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Add a streaming example using a callback buffer

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This commit is contained in:
Chris Robinson
2020-02-20 17:53:09 -08:00
parent 3aad01d3ba
commit 642ef4edc9
2 changed files with 442 additions and 0 deletions
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CMakeLists.txt
+6
View File
@@ -1492,6 +1492,12 @@ IF(ALSOFT_EXAMPLES)
TARGET_LINK_LIBRARIES(alstream
PRIVATE ${LINKER_FLAGS} ${SDL_SOUND_LIBRARIES} ${SDL2_LIBRARY} ex-common)
ADD_EXECUTABLE(alstreamcb examples/alstreamcb.cpp)
TARGET_INCLUDE_DIRECTORIES(alstreamcb
PRIVATE ${SDL2_INCLUDE_DIR} ${SDL_SOUND_INCLUDE_DIR})
TARGET_LINK_LIBRARIES(alstreamcb
PRIVATE ${LINKER_FLAGS} ${SDL_SOUND_LIBRARIES} ${SDL2_LIBRARY} ex-common)
ADD_EXECUTABLE(alreverb examples/alreverb.c)
TARGET_INCLUDE_DIRECTORIES(alreverb
PRIVATE ${SDL2_INCLUDE_DIR} ${SDL_SOUND_INCLUDE_DIR})
examples/alstreamcb.cpp
+436
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@@ -0,0 +1,436 @@
/*
* OpenAL Callback-based Stream Example
*
* Copyright (c) 2020 by Chris Robinson <chris.kcat@gmail.com>
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/* This file contains a streaming audio player using a callback buffer. */
#include <string.h>
#include <stdlib.h>
#include <stdio.h>
#include <atomic>
#include <chrono>
#include <memory>
#include <stdexcept>
#include <string>
#include <thread>
#include <vector>
#include "SDL_sound.h"
#include "SDL_audio.h"
#include "SDL_stdinc.h"
#include "AL/al.h"
#include "AL/alc.h"
#include "common/alhelpers.h"
#ifndef SDL_AUDIO_MASK_BITSIZE
#define SDL_AUDIO_MASK_BITSIZE (0xFF)
#endif
#ifndef SDL_AUDIO_BITSIZE
#define SDL_AUDIO_BITSIZE(x) (x & SDL_AUDIO_MASK_BITSIZE)
#endif
#ifndef AL_SOFT_callback_buffer
#define AL_SOFT_callback_buffer
typedef unsigned int ALbitfieldSOFT;
#define AL_BUFFER_CALLBACK_FUNCTION_SOFT 0x19A0
#define AL_BUFFER_CALLBACK_USER_PARAM_SOFT 0x19A1
typedef ALsizei (AL_APIENTRY*LPALBUFFERCALLBACKTYPESOFT)(ALvoid *userptr, ALvoid *sampledata, ALsizei numsamples);
typedef void (AL_APIENTRY*LPALBUFFERCALLBACKSOFT)(ALuint buffer, ALenum format, ALsizei freq, LPALBUFFERCALLBACKTYPESOFT callback, ALvoid *userptr, ALbitfieldSOFT flags);
typedef void (AL_APIENTRY*LPALGETBUFFERPTRSOFT)(ALuint buffer, ALenum param, ALvoid **value);
typedef void (AL_APIENTRY*LPALGETBUFFER3PTRSOFT)(ALuint buffer, ALenum param, ALvoid **value1, ALvoid **value2, ALvoid **value3);
typedef void (AL_APIENTRY*LPALGETBUFFERPTRVSOFT)(ALuint buffer, ALenum param, ALvoid **values);
#endif
namespace {
using std::chrono::seconds;
using std::chrono::nanoseconds;
LPALBUFFERCALLBACKSOFT alBufferCallbackSOFT;
struct StreamPlayer {
/* A lockless ring-buffer (supports single-provider, single-consumer
* operation).
*/
std::unique_ptr<ALbyte[]> mBufferData;
size_t mBufferDataSize{0};
std::atomic<size_t> mReadPos{0};
std::atomic<size_t> mWritePos{0};
/* The buffer to get the callback, and source to play with. */
ALuint mBuffer{0}, mSource{0};
size_t mStartOffset{0};
/* Handle for the audio file to decode. */
Sound_Sample *mSample{nullptr};
Uint32 mAvailableData{0};
size_t mDecoderOffset{0};
/* The format of the callback samples. */
ALenum mFormat;
ALsizei mSampleRate;
StreamPlayer()
{
alGenBuffers(1, &mBuffer);
if(ALenum err{alGetError()})
throw std::runtime_error{"alGenBuffers failed"};
alGenSources(1, &mSource);
if(ALenum err{alGetError()})
{
alDeleteBuffers(1, &mBuffer);
throw std::runtime_error{"alGenSources failed"};
}
}
~StreamPlayer()
{
alDeleteSources(1, &mSource);
alDeleteBuffers(1, &mBuffer);
if(mSample)
Sound_FreeSample(mSample);
}
void close()
{
if(mSample)
{
alSourceRewind(mSource);
alSourcei(mSource, AL_BUFFER, 0);
Sound_FreeSample(mSample);
mSample = nullptr;
}
}
bool open(const char *filename)
{
close();
/* Open the file in its normal format. */
mSample = Sound_NewSampleFromFile(filename, nullptr, 0);
if(!mSample)
{
fprintf(stderr, "Could not open audio in %s\n", filename);
return false;
}
/* Figure out the OpenAL format from the sample's format. */
mFormat = AL_NONE;
if(mSample->actual.channels == 1)
{
if(mSample->actual.format == AUDIO_U8)
mFormat = AL_FORMAT_MONO8;
else if(mSample->actual.format == AUDIO_S16SYS)
mFormat = AL_FORMAT_MONO16;
}
else if(mSample->actual.channels == 2)
{
if(mSample->actual.format == AUDIO_U8)
mFormat = AL_FORMAT_STEREO8;
else if(mSample->actual.format == AUDIO_S16SYS)
mFormat = AL_FORMAT_STEREO16;
}
if(!mFormat)
{
fprintf(stderr, "Unsupported sample format: 0x%04x, %d channels\n",
mSample->actual.format, mSample->actual.channels);
Sound_FreeSample(mSample);
mSample = nullptr;
return false;
}
mSampleRate = static_cast<ALsizei>(mSample->actual.rate);
const auto frame_size = Uint32{mSample->actual.channels} *
SDL_AUDIO_BITSIZE(mSample->actual.format) / 8;
/* Set a 50ms decode buffer size. */
Sound_SetBufferSize(mSample, static_cast<Uint32>(mSampleRate)*50/1000 * frame_size);
mAvailableData = 0;
/* Set a 1s ring buffer size. */
mBufferDataSize = static_cast<Uint32>(mSampleRate) * size_t{frame_size};
mBufferData.reset(new ALbyte[mBufferDataSize]);
mReadPos.store(0, std::memory_order_relaxed);
mWritePos.store(0, std::memory_order_relaxed);
mDecoderOffset = 0;
return true;
}
/* The actual C-style callback just forwards to the non-static method. Not
* strictly needed and the compiler will optimize it to a normal function,
* but it allows the callback implementation to have a nice 'this' pointer
* with normal member access.
*/
static ALsizei AL_APIENTRY bufferCallbackC(void *userptr, void *data, ALsizei size)
{ return static_cast<StreamPlayer*>(userptr)->bufferCallback(data, size); }
ALsizei bufferCallback(void *data, ALsizei size)
{
/* NOTE: The callback *MUST* be real-time safe! That means no blocking,
* no allocations or deallocations, no I/O, no page faults, or calls to
* functions that could do these things (this includes calling to
* libraries like SDL_sound, libsndfile, ffmpeg, etc). Nothing should
* unexpectedly stall this call since the audio has to get to the
* device on time.
*/
ALsizei got{0};
size_t roffset{mReadPos.load(std::memory_order_acquire)};
while(got < size)
{
/* If the write offset == read offset, there's nothing left in the
* ring-buffer. Break from the loop and give what has been written.
*/
const size_t woffset{mWritePos.load(std::memory_order_relaxed)};
if(woffset == roffset) break;
/* If the write offset is behind the read offset, the readable
* portion wrapped around. Just read up to the end of the buffer in
* that case, otherwise read up to the write offset. Also limit the
* amount to copy given how much is remaining to write.
*/
size_t todo{((woffset < roffset) ? mBufferDataSize : woffset) - roffset};
todo = std::min<size_t>(todo, static_cast<ALuint>(size-got));
/* Copy from the ring buffer to the provided output buffer. Wrap
* the resulting read offset if it reached the end of the ring-
* buffer.
*/
memcpy(data, &mBufferData[roffset], todo);
data = static_cast<ALbyte*>(data) + todo;
got += static_cast<ALsizei>(todo);
roffset += todo;
if(roffset == mBufferDataSize)
roffset = 0;
}
/* Finally, store the updated read offset, and return how many bytes
* have been written.
*/
mReadPos.store(roffset, std::memory_order_release);
return got;
}
bool prepare()
{
alBufferCallbackSOFT(mBuffer, mFormat, mSampleRate, bufferCallbackC, this, 0);
alSourcei(mSource, AL_BUFFER, static_cast<ALint>(mBuffer));
if(ALenum err{alGetError()})
{
fprintf(stderr, "Failed to set callback: %s (0x%04x)\n", alGetString(err), err);
return false;
}
mAvailableData = Sound_Decode(mSample);
if(!mAvailableData)
fprintf(stderr, "Failed to decode any samples: %s\n", Sound_GetError());
return mAvailableData != 0;
}
bool update()
{
constexpr int BadFlags{SOUND_SAMPLEFLAG_EOF | SOUND_SAMPLEFLAG_ERROR};
ALenum state;
ALint pos;
alGetSourcei(mSource, AL_SAMPLE_OFFSET, &pos);
alGetSourcei(mSource, AL_SOURCE_STATE, &state);
size_t woffset{mWritePos.load(std::memory_order_acquire)};
if(state != AL_INITIAL)
{
const auto frame_size = Uint32{mSample->actual.channels} *
SDL_AUDIO_BITSIZE(mSample->actual.format) / 8;
const size_t roffset{mReadPos.load(std::memory_order_relaxed)};
const size_t readable{((woffset >= roffset) ? woffset : (mBufferDataSize+woffset)) -
roffset};
/* For a stopped (underrun) source, the current playback offset is
* the current decoder offset excluding the readable buffered data.
* For a playing/paused source, it's the source's offset including
* the playback offset the source was started with.
*/
const size_t curtime{((state==AL_STOPPED) ? (mDecoderOffset-readable) / frame_size
: (static_cast<ALuint>(pos) + mStartOffset/frame_size)) / mSample->actual.rate};
printf("\r%3zus (%3zu%% full)", curtime, readable * 100 / mBufferDataSize);
}
else
fputs("Starting...", stdout);
fflush(stdout);
while(mAvailableData > 0)
{
const size_t roffset{mReadPos.load(std::memory_order_relaxed)};
if(roffset > woffset)
{
/* Note that the ring buffer's writable space is one byte less
* than the available area because the write offset ending up
* at the read offset would be interpreted as being empty
* instead of full.
*/
const size_t writable{roffset-woffset-1};
/* Don't copy the sample data if it can't all fit. */
if(writable < mAvailableData) break;
memcpy(&mBufferData[woffset], mSample->buffer, mAvailableData);
woffset += mAvailableData;
}
else
{
/* If the read offset is at or behind the write offset, the
* writeable area (might) wrap around. Make sure the sample
* data can fit, and calculate how much goes in front and in
* back.
*/
const size_t writable{mBufferDataSize+roffset-woffset-1};
if(writable < mAvailableData) break;
const size_t todo1{std::min<size_t>(mAvailableData, mBufferDataSize-woffset)};
const size_t todo2{mAvailableData - todo1};
memcpy(&mBufferData[woffset], mSample->buffer, todo1);
woffset += todo1;
if(woffset == mBufferDataSize)
{
woffset = 0;
if(todo2 > 0)
{
memcpy(&mBufferData[woffset], static_cast<ALbyte*>(mSample->buffer)+todo1,
todo2);
woffset += todo2;
}
}
}
mWritePos.store(woffset, std::memory_order_release);
mDecoderOffset += mAvailableData;
if(!(mSample->flags&BadFlags))
mAvailableData = Sound_Decode(mSample);
else
mAvailableData = 0;
}
if(state != AL_PLAYING && state != AL_PAUSED)
{
/* If the source is not playing or paused, it either underrun
* (AL_STOPPED) or is just getting started (AL_INITIAL). If the
* ring buffer is empty, it's done, otherwise play the source with
* what's available.
*/
const size_t roffset{mReadPos.load(std::memory_order_relaxed)};
const size_t readable{((woffset >= roffset) ? woffset : (mBufferDataSize+woffset)) -
roffset};
if(readable == 0)
return false;
/* Store the playback offset that the source will start reading
* from, so it can be tracked during playback.
*/
mStartOffset = mDecoderOffset - readable;
alSourcePlay(mSource);
if(alGetError() != AL_NO_ERROR)
return false;
}
return true;
}
};
} // namespace
int main(int argc, char **argv)
{
/* A simple RAII container for OpenAL and SDL_sound startup and shutdown. */
struct AudioManager {
AudioManager(char ***argv_, int *argc_)
{
if(InitAL(argv_, argc_) != 0)
throw std::runtime_error{"Failed to initialize OpenAL"};
Sound_Init();
}
~AudioManager() { Sound_Quit(); CloseAL(); }
};
/* Print out usage if no arguments were specified */
if(argc < 2)
{
fprintf(stderr, "Usage: %s [-device <name>] <filenames...>\n", argv[0]);
return 1;
}
argv++; argc--;
AudioManager almgr{&argv, &argc};
if(!alIsExtensionPresent("AL_SOFTX_callback_buffer"))
{
fprintf(stderr, "AL_SOFT_callback_buffer extension not available\n");
return 1;
}
alBufferCallbackSOFT = reinterpret_cast<LPALBUFFERCALLBACKSOFT>(
alGetProcAddress("alBufferCallbackSOFT"));
ALCint refresh{25};
alcGetIntegerv(alcGetContextsDevice(alcGetCurrentContext()), ALC_REFRESH, 1, &refresh);
std::unique_ptr<StreamPlayer> player{new StreamPlayer{}};
/* Play each file listed on the command line */
for(int i{0};i < argc;++i)
{
if(!player->open(argv[i]))
continue;
/* Get the name portion, without the path, for display. */
const char *namepart{strrchr(argv[i], '/')};
if(namepart || (namepart=strrchr(argv[i], '\\')))
++namepart;
else
namepart = argv[i];
printf("Playing: %s (%s, %dhz)\n", namepart, FormatName(player->mFormat),
player->mSampleRate);
fflush(stdout);
if(!player->prepare())
{
player->close();
continue;
}
while(player->update())
std::this_thread::sleep_for(nanoseconds{seconds{1}} / refresh);
putc('\n', stdout);
/* All done with this file. Close it and go to the next */
player->close();
}
/* All done. */
printf("Done.\n");
return 0;
}