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Corrected audio bugs and improved examples

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raysan5
2016-09-15 11:53:16 +02:00
parent 9923fe51a7
commit 79c8eb543e
3 changed files with 93 additions and 60 deletions
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70
src/audio.c
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@@ -225,12 +225,16 @@ Wave LoadWaveEx(float *data, int sampleCount, int sampleRate, int sampleSize, in
wave.data = data;
wave.sampleCount = sampleCount;
wave.sampleRate = sampleRate;
wave.sampleSize = sampleSize;
wave.sampleSize = 32;
wave.channels = channels;
WaveFormat(&wave, sampleRate, sampleSize, channels);
// NOTE: Copy wave data to work with,
// user is responsible of input data to free
Wave cwave = WaveCopy(wave);
return wave;
WaveFormat(&cwave, sampleRate, sampleSize, channels);
return cwave;
}
// Load sound to memory
@@ -578,6 +582,8 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
}
}
else TraceLog(WARNING, "Wave formatting: Sample size not supported");
free(samples);
}
// NOTE: Only supported 1 or 2 channels (mono or stereo)
@@ -615,7 +621,8 @@ Wave WaveCopy(Wave wave)
// NOTE: Security check in case of out-of-range
void WaveCrop(Wave *wave, int initSample, int finalSample)
{
if ((initSample >= 0) && (finalSample > 0) && (finalSample < wave->sampleCount))
if ((initSample >= 0) && (initSample < finalSample) &&
(finalSample > 0) && (finalSample < wave->sampleCount))
{
// TODO: Review cropping (it could be simplified...)
@@ -636,6 +643,7 @@ void WaveCrop(Wave *wave, int initSample, int finalSample)
// Get samples data from wave as a floats array
// NOTE: Returned sample values are normalized to range [-1..1]
// TODO: Consider multiple channels (mono - stereo)
float *GetWaveData(Wave wave)
{
float *samples = (float *)malloc(wave.sampleCount*sizeof(float));
@@ -759,26 +767,37 @@ void ResumeMusicStream(Music music)
}
// Stop music playing (close stream)
// TODO: Restart XM context
void StopMusicStream(Music music)
{
alSourceStop(music->stream.source);
switch (music->ctxType)
{
case MUSIC_AUDIO_OGG: stb_vorbis_seek_start(music->ctxOgg); break;
case MUSIC_MODULE_XM: break;
case MUSIC_MODULE_MOD: jar_mod_seek_start(&music->ctxMod); break;
default: break;
}
music->samplesLeft = music->totalSamples;
}
// Update (re-fill) music buffers if data already processed
void UpdateMusicStream(Music music)
{
ALenum state;
ALint processed = 0;
// Determine if music stream is ready to be written
alGetSourcei(music->stream.source, AL_BUFFERS_PROCESSED, &processed);
int numBuffersToProcess = processed;
alGetSourcei(music->stream.source, AL_SOURCE_STATE, &state); // Get music stream state
alGetSourcei(music->stream.source, AL_BUFFERS_PROCESSED, &processed); // Get processed buffers
if (processed > 0)
{
bool active = true;
short pcm[AUDIO_BUFFER_SIZE];
float pcmf[AUDIO_BUFFER_SIZE];
int numBuffersToProcess = processed;
int numSamples = 0; // Total size of data steamed in L+R samples for xm floats,
// individual L or R for ogg shorts
@@ -833,28 +852,23 @@ void UpdateMusicStream(Music music)
break;
}
}
// Reset audio stream for looping
if (!active && music->loop)
{
// Restart music context (if required)
//if (music->ctxType == MUSIC_MODULE_XM)
if (music->ctxType == MUSIC_MODULE_MOD) jar_mod_seek_start(&music->ctxMod);
else if (music->ctxType == MUSIC_AUDIO_OGG) stb_vorbis_seek_start(music->ctxOgg);
// Reset samples left to total samples
music->samplesLeft = music->totalSamples;
}
// This error is registered when UpdateAudioStream() fails
if (alGetError() == AL_INVALID_VALUE) TraceLog(WARNING, "OpenAL: Error buffering data...");
ALenum state;
alGetSourcei(music->stream.source, AL_SOURCE_STATE, &state);
if (state != AL_PLAYING && active) alSourcePlay(music->stream.source);
if (!active) StopMusicStream(music);
// Reset audio stream for looping
if (!active)
{
StopMusicStream(music); // Stop music (and reset)
if (music->loop) PlayMusicStream(music); // Play again
}
else
{
// NOTE: In case window is minimized, music stream is stopped,
// just make sure to play again on window restore
if (state != AL_PLAYING) PlayMusicStream(music);
}
}
}