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WARNING: Multiple funcs reviewed!

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There were some problems about frameCount vs sampleCount that could cause some breaks.

raylib audio structs stores sampleCount = frameCount*channels.

Most libraries return framesCount instead of sampleCount.

stb_vorbis seems to refer to framesCount as samples.

All required functions have been reviewed.
This commit is contained in:
Ray 2020-11-19 20:11:11 +01:00
parent 72e51cd6de
commit 7f6cd93d62
2 changed files with 28 additions and 27 deletions
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53
src/raudio.c
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@ -1,6 +1,6 @@
/**********************************************************************************************
*
* raudio - A simple and easy-to-use audio library based on miniaudio
* raudio v1.0 - A simple and easy-to-use audio library based on miniaudio
*
* FEATURES:
* - Manage audio device (init/close)
@ -824,7 +824,7 @@ void ExportWave(Wave wave, const char *fileName)
{
// Export raw sample data (without header)
// NOTE: It's up to the user to track wave parameters
SaveFileData(fileName, wave.data, wave.sampleCount*wave.channels*wave.sampleSize/8);
SaveFileData(fileName, wave.data, wave.sampleCount*wave.sampleSize/8);
success = true;
}
@ -1007,10 +1007,10 @@ void SetSoundPitch(Sound sound, float pitch)
// Convert wave data to desired format
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_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; // Is wave->sampleCount actually the frame count? That terminology needs to change, if so.
ma_uint32 frameCountIn = wave->sampleCount/wave->channels;
ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, formatOut, channels, sampleRate, NULL, frameCountIn, formatIn, wave->channels, wave->sampleRate);
if (frameCount == 0)
@ -1028,7 +1028,7 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
return;
}
wave->sampleCount = frameCount;
wave->sampleCount = frameCount*channels;
wave->sampleSize = sampleSize;
wave->sampleRate = sampleRate;
wave->channels = channels;
@ -1041,12 +1041,12 @@ Wave WaveCopy(Wave wave)
{
Wave newWave = { 0 };
newWave.data = RL_MALLOC(wave.sampleCount*wave.sampleSize/8*wave.channels);
newWave.data = RL_MALLOC(wave.sampleCount*wave.sampleSize/8);
if (newWave.data != NULL)
{
// NOTE: Size must be provided in bytes
memcpy(newWave.data, wave.data, wave.sampleCount*wave.channels*wave.sampleSize/8);
memcpy(newWave.data, wave.data, wave.sampleCount*wave.sampleSize/8);
newWave.sampleCount = wave.sampleCount;
newWave.sampleRate = wave.sampleRate;
@ -1066,9 +1066,9 @@ void WaveCrop(Wave *wave, int initSample, int finalSample)
{
int sampleCount = finalSample - initSample;
void *data = RL_MALLOC(sampleCount*wave->sampleSize/8*wave->channels);
void *data = RL_MALLOC(sampleCount*wave->sampleSize/8);
memcpy(data, (unsigned char *)wave->data + (initSample*wave->channels*wave->sampleSize/8), sampleCount*wave->channels*wave->sampleSize/8);
memcpy(data, (unsigned char *)wave->data + (initSample*wave->channels*wave->sampleSize/8), sampleCount*wave->sampleSize/8);
RL_FREE(wave->data);
wave->data = data;
@ -1080,16 +1080,15 @@ void WaveCrop(Wave *wave, int initSample, int finalSample)
// NOTE: Returned sample values are normalized to range [-1..1]
float *GetWaveData(Wave wave)
{
float *samples = (float *)RL_MALLOC(wave.sampleCount*wave.channels*sizeof(float));
float *samples = (float *)RL_MALLOC(wave.sampleCount*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 j = 0; j < wave.channels; j++)
{
if (wave.sampleSize == 8) samples[wave.channels*i + j] = (float)(((unsigned char *)wave.data)[wave.channels*i + j] - 127)/256.0f;
else if (wave.sampleSize == 16) samples[wave.channels*i + j] = (float)((short *)wave.data)[wave.channels*i + j]/32767.0f;
else if (wave.sampleSize == 32) samples[wave.channels*i + j] = ((float *)wave.data)[wave.channels*i + j];
}
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;
else if (wave.sampleSize == 32) samples[i] = ((float *)wave.data)[i];
}
return samples;
@ -1140,6 +1139,8 @@ Music LoadMusicStream(const char *fileName)
// OGG bit rate defaults to 16 bit, it's enough for compressed format
music.stream = InitAudioStream(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.looping = true; // Looping enabled by default
musicLoaded = true;
@ -1157,7 +1158,7 @@ Music LoadMusicStream(const char *fileName)
drflac *ctxFlac = (drflac *)music.ctxData;
music.stream = InitAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
music.sampleCount = (unsigned int)ctxFlac->totalPCMFrameCount;
music.sampleCount = (unsigned int)ctxFlac->totalPCMFrameCount*ctxFlac->channels;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
@ -1196,7 +1197,7 @@ Music LoadMusicStream(const char *fileName)
// NOTE: Only stereo is supported for XM
music.stream = InitAudioStream(48000, 16, 2);
music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm)*2;
music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm)*2; // 2 channels
music.looping = true; // Looping enabled by default
jar_xm_reset(ctxXm); // make sure we start at the beginning of the song
musicLoaded = true;
@ -1219,7 +1220,7 @@ Music LoadMusicStream(const char *fileName)
// NOTE: Only stereo is supported for MOD
music.stream = InitAudioStream(48000, 16, 2);
music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod)*2;
music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod)*2; // 2 channels
music.looping = true; // Looping enabled by default
musicLoaded = true;
@ -1493,7 +1494,7 @@ float GetMusicTimePlayed(Music music)
{
//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;
secondsPlayed = (float)samplesPlayed / (music.stream.sampleRate*music.stream.channels);
secondsPlayed = (float)samplesPlayed/(music.stream.sampleRate*music.stream.channels);
}
return secondsPlayed;
@ -1983,10 +1984,10 @@ static Wave LoadOGG(const unsigned char *fileData, unsigned int fileSize)
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*wave.channels*sizeof(short));
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*wave.channels);
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);
@ -2005,12 +2006,12 @@ 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 totalSampleCount = 0;
wave.data = drflac_open_memory_and_read_pcm_frames_s16(fileData, fileSize, &wave.channels, &wave.sampleRate, &totalSampleCount, NULL);
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)totalSampleCount;
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");

2
src/raudio.h
View File

@ -1,6 +1,6 @@
/**********************************************************************************************
*
* raudio - A simple and easy-to-use audio library based on miniaudio
* raudio v1.0 - A simple and easy-to-use audio library based on miniaudio
*
* FEATURES:
* - Manage audio device (init/close)