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Add audio-streaming and loop support

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This commit is contained in:
Doyle Thai 2016-07-26 18:47:55 +10:00
parent 6b2e759257
commit 99a700ca13
1 changed files with 78 additions and 15 deletions
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93
src/dengine.c
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@ -149,7 +149,11 @@ int main()
******************* *******************
*/ */
alGetError(); alGetError();
// TODO(doyle): Read this http://www.gamedev.net/page/resources/_/technical/game-programming/basic-openal-sound-manager-for-your-project-r3791 // TODO(doyle): Read this
// http://www.gamedev.net/page/resources/_/technical/game-programming/basic-openal-sound-manager-for-your-project-r3791
// https://gist.github.com/Oddity007/965399
// https://jogamp.org/joal-demos/www/devmaster/lesson8.html
// http://basic-converter.proboards.com/thread/818/play-files-using-vorbis-openal
ALCdevice *deviceAL = alcOpenDevice(NULL); ALCdevice *deviceAL = alcOpenDevice(NULL);
if (!deviceAL) if (!deviceAL)
{ {
@ -168,32 +172,33 @@ int main()
/* Open audio file */ /* Open audio file */
PlatformFileRead fileRead = {0}; PlatformFileRead fileRead = {0};
#if 0
platform_readFileToBuffer(&worldTraveller.arena, platform_readFileToBuffer(&worldTraveller.arena,
"data/audio/Yuki Kajiura - Swordland.ogg", "data/audio/Yuki Kajiura - Swordland.ogg",
&fileRead); &fileRead);
#else
platform_readFileToBuffer(&worldTraveller.arena,
"data/audio/Nobuo Uematsu - Battle 1.ogg",
&fileRead);
#endif
i32 channels, sampleRate, numSamples; i32 error;
ALshort *vorbisData = NULL; stb_vorbis *vorbisFile = stb_vorbis_open_memory(fileRead.buffer, fileRead.size,
numSamples = stb_vorbis_decode_memory(fileRead.buffer, fileRead.size, &error, NULL);
&channels, &sampleRate, &vorbisData); stb_vorbis_info vorbisInfo = stb_vorbis_get_info(vorbisFile);
platform_closeFileRead(&worldTraveller.arena, &fileRead); //platform_closeFileRead(&worldTraveller.arena, &fileRead);
/* Number of concurrent audio files */ /* Generate number of concurrent audio file listeners */
ALuint audioSourceId; ALuint audioSourceId;
alGenSources(1, &audioSourceId); alGenSources(1, &audioSourceId);
AL_CHECK_ERROR(); AL_CHECK_ERROR();
/* Audio data buffers */ /* Generate audio data buffers */
ALuint audioBufferId; ALuint audioBufferId[4];
alGenBuffers(1, &audioBufferId); alGenBuffers(ARRAY_COUNT(audioBufferId), audioBufferId);
AL_CHECK_ERROR(); AL_CHECK_ERROR();
alBufferData(audioBufferId, AL_FORMAT_STEREO16, vorbisData,
numSamples * channels * sizeof(i16), sampleRate);
alSourceQueueBuffers(audioSourceId, 1, &audioBufferId);
alSourcePlay(audioSourceId);
#if 0 #if 0
ALuint audioFormat = AL_FORMAT_MONO16; ALuint audioFormat = AL_FORMAT_MONO16;
@ -239,6 +244,64 @@ int main()
f32 endTime = CAST(f32)glfwGetTime(); f32 endTime = CAST(f32)glfwGetTime();
secondsElapsed = endTime - startTime; secondsElapsed = endTime - startTime;
#define AUDIO_CHUNK_SIZE 65536
ALint audioState;
alGetSourcei(audioSourceId, AL_SOURCE_STATE, &audioState);
if (audioState == AL_STOPPED || audioState == AL_INITIAL)
{
// TODO(doyle): This fixes clicking when reusing old buffers
if (audioState == AL_STOPPED)
{
alDeleteBuffers(ARRAY_COUNT(audioBufferId), audioBufferId);
alGenBuffers(ARRAY_COUNT(audioBufferId), audioBufferId);
}
stb_vorbis_seek_start(vorbisFile);
for (i32 i = 0; i < ARRAY_COUNT(audioBufferId); i++)
{
i16 audioChunk[AUDIO_CHUNK_SIZE] = {0};
stb_vorbis_get_samples_short_interleaved(
vorbisFile, vorbisInfo.channels, audioChunk,
AUDIO_CHUNK_SIZE);
alBufferData(audioBufferId[i], AL_FORMAT_STEREO16, audioChunk,
AUDIO_CHUNK_SIZE * sizeof(i16),
vorbisInfo.sample_rate);
}
alSourceQueueBuffers(audioSourceId, ARRAY_COUNT(audioBufferId),
audioBufferId);
alSourcePlay(audioSourceId);
}
else if (audioState == AL_PLAYING)
{
ALint numProcessedBuffers;
alGetSourcei(audioSourceId, AL_BUFFERS_PROCESSED,
&numProcessedBuffers);
if (numProcessedBuffers > 0)
{
ALint numBuffersToUnqueue = 1;
ALuint emptyBufferId;
alSourceUnqueueBuffers(audioSourceId, numBuffersToUnqueue,
&emptyBufferId);
i16 audioChunk[AUDIO_CHUNK_SIZE] = {0};
i32 sampleCount = stb_vorbis_get_samples_short_interleaved(
vorbisFile, vorbisInfo.channels, audioChunk,
AUDIO_CHUNK_SIZE);
/* There are still samples to play */
if (sampleCount > 0)
{
DEBUG_LOG("Buffering new audio data");
alBufferData(emptyBufferId, AL_FORMAT_STEREO16, audioChunk,
sampleCount * vorbisInfo.channels *
sizeof(i16),
vorbisInfo.sample_rate);
alSourceQueueBuffers(audioSourceId, 1, &emptyBufferId);
}
}
}
#if 0 #if 0
// TODO(doyle): Busy waiting, should sleep // TODO(doyle): Busy waiting, should sleep
while (secondsElapsed < targetSecondsPerFrame) while (secondsElapsed < targetSecondsPerFrame)