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Update raudio.c

pull/5619/head
Ray 3 days ago
parent
23f86689dc
commit
d0f899721b
1 changed files with 27 additions and 26 deletions
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  1. +27
    -26
      src/raudio.c

+ 27
- 26
src/raudio.c View File

@ -947,9 +947,9 @@ Sound LoadSoundFromWave(Wave wave)
if (wave.data != NULL)
{
// When using miniaudio we need to do our own mixing
// To simplify this we need convert the format of each sound to be consistent with
// the format used to open the playback AUDIO.System.device. We can do this two ways:
// When using miniaudio mixing needs to b done manually
// To simplify this, the format of each sound needs to be converted to be consistent with
// the format used to open the playback AUDIO.System.device. It can be done in two ways:
//
// 1) Convert the whole sound in one go at load time (here)
// 2) Convert the audio data in chunks at mixing time
@ -1394,9 +1394,9 @@ Music LoadMusicStream(const char *fileName)
// OGG bit rate defaults to 16 bit, it's enough for compressed format
music.stream = LoadAudioStream(info.sample_rate, 16, info.channels);
// WARNING: It seems this function returns length in frames, not samples, so we multiply by channels
// WARNING: It seems this function returns length in frames, not samples, so multiply by channels
music.frameCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData);
music.looping = true; // Looping enabled by default
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
else
@ -1435,8 +1435,8 @@ Music LoadMusicStream(const char *fileName)
{
music.ctxType = MUSIC_AUDIO_QOA;
music.ctxData = ctxQoa;
// NOTE: We are loading samples are 32bit float normalized data, so,
// we configure the output audio stream to also use float 32bit
// NOTE: Loading samples as 32bit float normalized data, so,
// configure the output audio stream to also use float 32bit
music.stream = LoadAudioStream(ctxQoa->info.samplerate, 32, ctxQoa->info.channels);
music.frameCount = ctxQoa->info.samples;
music.looping = true; // Looping enabled by default
@ -1487,7 +1487,7 @@ Music LoadMusicStream(const char *fileName)
music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, bits, AUDIO_DEVICE_CHANNELS);
music.frameCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm); // NOTE: Always 2 channels (stereo)
music.looping = true; // Looping enabled by default
jar_xm_reset(ctxXm); // Make sure we start at the beginning of the song
jar_xm_reset(ctxXm); // Make sure to start at the beginning of the song
musicLoaded = true;
}
else
@ -1588,7 +1588,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, const unsigned char *data,
// OGG bit rate defaults to 16 bit, it's enough for compressed format
music.stream = LoadAudioStream(info.sample_rate, 16, info.channels);
// WARNING: It seems this function returns length in frames, not samples, so we multiply by channels
// WARNING: It seems this function returns length in frames, not samples, so multiply by channels
music.frameCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData);
music.looping = true; // Looping enabled by default
musicLoaded = true;
@ -1634,8 +1634,9 @@ Music LoadMusicStreamFromMemory(const char *fileType, const unsigned char *data,
{
music.ctxType = MUSIC_AUDIO_QOA;
music.ctxData = ctxQoa;
// NOTE: We are loading samples are 32bit float normalized data, so,
// we configure the output audio stream to also use float 32bit
// NOTE: Loading samples are 32bit float normalized data, so,
// configure the output audio stream to also use float 32bit
music.stream = LoadAudioStream(ctxQoa->info.samplerate, 32, ctxQoa->info.channels);
music.frameCount = ctxQoa->info.samples;
music.looping = true; // Looping enabled by default
@ -1685,7 +1686,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, const unsigned char *data,
music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, bits, 2);
music.frameCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm); // NOTE: Always 2 channels (stereo)
music.looping = true; // Looping enabled by default
jar_xm_reset(ctxXm); // Make sure we start at the beginning of the song
jar_xm_reset(ctxXm); // Make sure to start at the beginning of the song
musicLoaded = true;
}
@ -1870,7 +1871,7 @@ void SeekMusicStream(Music music, float position)
int qoaFrame = positionInFrames/QOA_FRAME_LEN;
qoaplay_seek_frame((qoaplay_desc *)music.ctxData, qoaFrame); // Seeks to QOA frame, not PCM frame
// We need to compute QOA frame number and update positionInFrames
// Compute QOA frame number and update positionInFrames
positionInFrames = ((qoaplay_desc *)music.ctxData)->sample_position;
} break;
#endif
@ -1897,7 +1898,7 @@ void UpdateMusicStream(Music music)
unsigned int subBufferSizeInFrames = music.stream.buffer->sizeInFrames/2;
// On first call of this function we lazily pre-allocated a temp buffer to read audio files/memory data in
// On first call of this function, lazily pre-allocated a temp buffer to read audio files/memory data in
int frameSize = music.stream.channels*music.stream.sampleSize/8;
unsigned int pcmSize = subBufferSizeInFrames*frameSize;
@ -2024,7 +2025,7 @@ void UpdateMusicStream(Music music)
#if SUPPORT_FILEFORMAT_XM
case MUSIC_MODULE_XM:
{
// NOTE: Internally we consider 2 channels generation, so sampleCount/2
// NOTE: Internally considering 2 channels generation, so sampleCount/2
if (AUDIO_DEVICE_FORMAT == ma_format_f32) jar_xm_generate_samples((jar_xm_context_t *)music.ctxData, (float *)AUDIO.System.pcmBuffer, framesToStream);
else if (AUDIO_DEVICE_FORMAT == ma_format_s16) jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)AUDIO.System.pcmBuffer, framesToStream);
else if (AUDIO_DEVICE_FORMAT == ma_format_u8) jar_xm_generate_samples_8bit((jar_xm_context_t *)music.ctxData, (char *)AUDIO.System.pcmBuffer, framesToStream);
@ -2262,7 +2263,7 @@ void SetAudioStreamCallback(AudioStream stream, AudioCallback callback)
// Add processor to audio stream. Contrary to buffers, the order of processors is important
// The new processor must be added at the end. As there aren't supposed to be a lot of processors attached to
// a given stream, we iterate through the list to find the end. That way we don't need a pointer to the last element
// a given stream, iterate through the list to find the end. That way there is no need to keep a pointer to the last element
void AttachAudioStreamProcessor(AudioStream stream, AudioCallback process)
{
ma_mutex_lock(&AUDIO.System.lock);
@ -2396,20 +2397,20 @@ static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer,
if (currentSubBufferIndex > 1) return 0;
// Another thread can update the processed state of buffers, so
// we just take a copy here to try and avoid potential synchronization problems
// just 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];
ma_uint32 frameSizeInBytes = ma_get_bytes_per_frame(audioBuffer->converter.formatIn, audioBuffer->converter.channelsIn);
// Fill out every frame until we find a buffer that's marked as processed. Then fill the remainder with 0
// Fill out every frame until a buffer that's marked as processed is found, then fill the remainder with 0
ma_uint32 framesRead = 0;
while (1)
{
// We break from this loop differently depending on the buffer's usage
// - For static buffers, we simply fill as much data as we can
// - For streaming buffers we only fill half of the buffer that are processed
// Break from this loop differently depending on the buffer's usage
// - For static buffers, simply fill as much data as possible
// - For streaming buffers, only fill half of the buffer that are processed
// Unprocessed halves must keep their audio data in-tact
if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC)
{
@ -2464,8 +2465,8 @@ static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer,
{
memset((unsigned char *)framesOut + (framesRead*frameSizeInBytes), 0, totalFramesRemaining*frameSizeInBytes);
// For static buffers we can fill the remaining frames with silence for safety, but we don't want
// to report those frames as "read". The reason for this is that the caller uses the return value
// For static buffers, fill the remaining frames with silence for safety, but don't report those
// frames as "read"; The reason for this is that the caller uses the return value
// to know whether a non-looping sound has finished playback
if (audioBuffer->usage != AUDIO_BUFFER_USAGE_STATIC) framesRead += totalFramesRemaining;
}
@ -2558,7 +2559,7 @@ static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const
{
(void)pDevice;
// Mixing is basically just an accumulation, we need to initialize the output buffer to 0
// Mixing is basically just 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
@ -2576,7 +2577,7 @@ static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const
{
if (framesRead >= frameCount) break;
// Just read as much data as we can from the stream
// Just read as much data as possible from the stream
ma_uint32 framesToRead = (frameCount - framesRead);
while (framesToRead > 0)
@ -2615,7 +2616,7 @@ static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const
break;
}
// If we weren't able to read all the frames we requested, break
// If all the frames requested can't be read, break
if (framesJustRead < framesToReadRightNow)
{
if (!audioBuffer->looping)

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