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-rw-r--r--src/raudio.c304
-rw-r--r--src/raudio.h164
2 files changed, 215 insertions, 253 deletions
diff --git a/src/raudio.c b/src/raudio.c
index c63c5e9d..bb4638dc 100644
--- a/src/raudio.c
+++ b/src/raudio.c
@@ -236,16 +236,16 @@ static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 fr
// AudioBuffer management functions declaration
// NOTE: Those functions are not exposed by raylib... for the moment
AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 bufferSizeInFrames, int usage);
-void CloseAudioBuffer(AudioBuffer *audioBuffer);
-bool IsAudioBufferPlaying(AudioBuffer *audioBuffer);
-void PlayAudioBuffer(AudioBuffer *audioBuffer);
-void StopAudioBuffer(AudioBuffer *audioBuffer);
-void PauseAudioBuffer(AudioBuffer *audioBuffer);
-void ResumeAudioBuffer(AudioBuffer *audioBuffer);
-void SetAudioBufferVolume(AudioBuffer *audioBuffer, float volume);
-void SetAudioBufferPitch(AudioBuffer *audioBuffer, float pitch);
-void TrackAudioBuffer(AudioBuffer *audioBuffer);
-void UntrackAudioBuffer(AudioBuffer *audioBuffer);
+void CloseAudioBuffer(AudioBuffer *buffer);
+bool IsAudioBufferPlaying(AudioBuffer *buffer);
+void PlayAudioBuffer(AudioBuffer *buffer);
+void StopAudioBuffer(AudioBuffer *buffer);
+void PauseAudioBuffer(AudioBuffer *buffer);
+void ResumeAudioBuffer(AudioBuffer *buffer);
+void SetAudioBufferVolume(AudioBuffer *buffer, float volume);
+void SetAudioBufferPitch(AudioBuffer *buffer, float pitch);
+void TrackAudioBuffer(AudioBuffer *buffer);
+void UntrackAudioBuffer(AudioBuffer *buffer);
//----------------------------------------------------------------------------------
// Multi channel playback globals
@@ -644,151 +644,127 @@ AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sam
}
// Delete an audio buffer
-void CloseAudioBuffer(AudioBuffer *audioBuffer)
+void CloseAudioBuffer(AudioBuffer *buffer)
{
- if (audioBuffer == NULL)
+ if (buffer != NULL)
{
- TraceLog(LOG_ERROR, "CloseAudioBuffer() : No audio buffer");
- return;
+ UntrackAudioBuffer(buffer);
+ RL_FREE(buffer->buffer);
+ RL_FREE(buffer);
}
-
- UntrackAudioBuffer(audioBuffer);
- RL_FREE(audioBuffer->buffer);
- RL_FREE(audioBuffer);
+ else TraceLog(LOG_ERROR, "CloseAudioBuffer() : No audio buffer");
}
// Check if an audio buffer is playing
-bool IsAudioBufferPlaying(AudioBuffer *audioBuffer)
+bool IsAudioBufferPlaying(AudioBuffer *buffer)
{
- if (audioBuffer == NULL)
- {
- TraceLog(LOG_ERROR, "IsAudioBufferPlaying() : No audio buffer");
- return false;
- }
+ bool result = false;
+
+ if (buffer != NULL) result = (buffer->playing && !buffer->paused);
+ else TraceLog(LOG_ERROR, "IsAudioBufferPlaying() : No audio buffer");
- return audioBuffer->playing && !audioBuffer->paused;
+ return result;
}
// Play an audio buffer
// NOTE: Buffer is restarted to the start.
// Use PauseAudioBuffer() and ResumeAudioBuffer() if the playback position should be maintained.
-void PlayAudioBuffer(AudioBuffer *audioBuffer)
+void PlayAudioBuffer(AudioBuffer *buffer)
{
- if (audioBuffer == NULL)
+ if (buffer != NULL)
{
- TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer");
- return;
+ buffer->playing = true;
+ buffer->paused = false;
+ buffer->frameCursorPos = 0;
}
-
- audioBuffer->playing = true;
- audioBuffer->paused = false;
- audioBuffer->frameCursorPos = 0;
+ else TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer");
}
// Stop an audio buffer
-void StopAudioBuffer(AudioBuffer *audioBuffer)
+void StopAudioBuffer(AudioBuffer *buffer)
{
- if (audioBuffer == NULL)
+ if (buffer != NULL)
{
- TraceLog(LOG_ERROR, "StopAudioBuffer() : No audio buffer");
- return;
+ if (IsAudioBufferPlaying(buffer))
+ {
+ buffer->playing = false;
+ buffer->paused = false;
+ buffer->frameCursorPos = 0;
+ buffer->isSubBufferProcessed[0] = true;
+ buffer->isSubBufferProcessed[1] = true;
+ }
}
-
- // Don't do anything if the audio buffer is already stopped.
- if (!IsAudioBufferPlaying(audioBuffer)) return;
-
- audioBuffer->playing = false;
- audioBuffer->paused = false;
- audioBuffer->frameCursorPos = 0;
- audioBuffer->isSubBufferProcessed[0] = true;
- audioBuffer->isSubBufferProcessed[1] = true;
+ else TraceLog(LOG_ERROR, "StopAudioBuffer() : No audio buffer");
}
// Pause an audio buffer
-void PauseAudioBuffer(AudioBuffer *audioBuffer)
+void PauseAudioBuffer(AudioBuffer *buffer)
{
- if (audioBuffer == NULL)
- {
- TraceLog(LOG_ERROR, "PauseAudioBuffer() : No audio buffer");
- return;
- }
-
- audioBuffer->paused = true;
+ if (buffer != NULL) buffer->paused = true;
+ else TraceLog(LOG_ERROR, "PauseAudioBuffer() : No audio buffer");
}
// Resume an audio buffer
-void ResumeAudioBuffer(AudioBuffer *audioBuffer)
+void ResumeAudioBuffer(AudioBuffer *buffer)
{
- if (audioBuffer == NULL)
- {
- TraceLog(LOG_ERROR, "ResumeAudioBuffer() : No audio buffer");
- return;
- }
-
- audioBuffer->paused = false;
+ if (buffer != NULL) buffer->paused = false;
+ else TraceLog(LOG_ERROR, "ResumeAudioBuffer() : No audio buffer");
}
// Set volume for an audio buffer
-void SetAudioBufferVolume(AudioBuffer *audioBuffer, float volume)
+void SetAudioBufferVolume(AudioBuffer *buffer, float volume)
{
- if (audioBuffer == NULL)
- {
- TraceLog(LOG_WARNING, "SetAudioBufferVolume() : No audio buffer");
- return;
- }
-
- audioBuffer->volume = volume;
+ if (buffer != NULL) buffer->volume = volume;
+ else TraceLog(LOG_WARNING, "SetAudioBufferVolume() : No audio buffer");
}
// Set pitch for an audio buffer
-void SetAudioBufferPitch(AudioBuffer *audioBuffer, float pitch)
+void SetAudioBufferPitch(AudioBuffer *buffer, float pitch)
{
- if (audioBuffer == NULL)
+ if (buffer != NULL)
{
- TraceLog(LOG_WARNING, "SetAudioBufferPitch() : No audio buffer");
- return;
- }
-
- float pitchMul = pitch/audioBuffer->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.
- ma_uint32 newOutputSampleRate = (ma_uint32)((float)audioBuffer->dsp.src.config.sampleRateOut / pitchMul);
- audioBuffer->pitch *= (float)audioBuffer->dsp.src.config.sampleRateOut / newOutputSampleRate;
+ // 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;
- ma_pcm_converter_set_output_sample_rate(&audioBuffer->dsp, newOutputSampleRate);
+ ma_pcm_converter_set_output_sample_rate(&buffer->dsp, newOutputSampleRate);
+ }
+ else TraceLog(LOG_WARNING, "SetAudioBufferPitch() : No audio buffer");
}
// Track audio buffer to linked list next position
-void TrackAudioBuffer(AudioBuffer *audioBuffer)
+void TrackAudioBuffer(AudioBuffer *buffer)
{
ma_mutex_lock(&audioLock);
{
- if (firstAudioBuffer == NULL) firstAudioBuffer = audioBuffer;
+ if (firstAudioBuffer == NULL) firstAudioBuffer = buffer;
else
{
- lastAudioBuffer->next = audioBuffer;
- audioBuffer->prev = lastAudioBuffer;
+ lastAudioBuffer->next = buffer;
+ buffer->prev = lastAudioBuffer;
}
- lastAudioBuffer = audioBuffer;
+ lastAudioBuffer = buffer;
}
ma_mutex_unlock(&audioLock);
}
// Untrack audio buffer from linked list
-void UntrackAudioBuffer(AudioBuffer *audioBuffer)
+void UntrackAudioBuffer(AudioBuffer *buffer)
{
ma_mutex_lock(&audioLock);
{
- if (audioBuffer->prev == NULL) firstAudioBuffer = audioBuffer->next;
- else audioBuffer->prev->next = audioBuffer->next;
+ if (buffer->prev == NULL) firstAudioBuffer = buffer->next;
+ else buffer->prev->next = buffer->next;
- if (audioBuffer->next == NULL) lastAudioBuffer = audioBuffer->prev;
- else audioBuffer->next->prev = audioBuffer->prev;
+ if (buffer->next == NULL) lastAudioBuffer = buffer->prev;
+ else buffer->next->prev = buffer->prev;
- audioBuffer->prev = NULL;
- audioBuffer->next = NULL;
+ buffer->prev = NULL;
+ buffer->next = NULL;
}
ma_mutex_unlock(&audioLock);
}
@@ -802,10 +778,9 @@ Wave LoadWave(const char *fileName)
{
Wave wave = { 0 };
+ if (false) { }
#if defined(SUPPORT_FILEFORMAT_WAV)
- if (IsFileExtension(fileName, ".wav")) wave = LoadWAV(fileName);
-#else
- if (false) {}
+ else if (IsFileExtension(fileName, ".wav")) wave = LoadWAV(fileName);
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
else if (IsFileExtension(fileName, ".ogg")) wave = LoadOGG(fileName);
@@ -821,25 +796,6 @@ Wave LoadWave(const char *fileName)
return wave;
}
-// Load wave data from raw array data
-Wave LoadWaveEx(void *data, int sampleCount, int sampleRate, int sampleSize, int channels)
-{
- Wave wave;
-
- wave.data = data;
- wave.sampleCount = sampleCount;
- wave.sampleRate = sampleRate;
- wave.sampleSize = sampleSize;
- wave.channels = channels;
-
- // NOTE: Copy wave data to work with, user is responsible of input data to free
- Wave cwave = WaveCopy(wave);
-
- WaveFormat(&cwave, sampleRate, sampleSize, channels);
-
- return cwave;
-}
-
// Load sound from file
// NOTE: The entire file is loaded to memory to be played (no-streaming)
Sound LoadSound(const char *fileName)
@@ -903,7 +859,7 @@ void UnloadWave(Wave wave)
// Unload sound
void UnloadSound(Sound sound)
{
- CloseAudioBuffer((AudioBuffer *)sound.stream.buffer);
+ CloseAudioBuffer(sound.stream.buffer);
TraceLog(LOG_INFO, "Unloaded sound data from RAM");
}
@@ -911,7 +867,7 @@ void UnloadSound(Sound sound)
// Update sound buffer with new data
void UpdateSound(Sound sound, const void *data, int samplesCount)
{
- AudioBuffer *audioBuffer = (AudioBuffer *)sound.stream.buffer;
+ AudioBuffer *audioBuffer = sound.stream.buffer;
if (audioBuffer == NULL)
{
@@ -930,10 +886,9 @@ void ExportWave(Wave wave, const char *fileName)
{
bool success = false;
+ if (false) { }
#if defined(SUPPORT_FILEFORMAT_WAV)
- if (IsFileExtension(fileName, ".wav")) success = SaveWAV(wave, fileName);
-#else
- if (false) {}
+ else if (IsFileExtension(fileName, ".wav")) success = SaveWAV(wave, fileName);
#endif
else if (IsFileExtension(fileName, ".raw"))
{
@@ -994,7 +949,7 @@ void ExportWaveAsCode(Wave wave, const char *fileName)
// Play a sound
void PlaySound(Sound sound)
{
- PlayAudioBuffer((AudioBuffer *)sound.stream.buffer);
+ PlayAudioBuffer(sound.stream.buffer);
}
// Play a sound in the multichannel buffer pool
@@ -1046,17 +1001,16 @@ void PlaySoundMulti(Sound sound)
audioBufferPoolChannels[index] = audioBufferPoolCounter;
audioBufferPoolCounter++;
- audioBufferPool[index]->volume = ((AudioBuffer*)sound.stream.buffer)->volume;
- audioBufferPool[index]->pitch = ((AudioBuffer*)sound.stream.buffer)->pitch;
- audioBufferPool[index]->looping = ((AudioBuffer*)sound.stream.buffer)->looping;
- audioBufferPool[index]->usage = ((AudioBuffer*)sound.stream.buffer)->usage;
+ audioBufferPool[index]->volume = sound.stream.buffer->volume;
+ audioBufferPool[index]->pitch = sound.stream.buffer->pitch;
+ audioBufferPool[index]->looping = sound.stream.buffer->looping;
+ audioBufferPool[index]->usage = sound.stream.buffer->usage;
audioBufferPool[index]->isSubBufferProcessed[0] = false;
audioBufferPool[index]->isSubBufferProcessed[1] = false;
- audioBufferPool[index]->bufferSizeInFrames = ((AudioBuffer*)sound.stream.buffer)->bufferSizeInFrames;
- audioBufferPool[index]->buffer = ((AudioBuffer*)sound.stream.buffer)->buffer;
+ audioBufferPool[index]->bufferSizeInFrames = sound.stream.buffer->bufferSizeInFrames;
+ audioBufferPool[index]->buffer = sound.stream.buffer->buffer;
PlayAudioBuffer(audioBufferPool[index]);
-
}
// Stop any sound played with PlaySoundMulti()
@@ -1081,37 +1035,37 @@ int GetSoundsPlaying(void)
// Pause a sound
void PauseSound(Sound sound)
{
- PauseAudioBuffer((AudioBuffer *)sound.stream.buffer);
+ PauseAudioBuffer(sound.stream.buffer);
}
// Resume a paused sound
void ResumeSound(Sound sound)
{
- ResumeAudioBuffer((AudioBuffer *)sound.stream.buffer);
+ ResumeAudioBuffer(sound.stream.buffer);
}
// Stop reproducing a sound
void StopSound(Sound sound)
{
- StopAudioBuffer((AudioBuffer *)sound.stream.buffer);
+ StopAudioBuffer(sound.stream.buffer);
}
// Check if a sound is playing
bool IsSoundPlaying(Sound sound)
{
- return IsAudioBufferPlaying((AudioBuffer *)sound.stream.buffer);
+ return IsAudioBufferPlaying(sound.stream.buffer);
}
// Set volume for a sound
void SetSoundVolume(Sound sound, float volume)
{
- SetAudioBufferVolume((AudioBuffer *)sound.stream.buffer, volume);
+ SetAudioBufferVolume(sound.stream.buffer, volume);
}
// Set pitch for a sound
void SetSoundPitch(Sound sound, float pitch)
{
- SetAudioBufferPitch((AudioBuffer *)sound.stream.buffer, pitch);
+ SetAudioBufferPitch(sound.stream.buffer, pitch);
}
// Convert wave data to desired format
@@ -1213,25 +1167,26 @@ float *GetWaveData(Wave wave)
Music LoadMusicStream(const char *fileName)
{
Music music = (MusicStream *)RL_MALLOC(sizeof(MusicStream));
- bool musicLoaded = true;
+ bool musicLoaded = false;
+ if (false) { }
#if defined(SUPPORT_FILEFORMAT_OGG)
- if (IsFileExtension(fileName, ".ogg"))
+ else if (IsFileExtension(fileName, ".ogg"))
{
// Open ogg audio stream
music->ctxData = stb_vorbis_open_filename(fileName, NULL, NULL);
- if (music->ctxData == NULL) musicLoaded = false;
- else
+ if (music->ctxData != NULL)
{
+ music->ctxType = MUSIC_AUDIO_OGG;
stb_vorbis_info info = stb_vorbis_get_info((stb_vorbis *)music->ctxData); // Get Ogg file info
// 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->ctxType = MUSIC_AUDIO_OGG;
- music->loopCount = 0; // Infinite loop by default
+ music->loopCount = 0; // Infinite loop by default
+ musicLoaded = true;
TraceLog(LOG_DEBUG, "[%s] OGG total samples: %i", fileName, music->sampleCount);
TraceLog(LOG_DEBUG, "[%s] OGG sample rate: %i", fileName, info.sample_rate);
@@ -1239,24 +1194,22 @@ Music LoadMusicStream(const char *fileName)
TraceLog(LOG_DEBUG, "[%s] OGG memory required: %i", fileName, info.temp_memory_required);
}
}
-#else
- if (false) {}
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
else if (IsFileExtension(fileName, ".flac"))
{
music->ctxData = drflac_open_file(fileName);
- if (music->ctxData == NULL) musicLoaded = false;
- else
+ if (music->ctxData != NULL)
{
+ music->ctxType = MUSIC_AUDIO_FLAC;
drflac *ctxFlac = (drflac *)music->ctxData;
-
+
music->stream = InitAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
music->sampleCount = (unsigned int)ctxFlac->totalSampleCount;
music->sampleLeft = music->sampleCount;
- music->ctxType = MUSIC_AUDIO_FLAC;
- music->loopCount = 0; // Infinite loop by default
+ 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);
@@ -1273,19 +1226,19 @@ Music LoadMusicStream(const char *fileName)
int result = drmp3_init_file(ctxMp3, fileName, NULL);
- if (!result) musicLoaded = false;
- else
+ if (result > 0)
{
- 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);
+ music->ctxType = MUSIC_AUDIO_MP3;
music->stream = InitAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels);
music->sampleCount = drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels;
music->sampleLeft = music->sampleCount;
- music->ctxType = MUSIC_AUDIO_MP3;
- music->loopCount = 0; // Infinite loop by default
+ 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);
}
}
@@ -1297,23 +1250,23 @@ Music LoadMusicStream(const char *fileName)
int result = jar_xm_create_context_from_file(&ctxXm, 48000, fileName);
- if (!result) // XM context created successfully
+ if (result > 0) // XM context created successfully
{
- jar_xm_set_max_loop_count(ctxXm, 0); // Set infinite number of loops
+ music->ctxType = MUSIC_MODULE_XM;
+ jar_xm_set_max_loop_count(ctxXm, 0); // Set infinite number of loops
// 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->ctxType = MUSIC_MODULE_XM;
- music->loopCount = 0; // Infinite loop by default
+ 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);
-
- music->ctxData = ctxXm;
}
- else musicLoaded = false;
}
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
@@ -1323,30 +1276,30 @@ Music LoadMusicStream(const char *fileName)
music->ctxData = ctxMod;
jar_mod_init(ctxMod);
+ int result = jar_mod_load_file(ctxMod, fileName);
- if (jar_mod_load_file(ctxMod, fileName))
+ if (result > 0)
{
+ music->ctxType = MUSIC_MODULE_MOD;
+
// 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->ctxType = MUSIC_MODULE_MOD;
- music->loopCount = 0; // Infinite loop by default
-
+ music->loopCount = 0; // Infinite loop by default
+ musicLoaded = true;
+
TraceLog(LOG_INFO, "[%s] MOD number of samples: %i", fileName, music->sampleLeft);
TraceLog(LOG_INFO, "[%s] MOD track length: %11.6f sec", fileName, (float)music->sampleCount/48000.0f);
}
- else musicLoaded = false;
}
#endif
- else musicLoaded = false;
if (!musicLoaded)
{
+ if (false) { }
#if defined(SUPPORT_FILEFORMAT_OGG)
- if (music->ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music->ctxData);
- #else
- if (false) {}
+ else if (music->ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music->ctxData);
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
else if (music->ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music->ctxData);
@@ -1377,10 +1330,9 @@ void UnloadMusicStream(Music music)
CloseAudioStream(music->stream);
+ if (false) { }
#if defined(SUPPORT_FILEFORMAT_OGG)
- if (music->ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music->ctxData);
-#else
- if (false) {}
+ else if (music->ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music->ctxData);
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
else if (music->ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music->ctxData);
@@ -1403,7 +1355,7 @@ void PlayMusicStream(Music music)
{
if (music != NULL)
{
- AudioBuffer *audioBuffer = (AudioBuffer *)music->stream.buffer;
+ AudioBuffer *audioBuffer = music->stream.buffer;
if (audioBuffer == NULL)
{
@@ -1473,7 +1425,7 @@ void UpdateMusicStream(Music music)
bool streamEnding = false;
- unsigned int subBufferSizeInFrames = ((AudioBuffer *)music->stream.buffer)->bufferSizeInFrames/2;
+ unsigned int subBufferSizeInFrames = music->stream.buffer->bufferSizeInFrames/2;
// NOTE: Using dynamic allocation because it could require more than 16KB
void *pcm = RL_CALLOC(subBufferSizeInFrames*music->stream.channels*music->stream.sampleSize/8, 1);
diff --git a/src/raudio.h b/src/raudio.h
index e032d09b..7a3b7515 100644
--- a/src/raudio.h
+++ b/src/raudio.h
@@ -80,40 +80,44 @@
// Wave type, defines audio wave data
typedef struct Wave {
- unsigned int sampleCount; // Number of samples
- unsigned int sampleRate; // Frequency (samples per second)
- unsigned int sampleSize; // Bit depth (bits per sample): 8, 16, 32 (24 not supported)
- unsigned int channels; // Number of channels (1-mono, 2-stereo)
- void *data; // Buffer data pointer
+ unsigned int sampleCount; // Total number of samples
+ unsigned int sampleRate; // Frequency (samples per second)
+ unsigned int sampleSize; // Bit depth (bits per sample): 8, 16, 32 (24 not supported)
+ unsigned int channels; // Number of channels (1-mono, 2-stereo)
+ void *data; // Buffer data pointer
} Wave;
-// Sound source type
-typedef struct Sound {
- void *audioBuffer; // Pointer to internal data used by the audio system
-
- unsigned int source; // Audio source id
- unsigned int buffer; // Audio buffer id
- int format; // Audio format specifier
-} Sound;
-
-// Music type (file streaming from memory)
-// NOTE: Anything longer than ~10 seconds should be streamed
-typedef struct MusicData *Music;
+typedef struct rAudioBuffer rAudioBuffer;
// Audio stream type
// NOTE: Useful to create custom audio streams not bound to a specific file
typedef struct AudioStream {
- unsigned int sampleRate; // Frequency (samples per second)
- unsigned int sampleSize; // Bit depth (bits per sample): 8, 16, 32 (24 not supported)
- unsigned int channels; // Number of channels (1-mono, 2-stereo)
+ unsigned int sampleRate; // Frequency (samples per second)
+ unsigned int sampleSize; // Bit depth (bits per sample): 8, 16, 32 (24 not supported)
+ unsigned int channels; // Number of channels (1-mono, 2-stereo)
- void *audioBuffer; // Pointer to internal data used by the audio system.
-
- int format; // Audio format specifier
- unsigned int source; // Audio source id
- unsigned int buffers[2]; // Audio buffers (double buffering)
+ rAudioBuffer *buffer; // Pointer to internal data used by the audio system
} AudioStream;
+// Sound source type
+typedef struct Sound {
+ unsigned int sampleCount; // Total number of samples
+ AudioStream stream; // Audio stream
+} Sound;
+
+// Music stream type (audio file streaming from memory)
+// NOTE: Anything longer than ~10 seconds should be streamed
+typedef struct MusicStream {
+ int ctxType; // Type of music context (audio filetype)
+ 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
+} MusicStream, *Music;
+
#ifdef __cplusplus
extern "C" { // Prevents name mangling of functions
#endif
@@ -126,60 +130,66 @@ extern "C" { // Prevents name mangling of functions
//----------------------------------------------------------------------------------
// Module Functions Declaration
//----------------------------------------------------------------------------------
-void InitAudioDevice(void); // Initialize audio device and context
-void CloseAudioDevice(void); // Close the audio device and context
-bool IsAudioDeviceReady(void); // Check if audio device has been initialized successfully
-void SetMasterVolume(float volume); // Set master volume (listener)
-
-Wave LoadWave(const char *fileName); // Load wave data from file
-Wave LoadWaveEx(void *data, int sampleCount, int sampleRate, int sampleSize, int channels); // Load wave data from raw array data
-Sound LoadSound(const char *fileName); // Load sound from file
-Sound LoadSoundFromWave(Wave wave); // Load sound from wave data
-void UpdateSound(Sound sound, const void *data, int samplesCount);// Update sound buffer with new data
-void UnloadWave(Wave wave); // Unload wave data
-void UnloadSound(Sound sound); // Unload sound
-void PlaySound(Sound sound); // Play a sound
-void PlaySoundMulti(Sound sound); // Play a sound using the multi channel buffer pool
-int GetSoundsPlaying(void); // Get number of sounds playing in the multichannel buffer pool
-void PauseSound(Sound sound); // Pause a sound
-void ResumeSound(Sound sound); // Resume a paused sound
-void StopSound(Sound sound); // Stop playing a sound
-void StopSoundMulti(void); // Stop any sound played with PlaySoundMulti()
-bool IsSoundPlaying(Sound sound); // Check if a sound is currently playing
-void SetSoundVolume(Sound sound, float volume); // Set volume for a sound (1.0 is max level)
-void SetSoundPitch(Sound sound, float pitch); // Set pitch for a sound (1.0 is base level)
-void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels); // Convert wave data to desired format
-Wave WaveCopy(Wave wave); // Copy a wave to a new wave
-void WaveCrop(Wave *wave, int initSample, int finalSample); // Crop a wave to defined samples range
-float *GetWaveData(Wave wave); // Get samples data from wave as a floats array
-Music LoadMusicStream(const char *fileName); // Load music stream from file
-void UnloadMusicStream(Music music); // Unload music stream
-void PlayMusicStream(Music music); // Start music playing
-void UpdateMusicStream(Music music); // Updates buffers for music streaming
-void StopMusicStream(Music music); // Stop music playing
-void PauseMusicStream(Music music); // Pause music playing
-void ResumeMusicStream(Music music); // Resume playing paused music
-bool IsMusicPlaying(Music music); // Check if music is playing
-void SetMusicVolume(Music music, float volume); // Set volume for music (1.0 is max level)
-void SetMusicPitch(Music music, float pitch); // Set pitch for a music (1.0 is base level)
-void SetMusicLoopCount(Music music, int count); // Set music loop count (loop repeats)
-float GetMusicTimeLength(Music music); // Get music time length (in seconds)
-float GetMusicTimePlayed(Music music); // Get current music time played (in seconds)
+
+// Audio device management functions
+RLAPI void InitAudioDevice(void); // Initialize audio device and context
+RLAPI void CloseAudioDevice(void); // Close the audio device and context
+RLAPI bool IsAudioDeviceReady(void); // Check if audio device has been initialized successfully
+RLAPI void SetMasterVolume(float volume); // Set master volume (listener)
+
+// Wave/Sound loading/unloading functions
+RLAPI Wave LoadWave(const char *fileName); // Load wave data from file
+RLAPI Sound LoadSound(const char *fileName); // Load sound from file
+RLAPI Sound LoadSoundFromWave(Wave wave); // Load sound from wave data
+RLAPI void UpdateSound(Sound sound, const void *data, int samplesCount);// Update sound buffer with new data
+RLAPI void UnloadWave(Wave wave); // Unload wave data
+RLAPI void UnloadSound(Sound sound); // Unload sound
+RLAPI void ExportWave(Wave wave, const char *fileName); // Export wave data to file
+RLAPI void ExportWaveAsCode(Wave wave, const char *fileName); // Export wave sample data to code (.h)
+
+// Wave/Sound management functions
+RLAPI void PlaySound(Sound sound); // Play a sound
+RLAPI void StopSound(Sound sound); // Stop playing a sound
+RLAPI void PauseSound(Sound sound); // Pause a sound
+RLAPI void ResumeSound(Sound sound); // Resume a paused sound
+RLAPI void PlaySoundMulti(Sound sound); // Play a sound (using multichannel buffer pool)
+RLAPI void StopSoundMulti(void); // Stop any sound playing (using multichannel buffer pool)
+RLAPI int GetSoundsPlaying(void); // Get number of sounds playing in the multichannel
+RLAPI bool IsSoundPlaying(Sound sound); // Check if a sound is currently playing
+RLAPI void SetSoundVolume(Sound sound, float volume); // Set volume for a sound (1.0 is max level)
+RLAPI void SetSoundPitch(Sound sound, float pitch); // Set pitch for a sound (1.0 is base level)
+RLAPI void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels); // Convert wave data to desired format
+RLAPI Wave WaveCopy(Wave wave); // Copy a wave to a new wave
+RLAPI void WaveCrop(Wave *wave, int initSample, int finalSample); // Crop a wave to defined samples range
+RLAPI float *GetWaveData(Wave wave); // Get samples data from wave as a floats array
+
+// Music management functions
+RLAPI Music LoadMusicStream(const char *fileName); // Load music stream from file
+RLAPI void UnloadMusicStream(Music music); // Unload music stream
+RLAPI void PlayMusicStream(Music music); // Start music playing
+RLAPI void UpdateMusicStream(Music music); // Updates buffers for music streaming
+RLAPI void StopMusicStream(Music music); // Stop music playing
+RLAPI void PauseMusicStream(Music music); // Pause music playing
+RLAPI void ResumeMusicStream(Music music); // Resume playing paused music
+RLAPI bool IsMusicPlaying(Music music); // Check if music is playing
+RLAPI void SetMusicVolume(Music music, float volume); // Set volume for music (1.0 is max level)
+RLAPI void SetMusicPitch(Music music, float pitch); // Set pitch for a music (1.0 is base level)
+RLAPI void SetMusicLoopCount(Music music, int count); // Set music loop count (loop repeats)
+RLAPI float GetMusicTimeLength(Music music); // Get music time length (in seconds)
+RLAPI float GetMusicTimePlayed(Music music); // Get current music time played (in seconds)
// AudioStream management functions
-AudioStream InitAudioStream(unsigned int sampleRate,
- unsigned int sampleSize,
- unsigned int channels); // Init audio stream (to stream raw audio pcm data)
-void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount); // Update audio stream buffers with data
-void CloseAudioStream(AudioStream stream); // Close audio stream and free memory
-bool IsAudioBufferProcessed(AudioStream stream); // Check if any audio stream buffers requires refill
-void PlayAudioStream(AudioStream stream); // Play audio stream
-void PauseAudioStream(AudioStream stream); // Pause audio stream
-void ResumeAudioStream(AudioStream stream); // Resume audio stream
-bool IsAudioStreamPlaying(AudioStream stream); // Check if audio stream is playing
-void StopAudioStream(AudioStream stream); // Stop audio stream
-void SetAudioStreamVolume(AudioStream stream, float volume); // Set volume for audio stream (1.0 is max level)
-void SetAudioStreamPitch(AudioStream stream, float pitch); // Set pitch for audio stream (1.0 is base level)
+RLAPI AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, unsigned int channels); // Init audio stream (to stream raw audio pcm data)
+RLAPI void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount); // Update audio stream buffers with data
+RLAPI void CloseAudioStream(AudioStream stream); // Close audio stream and free memory
+RLAPI bool IsAudioBufferProcessed(AudioStream stream); // Check if any audio stream buffers requires refill
+RLAPI void PlayAudioStream(AudioStream stream); // Play audio stream
+RLAPI void PauseAudioStream(AudioStream stream); // Pause audio stream
+RLAPI void ResumeAudioStream(AudioStream stream); // Resume audio stream
+RLAPI bool IsAudioStreamPlaying(AudioStream stream); // Check if audio stream is playing
+RLAPI void StopAudioStream(AudioStream stream); // Stop audio stream
+RLAPI void SetAudioStreamVolume(AudioStream stream, float volume); // Set volume for audio stream (1.0 is max level)
+RLAPI void SetAudioStreamPitch(AudioStream stream, float pitch); // Set pitch for audio stream (1.0 is base level)
#ifdef __cplusplus
}
generated by cgit v1.2.3 (git 2.25.1) at 2026-02-22 01:02:45 +0000