mirror of
https://github.com/raysan5/raylib.git
synced 2025-12-25 10:22:33 -05:00
Remove unecessary spaces...
This commit is contained in:
Ray
98
src/audio.c
98
src/audio.c
@ -231,9 +231,9 @@ Wave LoadWave(const char *fileName)
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else if (strcmp(GetExtension(fileName),"rres") == 0)
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{
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RRESData rres = LoadResource(fileName);
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// NOTE: Parameters for RRES_WAVE type are: sampleCount, sampleRate, sampleSize, channels
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if (rres.type == RRES_WAVE) wave = LoadWaveEx(rres.data, rres.param1, rres.param2, rres.param3, rres.param4);
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else TraceLog(WARNING, "[%s] Resource file does not contain wave data", fileName);
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@ -248,18 +248,18 @@ Wave LoadWave(const char *fileName)
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Wave LoadWaveEx(void *data, int sampleCount, int sampleRate, int sampleSize, int channels)
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{
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Wave wave;
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wave.data = data;
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wave.sampleCount = sampleCount;
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wave.sampleRate = sampleRate;
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wave.sampleSize = sampleSize;
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wave.channels = channels;
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// NOTE: Copy wave data to work with, user is responsible of input data to free
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Wave cwave = WaveCopy(wave);
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WaveFormat(&cwave, sampleRate, sampleSize, channels);
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return cwave;
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}
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@ -268,9 +268,9 @@ Wave LoadWaveEx(void *data, int sampleCount, int sampleRate, int sampleSize, int
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Sound LoadSound(const char *fileName)
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{
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Wave wave = LoadWave(fileName);
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Sound sound = LoadSoundFromWave(wave);
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UnloadWave(wave); // Sound is loaded, we can unload wave
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return sound;
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@ -354,7 +354,7 @@ void UnloadWave(Wave wave)
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void UnloadSound(Sound sound)
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{
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alSourceStop(sound.source);
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alDeleteSources(1, &sound.source);
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alDeleteBuffers(1, &sound.buffer);
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@ -369,13 +369,13 @@ void UpdateSound(Sound sound, const void *data, int numSamples)
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alGetBufferi(sound.buffer, AL_FREQUENCY, &sampleRate);
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alGetBufferi(sound.buffer, AL_BITS, &sampleSize); // It could also be retrieved from sound.format
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alGetBufferi(sound.buffer, AL_CHANNELS, &channels); // It could also be retrieved from sound.format
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TraceLog(DEBUG, "UpdateSound() : AL_FREQUENCY: %i", sampleRate);
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TraceLog(DEBUG, "UpdateSound() : AL_BITS: %i", sampleSize);
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TraceLog(DEBUG, "UpdateSound() : AL_CHANNELS: %i", channels);
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unsigned int dataSize = numSamples*channels*sampleSize/8; // Size of data in bytes
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alSourceStop(sound.source); // Stop sound
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alSourcei(sound.source, AL_BUFFER, 0); // Unbind buffer from sound to update
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//alDeleteBuffers(1, &sound.buffer); // Delete current buffer data
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@ -463,18 +463,18 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
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if (wave->sampleRate != sampleRate)
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{
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// TODO: Resample wave data (upsampling or downsampling)
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// NOTE 1: To downsample, you have to drop samples or average them.
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// NOTE 1: To downsample, you have to drop samples or average them.
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// NOTE 2: To upsample, you have to interpolate new samples.
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wave->sampleRate = sampleRate;
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}
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// Format sample size
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// NOTE: Only supported 8 bit <--> 16 bit <--> 32 bit
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if (wave->sampleSize != sampleSize)
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{
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void *data = malloc(wave->sampleCount*wave->channels*sampleSize/8);
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for (int i = 0; i < wave->sampleCount; i++)
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{
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for (int j = 0; j < wave->channels; j++)
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@ -484,30 +484,30 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
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if (wave->sampleSize == 16) ((unsigned char *)data)[wave->channels*i + j] = (unsigned char)(((float)(((short *)wave->data)[wave->channels*i + j])/32767.0f)*256);
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else if (wave->sampleSize == 32) ((unsigned char *)data)[wave->channels*i + j] = (unsigned char)(((float *)wave->data)[wave->channels*i + j]*127.0f + 127);
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}
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else if (sampleSize == 16)
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else if (sampleSize == 16)
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{
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if (wave->sampleSize == 8) ((short *)data)[wave->channels*i + j] = (short)(((float)(((unsigned char *)wave->data)[wave->channels*i + j] - 127)/256.0f)*32767);
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else if (wave->sampleSize == 32) ((short *)data)[wave->channels*i + j] = (short)((((float *)wave->data)[wave->channels*i + j])*32767);
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}
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else if (sampleSize == 32)
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else if (sampleSize == 32)
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{
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if (wave->sampleSize == 8) ((float *)data)[wave->channels*i + j] = (float)(((unsigned char *)wave->data)[wave->channels*i + j] - 127)/256.0f;
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else if (wave->sampleSize == 16) ((float *)data)[wave->channels*i + j] = (float)(((short *)wave->data)[wave->channels*i + j])/32767.0f;
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}
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}
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}
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wave->sampleSize = sampleSize;
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free(wave->data);
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wave->data = data;
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}
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// Format channels (interlaced mode)
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// NOTE: Only supported mono <--> stereo
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if (wave->channels != channels)
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{
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void *data = malloc(wave->sampleCount*channels*wave->sampleSize/8);
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if ((wave->channels == 1) && (channels == 2)) // mono ---> stereo (duplicate mono information)
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{
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for (int i = 0; i < wave->sampleCount; i++)
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@ -529,7 +529,7 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
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else if (wave->sampleSize == 32) ((float *)data)[i] = (((float *)wave->data)[j] + ((float *)wave->data)[j + 1])/2.0f;
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}
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}
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// TODO: Add/remove additional interlaced channels
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wave->channels = channels;
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@ -563,15 +563,15 @@ Wave WaveCopy(Wave wave)
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// NOTE: Security check in case of out-of-range
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void WaveCrop(Wave *wave, int initSample, int finalSample)
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{
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if ((initSample >= 0) && (initSample < finalSample) &&
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if ((initSample >= 0) && (initSample < finalSample) &&
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(finalSample > 0) && (finalSample < wave->sampleCount))
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{
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int sampleCount = finalSample - initSample;
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void *data = malloc(sampleCount*wave->channels*wave->sampleSize/8);
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memcpy(data, wave->data + (initSample*wave->channels*wave->sampleSize/8), sampleCount*wave->channels*wave->sampleSize/8);
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free(wave->data);
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wave->data = data;
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}
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@ -583,7 +583,7 @@ void WaveCrop(Wave *wave, int initSample, int finalSample)
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float *GetWaveData(Wave wave)
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{
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float *samples = (float *)malloc(wave.sampleCount*wave.channels*sizeof(float));
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for (int i = 0; i < wave.sampleCount; i++)
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{
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for (int j = 0; j < wave.channels; j++)
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@ -593,7 +593,7 @@ float *GetWaveData(Wave wave)
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else if (wave.sampleSize == 32) samples[wave.channels*i + j] = ((float *)wave.data)[wave.channels*i + j];
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}
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}
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return samples;
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}
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@ -632,7 +632,7 @@ Music LoadMusicStream(const char *fileName)
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else if (strcmp(GetExtension(fileName), "flac") == 0)
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{
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music->ctxFlac = drflac_open_file(fileName);
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if (music->ctxFlac == NULL) TraceLog(WARNING, "[%s] FLAC audio file could not be opened", fileName);
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else
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{
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@ -641,7 +641,7 @@ Music LoadMusicStream(const char *fileName)
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music->samplesLeft = music->totalSamples;
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music->ctxType = MUSIC_AUDIO_FLAC;
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music->loop = true; // We loop by default
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TraceLog(DEBUG, "[%s] FLAC total samples: %i", fileName, music->totalSamples);
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TraceLog(DEBUG, "[%s] FLAC sample rate: %i", fileName, music->ctxFlac->sampleRate);
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TraceLog(DEBUG, "[%s] FLAC bits per sample: %i", fileName, music->ctxFlac->bitsPerSample);
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@ -728,7 +728,7 @@ void ResumeMusicStream(Music music)
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void StopMusicStream(Music music)
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{
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alSourceStop(music->stream.source);
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switch (music->ctxType)
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{
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case MUSIC_AUDIO_OGG: stb_vorbis_seek_start(music->ctxOgg); break;
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@ -736,7 +736,7 @@ void StopMusicStream(Music music)
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case MUSIC_MODULE_MOD: jar_mod_seek_start(&music->ctxMod); break;
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default: break;
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}
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music->samplesLeft = music->totalSamples;
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}
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@ -745,14 +745,14 @@ void UpdateMusicStream(Music music)
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{
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ALenum state;
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ALint processed = 0;
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alGetSourcei(music->stream.source, AL_SOURCE_STATE, &state); // Get music stream state
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alGetSourcei(music->stream.source, AL_BUFFERS_PROCESSED, &processed); // Get processed buffers
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if (processed > 0)
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{
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bool active = true;
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// NOTE: Using dynamic allocation because it could require more than 16KB
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void *pcm = calloc(AUDIO_BUFFER_SIZE*music->stream.channels*music->stream.sampleSize/8, 1);
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@ -764,7 +764,7 @@ void UpdateMusicStream(Music music)
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{
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if (music->samplesLeft >= AUDIO_BUFFER_SIZE) numSamples = AUDIO_BUFFER_SIZE;
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else numSamples = music->samplesLeft;
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// TODO: Really don't like ctxType thingy...
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switch (music->ctxType)
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{
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@ -784,7 +784,7 @@ void UpdateMusicStream(Music music)
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case MUSIC_MODULE_MOD: jar_mod_fillbuffer(&music->ctxMod, pcm, numSamples, 0); break;
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default: break;
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}
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UpdateAudioStream(music->stream, pcm, numSamples);
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music->samplesLeft -= numSamples;
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@ -794,12 +794,12 @@ void UpdateMusicStream(Music music)
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break;
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}
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}
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// This error is registered when UpdateAudioStream() fails
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if (alGetError() == AL_INVALID_VALUE) TraceLog(WARNING, "OpenAL: Error buffering data...");
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// Reset audio stream for looping
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if (!active)
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if (!active)
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{
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StopMusicStream(music); // Stop music (and reset)
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if (music->loop) PlayMusicStream(music); // Play again
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@ -810,7 +810,7 @@ void UpdateMusicStream(Music music)
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// just make sure to play again on window restore
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if (state != AL_PLAYING) PlayMusicStream(music);
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}
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free(pcm);
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}
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}
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@ -866,7 +866,7 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un
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stream.sampleRate = sampleRate;
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stream.sampleSize = sampleSize;
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// Only mono and stereo channels are supported, more channels require AL_EXT_MCFORMATS extension
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if ((channels > 0) && (channels < 3)) stream.channels = channels;
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else
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@ -910,12 +910,12 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un
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// Initialize buffer with zeros by default
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// NOTE: Using dynamic allocation because it requires more than 16KB
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void *pcm = calloc(AUDIO_BUFFER_SIZE*stream.sampleSize/8*stream.channels, 1);
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for (int i = 0; i < MAX_STREAM_BUFFERS; i++)
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{
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alBufferData(stream.buffers[i], stream.format, pcm, AUDIO_BUFFER_SIZE*stream.sampleSize/8*stream.channels, stream.sampleRate);
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}
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free(pcm);
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alSourceQueueBuffers(stream.source, MAX_STREAM_BUFFERS, stream.buffers);
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@ -1095,7 +1095,7 @@ static Wave LoadWAV(const char *fileName)
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wave.sampleRate = wavFormat.sampleRate;
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wave.sampleSize = wavFormat.bitsPerSample;
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wave.channels = wavFormat.numChannels;
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// NOTE: Only support 8 bit, 16 bit and 32 bit sample sizes
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if ((wave.sampleSize != 8) && (wave.sampleSize != 16) && (wave.sampleSize != 32))
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{
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@ -1104,16 +1104,16 @@ static Wave LoadWAV(const char *fileName)
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}
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// NOTE: Only support up to 2 channels (mono, stereo)
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if (wave.channels > 2)
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if (wave.channels > 2)
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{
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WaveFormat(&wave, wave.sampleRate, wave.sampleSize, 2);
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TraceLog(WARNING, "[%s] WAV channels number (%i) not supported, converted to 2 channels", fileName, wave.channels);
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}
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// NOTE: subChunkSize comes in bytes, we need to translate it to number of samples
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wave.sampleCount = (wavData.subChunkSize/(wave.sampleSize/8))/wave.channels;
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TraceLog(INFO, "[%s] WAV file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo");
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TraceLog(INFO, "[%s] WAV file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo");
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}
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}
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}
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@ -1145,7 +1145,7 @@ static Wave LoadOGG(const char *fileName)
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wave.sampleSize = 16; // 16 bit per sample (short)
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wave.channels = info.channels;
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wave.sampleCount = (int)stb_vorbis_stream_length_in_samples(oggFile);
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float totalSeconds = stb_vorbis_stream_length_in_seconds(oggFile);
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if (totalSeconds > 10) TraceLog(WARNING, "[%s] Ogg audio lenght is larger than 10 seconds (%f), that's a big file in memory, consider music streaming", fileName, totalSeconds);
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@ -1173,16 +1173,16 @@ static Wave LoadFLAC(const char *fileName)
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// Decode an entire FLAC file in one go
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uint64_t totalSampleCount;
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wave.data = drflac_open_and_decode_file_s16(fileName, &wave.channels, &wave.sampleRate, &totalSampleCount);
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wave.sampleCount = (int)totalSampleCount/wave.channels;
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wave.sampleSize = 16;
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// NOTE: Only support up to 2 channels (mono, stereo)
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if (wave.channels > 2) TraceLog(WARNING, "[%s] FLAC channels number (%i) not supported", fileName, wave.channels);
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if (wave.data == NULL) TraceLog(WARNING, "[%s] FLAC data could not be loaded", fileName);
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else TraceLog(INFO, "[%s] FLAC file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo");
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return wave;
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}
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