diff options
Diffstat (limited to 'src/audio.c')
| -rw-r--r-- | src/audio.c | 98 |
1 files changed, 49 insertions, 49 deletions
diff --git a/src/audio.c b/src/audio.c index 74a54b04..22da74be 100644 --- a/src/audio.c +++ b/src/audio.c @@ -231,9 +231,9 @@ Wave LoadWave(const char *fileName) else if (strcmp(GetExtension(fileName),"rres") == 0) { RRESData rres = LoadResource(fileName); - + // NOTE: Parameters for RRES_WAVE type are: sampleCount, sampleRate, sampleSize, channels - + if (rres.type == RRES_WAVE) wave = LoadWaveEx(rres.data, rres.param1, rres.param2, rres.param3, rres.param4); else TraceLog(WARNING, "[%s] Resource file does not contain wave data", fileName); @@ -248,18 +248,18 @@ Wave LoadWave(const char *fileName) Wave LoadWaveEx(void *data, int sampleCount, int sampleRate, int sampleSize, int channels) { Wave wave; - + wave.data = data; wave.sampleCount = sampleCount; wave.sampleRate = sampleRate; wave.sampleSize = sampleSize; wave.channels = channels; - + // NOTE: Copy wave data to work with, user is responsible of input data to free Wave cwave = WaveCopy(wave); - + WaveFormat(&cwave, sampleRate, sampleSize, channels); - + return cwave; } @@ -268,9 +268,9 @@ Wave LoadWaveEx(void *data, int sampleCount, int sampleRate, int sampleSize, int Sound LoadSound(const char *fileName) { Wave wave = LoadWave(fileName); - + Sound sound = LoadSoundFromWave(wave); - + UnloadWave(wave); // Sound is loaded, we can unload wave return sound; @@ -354,7 +354,7 @@ void UnloadWave(Wave wave) void UnloadSound(Sound sound) { alSourceStop(sound.source); - + alDeleteSources(1, &sound.source); alDeleteBuffers(1, &sound.buffer); @@ -369,13 +369,13 @@ void UpdateSound(Sound sound, const void *data, int numSamples) alGetBufferi(sound.buffer, AL_FREQUENCY, &sampleRate); alGetBufferi(sound.buffer, AL_BITS, &sampleSize); // It could also be retrieved from sound.format alGetBufferi(sound.buffer, AL_CHANNELS, &channels); // It could also be retrieved from sound.format - + TraceLog(DEBUG, "UpdateSound() : AL_FREQUENCY: %i", sampleRate); TraceLog(DEBUG, "UpdateSound() : AL_BITS: %i", sampleSize); TraceLog(DEBUG, "UpdateSound() : AL_CHANNELS: %i", channels); unsigned int dataSize = numSamples*channels*sampleSize/8; // Size of data in bytes - + alSourceStop(sound.source); // Stop sound alSourcei(sound.source, AL_BUFFER, 0); // Unbind buffer from sound to update //alDeleteBuffers(1, &sound.buffer); // Delete current buffer data @@ -463,18 +463,18 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels) if (wave->sampleRate != sampleRate) { // TODO: Resample wave data (upsampling or downsampling) - // NOTE 1: To downsample, you have to drop samples or average them. + // NOTE 1: To downsample, you have to drop samples or average them. // NOTE 2: To upsample, you have to interpolate new samples. - + wave->sampleRate = sampleRate; } - + // Format sample size // NOTE: Only supported 8 bit <--> 16 bit <--> 32 bit if (wave->sampleSize != sampleSize) { void *data = malloc(wave->sampleCount*wave->channels*sampleSize/8); - + for (int i = 0; i < wave->sampleCount; i++) { for (int j = 0; j < wave->channels; j++) @@ -484,30 +484,30 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels) if (wave->sampleSize == 16) ((unsigned char *)data)[wave->channels*i + j] = (unsigned char)(((float)(((short *)wave->data)[wave->channels*i + j])/32767.0f)*256); else if (wave->sampleSize == 32) ((unsigned char *)data)[wave->channels*i + j] = (unsigned char)(((float *)wave->data)[wave->channels*i + j]*127.0f + 127); } - else if (sampleSize == 16) + else if (sampleSize == 16) { if (wave->sampleSize == 8) ((short *)data)[wave->channels*i + j] = (short)(((float)(((unsigned char *)wave->data)[wave->channels*i + j] - 127)/256.0f)*32767); else if (wave->sampleSize == 32) ((short *)data)[wave->channels*i + j] = (short)((((float *)wave->data)[wave->channels*i + j])*32767); } - else if (sampleSize == 32) + else if (sampleSize == 32) { if (wave->sampleSize == 8) ((float *)data)[wave->channels*i + j] = (float)(((unsigned char *)wave->data)[wave->channels*i + j] - 127)/256.0f; else if (wave->sampleSize == 16) ((float *)data)[wave->channels*i + j] = (float)(((short *)wave->data)[wave->channels*i + j])/32767.0f; } } } - + wave->sampleSize = sampleSize; free(wave->data); wave->data = data; } - + // Format channels (interlaced mode) // NOTE: Only supported mono <--> stereo if (wave->channels != channels) { void *data = malloc(wave->sampleCount*channels*wave->sampleSize/8); - + if ((wave->channels == 1) && (channels == 2)) // mono ---> stereo (duplicate mono information) { for (int i = 0; i < wave->sampleCount; i++) @@ -529,7 +529,7 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels) else if (wave->sampleSize == 32) ((float *)data)[i] = (((float *)wave->data)[j] + ((float *)wave->data)[j + 1])/2.0f; } } - + // TODO: Add/remove additional interlaced channels wave->channels = channels; @@ -563,15 +563,15 @@ Wave WaveCopy(Wave wave) // NOTE: Security check in case of out-of-range void WaveCrop(Wave *wave, int initSample, int finalSample) { - if ((initSample >= 0) && (initSample < finalSample) && + if ((initSample >= 0) && (initSample < finalSample) && (finalSample > 0) && (finalSample < wave->sampleCount)) { int sampleCount = finalSample - initSample; - + void *data = malloc(sampleCount*wave->channels*wave->sampleSize/8); - + memcpy(data, wave->data + (initSample*wave->channels*wave->sampleSize/8), sampleCount*wave->channels*wave->sampleSize/8); - + free(wave->data); wave->data = data; } @@ -583,7 +583,7 @@ void WaveCrop(Wave *wave, int initSample, int finalSample) float *GetWaveData(Wave wave) { float *samples = (float *)malloc(wave.sampleCount*wave.channels*sizeof(float)); - + for (int i = 0; i < wave.sampleCount; i++) { for (int j = 0; j < wave.channels; j++) @@ -593,7 +593,7 @@ float *GetWaveData(Wave wave) else if (wave.sampleSize == 32) samples[wave.channels*i + j] = ((float *)wave.data)[wave.channels*i + j]; } } - + return samples; } @@ -632,7 +632,7 @@ Music LoadMusicStream(const char *fileName) else if (strcmp(GetExtension(fileName), "flac") == 0) { music->ctxFlac = drflac_open_file(fileName); - + if (music->ctxFlac == NULL) TraceLog(WARNING, "[%s] FLAC audio file could not be opened", fileName); else { @@ -641,7 +641,7 @@ Music LoadMusicStream(const char *fileName) music->samplesLeft = music->totalSamples; music->ctxType = MUSIC_AUDIO_FLAC; music->loop = true; // We loop by default - + TraceLog(DEBUG, "[%s] FLAC total samples: %i", fileName, music->totalSamples); TraceLog(DEBUG, "[%s] FLAC sample rate: %i", fileName, music->ctxFlac->sampleRate); TraceLog(DEBUG, "[%s] FLAC bits per sample: %i", fileName, music->ctxFlac->bitsPerSample); @@ -728,7 +728,7 @@ void ResumeMusicStream(Music music) void StopMusicStream(Music music) { alSourceStop(music->stream.source); - + switch (music->ctxType) { case MUSIC_AUDIO_OGG: stb_vorbis_seek_start(music->ctxOgg); break; @@ -736,7 +736,7 @@ void StopMusicStream(Music music) case MUSIC_MODULE_MOD: jar_mod_seek_start(&music->ctxMod); break; default: break; } - + music->samplesLeft = music->totalSamples; } @@ -745,14 +745,14 @@ void UpdateMusicStream(Music music) { ALenum state; ALint processed = 0; - + alGetSourcei(music->stream.source, AL_SOURCE_STATE, &state); // Get music stream state alGetSourcei(music->stream.source, AL_BUFFERS_PROCESSED, &processed); // Get processed buffers if (processed > 0) { bool active = true; - + // NOTE: Using dynamic allocation because it could require more than 16KB void *pcm = calloc(AUDIO_BUFFER_SIZE*music->stream.channels*music->stream.sampleSize/8, 1); @@ -764,7 +764,7 @@ void UpdateMusicStream(Music music) { if (music->samplesLeft >= AUDIO_BUFFER_SIZE) numSamples = AUDIO_BUFFER_SIZE; else numSamples = music->samplesLeft; - + // TODO: Really don't like ctxType thingy... switch (music->ctxType) { @@ -784,7 +784,7 @@ void UpdateMusicStream(Music music) case MUSIC_MODULE_MOD: jar_mod_fillbuffer(&music->ctxMod, pcm, numSamples, 0); break; default: break; } - + UpdateAudioStream(music->stream, pcm, numSamples); music->samplesLeft -= numSamples; @@ -794,12 +794,12 @@ void UpdateMusicStream(Music music) break; } } - + // This error is registered when UpdateAudioStream() fails if (alGetError() == AL_INVALID_VALUE) TraceLog(WARNING, "OpenAL: Error buffering data..."); // Reset audio stream for looping - if (!active) + if (!active) { StopMusicStream(music); // Stop music (and reset) if (music->loop) PlayMusicStream(music); // Play again @@ -810,7 +810,7 @@ void UpdateMusicStream(Music music) // just make sure to play again on window restore if (state != AL_PLAYING) PlayMusicStream(music); } - + free(pcm); } } @@ -866,7 +866,7 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un stream.sampleRate = sampleRate; stream.sampleSize = sampleSize; - + // Only mono and stereo channels are supported, more channels require AL_EXT_MCFORMATS extension if ((channels > 0) && (channels < 3)) stream.channels = channels; else @@ -910,12 +910,12 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un // Initialize buffer with zeros by default // NOTE: Using dynamic allocation because it requires more than 16KB void *pcm = calloc(AUDIO_BUFFER_SIZE*stream.sampleSize/8*stream.channels, 1); - + for (int i = 0; i < MAX_STREAM_BUFFERS; i++) { alBufferData(stream.buffers[i], stream.format, pcm, AUDIO_BUFFER_SIZE*stream.sampleSize/8*stream.channels, stream.sampleRate); } - + free(pcm); alSourceQueueBuffers(stream.source, MAX_STREAM_BUFFERS, stream.buffers); @@ -1095,7 +1095,7 @@ static Wave LoadWAV(const char *fileName) wave.sampleRate = wavFormat.sampleRate; wave.sampleSize = wavFormat.bitsPerSample; wave.channels = wavFormat.numChannels; - + // NOTE: Only support 8 bit, 16 bit and 32 bit sample sizes if ((wave.sampleSize != 8) && (wave.sampleSize != 16) && (wave.sampleSize != 32)) { @@ -1104,16 +1104,16 @@ static Wave LoadWAV(const char *fileName) } // NOTE: Only support up to 2 channels (mono, stereo) - if (wave.channels > 2) + if (wave.channels > 2) { WaveFormat(&wave, wave.sampleRate, wave.sampleSize, 2); TraceLog(WARNING, "[%s] WAV channels number (%i) not supported, converted to 2 channels", fileName, wave.channels); } - + // NOTE: subChunkSize comes in bytes, we need to translate it to number of samples wave.sampleCount = (wavData.subChunkSize/(wave.sampleSize/8))/wave.channels; - TraceLog(INFO, "[%s] WAV file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo"); + TraceLog(INFO, "[%s] WAV file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo"); } } } @@ -1145,7 +1145,7 @@ static Wave LoadOGG(const char *fileName) wave.sampleSize = 16; // 16 bit per sample (short) wave.channels = info.channels; wave.sampleCount = (int)stb_vorbis_stream_length_in_samples(oggFile); - + float totalSeconds = stb_vorbis_stream_length_in_seconds(oggFile); 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); @@ -1173,16 +1173,16 @@ static Wave LoadFLAC(const char *fileName) // Decode an entire FLAC file in one go uint64_t totalSampleCount; wave.data = drflac_open_and_decode_file_s16(fileName, &wave.channels, &wave.sampleRate, &totalSampleCount); - + wave.sampleCount = (int)totalSampleCount/wave.channels; wave.sampleSize = 16; - + // NOTE: Only support up to 2 channels (mono, stereo) if (wave.channels > 2) TraceLog(WARNING, "[%s] FLAC channels number (%i) not supported", fileName, wave.channels); if (wave.data == NULL) TraceLog(WARNING, "[%s] FLAC data could not be loaded", fileName); else TraceLog(INFO, "[%s] FLAC file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo"); - + return wave; } |