
almtx_unlock(&context->PropLock);
ALCcontext_DecRef(context);
}
AL_API ALvoid AL_APIENTRY alGetListeneri(ALenum param, ALint *value)
{
ALCcontext *context;
context = GetContextRef();
if(!context) return;
almtx_lock(&context->PropLock);
if(!value)
alSetError(context, AL_INVALID_VALUE, "NULL pointer");
else switch(param)
{
default:
alSetError(context, AL_INVALID_ENUM, "Invalid listener integer property");
}
almtx_unlock(&context->PropLock);
ALCcontext_DecRef(context);
}
AL_API void AL_APIENTRY alGetListener3i(ALenum param, ALint *value1, ALint *value2, ALint *value3)
{
ALCcontext *context;
context = GetContextRef();
if(!context) return;
almtx_lock(&context->PropLock);
if(!value1 || !value2 || !value3)
alSetError(context, AL_INVALID_VALUE, "NULL pointer");
else switch(param)
{
case AL_POSITION:
*value1 = (ALint)context->Listener->Position[0];
*value2 = (ALint)context->Listener->Position[1];
*value3 = (ALint)context->Listener->Position[2];
break;
case AL_VELOCITY:
*value1 = (ALint)context->Listener->Velocity[0];
*value2 = (ALint)context->Listener->Velocity[1];
*value3 = (ALint)context->Listener->Velocity[2];
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid listener 3-integer property");
}
almtx_unlock(&context->PropLock);
ALCcontext_DecRef(context);
}
AL_API void AL_APIENTRY alGetListeneriv(ALenum param, ALint* values)
{
ALCcontext *context;
switch(param)
{
case AL_POSITION:
case AL_VELOCITY:
alGetListener3i(param, values+0, values+1, values+2);
return;
}
context = GetContextRef();
if(!context) return;
almtx_lock(&context->PropLock);
if(!values)
alSetError(context, AL_INVALID_VALUE, "NULL pointer");
else switch(param)
{
case AL_ORIENTATION:
// AT then UP
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];
break;
default:
alSetError(context, AL_INVALID_ENUM, "Invalid listener integer-vector property");
}
almtx_unlock(&context->PropLock);
ALCcontext_DecRef(context);
}
void UpdateListenerProps(ALCcontext *context)
{
ALlistener *listener = context->Listener;
struct ALlistenerProps *props;
/* Get an unused proprty container, or allocate a new one as needed. */
props = ATOMIC_LOAD(&context->FreeListenerProps, almemory_order_acquire);
if(!props)
props = al_calloc(16, sizeof(*props));
else
{
struct ALlistenerProps *next;
do {
next = ATOMIC_LOAD(&props->next, almemory_order_relaxed);
} while(ATOMIC_COMPARE_EXCHANGE_PTR_WEAK(&context->FreeListenerProps, &props, next,
almemory_order_seq_cst, almemory_order_acquire) == 0);
}
/* Copy in current property values. */
props->Position[0] = listener->Position[0];
props->Position[1] = listener->Position[1];
props->Position[2] = listener->Position[2];
props->Velocity[0] = listener->Velocity[0];
props->Velocity[1] = listener->Velocity[1];
props->Velocity[2] = listener->Velocity[2];
props->Forward[0] = listener->Forward[0];
props->Forward[1] = listener->Forward[1];
props->Forward[2] = listener->Forward[2];
props->Up[0] = listener->Up[0];
props->Up[1] = listener->Up[1];
props->Up[2] = listener->Up[2];
props->Gain = listener->Gain;
/* Set the new container for updating internal parameters. */
props = ATOMIC_EXCHANGE_PTR(&listener->Update, props, almemory_order_acq_rel);
if(props)
{
/* If there was an unused update container, put it back in the
* freelist.
*/
ATOMIC_REPLACE_HEAD(struct ALlistenerProps*, &context->FreeListenerProps, props);
}
}
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-900
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@@ -1,900 +0,0 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2000 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include "version.h"
#include <stdlib.h>
#include "alMain.h"
#include "AL/alc.h"
#include "AL/al.h"
#include "AL/alext.h"
#include "alError.h"
#include "alListener.h"
#include "alSource.h"
#include "alAuxEffectSlot.h"
#include "backends/base.h"
static const ALchar alVendor[] = "OpenAL Community";
static const ALchar alVersion[] = "1.1 ALSOFT "ALSOFT_VERSION;
static const ALchar alRenderer[] = "OpenAL Soft";
// Error Messages
static const ALchar alNoError[] = "No Error";
static const ALchar alErrInvalidName[] = "Invalid Name";
static const ALchar alErrInvalidEnum[] = "Invalid Enum";
static const ALchar alErrInvalidValue[] = "Invalid Value";
static const ALchar alErrInvalidOp[] = "Invalid Operation";
static const ALchar alErrOutOfMemory[] = "Out of Memory";
/* Resampler strings */
static const ALchar alPointResampler[] = "Nearest";
static const ALchar alLinearResampler[] = "Linear";
static const ALchar alCubicResampler[] = "Cubic";
static const ALchar alBSinc12Resampler[] = "11th order Sinc";
static const ALchar alBSinc24Resampler[] = "23rd order Sinc";
/* WARNING: Non-standard export! Not part of any extension, or exposed in the
* alcFunctions list.
*/
AL_API const ALchar* AL_APIENTRY alsoft_get_version(void)
{
const char *spoof = getenv("ALSOFT_SPOOF_VERSION");
if(spoof && spoof[0] != '\0') return spoof;
return ALSOFT_VERSION;
}
#define DO_UPDATEPROPS() do { \
if(!ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire)) \
UpdateContextProps(context); \
else \
ATOMIC_FLAG_CLEAR(&context->PropsClean, almemory_order_release); \
} while(0)
AL_API ALvoid AL_APIENTRY alEnable(ALenum capability)
{
ALCcontext *context;
context = GetContextRef();
if(!context) return;
almtx_lock(&context->PropLock);
switch(capability)
{
case AL_SOURCE_DISTANCE_MODEL:
context->SourceDistanceModel = AL_TRUE;
DO_UPDATEPROPS();
break;
default:
alSetError(context, AL_INVALID_VALUE, "Invalid enable property 0x%04x", capability);
}
almtx_unlock(&context->PropLock);
ALCcontext_DecRef(context);
}
AL_API ALvoid AL_APIENTRY alDisable(ALenum capability)
{
ALCcontext *context;
context = GetContextRef();
if(!context) return;
almtx_lock(&context->PropLock);
switch(capability)
{
case AL_SOURCE_DISTANCE_MODEL:
context->SourceDistanceModel = AL_FALSE;
DO_UPDATEPROPS();
break;
default:
alSetError(context, AL_INVALID_VALUE, "Invalid disable property 0x%04x", capability);
}
almtx_unlock(&context->PropLock);
ALCcontext_DecRef(context);
}
AL_API ALboolean AL_APIENTRY alIsEnabled(ALenum capability)
{
ALCcontext *context;
ALboolean value=AL_FALSE;
context = GetContextRef();
if(!context) return AL_FALSE;
almtx_lock(&context->PropLock);
switch(capability)
{
case AL_SOURCE_DISTANCE_MODEL:
value = context->SourceDistanceModel;
break;
default:
alSetError(context, AL_INVALID_VALUE, "Invalid is enabled property 0x%04x", capability);
}
almtx_unlock(&context->PropLock);
ALCcontext_DecRef(context);
return value;
}
AL_API ALboolean AL_APIENTRY alGetBoolean(ALenum pname)
{
ALCcontext *context;
ALboolean value=AL_FALSE;
context = GetContextRef();
if(!context) return AL_FALSE;
almtx_lock(&context->PropLock);
switch(pname)
{
case AL_DOPPLER_FACTOR:
if(context->DopplerFactor != 0.0f)
value = AL_TRUE;
break;
case AL_DOPPLER_VELOCITY:
if(context->DopplerVelocity != 0.0f)
value = AL_TRUE;
break;
case AL_DISTANCE_MODEL:
if(context->DistanceModel == AL_INVERSE_DISTANCE_CLAMPED)
value = AL_TRUE;
break;
case AL_SPEED_OF_SOUND:
if(context->SpeedOfSound != 0.0f)
value = AL_TRUE;
break;
case AL_DEFERRED_UPDATES_SOFT:
if(ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
value = AL_TRUE;
break;
case AL_GAIN_LIMIT_SOFT:
if(GAIN_MIX_MAX/context->GainBoost != 0.0f)
value = AL_TRUE;
break;
case AL_NUM_RESAMPLERS_SOFT:
/* Always non-0. */
value = AL_TRUE;
break;
case AL_DEFAULT_RESAMPLER_SOFT:
value = ResamplerDefault ? AL_TRUE : AL_FALSE;
break;
default:
alSetError(context, AL_INVALID_VALUE, "Invalid boolean property 0x%04x", pname);
}
almtx_unlock(&context->PropLock);
ALCcontext_DecRef(context);
return value;
}
AL_API ALdouble AL_APIENTRY alGetDouble(ALenum pname)
{
ALCcontext *context;
ALdouble value = 0.0;
context = GetContextRef();
if(!context) return 0.0;
almtx_lock(&context->PropLock);
switch(pname)
{
case AL_DOPPLER_FACTOR:
value = (ALdouble)context->DopplerFactor;
break;
case AL_DOPPLER_VELOCITY:
value = (ALdouble)context->DopplerVelocity;
break;
case AL_DISTANCE_MODEL:
value = (ALdouble)context->DistanceModel;
break;
case AL_SPEED_OF_SOUND:
value = (ALdouble)context->SpeedOfSound;
break;
case AL_DEFERRED_UPDATES_SOFT:
if(ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
value = (ALdouble)AL_TRUE;
break;
case AL_GAIN_LIMIT_SOFT:
value = (ALdouble)GAIN_MIX_MAX/context->GainBoost;
break;
case AL_NUM_RESAMPLERS_SOFT:
value = (ALdouble)(ResamplerMax + 1);
break;
case AL_DEFAULT_RESAMPLER_SOFT:
value = (ALdouble)ResamplerDefault;
break;
default:
alSetError(context, AL_INVALID_VALUE, "Invalid double property 0x%04x", pname);
}
almtx_unlock(&context->PropLock);
ALCcontext_DecRef(context);
return value;
}
AL_API ALfloat AL_APIENTRY alGetFloat(ALenum pname)
{
ALCcontext *context;
ALfloat value = 0.0f;
context = GetContextRef();
if(!context) return 0.0f;
almtx_lock(&context->PropLock);
switch(pname)
{
case AL_DOPPLER_FACTOR:
value = context->DopplerFactor;
break;
case AL_DOPPLER_VELOCITY:
value = context->DopplerVelocity;
break;
case AL_DISTANCE_MODEL:
value = (ALfloat)context->DistanceModel;
break;
case AL_SPEED_OF_SOUND:
value = context->SpeedOfSound;
break;
case AL_DEFERRED_UPDATES_SOFT:
if(ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
value = (ALfloat)AL_TRUE;
break;
case AL_GAIN_LIMIT_SOFT:
value = GAIN_MIX_MAX/context->GainBoost;
break;
case AL_NUM_RESAMPLERS_SOFT:
value = (ALfloat)(ResamplerMax + 1);
break;
case AL_DEFAULT_RESAMPLER_SOFT:
value = (ALfloat)ResamplerDefault;
break;
default:
alSetError(context, AL_INVALID_VALUE, "Invalid float property 0x%04x", pname);
}
almtx_unlock(&context->PropLock);
ALCcontext_DecRef(context);
return value;
}
AL_API ALint AL_APIENTRY alGetInteger(ALenum pname)
{
ALCcontext *context;
ALint value = 0;
context = GetContextRef();
if(!context) return 0;
almtx_lock(&context->PropLock);
switch(pname)
{
case AL_DOPPLER_FACTOR:
value = (ALint)context->DopplerFactor;
break;
case AL_DOPPLER_VELOCITY:
value = (ALint)context->DopplerVelocity;
break;
case AL_DISTANCE_MODEL:
value = (ALint)context->DistanceModel;
break;
case AL_SPEED_OF_SOUND:
value = (ALint)context->SpeedOfSound;
break;
case AL_DEFERRED_UPDATES_SOFT:
if(ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
value = (ALint)AL_TRUE;
break;
case AL_GAIN_LIMIT_SOFT:
value = (ALint)(GAIN_MIX_MAX/context->GainBoost);
break;
case AL_NUM_RESAMPLERS_SOFT:
value = ResamplerMax + 1;
break;
case AL_DEFAULT_RESAMPLER_SOFT:
value = ResamplerDefault;
break;
default:
alSetError(context, AL_INVALID_VALUE, "Invalid integer property 0x%04x", pname);
}
almtx_unlock(&context->PropLock);
ALCcontext_DecRef(context);
return value;
}
AL_API ALint64SOFT AL_APIENTRY alGetInteger64SOFT(ALenum pname)
{
ALCcontext *context;
ALint64SOFT value = 0;
context = GetContextRef();
if(!context) return 0;
almtx_lock(&context->PropLock);
switch(pname)
{
case AL_DOPPLER_FACTOR:
value = (ALint64SOFT)context->DopplerFactor;
break;
case AL_DOPPLER_VELOCITY:
value = (ALint64SOFT)context->DopplerVelocity;
break;
case AL_DISTANCE_MODEL:
value = (ALint64SOFT)context->DistanceModel;
break;
case AL_SPEED_OF_SOUND:
value = (ALint64SOFT)context->SpeedOfSound;
break;
case AL_DEFERRED_UPDATES_SOFT:
if(ATOMIC_LOAD(&context->DeferUpdates, almemory_order_acquire))
value = (ALint64SOFT)AL_TRUE;
break;
case AL_GAIN_LIMIT_SOFT:
value = (ALint64SOFT)(GAIN_MIX_MAX/context->GainBoost);
break;
case AL_NUM_RESAMPLERS_SOFT:
value = (ALint64SOFT)(ResamplerMax + 1);
break;
case AL_DEFAULT_RESAMPLER_SOFT:
value = (ALint64SOFT)ResamplerDefault;
break;
default:
alSetError(context, AL_INVALID_VALUE, "Invalid integer64 property 0x%04x", pname);
}
almtx_unlock(&context->PropLock);
ALCcontext_DecRef(context);
return value;
}
AL_API void* AL_APIENTRY alGetPointerSOFT(ALenum pname)
{
ALCcontext *context;
void *value = NULL;
context = GetContextRef();
if(!context) return NULL;
almtx_lock(&context->PropLock);
switch(pname)
{
case AL_EVENT_CALLBACK_FUNCTION_SOFT:
value = context->EventCb;
break;
case AL_EVENT_CALLBACK_USER_PARAM_SOFT:
value = context->EventParam;
break;
default:
alSetError(context, AL_INVALID_VALUE, "Invalid pointer property 0x%04x", pname);
}
almtx_unlock(&context->PropLock);
ALCcontext_DecRef(context);
return value;
}
AL_API ALvoid AL_APIENTRY alGetBooleanv(ALenum pname, ALboolean *values)
{
ALCcontext *context;
if(values)
{
switch(pname)
{
case AL_DOPPLER_FACTOR:
case AL_DOPPLER_VELOCITY:
case AL_DISTANCE_MODEL:
case AL_SPEED_OF_SOUND:
case AL_DEFERRED_UPDATES_SOFT:
case AL_GAIN_LIMIT_SOFT:
case AL_NUM_RESAMPLERS_SOFT:
case AL_DEFAULT_RESAMPLER_SOFT:
values[0] = alGetBoolean(pname);
return;
}
}
context = GetContextRef();
if(!context) return;
if(!values)
alSetError(context, AL_INVALID_VALUE, "NULL pointer");
switch(pname)
{
default:
alSetError(context, AL_INVALID_VALUE, "Invalid boolean-vector property 0x%04x", pname);
}
ALCcontext_DecRef(context);
}
AL_API ALvoid AL_APIENTRY alGetDoublev(ALenum pname, ALdouble *values)
{
ALCcontext *context;
if(values)
{
switch(pname)
{
case AL_DOPPLER_FACTOR:
case AL_DOPPLER_VELOCITY:
case AL_DISTANCE_MODEL:
case AL_SPEED_OF_SOUND:
case AL_DEFERRED_UPDATES_SOFT:
case AL_GAIN_LIMIT_SOFT:
case AL_NUM_RESAMPLERS_SOFT:
case AL_DEFAULT_RESAMPLER_SOFT:
values[0] = alGetDouble(pname);
return;
}
}
context = GetContextRef();
if(!context) return;
if(!values)
alSetError(context, AL_INVALID_VALUE, "NULL pointer");
switch(pname)
{
default:
alSetError(context, AL_INVALID_VALUE, "Invalid double-vector property 0x%04x", pname);
}
ALCcontext_DecRef(context);
}
AL_API ALvoid AL_APIENTRY alGetFloatv(ALenum pname, ALfloat *values)
{
ALCcontext *context;
if(values)
{
switch(pname)
{
case AL_DOPPLER_FACTOR:
case AL_DOPPLER_VELOCITY:
case AL_DISTANCE_MODEL:
case AL_SPEED_OF_SOUND:
case AL_DEFERRED_UPDATES_SOFT:
case AL_GAIN_LIMIT_SOFT:
case AL_NUM_RESAMPLERS_SOFT:
case AL_DEFAULT_RESAMPLER_SOFT:
values[0] = alGetFloat(pname);
return;
}
}
context = GetContextRef();
if(!context) return;
if(!values)
alSetError(context, AL_INVALID_VALUE, "NULL pointer");
switch(pname)
{
default:
alSetError(context, AL_INVALID_VALUE, "Invalid float-vector property 0x%04x", pname);
}
ALCcontext_DecRef(context);
}
AL_API ALvoid AL_APIENTRY alGetIntegerv(ALenum pname, ALint *values)
{
ALCcontext *context;
if(values)
{
switch(pname)
{
case AL_DOPPLER_FACTOR:
case AL_DOPPLER_VELOCITY:
case AL_DISTANCE_MODEL:
case AL_SPEED_OF_SOUND:
case AL_DEFERRED_UPDATES_SOFT:
case AL_GAIN_LIMIT_SOFT:
case AL_NUM_RESAMPLERS_SOFT:
case AL_DEFAULT_RESAMPLER_SOFT:
values[0] = alGetInteger(pname);
return;
}
}
context = GetContextRef();
if(!context) return;
if(!values)
alSetError(context, AL_INVALID_VALUE, "NULL pointer");
switch(pname)
{
default:
alSetError(context, AL_INVALID_VALUE, "Invalid integer-vector property 0x%04x", pname);
}
ALCcontext_DecRef(context);
}
AL_API void AL_APIENTRY alGetInteger64vSOFT(ALenum pname, ALint64SOFT *values)
{
ALCcontext *context;
if(values)
{
switch(pname)
{
case AL_DOPPLER_FACTOR:
case AL_DOPPLER_VELOCITY:
case AL_DISTANCE_MODEL:
case AL_SPEED_OF_SOUND:
case AL_DEFERRED_UPDATES_SOFT:
case AL_GAIN_LIMIT_SOFT:
case AL_NUM_RESAMPLERS_SOFT:
case AL_DEFAULT_RESAMPLER_SOFT:
values[0] = alGetInteger64SOFT(pname);
return;
}
}
context = GetContextRef();
if(!context) return;
if(!values)
alSetError(context, AL_INVALID_VALUE, "NULL pointer");
switch(pname)
{
default:
alSetError(context, AL_INVALID_VALUE, "Invalid integer64-vector property 0x%04x", pname);
}
ALCcontext_DecRef(context);
}
AL_API void AL_APIENTRY alGetPointervSOFT(ALenum pname, void **values)
{
ALCcontext *context;
if(values)
{
switch(pname)
{
case AL_EVENT_CALLBACK_FUNCTION_SOFT:
case AL_EVENT_CALLBACK_USER_PARAM_SOFT:
values[0] = alGetPointerSOFT(pname);
return;
}
}
context = GetContextRef();
if(!context) return;
if(!values)
alSetError(context, AL_INVALID_VALUE, "NULL pointer");
switch(pname)
{
default:
alSetError(context, AL_INVALID_VALUE, "Invalid pointer-vector property 0x%04x", pname);
}
ALCcontext_DecRef(context);
}
AL_API const ALchar* AL_APIENTRY alGetString(ALenum pname)
{
const ALchar *value = NULL;
ALCcontext *context;
context = GetContextRef();
if(!context) return NULL;
switch(pname)
{
case AL_VENDOR:
value = alVendor;
break;
case AL_VERSION:
value = alVersion;
break;
case AL_RENDERER:
value = alRenderer;
break;
case AL_EXTENSIONS:
value = context->ExtensionList;
break;
case AL_NO_ERROR:
value = alNoError;
break;
case AL_INVALID_NAME:
value = alErrInvalidName;
break;
case AL_INVALID_ENUM:
value = alErrInvalidEnum;
break;
case AL_INVALID_VALUE:
value = alErrInvalidValue;
break;
case AL_INVALID_OPERATION:
value = alErrInvalidOp;
break;
case AL_OUT_OF_MEMORY:
value = alErrOutOfMemory;
break;
default:
alSetError(context, AL_INVALID_VALUE, "Invalid string property 0x%04x", pname);
}
ALCcontext_DecRef(context);
return value;
}
AL_API ALvoid AL_APIENTRY alDopplerFactor(ALfloat value)
{
ALCcontext *context;
context = GetContextRef();
if(!context) return;
if(!(value >= 0.0f && isfinite(value)))
alSetError(context, AL_INVALID_VALUE, "Doppler factor %f out of range", value);
else
{
almtx_lock(&context->PropLock);
context->DopplerFactor = value;
DO_UPDATEPROPS();
almtx_unlock(&context->PropLock);
}
ALCcontext_DecRef(context);
}
AL_API ALvoid AL_APIENTRY alDopplerVelocity(ALfloat value)
{
ALCcontext *context;
context = GetContextRef();
if(!context) return;
if((ATOMIC_LOAD(&context->EnabledEvts, almemory_order_relaxed)&EventType_Deprecated))
{
static const ALCchar msg[] =
"alDopplerVelocity is deprecated in AL1.1, use alSpeedOfSound";
const ALsizei msglen = (ALsizei)strlen(msg);
ALbitfieldSOFT enabledevts;
almtx_lock(&context->EventCbLock);
enabledevts = ATOMIC_LOAD(&context->EnabledEvts, almemory_order_relaxed);
if((enabledevts&EventType_Deprecated) && context->EventCb)
(*context->EventCb)(AL_EVENT_TYPE_DEPRECATED_SOFT, 0, 0, msglen, msg,
context->EventParam);
almtx_unlock(&context->EventCbLock);
}
if(!(value >= 0.0f && isfinite(value)))
alSetError(context, AL_INVALID_VALUE, "Doppler velocity %f out of range", value);
else
{
almtx_lock(&context->PropLock);
context->DopplerVelocity = value;
DO_UPDATEPROPS();
almtx_unlock(&context->PropLock);
}
ALCcontext_DecRef(context);
}
AL_API ALvoid AL_APIENTRY alSpeedOfSound(ALfloat value)
{
ALCcontext *context;
context = GetContextRef();
if(!context) return;
if(!(value > 0.0f && isfinite(value)))
alSetError(context, AL_INVALID_VALUE, "Speed of sound %f out of range", value);
else
{
almtx_lock(&context->PropLock);
context->SpeedOfSound = value;
DO_UPDATEPROPS();
almtx_unlock(&context->PropLock);
}
ALCcontext_DecRef(context);
}
AL_API ALvoid AL_APIENTRY alDistanceModel(ALenum value)
{
ALCcontext *context;
context = GetContextRef();
if(!context) return;
if(!(value == AL_INVERSE_DISTANCE || value == AL_INVERSE_DISTANCE_CLAMPED ||
value == AL_LINEAR_DISTANCE || value == AL_LINEAR_DISTANCE_CLAMPED ||
value == AL_EXPONENT_DISTANCE || value == AL_EXPONENT_DISTANCE_CLAMPED ||
value == AL_NONE))
alSetError(context, AL_INVALID_VALUE, "Distance model 0x%04x out of range", value);
else
{
almtx_lock(&context->PropLock);
context->DistanceModel = value;
if(!context->SourceDistanceModel)
DO_UPDATEPROPS();
almtx_unlock(&context->PropLock);
}
ALCcontext_DecRef(context);
}
AL_API ALvoid AL_APIENTRY alDeferUpdatesSOFT(void)
{
ALCcontext *context;
context = GetContextRef();
if(!context) return;
ALCcontext_DeferUpdates(context);
ALCcontext_DecRef(context);
}
AL_API ALvoid AL_APIENTRY alProcessUpdatesSOFT(void)
{
ALCcontext *context;
context = GetContextRef();
if(!context) return;
ALCcontext_ProcessUpdates(context);
ALCcontext_DecRef(context);
}
AL_API const ALchar* AL_APIENTRY alGetStringiSOFT(ALenum pname, ALsizei index)
{
const char *ResamplerNames[] = {
alPointResampler, alLinearResampler,
alCubicResampler, alBSinc12Resampler,
alBSinc24Resampler,
};
const ALchar *value = NULL;
ALCcontext *context;
static_assert(COUNTOF(ResamplerNames) == ResamplerMax+1, "Incorrect ResamplerNames list");
context = GetContextRef();
if(!context) return NULL;
switch(pname)
{
case AL_RESAMPLER_NAME_SOFT:
if(index < 0 || (size_t)index >= COUNTOF(ResamplerNames))
SETERR_GOTO(context, AL_INVALID_VALUE, done, "Resampler name index %d out of range",
index);
value = ResamplerNames[index];
break;
default:
alSetError(context, AL_INVALID_VALUE, "Invalid string indexed property");
}
done:
ALCcontext_DecRef(context);
return value;
}
void UpdateContextProps(ALCcontext *context)
{
struct ALcontextProps *props;
/* Get an unused proprty container, or allocate a new one as needed. */
props = ATOMIC_LOAD(&context->FreeContextProps, almemory_order_acquire);
if(!props)
props = al_calloc(16, sizeof(*props));
else
{
struct ALcontextProps *next;
do {
next = ATOMIC_LOAD(&props->next, almemory_order_relaxed);
} while(ATOMIC_COMPARE_EXCHANGE_PTR_WEAK(&context->FreeContextProps, &props, next,
almemory_order_seq_cst, almemory_order_acquire) == 0);
}
/* Copy in current property values. */
props->MetersPerUnit = context->MetersPerUnit;
props->DopplerFactor = context->DopplerFactor;
props->DopplerVelocity = context->DopplerVelocity;
props->SpeedOfSound = context->SpeedOfSound;
props->SourceDistanceModel = context->SourceDistanceModel;
props->DistanceModel = context->DistanceModel;
/* Set the new container for updating internal parameters. */
props = ATOMIC_EXCHANGE_PTR(&context->Update, props, almemory_order_acq_rel);
if(props)
{
/* If there was an unused update container, put it back in the
* freelist.
*/
ATOMIC_REPLACE_HEAD(struct ALcontextProps*, &context->FreeContextProps, props);
}
}
-127
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@@ -1,127 +0,0 @@
#include "config.h"
#include "AL/alc.h"
#include "AL/al.h"
#include "AL/alext.h"
#include "alMain.h"
#include "alError.h"
#include "alAuxEffectSlot.h"
#include "ringbuffer.h"
int EventThread(void *arg)
{
ALCcontext *context = arg;
bool quitnow = false;
while(!quitnow)
{
ALbitfieldSOFT enabledevts;
AsyncEvent evt;
if(ll_ringbuffer_read(context->AsyncEvents, (char*)&evt, 1) == 0)
{
alsem_wait(&context->EventSem);
continue;
}
almtx_lock(&context->EventCbLock);
do {
quitnow = evt.EnumType == EventType_KillThread;
if(quitnow) break;
if(evt.EnumType == EventType_ReleaseEffectState)
{
ALeffectState_DecRef(evt.u.EffectState);
continue;
}
enabledevts = ATOMIC_LOAD(&context->EnabledEvts, almemory_order_acquire);
if(context->EventCb && (enabledevts&evt.EnumType) == evt.EnumType)
context->EventCb(evt.u.user.type, evt.u.user.id, evt.u.user.param,
(ALsizei)strlen(evt.u.user.msg), evt.u.user.msg, context->EventParam
);
} while(ll_ringbuffer_read(context->AsyncEvents, (char*)&evt, 1) != 0);
almtx_unlock(&context->EventCbLock);
}
return 0;
}
AL_API void AL_APIENTRY alEventControlSOFT(ALsizei count, const ALenum *types, ALboolean enable)
{
ALCcontext *context;
ALbitfieldSOFT enabledevts;
ALbitfieldSOFT flags = 0;
ALsizei i;
context = GetContextRef();
if(!context) return;
if(count < 0) SETERR_GOTO(context, AL_INVALID_VALUE, done, "Controlling %d events", count);
if(count == 0) goto done;
if(!types) SETERR_GOTO(context, AL_INVALID_VALUE, done, "NULL pointer");
for(i = 0;i < count;i++)
{
if(types[i] == AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT)
flags |= EventType_BufferCompleted;
else if(types[i] == AL_EVENT_TYPE_SOURCE_STATE_CHANGED_SOFT)
flags |= EventType_SourceStateChange;
else if(types[i] == AL_EVENT_TYPE_ERROR_SOFT)
flags |= EventType_Error;
else if(types[i] == AL_EVENT_TYPE_PERFORMANCE_SOFT)
flags |= EventType_Performance;
else if(types[i] == AL_EVENT_TYPE_DEPRECATED_SOFT)
flags |= EventType_Deprecated;
else if(types[i] == AL_EVENT_TYPE_DISCONNECTED_SOFT)
flags |= EventType_Disconnected;
else
SETERR_GOTO(context, AL_INVALID_ENUM, done, "Invalid event type 0x%04x", types[i]);
}
if(enable)
{
enabledevts = ATOMIC_LOAD(&context->EnabledEvts, almemory_order_relaxed);
while(ATOMIC_COMPARE_EXCHANGE_WEAK(&context->EnabledEvts, &enabledevts, enabledevts|flags,
almemory_order_acq_rel, almemory_order_acquire) == 0)
{
/* enabledevts is (re-)filled with the current value on failure, so
* just try again.
*/
}
}
else
{
enabledevts = ATOMIC_LOAD(&context->EnabledEvts, almemory_order_relaxed);
while(ATOMIC_COMPARE_EXCHANGE_WEAK(&context->EnabledEvts, &enabledevts, enabledevts&~flags,
almemory_order_acq_rel, almemory_order_acquire) == 0)
{
}
/* Wait to ensure the event handler sees the changed flags before
* returning.
*/
almtx_lock(&context->EventCbLock);
almtx_unlock(&context->EventCbLock);
}
done:
ALCcontext_DecRef(context);
}
AL_API void AL_APIENTRY alEventCallbackSOFT(ALEVENTPROCSOFT callback, void *userParam)
{
ALCcontext *context;
context = GetContextRef();
if(!context) return;
almtx_lock(&context->PropLock);
almtx_lock(&context->EventCbLock);
context->EventCb = callback;
context->EventParam = userParam;
almtx_unlock(&context->EventCbLock);
almtx_unlock(&context->PropLock);
ALCcontext_DecRef(context);
}
-276
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@@ -1,276 +0,0 @@
#include "config.h"
#include "sample_cvt.h"
#include "AL/al.h"
#include "alu.h"
#include "alBuffer.h"
/* IMA ADPCM Stepsize table */
static const int IMAStep_size[89] = {
7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19,
21, 23, 25, 28, 31, 34, 37, 41, 45, 50, 55,
60, 66, 73, 80, 88, 97, 107, 118, 130, 143, 157,
173, 190, 209, 230, 253, 279, 307, 337, 371, 408, 449,
494, 544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282,
1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327, 3660,
4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630, 9493,10442,
11487,12635,13899,15289,16818,18500,20350,22358,24633,27086,29794,
32767
};
/* IMA4 ADPCM Codeword decode table */
static const int IMA4Codeword[16] = {
1, 3, 5, 7, 9, 11, 13, 15,
-1,-3,-5,-7,-9,-11,-13,-15,
};
/* IMA4 ADPCM Step index adjust decode table */
static const int IMA4Index_adjust[16] = {
-1,-1,-1,-1, 2, 4, 6, 8,
-1,-1,-1,-1, 2, 4, 6, 8
};
/* MSADPCM Adaption table */
static const int MSADPCMAdaption[16] = {
230, 230, 230, 230, 307, 409, 512, 614,
768, 614, 512, 409, 307, 230, 230, 230
};
/* MSADPCM Adaption Coefficient tables */
static const int MSADPCMAdaptionCoeff[7][2] = {
{ 256, 0 },
{ 512, -256 },
{ 0, 0 },
{ 192, 64 },
{ 240, 0 },
{ 460, -208 },
{ 392, -232 }
};
/* A quick'n'dirty lookup table to decode a muLaw-encoded byte sample into a
* signed 16-bit sample */
const ALshort muLawDecompressionTable[256] = {
-32124,-31100,-30076,-29052,-28028,-27004,-25980,-24956,
-23932,-22908,-21884,-20860,-19836,-18812,-17788,-16764,
-15996,-15484,-14972,-14460,-13948,-13436,-12924,-12412,
-11900,-11388,-10876,-10364, -9852, -9340, -8828, -8316,
-7932, -7676, -7420, -7164, -6908, -6652, -6396, -6140,
-5884, -5628, -5372, -5116, -4860, -4604, -4348, -4092,
-3900, -3772, -3644, -3516, -3388, -3260, -3132, -3004,
-2876, -2748, -2620, -2492, -2364, -2236, -2108, -1980,
-1884, -1820, -1756, -1692, -1628, -1564, -1500, -1436,
-1372, -1308, -1244, -1180, -1116, -1052, -988, -924,
-876, -844, -812, -780, -748, -716, -684, -652,
-620, -588, -556, -524, -492, -460, -428, -396,
-372, -356, -340, -324, -308, -292, -276, -260,
-244, -228, -212, -196, -180, -164, -148, -132,
-120, -112, -104, -96, -88, -80, -72, -64,
-56, -48, -40, -32, -24, -16, -8, 0,
32124, 31100, 30076, 29052, 28028, 27004, 25980, 24956,
23932, 22908, 21884, 20860, 19836, 18812, 17788, 16764,
15996, 15484, 14972, 14460, 13948, 13436, 12924, 12412,
11900, 11388, 10876, 10364, 9852, 9340, 8828, 8316,
7932, 7676, 7420, 7164, 6908, 6652, 6396, 6140,
5884, 5628, 5372, 5116, 4860, 4604, 4348, 4092,
3900, 3772, 3644, 3516, 3388, 3260, 3132, 3004,
2876, 2748, 2620, 2492, 2364, 2236, 2108, 1980,
1884, 1820, 1756, 1692, 1628, 1564, 1500, 1436,
1372, 1308, 1244, 1180, 1116, 1052, 988, 924,
876, 844, 812, 780, 748, 716, 684, 652,
620, 588, 556, 524, 492, 460, 428, 396,
372, 356, 340, 324, 308, 292, 276, 260,
244, 228, 212, 196, 180, 164, 148, 132,
120, 112, 104, 96, 88, 80, 72, 64,
56, 48, 40, 32, 24, 16, 8, 0
};
/* A quick'n'dirty lookup table to decode an aLaw-encoded byte sample into a
* signed 16-bit sample */
const ALshort aLawDecompressionTable[256] = {
-5504, -5248, -6016, -5760, -4480, -4224, -4992, -4736,
-7552, -7296, -8064, -7808, -6528, -6272, -7040, -6784,
-2752, -2624, -3008, -2880, -2240, -2112, -2496, -2368,
-3776, -3648, -4032, -3904, -3264, -3136, -3520, -3392,
-22016,-20992,-24064,-23040,-17920,-16896,-19968,-18944,
-30208,-29184,-32256,-31232,-26112,-25088,-28160,-27136,
-11008,-10496,-12032,-11520, -8960, -8448, -9984, -9472,
-15104,-14592,-16128,-15616,-13056,-12544,-14080,-13568,
-344, -328, -376, -360, -280, -264, -312, -296,
-472, -456, -504, -488, -408, -392, -440, -424,
-88, -72, -120, -104, -24, -8, -56, -40,
-216, -200, -248, -232, -152, -136, -184, -168,
-1376, -1312, -1504, -1440, -1120, -1056, -1248, -1184,
-1888, -1824, -2016, -1952, -1632, -1568, -1760, -1696,
-688, -656, -752, -720, -560, -528, -624, -592,
-944, -912, -1008, -976, -816, -784, -880, -848,
5504, 5248, 6016, 5760, 4480, 4224, 4992, 4736,
7552, 7296, 8064, 7808, 6528, 6272, 7040, 6784,
2752, 2624, 3008, 2880, 2240, 2112, 2496, 2368,
3776, 3648, 4032, 3904, 3264, 3136, 3520, 3392,
22016, 20992, 24064, 23040, 17920, 16896, 19968, 18944,
30208, 29184, 32256, 31232, 26112, 25088, 28160, 27136,
11008, 10496, 12032, 11520, 8960, 8448, 9984, 9472,
15104, 14592, 16128, 15616, 13056, 12544, 14080, 13568,
344, 328, 376, 360, 280, 264, 312, 296,
472, 456, 504, 488, 408, 392, 440, 424,
88, 72, 120, 104, 24, 8, 56, 40,
216, 200, 248, 232, 152, 136, 184, 168,
1376, 1312, 1504, 1440, 1120, 1056, 1248, 1184,
1888, 1824, 2016, 1952, 1632, 1568, 1760, 1696,
688, 656, 752, 720, 560, 528, 624, 592,
944, 912, 1008, 976, 816, 784, 880, 848
};
static void DecodeIMA4Block(ALshort *dst, const ALubyte *src, ALint numchans, ALsizei align)
{
ALint sample[MAX_INPUT_CHANNELS] = { 0 };
ALint index[MAX_INPUT_CHANNELS] = { 0 };
ALuint code[MAX_INPUT_CHANNELS] = { 0 };
ALsizei c, i;
for(c = 0;c < numchans;c++)
{
sample[c] = *(src++);
sample[c] |= *(src++) << 8;
sample[c] = (sample[c]^0x8000) - 32768;
index[c] = *(src++);
index[c] |= *(src++) << 8;
index[c] = (index[c]^0x8000) - 32768;
index[c] = clampi(index[c], 0, 88);
dst[c] = sample[c];
}
for(i = 1;i < align;i++)
{
if((i&7) == 1)
{
for(c = 0;c < numchans;c++)
{
code[c] = *(src++);
code[c] |= *(src++) << 8;
code[c] |= *(src++) << 16;
code[c] |= *(src++) << 24;
}
}
for(c = 0;c < numchans;c++)
{
int nibble = code[c]&0xf;
code[c] >>= 4;
sample[c] += IMA4Codeword[nibble] * IMAStep_size[index[c]] / 8;
sample[c] = clampi(sample[c], -32768, 32767);
index[c] += IMA4Index_adjust[nibble];
index[c] = clampi(index[c], 0, 88);
*(dst++) = sample[c];
}
}
}
static void DecodeMSADPCMBlock(ALshort *dst, const ALubyte *src, ALint numchans, ALsizei align)
{
ALubyte blockpred[MAX_INPUT_CHANNELS] = { 0 };
ALint delta[MAX_INPUT_CHANNELS] = { 0 };
ALshort samples[MAX_INPUT_CHANNELS][2] = { { 0, 0 } };
ALint c, i;
for(c = 0;c < numchans;c++)
{
blockpred[c] = *(src++);
blockpred[c] = minu(blockpred[c], 6);
}
for(c = 0;c < numchans;c++)
{
delta[c] = *(src++);
delta[c] |= *(src++) << 8;
delta[c] = (delta[c]^0x8000) - 32768;
}
for(c = 0;c < numchans;c++)
{
samples[c][0] = *(src++);
samples[c][0] |= *(src++) << 8;
samples[c][0] = (samples[c][0]^0x8000) - 32768;
}
for(c = 0;c < numchans;c++)
{
samples[c][1] = *(src++);
samples[c][1] |= *(src++) << 8;
samples[c][1] = (samples[c][1]^0x8000) - 0x8000;
}
/* Second sample is written first. */
for(c = 0;c < numchans;c++)
*(dst++) = samples[c][1];
for(c = 0;c < numchans;c++)
*(dst++) = samples[c][0];
for(i = 2;i < align;i++)
{
for(c = 0;c < numchans;c++)
{
const ALint num = (i*numchans) + c;
ALint nibble, pred;
/* Read the nibble (first is in the upper bits). */
if(!(num&1))
nibble = (*src>>4)&0x0f;
else
nibble = (*(src++))&0x0f;
pred = (samples[c][0]*MSADPCMAdaptionCoeff[blockpred[c]][0] +
samples[c][1]*MSADPCMAdaptionCoeff[blockpred[c]][1]) / 256;
pred += ((nibble^0x08) - 0x08) * delta[c];
pred = clampi(pred, -32768, 32767);
samples[c][1] = samples[c][0];
samples[c][0] = pred;
delta[c] = (MSADPCMAdaption[nibble] * delta[c]) / 256;
delta[c] = maxi(16, delta[c]);
*(dst++) = pred;
}
}
}
void Convert_ALshort_ALima4(ALshort *dst, const ALubyte *src, ALsizei numchans, ALsizei len,
ALsizei align)
{
ALsizei byte_align = ((align-1)/2 + 4) * numchans;
ALsizei i;
assert(align > 0 && (len%align) == 0);
for(i = 0;i < len;i += align)
{
DecodeIMA4Block(dst, src, numchans, align);
src += byte_align;
dst += align*numchans;
}
}
void Convert_ALshort_ALmsadpcm(ALshort *dst, const ALubyte *src, ALsizei numchans, ALsizei len,
ALsizei align)
{
ALsizei byte_align = ((align-2)/2 + 7) * numchans;
ALsizei i;
assert(align > 1 && (len%align) == 0);
for(i = 0;i < len;i += align)
{
DecodeMSADPCMBlock(dst, src, numchans, align);
src += byte_align;
dst += align*numchans;
}
}
+792
View File
@@ -0,0 +1,792 @@
/**
* 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include "auxeffectslot.h"
#include <algorithm>
#include <cstdint>
#include <iterator>
#include <memory>
#include <mutex>
#include <numeric>
#include <thread>
#include "AL/al.h"
#include "AL/alc.h"
#include "AL/efx.h"
#include "alcmain.h"
#include "alcontext.h"
#include "alexcpt.h"
#include "almalloc.h"
#include "alnumeric.h"
#include "alspan.h"
#include "alu.h"
#include "effect.h"
#include "fpu_modes.h"
#include "inprogext.h"
#include "logging.h"
#include "opthelpers.h"
namespace {
inline ALeffectslot *LookupEffectSlot(ALCcontext *context, ALuint id) noexcept
{
const size_t lidx{(id-1) >> 6};
const ALuint slidx{(id-1) & 0x3f};
if UNLIKELY(lidx >= context->mEffectSlotList.size())
return nullptr;
EffectSlotSubList &sublist{context->mEffectSlotList[lidx]};
if UNLIKELY(sublist.FreeMask & (1_u64 << slidx))
return nullptr;
return sublist.EffectSlots + slidx;
}
inline ALeffect *LookupEffect(ALCdevice *device, ALuint id) noexcept
{
const size_t lidx{(id-1) >> 6};
const ALuint slidx{(id-1) & 0x3f};
if UNLIKELY(lidx >= device->EffectList.size())
return nullptr;
EffectSubList &sublist = device->EffectList[lidx];
if UNLIKELY(sublist.FreeMask & (1_u64 << slidx))
return nullptr;
return sublist.Effects + slidx;
}
void AddActiveEffectSlots(const ALuint *slotids, size_t count, ALCcontext *context)
{
if(count < 1) return;
ALeffectslotArray *curarray{context->mActiveAuxSlots.load(std::memory_order_acquire)};
size_t newcount{curarray->size() + count};
/* Insert the new effect slots into the head of the array, followed by the
* existing ones.
*/
ALeffectslotArray *newarray = ALeffectslot::CreatePtrArray(newcount);
auto slotiter = std::transform(slotids, slotids+count, newarray->begin(),
[context](ALuint id) noexcept -> ALeffectslot*
{ return LookupEffectSlot(context, id); }
);
std::copy(curarray->begin(), curarray->end(), slotiter);
/* Remove any duplicates (first instance of each will be kept). */
auto last = newarray->end();
for(auto start=newarray->begin()+1;;)
{
last = std::remove(start, last, *(start-1));
if(start == last) break;
++start;
}
newcount = static_cast<size_t>(std::distance(newarray->begin(), last));
/* Reallocate newarray if the new size ended up smaller from duplicate
* removal.
*/
if UNLIKELY(newcount < newarray->size())
{
curarray = newarray;
newarray = ALeffectslot::CreatePtrArray(newcount);
std::copy_n(curarray->begin(), newcount, newarray->begin());
delete curarray;
curarray = nullptr;
}
curarray = context->mActiveAuxSlots.exchange(newarray, std::memory_order_acq_rel);
ALCdevice *device{context->mDevice.get()};
while((device->MixCount.load(std::memory_order_acquire)&1))
std::this_thread::yield();
delete curarray;
}
void RemoveActiveEffectSlots(const ALuint *slotids, size_t count, ALCcontext *context)
{
if(count < 1) return;
ALeffectslotArray *curarray{context->mActiveAuxSlots.load(std::memory_order_acquire)};
/* Don't shrink the allocated array size since we don't know how many (if
* any) of the effect slots to remove are in the array.
*/
ALeffectslotArray *newarray = ALeffectslot::CreatePtrArray(curarray->size());
/* Copy each element in curarray to newarray whose ID is not in slotids. */
const ALuint *slotids_end{slotids + count};
auto slotiter = std::copy_if(curarray->begin(), curarray->end(), newarray->begin(),
[slotids, slotids_end](const ALeffectslot *slot) -> bool
{ return std::find(slotids, slotids_end, slot->id) == slotids_end; }
);
/* Reallocate with the new size. */
auto newsize = static_cast<size_t>(std::distance(newarray->begin(), slotiter));
if LIKELY(newsize != newarray->size())
{
curarray = newarray;
newarray = ALeffectslot::CreatePtrArray(newsize);
std::copy_n(curarray->begin(), newsize, newarray->begin());
delete curarray;
curarray = nullptr;
}
curarray = context->mActiveAuxSlots.exchange(newarray, std::memory_order_acq_rel);
ALCdevice *device{context->mDevice.get()};
while((device->MixCount.load(std::memory_order_acquire)&1))
std::this_thread::yield();
delete curarray;
}
bool EnsureEffectSlots(ALCcontext *context, size_t needed)
{
size_t count{std::accumulate(context->mEffectSlotList.cbegin(),
context->mEffectSlotList.cend(), size_t{0},
[](size_t cur, const EffectSlotSubList &sublist) noexcept -> size_t
{ return cur + static_cast<ALuint>(POPCNT64(sublist.FreeMask)); }
)};
while(needed > count)
{
if UNLIKELY(context->mEffectSlotList.size() >= 1<<25)
return false;
context->mEffectSlotList.emplace_back();
auto sublist = context->mEffectSlotList.end() - 1;
sublist->FreeMask = ~0_u64;
sublist->EffectSlots = static_cast<ALeffectslot*>(
al_calloc(alignof(ALeffectslot), sizeof(ALeffectslot)*64));
if UNLIKELY(!sublist->EffectSlots)
{
context->mEffectSlotList.pop_back();
return false;
}
count += 64;
}
return true;
}
ALeffectslot *AllocEffectSlot(ALCcontext *context)
{
auto sublist = std::find_if(context->mEffectSlotList.begin(), context->mEffectSlotList.end(),
[](const EffectSlotSubList &entry) noexcept -> bool
{ return entry.FreeMask != 0; }
);
auto lidx = static_cast<ALuint>(std::distance(context->mEffectSlotList.begin(), sublist));
auto slidx = static_cast<ALuint>(CTZ64(sublist->FreeMask));
ALeffectslot *slot{::new (sublist->EffectSlots + slidx) ALeffectslot{}};
if(ALenum err{InitEffectSlot(slot)})
{
al::destroy_at(slot);
context->setError(err, "Effect slot object initialization failed");
return nullptr;
}
aluInitEffectPanning(slot, context->mDevice.get());
/* Add 1 to avoid source ID 0. */
slot->id = ((lidx<<6) | slidx) + 1;
context->mNumEffectSlots += 1;
sublist->FreeMask &= ~(1_u64 << slidx);
return slot;
}
void FreeEffectSlot(ALCcontext *context, ALeffectslot *slot)
{
const ALuint id{slot->id - 1};
const size_t lidx{id >> 6};
const ALuint slidx{id & 0x3f};
al::destroy_at(slot);
context->mEffectSlotList[lidx].FreeMask |= 1_u64 << slidx;
context->mNumEffectSlots--;
}
#define DO_UPDATEPROPS() do { \
if(!context->mDeferUpdates.load(std::memory_order_acquire)) \
UpdateEffectSlotProps(slot, context.get()); \
else \
slot->PropsClean.clear(std::memory_order_release); \
} while(0)
} // namespace
ALeffectslotArray *ALeffectslot::CreatePtrArray(size_t count) noexcept
{
/* Allocate space for twice as many pointers, so the mixer has scratch
* space to store a sorted list during mixing.
*/
void *ptr{al_calloc(alignof(ALeffectslotArray), ALeffectslotArray::Sizeof(count*2))};
return new (ptr) ALeffectslotArray{count};
}
AL_API ALvoid AL_APIENTRY alGenAuxiliaryEffectSlots(ALsizei n, ALuint *effectslots)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
if UNLIKELY(n < 0)
context->setError(AL_INVALID_VALUE, "Generating %d effect slots", n);
if UNLIKELY(n <= 0) return;
std::unique_lock<std::mutex> slotlock{context->mEffectSlotLock};
ALCdevice *device{context->mDevice.get()};
if(static_cast<ALuint>(n) > device->AuxiliaryEffectSlotMax-context->mNumEffectSlots)
{
context->setError(AL_OUT_OF_MEMORY, "Exceeding %u effect slot limit (%u + %d)",
device->AuxiliaryEffectSlotMax, context->mNumEffectSlots, n);
return;
}
if(!EnsureEffectSlots(context.get(), static_cast<ALuint>(n)))
{
context->setError(AL_OUT_OF_MEMORY, "Failed to allocate %d effectslot%s", n,
(n==1) ? "" : "s");
return;
}
if(n == 1)
{
ALeffectslot *slot{AllocEffectSlot(context.get())};
if(!slot) return;
effectslots[0] = slot->id;
}
else
{
al::vector<ALuint> ids;
ALsizei count{n};
ids.reserve(static_cast<ALuint>(count));
do {
ALeffectslot *slot{AllocEffectSlot(context.get())};
if(!slot)
{
slotlock.unlock();
alDeleteAuxiliaryEffectSlots(static_cast<ALsizei>(ids.size()), ids.data());
return;
}
ids.emplace_back(slot->id);
} while(--count);
std::copy(ids.cbegin(), ids.cend(), effectslots);
}
AddActiveEffectSlots(effectslots, static_cast<ALuint>(n), context.get());
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alDeleteAuxiliaryEffectSlots(ALsizei n, const ALuint *effectslots)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
if UNLIKELY(n < 0)
context->setError(AL_INVALID_VALUE, "Deleting %d effect slots", n);
if UNLIKELY(n <= 0) return;
std::lock_guard<std::mutex> _{context->mEffectSlotLock};
auto validate_slot = [&context](const ALuint id) -> bool
{
ALeffectslot *slot{LookupEffectSlot(context.get(), id)};
if UNLIKELY(!slot)
{
context->setError(AL_INVALID_NAME, "Invalid effect slot ID %u", id);
return false;
}
if UNLIKELY(ReadRef(slot->ref) != 0)
{
context->setError(AL_INVALID_OPERATION, "Deleting in-use effect slot %u", id);
return false;
}
return true;
};
auto effectslots_end = effectslots + n;
auto bad_slot = std::find_if_not(effectslots, effectslots_end, validate_slot);
if UNLIKELY(bad_slot != effectslots_end) return;
// All effectslots are valid, remove and delete them
RemoveActiveEffectSlots(effectslots, static_cast<ALuint>(n), context.get());
auto delete_slot = [&context](const ALuint sid) -> void
{
ALeffectslot *slot{LookupEffectSlot(context.get(), sid)};
if(slot) FreeEffectSlot(context.get(), slot);
};
std::for_each(effectslots, effectslots_end, delete_slot);
}
END_API_FUNC
AL_API ALboolean AL_APIENTRY alIsAuxiliaryEffectSlot(ALuint effectslot)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if LIKELY(context)
{
std::lock_guard<std::mutex> _{context->mEffectSlotLock};
if(LookupEffectSlot(context.get(), effectslot) != nullptr)
return AL_TRUE;
}
return AL_FALSE;
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSloti(ALuint effectslot, ALenum param, ALint value)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
std::lock_guard<std::mutex> _{context->mPropLock};
std::lock_guard<std::mutex> __{context->mEffectSlotLock};
ALeffectslot *slot = LookupEffectSlot(context.get(), effectslot);
if UNLIKELY(!slot)
SETERR_RETURN(context, AL_INVALID_NAME,, "Invalid effect slot ID %u", effectslot);
ALeffectslot *target{};
ALCdevice *device{};
ALenum err{};
switch(param)
{
case AL_EFFECTSLOT_EFFECT:
device = context->mDevice.get();
{ std::lock_guard<std::mutex> ___{device->EffectLock};
ALeffect *effect{value ? LookupEffect(device, static_cast<ALuint>(value)) : nullptr};
if(!(value == 0 || effect != nullptr))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Invalid effect ID %u", value);
err = InitializeEffect(context.get(), slot, effect);
}
if(err != AL_NO_ERROR)
{
context->setError(err, "Effect initialization failed");
return;
}
break;
case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO:
if(!(value == AL_TRUE || value == AL_FALSE))
SETERR_RETURN(context, AL_INVALID_VALUE,,
"Effect slot auxiliary send auto out of range");
slot->AuxSendAuto = static_cast<ALboolean>(value);
break;
case AL_EFFECTSLOT_TARGET_SOFT:
target = LookupEffectSlot(context.get(), static_cast<ALuint>(value));
if(value && !target)
SETERR_RETURN(context, AL_INVALID_VALUE,, "Invalid effect slot target ID");
if(target)
{
ALeffectslot *checker{target};
while(checker && checker != slot)
checker = checker->Target;
if(checker)
SETERR_RETURN(context, AL_INVALID_OPERATION,,
"Setting target of effect slot ID %u to %u creates circular chain", slot->id,
target->id);
}
if(ALeffectslot *oldtarget{slot->Target})
{
/* We must force an update if there was an existing effect slot
* target, in case it's about to be deleted.
*/
if(target) IncrementRef(target->ref);
DecrementRef(oldtarget->ref);
slot->Target = target;
UpdateEffectSlotProps(slot, context.get());
return;
}
if(target) IncrementRef(target->ref);
slot->Target = target;
break;
default:
SETERR_RETURN(context, AL_INVALID_ENUM,, "Invalid effect slot integer property 0x%04x",
param);
}
DO_UPDATEPROPS();
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotiv(ALuint effectslot, ALenum param, const ALint *values)
START_API_FUNC
{
switch(param)
{
case AL_EFFECTSLOT_EFFECT:
case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO:
case AL_EFFECTSLOT_TARGET_SOFT:
alAuxiliaryEffectSloti(effectslot, param, values[0]);
return;
}
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
std::lock_guard<std::mutex> _{context->mEffectSlotLock};
ALeffectslot *slot = LookupEffectSlot(context.get(), effectslot);
if UNLIKELY(!slot)
SETERR_RETURN(context, AL_INVALID_NAME,, "Invalid effect slot ID %u", effectslot);
switch(param)
{
default:
SETERR_RETURN(context, AL_INVALID_ENUM,,
"Invalid effect slot integer-vector property 0x%04x", param);
}
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotf(ALuint effectslot, ALenum param, ALfloat value)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
std::lock_guard<std::mutex> _{context->mPropLock};
std::lock_guard<std::mutex> __{context->mEffectSlotLock};
ALeffectslot *slot = LookupEffectSlot(context.get(), effectslot);
if UNLIKELY(!slot)
SETERR_RETURN(context, AL_INVALID_NAME,, "Invalid effect slot ID %u", effectslot);
switch(param)
{
case AL_EFFECTSLOT_GAIN:
if(!(value >= 0.0f && value <= 1.0f))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Effect slot gain out of range");
slot->Gain = value;
break;
default:
SETERR_RETURN(context, AL_INVALID_ENUM,, "Invalid effect slot float property 0x%04x",
param);
}
DO_UPDATEPROPS();
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alAuxiliaryEffectSlotfv(ALuint effectslot, ALenum param, const ALfloat *values)
START_API_FUNC
{
switch(param)
{
case AL_EFFECTSLOT_GAIN:
alAuxiliaryEffectSlotf(effectslot, param, values[0]);
return;
}
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
std::lock_guard<std::mutex> _{context->mEffectSlotLock};
ALeffectslot *slot = LookupEffectSlot(context.get(), effectslot);
if UNLIKELY(!slot)
SETERR_RETURN(context, AL_INVALID_NAME,, "Invalid effect slot ID %u", effectslot);
switch(param)
{
default:
SETERR_RETURN(context, AL_INVALID_ENUM,,
"Invalid effect slot float-vector property 0x%04x", param);
}
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSloti(ALuint effectslot, ALenum param, ALint *value)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
std::lock_guard<std::mutex> _{context->mEffectSlotLock};
ALeffectslot *slot = LookupEffectSlot(context.get(), effectslot);
if UNLIKELY(!slot)
SETERR_RETURN(context, AL_INVALID_NAME,, "Invalid effect slot ID %u", effectslot);
switch(param)
{
case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO:
*value = slot->AuxSendAuto;
break;
case AL_EFFECTSLOT_TARGET_SOFT:
if(auto *target = slot->Target)
*value = static_cast<ALint>(target->id);
else
*value = 0;
break;
default:
context->setError(AL_INVALID_ENUM, "Invalid effect slot integer property 0x%04x", param);
}
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotiv(ALuint effectslot, ALenum param, ALint *values)
START_API_FUNC
{
switch(param)
{
case AL_EFFECTSLOT_EFFECT:
case AL_EFFECTSLOT_AUXILIARY_SEND_AUTO:
case AL_EFFECTSLOT_TARGET_SOFT:
alGetAuxiliaryEffectSloti(effectslot, param, values);
return;
}
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
std::lock_guard<std::mutex> _{context->mEffectSlotLock};
ALeffectslot *slot = LookupEffectSlot(context.get(), effectslot);
if UNLIKELY(!slot)
SETERR_RETURN(context, AL_INVALID_NAME,, "Invalid effect slot ID %u", effectslot);
switch(param)
{
default:
context->setError(AL_INVALID_ENUM, "Invalid effect slot integer-vector property 0x%04x",
param);
}
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotf(ALuint effectslot, ALenum param, ALfloat *value)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
std::lock_guard<std::mutex> _{context->mEffectSlotLock};
ALeffectslot *slot = LookupEffectSlot(context.get(), effectslot);
if UNLIKELY(!slot)
SETERR_RETURN(context, AL_INVALID_NAME,, "Invalid effect slot ID %u", effectslot);
switch(param)
{
case AL_EFFECTSLOT_GAIN:
*value = slot->Gain;
break;
default:
context->setError(AL_INVALID_ENUM, "Invalid effect slot float property 0x%04x", param);
}
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alGetAuxiliaryEffectSlotfv(ALuint effectslot, ALenum param, ALfloat *values)
START_API_FUNC
{
switch(param)
{
case AL_EFFECTSLOT_GAIN:
alGetAuxiliaryEffectSlotf(effectslot, param, values);
return;
}
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
std::lock_guard<std::mutex> _{context->mEffectSlotLock};
ALeffectslot *slot = LookupEffectSlot(context.get(), effectslot);
if UNLIKELY(!slot)
SETERR_RETURN(context, AL_INVALID_NAME,, "Invalid effect slot ID %u", effectslot);
switch(param)
{
default:
context->setError(AL_INVALID_ENUM, "Invalid effect slot float-vector property 0x%04x",
param);
}
}
END_API_FUNC
ALenum InitializeEffect(ALCcontext *Context, ALeffectslot *EffectSlot, ALeffect *effect)
{
ALenum newtype{effect ? effect->type : AL_EFFECT_NULL};
if(newtype != EffectSlot->Effect.Type)
{
EffectStateFactory *factory{getFactoryByType(newtype)};
if(!factory)
{
ERR("Failed to find factory for effect type 0x%04x\n", newtype);
return AL_INVALID_ENUM;
}
EffectState *State{factory->create()};
if(!State) return AL_OUT_OF_MEMORY;
FPUCtl mixer_mode{};
ALCdevice *Device{Context->mDevice.get()};
std::unique_lock<std::mutex> statelock{Device->StateLock};
State->mOutTarget = Device->Dry.Buffer;
if(State->deviceUpdate(Device) == AL_FALSE)
{
statelock.unlock();
mixer_mode.leave();
State->release();
return AL_OUT_OF_MEMORY;
}
mixer_mode.leave();
if(!effect)
{
EffectSlot->Effect.Type = AL_EFFECT_NULL;
EffectSlot->Effect.Props = EffectProps {};
}
else
{
EffectSlot->Effect.Type = effect->type;
EffectSlot->Effect.Props = effect->Props;
}
EffectSlot->Effect.State->release();
EffectSlot->Effect.State = State;
}
else if(effect)
EffectSlot->Effect.Props = effect->Props;
/* Remove state references from old effect slot property updates. */
ALeffectslotProps *props{Context->mFreeEffectslotProps.load()};
while(props)
{
if(props->State)
props->State->release();
props->State = nullptr;
props = props->next.load(std::memory_order_relaxed);
}
return AL_NO_ERROR;
}
ALenum InitEffectSlot(ALeffectslot *slot)
{
EffectStateFactory *factory{getFactoryByType(slot->Effect.Type)};
if(!factory) return AL_INVALID_VALUE;
slot->Effect.State = factory->create();
if(!slot->Effect.State) return AL_OUT_OF_MEMORY;
slot->Effect.State->add_ref();
slot->Params.mEffectState = slot->Effect.State;
return AL_NO_ERROR;
}
ALeffectslot::~ALeffectslot()
{
if(Target)
DecrementRef(Target->ref);
Target = nullptr;
ALeffectslotProps *props{Params.Update.load()};
if(props)
{
if(props->State) props->State->release();
TRACE("Freed unapplied AuxiliaryEffectSlot update %p\n",
decltype(std::declval<void*>()){props});
delete props;
}
if(Effect.State)
Effect.State->release();
if(Params.mEffectState)
Params.mEffectState->release();
}
void UpdateEffectSlotProps(ALeffectslot *slot, ALCcontext *context)
{
/* Get an unused property container, or allocate a new one as needed. */
ALeffectslotProps *props{context->mFreeEffectslotProps.load(std::memory_order_relaxed)};
if(!props)
props = new ALeffectslotProps{};
else
{
ALeffectslotProps *next;
do {
next = props->next.load(std::memory_order_relaxed);
} while(context->mFreeEffectslotProps.compare_exchange_weak(props, next,
std::memory_order_seq_cst, std::memory_order_acquire) == 0);
}
/* Copy in current property values. */
props->Gain = slot->Gain;
props->AuxSendAuto = slot->AuxSendAuto;
props->Target = slot->Target;
props->Type = slot->Effect.Type;
props->Props = slot->Effect.Props;
/* Swap out any stale effect state object there may be in the container, to
* delete it.
*/
EffectState *oldstate{props->State};
slot->Effect.State->add_ref();
props->State = slot->Effect.State;
/* Set the new container for updating internal parameters. */
props = slot->Params.Update.exchange(props, std::memory_order_acq_rel);
if(props)
{
/* If there was an unused update container, put it back in the
* freelist.
*/
if(props->State)
props->State->release();
props->State = nullptr;
AtomicReplaceHead(context->mFreeEffectslotProps, props);
}
if(oldstate)
oldstate->release();
}
void UpdateAllEffectSlotProps(ALCcontext *context)
{
std::lock_guard<std::mutex> _{context->mEffectSlotLock};
ALeffectslotArray *auxslots{context->mActiveAuxSlots.load(std::memory_order_acquire)};
for(ALeffectslot *slot : *auxslots)
{
if(!slot->PropsClean.test_and_set(std::memory_order_acq_rel))
UpdateEffectSlotProps(slot, context);
}
}
EffectSlotSubList::~EffectSlotSubList()
{
uint64_t usemask{~FreeMask};
while(usemask)
{
ALsizei idx{CTZ64(usemask)};
al::destroy_at(EffectSlots+idx);
usemask &= ~(1_u64 << idx);
}
FreeMask = ~usemask;
al_free(EffectSlots);
EffectSlots = nullptr;
}
+105
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#ifndef AL_AUXEFFECTSLOT_H
#define AL_AUXEFFECTSLOT_H
#include <atomic>
#include <cstddef>
#include "AL/al.h"
#include "AL/alc.h"
#include "AL/efx.h"
#include "alcmain.h"
#include "almalloc.h"
#include "atomic.h"
#include "effects/base.h"
#include "vector.h"
struct ALeffect;
struct ALeffectslot;
using ALeffectslotArray = al::FlexArray<ALeffectslot*>;
struct ALeffectslotProps {
ALfloat Gain;
ALboolean AuxSendAuto;
ALeffectslot *Target;
ALenum Type;
EffectProps Props;
EffectState *State;
std::atomic<ALeffectslotProps*> next;
DEF_NEWDEL(ALeffectslotProps)
};
struct ALeffectslot {
ALfloat Gain{1.0f};
ALboolean AuxSendAuto{AL_TRUE};
ALeffectslot *Target{nullptr};
struct {
ALenum Type{AL_EFFECT_NULL};
EffectProps Props{};
EffectState *State{nullptr};
} Effect;
std::atomic_flag PropsClean;
RefCount ref{0u};
struct {
std::atomic<ALeffectslotProps*> Update{nullptr};
ALfloat Gain{1.0f};
ALboolean AuxSendAuto{AL_TRUE};
ALeffectslot *Target{nullptr};
ALenum EffectType{AL_EFFECT_NULL};
EffectProps mEffectProps{};
EffectState *mEffectState{nullptr};
ALfloat RoomRolloff{0.0f}; /* Added to the source's room rolloff, not multiplied. */
ALfloat DecayTime{0.0f};
ALfloat DecayLFRatio{0.0f};
ALfloat DecayHFRatio{0.0f};
ALboolean DecayHFLimit{AL_FALSE};
ALfloat AirAbsorptionGainHF{1.0f};
} Params;
/* Self ID */
ALuint id{};
/* Mixing buffer used by the Wet mix. */
al::vector<FloatBufferLine, 16> MixBuffer;
/* Wet buffer configuration is ACN channel order with N3D scaling.
* Consequently, effects that only want to work with mono input can use
* channel 0 by itself. Effects that want multichannel can process the
* ambisonics signal and make a B-Format source pan.
*/
MixParams Wet;
ALeffectslot() { PropsClean.test_and_set(std::memory_order_relaxed); }
ALeffectslot(const ALeffectslot&) = delete;
ALeffectslot& operator=(const ALeffectslot&) = delete;
~ALeffectslot();
static ALeffectslotArray *CreatePtrArray(size_t count) noexcept;
DEF_NEWDEL(ALeffectslot)
};
ALenum InitEffectSlot(ALeffectslot *slot);
void UpdateEffectSlotProps(ALeffectslot *slot, ALCcontext *context);
void UpdateAllEffectSlotProps(ALCcontext *context);
ALenum InitializeEffect(ALCcontext *Context, ALeffectslot *EffectSlot, ALeffect *effect);
#endif
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+99
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#ifndef AL_BUFFER_H
#define AL_BUFFER_H
#include <atomic>
#include "AL/al.h"
#include "albyte.h"
#include "almalloc.h"
#include "atomic.h"
#include "inprogext.h"
#include "vector.h"
/* User formats */
enum UserFmtType : unsigned char {
UserFmtUByte,
UserFmtShort,
UserFmtFloat,
UserFmtDouble,
UserFmtMulaw,
UserFmtAlaw,
UserFmtIMA4,
UserFmtMSADPCM,
};
enum UserFmtChannels : unsigned char {
UserFmtMono,
UserFmtStereo,
UserFmtRear,
UserFmtQuad,
UserFmtX51, /* (WFX order) */
UserFmtX61, /* (WFX order) */
UserFmtX71, /* (WFX order) */
UserFmtBFormat2D, /* WXY */
UserFmtBFormat3D, /* WXYZ */
};
/* Storable formats */
enum FmtType : unsigned char {
FmtUByte = UserFmtUByte,
FmtShort = UserFmtShort,
FmtFloat = UserFmtFloat,
FmtDouble = UserFmtDouble,
FmtMulaw = UserFmtMulaw,
FmtAlaw = UserFmtAlaw,
};
enum FmtChannels : unsigned char {
FmtMono = UserFmtMono,
FmtStereo = UserFmtStereo,
FmtRear = UserFmtRear,
FmtQuad = UserFmtQuad,
FmtX51 = UserFmtX51,
FmtX61 = UserFmtX61,
FmtX71 = UserFmtX71,
FmtBFormat2D = UserFmtBFormat2D,
FmtBFormat3D = UserFmtBFormat3D,
};
#define MAX_INPUT_CHANNELS (8)
ALuint BytesFromFmt(FmtType type);
ALuint ChannelsFromFmt(FmtChannels chans);
inline ALuint FrameSizeFromFmt(FmtChannels chans, FmtType type)
{ return ChannelsFromFmt(chans) * BytesFromFmt(type); }
struct ALbuffer {
al::vector<al::byte,16> mData;
ALuint Frequency{0u};
ALbitfieldSOFT Access{0u};
ALuint SampleLen{0u};
FmtChannels mFmtChannels{};
FmtType mFmtType{};
UserFmtType OriginalType{};
ALuint OriginalSize{0};
ALuint OriginalAlign{0};
ALuint LoopStart{0u};
ALuint LoopEnd{0u};
ALuint UnpackAlign{0};
ALuint PackAlign{0};
ALbitfieldSOFT MappedAccess{0u};
ALsizei MappedOffset{0};
ALsizei MappedSize{0};
/* Number of times buffer was attached to a source (deletion can only occur when 0) */
RefCount ref{0u};
/* Self ID */
ALuint id{0};
};
#endif
+725
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@@ -0,0 +1,725 @@
/**
* 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include "effect.h"
#include <algorithm>
#include <cstdint>
#include <cstring>
#include <iterator>
#include <memory>
#include <mutex>
#include <new>
#include <numeric>
#include <utility>
#include "AL/al.h"
#include "AL/alc.h"
#include "AL/alext.h"
#include "AL/efx-presets.h"
#include "AL/efx.h"
#include "alcmain.h"
#include "alcontext.h"
#include "alexcpt.h"
#include "almalloc.h"
#include "alnumeric.h"
#include "alstring.h"
#include "effects/base.h"
#include "logging.h"
#include "opthelpers.h"
#include "vector.h"
const EffectList gEffectList[15]{
{ "eaxreverb", EAXREVERB_EFFECT, AL_EFFECT_EAXREVERB },
{ "reverb", REVERB_EFFECT, AL_EFFECT_REVERB },
{ "autowah", AUTOWAH_EFFECT, AL_EFFECT_AUTOWAH },
{ "chorus", CHORUS_EFFECT, AL_EFFECT_CHORUS },
{ "compressor", COMPRESSOR_EFFECT, AL_EFFECT_COMPRESSOR },
{ "distortion", DISTORTION_EFFECT, AL_EFFECT_DISTORTION },
{ "echo", ECHO_EFFECT, AL_EFFECT_ECHO },
{ "equalizer", EQUALIZER_EFFECT, AL_EFFECT_EQUALIZER },
{ "flanger", FLANGER_EFFECT, AL_EFFECT_FLANGER },
{ "fshifter", FSHIFTER_EFFECT, AL_EFFECT_FREQUENCY_SHIFTER },
{ "modulator", MODULATOR_EFFECT, AL_EFFECT_RING_MODULATOR },
{ "pshifter", PSHIFTER_EFFECT, AL_EFFECT_PITCH_SHIFTER },
{ "vmorpher", VMORPHER_EFFECT, AL_EFFECT_VOCAL_MORPHER },
{ "dedicated", DEDICATED_EFFECT, AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT },
{ "dedicated", DEDICATED_EFFECT, AL_EFFECT_DEDICATED_DIALOGUE },
};
ALboolean DisabledEffects[MAX_EFFECTS];
namespace {
constexpr struct FactoryItem {
ALenum Type;
EffectStateFactory* (&GetFactory)(void);
} FactoryList[] = {
{ AL_EFFECT_NULL, NullStateFactory_getFactory },
{ AL_EFFECT_EAXREVERB, ReverbStateFactory_getFactory },
{ AL_EFFECT_REVERB, StdReverbStateFactory_getFactory },
{ AL_EFFECT_AUTOWAH, AutowahStateFactory_getFactory },
{ AL_EFFECT_CHORUS, ChorusStateFactory_getFactory },
{ AL_EFFECT_COMPRESSOR, CompressorStateFactory_getFactory },
{ AL_EFFECT_DISTORTION, DistortionStateFactory_getFactory },
{ AL_EFFECT_ECHO, EchoStateFactory_getFactory },
{ AL_EFFECT_EQUALIZER, EqualizerStateFactory_getFactory },
{ AL_EFFECT_FLANGER, FlangerStateFactory_getFactory },
{ AL_EFFECT_FREQUENCY_SHIFTER, FshifterStateFactory_getFactory },
{ AL_EFFECT_RING_MODULATOR, ModulatorStateFactory_getFactory },
{ AL_EFFECT_PITCH_SHIFTER, PshifterStateFactory_getFactory},
{ AL_EFFECT_VOCAL_MORPHER, VmorpherStateFactory_getFactory},
{ AL_EFFECT_DEDICATED_DIALOGUE, DedicatedStateFactory_getFactory },
{ AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT, DedicatedStateFactory_getFactory }
};
template<typename... T>
void ALeffect_setParami(ALeffect *effect, T&& ...args)
{ effect->vtab->setParami(&effect->Props, std::forward<T>(args)...); }
template<typename... T>
void ALeffect_setParamiv(ALeffect *effect, T&& ...args)
{ effect->vtab->setParamiv(&effect->Props, std::forward<T>(args)...); }
template<typename... T>
void ALeffect_setParamf(ALeffect *effect, T&& ...args)
{ effect->vtab->setParamf(&effect->Props, std::forward<T>(args)...); }
template<typename... T>
void ALeffect_setParamfv(ALeffect *effect, T&& ...args)
{ effect->vtab->setParamfv(&effect->Props, std::forward<T>(args)...); }
template<typename... T>
void ALeffect_getParami(const ALeffect *effect, T&& ...args)
{ effect->vtab->getParami(&effect->Props, std::forward<T>(args)...); }
template<typename... T>
void ALeffect_getParamiv(const ALeffect *effect, T&& ...args)
{ effect->vtab->getParamiv(&effect->Props, std::forward<T>(args)...); }
template<typename... T>
void ALeffect_getParamf(const ALeffect *effect, T&& ...args)
{ effect->vtab->getParamf(&effect->Props, std::forward<T>(args)...); }
template<typename... T>
void ALeffect_getParamfv(const ALeffect *effect, T&& ...args)
{ effect->vtab->getParamfv(&effect->Props, std::forward<T>(args)...); }
void InitEffectParams(ALeffect *effect, ALenum type)
{
EffectStateFactory *factory = getFactoryByType(type);
if(factory)
{
effect->Props = factory->getDefaultProps();
effect->vtab = factory->getEffectVtable();
}
else
{
effect->Props = EffectProps{};
effect->vtab = nullptr;
}
effect->type = type;
}
bool EnsureEffects(ALCdevice *device, size_t needed)
{
size_t count{std::accumulate(device->EffectList.cbegin(), device->EffectList.cend(), size_t{0},
[](size_t cur, const EffectSubList &sublist) noexcept -> size_t
{ return cur + static_cast<ALuint>(POPCNT64(sublist.FreeMask)); }
)};
while(needed > count)
{
if UNLIKELY(device->EffectList.size() >= 1<<25)
return false;
device->EffectList.emplace_back();
auto sublist = device->EffectList.end() - 1;
sublist->FreeMask = ~0_u64;
sublist->Effects = static_cast<ALeffect*>(al_calloc(alignof(ALeffect), sizeof(ALeffect)*64));
if UNLIKELY(!sublist->Effects)
{
device->EffectList.pop_back();
return false;
}
count += 64;
}
return true;
}
ALeffect *AllocEffect(ALCdevice *device)
{
auto sublist = std::find_if(device->EffectList.begin(), device->EffectList.end(),
[](const EffectSubList &entry) noexcept -> bool
{ return entry.FreeMask != 0; }
);
auto lidx = static_cast<ALuint>(std::distance(device->EffectList.begin(), sublist));
auto slidx = static_cast<ALuint>(CTZ64(sublist->FreeMask));
ALeffect *effect{::new (sublist->Effects + slidx) ALeffect{}};
InitEffectParams(effect, AL_EFFECT_NULL);
/* Add 1 to avoid effect ID 0. */
effect->id = ((lidx<<6) | slidx) + 1;
sublist->FreeMask &= ~(1_u64 << slidx);
return effect;
}
void FreeEffect(ALCdevice *device, ALeffect *effect)
{
const ALuint id{effect->id - 1};
const size_t lidx{id >> 6};
const ALuint slidx{id & 0x3f};
al::destroy_at(effect);
device->EffectList[lidx].FreeMask |= 1_u64 << slidx;
}
inline ALeffect *LookupEffect(ALCdevice *device, ALuint id)
{
const size_t lidx{(id-1) >> 6};
const ALuint slidx{(id-1) & 0x3f};
if UNLIKELY(lidx >= device->EffectList.size())
return nullptr;
EffectSubList &sublist = device->EffectList[lidx];
if UNLIKELY(sublist.FreeMask & (1_u64 << slidx))
return nullptr;
return sublist.Effects + slidx;
}
} // namespace
AL_API ALvoid AL_APIENTRY alGenEffects(ALsizei n, ALuint *effects)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
if UNLIKELY(n < 0)
context->setError(AL_INVALID_VALUE, "Generating %d effects", n);
if UNLIKELY(n <= 0) return;
ALCdevice *device{context->mDevice.get()};
std::lock_guard<std::mutex> _{device->EffectLock};
if(!EnsureEffects(device, static_cast<ALuint>(n)))
{
context->setError(AL_OUT_OF_MEMORY, "Failed to allocate %d effect%s", n, (n==1)?"":"s");
return;
}
if LIKELY(n == 1)
{
/* Special handling for the easy and normal case. */
ALeffect *effect{AllocEffect(device)};
effects[0] = effect->id;
}
else
{
/* Store the allocated buffer IDs in a separate local list, to avoid
* modifying the user storage in case of failure.
*/
al::vector<ALuint> ids;
ids.reserve(static_cast<ALuint>(n));
do {
ALeffect *effect{AllocEffect(device)};
ids.emplace_back(effect->id);
} while(--n);
std::copy(ids.cbegin(), ids.cend(), effects);
}
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alDeleteEffects(ALsizei n, const ALuint *effects)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
if UNLIKELY(n < 0)
context->setError(AL_INVALID_VALUE, "Deleting %d effects", n);
if UNLIKELY(n <= 0) return;
ALCdevice *device{context->mDevice.get()};
std::lock_guard<std::mutex> _{device->EffectLock};
/* First try to find any effects that are invalid. */
auto validate_effect = [device](const ALuint eid) -> bool
{ return !eid || LookupEffect(device, eid) != nullptr; };
const ALuint *effects_end = effects + n;
auto inveffect = std::find_if_not(effects, effects_end, validate_effect);
if UNLIKELY(inveffect != effects_end)
{
context->setError(AL_INVALID_NAME, "Invalid effect ID %u", *inveffect);
return;
}
/* All good. Delete non-0 effect IDs. */
auto delete_effect = [device](ALuint eid) -> void
{
ALeffect *effect{eid ? LookupEffect(device, eid) : nullptr};
if(effect) FreeEffect(device, effect);
};
std::for_each(effects, effects_end, delete_effect);
}
END_API_FUNC
AL_API ALboolean AL_APIENTRY alIsEffect(ALuint effect)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if LIKELY(context)
{
ALCdevice *device{context->mDevice.get()};
std::lock_guard<std::mutex> _{device->EffectLock};
if(!effect || LookupEffect(device, effect))
return AL_TRUE;
}
return AL_FALSE;
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alEffecti(ALuint effect, ALenum param, ALint value)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
ALCdevice *device{context->mDevice.get()};
std::lock_guard<std::mutex> _{device->EffectLock};
ALeffect *aleffect{LookupEffect(device, effect)};
if UNLIKELY(!aleffect)
context->setError(AL_INVALID_NAME, "Invalid effect ID %u", effect);
else
{
if(param == AL_EFFECT_TYPE)
{
ALboolean isOk{value == AL_EFFECT_NULL};
if(!isOk)
{
for(const EffectList &effectitem : gEffectList)
{
if(value == effectitem.val && !DisabledEffects[effectitem.type])
{
isOk = AL_TRUE;
break;
}
}
}
if(isOk)
InitEffectParams(aleffect, value);
else
context->setError(AL_INVALID_VALUE, "Effect type 0x%04x not supported", value);
}
else
{
/* Call the appropriate handler */
ALeffect_setParami(aleffect, context.get(), param, value);
}
}
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alEffectiv(ALuint effect, ALenum param, const ALint *values)
START_API_FUNC
{
switch(param)
{
case AL_EFFECT_TYPE:
alEffecti(effect, param, values[0]);
return;
}
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
ALCdevice *device{context->mDevice.get()};
std::lock_guard<std::mutex> _{device->EffectLock};
ALeffect *aleffect{LookupEffect(device, effect)};
if UNLIKELY(!aleffect)
context->setError(AL_INVALID_NAME, "Invalid effect ID %u", effect);
else
{
/* Call the appropriate handler */
ALeffect_setParamiv(aleffect, context.get(), param, values);
}
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alEffectf(ALuint effect, ALenum param, ALfloat value)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
ALCdevice *device{context->mDevice.get()};
std::lock_guard<std::mutex> _{device->EffectLock};
ALeffect *aleffect{LookupEffect(device, effect)};
if UNLIKELY(!aleffect)
context->setError(AL_INVALID_NAME, "Invalid effect ID %u", effect);
else
{
/* Call the appropriate handler */
ALeffect_setParamf(aleffect, context.get(), param, value);
}
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alEffectfv(ALuint effect, ALenum param, const ALfloat *values)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
ALCdevice *device{context->mDevice.get()};
std::lock_guard<std::mutex> _{device->EffectLock};
ALeffect *aleffect{LookupEffect(device, effect)};
if UNLIKELY(!aleffect)
context->setError(AL_INVALID_NAME, "Invalid effect ID %u", effect);
else
{
/* Call the appropriate handler */
ALeffect_setParamfv(aleffect, context.get(), param, values);
}
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alGetEffecti(ALuint effect, ALenum param, ALint *value)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
ALCdevice *device{context->mDevice.get()};
std::lock_guard<std::mutex> _{device->EffectLock};
const ALeffect *aleffect{LookupEffect(device, effect)};
if UNLIKELY(!aleffect)
context->setError(AL_INVALID_NAME, "Invalid effect ID %u", effect);
else
{
if(param == AL_EFFECT_TYPE)
*value = aleffect->type;
else
{
/* Call the appropriate handler */
ALeffect_getParami(aleffect, context.get(), param, value);
}
}
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alGetEffectiv(ALuint effect, ALenum param, ALint *values)
START_API_FUNC
{
switch(param)
{
case AL_EFFECT_TYPE:
alGetEffecti(effect, param, values);
return;
}
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
ALCdevice *device{context->mDevice.get()};
std::lock_guard<std::mutex> _{device->EffectLock};
const ALeffect *aleffect{LookupEffect(device, effect)};
if UNLIKELY(!aleffect)
context->setError(AL_INVALID_NAME, "Invalid effect ID %u", effect);
else
{
/* Call the appropriate handler */
ALeffect_getParamiv(aleffect, context.get(), param, values);
}
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alGetEffectf(ALuint effect, ALenum param, ALfloat *value)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
ALCdevice *device{context->mDevice.get()};
std::lock_guard<std::mutex> _{device->EffectLock};
const ALeffect *aleffect{LookupEffect(device, effect)};
if UNLIKELY(!aleffect)
context->setError(AL_INVALID_NAME, "Invalid effect ID %u", effect);
else
{
/* Call the appropriate handler */
ALeffect_getParamf(aleffect, context.get(), param, value);
}
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alGetEffectfv(ALuint effect, ALenum param, ALfloat *values)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
ALCdevice *device{context->mDevice.get()};
std::lock_guard<std::mutex> _{device->EffectLock};
const ALeffect *aleffect{LookupEffect(device, effect)};
if UNLIKELY(!aleffect)
context->setError(AL_INVALID_NAME, "Invalid effect ID %u", effect);
else
{
/* Call the appropriate handler */
ALeffect_getParamfv(aleffect, context.get(), param, values);
}
}
END_API_FUNC
void InitEffect(ALeffect *effect)
{
InitEffectParams(effect, AL_EFFECT_NULL);
}
EffectSubList::~EffectSubList()
{
uint64_t usemask{~FreeMask};
while(usemask)
{
ALsizei idx = CTZ64(usemask);
al::destroy_at(Effects+idx);
usemask &= ~(1_u64 << idx);
}
FreeMask = ~usemask;
al_free(Effects);
Effects = nullptr;
}
EffectStateFactory *getFactoryByType(ALenum type)
{
auto iter = std::find_if(std::begin(FactoryList), std::end(FactoryList),
[type](const FactoryItem &item) noexcept -> bool
{ return item.Type == type; }
);
return (iter != std::end(FactoryList)) ? iter->GetFactory() : nullptr;
}
#define DECL(x) { #x, EFX_REVERB_PRESET_##x }
static const struct {
const char name[32];
EFXEAXREVERBPROPERTIES props;
} reverblist[] = {
DECL(GENERIC),
DECL(PADDEDCELL),
DECL(ROOM),
DECL(BATHROOM),
DECL(LIVINGROOM),
DECL(STONEROOM),
DECL(AUDITORIUM),
DECL(CONCERTHALL),
DECL(CAVE),
DECL(ARENA),
DECL(HANGAR),
DECL(CARPETEDHALLWAY),
DECL(HALLWAY),
DECL(STONECORRIDOR),
DECL(ALLEY),
DECL(FOREST),
DECL(CITY),
DECL(MOUNTAINS),
DECL(QUARRY),
DECL(PLAIN),
DECL(PARKINGLOT),
DECL(SEWERPIPE),
DECL(UNDERWATER),
DECL(DRUGGED),
DECL(DIZZY),
DECL(PSYCHOTIC),
DECL(CASTLE_SMALLROOM),
DECL(CASTLE_SHORTPASSAGE),
DECL(CASTLE_MEDIUMROOM),
DECL(CASTLE_LARGEROOM),
DECL(CASTLE_LONGPASSAGE),
DECL(CASTLE_HALL),
DECL(CASTLE_CUPBOARD),
DECL(CASTLE_COURTYARD),
DECL(CASTLE_ALCOVE),
DECL(FACTORY_SMALLROOM),
DECL(FACTORY_SHORTPASSAGE),
DECL(FACTORY_MEDIUMROOM),
DECL(FACTORY_LARGEROOM),
DECL(FACTORY_LONGPASSAGE),
DECL(FACTORY_HALL),
DECL(FACTORY_CUPBOARD),
DECL(FACTORY_COURTYARD),
DECL(FACTORY_ALCOVE),
DECL(ICEPALACE_SMALLROOM),
DECL(ICEPALACE_SHORTPASSAGE),
DECL(ICEPALACE_MEDIUMROOM),
DECL(ICEPALACE_LARGEROOM),
DECL(ICEPALACE_LONGPASSAGE),
DECL(ICEPALACE_HALL),
DECL(ICEPALACE_CUPBOARD),
DECL(ICEPALACE_COURTYARD),
DECL(ICEPALACE_ALCOVE),
DECL(SPACESTATION_SMALLROOM),
DECL(SPACESTATION_SHORTPASSAGE),
DECL(SPACESTATION_MEDIUMROOM),
DECL(SPACESTATION_LARGEROOM),
DECL(SPACESTATION_LONGPASSAGE),
DECL(SPACESTATION_HALL),
DECL(SPACESTATION_CUPBOARD),
DECL(SPACESTATION_ALCOVE),
DECL(WOODEN_SMALLROOM),
DECL(WOODEN_SHORTPASSAGE),
DECL(WOODEN_MEDIUMROOM),
DECL(WOODEN_LARGEROOM),
DECL(WOODEN_LONGPASSAGE),
DECL(WOODEN_HALL),
DECL(WOODEN_CUPBOARD),
DECL(WOODEN_COURTYARD),
DECL(WOODEN_ALCOVE),
DECL(SPORT_EMPTYSTADIUM),
DECL(SPORT_SQUASHCOURT),
DECL(SPORT_SMALLSWIMMINGPOOL),
DECL(SPORT_LARGESWIMMINGPOOL),
DECL(SPORT_GYMNASIUM),
DECL(SPORT_FULLSTADIUM),
DECL(SPORT_STADIUMTANNOY),
DECL(PREFAB_WORKSHOP),
DECL(PREFAB_SCHOOLROOM),
DECL(PREFAB_PRACTISEROOM),
DECL(PREFAB_OUTHOUSE),
DECL(PREFAB_CARAVAN),
DECL(DOME_TOMB),
DECL(PIPE_SMALL),
DECL(DOME_SAINTPAULS),
DECL(PIPE_LONGTHIN),
DECL(PIPE_LARGE),
DECL(PIPE_RESONANT),
DECL(OUTDOORS_BACKYARD),
DECL(OUTDOORS_ROLLINGPLAINS),
DECL(OUTDOORS_DEEPCANYON),
DECL(OUTDOORS_CREEK),
DECL(OUTDOORS_VALLEY),
DECL(MOOD_HEAVEN),
DECL(MOOD_HELL),
DECL(MOOD_MEMORY),
DECL(DRIVING_COMMENTATOR),
DECL(DRIVING_PITGARAGE),
DECL(DRIVING_INCAR_RACER),
DECL(DRIVING_INCAR_SPORTS),
DECL(DRIVING_INCAR_LUXURY),
DECL(DRIVING_FULLGRANDSTAND),
DECL(DRIVING_EMPTYGRANDSTAND),
DECL(DRIVING_TUNNEL),
DECL(CITY_STREETS),
DECL(CITY_SUBWAY),
DECL(CITY_MUSEUM),
DECL(CITY_LIBRARY),
DECL(CITY_UNDERPASS),
DECL(CITY_ABANDONED),
DECL(DUSTYROOM),
DECL(CHAPEL),
DECL(SMALLWATERROOM),
};
#undef DECL
void LoadReverbPreset(const char *name, ALeffect *effect)
{
if(al::strcasecmp(name, "NONE") == 0)
{
InitEffectParams(effect, AL_EFFECT_NULL);
TRACE("Loading reverb '%s'\n", "NONE");
return;
}
if(!DisabledEffects[EAXREVERB_EFFECT])
InitEffectParams(effect, AL_EFFECT_EAXREVERB);
else if(!DisabledEffects[REVERB_EFFECT])
InitEffectParams(effect, AL_EFFECT_REVERB);
else
InitEffectParams(effect, AL_EFFECT_NULL);
for(const auto &reverbitem : reverblist)
{
const EFXEAXREVERBPROPERTIES *props;
if(al::strcasecmp(name, reverbitem.name) != 0)
continue;
TRACE("Loading reverb '%s'\n", reverbitem.name);
props = &reverbitem.props;
effect->Props.Reverb.Density = props->flDensity;
effect->Props.Reverb.Diffusion = props->flDiffusion;
effect->Props.Reverb.Gain = props->flGain;
effect->Props.Reverb.GainHF = props->flGainHF;
effect->Props.Reverb.GainLF = props->flGainLF;
effect->Props.Reverb.DecayTime = props->flDecayTime;
effect->Props.Reverb.DecayHFRatio = props->flDecayHFRatio;
effect->Props.Reverb.DecayLFRatio = props->flDecayLFRatio;
effect->Props.Reverb.ReflectionsGain = props->flReflectionsGain;
effect->Props.Reverb.ReflectionsDelay = props->flReflectionsDelay;
effect->Props.Reverb.ReflectionsPan[0] = props->flReflectionsPan[0];
effect->Props.Reverb.ReflectionsPan[1] = props->flReflectionsPan[1];
effect->Props.Reverb.ReflectionsPan[2] = props->flReflectionsPan[2];
effect->Props.Reverb.LateReverbGain = props->flLateReverbGain;
effect->Props.Reverb.LateReverbDelay = props->flLateReverbDelay;
effect->Props.Reverb.LateReverbPan[0] = props->flLateReverbPan[0];
effect->Props.Reverb.LateReverbPan[1] = props->flLateReverbPan[1];
effect->Props.Reverb.LateReverbPan[2] = props->flLateReverbPan[2];
effect->Props.Reverb.EchoTime = props->flEchoTime;
effect->Props.Reverb.EchoDepth = props->flEchoDepth;
effect->Props.Reverb.ModulationTime = props->flModulationTime;
effect->Props.Reverb.ModulationDepth = props->flModulationDepth;
effect->Props.Reverb.AirAbsorptionGainHF = props->flAirAbsorptionGainHF;
effect->Props.Reverb.HFReference = props->flHFReference;
effect->Props.Reverb.LFReference = props->flLFReference;
effect->Props.Reverb.RoomRolloffFactor = props->flRoomRolloffFactor;
effect->Props.Reverb.DecayHFLimit = props->iDecayHFLimit ? AL_TRUE : AL_FALSE;
return;
}
WARN("Reverb preset '%s' not found\n", name);
}
+61
View File
@@ -0,0 +1,61 @@
#ifndef AL_EFFECT_H
#define AL_EFFECT_H
#include "AL/al.h"
#include "AL/efx.h"
#include "effects/base.h"
enum {
EAXREVERB_EFFECT = 0,
REVERB_EFFECT,
AUTOWAH_EFFECT,
CHORUS_EFFECT,
COMPRESSOR_EFFECT,
DISTORTION_EFFECT,
ECHO_EFFECT,
EQUALIZER_EFFECT,
FLANGER_EFFECT,
FSHIFTER_EFFECT,
MODULATOR_EFFECT,
PSHIFTER_EFFECT,
VMORPHER_EFFECT,
DEDICATED_EFFECT,
MAX_EFFECTS
};
extern ALboolean DisabledEffects[MAX_EFFECTS];
extern ALfloat ReverbBoost;
struct EffectList {
const char name[16];
int type;
ALenum val;
};
extern const EffectList gEffectList[15];
struct ALeffect {
// Effect type (AL_EFFECT_NULL, ...)
ALenum type{AL_EFFECT_NULL};
EffectProps Props{};
const EffectVtable *vtab{nullptr};
/* Self ID */
ALuint id{0u};
};
inline ALboolean IsReverbEffect(ALenum type)
{ return type == AL_EFFECT_REVERB || type == AL_EFFECT_EAXREVERB; }
EffectStateFactory *getFactoryByType(ALenum type);
void InitEffect(ALeffect *effect);
void LoadReverbPreset(const char *name, ALeffect *effect);
#endif
+49 -45
View File
@@ -20,46 +20,55 @@
#include "config.h"
#include <signal.h>
#include <stdarg.h>
#ifdef HAVE_WINDOWS_H
#ifdef _WIN32
#define WIN32_LEAN_AND_MEAN
#include <windows.h>
#endif
#include "alMain.h"
#include <atomic>
#include <csignal>
#include <cstdarg>
#include <cstdio>
#include <cstring>
#include <mutex>
#include "AL/al.h"
#include "AL/alc.h"
#include "alError.h"
ALboolean TrapALError = AL_FALSE;
#include "alcontext.h"
#include "alexcpt.h"
#include "almalloc.h"
#include "event.h"
#include "inprogext.h"
#include "logging.h"
#include "opthelpers.h"
#include "vector.h"
void alSetError(ALCcontext *context, ALenum errorCode, const char *msg, ...)
bool TrapALError{false};
void ALCcontext::setError(ALenum errorCode, const char *msg, ...)
{
ALenum curerr = AL_NO_ERROR;
char message[1024] = { 0 };
va_list args;
int msglen;
auto message = al::vector<char>(256);
va_list args, args2;
va_start(args, msg);
msglen = vsnprintf(message, sizeof(message), msg, args);
va_copy(args2, args);
int msglen{std::vsnprintf(message.data(), message.size(), msg, args)};
if(msglen >= 0 && static_cast<size_t>(msglen) >= message.size())
{
message.resize(static_cast<size_t>(msglen) + 1u);
msglen = std::vsnprintf(message.data(), message.size(), msg, args2);
}
va_end(args2);
va_end(args);
if(msglen < 0 || (size_t)msglen >= sizeof(message))
{
message[sizeof(message)-1] = 0;
msglen = (int)strlen(message);
}
if(msglen > 0)
msg = message;
else
{
msg = "<internal error constructing message>";
msglen = (int)strlen(msg);
}
if(msglen >= 0) msg = message.data();
else msg = "<internal error constructing message>";
msglen = static_cast<int>(strlen(msg));
WARN("Error generated on context %p, code 0x%04x, \"%s\"\n",
context, errorCode, message);
decltype(std::declval<void*>()){this}, errorCode, msg);
if(TrapALError)
{
#ifdef _WIN32
@@ -71,28 +80,25 @@ void alSetError(ALCcontext *context, ALenum errorCode, const char *msg, ...)
#endif
}
ATOMIC_COMPARE_EXCHANGE_STRONG_SEQ(&context->LastError, &curerr, errorCode);
if((ATOMIC_LOAD(&context->EnabledEvts, almemory_order_relaxed)&EventType_Error))
ALenum curerr{AL_NO_ERROR};
mLastError.compare_exchange_strong(curerr, errorCode);
if((mEnabledEvts.load(std::memory_order_relaxed)&EventType_Error))
{
ALbitfieldSOFT enabledevts;
almtx_lock(&context->EventCbLock);
enabledevts = ATOMIC_LOAD(&context->EnabledEvts, almemory_order_relaxed);
if((enabledevts&EventType_Error) && context->EventCb)
(*context->EventCb)(AL_EVENT_TYPE_ERROR_SOFT, 0, errorCode, msglen, msg,
context->EventParam);
almtx_unlock(&context->EventCbLock);
std::lock_guard<std::mutex> _{mEventCbLock};
ALbitfieldSOFT enabledevts{mEnabledEvts.load(std::memory_order_relaxed)};
if((enabledevts&EventType_Error) && mEventCb)
(*mEventCb)(AL_EVENT_TYPE_ERROR_SOFT, 0, static_cast<ALuint>(errorCode), msglen, msg,
mEventParam);
}
}
AL_API ALenum AL_APIENTRY alGetError(void)
START_API_FUNC
{
ALCcontext *context;
ALenum errorCode;
context = GetContextRef();
if(!context)
ContextRef context{GetContextRef()};
if UNLIKELY(!context)
{
const ALenum deferror = AL_INVALID_OPERATION;
constexpr ALenum deferror{AL_INVALID_OPERATION};
WARN("Querying error state on null context (implicitly 0x%04x)\n", deferror);
if(TrapALError)
{
@@ -106,8 +112,6 @@ AL_API ALenum AL_APIENTRY alGetError(void)
return deferror;
}
errorCode = ATOMIC_EXCHANGE_SEQ(&context->LastError, AL_NO_ERROR);
ALCcontext_DecRef(context);
return errorCode;
return context->mLastError.exchange(AL_NO_ERROR);
}
END_API_FUNC
+205
View File
@@ -0,0 +1,205 @@
#include "config.h"
#include "event.h"
#include <algorithm>
#include <atomic>
#include <cstring>
#include <exception>
#include <memory>
#include <mutex>
#include <new>
#include <string>
#include <thread>
#include <utility>
#include "AL/al.h"
#include "AL/alc.h"
#include "albyte.h"
#include "alcontext.h"
#include "alexcpt.h"
#include "almalloc.h"
#include "effects/base.h"
#include "inprogext.h"
#include "logging.h"
#include "opthelpers.h"
#include "ringbuffer.h"
#include "threads.h"
static int EventThread(ALCcontext *context)
{
RingBuffer *ring{context->mAsyncEvents.get()};
bool quitnow{false};
while LIKELY(!quitnow)
{
auto evt_data = ring->getReadVector().first;
if(evt_data.len == 0)
{
context->mEventSem.wait();
continue;
}
std::lock_guard<std::mutex> _{context->mEventCbLock};
do {
auto *evt_ptr = reinterpret_cast<AsyncEvent*>(evt_data.buf);
evt_data.buf += sizeof(AsyncEvent);
evt_data.len -= 1;
AsyncEvent evt{*evt_ptr};
al::destroy_at(evt_ptr);
ring->readAdvance(1);
quitnow = evt.EnumType == EventType_KillThread;
if UNLIKELY(quitnow) break;
if(evt.EnumType == EventType_ReleaseEffectState)
{
evt.u.mEffectState->release();
continue;
}
ALbitfieldSOFT enabledevts{context->mEnabledEvts.load(std::memory_order_acquire)};
if(!context->mEventCb) continue;
if(evt.EnumType == EventType_SourceStateChange)
{
if(!(enabledevts&EventType_SourceStateChange))
continue;
std::string msg{"Source ID " + std::to_string(evt.u.srcstate.id)};
msg += " state has changed to ";
msg += (evt.u.srcstate.state==AL_INITIAL) ? "AL_INITIAL" :
(evt.u.srcstate.state==AL_PLAYING) ? "AL_PLAYING" :
(evt.u.srcstate.state==AL_PAUSED) ? "AL_PAUSED" :
(evt.u.srcstate.state==AL_STOPPED) ? "AL_STOPPED" : "<unknown>";
context->mEventCb(AL_EVENT_TYPE_SOURCE_STATE_CHANGED_SOFT, evt.u.srcstate.id,
static_cast<ALuint>(evt.u.srcstate.state), static_cast<ALsizei>(msg.length()),
msg.c_str(), context->mEventParam);
}
else if(evt.EnumType == EventType_BufferCompleted)
{
if(!(enabledevts&EventType_BufferCompleted))
continue;
std::string msg{std::to_string(evt.u.bufcomp.count)};
if(evt.u.bufcomp.count == 1) msg += " buffer completed";
else msg += " buffers completed";
context->mEventCb(AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT, evt.u.bufcomp.id,
evt.u.bufcomp.count, static_cast<ALsizei>(msg.length()), msg.c_str(),
context->mEventParam);
}
else if((enabledevts&evt.EnumType) == evt.EnumType)
context->mEventCb(evt.u.user.type, evt.u.user.id, evt.u.user.param,
static_cast<ALsizei>(strlen(evt.u.user.msg)), evt.u.user.msg,
context->mEventParam);
} while(evt_data.len != 0);
}
return 0;
}
void StartEventThrd(ALCcontext *ctx)
{
try {
ctx->mEventThread = std::thread{EventThread, ctx};
}
catch(std::exception& e) {
ERR("Failed to start event thread: %s\n", e.what());
}
catch(...) {
ERR("Failed to start event thread! Expect problems.\n");
}
}
void StopEventThrd(ALCcontext *ctx)
{
RingBuffer *ring{ctx->mAsyncEvents.get()};
auto evt_data = ring->getWriteVector().first;
if(evt_data.len == 0)
{
do {
std::this_thread::yield();
evt_data = ring->getWriteVector().first;
} while(evt_data.len == 0);
}
new (evt_data.buf) AsyncEvent{EventType_KillThread};
ring->writeAdvance(1);
ctx->mEventSem.post();
if(ctx->mEventThread.joinable())
ctx->mEventThread.join();
}
AL_API void AL_APIENTRY alEventControlSOFT(ALsizei count, const ALenum *types, ALboolean enable)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
if(count < 0) context->setError(AL_INVALID_VALUE, "Controlling %d events", count);
if(count <= 0) return;
if(!types) SETERR_RETURN(context, AL_INVALID_VALUE,, "NULL pointer");
ALbitfieldSOFT flags{0};
const ALenum *types_end = types+count;
auto bad_type = std::find_if_not(types, types_end,
[&flags](ALenum type) noexcept -> bool
{
if(type == AL_EVENT_TYPE_BUFFER_COMPLETED_SOFT)
flags |= EventType_BufferCompleted;
else if(type == AL_EVENT_TYPE_SOURCE_STATE_CHANGED_SOFT)
flags |= EventType_SourceStateChange;
else if(type == AL_EVENT_TYPE_ERROR_SOFT)
flags |= EventType_Error;
else if(type == AL_EVENT_TYPE_PERFORMANCE_SOFT)
flags |= EventType_Performance;
else if(type == AL_EVENT_TYPE_DEPRECATED_SOFT)
flags |= EventType_Deprecated;
else if(type == AL_EVENT_TYPE_DISCONNECTED_SOFT)
flags |= EventType_Disconnected;
else
return false;
return true;
}
);
if(bad_type != types_end)
SETERR_RETURN(context, AL_INVALID_ENUM,, "Invalid event type 0x%04x", *bad_type);
if(enable)
{
ALbitfieldSOFT enabledevts{context->mEnabledEvts.load(std::memory_order_relaxed)};
while(context->mEnabledEvts.compare_exchange_weak(enabledevts, enabledevts|flags,
std::memory_order_acq_rel, std::memory_order_acquire) == 0)
{
/* enabledevts is (re-)filled with the current value on failure, so
* just try again.
*/
}
}
else
{
ALbitfieldSOFT enabledevts{context->mEnabledEvts.load(std::memory_order_relaxed)};
while(context->mEnabledEvts.compare_exchange_weak(enabledevts, enabledevts&~flags,
std::memory_order_acq_rel, std::memory_order_acquire) == 0)
{
}
/* Wait to ensure the event handler sees the changed flags before
* returning.
*/
std::lock_guard<std::mutex>{context->mEventCbLock};
}
}
END_API_FUNC
AL_API void AL_APIENTRY alEventCallbackSOFT(ALEVENTPROCSOFT callback, void *userParam)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
std::lock_guard<std::mutex> _{context->mPropLock};
std::lock_guard<std::mutex> __{context->mEventCbLock};
context->mEventCb = callback;
context->mEventParam = userParam;
}
END_API_FUNC
+55
View File
@@ -0,0 +1,55 @@
#ifndef AL_EVENT_H
#define AL_EVENT_H
#include "AL/al.h"
#include "AL/alc.h"
struct EffectState;
enum {
/* End event thread processing. */
EventType_KillThread = 0,
/* User event types. */
EventType_SourceStateChange = 1<<0,
EventType_BufferCompleted = 1<<1,
EventType_Error = 1<<2,
EventType_Performance = 1<<3,
EventType_Deprecated = 1<<4,
EventType_Disconnected = 1<<5,
/* Internal events. */
EventType_ReleaseEffectState = 65536,
};
struct AsyncEvent {
unsigned int EnumType{0u};
union {
char dummy;
struct {
ALuint id;
ALenum state;
} srcstate;
struct {
ALuint id;
ALuint count;
} bufcomp;
struct {
ALenum type;
ALuint id;
ALuint param;
ALchar msg[232];
} user;
EffectState *mEffectState;
} u{};
AsyncEvent() noexcept = default;
constexpr AsyncEvent(unsigned int type) noexcept : EnumType{type} { }
};
void StartEventThrd(ALCcontext *ctx);
void StopEventThrd(ALCcontext *ctx);
#endif
+28 -36
View File
@@ -20,45 +20,36 @@
#include "config.h"
#include <stdlib.h>
#include <string.h>
#include <ctype.h>
#include <cctype>
#include <cstdlib>
#include <cstring>
#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"
#include "alcontext.h"
#include "alexcpt.h"
#include "alstring.h"
#include "opthelpers.h"
AL_API ALboolean AL_APIENTRY alIsExtensionPresent(const ALchar *extName)
START_API_FUNC
{
ALboolean ret = AL_FALSE;
ALCcontext *context;
const char *ptr;
size_t len;
context = GetContextRef();
if(!context) return AL_FALSE;
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return AL_FALSE;
if(!extName)
SETERR_GOTO(context, AL_INVALID_VALUE, done, "NULL pointer");
SETERR_RETURN(context, AL_INVALID_VALUE, AL_FALSE, "NULL pointer");
len = strlen(extName);
ptr = context->ExtensionList;
size_t len{strlen(extName)};
const char *ptr{context->mExtensionList};
while(ptr && *ptr)
{
if(strncasecmp(ptr, extName, len) == 0 &&
(ptr[len] == '\0' || isspace(ptr[len])))
{
ret = AL_TRUE;
break;
}
if((ptr=strchr(ptr, ' ')) != NULL)
if(al::strncasecmp(ptr, extName, len) == 0 && (ptr[len] == '\0' || isspace(ptr[len])))
return AL_TRUE;
if((ptr=strchr(ptr, ' ')) != nullptr)
{
do {
++ptr;
@@ -66,22 +57,23 @@ AL_API ALboolean AL_APIENTRY alIsExtensionPresent(const ALchar *extName)
}
}
done:
ALCcontext_DecRef(context);
return ret;
return AL_FALSE;
}
END_API_FUNC
AL_API ALvoid* AL_APIENTRY alGetProcAddress(const ALchar *funcName)
START_API_FUNC
{
if(!funcName)
return NULL;
return alcGetProcAddress(NULL, funcName);
if(!funcName) return nullptr;
return alcGetProcAddress(nullptr, funcName);
}
END_API_FUNC
AL_API ALenum AL_APIENTRY alGetEnumValue(const ALchar *enumName)
START_API_FUNC
{
if(!enumName)
return (ALenum)0;
return alcGetEnumValue(NULL, enumName);
if(!enumName) return static_cast<ALenum>(0);
return alcGetEnumValue(nullptr, enumName);
}
END_API_FUNC
+650
View File
@@ -0,0 +1,650 @@
/**
* 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include "filter.h"
#include <algorithm>
#include <cstdint>
#include <iterator>
#include <memory>
#include <mutex>
#include <new>
#include <numeric>
#include "AL/al.h"
#include "AL/alc.h"
#include "AL/efx.h"
#include "alcmain.h"
#include "alcontext.h"
#include "alexcpt.h"
#include "almalloc.h"
#include "alnumeric.h"
#include "opthelpers.h"
#include "vector.h"
namespace {
#define FILTER_MIN_GAIN 0.0f
#define FILTER_MAX_GAIN 4.0f /* +12dB */
void ALlowpass_setParami(ALfilter*, ALCcontext *context, ALenum param, ALint)
{ context->setError(AL_INVALID_ENUM, "Invalid low-pass integer property 0x%04x", param); }
void ALlowpass_setParamiv(ALfilter*, ALCcontext *context, ALenum param, const ALint*)
{ context->setError(AL_INVALID_ENUM, "Invalid low-pass integer-vector property 0x%04x", param); }
void ALlowpass_setParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val)
{
switch(param)
{
case AL_LOWPASS_GAIN:
if(!(val >= FILTER_MIN_GAIN && val <= FILTER_MAX_GAIN))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Low-pass gain %f out of range", val);
filter->Gain = val;
break;
case AL_LOWPASS_GAINHF:
if(!(val >= AL_LOWPASS_MIN_GAINHF && val <= AL_LOWPASS_MAX_GAINHF))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Low-pass gainhf %f out of range", val);
filter->GainHF = val;
break;
default:
context->setError(AL_INVALID_ENUM, "Invalid low-pass float property 0x%04x", param);
}
}
void ALlowpass_setParamfv(ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals)
{ ALlowpass_setParamf(filter, context, param, vals[0]); }
void ALlowpass_getParami(ALfilter*, ALCcontext *context, ALenum param, ALint*)
{ context->setError(AL_INVALID_ENUM, "Invalid low-pass integer property 0x%04x", param); }
void ALlowpass_getParamiv(ALfilter*, ALCcontext *context, ALenum param, ALint*)
{ context->setError(AL_INVALID_ENUM, "Invalid low-pass integer-vector property 0x%04x", param); }
void ALlowpass_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:
context->setError(AL_INVALID_ENUM, "Invalid low-pass float property 0x%04x", param);
}
}
void ALlowpass_getParamfv(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals)
{ ALlowpass_getParamf(filter, context, param, vals); }
DEFINE_ALFILTER_VTABLE(ALlowpass);
void ALhighpass_setParami(ALfilter*, ALCcontext *context, ALenum param, ALint)
{ context->setError(AL_INVALID_ENUM, "Invalid high-pass integer property 0x%04x", param); }
void ALhighpass_setParamiv(ALfilter*, ALCcontext *context, ALenum param, const ALint*)
{ context->setError(AL_INVALID_ENUM, "Invalid high-pass integer-vector property 0x%04x", param); }
void ALhighpass_setParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val)
{
switch(param)
{
case AL_HIGHPASS_GAIN:
if(!(val >= FILTER_MIN_GAIN && val <= FILTER_MAX_GAIN))
SETERR_RETURN(context, AL_INVALID_VALUE,, "High-pass gain out of range");
filter->Gain = val;
break;
case AL_HIGHPASS_GAINLF:
if(!(val >= AL_HIGHPASS_MIN_GAINLF && val <= AL_HIGHPASS_MAX_GAINLF))
SETERR_RETURN(context, AL_INVALID_VALUE,, "High-pass gainlf out of range");
filter->GainLF = val;
break;
default:
context->setError(AL_INVALID_ENUM, "Invalid high-pass float property 0x%04x", param);
}
}
void ALhighpass_setParamfv(ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals)
{ ALhighpass_setParamf(filter, context, param, vals[0]); }
void ALhighpass_getParami(ALfilter*, ALCcontext *context, ALenum param, ALint*)
{ context->setError(AL_INVALID_ENUM, "Invalid high-pass integer property 0x%04x", param); }
void ALhighpass_getParamiv(ALfilter*, ALCcontext *context, ALenum param, ALint*)
{ context->setError(AL_INVALID_ENUM, "Invalid high-pass integer-vector property 0x%04x", param); }
void ALhighpass_getParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *val)
{
switch(param)
{
case AL_HIGHPASS_GAIN:
*val = filter->Gain;
break;
case AL_HIGHPASS_GAINLF:
*val = filter->GainLF;
break;
default:
context->setError(AL_INVALID_ENUM, "Invalid high-pass float property 0x%04x", param);
}
}
void ALhighpass_getParamfv(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals)
{ ALhighpass_getParamf(filter, context, param, vals); }
DEFINE_ALFILTER_VTABLE(ALhighpass);
void ALbandpass_setParami(ALfilter*, ALCcontext *context, ALenum param, ALint)
{ context->setError(AL_INVALID_ENUM, "Invalid band-pass integer property 0x%04x", param); }
void ALbandpass_setParamiv(ALfilter*, ALCcontext *context, ALenum param, const ALint*)
{ context->setError(AL_INVALID_ENUM, "Invalid band-pass integer-vector property 0x%04x", param); }
void ALbandpass_setParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat val)
{
switch(param)
{
case AL_BANDPASS_GAIN:
if(!(val >= FILTER_MIN_GAIN && val <= FILTER_MAX_GAIN))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Band-pass gain out of range");
filter->Gain = val;
break;
case AL_BANDPASS_GAINHF:
if(!(val >= AL_BANDPASS_MIN_GAINHF && val <= AL_BANDPASS_MAX_GAINHF))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Band-pass gainhf out of range");
filter->GainHF = val;
break;
case AL_BANDPASS_GAINLF:
if(!(val >= AL_BANDPASS_MIN_GAINLF && val <= AL_BANDPASS_MAX_GAINLF))
SETERR_RETURN(context, AL_INVALID_VALUE,, "Band-pass gainlf out of range");
filter->GainLF = val;
break;
default:
context->setError(AL_INVALID_ENUM, "Invalid band-pass float property 0x%04x", param);
}
}
void ALbandpass_setParamfv(ALfilter *filter, ALCcontext *context, ALenum param, const ALfloat *vals)
{ ALbandpass_setParamf(filter, context, param, vals[0]); }
void ALbandpass_getParami(ALfilter*, ALCcontext *context, ALenum param, ALint*)
{ context->setError(AL_INVALID_ENUM, "Invalid band-pass integer property 0x%04x", param); }
void ALbandpass_getParamiv(ALfilter*, ALCcontext *context, ALenum param, ALint*)
{ context->setError(AL_INVALID_ENUM, "Invalid band-pass integer-vector property 0x%04x", param); }
void ALbandpass_getParamf(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *val)
{
switch(param)
{
case AL_BANDPASS_GAIN:
*val = filter->Gain;
break;
case AL_BANDPASS_GAINHF:
*val = filter->GainHF;
break;
case AL_BANDPASS_GAINLF:
*val = filter->GainLF;
break;
default:
context->setError(AL_INVALID_ENUM, "Invalid band-pass float property 0x%04x", param);
}
}
void ALbandpass_getParamfv(ALfilter *filter, ALCcontext *context, ALenum param, ALfloat *vals)
{ ALbandpass_getParamf(filter, context, param, vals); }
DEFINE_ALFILTER_VTABLE(ALbandpass);
void ALnullfilter_setParami(ALfilter*, ALCcontext *context, ALenum param, ALint)
{ context->setError(AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
void ALnullfilter_setParamiv(ALfilter*, ALCcontext *context, ALenum param, const ALint*)
{ context->setError(AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
void ALnullfilter_setParamf(ALfilter*, ALCcontext *context, ALenum param, ALfloat)
{ context->setError(AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
void ALnullfilter_setParamfv(ALfilter*, ALCcontext *context, ALenum param, const ALfloat*)
{ context->setError(AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
void ALnullfilter_getParami(ALfilter*, ALCcontext *context, ALenum param, ALint*)
{ context->setError(AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
void ALnullfilter_getParamiv(ALfilter*, ALCcontext *context, ALenum param, ALint*)
{ context->setError(AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
void ALnullfilter_getParamf(ALfilter*, ALCcontext *context, ALenum param, ALfloat*)
{ context->setError(AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
void ALnullfilter_getParamfv(ALfilter*, ALCcontext *context, ALenum param, ALfloat*)
{ context->setError(AL_INVALID_ENUM, "Invalid null filter property 0x%04x", param); }
DEFINE_ALFILTER_VTABLE(ALnullfilter);
void InitFilterParams(ALfilter *filter, ALenum type)
{
if(type == AL_FILTER_LOWPASS)
{
filter->Gain = AL_LOWPASS_DEFAULT_GAIN;
filter->GainHF = AL_LOWPASS_DEFAULT_GAINHF;
filter->HFReference = LOWPASSFREQREF;
filter->GainLF = 1.0f;
filter->LFReference = HIGHPASSFREQREF;
filter->vtab = &ALlowpass_vtable;
}
else if(type == AL_FILTER_HIGHPASS)
{
filter->Gain = AL_HIGHPASS_DEFAULT_GAIN;
filter->GainHF = 1.0f;
filter->HFReference = LOWPASSFREQREF;
filter->GainLF = AL_HIGHPASS_DEFAULT_GAINLF;
filter->LFReference = HIGHPASSFREQREF;
filter->vtab = &ALhighpass_vtable;
}
else if(type == AL_FILTER_BANDPASS)
{
filter->Gain = AL_BANDPASS_DEFAULT_GAIN;
filter->GainHF = AL_BANDPASS_DEFAULT_GAINHF;
filter->HFReference = LOWPASSFREQREF;
filter->GainLF = AL_BANDPASS_DEFAULT_GAINLF;
filter->LFReference = HIGHPASSFREQREF;
filter->vtab = &ALbandpass_vtable;
}
else
{
filter->Gain = 1.0f;
filter->GainHF = 1.0f;
filter->HFReference = LOWPASSFREQREF;
filter->GainLF = 1.0f;
filter->LFReference = HIGHPASSFREQREF;
filter->vtab = &ALnullfilter_vtable;
}
filter->type = type;
}
bool EnsureFilters(ALCdevice *device, size_t needed)
{
size_t count{std::accumulate(device->FilterList.cbegin(), device->FilterList.cend(), size_t{0},
[](size_t cur, const FilterSubList &sublist) noexcept -> size_t
{ return cur + static_cast<ALuint>(POPCNT64(sublist.FreeMask)); }
)};
while(needed > count)
{
if UNLIKELY(device->FilterList.size() >= 1<<25)
return false;
device->FilterList.emplace_back();
auto sublist = device->FilterList.end() - 1;
sublist->FreeMask = ~0_u64;
sublist->Filters = static_cast<ALfilter*>(al_calloc(alignof(ALfilter), sizeof(ALfilter)*64));
if UNLIKELY(!sublist->Filters)
{
device->FilterList.pop_back();
return false;
}
count += 64;
}
return true;
}
ALfilter *AllocFilter(ALCdevice *device)
{
auto sublist = std::find_if(device->FilterList.begin(), device->FilterList.end(),
[](const FilterSubList &entry) noexcept -> bool
{ return entry.FreeMask != 0; }
);
auto lidx = static_cast<ALuint>(std::distance(device->FilterList.begin(), sublist));
auto slidx = static_cast<ALuint>(CTZ64(sublist->FreeMask));
ALfilter *filter{::new (sublist->Filters + slidx) ALfilter{}};
InitFilterParams(filter, AL_FILTER_NULL);
/* Add 1 to avoid filter ID 0. */
filter->id = ((lidx<<6) | slidx) + 1;
sublist->FreeMask &= ~(1_u64 << slidx);
return filter;
}
void FreeFilter(ALCdevice *device, ALfilter *filter)
{
const ALuint id{filter->id - 1};
const size_t lidx{id >> 6};
const ALuint slidx{id & 0x3f};
al::destroy_at(filter);
device->FilterList[lidx].FreeMask |= 1_u64 << slidx;
}
inline ALfilter *LookupFilter(ALCdevice *device, ALuint id)
{
const size_t lidx{(id-1) >> 6};
const ALuint slidx{(id-1) & 0x3f};
if UNLIKELY(lidx >= device->FilterList.size())
return nullptr;
FilterSubList &sublist = device->FilterList[lidx];
if UNLIKELY(sublist.FreeMask & (1_u64 << slidx))
return nullptr;
return sublist.Filters + slidx;
}
} // namespace
AL_API ALvoid AL_APIENTRY alGenFilters(ALsizei n, ALuint *filters)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
if UNLIKELY(n < 0)
context->setError(AL_INVALID_VALUE, "Generating %d filters", n);
if UNLIKELY(n <= 0) return;
ALCdevice *device{context->mDevice.get()};
std::lock_guard<std::mutex> _{device->EffectLock};
if(!EnsureFilters(device, static_cast<ALuint>(n)))
{
context->setError(AL_OUT_OF_MEMORY, "Failed to allocate %d filter%s", n, (n==1)?"":"s");
return;
}
if LIKELY(n == 1)
{
/* Special handling for the easy and normal case. */
ALfilter *filter{AllocFilter(device)};
if(filter) filters[0] = filter->id;
}
else
{
/* Store the allocated buffer IDs in a separate local list, to avoid
* modifying the user storage in case of failure.
*/
al::vector<ALuint> ids;
ids.reserve(static_cast<ALuint>(n));
do {
ALfilter *filter{AllocFilter(device)};
ids.emplace_back(filter->id);
} while(--n);
std::copy(ids.begin(), ids.end(), filters);
}
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alDeleteFilters(ALsizei n, const ALuint *filters)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
if UNLIKELY(n < 0)
context->setError(AL_INVALID_VALUE, "Deleting %d filters", n);
if UNLIKELY(n <= 0) return;
ALCdevice *device{context->mDevice.get()};
std::lock_guard<std::mutex> _{device->FilterLock};
/* First try to find any filters that are invalid. */
auto validate_filter = [device](const ALuint fid) -> bool
{ return !fid || LookupFilter(device, fid) != nullptr; };
const ALuint *filters_end = filters + n;
auto invflt = std::find_if_not(filters, filters_end, validate_filter);
if UNLIKELY(invflt != filters_end)
{
context->setError(AL_INVALID_NAME, "Invalid filter ID %u", *invflt);
return;
}
/* All good. Delete non-0 filter IDs. */
auto delete_filter = [device](const ALuint fid) -> void
{
ALfilter *filter{fid ? LookupFilter(device, fid) : nullptr};
if(filter) FreeFilter(device, filter);
};
std::for_each(filters, filters_end, delete_filter);
}
END_API_FUNC
AL_API ALboolean AL_APIENTRY alIsFilter(ALuint filter)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if LIKELY(context)
{
ALCdevice *device{context->mDevice.get()};
std::lock_guard<std::mutex> _{device->FilterLock};
if(!filter || LookupFilter(device, filter))
return AL_TRUE;
}
return AL_FALSE;
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alFilteri(ALuint filter, ALenum param, ALint value)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
ALCdevice *device{context->mDevice.get()};
std::lock_guard<std::mutex> _{device->FilterLock};
ALfilter *alfilt{LookupFilter(device, filter)};
if UNLIKELY(!alfilt)
context->setError(AL_INVALID_NAME, "Invalid filter ID %u", filter);
else
{
if(param == AL_FILTER_TYPE)
{
if(value == AL_FILTER_NULL || value == AL_FILTER_LOWPASS ||
value == AL_FILTER_HIGHPASS || value == AL_FILTER_BANDPASS)
InitFilterParams(alfilt, value);
else
context->setError(AL_INVALID_VALUE, "Invalid filter type 0x%04x", value);
}
else
{
/* Call the appropriate handler */
ALfilter_setParami(alfilt, context.get(), param, value);
}
}
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alFilteriv(ALuint filter, ALenum param, const ALint *values)
START_API_FUNC
{
switch(param)
{
case AL_FILTER_TYPE:
alFilteri(filter, param, values[0]);
return;
}
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
ALCdevice *device{context->mDevice.get()};
std::lock_guard<std::mutex> _{device->FilterLock};
ALfilter *alfilt{LookupFilter(device, filter)};
if UNLIKELY(!alfilt)
context->setError(AL_INVALID_NAME, "Invalid filter ID %u", filter);
else
{
/* Call the appropriate handler */
ALfilter_setParamiv(alfilt, context.get(), param, values);
}
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alFilterf(ALuint filter, ALenum param, ALfloat value)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
ALCdevice *device{context->mDevice.get()};
std::lock_guard<std::mutex> _{device->FilterLock};
ALfilter *alfilt{LookupFilter(device, filter)};
if UNLIKELY(!alfilt)
context->setError(AL_INVALID_NAME, "Invalid filter ID %u", filter);
else
{
/* Call the appropriate handler */
ALfilter_setParamf(alfilt, context.get(), param, value);
}
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alFilterfv(ALuint filter, ALenum param, const ALfloat *values)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
ALCdevice *device{context->mDevice.get()};
std::lock_guard<std::mutex> _{device->FilterLock};
ALfilter *alfilt{LookupFilter(device, filter)};
if UNLIKELY(!alfilt)
context->setError(AL_INVALID_NAME, "Invalid filter ID %u", filter);
else
{
/* Call the appropriate handler */
ALfilter_setParamfv(alfilt, context.get(), param, values);
}
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alGetFilteri(ALuint filter, ALenum param, ALint *value)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
ALCdevice *device{context->mDevice.get()};
std::lock_guard<std::mutex> _{device->FilterLock};
ALfilter *alfilt{LookupFilter(device, filter)};
if UNLIKELY(!alfilt)
context->setError(AL_INVALID_NAME, "Invalid filter ID %u", filter);
else
{
if(param == AL_FILTER_TYPE)
*value = alfilt->type;
else
{
/* Call the appropriate handler */
ALfilter_getParami(alfilt, context.get(), param, value);
}
}
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alGetFilteriv(ALuint filter, ALenum param, ALint *values)
START_API_FUNC
{
switch(param)
{
case AL_FILTER_TYPE:
alGetFilteri(filter, param, values);
return;
}
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
ALCdevice *device{context->mDevice.get()};
std::lock_guard<std::mutex> _{device->FilterLock};
ALfilter *alfilt{LookupFilter(device, filter)};
if UNLIKELY(!alfilt)
context->setError(AL_INVALID_NAME, "Invalid filter ID %u", filter);
else
{
/* Call the appropriate handler */
ALfilter_getParamiv(alfilt, context.get(), param, values);
}
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alGetFilterf(ALuint filter, ALenum param, ALfloat *value)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
ALCdevice *device{context->mDevice.get()};
std::lock_guard<std::mutex> _{device->FilterLock};
ALfilter *alfilt{LookupFilter(device, filter)};
if UNLIKELY(!alfilt)
context->setError(AL_INVALID_NAME, "Invalid filter ID %u", filter);
else
{
/* Call the appropriate handler */
ALfilter_getParamf(alfilt, context.get(), param, value);
}
}
END_API_FUNC
AL_API ALvoid AL_APIENTRY alGetFilterfv(ALuint filter, ALenum param, ALfloat *values)
START_API_FUNC
{
ContextRef context{GetContextRef()};
if UNLIKELY(!context) return;
ALCdevice *device{context->mDevice.get()};
std::lock_guard<std::mutex> _{device->FilterLock};
ALfilter *alfilt{LookupFilter(device, filter)};
if UNLIKELY(!alfilt)
context->setError(AL_INVALID_NAME, "Invalid filter ID %u", filter);
else
{
/* Call the appropriate handler */
ALfilter_getParamfv(alfilt, context.get(), param, values);
}
}
END_API_FUNC
FilterSubList::~FilterSubList()
{
uint64_t usemask{~FreeMask};
while(usemask)
{
ALsizei idx = CTZ64(usemask);
al::destroy_at(Filters+idx);
usemask &= ~(1_u64 << idx);
}
FreeMask = ~usemask;
al_free(Filters);
Filters = nullptr;
}

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