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REVIEWED: Simplified code to avoid extra functions calls

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This commit is contained in:
Ray 2021-06-26 13:06:22 +02:00
parent e0720a0a55
commit 7cbfca8bd1
1 changed files with 109 additions and 195 deletions
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304
src/raudio.c
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@ -256,10 +256,10 @@ typedef struct tagBITMAPINFOHEADER {
#ifndef AUDIO_DEVICE_CHANNELS #ifndef AUDIO_DEVICE_CHANNELS
#define AUDIO_DEVICE_CHANNELS 2 // Device output channels: stereo #define AUDIO_DEVICE_CHANNELS 2 // Device output channels: stereo
#endif #endif
#ifndef AUDIO_DEVICE_SAMPLE_RATE #ifndef AUDIO_DEVICE_SAMPLE_RATE
#define AUDIO_DEVICE_SAMPLE_RATE 0 // Device output channels: stereo #define AUDIO_DEVICE_SAMPLE_RATE 0 // Device output sample rate
#endif #endif
#ifndef MAX_AUDIO_BUFFER_POOL_CHANNELS #ifndef MAX_AUDIO_BUFFER_POOL_CHANNELS
#define MAX_AUDIO_BUFFER_POOL_CHANNELS 16 // Audio pool channels #define MAX_AUDIO_BUFFER_POOL_CHANNELS 16 // Audio pool channels
#endif #endif
@ -322,7 +322,7 @@ struct rAudioBuffer {
bool isSubBufferProcessed[2]; // SubBuffer processed (virtual double buffer) bool isSubBufferProcessed[2]; // SubBuffer processed (virtual double buffer)
unsigned int sizeInFrames; // Total buffer size in frames unsigned int sizeInFrames; // Total buffer size in frames
unsigned int frameCursorPos; // Frame cursor position unsigned int frameCursorPos; // Frame cursor position
unsigned int totalFramesProcessed; // Total frames processed in this buffer (required for play timing) unsigned int framesProcessed; // Total frames processed in this buffer (required for play timing)
unsigned char *data; // Data buffer, on music stream keeps filling unsigned char *data; // Data buffer, on music stream keeps filling
@ -372,18 +372,8 @@ static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const
static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, float localVolume); static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, float localVolume);
#if defined(SUPPORT_FILEFORMAT_WAV) #if defined(SUPPORT_FILEFORMAT_WAV)
static Wave LoadWAV(const unsigned char *fileData, unsigned int fileSize); // Load WAV file
static int SaveWAV(Wave wave, const char *fileName); // Save wave data as WAV file static int SaveWAV(Wave wave, const char *fileName); // Save wave data as WAV file
#endif #endif
#if defined(SUPPORT_FILEFORMAT_OGG)
static Wave LoadOGG(const unsigned char *fileData, unsigned int fileSize); // Load OGG file
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
static Wave LoadFLAC(const unsigned char *fileData, unsigned int fileSize); // Load FLAC file
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
static Wave LoadMP3(const unsigned char *fileData, unsigned int fileSize); // Load MP3 file
#endif
#if defined(RAUDIO_STANDALONE) #if defined(RAUDIO_STANDALONE)
static bool IsFileExtension(const char *fileName, const char *ext); // Check file extension static bool IsFileExtension(const char *fileName, const char *ext); // Check file extension
@ -630,7 +620,7 @@ void StopAudioBuffer(AudioBuffer *buffer)
buffer->playing = false; buffer->playing = false;
buffer->paused = false; buffer->paused = false;
buffer->frameCursorPos = 0; buffer->frameCursorPos = 0;
buffer->totalFramesProcessed = 0; buffer->framesProcessed = 0;
buffer->isSubBufferProcessed[0] = true; buffer->isSubBufferProcessed[0] = true;
buffer->isSubBufferProcessed[1] = true; buffer->isSubBufferProcessed[1] = true;
} }
@ -718,13 +708,10 @@ Wave LoadWave(const char *fileName)
unsigned int fileSize = 0; unsigned int fileSize = 0;
unsigned char *fileData = LoadFileData(fileName, &fileSize); unsigned char *fileData = LoadFileData(fileName, &fileSize);
if (fileData != NULL) // Loading wave from memory data
{ if (fileData != NULL) wave = LoadWaveFromMemory(GetFileExtension(fileName), fileData, fileSize);
// Loading wave from memory data
wave = LoadWaveFromMemory(GetFileExtension(fileName), fileData, fileSize);
RL_FREE(fileData); RL_FREE(fileData);
}
return wave; return wave;
} }
@ -739,18 +726,85 @@ Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int
if (false) { } if (false) { }
#if defined(SUPPORT_FILEFORMAT_WAV) #if defined(SUPPORT_FILEFORMAT_WAV)
else if (TextIsEqual(fileExtLower, ".wav")) wave = LoadWAV(fileData, dataSize); else if (TextIsEqual(fileExtLower, ".wav"))
{
drwav wav = { 0 };
bool success = drwav_init_memory(&wav, fileData, dataSize, NULL);
if (success)
{
wave.sampleCount = (unsigned int)wav.totalPCMFrameCount*wav.channels;
wave.sampleRate = wav.sampleRate;
wave.sampleSize = 16;
wave.channels = wav.channels;
wave.data = (short *)RL_MALLOC(wave.sampleCount*sizeof(short));
// NOTE: We are forcing conversion to 16bit sample size on reading
drwav_read_pcm_frames_s16(&wav, wav.totalPCMFrameCount, wave.data);
}
else TRACELOG(LOG_WARNING, "WAVE: Failed to load WAV data");
drwav_uninit(&wav);
}
#endif #endif
#if defined(SUPPORT_FILEFORMAT_OGG) #if defined(SUPPORT_FILEFORMAT_OGG)
else if (TextIsEqual(fileExtLower, ".ogg")) wave = LoadOGG(fileData, dataSize); else if (TextIsEqual(fileExtLower, ".ogg"))
{
stb_vorbis *oggData = stb_vorbis_open_memory((unsigned char *)fileData, dataSize, NULL, NULL);
if (oggData != NULL)
{
stb_vorbis_info info = stb_vorbis_get_info(oggData);
wave.sampleRate = info.sample_rate;
wave.sampleSize = 16; // By default, ogg data is 16 bit per sample (short)
wave.channels = info.channels;
wave.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples(oggData)*info.channels; // Independent by channel
wave.data = (short *)RL_MALLOC(wave.sampleCount*sizeof(short));
// NOTE: Get the number of samples to process (be careful! we ask for number of shorts!)
stb_vorbis_get_samples_short_interleaved(oggData, info.channels, (short *)wave.data, wave.sampleCount);
stb_vorbis_close(oggData);
}
else TRACELOG(LOG_WARNING, "WAVE: Failed to load OGG data");
}
#endif #endif
#if defined(SUPPORT_FILEFORMAT_FLAC) #if defined(SUPPORT_FILEFORMAT_FLAC)
else if (TextIsEqual(fileExtLower, ".flac")) wave = LoadFLAC(fileData, dataSize); else if (TextIsEqual(fileExtLower, ".flac"))
{
unsigned long long int totalFrameCount = 0;
// NOTE: We are forcing conversion to 16bit sample size on reading
wave.data = drflac_open_memory_and_read_pcm_frames_s16(fileData, dataSize, &wave.channels, &wave.sampleRate, &totalFrameCount, NULL);
wave.sampleSize = 16;
if (wave.data != NULL) wave.sampleCount = (unsigned int)totalFrameCount*wave.channels;
else TRACELOG(LOG_WARNING, "WAVE: Failed to load FLAC data");
}
#endif #endif
#if defined(SUPPORT_FILEFORMAT_MP3) #if defined(SUPPORT_FILEFORMAT_MP3)
else if (TextIsEqual(fileExtLower, ".mp3")) wave = LoadMP3(fileData, dataSize); else if (TextIsEqual(fileExtLower, ".mp3"))
{
drmp3_config config = { 0 };
unsigned long long int totalFrameCount = 0;
// NOTE: We are forcing conversion to 32bit float sample size on reading
wave.data = drmp3_open_memory_and_read_pcm_frames_f32(fileData, dataSize, &config, &totalFrameCount, NULL);
wave.sampleSize = 32;
if (wave.data != NULL)
{
wave.channels = config.channels;
wave.sampleRate = config.sampleRate;
wave.sampleCount = (int)totalFrameCount*wave.channels;
}
else TRACELOG(LOG_WARNING, "WAVE: Failed to load MP3 data");
}
#endif #endif
else TRACELOG(LOG_WARNING, "WAVE: File format not supported"); else TRACELOG(LOG_WARNING, "WAVE: Data format not supported");
TRACELOG(LOG_INFO, "WAVE: Data loaded successfully (%i Hz, %i bit, %i channels)", wave.sampleRate, wave.sampleSize, wave.channels);
return wave; return wave;
} }
@ -846,7 +900,26 @@ bool ExportWave(Wave wave, const char *fileName)
if (false) { } if (false) { }
#if defined(SUPPORT_FILEFORMAT_WAV) #if defined(SUPPORT_FILEFORMAT_WAV)
else if (IsFileExtension(fileName, ".wav")) success = SaveWAV(wave, fileName); else if (IsFileExtension(fileName, ".wav"))
{
drwav wav = { 0 };
drwav_data_format format = { 0 };
format.container = drwav_container_riff;
format.format = DR_WAVE_FORMAT_PCM;
format.channels = wave.channels;
format.sampleRate = wave.sampleRate;
format.bitsPerSample = wave.sampleSize;
void *fileData = NULL;
size_t fileDataSize = 0;
success = drwav_init_memory_write(&wav, &fileData, &fileDataSize, &format, NULL);
if (success) success = (int)drwav_write_pcm_frames(&wav, wave.sampleCount/wave.channels, wave.data);
drwav_result result = drwav_uninit(&wav);
if (result == DRWAV_SUCCESS) success = SaveFileData(fileName, (unsigned char *)fileData, (unsigned int)fileDataSize);
drwav_free(fileData, NULL);
}
#endif #endif
else if (IsFileExtension(fileName, ".raw")) else if (IsFileExtension(fileName, ".raw"))
{ {
@ -1236,10 +1309,8 @@ Music LoadMusicStream(const char *fileName)
jar_xm_set_max_loop_count(ctxXm, 0); // Set infinite number of loops jar_xm_set_max_loop_count(ctxXm, 0); // Set infinite number of loops
unsigned int bits = 32; unsigned int bits = 32;
if (AUDIO_DEVICE_FORMAT == ma_format_s16) if (AUDIO_DEVICE_FORMAT == ma_format_s16) bits = 16;
bits = 16; else if (AUDIO_DEVICE_FORMAT == ma_format_u8) bits = 8;
else if (AUDIO_DEVICE_FORMAT == ma_format_u8)
bits = 8;
// NOTE: Only stereo is supported for XM // NOTE: Only stereo is supported for XM
music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, bits, AUDIO_DEVICE_CHANNELS); music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, bits, AUDIO_DEVICE_CHANNELS);
@ -1607,9 +1678,9 @@ void UpdateMusicStream(Music music)
int samplesCount = 0; // Total size of data streamed in L+R samples for xm floats, individual L or R for ogg shorts int samplesCount = 0; // Total size of data streamed in L+R samples for xm floats, individual L or R for ogg shorts
// TODO: Get the sampleLeft using totalFramesProcessed... but first, get total frames processed correctly... // TODO: Get the sampleLeft using framesProcessed... but first, get total frames processed correctly...
//ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels; //ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
int sampleLeft = music.sampleCount - (music.stream.buffer->totalFramesProcessed*music.stream.channels); int sampleLeft = music.sampleCount - (music.stream.buffer->framesProcessed*music.stream.channels);
if (music.ctxType == MUSIC_MODULE_XM && music.looping) sampleLeft = subBufferSizeInFrames*4; if (music.ctxType == MUSIC_MODULE_XM && music.looping) sampleLeft = subBufferSizeInFrames*4;
@ -1656,23 +1727,10 @@ void UpdateMusicStream(Music music)
#if defined(SUPPORT_FILEFORMAT_XM) #if defined(SUPPORT_FILEFORMAT_XM)
case MUSIC_MODULE_XM: case MUSIC_MODULE_XM:
{ {
switch (AUDIO_DEVICE_FORMAT) // NOTE: Internally we consider 2 channels generation, so samplesCount/2
{ if (AUDIO_DEVICE_FORMAT == ma_format_f32) jar_xm_generate_samples((jar_xm_context_t*)music.ctxData, (float *)pcm, samplesCount/2);
case ma_format_f32: else if (AUDIO_DEVICE_FORMAT == ma_format_s16) jar_xm_generate_samples_16bit((jar_xm_context_t*)music.ctxData, (short *)pcm, samplesCount/2);
// NOTE: Internally this function considers 2 channels generation, so samplesCount/2 else if (AUDIO_DEVICE_FORMAT == ma_format_u8) jar_xm_generate_samples_8bit((jar_xm_context_t*)music.ctxData, (char *)pcm, samplesCount/2);
jar_xm_generate_samples((jar_xm_context_t*)music.ctxData, (float *)pcm, samplesCount/2);
break;
case ma_format_s16:
// NOTE: Internally this function considers 2 channels generation, so samplesCount/2
jar_xm_generate_samples_16bit((jar_xm_context_t*)music.ctxData, (short *)pcm, samplesCount/2);
break;
case ma_format_u8:
// NOTE: Internally this function considers 2 channels generation, so samplesCount/2
jar_xm_generate_samples_8bit((jar_xm_context_t*)music.ctxData, (char *)pcm, samplesCount/2);
break;
}
} break; } break;
#endif #endif
@ -1764,7 +1822,7 @@ float GetMusicTimePlayed(Music music)
if (music.stream.buffer != NULL) if (music.stream.buffer != NULL)
{ {
//ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels; //ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
unsigned int samplesPlayed = music.stream.buffer->totalFramesProcessed*music.stream.channels; unsigned int samplesPlayed = music.stream.buffer->framesProcessed*music.stream.channels;
secondsPlayed = (float)samplesPlayed/(music.stream.sampleRate*music.stream.channels); secondsPlayed = (float)samplesPlayed/(music.stream.sampleRate*music.stream.channels);
} }
@ -1839,7 +1897,7 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
unsigned char *subBuffer = stream.buffer->data + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate); unsigned char *subBuffer = stream.buffer->data + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate);
// TODO: Get total frames processed on this buffer... DOES NOT WORK. // TODO: Get total frames processed on this buffer... DOES NOT WORK.
stream.buffer->totalFramesProcessed += subBufferSizeInFrames; stream.buffer->framesProcessed += subBufferSizeInFrames;
// Does this API expect a whole buffer to be updated in one go? // Does this API expect a whole buffer to be updated in one go?
// Assuming so, but if not will need to change this logic. // Assuming so, but if not will need to change this logic.
@ -2166,150 +2224,6 @@ static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 fr
} }
} }
#if defined(SUPPORT_FILEFORMAT_WAV)
// Load WAV file data into Wave structure
// NOTE: Using dr_wav library
static Wave LoadWAV(const unsigned char *fileData, unsigned int fileSize)
{
Wave wave = { 0 };
drwav wav = { 0 };
bool success = drwav_init_memory(&wav, fileData, fileSize, NULL);
if (success)
{
wave.sampleCount = (unsigned int)wav.totalPCMFrameCount*wav.channels;
wave.sampleRate = wav.sampleRate;
wave.sampleSize = 16; // NOTE: We are forcing conversion to 16bit
wave.channels = wav.channels;
wave.data = (short *)RL_MALLOC(wave.sampleCount*sizeof(short));
drwav_read_pcm_frames_s16(&wav, wav.totalPCMFrameCount, wave.data);
}
else TRACELOG(LOG_WARNING, "WAVE: Failed to load WAV data");
drwav_uninit(&wav);
return wave;
}
// Save wave data as WAV file
// NOTE: Using dr_wav library
static int SaveWAV(Wave wave, const char *fileName)
{
int success = false;
drwav wav = { 0 };
drwav_data_format format = { 0 };
format.container = drwav_container_riff;
format.format = DR_WAVE_FORMAT_PCM;
format.channels = wave.channels;
format.sampleRate = wave.sampleRate;
format.bitsPerSample = wave.sampleSize;
void *fileData = NULL;
size_t fileDataSize = 0;
success = drwav_init_memory_write(&wav, &fileData, &fileDataSize, &format, NULL);
if (success) success = (int)drwav_write_pcm_frames(&wav, wave.sampleCount/wave.channels, wave.data);
drwav_result result = drwav_uninit(&wav);
if (result == DRWAV_SUCCESS) success = SaveFileData(fileName, (unsigned char *)fileData, (unsigned int)fileDataSize);
drwav_free(fileData, NULL);
return success;
}
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
// Load OGG file data into Wave structure
// NOTE: Using stb_vorbis library
static Wave LoadOGG(const unsigned char *fileData, unsigned int fileSize)
{
Wave wave = { 0 };
stb_vorbis *oggData = stb_vorbis_open_memory((unsigned char *)fileData, fileSize, NULL, NULL);
if (oggData != NULL)
{
stb_vorbis_info info = stb_vorbis_get_info(oggData);
wave.sampleRate = info.sample_rate;
wave.sampleSize = 16; // 16 bit per sample (short)
wave.channels = info.channels;
wave.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples(oggData)*info.channels; // Independent by channel
float totalSeconds = stb_vorbis_stream_length_in_seconds(oggData);
if (totalSeconds > 10) TRACELOG(LOG_WARNING, "WAVE: OGG audio length larger than 10 seconds (%f sec.), that's a big file in memory, consider music streaming", totalSeconds);
wave.data = (short *)RL_MALLOC(wave.sampleCount*sizeof(short));
// NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
stb_vorbis_get_samples_short_interleaved(oggData, info.channels, (short *)wave.data, wave.sampleCount);
TRACELOG(LOG_INFO, "WAVE: OGG data loaded successfully (%i Hz, %i bit, %s)", wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
stb_vorbis_close(oggData);
}
else TRACELOG(LOG_WARNING, "WAVE: Failed to load OGG data");
return wave;
}
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
// Load FLAC file data into Wave structure
// NOTE: Using dr_flac library
static Wave LoadFLAC(const unsigned char *fileData, unsigned int fileSize)
{
Wave wave = { 0 };
// Decode the entire FLAC file in one go
unsigned long long int totalFrameCount = 0;
wave.data = drflac_open_memory_and_read_pcm_frames_s16(fileData, fileSize, &wave.channels, &wave.sampleRate, &totalFrameCount, NULL);
if (wave.data != NULL)
{
wave.sampleCount = (unsigned int)totalFrameCount*wave.channels;
wave.sampleSize = 16;
TRACELOG(LOG_INFO, "WAVE: FLAC data loaded successfully (%i Hz, %i bit, %s)", wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
}
else TRACELOG(LOG_WARNING, "WAVE: Failed to load FLAC data");
return wave;
}
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
// Load MP3 file data into Wave structure
// NOTE: Using dr_mp3 library
static Wave LoadMP3(const unsigned char *fileData, unsigned int fileSize)
{
Wave wave = { 0 };
drmp3_config config = { 0 };
// Decode the entire MP3 file in one go
unsigned long long int totalFrameCount = 0;
wave.data = drmp3_open_memory_and_read_pcm_frames_f32(fileData, fileSize, &config, &totalFrameCount, NULL);
if (wave.data != NULL)
{
wave.channels = config.channels;
wave.sampleRate = config.sampleRate;
wave.sampleCount = (int)totalFrameCount*wave.channels;
wave.sampleSize = 32;
// NOTE: Only support up to 2 channels (mono, stereo)
// TODO: Really?
if (wave.channels > 2) TRACELOG(LOG_WARNING, "WAVE: MP3 channels number (%i) not supported", wave.channels);
TRACELOG(LOG_INFO, "WAVE: MP3 file loaded successfully (%i Hz, %i bit, %s)", wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
}
else TRACELOG(LOG_WARNING, "WAVE: Failed to load MP3 data");
return wave;
}
#endif
// Some required functions for audio standalone module version // Some required functions for audio standalone module version
#if defined(RAUDIO_STANDALONE) #if defined(RAUDIO_STANDALONE)
// Check file extension // Check file extension