Some variables renamed
This commit is contained in:
Ray
parent
8730f22371
commit
ae95111006
2 changed files with 215 additions and 253 deletions
306
src/raudio.c
306
src/raudio.c
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@ -236,16 +236,16 @@ static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 fr
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// AudioBuffer management functions declaration
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// NOTE: Those functions are not exposed by raylib... for the moment
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AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 bufferSizeInFrames, int usage);
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void CloseAudioBuffer(AudioBuffer *audioBuffer);
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bool IsAudioBufferPlaying(AudioBuffer *audioBuffer);
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void PlayAudioBuffer(AudioBuffer *audioBuffer);
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void StopAudioBuffer(AudioBuffer *audioBuffer);
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void PauseAudioBuffer(AudioBuffer *audioBuffer);
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void ResumeAudioBuffer(AudioBuffer *audioBuffer);
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void SetAudioBufferVolume(AudioBuffer *audioBuffer, float volume);
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void SetAudioBufferPitch(AudioBuffer *audioBuffer, float pitch);
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void TrackAudioBuffer(AudioBuffer *audioBuffer);
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void UntrackAudioBuffer(AudioBuffer *audioBuffer);
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void CloseAudioBuffer(AudioBuffer *buffer);
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bool IsAudioBufferPlaying(AudioBuffer *buffer);
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void PlayAudioBuffer(AudioBuffer *buffer);
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void StopAudioBuffer(AudioBuffer *buffer);
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void PauseAudioBuffer(AudioBuffer *buffer);
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void ResumeAudioBuffer(AudioBuffer *buffer);
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void SetAudioBufferVolume(AudioBuffer *buffer, float volume);
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void SetAudioBufferPitch(AudioBuffer *buffer, float pitch);
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void TrackAudioBuffer(AudioBuffer *buffer);
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void UntrackAudioBuffer(AudioBuffer *buffer);
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//----------------------------------------------------------------------------------
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// Multi channel playback globals
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@ -644,151 +644,127 @@ AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sam
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}
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// Delete an audio buffer
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void CloseAudioBuffer(AudioBuffer *audioBuffer)
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void CloseAudioBuffer(AudioBuffer *buffer)
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{
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if (audioBuffer == NULL)
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if (buffer != NULL)
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{
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TraceLog(LOG_ERROR, "CloseAudioBuffer() : No audio buffer");
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return;
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UntrackAudioBuffer(buffer);
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RL_FREE(buffer->buffer);
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RL_FREE(buffer);
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}
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UntrackAudioBuffer(audioBuffer);
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RL_FREE(audioBuffer->buffer);
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RL_FREE(audioBuffer);
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else TraceLog(LOG_ERROR, "CloseAudioBuffer() : No audio buffer");
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}
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// Check if an audio buffer is playing
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bool IsAudioBufferPlaying(AudioBuffer *audioBuffer)
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bool IsAudioBufferPlaying(AudioBuffer *buffer)
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{
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if (audioBuffer == NULL)
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{
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TraceLog(LOG_ERROR, "IsAudioBufferPlaying() : No audio buffer");
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return false;
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}
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bool result = false;
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if (buffer != NULL) result = (buffer->playing && !buffer->paused);
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else TraceLog(LOG_ERROR, "IsAudioBufferPlaying() : No audio buffer");
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return audioBuffer->playing && !audioBuffer->paused;
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return result;
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}
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// Play an audio buffer
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// NOTE: Buffer is restarted to the start.
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// Use PauseAudioBuffer() and ResumeAudioBuffer() if the playback position should be maintained.
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void PlayAudioBuffer(AudioBuffer *audioBuffer)
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void PlayAudioBuffer(AudioBuffer *buffer)
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{
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if (audioBuffer == NULL)
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if (buffer != NULL)
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{
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TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer");
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return;
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buffer->playing = true;
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buffer->paused = false;
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buffer->frameCursorPos = 0;
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}
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audioBuffer->playing = true;
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audioBuffer->paused = false;
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audioBuffer->frameCursorPos = 0;
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else TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer");
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}
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// Stop an audio buffer
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void StopAudioBuffer(AudioBuffer *audioBuffer)
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void StopAudioBuffer(AudioBuffer *buffer)
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{
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if (audioBuffer == NULL)
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if (buffer != NULL)
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{
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TraceLog(LOG_ERROR, "StopAudioBuffer() : No audio buffer");
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return;
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if (IsAudioBufferPlaying(buffer))
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{
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buffer->playing = false;
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buffer->paused = false;
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buffer->frameCursorPos = 0;
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buffer->isSubBufferProcessed[0] = true;
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buffer->isSubBufferProcessed[1] = true;
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}
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}
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// Don't do anything if the audio buffer is already stopped.
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if (!IsAudioBufferPlaying(audioBuffer)) return;
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audioBuffer->playing = false;
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audioBuffer->paused = false;
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audioBuffer->frameCursorPos = 0;
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audioBuffer->isSubBufferProcessed[0] = true;
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audioBuffer->isSubBufferProcessed[1] = true;
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else TraceLog(LOG_ERROR, "StopAudioBuffer() : No audio buffer");
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}
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// Pause an audio buffer
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void PauseAudioBuffer(AudioBuffer *audioBuffer)
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void PauseAudioBuffer(AudioBuffer *buffer)
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{
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if (audioBuffer == NULL)
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{
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TraceLog(LOG_ERROR, "PauseAudioBuffer() : No audio buffer");
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return;
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}
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audioBuffer->paused = true;
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if (buffer != NULL) buffer->paused = true;
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else TraceLog(LOG_ERROR, "PauseAudioBuffer() : No audio buffer");
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}
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// Resume an audio buffer
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void ResumeAudioBuffer(AudioBuffer *audioBuffer)
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void ResumeAudioBuffer(AudioBuffer *buffer)
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{
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if (audioBuffer == NULL)
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{
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TraceLog(LOG_ERROR, "ResumeAudioBuffer() : No audio buffer");
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return;
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}
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audioBuffer->paused = false;
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if (buffer != NULL) buffer->paused = false;
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else TraceLog(LOG_ERROR, "ResumeAudioBuffer() : No audio buffer");
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}
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// Set volume for an audio buffer
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void SetAudioBufferVolume(AudioBuffer *audioBuffer, float volume)
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void SetAudioBufferVolume(AudioBuffer *buffer, float volume)
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{
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if (audioBuffer == NULL)
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{
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TraceLog(LOG_WARNING, "SetAudioBufferVolume() : No audio buffer");
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return;
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}
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audioBuffer->volume = volume;
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if (buffer != NULL) buffer->volume = volume;
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else TraceLog(LOG_WARNING, "SetAudioBufferVolume() : No audio buffer");
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}
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// Set pitch for an audio buffer
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void SetAudioBufferPitch(AudioBuffer *audioBuffer, float pitch)
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void SetAudioBufferPitch(AudioBuffer *buffer, float pitch)
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{
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if (audioBuffer == NULL)
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if (buffer != NULL)
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{
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TraceLog(LOG_WARNING, "SetAudioBufferPitch() : No audio buffer");
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return;
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float pitchMul = pitch/buffer->pitch;
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// Pitching is just an adjustment of the sample rate. Note that this changes the duration of the sound - higher pitches
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// will make the sound faster; lower pitches make it slower.
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ma_uint32 newOutputSampleRate = (ma_uint32)((float)buffer->dsp.src.config.sampleRateOut/pitchMul);
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buffer->pitch *= (float)buffer->dsp.src.config.sampleRateOut/newOutputSampleRate;
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ma_pcm_converter_set_output_sample_rate(&buffer->dsp, newOutputSampleRate);
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}
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float pitchMul = pitch/audioBuffer->pitch;
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// Pitching is just an adjustment of the sample rate. Note that this changes the duration of the sound - higher pitches
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// will make the sound faster; lower pitches make it slower.
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ma_uint32 newOutputSampleRate = (ma_uint32)((float)audioBuffer->dsp.src.config.sampleRateOut / pitchMul);
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audioBuffer->pitch *= (float)audioBuffer->dsp.src.config.sampleRateOut / newOutputSampleRate;
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ma_pcm_converter_set_output_sample_rate(&audioBuffer->dsp, newOutputSampleRate);
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else TraceLog(LOG_WARNING, "SetAudioBufferPitch() : No audio buffer");
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}
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// Track audio buffer to linked list next position
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void TrackAudioBuffer(AudioBuffer *audioBuffer)
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void TrackAudioBuffer(AudioBuffer *buffer)
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{
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ma_mutex_lock(&audioLock);
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{
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if (firstAudioBuffer == NULL) firstAudioBuffer = audioBuffer;
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if (firstAudioBuffer == NULL) firstAudioBuffer = buffer;
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else
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{
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lastAudioBuffer->next = audioBuffer;
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audioBuffer->prev = lastAudioBuffer;
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lastAudioBuffer->next = buffer;
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buffer->prev = lastAudioBuffer;
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}
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lastAudioBuffer = audioBuffer;
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lastAudioBuffer = buffer;
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}
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ma_mutex_unlock(&audioLock);
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}
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// Untrack audio buffer from linked list
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void UntrackAudioBuffer(AudioBuffer *audioBuffer)
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void UntrackAudioBuffer(AudioBuffer *buffer)
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{
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ma_mutex_lock(&audioLock);
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{
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if (audioBuffer->prev == NULL) firstAudioBuffer = audioBuffer->next;
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else audioBuffer->prev->next = audioBuffer->next;
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if (buffer->prev == NULL) firstAudioBuffer = buffer->next;
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else buffer->prev->next = buffer->next;
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if (audioBuffer->next == NULL) lastAudioBuffer = audioBuffer->prev;
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else audioBuffer->next->prev = audioBuffer->prev;
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if (buffer->next == NULL) lastAudioBuffer = buffer->prev;
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else buffer->next->prev = buffer->prev;
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audioBuffer->prev = NULL;
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audioBuffer->next = NULL;
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buffer->prev = NULL;
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buffer->next = NULL;
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}
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ma_mutex_unlock(&audioLock);
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}
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@ -802,10 +778,9 @@ Wave LoadWave(const char *fileName)
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{
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Wave wave = { 0 };
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if (false) { }
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#if defined(SUPPORT_FILEFORMAT_WAV)
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if (IsFileExtension(fileName, ".wav")) wave = LoadWAV(fileName);
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#else
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if (false) {}
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else if (IsFileExtension(fileName, ".wav")) wave = LoadWAV(fileName);
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#endif
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#if defined(SUPPORT_FILEFORMAT_OGG)
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else if (IsFileExtension(fileName, ".ogg")) wave = LoadOGG(fileName);
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@ -821,25 +796,6 @@ Wave LoadWave(const char *fileName)
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return wave;
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}
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// Load wave data from raw array data
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Wave LoadWaveEx(void *data, int sampleCount, int sampleRate, int sampleSize, int channels)
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{
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Wave wave;
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wave.data = data;
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wave.sampleCount = sampleCount;
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wave.sampleRate = sampleRate;
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wave.sampleSize = sampleSize;
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wave.channels = channels;
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// NOTE: Copy wave data to work with, user is responsible of input data to free
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Wave cwave = WaveCopy(wave);
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WaveFormat(&cwave, sampleRate, sampleSize, channels);
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return cwave;
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}
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// Load sound from file
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// NOTE: The entire file is loaded to memory to be played (no-streaming)
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Sound LoadSound(const char *fileName)
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@ -903,7 +859,7 @@ void UnloadWave(Wave wave)
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// Unload sound
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void UnloadSound(Sound sound)
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{
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CloseAudioBuffer((AudioBuffer *)sound.stream.buffer);
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CloseAudioBuffer(sound.stream.buffer);
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TraceLog(LOG_INFO, "Unloaded sound data from RAM");
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}
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@ -911,7 +867,7 @@ void UnloadSound(Sound sound)
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// Update sound buffer with new data
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void UpdateSound(Sound sound, const void *data, int samplesCount)
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{
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AudioBuffer *audioBuffer = (AudioBuffer *)sound.stream.buffer;
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AudioBuffer *audioBuffer = sound.stream.buffer;
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if (audioBuffer == NULL)
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{
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@ -930,10 +886,9 @@ void ExportWave(Wave wave, const char *fileName)
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{
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bool success = false;
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if (false) { }
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#if defined(SUPPORT_FILEFORMAT_WAV)
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if (IsFileExtension(fileName, ".wav")) success = SaveWAV(wave, fileName);
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#else
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if (false) {}
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else if (IsFileExtension(fileName, ".wav")) success = SaveWAV(wave, fileName);
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#endif
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else if (IsFileExtension(fileName, ".raw"))
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{
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@ -994,7 +949,7 @@ void ExportWaveAsCode(Wave wave, const char *fileName)
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// Play a sound
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void PlaySound(Sound sound)
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{
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PlayAudioBuffer((AudioBuffer *)sound.stream.buffer);
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PlayAudioBuffer(sound.stream.buffer);
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}
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// Play a sound in the multichannel buffer pool
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@ -1046,17 +1001,16 @@ void PlaySoundMulti(Sound sound)
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audioBufferPoolChannels[index] = audioBufferPoolCounter;
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audioBufferPoolCounter++;
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audioBufferPool[index]->volume = ((AudioBuffer*)sound.stream.buffer)->volume;
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audioBufferPool[index]->pitch = ((AudioBuffer*)sound.stream.buffer)->pitch;
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audioBufferPool[index]->looping = ((AudioBuffer*)sound.stream.buffer)->looping;
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audioBufferPool[index]->usage = ((AudioBuffer*)sound.stream.buffer)->usage;
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audioBufferPool[index]->volume = sound.stream.buffer->volume;
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audioBufferPool[index]->pitch = sound.stream.buffer->pitch;
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audioBufferPool[index]->looping = sound.stream.buffer->looping;
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audioBufferPool[index]->usage = sound.stream.buffer->usage;
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audioBufferPool[index]->isSubBufferProcessed[0] = false;
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audioBufferPool[index]->isSubBufferProcessed[1] = false;
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audioBufferPool[index]->bufferSizeInFrames = ((AudioBuffer*)sound.stream.buffer)->bufferSizeInFrames;
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audioBufferPool[index]->buffer = ((AudioBuffer*)sound.stream.buffer)->buffer;
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audioBufferPool[index]->bufferSizeInFrames = sound.stream.buffer->bufferSizeInFrames;
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audioBufferPool[index]->buffer = sound.stream.buffer->buffer;
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PlayAudioBuffer(audioBufferPool[index]);
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}
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// Stop any sound played with PlaySoundMulti()
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@ -1081,37 +1035,37 @@ int GetSoundsPlaying(void)
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// Pause a sound
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void PauseSound(Sound sound)
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{
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PauseAudioBuffer((AudioBuffer *)sound.stream.buffer);
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PauseAudioBuffer(sound.stream.buffer);
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}
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// Resume a paused sound
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void ResumeSound(Sound sound)
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{
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ResumeAudioBuffer((AudioBuffer *)sound.stream.buffer);
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ResumeAudioBuffer(sound.stream.buffer);
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}
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// Stop reproducing a sound
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void StopSound(Sound sound)
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{
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StopAudioBuffer((AudioBuffer *)sound.stream.buffer);
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StopAudioBuffer(sound.stream.buffer);
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}
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// Check if a sound is playing
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bool IsSoundPlaying(Sound sound)
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{
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return IsAudioBufferPlaying((AudioBuffer *)sound.stream.buffer);
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return IsAudioBufferPlaying(sound.stream.buffer);
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}
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// Set volume for a sound
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void SetSoundVolume(Sound sound, float volume)
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{
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SetAudioBufferVolume((AudioBuffer *)sound.stream.buffer, volume);
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SetAudioBufferVolume(sound.stream.buffer, volume);
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}
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// Set pitch for a sound
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void SetSoundPitch(Sound sound, float pitch)
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{
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SetAudioBufferPitch((AudioBuffer *)sound.stream.buffer, pitch);
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SetAudioBufferPitch(sound.stream.buffer, pitch);
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}
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// Convert wave data to desired format
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@ -1213,25 +1167,26 @@ float *GetWaveData(Wave wave)
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Music LoadMusicStream(const char *fileName)
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{
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Music music = (MusicStream *)RL_MALLOC(sizeof(MusicStream));
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bool musicLoaded = true;
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bool musicLoaded = false;
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if (false) { }
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#if defined(SUPPORT_FILEFORMAT_OGG)
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if (IsFileExtension(fileName, ".ogg"))
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else if (IsFileExtension(fileName, ".ogg"))
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{
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// Open ogg audio stream
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music->ctxData = stb_vorbis_open_filename(fileName, NULL, NULL);
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if (music->ctxData == NULL) musicLoaded = false;
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else
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if (music->ctxData != NULL)
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{
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music->ctxType = MUSIC_AUDIO_OGG;
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stb_vorbis_info info = stb_vorbis_get_info((stb_vorbis *)music->ctxData); // Get Ogg file info
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// OGG bit rate defaults to 16 bit, it's enough for compressed format
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music->stream = InitAudioStream(info.sample_rate, 16, info.channels);
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music->sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music->ctxData)*info.channels;
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music->sampleLeft = music->sampleCount;
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music->ctxType = MUSIC_AUDIO_OGG;
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music->loopCount = 0; // Infinite loop by default
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music->loopCount = 0; // Infinite loop by default
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musicLoaded = true;
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TraceLog(LOG_DEBUG, "[%s] OGG total samples: %i", fileName, music->sampleCount);
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TraceLog(LOG_DEBUG, "[%s] OGG sample rate: %i", fileName, info.sample_rate);
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@ -1239,24 +1194,22 @@ Music LoadMusicStream(const char *fileName)
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TraceLog(LOG_DEBUG, "[%s] OGG memory required: %i", fileName, info.temp_memory_required);
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}
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}
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#else
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if (false) {}
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#endif
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#if defined(SUPPORT_FILEFORMAT_FLAC)
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else if (IsFileExtension(fileName, ".flac"))
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{
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music->ctxData = drflac_open_file(fileName);
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if (music->ctxData == NULL) musicLoaded = false;
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else
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if (music->ctxData != NULL)
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{
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music->ctxType = MUSIC_AUDIO_FLAC;
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drflac *ctxFlac = (drflac *)music->ctxData;
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music->stream = InitAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
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music->sampleCount = (unsigned int)ctxFlac->totalSampleCount;
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music->sampleLeft = music->sampleCount;
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music->ctxType = MUSIC_AUDIO_FLAC;
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music->loopCount = 0; // Infinite loop by default
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music->loopCount = 0; // Infinite loop by default
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musicLoaded = true;
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TraceLog(LOG_DEBUG, "[%s] FLAC total samples: %i", fileName, music->sampleCount);
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TraceLog(LOG_DEBUG, "[%s] FLAC sample rate: %i", fileName, ctxFlac->sampleRate);
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@ -1273,19 +1226,19 @@ Music LoadMusicStream(const char *fileName)
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int result = drmp3_init_file(ctxMp3, fileName, NULL);
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if (!result) musicLoaded = false;
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else
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if (result > 0)
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{
|
||||
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);
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue