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Ray
2026-03-04 01:14:26 +01:00
parent 23c06bc6f1
commit faf42366ec
23 changed files with 213 additions and 223 deletions
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64
src/raudio.c
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@ -26,7 +26,7 @@
* #define SUPPORT_FILEFORMAT_XM
* #define SUPPORT_FILEFORMAT_MOD
* Selected desired fileformats to be supported for loading. Some of those formats are
* supported by default, to remove support, just comment unrequired #define in this module
* supported by default, to remove support, comment unrequired #define in this module
*
* DEPENDENCIES:
* miniaudio.h - Audio device management lib (https://github.com/mackron/miniaudio)
@ -299,7 +299,7 @@ typedef struct tagBITMAPINFOHEADER {
#endif
#ifndef AUDIO_BUFFER_RESIDUAL_CAPACITY
#define AUDIO_BUFFER_RESIDUAL_CAPACITY 8 // In PCM frames. For resampling and pitch shifting.
#define AUDIO_BUFFER_RESIDUAL_CAPACITY 8 // In PCM frames, for resampling and pitch shifting
#endif
//----------------------------------------------------------------------------------
@ -406,7 +406,7 @@ static AudioData AUDIO = { // Global AUDIO context
// NOTE: Music buffer size is defined by number of samples, independent of sample size and channels number
// 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
// In case of music-stalls, increase this number
.Buffer.defaultSize = 0,
.mixedProcessor = NULL
};
@ -601,7 +601,7 @@ AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sam
// be consumed. Any residual input frames need to be kept track of to
// ensure there are no discontinuities. Since raylib supports pitch
// shifting, which is done through resampling, a cache will always be
// required. This will be kept relatively small to avoid too much wastage.
// required. This will be kept relatively small to avoid too much wastage
audioBuffer->converterResidualCount = 0;
audioBuffer->converterResidual = (unsigned char*)RL_CALLOC(AUDIO_BUFFER_RESIDUAL_CAPACITY*ma_get_bytes_per_frame(format, channels), 1);
@ -721,7 +721,7 @@ void SetAudioBufferPitch(AudioBuffer *buffer, float pitch)
if ((buffer != NULL) && (pitch > 0.0f))
{
ma_mutex_lock(&AUDIO.System.lock);
// Pitching is just an adjustment of the sample rate
// Pitching is 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
@ -1046,7 +1046,7 @@ void UnloadSound(Sound sound)
void UnloadSoundAlias(Sound alias)
{
// Untrack and unload just the sound buffer, not the sample data, it is shared with the source for the alias
// Untrack and unload the sound buffer, not the sample data, it is shared with the source for the alias
if (alias.stream.buffer != NULL)
{
UntrackAudioBuffer(alias.stream.buffer);
@ -2316,7 +2316,7 @@ void DetachAudioStreamProcessor(AudioStream stream, AudioCallback process)
// Add processor to audio pipeline. Order of processors is important
// Works the same way as {Attach,Detach}AudioStreamProcessor() functions, except
// these two work on the already mixed output just before sending it to the sound hardware
// these two work on the already mixed output before sending it to the sound hardware
void AttachAudioMixedProcessor(AudioCallback process)
{
ma_mutex_lock(&AUDIO.System.lock);
@ -2397,7 +2397,7 @@ static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer,
if (currentSubBufferIndex > 1) return 0;
// Another thread can update the processed state of buffers, so
// just take a copy here to try and avoid potential synchronization problems
// take a copy here to try and avoid potential synchronization problems
bool isSubBufferProcessed[2] = { 0 };
isSubBufferProcessed[0] = audioBuffer->isSubBufferProcessed[0];
isSubBufferProcessed[1] = audioBuffer->isSubBufferProcessed[1];
@ -2478,8 +2478,8 @@ static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer,
static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, float *framesOut, ma_uint32 frameCount)
{
// NOTE: Continuously converting data from the AudioBuffer's internal format to the mixing format,
// which should be defined by the output format of the data converter.
// This is done until frameCount frames have been output.
// which should be defined by the output format of the data converter
// This is done until frameCount frames have been output
ma_uint32 bpf = ma_get_bytes_per_frame(audioBuffer->converter.formatIn, audioBuffer->converter.channelsIn);
ma_uint8 inputBuffer[4096] = { 0 };
ma_uint32 inputBufferFrameCap = sizeof(inputBuffer)/bpf;
@ -2491,34 +2491,34 @@ static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, f
ma_uint64 outputFramesToProcessThisIteration = frameCount - totalOutputFramesProcessed;
//ma_uint64 inputFramesToProcessThisIteration = 0;
// Process any residual input frames from the previous read first.
// Process any residual input frames from the previous read first
if (audioBuffer->converterResidualCount > 0)
{
ma_uint64 inputFramesProcessedThisIteration = audioBuffer->converterResidualCount;
ma_uint64 outputFramesProcessedThisIteration = outputFramesToProcessThisIteration;
ma_data_converter_process_pcm_frames(&audioBuffer->converter, audioBuffer->converterResidual, &inputFramesProcessedThisIteration, runningFramesOut, &outputFramesProcessedThisIteration);
// Make sure the data in the cache is consumed. This can be optimized to use a cursor instead of a memmove().
memmove(audioBuffer->converterResidual, audioBuffer->converterResidual + inputFramesProcessedThisIteration*bpf, (size_t)(AUDIO_BUFFER_RESIDUAL_CAPACITY - inputFramesProcessedThisIteration) * bpf);
// Make sure the data in the cache is consumed, this can be optimized to use a cursor instead of a memmove()
memmove(audioBuffer->converterResidual, audioBuffer->converterResidual + inputFramesProcessedThisIteration*bpf, (size_t)(AUDIO_BUFFER_RESIDUAL_CAPACITY - inputFramesProcessedThisIteration)*bpf);
audioBuffer->converterResidualCount -= (ma_uint32)inputFramesProcessedThisIteration; // Safe cast
totalOutputFramesProcessed += (ma_uint32)outputFramesProcessedThisIteration; // Safe cast
}
else
{
// Getting here means there are no residual frames from the previous read. Fresh data can now be
// pulled from the AudioBuffer and processed.
// Getting here means there are no residual frames from the previous read
// Fresh data can now be pulled from the AudioBuffer and processed
//
// A best guess needs to be used made to determine how many input frames to pull from the
// buffer. There are three possible outcomes: 1) exact; 2) underestimated; 3) overestimated.
// A best guess needs to be used made to determine how many input frames to pull from the buffer
// There are three possible outcomes: 1) exact; 2) underestimated; 3) overestimated
//
// When the guess is exactly correct or underestimated there is nothing special to handle - it'll be
// handled naturally by the loop.
// When the guess is exactly correct or underestimated there is nothing special to handle,
// it'll be handled naturally by the loop
//
// When the guess is overestimated, that's when it gets more complicated. In this case, any overflow
// needs to be stored in a buffer for later processing by the next read.
ma_uint32 estimatedInputFrameCount = (ma_uint32)(((float)audioBuffer->converter.resampler.sampleRateIn / audioBuffer->converter.resampler.sampleRateOut) * outputFramesToProcessThisIteration);
if (estimatedInputFrameCount == 0) estimatedInputFrameCount = 1; // Make sure at least one input frame is read.
// When the guess is overestimated, that's when it gets more complicated
// In this case, any overflow needs to be stored in a buffer for later processing by the next read
ma_uint32 estimatedInputFrameCount = (ma_uint32)(((float)audioBuffer->converter.resampler.sampleRateIn / audioBuffer->converter.resampler.sampleRateOut)*outputFramesToProcessThisIteration);
if (estimatedInputFrameCount == 0) estimatedInputFrameCount = 1; // Make sure at least one input frame is read
if (estimatedInputFrameCount > inputBufferFrameCap) estimatedInputFrameCount = inputBufferFrameCap;
ma_uint32 inputFramesInInternalFormatCount = ReadAudioBufferFramesInInternalFormat(audioBuffer, inputBuffer, estimatedInputFrameCount);
@ -2531,17 +2531,17 @@ static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, f
if (inputFramesInInternalFormatCount > inputFramesProcessedThisIteration)
{
// Getting here means the estimated input frame count was overestimated. The residual needs
// be stored for later use.
// Getting here means the estimated input frame count was overestimated
// The residual needs be stored for later use
ma_uint64 residualFrameCount = inputFramesInInternalFormatCount - inputFramesProcessedThisIteration;
// A safety check to make sure the capacity of the residual cache is not exceeded.
// A safety check to make sure the capacity of the residual cache is not exceeded
if (residualFrameCount > AUDIO_BUFFER_RESIDUAL_CAPACITY)
{
residualFrameCount = AUDIO_BUFFER_RESIDUAL_CAPACITY;
}
memcpy(audioBuffer->converterResidual, inputBuffer + inputFramesProcessedThisIteration*bpf, (size_t)(residualFrameCount * bpf));
memcpy(audioBuffer->converterResidual, inputBuffer + inputFramesProcessedThisIteration*bpf, (size_t)(residualFrameCount*bpf));
audioBuffer->converterResidualCount = (unsigned int)residualFrameCount;
}
@ -2559,7 +2559,7 @@ static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const
{
(void)pDevice;
// Mixing is basically just an accumulation, need to initialize the output buffer to 0
// Mixing is basically an accumulation, need to initialize the output buffer to 0
memset(pFramesOut, 0, frameCount*pDevice->playback.channels*ma_get_bytes_per_sample(pDevice->playback.format));
// Using a mutex here for thread-safety which makes things not real-time
@ -2577,7 +2577,7 @@ static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const
{
if (framesRead >= frameCount) break;
// Just read as much data as possible from the stream
// Read as much data as possible from the stream
ma_uint32 framesToRead = (frameCount - framesRead);
while (framesToRead > 0)
@ -2626,7 +2626,7 @@ static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const
}
else
{
// Should never get here, but just for safety,
// Should never get here, but for safety,
// move the cursor position back to the start and continue the loop
audioBuffer->frameCursorPos = 0;
continue;
@ -2651,7 +2651,7 @@ static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const
ma_mutex_unlock(&AUDIO.System.lock);
}
// Main mixing function, pretty simple in this project, just an accumulation
// Main mixing function, pretty simple in this project, only an accumulation
// NOTE: framesOut is both an input and an output, it is initially filled with zeros outside of this function
static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, AudioBuffer *buffer)
{
@ -2739,7 +2739,7 @@ static void UpdateAudioStreamInLockedState(AudioStream stream, const void *data,
}
else
{
// Just update whichever sub-buffer is processed
// Update whichever sub-buffer is processed
subBufferToUpdate = (stream.buffer->isSubBufferProcessed[0])? 0 : 1;
}