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Review audio module and examples

pull/172/head
raysan5 8 years ago
parent
36cf1f7dfd
commit
58d2f70b7e
4 changed files with 75 additions and 124 deletions
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  1. +0
    -1
      examples/audio_music_stream.c
  2. +7
    -4
      examples/audio_standalone.c
  3. +63
    -116
      src/audio.c
  4. +5
    -3
      templates/android_project/jni/basic_game.c

+ 0
- 1
examples/audio_music_stream.c View File

@ -54,7 +54,6 @@ int main()
{
volume = 1.0;
framesCounter = 0;
PlayMusicStream(1, "resources/audio/another_file.ogg");
}
SetMusicVolume(volume);

+ 7
- 4
examples/audio_standalone.c View File

@ -39,7 +39,8 @@ int main()
Sound fxWav = LoadSound("resources/audio/weird.wav"); // Load WAV audio file
Sound fxOgg = LoadSound("resources/audio/tanatana.ogg"); // Load OGG audio file
PlayMusicStream(0, "resources/audio/guitar_noodling.ogg");
Music music = LoadMusicStream("resources/audio/guitar_noodling.ogg");
PlayMusicStream(music);
printf("\nPress s or d to play sounds...\n");
@ -59,11 +60,13 @@ int main()
key = 0;
}
UpdateMusicStream(mi">0);
UpdateMusicStream(n">music);
}
UnloadSound(fxWav); // Unload sound data
UnloadSound(fxOgg); // Unload sound data
UnloadSound(fxWav); // Unload sound data
UnloadSound(fxOgg); // Unload sound data
UnloadMusicStream(music); // Unload music stream data
CloseAudioDevice();

+ 63
- 116
src/audio.c View File

@ -86,13 +86,13 @@
//----------------------------------------------------------------------------------
// Defines and Macros
//----------------------------------------------------------------------------------
#define MAX_STREAM_BUFFERS 2 // Number of buffers for each audio stream
#define MAX_STREAM_BUFFERS 2 // Number of buffers for each audio stream
// NOTE: Music buffer size is defined by number of samples, independent of sample size
// After some math, considering a sampleRate of 48000, a buffer refill rate of 1/60 seconds
// and double-buffering system, I concluded that a 4096 samples buffer should be enough
// In case of music-stalls, just inclease this number
#define AUDIO_BUFFER_SIZE 4096 // PCM data samples (i.e. short: 32Kb)
// In case of music-stalls, just increase this number
#define AUDIO_BUFFER_SIZE 4096 // PCM data samples (i.e. short: 32Kb)
//----------------------------------------------------------------------------------
// Types and Structures Definition
@ -141,12 +141,12 @@ 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 AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, unsigned int channels);
static void CloseAudioStream(AudioStream stream); // Frees mix channel
static int BufferAudioStream(AudioStream stream, void *data, int numberElements); // Pushes more audio data into mix channel
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)
@ -492,27 +492,23 @@ Music LoadMusicStream(char *fileName)
// Open ogg audio stream
music->ctxOgg = stb_vorbis_open_filename(fileName, NULL, NULL);
if (music->ctxOgg == NULL)
{
TraceLog(WARNING, "[%s] OGG audio file could not be opened", fileName);
}
if (music->ctxOgg == NULL) TraceLog(WARNING, "[%s] OGG audio file could not be opened", fileName);
else
{
stb_vorbis_info info = stb_vorbis_get_info(music->ctxOgg); // Get Ogg file info
TraceLog(DEBUG, "[%s] Ogg sample rate: %i", fileName, info.sample_rate);
TraceLog(DEBUG, "[%s] Ogg channels: %i", fileName, info.channels);
TraceLog(DEBUG, "[%s] Temp memory required: %i", fileName, info.temp_memory_required);
//float totalLengthSeconds = stb_vorbis_stream_length_in_seconds(music->ctxOgg);
// TODO: Support 32-bit sampleSize OGGs
music->stream = InitAudioStream(info.sample_rate, 16, info.channels);
music->totalSamples = (unsigned int)stb_vorbis_stream_length_in_samples(music->ctxOgg)*info.channels;
music->samplesLeft = music->totalSamples;
//float totalLengthSeconds = stb_vorbis_stream_length_in_seconds(music->ctxOgg);
music->ctxType = MUSIC_AUDIO_OGG;
music->loop = true; // We loop by default
TraceLog(DEBUG, "[%s] OGG sample rate: %i", fileName, info.sample_rate);
TraceLog(DEBUG, "[%s] OGG channels: %i", fileName, info.channels);
TraceLog(DEBUG, "[%s] OGG memory required: %i", fileName, info.temp_memory_required);
}
}
else if (strcmp(GetExtension(fileName), "xm") == 0)
@ -523,17 +519,15 @@ Music LoadMusicStream(char *fileName)
{
jar_xm_set_max_loop_count(music->ctxXm, 0); // Set infinite number of loops
music->totalSamples = (unsigned int)jar_xm_get_remaining_samples(music->ctxXm);
music->samplesLeft = music->totalSamples;
TraceLog(INFO, "[%s] XM number of samples: %i", fileName, music->totalSamples);
TraceLog(INFO, "[%s] XM track length: %11.6f sec", fileName, (float)music->totalSamples/48000.0f);
// NOTE: Only stereo is supported for XM
music->stream = InitAudioStream(48000, 32, 2);
music->totalSamples = (unsigned int)jar_xm_get_remaining_samples(music->ctxXm);
music->samplesLeft = music->totalSamples;
music->ctxType = MUSIC_MODULE_XM;
music->loop = true;
TraceLog(DEBUG, "[%s] XM number of samples: %i", fileName, music->totalSamples);
TraceLog(DEBUG, "[%s] XM track length: %11.6f sec", fileName, (float)music->totalSamples/48000.0f);
}
else TraceLog(WARNING, "[%s] XM file could not be opened", fileName);
}
@ -543,16 +537,14 @@ Music LoadMusicStream(char *fileName)
if (jar_mod_load_file(&music->ctxMod, fileName))
{
music->stream = InitAudioStream(48000, 16, 2);
music->totalSamples = (unsigned int)jar_mod_max_samples(&music->ctxMod);
music->samplesLeft = music->totalSamples;
music->ctxType = MUSIC_MODULE_MOD;
music->loop = true;
TraceLog(INFO, "[%s] MOD number of samples: %i", fileName, music->samplesLeft);
TraceLog(INFO, "[%s] MOD track length: %11.6f sec", fileName, (float)music->totalSamples/48000.0f);
music->stream = InitAudioStream(48000, 16, 2);
music->ctxType = MUSIC_MODULE_MOD;
music->loop = true;
}
else TraceLog(WARNING, "[%s] MOD file could not be opened", fileName);
}
@ -799,42 +791,18 @@ static void CloseAudioStream(AudioStream stream)
}
// Push more audio data into audio stream, only one buffer per call
// NOTE: Returns number of samples that were processed
static int BufferAudioStream(AudioStream stream, void *data, int numberElements)
{
if (!data || !numberElements)
{
// Pauses audio until data is given
alSourcePause(stream.source);
return 0;
}
static void BufferAudioStream(AudioStream stream, void *data, int numSamples)
{
ALuint buffer = 0;
alSourceUnqueueBuffers(stream.source, 1, &buffer);
if (!buffer) return 0;
// Reference
//void alBufferData(ALuint bufferName, ALenum format, const ALvoid *data, ALsizei size, ALsizei frequency);
// ALuint bufferName: buffer id
// ALenum format: Valid formats are
// AL_FORMAT_MONO8, // unsigned char
// AL_FORMAT_MONO16, // short
// AL_FORMAT_STEREO8,
// AL_FORMAT_STEREO16 // stereo data is interleaved: left+right channels sample
// AL_FORMAT_MONO_FLOAT32 (extension)
// AL_FORMAT_STEREO_FLOAT32 (extension)
// ALsizei size: Number of bytes, must be coherent with format
// ALsizei frequency: sample rate
if (stream.sampleSize == 8) alBufferData(buffer, stream.format, (unsigned char *)data, numberElements*sizeof(unsigned char), stream.sampleRate);
else if (stream.sampleSize == 16) alBufferData(buffer, stream.format, (short *)data, numberElements*sizeof(short), stream.sampleRate);
else if (stream.sampleSize == 32) alBufferData(buffer, stream.format, (float *)data, numberElements*sizeof(float), stream.sampleRate);
//TraceLog(DEBUG, "Buffer to refill: %i", buffer);
alSourceQueueBuffers(stream.source, 1, &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);
return numberElements;
alSourceQueueBuffers(stream.source, 1, &buffer);
}
// Fill music buffers with new data from music stream
@ -845,70 +813,49 @@ static bool BufferMusicStream(Music music, int numBuffersToProcess)
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)
if (music->ctxType == MUSIC_MODULE_XM) // There is no end of stream for xmfiles, once the end is reached zeros are generated for non looped chiptunes.
for (int i = 0; i < numBuffersToProcess; i++)
{
for (int i = 0; i < numBuffersToProcess; i++)
{
if (music->samplesLeft >= AUDIO_BUFFER_SIZE) size = AUDIO_BUFFER_SIZE/2;
else size = music->samplesLeft/2;
// Read 2*shorts and moves them to buffer+size memory location
jar_xm_generate_samples(music->ctxXm, pcmf, size);
BufferAudioStream(music->stream, pcmf, size*2);
music->samplesLeft -= size;
if (music->samplesLeft >= AUDIO_BUFFER_SIZE) size = AUDIO_BUFFER_SIZE;
else size = music->samplesLeft;
if (music->samplesLeft <= 0)
{
active = false;
break;
}
}
}
else if (music->ctxType == MUSIC_MODULE_MOD)
{
for (int i = 0; i < numBuffersToProcess; i++)
switch (music->ctxType)
{
if (music->samplesLeft >= AUDIO_BUFFER_SIZE) size = AUDIO_BUFFER_SIZE/2;
else size = music->samplesLeft/2;
jar_mod_fillbuffer(&music->ctxMod, pcm, size, 0);
BufferAudioStream(music->stream, pcm, size*2);
music->samplesLeft -= size;
if (music->samplesLeft <= 0)
case MUSIC_AUDIO_OGG:
{
active = false;
break;
}
// 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;
}
}
else if (music->ctxType == MUSIC_AUDIO_OGG)
{
if (music->samplesLeft >= AUDIO_BUFFER_SIZE) size = AUDIO_BUFFER_SIZE;
else size = music->samplesLeft;
for (int i = 0; i < numBuffersToProcess; i++)
if (music->samplesLeft <= 0)
{
// NOTE: Returns the number of samples stored per channel
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);
if (music->samplesLeft <= 0)
{
active = false;
break;
}
active = false;
break;
}
}
return active;
}

+ 5
- 3
templates/android_project/jni/basic_game.c View File

@ -43,7 +43,8 @@ void android_main(struct android_app *app)
int framesCounter = 0; // Used to count frames
PlayMusicStream(0, "ambient.ogg");
Music ambient = LoadMusicStream("ambient.ogg");
PlayMusicStream(ambient);
SetTargetFPS(60); // Not required on Android, already locked to 60 fps
//--------------------------------------------------------------------------------------
@ -53,7 +54,7 @@ void android_main(struct android_app *app)
{
// Update
//----------------------------------------------------------------------------------
UpdateMusicStream(mi">0);
UpdateMusicStream(n">ambient);
switch(currentScreen)
{
@ -158,7 +159,8 @@ void android_main(struct android_app *app)
// TODO: Unload all loaded data (textures, fonts, audio) here!
UnloadSound(fx); // Unload sound data
UnloadSound(fx); // Unload sound data
UnloadMusicStream(ambient); // Unload music stream data
CloseAudioDevice(); // Close audio device (music streaming is automatically stopped)

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