twix/raylib
1
0
Fork 0
mirror of https://github.com/raysan5/raylib.git synced 2025-12-25 10:22:33 -05:00
Code Issues Packages Projects Releases Wiki Activity

Complete review and update

Browse Source 
Simplified module for Music and AudioStream
Added support for raw audio streaming (with example)
This commit is contained in:
raysan5
2016-08-02 17:32:24 +02:00
parent 58d2f70b7e
commit 68d647c1af
10 changed files with 283 additions and 127 deletions
Download Patch File Download Diff File Expand all files Collapse all files

242
src/audio.c
View File

@ -100,17 +100,6 @@
typedef enum { MUSIC_AUDIO_OGG = 0, MUSIC_MODULE_XM, MUSIC_MODULE_MOD } MusicContextType;
// Used to create custom audio streams that are not bound to a specific file.
typedef struct AudioStream {
unsigned int sampleRate; // Frequency (samples per second): default is 48000
unsigned int sampleSize; // BitDepth (bits per sample): 8, 16, 32 (24 not supported)
unsigned int channels; // Number of channels
ALenum format; // OpenAL format specifier
ALuint source; // OpenAL source
ALuint buffers[MAX_STREAM_BUFFERS]; // OpenAL buffers (double buffering)
} AudioStream;
// Music type (file streaming from memory)
typedef struct Music {
MusicContextType ctxType; // Type of music context (OGG, XM, MOD)
@ -118,7 +107,7 @@ typedef struct Music {
jar_xm_context_t *ctxXm; // XM chiptune context
jar_mod_context_t ctxMod; // MOD chiptune context
AudioStream stream; // Audio stream
AudioStream stream; // Audio stream (double buffering)
bool loop; // Repeat music after finish (loop)
unsigned int totalSamples; // Total number of samples
@ -141,12 +130,6 @@ static Wave LoadWAV(const char *fileName); // Load WAV file
static Wave LoadOGG(char *fileName); // Load OGG file
static void UnloadWave(Wave wave); // Unload wave data
static bool BufferMusicStream(Music music, int numBuffersToProcess); // Fill music buffers with data
static AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, unsigned int channels);
static void BufferAudioStream(AudioStream stream, void *data, int numSamples);
static void CloseAudioStream(AudioStream stream);
#if defined(AUDIO_STANDALONE)
const char *GetExtension(const char *fileName); // Get the extension for a filename
void TraceLog(int msgType, const char *text, ...); // Outputs a trace log message (INFO, ERROR, WARNING)
@ -595,33 +578,89 @@ void StopMusicStream(Music music)
// Update (re-fill) music buffers if data already processed
void UpdateMusicStream(Music music)
{
ALenum state;
bool active = true;
ALint processed = 0;
// Determine if music stream is ready to be written
alGetSourcei(music->stream.source, AL_BUFFERS_PROCESSED, &processed);
int numBuffersToProcess = processed;
if (processed > 0)
{
active = BufferMusicStream(music, processed);
bool active = true;
short pcm[AUDIO_BUFFER_SIZE];
float pcmf[AUDIO_BUFFER_SIZE];
int numSamples = 0; // Total size of data steamed in L+R samples for xm floats,
// individual L or R for ogg shorts
for (int i = 0; i < numBuffersToProcess; i++)
{
switch (music->ctxType)
{
case MUSIC_AUDIO_OGG:
{
if (music->samplesLeft >= AUDIO_BUFFER_SIZE) numSamples = AUDIO_BUFFER_SIZE;
else numSamples = music->samplesLeft;
// NOTE: Returns the number of samples to process (should be the same as numSamples -> it is)
int numSamplesOgg = stb_vorbis_get_samples_short_interleaved(music->ctxOgg, music->stream.channels, pcm, numSamples);
// TODO: Review stereo channels Ogg, not enough samples served!
UpdateAudioStream(music->stream, pcm, numSamples*music->stream.channels);
music->samplesLeft -= (numSamples*music->stream.channels);
} break;
case MUSIC_MODULE_XM:
{
if (music->samplesLeft >= AUDIO_BUFFER_SIZE/2) numSamples = AUDIO_BUFFER_SIZE/2;
else numSamples = music->samplesLeft;
// NOTE: Output buffer is 2*numsamples elements (left and right value for each sample)
jar_xm_generate_samples(music->ctxXm, pcmf, numSamples);
UpdateAudioStream(music->stream, pcmf, numSamples*2); // Using 32bit PCM data
music->samplesLeft -= numSamples;
//TraceLog(INFO, "Samples left: %i", music->samplesLeft);
} break;
case MUSIC_MODULE_MOD:
{
if (music->samplesLeft >= AUDIO_BUFFER_SIZE/2) numSamples = AUDIO_BUFFER_SIZE/2;
else numSamples = music->samplesLeft;
// NOTE: Output buffer size is nbsample*channels (default: 48000Hz, 16bit, Stereo)
jar_mod_fillbuffer(&music->ctxMod, pcm, numSamples, 0);
UpdateAudioStream(music->stream, pcm, numSamples*2);
music->samplesLeft -= numSamples;
} break;
default: break;
}
if (music->samplesLeft <= 0)
{
active = false;
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;
// Determine if music stream is ready to be written
alGetSourcei(music->stream.source, AL_BUFFERS_PROCESSED, &processed);
active = BufferMusicStream(music, processed);
}
if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data...");
// 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);
@ -668,36 +707,14 @@ float GetMusicTimePlayed(Music music)
{
float secondsPlayed = 0.0f;
if (music->ctxType == MUSIC_MODULE_XM)
{
uint64_t samplesPlayed;
jar_xm_get_position(music->ctxXm, NULL, NULL, NULL, &samplesPlayed);
// TODO: Not sure if this is the correct value
secondsPlayed = (float)samplesPlayed/(music->stream.sampleRate*music->stream.channels);
}
else if (music->ctxType == MUSIC_MODULE_MOD)
{
long samplesPlayed = jar_mod_current_samples(&music->ctxMod);
secondsPlayed = (float)samplesPlayed/music->stream.sampleRate;
}
else if (music->ctxType == MUSIC_AUDIO_OGG)
{
unsigned int samplesPlayed = music->totalSamples - music->samplesLeft;
secondsPlayed = (float)samplesPlayed/(music->stream.sampleRate*music->stream.channels);
}
unsigned int samplesPlayed = music->totalSamples - music->samplesLeft;
secondsPlayed = (float)samplesPlayed/(music->stream.sampleRate*music->stream.channels);
return secondsPlayed;
}
//----------------------------------------------------------------------------------
// Module specific Functions Definition
//----------------------------------------------------------------------------------
// Init audio stream (to stream audio pcm data)
static AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, unsigned int channels)
AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, unsigned int channels)
{
AudioStream stream = { 0 };
@ -735,7 +752,7 @@ static AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleS
alSource3f(stream.source, AL_POSITION, 0, 0, 0);
alSource3f(stream.source, AL_VELOCITY, 0, 0, 0);
// Create Buffers
// Create Buffers (double buffering)
alGenBuffers(MAX_STREAM_BUFFERS, stream.buffers);
// Initialize buffer with zeros by default
@ -766,7 +783,7 @@ static AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleS
}
// Close audio stream and free memory
static void CloseAudioStream(AudioStream stream)
void CloseAudioStream(AudioStream stream)
{
// Stop playing channel
alSourceStop(stream.source);
@ -790,75 +807,66 @@ static void CloseAudioStream(AudioStream stream)
TraceLog(INFO, "[AUD ID %i] Unloaded audio stream data", stream.source);
}
// Push more audio data into audio stream, only one buffer per call
static void BufferAudioStream(AudioStream stream, void *data, int numSamples)
{
// Update audio stream buffers with data
// NOTE: Only one buffer per call
void UpdateAudioStream(AudioStream stream, void *data, int numSamples)
{
ALuint buffer = 0;
alSourceUnqueueBuffers(stream.source, 1, &buffer);
//TraceLog(DEBUG, "Buffer to refill: %i", buffer);
if (stream.sampleSize == 8) alBufferData(buffer, stream.format, (unsigned char *)data, numSamples*sizeof(unsigned char), stream.sampleRate);
else if (stream.sampleSize == 16) alBufferData(buffer, stream.format, (short *)data, numSamples*sizeof(short), stream.sampleRate);
else if (stream.sampleSize == 32) alBufferData(buffer, stream.format, (float *)data, numSamples*sizeof(float), stream.sampleRate);
alSourceQueueBuffers(stream.source, 1, &buffer);
}
// Fill music buffers with new data from music stream
static bool BufferMusicStream(Music music, int numBuffersToProcess)
{
short pcm[AUDIO_BUFFER_SIZE];
float pcmf[AUDIO_BUFFER_SIZE];
int size = 0; // Total size of data steamed in L+R samples for xm floats, individual L or R for ogg shorts
bool active = true; // We can get more data from stream (not finished)
for (int i = 0; i < numBuffersToProcess; i++)
// Check if any buffer was available for unqueue
if (alGetError() != AL_INVALID_VALUE)
{
if (music->samplesLeft >= AUDIO_BUFFER_SIZE) size = AUDIO_BUFFER_SIZE;
else size = music->samplesLeft;
switch (music->ctxType)
{
case MUSIC_AUDIO_OGG:
{
// NOTE: Returns the number of samples to process (should be the same as size)
int numSamples = stb_vorbis_get_samples_short_interleaved(music->ctxOgg, music->stream.channels, pcm, size);
BufferAudioStream(music->stream, pcm, numSamples*music->stream.channels);
music->samplesLeft -= (numSamples*music->stream.channels);
} break;
case MUSIC_MODULE_XM:
{
// NOTE: Output buffer is 2*numsamples elements (left and right value for each sample)
jar_xm_generate_samples(music->ctxXm, pcmf, size/2);
BufferAudioStream(music->stream, pcmf, size); // Using 32bit PCM data
music->samplesLeft -= (size/2);
} break;
case MUSIC_MODULE_MOD:
{
// NOTE: Output buffer size is nbsample*channels (default: 48000Hz, 16bit, Stereo)
jar_mod_fillbuffer(&music->ctxMod, pcm, size/2, 0);
BufferAudioStream(music->stream, pcm, size);
music->samplesLeft -= (size/2);
} break;
default: break;
}
if (music->samplesLeft <= 0)
{
active = false;
break;
}
if (stream.sampleSize == 8) alBufferData(buffer, stream.format, (unsigned char *)data, numSamples*sizeof(unsigned char), stream.sampleRate);
else if (stream.sampleSize == 16) alBufferData(buffer, stream.format, (short *)data, numSamples*sizeof(short), stream.sampleRate);
else if (stream.sampleSize == 32) alBufferData(buffer, stream.format, (float *)data, numSamples*sizeof(float), stream.sampleRate);
alSourceQueueBuffers(stream.source, 1, &buffer);
}
return active;
}
// Check if any audio stream buffers requires refill
bool IsAudioBufferProcessed(AudioStream stream)
{
ALint processed = 0;
// Determine if music stream is ready to be written
alGetSourcei(stream.source, AL_BUFFERS_PROCESSED, &processed);
return (processed > 0);
}
// Play audio stream
void PlayAudioStream(AudioStream stream)
{
alSourcePlay(stream.source);
}
// Play audio stream
void PauseAudioStream(AudioStream stream)
{
alSourcePause(stream.source);
}
// Resume audio stream playing
void ResumeAudioStream(AudioStream stream)
{
ALenum state;
alGetSourcei(stream.source, AL_SOURCE_STATE, &state);
if (state == AL_PAUSED) alSourcePlay(stream.source);
}
// Stop audio stream
void StopAudioStream(AudioStream stream)
{
alSourceStop(stream.source);
}
//----------------------------------------------------------------------------------
// Module specific Functions Definition
//----------------------------------------------------------------------------------
// Load WAV file into Wave structure
static Wave LoadWAV(const char *fileName)
{