diff --git a/examples/audio_module_playing.c b/examples/audio_module_playing.c index 4582a561..08ae2b05 100644 --- a/examples/audio_module_playing.c +++ b/examples/audio_module_playing.c @@ -86,7 +86,7 @@ int main() } // Get timePlayed scaled to bar dimensions - timePlayed = (GetMusicTimePlayed(xm)/GetMusicTimeLength(xm)*(screenWidth - 40))*2; + timePlayed = GetMusicTimePlayed(xm)/GetMusicTimeLength(xm)*(screenWidth - 40); // Color circles animation for (int i = MAX_CIRCLES - 1; (i >= 0) && !pause; i--) diff --git a/examples/audio_music_stream.c b/examples/audio_music_stream.c index dc9d4355..9c1ca4df 100644 --- a/examples/audio_music_stream.c +++ b/examples/audio_music_stream.c @@ -58,7 +58,7 @@ int main() } // Get timePlayed scaled to bar dimensions (400 pixels) - timePlayed = GetMusicTimePlayed(music)/GetMusicTimeLength(music)*100*4; + timePlayed = GetMusicTimePlayed(music)/GetMusicTimeLength(music)*400; //---------------------------------------------------------------------------------- // Draw diff --git a/examples/audio_raw_stream.c b/examples/audio_raw_stream.c index c044a7e0..d1fd1794 100644 --- a/examples/audio_raw_stream.c +++ b/examples/audio_raw_stream.c @@ -16,7 +16,7 @@ #include // Required for: malloc(), free() #include // Required for: sinf() -#define MAX_SAMPLES 20000 +#define MAX_SAMPLES 22050 int main() { @@ -29,15 +29,15 @@ int main() InitAudioDevice(); // Initialize audio device - // Init raw audio stream (sample rate: 22050, sample size: 32bit-float, channels: 1-mono) - AudioStream stream = InitAudioStream(22050, 32, 1); + // Init raw audio stream (sample rate: 22050, sample size: 16bit-short, channels: 1-mono) + AudioStream stream = InitAudioStream(22050, 16, 1); // Fill audio stream with some samples (sine wave) - float *data = (float *)malloc(sizeof(float)*MAX_SAMPLES); + short *data = (short *)malloc(sizeof(short)*MAX_SAMPLES); for (int i = 0; i < MAX_SAMPLES; i++) { - data[i] = sinf(((2*PI*(float)i)/2)*DEG2RAD); + data[i] = (short)(sinf(((2*PI*(float)i)/2)*DEG2RAD)*32000); } // NOTE: The generated MAX_SAMPLES do not fit to close a perfect loop @@ -87,7 +87,7 @@ int main() for (int i = 0; i < GetScreenWidth(); i++) { position.x = i; - position.y = 250 + 50*data[i]; + position.y = 250 + 50*data[i]/32000; DrawPixelV(position, RED); } diff --git a/src/audio.c b/src/audio.c index aa89de02..04ff90da 100644 --- a/src/audio.c +++ b/src/audio.c @@ -19,6 +19,10 @@ * Module Configuration Flags: * AUDIO_STANDALONE - Use this module as standalone library (independently of raylib) * +* Some design decisions: +* Support only up to two channels: MONO and STEREO (for additional channels, AL_EXT_MCFORMATS) +* Support only the following sample sizes: 8bit PCM and 16bit PCM (for additional size, AL_EXT_FLOAT32) +* * Many thanks to Joshua Reisenauer (github: @kd7tck) for the following additions: * XM audio module support (jar_xm) * MOD audio module support (jar_mod) @@ -57,19 +61,15 @@ #include "AL/al.h" // OpenAL basic header #include "AL/alc.h" // OpenAL context header (like OpenGL, OpenAL requires a context to work) +//#include "AL/alext.h" // OpenAL extensions header, required for AL_EXT_FLOAT32 and AL_EXT_MCFORMATS + +// OpenAL extension: AL_EXT_FLOAT32 - Support for 32bit float samples +// OpenAL extension: AL_EXT_MCFORMATS - Support for multi-channel formats (Quad, 5.1, 6.1, 7.1) #include // Required for: malloc(), free() #include // Required for: strcmp(), strncmp() #include // Required for: FILE, fopen(), fclose(), fread() -// Tokens defined by OpenAL extension: AL_EXT_float32 -#ifndef AL_FORMAT_MONO_FLOAT32 - #define AL_FORMAT_MONO_FLOAT32 0x10010 -#endif -#ifndef AL_FORMAT_STEREO_FLOAT32 - #define AL_FORMAT_STEREO_FLOAT32 0x10011 -#endif - //#define STB_VORBIS_HEADER_ONLY #include "external/stb_vorbis.h" // OGG loading functions @@ -92,11 +92,11 @@ //---------------------------------------------------------------------------------- #define MAX_STREAM_BUFFERS 2 // Number of buffers for each audio stream -// NOTE: Music buffer size is defined by number of samples, independent of sample size +// NOTE: Music buffer size is defined by number of samples, independent of sample size and channels number // After some math, considering a sampleRate of 48000, a buffer refill rate of 1/60 seconds // and double-buffering system, I concluded that a 4096 samples buffer should be enough // In case of music-stalls, just increase this number -#define AUDIO_BUFFER_SIZE 4096 // PCM data samples (i.e. short: 32Kb) +#define AUDIO_BUFFER_SIZE 4096 // PCM data samples (i.e. 16bit, Mono: 8Kb) //---------------------------------------------------------------------------------- // Types and Structures Definition @@ -211,7 +211,7 @@ bool IsAudioDeviceReady(void) // Module Functions Definition - Sounds loading and playing (.WAV) //---------------------------------------------------------------------------------- -// Load wave data from file into RAM +// Load wave data from file Wave LoadWave(const char *fileName) { Wave wave = { 0 }; @@ -224,19 +224,18 @@ Wave LoadWave(const char *fileName) return wave; } -// Load wave data from float array data (32bit) -Wave LoadWaveEx(float *data, int sampleCount, int sampleRate, int sampleSize, int channels) +// Load wave data from raw array data +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 = 32; + wave.sampleSize = sampleSize; wave.channels = channels; - // NOTE: Copy wave data to work with, - // user is responsible of input data to free + // NOTE: Copy wave data to work with, user is responsible of input data to free Wave cwave = WaveCopy(wave); WaveFormat(&cwave, sampleRate, sampleSize, channels); @@ -244,7 +243,7 @@ Wave LoadWaveEx(float *data, int sampleCount, int sampleRate, int sampleSize, in return cwave; } -// Load sound to memory +// Load sound from file // NOTE: The entire file is loaded to memory to be played (no-streaming) Sound LoadSound(const char *fileName) { @@ -274,7 +273,7 @@ Sound LoadSoundFromWave(Wave wave) { case 8: format = AL_FORMAT_MONO8; break; case 16: format = AL_FORMAT_MONO16; break; - case 32: format = AL_FORMAT_MONO_FLOAT32; break; + case 32: //format = AL_FORMAT_MONO_FLOAT32; break; // Requires OpenAL extension: AL_EXT_FLOAT32 default: TraceLog(WARNING, "Wave sample size not supported: %i", wave.sampleSize); break; } } @@ -284,7 +283,7 @@ Sound LoadSoundFromWave(Wave wave) { case 8: format = AL_FORMAT_STEREO8; break; case 16: format = AL_FORMAT_STEREO16; break; - case 32: format = AL_FORMAT_STEREO_FLOAT32; break; + case 32: //format = AL_FORMAT_STEREO_FLOAT32; break; // Requires OpenAL extension: AL_EXT_FLOAT32 default: TraceLog(WARNING, "Wave sample size not supported: %i", wave.sampleSize); break; } } @@ -305,7 +304,7 @@ Sound LoadSoundFromWave(Wave wave) ALuint buffer; alGenBuffers(1, &buffer); // Generate pointer to buffer - unsigned int dataSize = wave.sampleCount*wave.sampleSize/8; // Size in bytes + unsigned int dataSize = wave.sampleCount*wave.sampleSize/8*wave.channels; // Size in bytes // Upload sound data to buffer alBufferData(buffer, format, wave.data, dataSize, wave.sampleRate); @@ -313,7 +312,7 @@ Sound LoadSoundFromWave(Wave wave) // Attach sound buffer to source alSourcei(source, AL_BUFFER, buffer); - TraceLog(INFO, "[SND ID %i][BUFR ID %i] Sound data loaded successfully (SampleRate: %i, SampleSize: %i, Channels: %i)", source, buffer, wave.sampleRate, wave.sampleSize, wave.channels); + TraceLog(INFO, "[SND ID %i][BUFR ID %i] Sound data loaded successfully (%i Hz, %i bit, %s)", source, buffer, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo"); sound.source = source; sound.buffer = buffer; @@ -323,7 +322,7 @@ Sound LoadSoundFromWave(Wave wave) return sound; } -// Unload Wave data +// Unload wave data void UnloadWave(Wave wave) { free(wave.data); @@ -346,14 +345,14 @@ void UpdateSound(Sound sound, const void *data, int numSamples) { ALint sampleRate, sampleSize, channels; 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 + 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*sampleSize/8; // Size of data in bytes + unsigned int dataSize = numSamples*sampleSize/8*channels; // Size of data in bytes alSourceStop(sound.source); // Stop sound alSourcei(sound.source, AL_BUFFER, 0); // Unbind buffer from sound to update @@ -435,69 +434,86 @@ void SetSoundPitch(Sound sound, float pitch) } // Convert wave data to desired format -// TODO: Consider channels (mono - stereo) void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels) { + // Format sample rate + if (wave->sampleRate != sampleRate) wave->sampleRate = sampleRate; + + // Format sample size + // NOTE: Only supported 8 bit <--> 16 bit <--> 32 bit if (wave->sampleSize != sampleSize) { - float *samples = GetWaveData(*wave); //Color *pixels = GetImageData(*image); - - free(wave->data); + void *data = malloc(wave->sampleCount*wave->channels*sampleSize/8); - wave->sampleSize = sampleSize; - - //sample *= 4.0f; // Arbitrary gain to get reasonable output volume... - //if (sample > 1.0f) sample = 1.0f; - //if (sample < -1.0f) sample = -1.0f; - - if (sampleSize == 8) + for (int i = 0; i < wave->sampleCount; i++) { - wave->data = (unsigned char *)malloc(wave->sampleCount*sizeof(unsigned char)); - - for (int i = 0; i < wave->sampleCount; i++) + for (int j = 0; j < wave->channels; j++) { - ((unsigned char *)wave->data)[i] = (unsigned char)((float)samples[i]*127 + 128); + if (sampleSize == 8) + { + 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) + { + 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) + { + 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; + } } } - else if (sampleSize == 16) + + 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) { - wave->data = (short *)malloc(wave->sampleCount*sizeof(short)); - for (int i = 0; i < wave->sampleCount; i++) { - ((short *)wave->data)[i] = (short)((float)samples[i]*32000); // SHRT_MAX = 32767 + for (int j = 0; j < channels; j++) + { + if (wave->sampleSize == 8) ((unsigned char *)data)[channels*i + j] = ((unsigned char *)wave->data)[i]; + else if (wave->sampleSize == 16) ((short *)data)[channels*i + j] = ((short *)wave->data)[i]; + else if (wave->sampleSize == 32) ((float *)data)[channels*i + j] = ((float *)wave->data)[i]; + } } } - else if (sampleSize == 32) + else if ((wave->channels == 2) && (channels == 1)) // stereo ---> mono (mix stereo channels) { - wave->data = (float *)malloc(wave->sampleCount*sizeof(float)); - - for (int i = 0; i < wave->sampleCount; i++) + for (int i = 0, j = 0; i < wave->sampleCount; i++, j += 2) { - ((float *)wave->data)[i] = (float)samples[i]; + if (wave->sampleSize == 8) ((unsigned char *)data)[i] = (((unsigned char *)wave->data)[j] + ((unsigned char *)wave->data)[j + 1])/2; + else if (wave->sampleSize == 16) ((short *)data)[i] = (((short *)wave->data)[j] + ((short *)wave->data)[j + 1])/2; + else if (wave->sampleSize == 32) ((float *)data)[i] = (((float *)wave->data)[j] + ((float *)wave->data)[j + 1])/2.0f; } } - else TraceLog(WARNING, "Wave formatting: Sample size not supported"); - free(samples); - } - - // NOTE: Only supported 1 or 2 channels (mono or stereo) - if ((channels > 0) && (channels < 3) && (wave->channels != channels)) - { - // TODO: Add/remove channels interlaced data if required... + // TODO: Add/remove additional interlaced channels + + wave->channels = channels; + free(wave->data); + wave->data = data; } } // Copy a wave to a new wave Wave WaveCopy(Wave wave) { - Wave newWave = { 0 }; + Wave newWave = { 0 }; - if (wave.sampleSize == 8) newWave.data = (unsigned char *)malloc(wave.sampleCount*wave.channels*sizeof(unsigned char)); - else if (wave.sampleSize == 16) newWave.data = (short *)malloc(wave.sampleCount*wave.channels*sizeof(short)); - else if (wave.sampleSize == 32) newWave.data = (float *)malloc(wave.sampleCount*wave.channels*sizeof(float)); - else TraceLog(WARNING, "Wave sample size not supported for copy"); + newWave.data = malloc(wave.sampleCount*wave.channels*wave.sampleSize/8); if (newWave.data != NULL) { @@ -520,35 +536,32 @@ void WaveCrop(Wave *wave, int initSample, int finalSample) if ((initSample >= 0) && (initSample < finalSample) && (finalSample > 0) && (finalSample < wave->sampleCount)) { - // TODO: Review cropping (it could be simplified...) + int sampleCount = finalSample - initSample; - float *samples = GetWaveData(*wave); - float *cropSamples = (float *)malloc((finalSample - initSample)*sizeof(float)); + 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); - for (int i = initSample; i < finalSample; i++) cropSamples[i] = samples[i]; - free(wave->data); - wave->data = cropSamples; - int sampleSize = wave->sampleSize; - wave->sampleSize = 32; - - WaveFormat(wave, wave->sampleRate, sampleSize, wave->channels); + wave->data = data; } else TraceLog(WARNING, "Wave crop range out of bounds"); } // Get samples data from wave as a floats array // NOTE: Returned sample values are normalized to range [-1..1] -// TODO: Consider multiple channels (mono - stereo) float *GetWaveData(Wave wave) { - float *samples = (float *)malloc(wave.sampleCount*sizeof(float)); + float *samples = (float *)malloc(wave.sampleCount*wave.channels*sizeof(float)); for (int i = 0; i < wave.sampleCount; 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; - else if (wave.sampleSize == 32) samples[i] = ((float *)wave.data)[i]; + for (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]; + } } return samples; @@ -572,11 +585,10 @@ Music LoadMusicStream(const char *fileName) else { stb_vorbis_info info = stb_vorbis_get_info(music->ctxOgg); // Get Ogg file info - //float totalLengthSeconds = stb_vorbis_stream_length_in_seconds(music->ctxOgg); - // TODO: Support 32-bit sampleSize OGGs + // OGG bit rate defaults to 16 bit, it's enough for compressed format music->stream = InitAudioStream(info.sample_rate, 16, info.channels); - music->totalSamples = (unsigned int)stb_vorbis_stream_length_in_samples(music->ctxOgg)*info.channels; + music->totalSamples = (unsigned int)stb_vorbis_stream_length_in_samples(music->ctxOgg); music->samplesLeft = music->totalSamples; music->ctxType = MUSIC_AUDIO_OGG; music->loop = true; // We loop by default @@ -584,7 +596,6 @@ Music LoadMusicStream(const char *fileName) TraceLog(DEBUG, "[%s] OGG sample rate: %i", fileName, info.sample_rate); TraceLog(DEBUG, "[%s] OGG channels: %i", fileName, info.channels); TraceLog(DEBUG, "[%s] OGG memory required: %i", fileName, info.temp_memory_required); - } } else if (strcmp(GetExtension(fileName), "flac") == 0) @@ -614,7 +625,7 @@ Music LoadMusicStream(const char *fileName) jar_xm_set_max_loop_count(music->ctxXm, 0); // Set infinite number of loops // NOTE: Only stereo is supported for XM - music->stream = InitAudioStream(48000, 32, 2); + music->stream = InitAudioStream(48000, 16, 2); music->totalSamples = (unsigned int)jar_xm_get_remaining_samples(music->ctxXm); music->samplesLeft = music->totalSamples; music->ctxType = MUSIC_MODULE_XM; @@ -637,8 +648,8 @@ Music LoadMusicStream(const char *fileName) music->ctxType = MUSIC_MODULE_MOD; music->loop = true; - TraceLog(INFO, "[%s] MOD number of samples: %i", fileName, music->samplesLeft); - TraceLog(INFO, "[%s] MOD track length: %11.6f sec", fileName, (float)music->totalSamples/48000.0f); + TraceLog(DEBUG, "[%s] MOD number of samples: %i", fileName, music->samplesLeft); + TraceLog(DEBUG, "[%s] MOD track length: %11.6f sec", fileName, (float)music->totalSamples/48000.0f); } else TraceLog(WARNING, "[%s] MOD file could not be opened", fileName); } @@ -682,7 +693,6 @@ void ResumeMusicStream(Music music) } // Stop music playing (close stream) -// TODO: Restart XM context void StopMusicStream(Music music) { alSourceStop(music->stream.source); @@ -690,7 +700,7 @@ void StopMusicStream(Music music) switch (music->ctxType) { case MUSIC_AUDIO_OGG: stb_vorbis_seek_start(music->ctxOgg); break; - case MUSIC_MODULE_XM: break; + case MUSIC_MODULE_XM: /* TODO: Restart XM context */ break; case MUSIC_MODULE_MOD: jar_mod_seek_start(&music->ctxMod); break; default: break; } @@ -710,70 +720,43 @@ void UpdateMusicStream(Music music) if (processed > 0) { bool active = true; - short pcm[AUDIO_BUFFER_SIZE]; // TODO: Dynamic allocation (uses more than 16KB of stack) - float pcmf[AUDIO_BUFFER_SIZE]; // TODO: Dynamic allocation (uses more than 16KB of stack) - int numBuffersToProcess = processed; + + // 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); + int numBuffersToProcess = processed; int numSamples = 0; // Total size of data steamed in L+R samples for xm floats, // individual L or R for ogg shorts for (int i = 0; i < numBuffersToProcess; i++) { + if (music->samplesLeft >= AUDIO_BUFFER_SIZE) numSamples = AUDIO_BUFFER_SIZE; + else numSamples = music->samplesLeft; + + // TODO: Really don't like ctxType thingy... switch (music->ctxType) { case MUSIC_AUDIO_OGG: { - if (music->samplesLeft >= AUDIO_BUFFER_SIZE) numSamples = AUDIO_BUFFER_SIZE; - else numSamples = music->samplesLeft; - - // NOTE: Returns the number of samples to process (should be the same as numSamples -> it is) - int numSamplesOgg = stb_vorbis_get_samples_short_interleaved(music->ctxOgg, music->stream.channels, pcm, numSamples); - - // TODO: Review stereo channels Ogg, not enough samples served! - UpdateAudioStream(music->stream, pcm, numSamplesOgg*music->stream.channels); - music->samplesLeft -= (numSamplesOgg*music->stream.channels); + // NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!) + int numSamplesOgg = stb_vorbis_get_samples_short_interleaved(music->ctxOgg, music->stream.channels, (short *)pcm, numSamples*music->stream.channels); } break; case MUSIC_AUDIO_FLAC: { - if (music->samplesLeft >= AUDIO_BUFFER_SIZE) numSamples = AUDIO_BUFFER_SIZE; - else numSamples = music->samplesLeft; - - int pcmi[AUDIO_BUFFER_SIZE]; - - // NOTE: Returns the number of samples to process (should be the same as numSamples) - unsigned int numSamplesFlac = (unsigned int)drflac_read_s32(music->ctxFlac, numSamples, pcmi); + // NOTE: Returns the number of samples to process + unsigned int numSamplesFlac = (unsigned int)drflac_read_s32(music->ctxFlac, numSamples/2, (int *)pcm); - UpdateAudioStream(music->stream, pcmi, numSamplesFlac*music->stream.channels); - music->samplesLeft -= (numSamples*music->stream.channels); - - } break; - case MUSIC_MODULE_XM: - { - if (music->samplesLeft >= AUDIO_BUFFER_SIZE/2) numSamples = AUDIO_BUFFER_SIZE/2; - else numSamples = music->samplesLeft; - - // NOTE: Output buffer is 2*numsamples elements (left and right value for each sample) - jar_xm_generate_samples(music->ctxXm, pcmf, numSamples); - UpdateAudioStream(music->stream, pcmf, numSamples*2); // Using 32bit PCM data - music->samplesLeft -= numSamples; - - //TraceLog(INFO, "Samples left: %i", music->samplesLeft); - - } break; - case MUSIC_MODULE_MOD: - { - if (music->samplesLeft >= AUDIO_BUFFER_SIZE/2) numSamples = AUDIO_BUFFER_SIZE/2; - else numSamples = music->samplesLeft; - - // NOTE: Output buffer size is nbsample*channels (default: 48000Hz, 16bit, Stereo) - jar_mod_fillbuffer(&music->ctxMod, pcm, numSamples, 0); - UpdateAudioStream(music->stream, pcm, numSamples*2); - music->samplesLeft -= numSamples; + // TODO: Samples should be provided as 16 bit instead of 32 bit! } break; + case MUSIC_MODULE_XM: jar_xm_generate_samples_16bit(music->ctxXm, pcm, numSamples); break; + case MUSIC_MODULE_MOD: jar_mod_fillbuffer(&music->ctxMod, pcm, numSamples, 0); break; default: break; } + + UpdateAudioStream(music->stream, pcm, numSamples); + music->samplesLeft -= numSamples; if (music->samplesLeft <= 0) { @@ -789,7 +772,6 @@ void UpdateMusicStream(Music music) if (!active) { StopMusicStream(music); // Stop music (and reset) - if (music->loop) PlayMusicStream(music); // Play again } else @@ -798,6 +780,8 @@ void UpdateMusicStream(Music music) // just make sure to play again on window restore if (state != AL_PLAYING) PlayMusicStream(music); } + + free(pcm); } } @@ -840,7 +824,7 @@ float GetMusicTimePlayed(Music music) float secondsPlayed = 0.0f; unsigned int samplesPlayed = music->totalSamples - music->samplesLeft; - secondsPlayed = (float)(samplesPlayed/(music->stream.sampleRate*music->stream.channels)); + secondsPlayed = (float)samplesPlayed/music->stream.sampleRate; return secondsPlayed; } @@ -852,30 +836,36 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un stream.sampleRate = sampleRate; stream.sampleSize = sampleSize; - stream.channels = channels; + + // Only mono and stereo channels are supported, more channels require AL_EXT_MCFORMATS extension + if ((channels > 0) && (channels < 3)) stream.channels = channels; + else + { + TraceLog(WARNING, "Init audio stream: Number of channels not supported: %i", channels); + stream.channels = 1; // Fallback to mono channel + } // Setup OpenAL format - if (channels == 1) + if (stream.channels == 1) { switch (sampleSize) { case 8: stream.format = AL_FORMAT_MONO8; break; case 16: stream.format = AL_FORMAT_MONO16; break; - case 32: stream.format = AL_FORMAT_MONO_FLOAT32; break; + case 32: //stream.format = AL_FORMAT_MONO_FLOAT32; break; // Requires OpenAL extension: AL_EXT_FLOAT32 default: TraceLog(WARNING, "Init audio stream: Sample size not supported: %i", sampleSize); break; } } - else if (channels == 2) + else if (stream.channels == 2) { switch (sampleSize) { case 8: stream.format = AL_FORMAT_STEREO8; break; case 16: stream.format = AL_FORMAT_STEREO16; break; - case 32: stream.format = AL_FORMAT_STEREO_FLOAT32; break; + case 32: //stream.format = AL_FORMAT_STEREO_FLOAT32; break; // Requires OpenAL extension: AL_EXT_FLOAT32 default: TraceLog(WARNING, "Init audio stream: Sample size not supported: %i", sampleSize); break; } } - else TraceLog(WARNING, "Init audio stream: Number of channels not supported: %i", channels); // Create an audio source alGenSources(1, &stream.source); @@ -888,28 +878,19 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un alGenBuffers(MAX_STREAM_BUFFERS, stream.buffers); // 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++) { - if (stream.sampleSize == 8) - { - unsigned char pcm[AUDIO_BUFFER_SIZE] = { 0 }; // TODO: Dynamic allocation (uses more than 16KB of stack) - alBufferData(stream.buffers[i], stream.format, pcm, AUDIO_BUFFER_SIZE*sizeof(unsigned char), stream.sampleRate); - } - else if (stream.sampleSize == 16) - { - short pcm[AUDIO_BUFFER_SIZE] = { 0 }; // TODO: Dynamic allocation (uses more than 16KB of stack) - alBufferData(stream.buffers[i], stream.format, pcm, AUDIO_BUFFER_SIZE*sizeof(short), stream.sampleRate); - } - else if (stream.sampleSize == 32) - { - float pcm[AUDIO_BUFFER_SIZE] = { 0.0f }; // TODO: Dynamic allocation (uses more than 16KB of stack) - alBufferData(stream.buffers[i], stream.format, pcm, AUDIO_BUFFER_SIZE*sizeof(float), stream.sampleRate); - } + 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); - TraceLog(INFO, "[AUD ID %i] Audio stream loaded successfully", stream.source); + TraceLog(INFO, "[AUD ID %i] Audio stream loaded successfully (%i Hz, %i bit, %s)", stream.source, stream.sampleRate, stream.sampleSize, (stream.channels == 1) ? "Mono" : "Stereo"); return stream; } @@ -940,8 +921,8 @@ void CloseAudioStream(AudioStream stream) } // Update audio stream buffers with data -// NOTE: Only one buffer per call -void UpdateAudioStream(AudioStream stream, void *data, int numSamples) +// NOTE: Only updates one buffer per call +void UpdateAudioStream(AudioStream stream, const void *data, int numSamples) { ALuint buffer = 0; alSourceUnqueueBuffers(stream.source, 1, &buffer); @@ -949,10 +930,7 @@ void UpdateAudioStream(AudioStream stream, void *data, int numSamples) // Check if any buffer was available for unqueue if (alGetError() != AL_INVALID_VALUE) { - if (stream.sampleSize == 8) alBufferData(buffer, stream.format, (unsigned char *)data, numSamples*sizeof(unsigned char), stream.sampleRate); - else if (stream.sampleSize == 16) alBufferData(buffer, stream.format, (short *)data, numSamples*sizeof(short), stream.sampleRate); - else if (stream.sampleSize == 32) alBufferData(buffer, stream.format, (float *)data, numSamples*sizeof(float), stream.sampleRate); - + alBufferData(buffer, stream.format, data, numSamples*stream.channels*stream.sampleSize/8, stream.sampleRate); alSourceQueueBuffers(stream.source, 1, &buffer); } } @@ -1007,7 +985,7 @@ static Wave LoadWAV(const char *fileName) char chunkID[4]; int chunkSize; char format[4]; - } RiffHeader; + } WavRiffHeader; typedef struct { char subChunkID[4]; @@ -1018,16 +996,16 @@ static Wave LoadWAV(const char *fileName) int byteRate; short blockAlign; short bitsPerSample; - } WaveFormat; + } WavFormat; typedef struct { char subChunkID[4]; int subChunkSize; - } WaveData; + } WavData; - RiffHeader riffHeader; - WaveFormat waveFormat; - WaveData waveData; + WavRiffHeader wavRiffHeader; + WavFormat wavFormat; + WavData wavData; Wave wave = { 0 }; FILE *wavFile; @@ -1042,56 +1020,70 @@ static Wave LoadWAV(const char *fileName) else { // Read in the first chunk into the struct - fread(&riffHeader, sizeof(RiffHeader), 1, wavFile); + fread(&wavRiffHeader, sizeof(WavRiffHeader), 1, wavFile); // Check for RIFF and WAVE tags - if (strncmp(riffHeader.chunkID, "RIFF", 4) || - strncmp(riffHeader.format, "WAVE", 4)) + if (strncmp(wavRiffHeader.chunkID, "RIFF", 4) || + strncmp(wavRiffHeader.format, "WAVE", 4)) { TraceLog(WARNING, "[%s] Invalid RIFF or WAVE Header", fileName); } else { // Read in the 2nd chunk for the wave info - fread(&waveFormat, sizeof(WaveFormat), 1, wavFile); + fread(&wavFormat, sizeof(WavFormat), 1, wavFile); // Check for fmt tag - if ((waveFormat.subChunkID[0] != 'f') || (waveFormat.subChunkID[1] != 'm') || - (waveFormat.subChunkID[2] != 't') || (waveFormat.subChunkID[3] != ' ')) + if ((wavFormat.subChunkID[0] != 'f') || (wavFormat.subChunkID[1] != 'm') || + (wavFormat.subChunkID[2] != 't') || (wavFormat.subChunkID[3] != ' ')) { TraceLog(WARNING, "[%s] Invalid Wave format", fileName); } else { // Check for extra parameters; - if (waveFormat.subChunkSize > 16) fseek(wavFile, sizeof(short), SEEK_CUR); + if (wavFormat.subChunkSize > 16) fseek(wavFile, sizeof(short), SEEK_CUR); // Read in the the last byte of data before the sound file - fread(&waveData, sizeof(WaveData), 1, wavFile); + fread(&wavData, sizeof(WavData), 1, wavFile); // Check for data tag - if ((waveData.subChunkID[0] != 'd') || (waveData.subChunkID[1] != 'a') || - (waveData.subChunkID[2] != 't') || (waveData.subChunkID[3] != 'a')) + if ((wavData.subChunkID[0] != 'd') || (wavData.subChunkID[1] != 'a') || + (wavData.subChunkID[2] != 't') || (wavData.subChunkID[3] != 'a')) { TraceLog(WARNING, "[%s] Invalid data header", fileName); } else { // Allocate memory for data - wave.data = (unsigned char *)malloc(sizeof(unsigned char)*waveData.subChunkSize); + wave.data = (unsigned char *)malloc(sizeof(unsigned char)*wavData.subChunkSize); // Read in the sound data into the soundData variable - fread(wave.data, waveData.subChunkSize, 1, wavFile); + fread(wave.data, wavData.subChunkSize, 1, wavFile); // Store wave parameters - wave.sampleRate = waveFormat.sampleRate; - wave.sampleSize = waveFormat.bitsPerSample; - wave.channels = waveFormat.numChannels; + wave.sampleRate = wavFormat.sampleRate; + wave.sampleSize = wavFormat.bitsPerSample; + wave.channels = wavFormat.numChannels; + + // NOTE: Only support up to 16 bit sample sizes + if (wave.sampleSize > 16) + { + WaveFormat(&wave, wave.sampleRate, 16, wave.channels); + TraceLog(WARNING, "[%s] WAV sample size (%ibit) not supported, converted to 16bit", fileName, wave.sampleSize); + } + + // NOTE: Only support up to 2 channels (mono, stereo) + 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 = waveData.subChunkSize/(waveFormat.bitsPerSample/8); + wave.sampleCount = (wavData.subChunkSize/(wave.sampleSize/8))/wave.channels; - TraceLog(INFO, "[%s] WAV file loaded successfully (SampleRate: %i, SampleSize: %i, Channels: %i)", fileName, wave.sampleRate, wave.sampleSize, wave.channels); + TraceLog(INFO, "[%s] WAV file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo"); } } } @@ -1137,7 +1129,7 @@ static Wave LoadOGG(const char *fileName) TraceLog(DEBUG, "[%s] Samples obtained: %i", fileName, samplesObtained); - TraceLog(INFO, "[%s] OGG file loaded successfully (SampleRate: %i, SampleSize: %i, Channels: %i)", fileName, wave.sampleRate, wave.sampleSize, wave.channels); + TraceLog(INFO, "[%s] OGG file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo"); stb_vorbis_close(oggFile); } @@ -1156,9 +1148,20 @@ static Wave LoadFLAC(const char *fileName) wave.data = drflac_open_and_decode_file_s32(fileName, &wave.channels, &wave.sampleRate, &totalSampleCount); wave.sampleCount = (int)totalSampleCount; - wave.sampleSize = 32; + wave.sampleSize = 32; // 32 bit per sample (float) + + // NOTE: By default, dr_flac returns 32bit float samples, needs to be converted to 16bit + WaveFormat(&wave, wave.sampleRate, 16, wave.channels); + + // NOTE: Only support up to 2 channels (mono, stereo) + if (wave.channels > 2) + { + WaveFormat(&wave, wave.sampleRate, wave.sampleSize, 2); + TraceLog(WARNING, "[%s] FLAC channels number (%i) not supported, converted to 2 channels", 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; } diff --git a/src/audio.h b/src/audio.h index db1bb694..6f0c235a 100644 --- a/src/audio.h +++ b/src/audio.h @@ -110,12 +110,11 @@ void InitAudioDevice(void); // Initialize au void CloseAudioDevice(void); // Close the audio device and context bool IsAudioDeviceReady(void); // Check if audio device has been initialized successfully -Wave LoadWave(const char *fileName); // Load wave data from file into RAM -Wave LoadWaveEx(float *data, int sampleCount, int sampleRate, int sampleSize, int channels); // Load wave data from float array data (32bit) -Sound LoadSound(const char *fileName); // Load sound to memory -Sound LoadSoundFromWave(Wave wave); // Load sound to memory from wave data -Sound LoadSoundFromRES(const char *rresName, int resId); // Load sound to memory from rRES file (raylib Resource) -void UpdateSound(Sound sound, const void *data, int numSamples); // Update sound buffer with new data +Wave LoadWave(const char *fileName); // Load wave data from file +Wave LoadWaveEx(void *data, int sampleCount, int sampleRate, int sampleSize, int channels); // Load wave data from raw array data +Sound LoadSound(const char *fileName); // Load sound from file +Sound LoadSoundFromWave(Wave wave); // Load sound from wave data +void UpdateSound(Sound sound, const void *data, int numSamples);// Update sound buffer with new data void UnloadWave(Wave wave); // Unload wave data void UnloadSound(Sound sound); // Unload sound void PlaySound(Sound sound); // Play a sound