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-rw-r--r--src/raudio.c266
-rw-r--r--src/raudio.h4
-rw-r--r--src/raylib.h1
3 files changed, 117 insertions, 154 deletions
diff --git a/src/raudio.c b/src/raudio.c
index 43e63bc9..2edf660a 100644
--- a/src/raudio.c
+++ b/src/raudio.c
@@ -124,7 +124,7 @@
// After some math, considering a sampleRate of 48000, a buffer refill rate of 1/60 seconds and a
// standard double-buffering system, a 4096 samples buffer has been chosen, it should be enough
// In case of music-stalls, just increase this number
-#define AUDIO_BUFFER_SIZE 4096 // PCM data samples (i.e. 16bit, Mono: 8Kb)
+#define AUDIO_BUFFER_SIZE 4096 // PCM data samples (i.e. 16bit, Mono: 8Kb)
//----------------------------------------------------------------------------------
// Types and Structures Definition
@@ -189,6 +189,8 @@ void TraceLog(int msgType, const char *text, ...); // Show trace lo
#define DEVICE_CHANNELS 2
#define DEVICE_SAMPLE_RATE 44100
+#define MAX_AUDIO_BUFFER_POOL_CHANNELS 16
+
typedef enum { AUDIO_BUFFER_USAGE_STATIC = 0, AUDIO_BUFFER_USAGE_STREAM } AudioBufferUsage;
// Audio buffer structure
@@ -205,16 +207,22 @@ struct rAudioBuffer {
bool looping; // Audio buffer looping, always true for AudioStreams
int usage; // Audio buffer usage mode: STATIC or STREAM
- bool isSubBufferProcessed[2];
- unsigned int frameCursorPos; // Samples processed?
- unsigned int bufferSizeInFrames;
+ bool isSubBufferProcessed[2]; // SubBuffer processed (virtual double buffer)
+ unsigned int frameCursorPos; // Frame cursor position
+ unsigned int bufferSizeInFrames; // Total buffer size in frames
+ unsigned int totalFramesProcessed; // Total frames processed in this buffer (required for play timming)
- rAudioBuffer *next;
- rAudioBuffer *prev;
- unsigned char *buffer;
+ unsigned char *buffer; // Data buffer, on music stream keeps filling
+
+ 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
static ma_context context;
@@ -223,9 +231,10 @@ static ma_mutex audioLock;
static bool isAudioInitialized = false;
static float masterVolume = 1.0f;
-// Audio buffers are tracked in a linked list
-static AudioBuffer *firstAudioBuffer = NULL;
-static AudioBuffer *lastAudioBuffer = NULL;
+// Multi channel playback global variables
+AudioBuffer *audioBufferPool[MAX_AUDIO_BUFFER_POOL_CHANNELS] = { 0 };
+unsigned int audioBufferPoolCounter = 0;
+unsigned int audioBufferPoolChannels[MAX_AUDIO_BUFFER_POOL_CHANNELS] = { 0 };
// miniaudio functions declaration
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 UntrackAudioBuffer(AudioBuffer *buffer);
-//----------------------------------------------------------------------------------
-// Multi channel playback globals
-//----------------------------------------------------------------------------------
-// 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 };
+//----------------------------------------------------------------------------------
+// miniaudio functions definitions
//----------------------------------------------------------------------------------
// 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 *framesIn = tempBuffer;
+
MixAudioFrames(framesOut, framesIn, framesJustRead, audioBuffer->volume);
framesToRead -= framesJustRead;
@@ -485,7 +485,6 @@ void InitAudioDevice(void)
{
// Init audio context
ma_context_config contextConfig = ma_context_config_init();
-
contextConfig.logCallback = OnLog;
ma_result result = ma_context_init(NULL, 0, &contextConfig, &context);
@@ -553,11 +552,7 @@ void InitAudioDevice(void)
// Close the audio device for all contexts
void CloseAudioDevice(void)
{
- if (!isAudioInitialized)
- {
- TraceLog(LOG_WARNING, "Could not close audio device because it is not currently initialized");
- }
- else
+ if (isAudioInitialized)
{
ma_mutex_uninit(&audioLock);
ma_device_uninit(&device);
@@ -567,6 +562,7 @@ void CloseAudioDevice(void)
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
@@ -588,11 +584,11 @@ void SetMasterVolume(float volume)
// 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 *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)
{
@@ -690,6 +686,7 @@ void StopAudioBuffer(AudioBuffer *buffer)
buffer->playing = false;
buffer->paused = false;
buffer->frameCursorPos = 0;
+ buffer->totalFramesProcessed = 0;
buffer->isSubBufferProcessed[0] = true;
buffer->isSubBufferProcessed[1] = true;
}
@@ -725,8 +722,10 @@ void SetAudioBufferPitch(AudioBuffer *buffer, float 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
- // will make the sound faster; lower pitches make it slower.
+ // Pitching is just an adjustment of the sample rate.
+ // 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);
buffer->pitch *= (float)buffer->dsp.src.config.sampleRateOut/newOutputSampleRate;
@@ -869,16 +868,14 @@ void UpdateSound(Sound sound, const void *data, int samplesCount)
{
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
- memcpy(audioBuffer->buffer, data, samplesCount*audioBuffer->dsp.formatConverterIn.config.channels*ma_get_bytes_per_sample(audioBuffer->dsp.formatConverterIn.config.formatIn));
+ // 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));
+ }
+ else TraceLog(LOG_ERROR, "UpdateSound() : Invalid sound - no audio buffer");
}
// Export wave data to file
@@ -956,7 +953,7 @@ void PlaySound(Sound sound)
void PlaySoundMulti(Sound sound)
{
int index = -1;
- unsigned long oldAge = 0;
+ unsigned int oldAge = 0;
int oldIndex = -1;
// 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
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.sampleLeft = music.sampleCount;
music.loopCount = 0; // Infinite loop by default
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
@@ -1207,14 +1198,8 @@ Music LoadMusicStream(const char *fileName)
music.stream = InitAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
music.sampleCount = (unsigned int)ctxFlac->totalSampleCount;
- music.sampleLeft = music.sampleCount;
music.loopCount = 0; // Infinite loop by default
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
@@ -1232,14 +1217,8 @@ Music LoadMusicStream(const char *fileName)
music.stream = InitAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels);
music.sampleCount = drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels;
- music.sampleLeft = music.sampleCount;
music.loopCount = 0; // Infinite loop by default
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
@@ -1258,14 +1237,10 @@ Music LoadMusicStream(const char *fileName)
// NOTE: Only stereo is supported for XM
music.stream = InitAudioStream(48000, 16, 2);
music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm);
- music.sampleLeft = music.sampleCount;
music.loopCount = 0; // Infinite loop by default
musicLoaded = true;
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
@@ -1285,12 +1260,8 @@ Music LoadMusicStream(const char *fileName)
// NOTE: Only stereo is supported for MOD
music.stream = InitAudioStream(48000, 16, 2);
music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod);
- music.sampleLeft = music.sampleCount;
music.loopCount = 0; // Infinite loop by default
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
@@ -1316,6 +1287,15 @@ Music LoadMusicStream(const char *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;
}
@@ -1348,21 +1328,18 @@ void PlayMusicStream(Music music)
{
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
+ // 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
+ // play again on window restore: if (IsMusicPlaying(music)) PlayMusicStream(music);
+ ma_uint32 frameCursorPos = audioBuffer->frameCursorPos;
+ PlayAudioStream(music.stream); // WARNING: This resets the cursor position.
+ audioBuffer->frameCursorPos = frameCursorPos;
}
+ else TraceLog(LOG_ERROR, "PlayMusicStream() : No audio buffer");
- // 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()
- // NOTE: In case window is minimized, music stream is stopped, just make sure to
- // play again on window restore: if (IsMusicPlaying(music)) PlayMusicStream(music);
- ma_uint32 frameCursorPos = audioBuffer->frameCursorPos;
-
- PlayAudioStream(music.stream); // <-- This resets the cursor position.
-
- audioBuffer->frameCursorPos = frameCursorPos;
}
// Pause music playing
@@ -1389,7 +1366,7 @@ void StopMusicStream(Music music)
case MUSIC_AUDIO_OGG: stb_vorbis_seek_start((stb_vorbis *)music.ctxData); break;
#endif
#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
#if defined(SUPPORT_FILEFORMAT_MP3)
case MUSIC_AUDIO_MP3: drmp3_seek_to_pcm_frame((drmp3 *)music.ctxData, 0); break;
@@ -1402,8 +1379,6 @@ void StopMusicStream(Music music)
#endif
default: break;
}
-
- music.sampleLeft = music.sampleCount;
}
// 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
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))
{
- if ((music.sampleLeft/music.stream.channels) >= subBufferSizeInFrames) samplesCount = subBufferSizeInFrames*music.stream.channels;
- else samplesCount = music.sampleLeft;
+ if ((sampleLeft/music.stream.channels) >= subBufferSizeInFrames) samplesCount = subBufferSizeInFrames*music.stream.channels;
+ else samplesCount = sampleLeft;
switch (music.ctxType)
{
@@ -1470,12 +1449,12 @@ void UpdateMusicStream(Music music)
if ((music.ctxType == MUSIC_MODULE_XM) || (music.ctxType == MUSIC_MODULE_MOD))
{
- if (samplesCount > 1) music.sampleLeft -= samplesCount/2;
- else music.sampleLeft -= samplesCount;
+ if (samplesCount > 1) sampleLeft -= samplesCount/2;
+ else sampleLeft -= samplesCount;
}
- else music.sampleLeft -= samplesCount;
+ else sampleLeft -= samplesCount;
- if (music.sampleLeft <= 0)
+ if (sampleLeft <= 0)
{
streamEnding = true;
break;
@@ -1493,13 +1472,10 @@ void UpdateMusicStream(Music music)
// Decrease loopCount to stop when required
if (music.loopCount > 1)
{
- music.loopCount--; // Decrease loop count
+ music.loopCount--; // Decrease loop count
PlayMusicStream(music); // Play again
}
- else
- {
- if (music.loopCount == 0) PlayMusicStream(music);
- }
+ else if (music.loopCount == 0) PlayMusicStream(music);
}
else
{
@@ -1549,7 +1525,8 @@ float GetMusicTimePlayed(Music music)
{
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);
return secondsPlayed;
@@ -1562,14 +1539,7 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un
stream.sampleRate = sampleRate;
stream.sampleSize = sampleSize;
-
- // 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
- }
+ stream.channels = channels;
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;
- 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");
- return stream;
+ stream.buffer->looping = true; // Always loop for streaming buffers
+ TraceLog(LOG_INFO, "Audio stream loaded successfully (%i Hz, %i bit, %s)", stream.sampleRate, stream.sampleSize, (stream.channels == 1)? "Mono" : "Stereo");
}
-
- 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");
+ else TraceLog(LOG_ERROR, "InitAudioStream() : Failed to create audio buffer");
return stream;
}
@@ -1610,56 +1576,54 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
{
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])
+ {
+ ma_uint32 subBufferToUpdate = 0;
- if (audioBuffer->isSubBufferProcessed[0] || audioBuffer->isSubBufferProcessed[1])
- {
- ma_uint32 subBufferToUpdate = 0;
+ if (audioBuffer->isSubBufferProcessed[0] && audioBuffer->isSubBufferProcessed[1])
+ {
+ // Both buffers are available for updating.
+ // Update the first one and make sure the cursor is moved back to the front.
+ subBufferToUpdate = 0;
+ audioBuffer->frameCursorPos = 0;
+ }
+ else
+ {
+ // Just update whichever sub-buffer is processed.
+ subBufferToUpdate = (audioBuffer->isSubBufferProcessed[0])? 0 : 1;
+ }
- if (audioBuffer->isSubBufferProcessed[0] && audioBuffer->isSubBufferProcessed[1])
- {
- // Both buffers are available for updating.
- // Update the first one and make sure the cursor is moved back to the front.
- subBufferToUpdate = 0;
- audioBuffer->frameCursorPos = 0;
- }
- else
- {
- // Just update whichever sub-buffer is processed.
- subBufferToUpdate = (audioBuffer->isSubBufferProcessed[0])? 0 : 1;
- }
+ ma_uint32 subBufferSizeInFrames = audioBuffer->bufferSizeInFrames/2;
+ unsigned char *subBuffer = audioBuffer->buffer + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate);
- ma_uint32 subBufferSizeInFrames = audioBuffer->bufferSizeInFrames/2;
- 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?
- // Assuming so, but if not will need to change this logic.
- if (subBufferSizeInFrames >= (ma_uint32)samplesCount/stream.channels)
- {
- ma_uint32 framesToWrite = subBufferSizeInFrames;
+ // Does this API expect a whole buffer to be updated in one go?
+ // Assuming so, but if not will need to change this logic.
+ if (subBufferSizeInFrames >= (ma_uint32)samplesCount/stream.channels)
+ {
+ ma_uint32 framesToWrite = subBufferSizeInFrames;
- if (framesToWrite > ((ma_uint32)samplesCount/stream.channels)) framesToWrite = (ma_uint32)samplesCount/stream.channels;
+ if (framesToWrite > ((ma_uint32)samplesCount/stream.channels)) framesToWrite = (ma_uint32)samplesCount/stream.channels;
- ma_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8);
- memcpy(subBuffer, data, bytesToWrite);
+ ma_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8);
+ memcpy(subBuffer, data, bytesToWrite);
- // Any leftover frames should be filled with zeros.
- ma_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite;
+ // Any leftover frames should be filled with zeros.
+ ma_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite;
- if (leftoverFrameCount > 0)
- {
- memset(subBuffer + bytesToWrite, 0, leftoverFrameCount*stream.channels*(stream.sampleSize/8));
- }
+ if (leftoverFrameCount > 0) 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, "UpdateAudioStream() : Audio buffer not available for updating");
}
- else TraceLog(LOG_ERROR, "Audio buffer not available for updating");
+ else TraceLog(LOG_ERROR, "UpdateAudioStream() : No audio buffer");
}
// Check if any audio stream buffers requires refill
diff --git a/src/raudio.h b/src/raudio.h
index 302993a9..e508d380 100644
--- a/src/raudio.h
+++ b/src/raudio.h
@@ -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:
* - Manage audio device (init/close)
@@ -20,7 +20,7 @@
*
* CONTRIBUTORS:
* David Reid (github: @mackron) (Nov. 2017):
-* - Complete port to mini_al library
+* - Complete port to miniaudio library
*
* Joshua Reisenauer (github: @kd7tck) (2015)
* - XM audio module support (jar_xm)
diff --git a/src/raylib.h b/src/raylib.h
index ec6fd8a1..ed0314d3 100644
--- a/src/raylib.h
+++ b/src/raylib.h
@@ -434,7 +434,6 @@ typedef struct Music {
void *ctxData; // Audio context data, depends on type
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
AudioStream stream; // Audio stream
generated by cgit v1.2.3 (git 2.25.1) at 2026-04-24 16:48:37 +0000