REVIEWED: Simplified code to avoid extra functions calls
This commit is contained in:
Ray
parent
e0720a0a55
commit
7cbfca8bd1
304
src/raudio.c
304
src/raudio.c
@ -256,10 +256,10 @@ typedef struct tagBITMAPINFOHEADER {
|
||||
#ifndef AUDIO_DEVICE_CHANNELS
|
||||
#define AUDIO_DEVICE_CHANNELS 2 // Device output channels: stereo
|
||||
#endif
|
||||
|
||||
#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
|
||||
|
||||
#ifndef MAX_AUDIO_BUFFER_POOL_CHANNELS
|
||||
#define MAX_AUDIO_BUFFER_POOL_CHANNELS 16 // Audio pool channels
|
||||
#endif
|
||||
@ -322,7 +322,7 @@ struct rAudioBuffer {
|
||||
bool isSubBufferProcessed[2]; // SubBuffer processed (virtual double buffer)
|
||||
unsigned int sizeInFrames; // Total buffer size in frames
|
||||
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
|
||||
|
||||
@ -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);
|
||||
|
||||
#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
|
||||
#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)
|
||||
static bool IsFileExtension(const char *fileName, const char *ext); // Check file extension
|
||||
@ -630,7 +620,7 @@ void StopAudioBuffer(AudioBuffer *buffer)
|
||||
buffer->playing = false;
|
||||
buffer->paused = false;
|
||||
buffer->frameCursorPos = 0;
|
||||
buffer->totalFramesProcessed = 0;
|
||||
buffer->framesProcessed = 0;
|
||||
buffer->isSubBufferProcessed[0] = true;
|
||||
buffer->isSubBufferProcessed[1] = true;
|
||||
}
|
||||
@ -718,13 +708,10 @@ Wave LoadWave(const char *fileName)
|
||||
unsigned int fileSize = 0;
|
||||
unsigned char *fileData = LoadFileData(fileName, &fileSize);
|
||||
|
||||
if (fileData != NULL)
|
||||
{
|
||||
// Loading wave from memory data
|
||||
wave = LoadWaveFromMemory(GetFileExtension(fileName), fileData, fileSize);
|
||||
// Loading wave from memory data
|
||||
if (fileData != NULL) wave = LoadWaveFromMemory(GetFileExtension(fileName), fileData, fileSize);
|
||||
|
||||
RL_FREE(fileData);
|
||||
}
|
||||
RL_FREE(fileData);
|
||||
|
||||
return wave;
|
||||
}
|
||||
@ -739,18 +726,85 @@ Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int
|
||||
|
||||
if (false) { }
|
||||
#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
|
||||
#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
|
||||
#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
|
||||
#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
|
||||
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;
|
||||
}
|
||||
@ -846,7 +900,26 @@ bool ExportWave(Wave wave, const char *fileName)
|
||||
|
||||
if (false) { }
|
||||
#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
|
||||
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
|
||||
|
||||
unsigned int bits = 32;
|
||||
if (AUDIO_DEVICE_FORMAT == ma_format_s16)
|
||||
bits = 16;
|
||||
else if (AUDIO_DEVICE_FORMAT == ma_format_u8)
|
||||
bits = 8;
|
||||
if (AUDIO_DEVICE_FORMAT == ma_format_s16) bits = 16;
|
||||
else if (AUDIO_DEVICE_FORMAT == ma_format_u8) bits = 8;
|
||||
|
||||
// NOTE: Only stereo is supported for XM
|
||||
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
|
||||
|
||||
// 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;
|
||||
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;
|
||||
|
||||
@ -1656,23 +1727,10 @@ void UpdateMusicStream(Music music)
|
||||
#if defined(SUPPORT_FILEFORMAT_XM)
|
||||
case MUSIC_MODULE_XM:
|
||||
{
|
||||
switch (AUDIO_DEVICE_FORMAT)
|
||||
{
|
||||
case ma_format_f32:
|
||||
// NOTE: Internally this function considers 2 channels generation, so 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;
|
||||
}
|
||||
// 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);
|
||||
else if (AUDIO_DEVICE_FORMAT == ma_format_s16) jar_xm_generate_samples_16bit((jar_xm_context_t*)music.ctxData, (short *)pcm, 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);
|
||||
|
||||
} break;
|
||||
#endif
|
||||
@ -1764,7 +1822,7 @@ float GetMusicTimePlayed(Music music)
|
||||
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;
|
||||
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);
|
||||
}
|
||||
|
||||
@ -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);
|
||||
|
||||
// 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?
|
||||
// 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
|
||||
#if defined(RAUDIO_STANDALONE)
|
||||
// Check file extension
|
||||
|
||||
Loading…
Reference in New Issue
Block a user