From 76ff4d220ee735b8b86bd4dae776665cf68e4fb4 Mon Sep 17 00:00:00 2001 From: Joshua Reisenauer Date: Sun, 15 May 2016 19:37:15 -0700 Subject: [PATCH] renamed everything so it is obvious what it does --- src/audio.c | 380 ++++++++++++++++++++++++--------------------------- src/audio.h | 15 +- src/raylib.h | 15 +- 3 files changed, 190 insertions(+), 220 deletions(-) diff --git a/src/audio.c b/src/audio.c index cc5ca1a27..584d3ad14 100644 --- a/src/audio.c +++ b/src/audio.c @@ -77,10 +77,10 @@ // Types and Structures Definition //---------------------------------------------------------------------------------- -// Audio Context, used to create custom audio streams that are not bound to a sound file. There can be -// no more than 4 concurrent audio contexts in use. This is due to each active context being tied to -// a dedicated mix channel. All audio is 32bit floating point in stereo. -typedef struct AudioContext_t { +// Used to create custom audio streams that are not bound to a specific file. There can be +// no more than 4 concurrent mixchannels in use. This is due to each active mixc being tied to +// a dedicated mix channel. +typedef struct MixChannel_t { unsigned short sampleRate; // default is 48000 unsigned char channels; // 1=mono,2=stereo unsigned char mixChannel; // 0-3 or mixA-mixD, each mix channel can receive up to one dedicated audio stream @@ -89,14 +89,14 @@ typedef struct AudioContext_t { ALenum alFormat; // openAL format specifier ALuint alSource; // openAL source ALuint alBuffer[MAX_STREAM_BUFFERS]; // openAL sample buffer -} AudioContext_t; +} MixChannel_t; // Music type (file streaming from memory) -// NOTE: Anything longer than ~10 seconds should be streamed... +// NOTE: Anything longer than ~10 seconds should be streamed into a mix channel... typedef struct Music { stb_vorbis *stream; - jar_xm_context_t *chipctx; // Stores jar_xm context - AudioContext_t *ctx; // audio context + jar_xm_context_t *chipctx; // Stores jar_xm mixc + MixChannel_t *mixc; // mix channel int totalSamplesLeft; float totalLengthSeconds; @@ -111,9 +111,9 @@ typedef enum { INFO = 0, ERROR, WARNING, DEBUG, OTHER } TraceLogType; //---------------------------------------------------------------------------------- // Global Variables Definition //---------------------------------------------------------------------------------- -static AudioContext_t* mixChannelsActive_g[MAX_AUDIO_CONTEXTS]; // What mix channels are currently active +static MixChannel_t* mixChannelsActive_g[MAX_AUDIO_CONTEXTS]; // What mix channels are currently active static bool musicEnabled_g = false; -static Music currentMusic[MAX_MUSIC_STREAMS]; // Current music loaded, up to two can play at the same time +static Music currentMusic[MAX_MUSIC_STREAMS]; // Current music loaded, up to two can play at the same time //---------------------------------------------------------------------------------- // Module specific Functions Declaration @@ -122,13 +122,17 @@ static Wave LoadWAV(const char *fileName); // Load WAV file static Wave LoadOGG(char *fileName); // Load OGG file static void UnloadWave(Wave wave); // Unload wave data -static bool BufferMusicStream(int index); // Fill music buffers with data -static void EmptyMusicStream(int index); // Empty music buffers +static bool BufferMusicStream(int index, int numBuffers); // Fill music buffers with data +static void EmptyMusicStream(int index); // Empty music buffers -static unsigned short FillAlBufferWithSilence(AudioContext_t *context, ALuint buffer);// fill buffer with zeros, returns number processed -static void ResampleShortToFloat(short *shorts, float *floats, unsigned short len); // pass two arrays of the same legnth in -static void ResampleByteToFloat(char *chars, float *floats, unsigned short len); // pass two arrays of same length in -static bool isMusicStreamReady(int index); // Checks if music buffer is ready to be refilled + +static MixChannel_t* InitMixChannel(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint); // For streaming into mix channels. +static void CloseMixChannel(MixChannel_t* mixc); // Frees mix channel +static unsigned short BufferMixChannel(MixChannel_t* mixc, void *data, int numberElements); // Pushes more audio data into mixc mix channel, if NULL is passed it pauses +static unsigned short FillAlBufferWithSilence(MixChannel_t *mixc, ALuint buffer); // Fill buffer with zeros, returns number processed +static void ResampleShortToFloat(short *shorts, float *floats, unsigned short len); // Pass two arrays of the same legnth in +static void ResampleByteToFloat(char *chars, float *floats, unsigned short len); // Pass two arrays of same length in +static int IsMusicStreamReadyForBuffering(int index); // Checks if music buffer is ready to be refilled #if defined(AUDIO_STANDALONE) const char *GetExtension(const char *fileName); // Get the extension for a filename @@ -139,7 +143,7 @@ void TraceLog(int msgType, const char *text, ...); // Outputs a trace log messa // Module Functions Definition - Audio Device initialization and Closing //---------------------------------------------------------------------------------- -// Initialize audio device and context +// Initialize audio device and mixc void InitAudioDevice(void) { // Open and initialize a device with default settings @@ -155,7 +159,7 @@ void InitAudioDevice(void) alcCloseDevice(device); - TraceLog(ERROR, "Could not setup audio context"); + TraceLog(ERROR, "Could not setup mix channel"); } TraceLog(INFO, "Audio device and context initialized successfully: %s", alcGetString(device, ALC_DEVICE_SPECIFIER)); @@ -171,14 +175,14 @@ void CloseAudioDevice(void) { for(int index=0; index= MAX_AUDIO_CONTEXTS) return NULL; if(!IsAudioDeviceReady()) InitAudioDevice(); if(!mixChannelsActive_g[mixChannel]){ - AudioContext_t *ac = (AudioContext_t*)malloc(sizeof(AudioContext_t)); - ac->sampleRate = sampleRate; - ac->channels = channels; - ac->mixChannel = mixChannel; - ac->floatingPoint = floatingPoint; - mixChannelsActive_g[mixChannel] = ac; + MixChannel_t *mixc = (MixChannel_t*)malloc(sizeof(MixChannel_t)); + mixc->sampleRate = sampleRate; + mixc->channels = channels; + mixc->mixChannel = mixChannel; + mixc->floatingPoint = floatingPoint; + mixChannelsActive_g[mixChannel] = mixc; // setup openAL format if(channels == 1) { if(floatingPoint) - ac->alFormat = AL_FORMAT_MONO_FLOAT32; + mixc->alFormat = AL_FORMAT_MONO_FLOAT32; else - ac->alFormat = AL_FORMAT_MONO16; + mixc->alFormat = AL_FORMAT_MONO16; } else if(channels == 2) { if(floatingPoint) - ac->alFormat = AL_FORMAT_STEREO_FLOAT32; + mixc->alFormat = AL_FORMAT_STEREO_FLOAT32; else - ac->alFormat = AL_FORMAT_STEREO16; + mixc->alFormat = AL_FORMAT_STEREO16; } // Create an audio source - alGenSources(1, &ac->alSource); - alSourcef(ac->alSource, AL_PITCH, 1); - alSourcef(ac->alSource, AL_GAIN, 1); - alSource3f(ac->alSource, AL_POSITION, 0, 0, 0); - alSource3f(ac->alSource, AL_VELOCITY, 0, 0, 0); + alGenSources(1, &mixc->alSource); + alSourcef(mixc->alSource, AL_PITCH, 1); + alSourcef(mixc->alSource, AL_GAIN, 1); + alSource3f(mixc->alSource, AL_POSITION, 0, 0, 0); + alSource3f(mixc->alSource, AL_VELOCITY, 0, 0, 0); // Create Buffer - alGenBuffers(MAX_STREAM_BUFFERS, ac->alBuffer); + alGenBuffers(MAX_STREAM_BUFFERS, mixc->alBuffer); //fill buffers int x; for(x=0;xalBuffer[x]); + FillAlBufferWithSilence(mixc, mixc->alBuffer[x]); - alSourceQueueBuffers(ac->alSource, MAX_STREAM_BUFFERS, ac->alBuffer); - alSourcePlay(ac->alSource); - ac->playing = true; + alSourceQueueBuffers(mixc->alSource, MAX_STREAM_BUFFERS, mixc->alBuffer); + mixc->playing = true; + alSourcePlay(mixc->alSource); - return ac; + return mixc; } return NULL; } -// Frees buffer in audio context -void CloseAudioContext(AudioContext ctx) +// Frees buffer in mix channel +static void CloseMixChannel(MixChannel_t* mixc) { - AudioContext_t *context = (AudioContext_t*)ctx; - if(context){ - alSourceStop(context->alSource); - context->playing = false; + if(mixc){ + alSourceStop(mixc->alSource); + mixc->playing = false; //flush out all queued buffers ALuint buffer = 0; int queued = 0; - alGetSourcei(context->alSource, AL_BUFFERS_QUEUED, &queued); + alGetSourcei(mixc->alSource, AL_BUFFERS_QUEUED, &queued); while (queued > 0) { - alSourceUnqueueBuffers(context->alSource, 1, &buffer); + alSourceUnqueueBuffers(mixc->alSource, 1, &buffer); queued--; } //delete source and buffers - alDeleteSources(1, &context->alSource); - alDeleteBuffers(MAX_STREAM_BUFFERS, context->alBuffer); - mixChannelsActive_g[context->mixChannel] = NULL; - free(context); - ctx = NULL; + alDeleteSources(1, &mixc->alSource); + alDeleteBuffers(MAX_STREAM_BUFFERS, mixc->alBuffer); + mixChannelsActive_g[mixc->mixChannel] = NULL; + free(mixc); + mixc = NULL; } } -// Pushes more audio data into context mix channel, if none are ever pushed then zeros are fed in. -// Call "UpdateAudioContext(ctx, NULL, 0)" if you want to pause the audio. +// Pushes more audio data into mixc mix channel, only one buffer per call +// Call "BufferMixChannel(mixc, NULL, 0)" if you want to pause the audio. // @Returns number of samples that where processed. -unsigned short UpdateAudioContext(AudioContext ctx, void *data, unsigned short numberElements) +static unsigned short BufferMixChannel(MixChannel_t* mixc, void *data, int numberElements) { - AudioContext_t *context = (AudioContext_t*)ctx; - - if(!context || (context->channels == 2 && numberElements % 2 != 0)) return 0; // when there is two channels there must be an even number of samples + if(!mixc || mixChannelsActive_g[mixc->mixChannel] != mixc) return 0; // when there is two channels there must be an even number of samples if (!data || !numberElements) { // pauses audio until data is given - alSourcePause(context->alSource); - context->playing = false; + if(mixc->playing){ + alSourcePause(mixc->alSource); + mixc->playing = false; + } return 0; } - else + else if(!mixc->playing) { // restart audio otherwise - ALint state; - alGetSourcei(context->alSource, AL_SOURCE_STATE, &state); - if (state != AL_PLAYING){ - alSourcePlay(context->alSource); - context->playing = true; - } + alSourcePlay(mixc->alSource); + mixc->playing = true; } - if (context && context->playing && mixChannelsActive_g[context->mixChannel] == context) + + ALuint buffer = 0; + + alSourceUnqueueBuffers(mixc->alSource, 1, &buffer); + if(!buffer) return 0; + if(mixc->floatingPoint) // process float buffers { - ALint processed = 0; - ALuint buffer = 0; - unsigned short numberProcessed = 0; - unsigned short numberRemaining = numberElements; - - - alGetSourcei(context->alSource, AL_BUFFERS_PROCESSED, &processed); // Get the number of already processed buffers (if any) - if(!processed) return 0; // nothing to process, queue is still full - - - while (processed > 0) - { - if(context->floatingPoint) // process float buffers - { - float *ptr = (float*)data; - alSourceUnqueueBuffers(context->alSource, 1, &buffer); - if(numberRemaining >= MUSIC_BUFFER_SIZE_FLOAT) - { - alBufferData(buffer, context->alFormat, &ptr[numberProcessed], MUSIC_BUFFER_SIZE_FLOAT*sizeof(float), context->sampleRate); - numberProcessed+=MUSIC_BUFFER_SIZE_FLOAT; - numberRemaining-=MUSIC_BUFFER_SIZE_FLOAT; - } - else - { - alBufferData(buffer, context->alFormat, &ptr[numberProcessed], numberRemaining*sizeof(float), context->sampleRate); - numberProcessed+=numberRemaining; - numberRemaining=0; - } - alSourceQueueBuffers(context->alSource, 1, &buffer); - processed--; - } - else if(!context->floatingPoint) // process short buffers - { - short *ptr = (short*)data; - alSourceUnqueueBuffers(context->alSource, 1, &buffer); - if(numberRemaining >= MUSIC_BUFFER_SIZE_SHORT) - { - alBufferData(buffer, context->alFormat, &ptr[numberProcessed], MUSIC_BUFFER_SIZE_FLOAT*sizeof(short), context->sampleRate); - numberProcessed+=MUSIC_BUFFER_SIZE_SHORT; - numberRemaining-=MUSIC_BUFFER_SIZE_SHORT; - } - else - { - alBufferData(buffer, context->alFormat, &ptr[numberProcessed], numberRemaining*sizeof(short), context->sampleRate); - numberProcessed+=numberRemaining; - numberRemaining=0; - } - alSourceQueueBuffers(context->alSource, 1, &buffer); - processed--; - } - else - break; - } - return numberProcessed; + float *ptr = (float*)data; + alBufferData(buffer, mixc->alFormat, ptr, numberElements*sizeof(float), mixc->sampleRate); } - return 0; + else // process short buffers + { + short *ptr = (short*)data; + alBufferData(buffer, mixc->alFormat, ptr, numberElements*sizeof(short), mixc->sampleRate); + } + alSourceQueueBuffers(mixc->alSource, 1, &buffer); + + return numberElements; } // fill buffer with zeros, returns number processed -static unsigned short FillAlBufferWithSilence(AudioContext_t *context, ALuint buffer) +static unsigned short FillAlBufferWithSilence(MixChannel_t *mixc, ALuint buffer) { - if(context->floatingPoint){ + if(mixc->floatingPoint){ float pcm[MUSIC_BUFFER_SIZE_FLOAT] = {0.f}; - alBufferData(buffer, context->alFormat, pcm, MUSIC_BUFFER_SIZE_FLOAT*sizeof(float), context->sampleRate); + alBufferData(buffer, mixc->alFormat, pcm, MUSIC_BUFFER_SIZE_FLOAT*sizeof(float), mixc->sampleRate); return MUSIC_BUFFER_SIZE_FLOAT; } else { short pcm[MUSIC_BUFFER_SIZE_SHORT] = {0}; - alBufferData(buffer, context->alFormat, pcm, MUSIC_BUFFER_SIZE_SHORT*sizeof(short), context->sampleRate); + alBufferData(buffer, mixc->alFormat, pcm, MUSIC_BUFFER_SIZE_SHORT*sizeof(short), mixc->sampleRate); return MUSIC_BUFFER_SIZE_SHORT; } } @@ -417,6 +376,28 @@ static void ResampleByteToFloat(char *chars, float *floats, unsigned short len) } } +// used to output raw audio streams, returns negative numbers on error +RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint) +{ + int mixIndex; + for(mixIndex = 0; mixIndex < MAX_AUDIO_CONTEXTS; mixIndex++) // find empty mix channel slot + { + if(mixChannelsActive_g[mixIndex] == NULL) break; + else if(mixIndex = MAX_AUDIO_CONTEXTS - 1) return -1; // error + } + + if(InitMixChannel(sampleRate, mixIndex, channels, floatingPoint)) + return mixIndex; + else + return -2; // error +} + +void CloseRawAudioContext(RawAudioContext ctx) +{ + if(mixChannelsActive_g[ctx]) + CloseMixChannel(mixChannelsActive_g[ctx]); +} + //---------------------------------------------------------------------------------- @@ -807,14 +788,14 @@ int PlayMusicStream(int musicIndex, char *fileName) currentMusic[musicIndex].totalLengthSeconds = stb_vorbis_stream_length_in_seconds(currentMusic[musicIndex].stream); if (info.channels == 2){ - currentMusic[musicIndex].ctx = InitAudioContext(info.sample_rate, mixIndex, 2, false); - currentMusic[musicIndex].ctx->playing = true; + currentMusic[musicIndex].mixc = InitMixChannel(info.sample_rate, mixIndex, 2, false); + currentMusic[musicIndex].mixc->playing = true; } else{ - currentMusic[musicIndex].ctx = InitAudioContext(info.sample_rate, mixIndex, 1, false); - currentMusic[musicIndex].ctx->playing = true; + currentMusic[musicIndex].mixc = InitMixChannel(info.sample_rate, mixIndex, 1, false); + currentMusic[musicIndex].mixc->playing = true; } - if(!currentMusic[musicIndex].ctx) return 4; // error + if(!currentMusic[musicIndex].mixc) return 4; // error } } else if (strcmp(GetExtension(fileName),"xm") == 0) @@ -832,9 +813,9 @@ int PlayMusicStream(int musicIndex, char *fileName) TraceLog(INFO, "[%s] XM number of samples: %i", fileName, currentMusic[musicIndex].totalSamplesLeft); TraceLog(INFO, "[%s] XM track length: %11.6f sec", fileName, currentMusic[musicIndex].totalLengthSeconds); - currentMusic[musicIndex].ctx = InitAudioContext(48000, mixIndex, 2, true); - if(!currentMusic[musicIndex].ctx) return 5; // error - currentMusic[musicIndex].ctx->playing = true; + currentMusic[musicIndex].mixc = InitMixChannel(48000, mixIndex, 2, false); + if(!currentMusic[musicIndex].mixc) return 5; // error + currentMusic[musicIndex].mixc->playing = true; } else { @@ -853,9 +834,9 @@ int PlayMusicStream(int musicIndex, char *fileName) // Stop music playing for individual music index of currentMusic array (close stream) void StopMusicStream(int index) { - if (index < MAX_MUSIC_STREAMS && currentMusic[index].ctx) + if (index < MAX_MUSIC_STREAMS && currentMusic[index].mixc) { - CloseAudioContext(currentMusic[index].ctx); + CloseMixChannel(currentMusic[index].mixc); if (currentMusic[index].chipTune) { @@ -889,11 +870,11 @@ int getMusicStreamCount(void) void PauseMusicStream(int index) { // Pause music stream if music available! - if (index < MAX_MUSIC_STREAMS && currentMusic[index].ctx && musicEnabled_g) + if (index < MAX_MUSIC_STREAMS && currentMusic[index].mixc && musicEnabled_g) { TraceLog(INFO, "Pausing music stream"); - alSourcePause(currentMusic[index].ctx->alSource); - currentMusic[index].ctx->playing = false; + alSourcePause(currentMusic[index].mixc->alSource); + currentMusic[index].mixc->playing = false; } } @@ -902,13 +883,13 @@ void ResumeMusicStream(int index) { // Resume music playing... if music available! ALenum state; - if(index < MAX_MUSIC_STREAMS && currentMusic[index].ctx){ - alGetSourcei(currentMusic[index].ctx->alSource, AL_SOURCE_STATE, &state); + if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){ + alGetSourcei(currentMusic[index].mixc->alSource, AL_SOURCE_STATE, &state); if (state == AL_PAUSED) { TraceLog(INFO, "Resuming music stream"); - alSourcePlay(currentMusic[index].ctx->alSource); - currentMusic[index].ctx->playing = true; + alSourcePlay(currentMusic[index].mixc->alSource); + currentMusic[index].mixc->playing = true; } } } @@ -919,8 +900,8 @@ bool IsMusicPlaying(int index) bool playing = false; ALint state; - if(index < MAX_MUSIC_STREAMS && currentMusic[index].ctx){ - alGetSourcei(currentMusic[index].ctx->alSource, AL_SOURCE_STATE, &state); + if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){ + alGetSourcei(currentMusic[index].mixc->alSource, AL_SOURCE_STATE, &state); if (state == AL_PLAYING) playing = true; } @@ -930,15 +911,15 @@ bool IsMusicPlaying(int index) // Set volume for music void SetMusicVolume(int index, float volume) { - if(index < MAX_MUSIC_STREAMS && currentMusic[index].ctx){ - alSourcef(currentMusic[index].ctx->alSource, AL_GAIN, volume); + if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){ + alSourcef(currentMusic[index].mixc->alSource, AL_GAIN, volume); } } void SetMusicPitch(int index, float pitch) { - if(index < MAX_MUSIC_STREAMS && currentMusic[index].ctx){ - alSourcef(currentMusic[index].ctx->alSource, AL_PITCH, pitch); + if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){ + alSourcef(currentMusic[index].mixc->alSource, AL_PITCH, pitch); } } @@ -962,19 +943,19 @@ float GetMusicTimeLength(int index) float GetMusicTimePlayed(int index) { float secondsPlayed; - if(index < MAX_MUSIC_STREAMS && currentMusic[index].ctx) + if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc) { if (currentMusic[index].chipTune) { uint64_t samples; jar_xm_get_position(currentMusic[index].chipctx, NULL, NULL, NULL, &samples); - secondsPlayed = (float)samples / (48000 * currentMusic[index].ctx->channels); // Not sure if this is the correct value + secondsPlayed = (float)samples / (48000 * currentMusic[index].mixc->channels); // Not sure if this is the correct value } else { - int totalSamples = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].ctx->channels; + int totalSamples = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].mixc->channels; int samplesPlayed = totalSamples - currentMusic[index].totalSamplesLeft; - secondsPlayed = (float)samplesPlayed / (currentMusic[index].ctx->sampleRate * currentMusic[index].ctx->channels); + secondsPlayed = (float)samplesPlayed / (currentMusic[index].mixc->sampleRate * currentMusic[index].mixc->channels); } } @@ -987,32 +968,32 @@ float GetMusicTimePlayed(int index) //---------------------------------------------------------------------------------- // Fill music buffers with new data from music stream -static bool BufferMusicStream(int index) +static bool BufferMusicStream(int index, int numBuffers) { short pcm[MUSIC_BUFFER_SIZE_SHORT]; float pcmf[MUSIC_BUFFER_SIZE_FLOAT]; - int size = 0; // Total size of data steamed in L+R samples + int size = 0; // Total size of data steamed in L+R samples for xm floats, individual L or R for ogg shorts bool active = true; // We can get more data from stream (not finished) - - - if (!currentMusic[index].ctx->playing && currentMusic[index].totalSamplesLeft > 0) - { - UpdateAudioContext(currentMusic[index].ctx, NULL, 0); - return true; // it is still active but it is paused - } - if (currentMusic[index].chipTune) // There is no end of stream for xmfiles, once the end is reached zeros are generated for non looped chiptunes. { - if(currentMusic[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_FLOAT / 2) - size = MUSIC_BUFFER_SIZE_FLOAT / 2; + if(currentMusic[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT) + size = MUSIC_BUFFER_SIZE_SHORT / 2; else size = currentMusic[index].totalSamplesLeft / 2; - - jar_xm_generate_samples(currentMusic[index].chipctx, pcmf, size); // reads 2*readlen shorts and moves them to buffer+size memory location - UpdateAudioContext(currentMusic[index].ctx, pcmf, size * 2); - currentMusic[index].totalSamplesLeft -= size * 2; + + for(int x=0; xchannels, pcm, size); - UpdateAudioContext(currentMusic[index].ctx, pcm, streamedBytes * currentMusic[index].ctx->channels); - currentMusic[index].totalSamplesLeft -= streamedBytes * currentMusic[index].ctx->channels; + for(int x=0; xchannels, pcm, size); + BufferMixChannel(currentMusic[index].mixc, pcm, streamedBytes * currentMusic[index].mixc->channels); + currentMusic[index].totalSamplesLeft -= streamedBytes * currentMusic[index].mixc->channels; + if(currentMusic[index].totalSamplesLeft <= 0) + { + active = false; + break; + } + } } - - TraceLog(DEBUG, "Buffering index:%i, chiptune:%i", index, (int)currentMusic[index].chipTune); - if(currentMusic[index].totalSamplesLeft <= 0) active = false; return active; } @@ -1038,25 +1024,22 @@ static void EmptyMusicStream(int index) ALuint buffer = 0; int queued = 0; - alGetSourcei(currentMusic[index].ctx->alSource, AL_BUFFERS_QUEUED, &queued); + alGetSourcei(currentMusic[index].mixc->alSource, AL_BUFFERS_QUEUED, &queued); while (queued > 0) { - alSourceUnqueueBuffers(currentMusic[index].ctx->alSource, 1, &buffer); + alSourceUnqueueBuffers(currentMusic[index].mixc->alSource, 1, &buffer); queued--; } } //determine if a music stream is ready to be written to -static bool isMusicStreamReady(int index) +static int IsMusicStreamReadyForBuffering(int index) { ALint processed = 0; - alGetSourcei(currentMusic[index].ctx->alSource, AL_BUFFERS_PROCESSED, &processed); - - if(processed) return true; - - return false; + alGetSourcei(currentMusic[index].mixc->alSource, AL_BUFFERS_PROCESSED, &processed); + return processed; } // Update (re-fill) music buffers if data already processed @@ -1064,21 +1047,22 @@ void UpdateMusicStream(int index) { ALenum state; bool active = true; - - if (index < MAX_MUSIC_STREAMS && musicEnabled_g && currentMusic[index].ctx && isMusicStreamReady(index)) + int numBuffers = IsMusicStreamReadyForBuffering(index); + + if (currentMusic[index].mixc->playing && index < MAX_MUSIC_STREAMS && musicEnabled_g && currentMusic[index].mixc && numBuffers) { - active = BufferMusicStream(index); + active = BufferMusicStream(index, numBuffers); - if (!active && currentMusic[index].loop && currentMusic[index].ctx->playing) + if (!active && currentMusic[index].loop) { if (currentMusic[index].chipTune) { - currentMusic[index].totalSamplesLeft = currentMusic[index].totalLengthSeconds * currentMusic[index].ctx->sampleRate; + currentMusic[index].totalSamplesLeft = currentMusic[index].totalLengthSeconds * 48000; } else { stb_vorbis_seek_start(currentMusic[index].stream); - currentMusic[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].ctx->channels; + currentMusic[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].mixc->channels; } active = true; } @@ -1086,9 +1070,9 @@ void UpdateMusicStream(int index) if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data..."); - alGetSourcei(currentMusic[index].ctx->alSource, AL_SOURCE_STATE, &state); + alGetSourcei(currentMusic[index].mixc->alSource, AL_SOURCE_STATE, &state); - if (state != AL_PLAYING && active && currentMusic[index].ctx->playing) alSourcePlay(currentMusic[index].ctx->alSource); + if (state != AL_PLAYING && active) alSourcePlay(currentMusic[index].mixc->alSource); if (!active) StopMusicStream(index); diff --git a/src/audio.h b/src/audio.h index 63c1c1365..d3276bf6e 100644 --- a/src/audio.h +++ b/src/audio.h @@ -61,10 +61,7 @@ typedef struct Wave { short channels; } Wave; -// Audio Context, used to create custom audio streams that are not bound to a sound file. There can be -// no more than 4 concurrent audio contexts in use. This is due to each active context being tied to -// a dedicated mix channel. -typedef void* AudioContext; +typedef int RawAudioContext; #ifdef __cplusplus extern "C" { // Prevents name mangling of functions @@ -82,13 +79,6 @@ void InitAudioDevice(void); // Initialize au void CloseAudioDevice(void); // Close the audio device and context (and music stream) bool IsAudioDeviceReady(void); // True if call to InitAudioDevice() was successful and CloseAudioDevice() has not been called yet -// Audio contexts are for outputing custom audio waveforms, This will shut down any other sound sources currently playing -// The mixChannel is what mix channel you want to operate on, 0-3 are the ones available. Each mix channel can only be used one at a time. -// exmple usage is InitAudioContext(48000, 0, 2, true); // mixchannel 1, 48khz, stereo, floating point -AudioContext InitAudioContext(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint); -void CloseAudioContext(AudioContext ctx); // Frees audio context -unsigned short UpdateAudioContext(AudioContext ctx, void *data, unsigned short numberElements); // Pushes more audio data into context mix channel, if NULL is passed to data then zeros are played - Sound LoadSound(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) @@ -112,6 +102,9 @@ float GetMusicTimePlayed(int index); // Get current m int getMusicStreamCount(void); void SetMusicPitch(int index, float pitch); +RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint); +void CloseRawAudioContext(RawAudioContext ctx); + #ifdef __cplusplus } #endif diff --git a/src/raylib.h b/src/raylib.h index ea9fbfcb6..6efde7102 100644 --- a/src/raylib.h +++ b/src/raylib.h @@ -451,10 +451,7 @@ typedef struct Wave { short channels; } Wave; -// Audio Context, used to create custom audio streams that are not bound to a sound file. There can be -// no more than 4 concurrent audio contexts in use. This is due to each active context being tied to -// a dedicated mix channel. -typedef void* AudioContext; +typedef int RawAudioContext; // Texture formats // NOTE: Support depends on OpenGL version and platform @@ -876,13 +873,6 @@ void InitAudioDevice(void); // Initialize au void CloseAudioDevice(void); // Close the audio device and context (and music stream) bool IsAudioDeviceReady(void); // True if call to InitAudioDevice() was successful and CloseAudioDevice() has not been called yet -// Audio contexts are for outputing custom audio waveforms, This will shut down any other sound sources currently playing -// The mixChannel is what mix channel you want to operate on, 0-3 are the ones available. Each mix channel can only be used one at a time. -// exmple usage is InitAudioContext(48000, 0, 2, true); // mixchannel 1, 48khz, stereo, floating point -AudioContext InitAudioContext(unsigned short sampleRate, unsigned char mixChannel, unsigned char channels, bool floatingPoint); -void CloseAudioContext(AudioContext ctx); // Frees audio context -unsigned short UpdateAudioContext(AudioContext ctx, void *data, unsigned short numberElements); // Pushes more audio data into context mix channel, if NULL is passed to data then zeros are played - Sound LoadSound(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) @@ -906,6 +896,9 @@ float GetMusicTimePlayed(int index); // Get current m int getMusicStreamCount(void); void SetMusicPitch(int index, float pitch); +RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint); // used to output raw audio streams, returns negative numbers on error +void CloseRawAudioContext(RawAudioContext ctx); + #ifdef __cplusplus } #endif