Archivist
/
raylib-src
mirror of https://github.com/raysan5/raylib
1
0
Fork 0
Code Issues 0 Releases 24 Wiki Activity
Browse Source

WARNING: BREAKING: Use `frameCount` on audio 

This is a big change for optimization and a more professional understanding of audio. Instead of dealing with samples, now we deal with frames, like miniaudio does, so, avoiding continuous conversions from samples to frames.
pull/1925/head
Ray 4 years ago
parent
e203fb58c6
commit
1b4c58b66f
3 changed files with 105 additions and 118 deletions
Split View
Diff Options
Show Stats Download Patch File Download Diff File
  1. +75
    -85
      src/raudio.c
  2. +24
    -27
      src/raudio.h
  3. +6
    -6
      src/raylib.h

+ 75
- 85
src/raudio.c View File

@ -728,11 +728,11 @@ Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int
if (success)
{
wave.sampleCount = (unsigned int)wav.totalPCMFrameCount*wav.channels;
wave.frameCount = (unsigned int)wav.totalPCMFrameCount;
wave.sampleRate = wav.sampleRate;
wave.sampleSize = 16;
wave.channels = wav.channels;
wave.data = (short *)RL_MALLOC(wave.sampleCount*sizeof(short));
wave.data = (short *)RL_MALLOC(wave.frameCount*wave.channels*sizeof(short));
// NOTE: We are forcing conversion to 16bit sample size on reading
drwav_read_pcm_frames_s16(&wav, wav.totalPCMFrameCount, wave.data);
@ -754,11 +754,11 @@ Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int
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)o">*info.channels; // Independent by channel
wave.data = (short *)RL_MALLOC(wave.sampleCount*sizeof(short));
wave.frameCount = (unsigned int)stb_vorbis_stream_length_in_samples(oggData)p">; // NOTE: It returns frames!
wave.data = (short *)RL_MALLOC(wave.frameCount*wave.channels*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);
// NOTE: Get the number of samples to process (be careful! we ask for number of shortsp">, not bytes!)
stb_vorbis_get_samples_short_interleaved(oggData, info.channels, (short *)wave.data, wave.frameCount*wave.channels);
stb_vorbis_close(oggData);
}
else TRACELOG(LOG_WARNING, "WAVE: Failed to load OGG data");
@ -773,7 +773,7 @@ Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int
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;
if (wave.data != NULL) wave.frameCount = (unsigned int)totalFrameCount;
else TRACELOG(LOG_WARNING, "WAVE: Failed to load FLAC data");
}
#endif
@ -791,7 +791,7 @@ Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int
{
wave.channels = config.channels;
wave.sampleRate = config.sampleRate;
wave.sampleCount = (int)totalFrameCount*wave.channels;
wave.frameCount = (int)totalFrameCount;
}
else TRACELOG(LOG_WARNING, "WAVE: Failed to load MP3 data");
@ -835,7 +835,7 @@ Sound LoadSoundFromWave(Wave wave)
// First option has been selected, format conversion is done on the loading stage.
// The downside is that it uses more memory if the original sound is u8 or s16.
ma_format formatIn = ((wave.sampleSize == 8)? ma_format_u8 : ((wave.sampleSize == 16)? ma_format_s16 : ma_format_f32));
ma_uint32 frameCountIn = wave.sampleCount/wave.channels;
ma_uint32 frameCountIn = wave.frameCount;
ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, NULL, frameCountIn, formatIn, wave.channels, wave.sampleRate);
if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed to get frame count for format conversion");
@ -850,7 +850,7 @@ Sound LoadSoundFromWave(Wave wave)
frameCount = (ma_uint32)ma_convert_frames(audioBuffer->data, frameCount, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, wave.data, frameCountIn, formatIn, wave.channels, wave.sampleRate);
if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed format conversion");
sound.sampleCount = frameCount*AUDIO_DEVICE_CHANNELS;
sound.frameCount = frameCount;
sound.stream.sampleRate = AUDIO.System.device.sampleRate;
sound.stream.sampleSize = 32;
sound.stream.channels = AUDIO_DEVICE_CHANNELS;
@ -908,7 +908,7 @@ bool ExportWave(Wave wave, const char *fileName)
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);
if (success) success = (int)drwav_write_pcm_frames(&wav, wave.frameCount, wave.data);
drwav_result result = drwav_uninit(&wav);
if (result == DRWAV_SUCCESS) success = SaveFileData(fileName, (unsigned char *)fileData, (unsigned int)fileDataSize);
@ -920,7 +920,7 @@ bool ExportWave(Wave wave, const char *fileName)
{
// Export raw sample data (without header)
// NOTE: It's up to the user to track wave parameters
success = SaveFileData(fileName, wave.data, wave.sampleCount*wave.sampleSize/8);
success = SaveFileData(fileName, wave.data, wave.frameCount*wave.channels*wave.sampleSize/8);
}
if (success) TRACELOG(LOG_INFO, "FILEIO: [%s] Wave data exported successfully", fileName);
@ -938,7 +938,7 @@ bool ExportWaveAsCode(Wave wave, const char *fileName)
#define TEXT_BYTES_PER_LINE 20
#endif
int waveDataSize = wave.sampleCount*wave.channels*wave.sampleSize/8;
int waveDataSize = wave.frameCount*wave.channels*wave.sampleSize/8;
// NOTE: Text data buffer size is estimated considering wave data size in bytes
// and requiring 6 char bytes for every byte: "0x00, "
@ -966,7 +966,8 @@ bool ExportWaveAsCode(Wave wave, const char *fileName)
#endif
bytesCount += sprintf(txtData + bytesCount, "// Wave data information\n");
bytesCount += sprintf(txtData + bytesCount, "#define %s_SAMPLE_COUNT %u\n", varFileName, wave.sampleCount);
bytesCount += sprintf(txtData + bytesCount, "#define %s_FRAME_COUNT %u\n", varFileName, wave.frameCount);
bytesCount += sprintf(txtData + bytesCount, "#define %s_SAMPLE_COUNT %u\n", varFileName, wave.frameCount*wave.channels);
bytesCount += sprintf(txtData + bytesCount, "#define %s_SAMPLE_RATE %u\n", varFileName, wave.sampleRate);
bytesCount += sprintf(txtData + bytesCount, "#define %s_SAMPLE_SIZE %u\n", varFileName, wave.sampleSize);
bytesCount += sprintf(txtData + bytesCount, "#define %s_CHANNELS %u\n\n", varFileName, wave.channels);
@ -1111,7 +1112,7 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
ma_format formatIn = ((wave->sampleSize == 8)? ma_format_u8 : ((wave->sampleSize == 16)? ma_format_s16 : ma_format_f32));
ma_format formatOut = ((sampleSize == 8)? ma_format_u8 : ((sampleSize == 16)? ma_format_s16 : ma_format_f32));
ma_uint32 frameCountIn = wave->sampleCount/wave->channels;
ma_uint32 frameCountIn = wave->frameCount;
ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, formatOut, channels, sampleRate, NULL, frameCountIn, formatIn, wave->channels, wave->sampleRate);
if (frameCount == 0)
@ -1129,7 +1130,7 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
return;
}
wave->sampleCount = frameCount*channels;
wave->frameCount = frameCount;
wave->sampleSize = sampleSize;
wave->sampleRate = sampleRate;
wave->channels = channels;
@ -1142,14 +1143,14 @@ Wave WaveCopy(Wave wave)
{
Wave newWave = { 0 };
newWave.data = RL_MALLOC(wave.sampleCount*wave.sampleSize/8);
newWave.data = RL_MALLOC(wave.frameCount*wave.channels*wave.sampleSize/8);
if (newWave.data != NULL)
{
// NOTE: Size must be provided in bytes
memcpy(newWave.data, wave.data, wave.sampleCount*wave.sampleSize/8);
memcpy(newWave.data, wave.data, wave.frameCount*wave.channels*wave.sampleSize/8);
newWave.sampleCount = wave.sampleCount;
newWave.frameCount = wave.frameCount;
newWave.sampleRate = wave.sampleRate;
newWave.sampleSize = wave.sampleSize;
newWave.channels = wave.channels;
@ -1163,7 +1164,7 @@ Wave WaveCopy(Wave wave)
void WaveCrop(Wave *wave, int initSample, int finalSample)
{
if ((initSample >= 0) && (initSample < finalSample) &&
(finalSample > 0) && ((unsigned int)finalSample < wave->sampleCount))
(finalSample > 0) && ((unsigned int)finalSample < p">(wave->frameCount*wave->channels)))
{
int sampleCount = finalSample - initSample;
@ -1182,11 +1183,11 @@ void WaveCrop(Wave *wave, int initSample, int finalSample)
// NOTE 2: Sample data allocated should be freed with UnloadWaveSamples()
float *LoadWaveSamples(Wave wave)
{
float *samples = (float *)RL_MALLOC(wave.sampleCount*sizeof(float));
float *samples = (float *)RL_MALLOC(wave.frameCount*wave.channels*sizeof(float));
// NOTE: sampleCount is the total number of interlaced samples (including channels)
for (unsigned int i = 0; i < wave.sampleCount; i++)
for (unsigned int i = 0; i < wave.frameCount*wave.channels; i++)
{
if (wave.sampleSize == 8) samples[i] = (float)(((unsigned char *)wave.data)[i] - 127)/256.0f;
else if (wave.sampleSize == 16) samples[i] = (float)(((short *)wave.data)[i])/32767.0f;
@ -1228,7 +1229,7 @@ Music LoadMusicStream(const char *fileName)
if (ctxWav->bitsPerSample == 24) sampleSize = 16; // Forcing conversion to s16 on UpdateMusicStream()
music.stream = LoadAudioStream(ctxWav->sampleRate, sampleSize, ctxWav->channels);
music.sampleCount = (unsigned int)ctxWav->totalPCMFrameCount*ctxWav->channels;
music.frameCount = (unsigned int)ctxWav->totalPCMFrameCount;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
@ -1249,7 +1250,7 @@ Music LoadMusicStream(const char *fileName)
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
music.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData)*info.channels;
music.frameCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData);
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
@ -1266,7 +1267,7 @@ Music LoadMusicStream(const char *fileName)
drflac *ctxFlac = (drflac *)music.ctxData;
music.stream = LoadAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
music.sampleCount = (unsigned int)ctxFlac->totalPCMFrameCount*ctxFlac->channels;
music.frameCount = (unsigned int)ctxFlac->totalPCMFrameCount;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
@ -1284,7 +1285,7 @@ Music LoadMusicStream(const char *fileName)
if (result > 0)
{
music.stream = LoadAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels);
music.sampleCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels;
music.frameCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3);
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
@ -1309,9 +1310,9 @@ Music LoadMusicStream(const char *fileName)
// NOTE: Only stereo is supported for XM
music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, bits, AUDIO_DEVICE_CHANNELS);
music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm)o">*2; // 2 channels
music.frameCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm)p">; // 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 we start at the beginning of the song
musicLoaded = true;
}
}
@ -1330,7 +1331,7 @@ Music LoadMusicStream(const char *fileName)
{
// NOTE: Only stereo is supported for MOD
music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, 16, AUDIO_DEVICE_CHANNELS);
music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod)o">*2; // 2 channels
music.frameCount = (unsigned int)jar_mod_max_samples(ctxMod)p">; // NOTE: Always 2 channels (stereo)
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
@ -1367,10 +1368,10 @@ Music LoadMusicStream(const char *fileName)
{
// Show some music stream info
TRACELOG(LOG_INFO, "FILEIO: [%s] Music file loaded successfully", fileName);
TRACELOG(LOG_INFO, " > Total samples: %i", music.sampleCount);
TRACELOG(LOG_INFO, " > Sample rate: %i Hz", music.stream.sampleRate);
TRACELOG(LOG_INFO, " > Sample size: %i bits", music.stream.sampleSize);
TRACELOG(LOG_INFO, " > Channels: %i (%s)", music.stream.channels, (music.stream.channels == 1)? "Mono" : (music.stream.channels == 2)? "Stereo" : "Multi");
TRACELOG(LOG_INFO, " > Total frames: %i", music.frameCount);
}
return music;
@ -1402,7 +1403,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char *data, int d
if (ctxWav->bitsPerSample == 24) sampleSize = 16; // Forcing conversion to s16 on UpdateMusicStream()
music.stream = LoadAudioStream(ctxWav->sampleRate, sampleSize, ctxWav->channels);
music.sampleCount = (unsigned int)ctxWav->totalPCMFrameCount*ctxWav->channels;
music.frameCount = (unsigned int)ctxWav->totalPCMFrameCount;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
@ -1419,7 +1420,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char *data, int d
drflac *ctxFlac = (drflac *)music.ctxData;
music.stream = LoadAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
music.sampleCount = (unsigned int)ctxFlac->totalPCMFrameCount*ctxFlac->channels;
music.frameCount = (unsigned int)ctxFlac->totalPCMFrameCount;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
@ -1437,7 +1438,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char *data, int d
if (success)
{
music.stream = LoadAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels);
music.sampleCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels;
music.frameCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3);
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
@ -1459,7 +1460,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char *data, int d
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
music.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData)*info.channels;
music.frameCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData);
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
@ -1483,7 +1484,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char *data, int d
// NOTE: Only stereo is supported for XM
music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, bits, 2);
music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm)o">*2; // 2 channels
music.frameCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm)p">; // 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
@ -1521,7 +1522,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char *data, int d
// NOTE: Only stereo is supported for MOD
music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, 16, 2);
music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod)o">*2; // 2 channels
music.frameCount = (unsigned int)jar_mod_max_samples(ctxMod)p">; // NOTE: Always 2 channels (stereo)
music.looping = true; // Looping enabled by default
musicLoaded = true;
@ -1561,10 +1562,10 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char *data, int d
{
// Show some music stream info
TRACELOG(LOG_INFO, "FILEIO: Music data loaded successfully");
TRACELOG(LOG_INFO, " > Total samples: %i", music.sampleCount);
TRACELOG(LOG_INFO, " > Sample rate: %i Hz", music.stream.sampleRate);
TRACELOG(LOG_INFO, " > Sample size: %i bits", music.stream.sampleSize);
TRACELOG(LOG_INFO, " > Channels: %i (%s)", music.stream.channels, (music.stream.channels == 1)? "Mono" : (music.stream.channels == 2)? "Stereo" : "Multi");
TRACELOG(LOG_INFO, " > Total frames: %i", music.frameCount);
}
return music;
@ -1660,29 +1661,22 @@ void UpdateMusicStream(Music music)
{
if (music.stream.buffer == NULL) return;
#if defined(SUPPORT_FILEFORMAT_XM)
if (music.ctxType == MUSIC_MODULE_XM) jar_xm_set_max_loop_count(music.ctxData, music.looping ? 0 : 1);
#endif
bool streamEnding = false;
unsigned int subBufferSizeInFrames = music.stream.buffer->sizeInFrames/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);
int samplesCount = 0; // Total size of data streamed in L+R samples for xm floats, individual L or R for ogg shorts
int frameCountToStream = 0; // Total size of data in frames to be streamed
// TODO: Get the sampleLeft using framesProcessed... but first, get total frames processed correctly...
// TODO: Get the framesLeft 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->framesProcessed*music.stream.channels);
if (music.ctxType == MUSIC_MODULE_XM && music.looping) sampleLeft = subBufferSizeInFrames*4;
int framesLeft = music.frameCount - music.stream.buffer->framesProcessed;
while (IsAudioStreamProcessed(music.stream))
{
if (p">(sampleLeft/music.stream.channels) >= subBufferSizeInFrames) samplesCount = subBufferSizeInFrames*music.stream.channels;
else samplesCount = sampleLeft;
if (n">framesLeft >= subBufferSizeInFrames) frameCountToStream = subBufferSizeInFrames;
else frameCountToStream = framesLeft;
switch (music.ctxType)
{
@ -1690,8 +1684,8 @@ void UpdateMusicStream(Music music)
case MUSIC_AUDIO_WAV:
{
// NOTE: Returns the number of samples to process (not required)
if (music.stream.sampleSize == 16) drwav_read_pcm_frames_s16((drwav *)music.ctxData, samplesCount/music.stream.channels, (short *)pcm);
else if (music.stream.sampleSize == 32) drwav_read_pcm_frames_f32((drwav *)music.ctxData, samplesCount/music.stream.channels, (float *)pcm);
if (music.stream.sampleSize == 16) drwav_read_pcm_frames_s16((drwav *)music.ctxData, frameCountToStream, (short *)pcm);
else if (music.stream.sampleSize == 32) drwav_read_pcm_frames_f32((drwav *)music.ctxData, frameCountToStream, (float *)pcm);
} break;
#endif
@ -1699,7 +1693,7 @@ void UpdateMusicStream(Music music)
case MUSIC_AUDIO_OGG:
{
// NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
stb_vorbis_get_samples_short_interleaved((stb_vorbis *)music.ctxData, music.stream.channels, (short *)pcm, samplesCount);
stb_vorbis_get_samples_short_interleaved((stb_vorbis *)music.ctxData, music.stream.channels, (short *)pcm, frameCountToStream*music.stream.channels);
} break;
#endif
@ -1707,15 +1701,14 @@ void UpdateMusicStream(Music music)
case MUSIC_AUDIO_FLAC:
{
// NOTE: Returns the number of samples to process (not required)
drflac_read_pcm_frames_s16((drflac *)music.ctxData, samplesCount, (short *)pcm);
drflac_read_pcm_frames_s16((drflac *)music.ctxData, frameCountToStream*music.stream.channels, (short *)pcm);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
case MUSIC_AUDIO_MP3:
{
// NOTE: samplesCount, actually refers to framesCount and returns the number of frames processed
drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, samplesCount/music.stream.channels, (float *)pcm);
drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, frameCountToStream, (float *)pcm);
} break;
#endif
@ -1723,9 +1716,9 @@ void UpdateMusicStream(Music music)
case MUSIC_MODULE_XM:
{
// 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);
if (AUDIO_DEVICE_FORMAT == ma_format_f32) jar_xm_generate_samples((jar_xm_context_t *)music.ctxData, (float *)pcm, frameCountToStream);
else if (AUDIO_DEVICE_FORMAT == ma_format_s16) jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)pcm, frameCountToStream);
else if (AUDIO_DEVICE_FORMAT == ma_format_u8) jar_xm_generate_samples_8bit((jar_xm_context_t *)music.ctxData, (char *)pcm, frameCountToStream);
} break;
#endif
@ -1733,22 +1726,17 @@ void UpdateMusicStream(Music music)
case MUSIC_MODULE_MOD:
{
// NOTE: 3rd parameter (nbsample) specify the number of stereo 16bits samples you want, so sampleCount/2
jar_mod_fillbuffer((jar_mod_context_t *)music.ctxData, (short *)pcm, samplesCount/2, 0);
jar_mod_fillbuffer((jar_mod_context_t *)music.ctxData, (short *)pcm, frameCountToStream, 0);
} break;
#endif
default: break;
}
UpdateAudioStream(music.stream, pcm, samplesCount);
UpdateAudioStream(music.stream, pcm, frameCountToStream);
framesLeft -= frameCountToStream;
if ((music.ctxType == MUSIC_MODULE_XM) || music.ctxType == MUSIC_MODULE_MOD)
{
if (samplesCount > 1) sampleLeft -= samplesCount/2;
else sampleLeft -= samplesCount;
}
else sampleLeft -= samplesCount;
if (sampleLeft <= 0)
if (framesLeft <= 0)
{
streamEnding = true;
break;
@ -1795,7 +1783,7 @@ float GetMusicTimeLength(Music music)
{
float totalSeconds = 0.0f;
totalSeconds = (float)music.sampleCount/(music.stream.sampleRate*music.stream.channels);
totalSeconds = (float)music.frameCount/music.stream.sampleRate;
return totalSeconds;
}
@ -1803,22 +1791,24 @@ float GetMusicTimeLength(Music music)
// Get current music time played (in seconds)
float GetMusicTimePlayed(Music music)
{
#if defined(SUPPORT_FILEFORMAT_XM)
if (music.ctxType == MUSIC_MODULE_XM)
{
uint64_t samples = 0;
jar_xm_get_position(music.ctxData, NULL, NULL, NULL, &samples);
samples = samples % (music.sampleCount);
return (float)(samples)/(music.stream.sampleRate*music.stream.channels);
}
#endif
float secondsPlayed = 0.0f;
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->framesProcessed*music.stream.channels;
secondsPlayed = (float)samplesPlayed/(music.stream.sampleRate*music.stream.channels);
#if defined(SUPPORT_FILEFORMAT_XM)
if (music.ctxType == MUSIC_MODULE_XM)
{
uint64_t framesPlayed = 0;
jar_xm_get_position(music.ctxData, NULL, NULL, NULL, &framesPlayed);
secondsPlayed = (float)framesPlayed/music.stream.sampleRate;
}
else
#endif
{
//ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
unsigned int framesPlayed = music.stream.buffer->framesProcessed;
secondsPlayed = (float)framesPlayed/music.stream.sampleRate;
}
}
return secondsPlayed;
@ -1867,7 +1857,7 @@ void UnloadAudioStream(AudioStream stream)
// Update audio stream buffers with data
// NOTE 1: Only updates one buffer of the stream source: unqueue -> update -> queue
// NOTE 2: To unqueue a buffer it needs to be processed: IsAudioStreamProcessed()
void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
void UpdateAudioStream(AudioStream stream, const void *data, int frameCount)
{
if (stream.buffer != NULL)
{
@ -1896,11 +1886,11 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
// Does this API expect a whole buffer to be updated in one go?
// Assuming so, but if not will need to change this logic.
if (subBufferSizeInFrames >= (ma_uint32)samplesCount/stream.channels)
if (subBufferSizeInFrames >= (ma_uint32)frameCount)
{
ma_uint32 framesToWrite = subBufferSizeInFrames;
if (framesToWrite > (p">(ma_uint32)samplesCount/stream.channels)) framesToWrite = (ma_uint32)samplesCount/stream.channels;
if (framesToWrite > (ma_uint32)frameCount) framesToWrite = (ma_uint32)frameCount;
ma_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8);
memcpy(subBuffer, data, bytesToWrite);

+ 24
- 27
src/raudio.h View File

@ -78,49 +78,42 @@
#endif
#endif
// Wave type, defines audio wave data
// Wave, audio wave data
typedef struct Wave {
unsigned int sampleCount; // Total number of samples
unsigned int sampleRate; // Frequency (samples per second)
unsigned int sampleSize; // Bit depth (bits per sample): 8, 16, 32 (24 not supported)
unsigned int channels; // Number of channels (1-mono, 2-stereo)
void *data; // Buffer data pointer
unsigned int frameCount; // Total number of frames (considering channels)
unsigned int sampleRate; // Frequency (samples per second)
unsigned int sampleSize; // Bit depth (bits per sample): 8, 16, 32 (24 not supported)
unsigned int channels; // Number of channels (1-mono, 2-stereo, ...)
void *data; // Buffer data pointer
} Wave;
typedef struct rAudioBuffer rAudioBuffer;
// Audio stream type
// NOTE: Useful to create custom audio streams not bound to a specific file
// AudioStream, custom audio stream
typedef struct AudioStream {
unsigned int sampleRate; // Frequency (samples per second)
unsigned int sampleSize; // Bit depth (bits per sample): 8, 16, 32 (24 not supported)
unsigned int channels; // Number of channels (1-mono, 2-stereo)
rAudioBuffer *buffer; // Pointer to internal data used by the audio system
rAudioBuffer *buffer; // Pointer to internal data used by the audio system
unsigned int sampleRate; // Frequency (samples per second)
unsigned int sampleSize; // Bit depth (bits per sample): 8, 16, 32 (24 not supported)
unsigned int channels; // Number of channels (1-mono, 2-stereo, ...)
} AudioStream;
// Sound source type
// Sound
typedef struct Sound {
kt">unsigned int sampleCount; // Total number of samples
n">AudioStream stream; // Audio stream
n">AudioStream stream; // Audio stream
kt">unsigned int frameCount; // Total number of frames (considering channels)
} Sound;
// Music stream type (audio file streaming from memory)
// NOTE: Anything longer than ~10 seconds should be streamed
// Music, audio stream, anything longer than ~10 seconds should be streamed
typedef struct Music {
int ctxType; // Type of music context (audio filetype)
void *ctxData; // Audio context data, depends on type
AudioStream stream; // Audio stream
unsigned int frameCount; // Total number of frames (considering channels)
bool looping; // Music looping enable
bool looping; // Music looping enable
unsigned int sampleCount; // Total number of samples
AudioStream stream; // Audio stream
int ctxType; // Type of music context (audio filetype)
void *ctxData; // Audio context data, depends on type
} Music;
#ifdef __cplusplus
extern "C" { // Prevents name mangling of functions
#endif
//----------------------------------------------------------------------------------
// Global Variables Definition
//----------------------------------------------------------------------------------
@ -130,6 +123,10 @@ extern "C" { // Prevents name mangling of functions
// Module Functions Declaration
//----------------------------------------------------------------------------------
#ifdef __cplusplus
extern "C" { // Prevents name mangling of functions
#endif
// Audio device management functions
void InitAudioDevice(void); // Initialize audio device and context
void CloseAudioDevice(void); // Close the audio device and context

+ 6
- 6
src/raylib.h View File

@ -423,10 +423,10 @@ typedef struct BoundingBox {
// Wave, audio wave data
typedef struct Wave {
unsigned int sampleCount; // Total number of samples (considering channels!)
unsigned int frameCount; // Total number of frames (considering channels)
unsigned int sampleRate; // Frequency (samples per second)
unsigned int sampleSize; // Bit depth (bits per sample): 8, 16, 32 (24 not supported)
unsigned int channels; // Number of channels (1-mono, 2-stereo)
unsigned int channels; // Number of channels (1-mono, 2-stereo, ...)
void *data; // Buffer data pointer
} Wave;
@ -438,19 +438,19 @@ typedef struct AudioStream {
unsigned int sampleRate; // Frequency (samples per second)
unsigned int sampleSize; // Bit depth (bits per sample): 8, 16, 32 (24 not supported)
unsigned int channels; // Number of channels (1-mono, 2-stereo)
unsigned int channels; // Number of channels (1-mono, 2-stereo, ...)
} AudioStream;
// Sound
typedef struct Sound {
AudioStream stream; // Audio stream
unsigned int sampleCount; // Total number of samples
unsigned int frameCount; // Total number of frames (considering channels)
} Sound;
// Music, audio stream, anything longer than ~10 seconds should be streamed
typedef struct Music {
AudioStream stream; // Audio stream
unsigned int sampleCount; // Total number of samples
unsigned int frameCount; // Total number of frames (considering channels)
bool looping; // Music looping enable
int ctxType; // Type of music context (audio filetype)
@ -1513,7 +1513,7 @@ RLAPI float GetMusicTimePlayed(Music music); // Get cur
// AudioStream management functions
RLAPI AudioStream LoadAudioStream(unsigned int sampleRate, unsigned int sampleSize, unsigned int channels); // Load audio stream (to stream raw audio pcm data)
RLAPI void UnloadAudioStream(AudioStream stream); // Unload audio stream and free memory
RLAPI void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount); // Update audio stream buffers with data
RLAPI void UpdateAudioStream(AudioStream stream, const void *data, int framesCount); // Update audio stream buffers with data
RLAPI bool IsAudioStreamProcessed(AudioStream stream); // Check if any audio stream buffers requires refill
RLAPI void PlayAudioStream(AudioStream stream); // Play audio stream
RLAPI void PauseAudioStream(AudioStream stream); // Pause audio stream

Write Preview
Loading…
Cancel
Save