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Remove sampleLeft from Music struct... -WIP-

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...moved to internal AudioBuffer structure as totalFramesProcessed, now time measure works again... but there is some problem with looping...
Also note some inconsistencies between frames and samples, it seems those concepts are not considered correctly in the code.
This commit is contained in:
Ray 2019-09-03 23:08:02 +02:00
parent dde1e0671f
commit fc95ceadf0
4 changed files with 132 additions and 169 deletions
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22
examples/audio/audio_module_playing.c
View file

@ -46,13 +46,13 @@ int main(void)
circles[i].radius = GetRandomValue(10, 40); circles[i].radius = GetRandomValue(10, 40);
circles[i].position.x = GetRandomValue(circles[i].radius, screenWidth - circles[i].radius); circles[i].position.x = GetRandomValue(circles[i].radius, screenWidth - circles[i].radius);
circles[i].position.y = GetRandomValue(circles[i].radius, screenHeight - circles[i].radius); circles[i].position.y = GetRandomValue(circles[i].radius, screenHeight - circles[i].radius);
circles[i].speed = (float)GetRandomValue(1, 100)/20000.0f; circles[i].speed = (float)GetRandomValue(1, 100)/2000.0f;
circles[i].color = colors[GetRandomValue(0, 13)]; circles[i].color = colors[GetRandomValue(0, 13)];
} }
Music xm = LoadMusicStream("resources/chiptun1.mod"); Music music = LoadMusicStream("resources/mini1111.xm");
PlayMusicStream(xm); PlayMusicStream(music);
float timePlayed = 0.0f; float timePlayed = 0.0f;
bool pause = false; bool pause = false;
@ -65,13 +65,13 @@ int main(void)
{ {
// Update // Update
//---------------------------------------------------------------------------------- //----------------------------------------------------------------------------------
UpdateMusicStream(xm); // Update music buffer with new stream data UpdateMusicStream(music); // Update music buffer with new stream data
// Restart music playing (stop and play) // Restart music playing (stop and play)
if (IsKeyPressed(KEY_SPACE)) if (IsKeyPressed(KEY_SPACE))
{ {
StopMusicStream(xm); StopMusicStream(music);
PlayMusicStream(xm); PlayMusicStream(music);
} }
// Pause/Resume music playing // Pause/Resume music playing
@ -79,12 +79,12 @@ int main(void)
{ {
pause = !pause; pause = !pause;
if (pause) PauseMusicStream(xm); if (pause) PauseMusicStream(music);
else ResumeMusicStream(xm); else ResumeMusicStream(music);
} }
// Get timePlayed scaled to bar dimensions // Get timePlayed scaled to bar dimensions
timePlayed = GetMusicTimePlayed(xm)/GetMusicTimeLength(xm)*(screenWidth - 40); timePlayed = GetMusicTimePlayed(music)/GetMusicTimeLength(music)*(screenWidth - 40);
// Color circles animation // Color circles animation
for (int i = MAX_CIRCLES - 1; (i >= 0) && !pause; i--) for (int i = MAX_CIRCLES - 1; (i >= 0) && !pause; i--)
@ -101,7 +101,7 @@ int main(void)
circles[i].position.x = GetRandomValue(circles[i].radius, screenWidth - circles[i].radius); circles[i].position.x = GetRandomValue(circles[i].radius, screenWidth - circles[i].radius);
circles[i].position.y = GetRandomValue(circles[i].radius, screenHeight - circles[i].radius); circles[i].position.y = GetRandomValue(circles[i].radius, screenHeight - circles[i].radius);
circles[i].color = colors[GetRandomValue(0, 13)]; circles[i].color = colors[GetRandomValue(0, 13)];
circles[i].speed = (float)GetRandomValue(1, 100)/20000.0f; circles[i].speed = (float)GetRandomValue(1, 100)/2000.0f;
} }
} }
//---------------------------------------------------------------------------------- //----------------------------------------------------------------------------------
@ -128,7 +128,7 @@ int main(void)
// De-Initialization // De-Initialization
//-------------------------------------------------------------------------------------- //--------------------------------------------------------------------------------------
UnloadMusicStream(xm); // Unload music stream buffers from RAM UnloadMusicStream(music); // Unload music stream buffers from RAM
CloseAudioDevice(); // Close audio device (music streaming is automatically stopped) CloseAudioDevice(); // Close audio device (music streaming is automatically stopped)

190
src/raudio.c
View file

@ -189,6 +189,8 @@ void TraceLog(int msgType, const char *text, ...); // Show trace lo
#define DEVICE_CHANNELS 2 #define DEVICE_CHANNELS 2
#define DEVICE_SAMPLE_RATE 44100 #define DEVICE_SAMPLE_RATE 44100
#define MAX_AUDIO_BUFFER_POOL_CHANNELS 16
typedef enum { AUDIO_BUFFER_USAGE_STATIC = 0, AUDIO_BUFFER_USAGE_STREAM } AudioBufferUsage; typedef enum { AUDIO_BUFFER_USAGE_STATIC = 0, AUDIO_BUFFER_USAGE_STREAM } AudioBufferUsage;
// Audio buffer structure // Audio buffer structure
@ -205,17 +207,23 @@ struct rAudioBuffer {
bool looping; // Audio buffer looping, always true for AudioStreams bool looping; // Audio buffer looping, always true for AudioStreams
int usage; // Audio buffer usage mode: STATIC or STREAM int usage; // Audio buffer usage mode: STATIC or STREAM
bool isSubBufferProcessed[2]; bool isSubBufferProcessed[2]; // SubBuffer processed (virtual double buffer)
unsigned int frameCursorPos; // Samples processed? unsigned int frameCursorPos; // Frame cursor position
unsigned int bufferSizeInFrames; unsigned int bufferSizeInFrames; // Total buffer size in frames
unsigned int totalFramesProcessed; // Total frames processed in this buffer (required for play timming)
rAudioBuffer *next; unsigned char *buffer; // Data buffer, on music stream keeps filling
rAudioBuffer *prev;
unsigned char *buffer; rAudioBuffer *next; // Next audio buffer on the list
rAudioBuffer *prev; // Previous audio buffer on the list
}; };
#define AudioBuffer rAudioBuffer // HACK: To avoid CoreAudio (macOS) symbol collision #define AudioBuffer rAudioBuffer // HACK: To avoid CoreAudio (macOS) symbol collision
// Audio buffers are tracked in a linked list
static AudioBuffer *firstAudioBuffer = NULL;
static AudioBuffer *lastAudioBuffer = NULL;
// miniaudio global variables // miniaudio global variables
static ma_context context; static ma_context context;
static ma_device device; static ma_device device;
@ -223,9 +231,10 @@ static ma_mutex audioLock;
static bool isAudioInitialized = false; static bool isAudioInitialized = false;
static float masterVolume = 1.0f; static float masterVolume = 1.0f;
// Audio buffers are tracked in a linked list // Multi channel playback global variables
static AudioBuffer *firstAudioBuffer = NULL; AudioBuffer *audioBufferPool[MAX_AUDIO_BUFFER_POOL_CHANNELS] = { 0 };
static AudioBuffer *lastAudioBuffer = NULL; unsigned int audioBufferPoolCounter = 0;
unsigned int audioBufferPoolChannels[MAX_AUDIO_BUFFER_POOL_CHANNELS] = { 0 };
// miniaudio functions declaration // miniaudio functions declaration
static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message); static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message);
@ -247,19 +256,9 @@ void SetAudioBufferPitch(AudioBuffer *buffer, float pitch);
void TrackAudioBuffer(AudioBuffer *buffer); void TrackAudioBuffer(AudioBuffer *buffer);
void UntrackAudioBuffer(AudioBuffer *buffer); void UntrackAudioBuffer(AudioBuffer *buffer);
//---------------------------------------------------------------------------------- //----------------------------------------------------------------------------------
// Multi channel playback globals // miniaudio functions definitions
//----------------------------------------------------------------------------------
// Number of channels in the audio pool
#define MAX_AUDIO_BUFFER_POOL_CHANNELS 16
// Audio buffer pool
AudioBuffer *audioBufferPool[MAX_AUDIO_BUFFER_POOL_CHANNELS] = { 0 };
// These are used to determine the oldest playing channel
unsigned long audioBufferPoolCounter = 0;
unsigned long audioBufferPoolChannels[MAX_AUDIO_BUFFER_POOL_CHANNELS] = { 0 };
//---------------------------------------------------------------------------------- //----------------------------------------------------------------------------------
// Log callback function // Log callback function
@ -319,6 +318,7 @@ static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const
{ {
float *framesOut = (float *)pFramesOut + (framesRead*device.playback.channels); float *framesOut = (float *)pFramesOut + (framesRead*device.playback.channels);
float *framesIn = tempBuffer; float *framesIn = tempBuffer;
MixAudioFrames(framesOut, framesIn, framesJustRead, audioBuffer->volume); MixAudioFrames(framesOut, framesIn, framesJustRead, audioBuffer->volume);
framesToRead -= framesJustRead; framesToRead -= framesJustRead;
@ -485,7 +485,6 @@ void InitAudioDevice(void)
{ {
// Init audio context // Init audio context
ma_context_config contextConfig = ma_context_config_init(); ma_context_config contextConfig = ma_context_config_init();
contextConfig.logCallback = OnLog; contextConfig.logCallback = OnLog;
ma_result result = ma_context_init(NULL, 0, &contextConfig, &context); ma_result result = ma_context_init(NULL, 0, &contextConfig, &context);
@ -553,11 +552,7 @@ void InitAudioDevice(void)
// Close the audio device for all contexts // Close the audio device for all contexts
void CloseAudioDevice(void) void CloseAudioDevice(void)
{ {
if (!isAudioInitialized) if (isAudioInitialized)
{
TraceLog(LOG_WARNING, "Could not close audio device because it is not currently initialized");
}
else
{ {
ma_mutex_uninit(&audioLock); ma_mutex_uninit(&audioLock);
ma_device_uninit(&device); ma_device_uninit(&device);
@ -567,6 +562,7 @@ void CloseAudioDevice(void)
TraceLog(LOG_INFO, "Audio device closed successfully"); TraceLog(LOG_INFO, "Audio device closed successfully");
} }
else TraceLog(LOG_WARNING, "Could not close audio device because it is not currently initialized");
} }
// Check if device has been initialized successfully // Check if device has been initialized successfully
@ -588,11 +584,11 @@ void SetMasterVolume(float volume)
// Module Functions Definition - Audio Buffer management // Module Functions Definition - Audio Buffer management
//---------------------------------------------------------------------------------- //----------------------------------------------------------------------------------
// Create a new audio buffer. Initially filled with silence // Initialize a new audio buffer (filled with silence)
AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 bufferSizeInFrames, int usage) AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 bufferSizeInFrames, int usage)
{ {
AudioBuffer *audioBuffer = (AudioBuffer *)RL_CALLOC(1, sizeof(AudioBuffer)); AudioBuffer *audioBuffer = (AudioBuffer *)RL_CALLOC(1, sizeof(AudioBuffer));
audioBuffer->buffer = RL_CALLOC((bufferSizeInFrames*channels*ma_get_bytes_per_sample(format)), 1); audioBuffer->buffer = RL_CALLOC(bufferSizeInFrames*channels*ma_get_bytes_per_sample(format), 1);
if (audioBuffer == NULL) if (audioBuffer == NULL)
{ {
@ -690,6 +686,7 @@ void StopAudioBuffer(AudioBuffer *buffer)
buffer->playing = false; buffer->playing = false;
buffer->paused = false; buffer->paused = false;
buffer->frameCursorPos = 0; buffer->frameCursorPos = 0;
buffer->totalFramesProcessed = 0;
buffer->isSubBufferProcessed[0] = true; buffer->isSubBufferProcessed[0] = true;
buffer->isSubBufferProcessed[1] = true; buffer->isSubBufferProcessed[1] = true;
} }
@ -725,8 +722,10 @@ void SetAudioBufferPitch(AudioBuffer *buffer, float pitch)
{ {
float pitchMul = pitch/buffer->pitch; float pitchMul = pitch/buffer->pitch;
// Pitching is just an adjustment of the sample rate. Note that this changes the duration of the sound - higher pitches // Pitching is just an adjustment of the sample rate.
// will make the sound faster; lower pitches make it slower. // Note that this changes the duration of the sound:
// - higher pitches will make the sound faster
// - lower pitches make it slower
ma_uint32 newOutputSampleRate = (ma_uint32)((float)buffer->dsp.src.config.sampleRateOut/pitchMul); ma_uint32 newOutputSampleRate = (ma_uint32)((float)buffer->dsp.src.config.sampleRateOut/pitchMul);
buffer->pitch *= (float)buffer->dsp.src.config.sampleRateOut/newOutputSampleRate; buffer->pitch *= (float)buffer->dsp.src.config.sampleRateOut/newOutputSampleRate;
@ -869,17 +868,15 @@ void UpdateSound(Sound sound, const void *data, int samplesCount)
{ {
AudioBuffer *audioBuffer = sound.stream.buffer; AudioBuffer *audioBuffer = sound.stream.buffer;
if (audioBuffer == NULL) if (audioBuffer != NULL)
{ {
TraceLog(LOG_ERROR, "UpdateSound() : Invalid sound - no audio buffer");
return;
}
StopAudioBuffer(audioBuffer); StopAudioBuffer(audioBuffer);
// TODO: May want to lock/unlock this since this data buffer is read at mixing time // TODO: May want to lock/unlock this since this data buffer is read at mixing time
memcpy(audioBuffer->buffer, data, samplesCount*audioBuffer->dsp.formatConverterIn.config.channels*ma_get_bytes_per_sample(audioBuffer->dsp.formatConverterIn.config.formatIn)); memcpy(audioBuffer->buffer, data, samplesCount*audioBuffer->dsp.formatConverterIn.config.channels*ma_get_bytes_per_sample(audioBuffer->dsp.formatConverterIn.config.formatIn));
} }
else TraceLog(LOG_ERROR, "UpdateSound() : Invalid sound - no audio buffer");
}
// Export wave data to file // Export wave data to file
void ExportWave(Wave wave, const char *fileName) void ExportWave(Wave wave, const char *fileName)
@ -956,7 +953,7 @@ void PlaySound(Sound sound)
void PlaySoundMulti(Sound sound) void PlaySoundMulti(Sound sound)
{ {
int index = -1; int index = -1;
unsigned long oldAge = 0; unsigned int oldAge = 0;
int oldIndex = -1; int oldIndex = -1;
// find the first non playing pool entry // find the first non playing pool entry
@ -1184,14 +1181,8 @@ Music LoadMusicStream(const char *fileName)
// OGG bit rate defaults to 16 bit, it's enough for compressed format // OGG bit rate defaults to 16 bit, it's enough for compressed format
music.stream = InitAudioStream(info.sample_rate, 16, info.channels); music.stream = InitAudioStream(info.sample_rate, 16, info.channels);
music.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData)*info.channels; music.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData)*info.channels;
music.sampleLeft = music.sampleCount;
music.loopCount = 0; // Infinite loop by default music.loopCount = 0; // Infinite loop by default
musicLoaded = true; musicLoaded = true;
TraceLog(LOG_INFO, "[%s] OGG total samples: %i", fileName, music.sampleCount);
TraceLog(LOG_INFO, "[%s] OGG sample rate: %i", fileName, info.sample_rate);
TraceLog(LOG_INFO, "[%s] OGG channels: %i", fileName, info.channels);
TraceLog(LOG_INFO, "[%s] OGG memory required: %i", fileName, info.temp_memory_required);
} }
} }
#endif #endif
@ -1207,14 +1198,8 @@ Music LoadMusicStream(const char *fileName)
music.stream = InitAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels); music.stream = InitAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
music.sampleCount = (unsigned int)ctxFlac->totalSampleCount; music.sampleCount = (unsigned int)ctxFlac->totalSampleCount;
music.sampleLeft = music.sampleCount;
music.loopCount = 0; // Infinite loop by default music.loopCount = 0; // Infinite loop by default
musicLoaded = true; musicLoaded = true;
TraceLog(LOG_DEBUG, "[%s] FLAC total samples: %i", fileName, music.sampleCount);
TraceLog(LOG_DEBUG, "[%s] FLAC sample rate: %i", fileName, ctxFlac->sampleRate);
TraceLog(LOG_DEBUG, "[%s] FLAC bits per sample: %i", fileName, ctxFlac->bitsPerSample);
TraceLog(LOG_DEBUG, "[%s] FLAC channels: %i", fileName, ctxFlac->channels);
} }
} }
#endif #endif
@ -1232,14 +1217,8 @@ Music LoadMusicStream(const char *fileName)
music.stream = InitAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels); music.stream = InitAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels);
music.sampleCount = drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels; music.sampleCount = drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels;
music.sampleLeft = music.sampleCount;
music.loopCount = 0; // Infinite loop by default music.loopCount = 0; // Infinite loop by default
musicLoaded = true; musicLoaded = true;
TraceLog(LOG_INFO, "[%s] MP3 sample rate: %i", fileName, ctxMp3->sampleRate);
TraceLog(LOG_INFO, "[%s] MP3 bits per sample: %i", fileName, 32);
TraceLog(LOG_INFO, "[%s] MP3 channels: %i", fileName, ctxMp3->channels);
TraceLog(LOG_INFO, "[%s] MP3 total samples: %i", fileName, music.sampleCount);
} }
} }
#endif #endif
@ -1258,14 +1237,10 @@ Music LoadMusicStream(const char *fileName)
// NOTE: Only stereo is supported for XM // NOTE: Only stereo is supported for XM
music.stream = InitAudioStream(48000, 16, 2); music.stream = InitAudioStream(48000, 16, 2);
music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm); music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm);
music.sampleLeft = music.sampleCount;
music.loopCount = 0; // Infinite loop by default music.loopCount = 0; // Infinite loop by default
musicLoaded = true; musicLoaded = true;
music.ctxData = ctxXm; music.ctxData = ctxXm;
TraceLog(LOG_INFO, "[%s] XM number of samples: %i", fileName, music.sampleCount);
TraceLog(LOG_INFO, "[%s] XM track length: %11.6f sec", fileName, (float)music.sampleCount/48000.0f);
} }
} }
#endif #endif
@ -1285,12 +1260,8 @@ Music LoadMusicStream(const char *fileName)
// NOTE: Only stereo is supported for MOD // NOTE: Only stereo is supported for MOD
music.stream = InitAudioStream(48000, 16, 2); music.stream = InitAudioStream(48000, 16, 2);
music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod); music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod);
music.sampleLeft = music.sampleCount;
music.loopCount = 0; // Infinite loop by default music.loopCount = 0; // Infinite loop by default
musicLoaded = true; musicLoaded = true;
TraceLog(LOG_INFO, "[%s] MOD number of samples: %i", fileName, music.sampleCount);
TraceLog(LOG_INFO, "[%s] MOD track length: %11.6f sec", fileName, (float)music.sampleCount/48000.0f);
} }
} }
#endif #endif
@ -1316,6 +1287,15 @@ Music LoadMusicStream(const char *fileName)
TraceLog(LOG_WARNING, "[%s] Music file could not be opened", fileName); TraceLog(LOG_WARNING, "[%s] Music file could not be opened", fileName);
} }
else
{
// Show some music stream info
TraceLog(LOG_INFO, "[%s] Music file successfully loaded:", fileName);
TraceLog(LOG_INFO, " Total samples: %i", music.sampleCount);
TraceLog(LOG_INFO, " Sample rate: %i Hz", music.stream.sampleRate);
TraceLog(LOG_INFO, " Sample size: %i bits", music.stream.sampleSize);
TraceLog(LOG_INFO, " Channels: %i (%s)", music.stream.channels, (music.stream.channels == 1)? "Mono" : (music.stream.channels == 2)? "Stereo" : "Multi");
}
return music; return music;
} }
@ -1348,22 +1328,19 @@ void PlayMusicStream(Music music)
{ {
AudioBuffer *audioBuffer = music.stream.buffer; AudioBuffer *audioBuffer = music.stream.buffer;
if (audioBuffer == NULL) if (audioBuffer != NULL)
{ {
TraceLog(LOG_ERROR, "PlayMusicStream() : No audio buffer");
return;
}
// For music streams, we need to make sure we maintain the frame cursor position // For music streams, we need to make sure we maintain the frame cursor position
// This is a hack for this section of code in UpdateMusicStream() // This is a hack for this section of code in UpdateMusicStream()
// NOTE: In case window is minimized, music stream is stopped, just make sure to // NOTE: In case window is minimized, music stream is stopped, just make sure to
// play again on window restore: if (IsMusicPlaying(music)) PlayMusicStream(music); // play again on window restore: if (IsMusicPlaying(music)) PlayMusicStream(music);
ma_uint32 frameCursorPos = audioBuffer->frameCursorPos; ma_uint32 frameCursorPos = audioBuffer->frameCursorPos;
PlayAudioStream(music.stream); // WARNING: This resets the cursor position.
PlayAudioStream(music.stream); // <-- This resets the cursor position.
audioBuffer->frameCursorPos = frameCursorPos; audioBuffer->frameCursorPos = frameCursorPos;
} }
else TraceLog(LOG_ERROR, "PlayMusicStream() : No audio buffer");
}
// Pause music playing // Pause music playing
void PauseMusicStream(Music music) void PauseMusicStream(Music music)
@ -1389,7 +1366,7 @@ void StopMusicStream(Music music)
case MUSIC_AUDIO_OGG: stb_vorbis_seek_start((stb_vorbis *)music.ctxData); break; case MUSIC_AUDIO_OGG: stb_vorbis_seek_start((stb_vorbis *)music.ctxData); break;
#endif #endif
#if defined(SUPPORT_FILEFORMAT_FLAC) #if defined(SUPPORT_FILEFORMAT_FLAC)
case MUSIC_AUDIO_FLAC: /* TODO: Restart FLAC context */ break; case MUSIC_AUDIO_FLAC: drflac_seek_to_sample((drflac *)music.ctxData, 0) break;
#endif #endif
#if defined(SUPPORT_FILEFORMAT_MP3) #if defined(SUPPORT_FILEFORMAT_MP3)
case MUSIC_AUDIO_MP3: drmp3_seek_to_pcm_frame((drmp3 *)music.ctxData, 0); break; case MUSIC_AUDIO_MP3: drmp3_seek_to_pcm_frame((drmp3 *)music.ctxData, 0); break;
@ -1402,8 +1379,6 @@ void StopMusicStream(Music music)
#endif #endif
default: break; default: break;
} }
music.sampleLeft = music.sampleCount;
} }
// Update (re-fill) music buffers if data already processed // Update (re-fill) music buffers if data already processed
@ -1416,12 +1391,16 @@ void UpdateMusicStream(Music music)
// NOTE: Using dynamic allocation because it could require more than 16KB // NOTE: Using dynamic allocation because it could require more than 16KB
void *pcm = RL_CALLOC(subBufferSizeInFrames*music.stream.channels*music.stream.sampleSize/8, 1); void *pcm = RL_CALLOC(subBufferSizeInFrames*music.stream.channels*music.stream.sampleSize/8, 1);
int samplesCount = 0; // Total size of data steamed in L+R samples for xm floats, individual L or R for ogg shorts int samplesCount = 0; // Total size of data streamed in L+R samples for xm floats, individual L or R for ogg shorts
// TODO: Get the sampleLeft using totalFramesProcessed... but first, get total frames processed correctly...
//ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
int sampleLeft = music.sampleCount - (music.stream.buffer->totalFramesProcessed*music.stream.channels);
while (IsAudioStreamProcessed(music.stream)) while (IsAudioStreamProcessed(music.stream))
{ {
if ((music.sampleLeft/music.stream.channels) >= subBufferSizeInFrames) samplesCount = subBufferSizeInFrames*music.stream.channels; if ((sampleLeft/music.stream.channels) >= subBufferSizeInFrames) samplesCount = subBufferSizeInFrames*music.stream.channels;
else samplesCount = music.sampleLeft; else samplesCount = sampleLeft;
switch (music.ctxType) switch (music.ctxType)
{ {
@ -1470,12 +1449,12 @@ void UpdateMusicStream(Music music)
if ((music.ctxType == MUSIC_MODULE_XM) || (music.ctxType == MUSIC_MODULE_MOD)) if ((music.ctxType == MUSIC_MODULE_XM) || (music.ctxType == MUSIC_MODULE_MOD))
{ {
if (samplesCount > 1) music.sampleLeft -= samplesCount/2; if (samplesCount > 1) sampleLeft -= samplesCount/2;
else music.sampleLeft -= samplesCount; else sampleLeft -= samplesCount;
} }
else music.sampleLeft -= samplesCount; else sampleLeft -= samplesCount;
if (music.sampleLeft <= 0) if (sampleLeft <= 0)
{ {
streamEnding = true; streamEnding = true;
break; break;
@ -1496,10 +1475,7 @@ void UpdateMusicStream(Music music)
music.loopCount--; // Decrease loop count music.loopCount--; // Decrease loop count
PlayMusicStream(music); // Play again PlayMusicStream(music); // Play again
} }
else else if (music.loopCount == 0) PlayMusicStream(music);
{
if (music.loopCount == 0) PlayMusicStream(music);
}
} }
else else
{ {
@ -1549,7 +1525,8 @@ float GetMusicTimePlayed(Music music)
{ {
float secondsPlayed = 0.0f; float secondsPlayed = 0.0f;
unsigned int samplesPlayed = music.sampleCount - music.sampleLeft; //ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
unsigned int samplesPlayed = music.stream.buffer->totalFramesProcessed*music.stream.channels;
secondsPlayed = (float)samplesPlayed/(music.stream.sampleRate*music.stream.channels); secondsPlayed = (float)samplesPlayed/(music.stream.sampleRate*music.stream.channels);
return secondsPlayed; return secondsPlayed;
@ -1562,14 +1539,7 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un
stream.sampleRate = sampleRate; stream.sampleRate = sampleRate;
stream.sampleSize = sampleSize; stream.sampleSize = sampleSize;
stream.channels = channels;
// Only mono and stereo channels are supported
if ((channels > 0) && (channels < 3)) stream.channels = channels;
else
{
TraceLog(LOG_WARNING, "Init audio stream: Number of channels not supported: %i", channels);
stream.channels = 1; // Fallback to mono channel
}
ma_format formatIn = ((stream.sampleSize == 8)? ma_format_u8 : ((stream.sampleSize == 16)? ma_format_s16 : ma_format_f32)); ma_format formatIn = ((stream.sampleSize == 8)? ma_format_u8 : ((stream.sampleSize == 16)? ma_format_s16 : ma_format_f32));
@ -1579,18 +1549,14 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un
if (subBufferSize < periodSize) subBufferSize = periodSize; if (subBufferSize < periodSize) subBufferSize = periodSize;
AudioBuffer *audioBuffer = InitAudioBuffer(formatIn, stream.channels, stream.sampleRate, subBufferSize*2, AUDIO_BUFFER_USAGE_STREAM); stream.buffer = InitAudioBuffer(formatIn, stream.channels, stream.sampleRate, subBufferSize*2, AUDIO_BUFFER_USAGE_STREAM);
if (audioBuffer == NULL) if (stream.buffer != NULL)
{ {
TraceLog(LOG_ERROR, "InitAudioStream() : Failed to create audio buffer"); stream.buffer->looping = true; // Always loop for streaming buffers
return stream;
}
audioBuffer->looping = true; // Always loop for streaming buffers
stream.buffer = audioBuffer;
TraceLog(LOG_INFO, "Audio stream loaded successfully (%i Hz, %i bit, %s)", stream.sampleRate, stream.sampleSize, (stream.channels == 1)? "Mono" : "Stereo"); TraceLog(LOG_INFO, "Audio stream loaded successfully (%i Hz, %i bit, %s)", stream.sampleRate, stream.sampleSize, (stream.channels == 1)? "Mono" : "Stereo");
}
else TraceLog(LOG_ERROR, "InitAudioStream() : Failed to create audio buffer");
return stream; return stream;
} }
@ -1610,12 +1576,8 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
{ {
AudioBuffer *audioBuffer = stream.buffer; AudioBuffer *audioBuffer = stream.buffer;
if (audioBuffer == NULL) if (audioBuffer != NULL)
{ {
TraceLog(LOG_ERROR, "UpdateAudioStream() : No audio buffer");
return;
}
if (audioBuffer->isSubBufferProcessed[0] || audioBuffer->isSubBufferProcessed[1]) if (audioBuffer->isSubBufferProcessed[0] || audioBuffer->isSubBufferProcessed[1])
{ {
ma_uint32 subBufferToUpdate = 0; ma_uint32 subBufferToUpdate = 0;
@ -1636,6 +1598,9 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
ma_uint32 subBufferSizeInFrames = audioBuffer->bufferSizeInFrames/2; ma_uint32 subBufferSizeInFrames = audioBuffer->bufferSizeInFrames/2;
unsigned char *subBuffer = audioBuffer->buffer + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate); unsigned char *subBuffer = audioBuffer->buffer + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate);
// TODO: Get total frames processed on this buffer... DOES NOT WORK.
audioBuffer->totalFramesProcessed += subBufferSizeInFrames;
// Does this API expect a whole buffer to be updated in one go? // Does this API expect a whole buffer to be updated in one go?
// Assuming so, but if not will need to change this logic. // Assuming so, but if not will need to change this logic.
if (subBufferSizeInFrames >= (ma_uint32)samplesCount/stream.channels) if (subBufferSizeInFrames >= (ma_uint32)samplesCount/stream.channels)
@ -1650,16 +1615,15 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
// Any leftover frames should be filled with zeros. // Any leftover frames should be filled with zeros.
ma_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite; ma_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite;
if (leftoverFrameCount > 0) if (leftoverFrameCount > 0) memset(subBuffer + bytesToWrite, 0, leftoverFrameCount*stream.channels*(stream.sampleSize/8));
{
memset(subBuffer + bytesToWrite, 0, leftoverFrameCount*stream.channels*(stream.sampleSize/8));
}
audioBuffer->isSubBufferProcessed[subBufferToUpdate] = false; audioBuffer->isSubBufferProcessed[subBufferToUpdate] = false;
} }
else TraceLog(LOG_ERROR, "UpdateAudioStream() : Attempting to write too many frames to buffer"); else TraceLog(LOG_ERROR, "UpdateAudioStream() : Attempting to write too many frames to buffer");
} }
else TraceLog(LOG_ERROR, "Audio buffer not available for updating"); else TraceLog(LOG_ERROR, "UpdateAudioStream() : Audio buffer not available for updating");
}
else TraceLog(LOG_ERROR, "UpdateAudioStream() : No audio buffer");
} }
// Check if any audio stream buffers requires refill // Check if any audio stream buffers requires refill

4
src/raudio.h
View file

@ -1,6 +1,6 @@
/********************************************************************************************** /**********************************************************************************************
* *
* raudio - A simple and easy-to-use audio library based on mini_al * raudio - A simple and easy-to-use audio library based on miniaudio
* *
* FEATURES: * FEATURES:
* - Manage audio device (init/close) * - Manage audio device (init/close)
@ -20,7 +20,7 @@
* *
* CONTRIBUTORS: * CONTRIBUTORS:
* David Reid (github: @mackron) (Nov. 2017): * David Reid (github: @mackron) (Nov. 2017):
* - Complete port to mini_al library * - Complete port to miniaudio library
* *
* Joshua Reisenauer (github: @kd7tck) (2015) * Joshua Reisenauer (github: @kd7tck) (2015)
* - XM audio module support (jar_xm) * - XM audio module support (jar_xm)

1
src/raylib.h
View file

@ -434,7 +434,6 @@ typedef struct Music {
void *ctxData; // Audio context data, depends on type void *ctxData; // Audio context data, depends on type
unsigned int sampleCount; // Total number of samples unsigned int sampleCount; // Total number of samples
unsigned int sampleLeft; // Number of samples left to end
unsigned int loopCount; // Loops count (times music will play), 0 means infinite loop unsigned int loopCount; // Loops count (times music will play), 0 means infinite loop
AudioStream stream; // Audio stream AudioStream stream; // Audio stream