diff --git a/src/audio.c b/src/audio.c index fbf53df6..43e8be14 100644 --- a/src/audio.c +++ b/src/audio.c @@ -59,8 +59,9 @@ //---------------------------------------------------------------------------------- // Defines and Macros //---------------------------------------------------------------------------------- -#define MAX_STREAM_BUFFERS 2 -#define MAX_AUDIO_CONTEXTS 4 // Number of open AL sources +#define MAX_STREAM_BUFFERS 2 // Number of buffers for each alSource +#define MAX_MIX_CHANNELS 4 // Number of open AL sources +#define MAX_MUSIC_STREAMS 2 // Number of simultanious music sources #if defined(PLATFORM_RPI) || defined(PLATFORM_ANDROID) // NOTE: On RPI and Android should be lower to avoid frame-stalls @@ -76,37 +77,32 @@ // Types and Structures Definition //---------------------------------------------------------------------------------- -// Music type (file streaming from memory) -// NOTE: Anything longer than ~10 seconds should be streamed... -typedef struct Music { - stb_vorbis *stream; - jar_xm_context_t *chipctx; // Stores jar_xm context - - ALuint buffers[MAX_STREAM_BUFFERS]; - ALuint source; - ALenum format; - - int channels; - int sampleRate; - int totalSamplesLeft; - float totalLengthSeconds; - bool loop; - bool chipTune; // True if chiptune is loaded -} Music; - -// 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 bool floatingPoint; // if false then the short datatype is used instead - bool playing; + bool playing; // false if paused 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 into a mix channel... +typedef struct Music { + stb_vorbis *stream; + jar_xm_context_t *chipctx; // Stores jar_xm mixc + MixChannel_t *mixc; // mix channel + + int totalSamplesLeft; + float totalLengthSeconds; + bool loop; + bool chipTune; // True if chiptune is loaded +} Music; #if defined(AUDIO_STANDALONE) typedef enum { INFO = 0, ERROR, WARNING, DEBUG, OTHER } TraceLogType; @@ -115,23 +111,28 @@ 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 bool musicEnabled = false; -static Music currentMusic; // Current music loaded - // NOTE: Only one music file playing at a time +static MixChannel_t* mixChannelsActive_g[MAX_MIX_CHANNELS]; // 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 + //---------------------------------------------------------------------------------- // Module specific Functions Declaration //---------------------------------------------------------------------------------- -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 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(ALuint buffer); // Fill music buffers with data -static void EmptyMusicStream(void); // 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 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 int BufferMixChannel(MixChannel_t* mixc, void *data, int numberElements); // Pushes more audio data into mixc mix channel, if NULL is passed it pauses +static int 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 @@ -142,7 +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 @@ -158,7 +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)); @@ -169,15 +170,19 @@ void InitAudioDevice(void) alListener3f(AL_ORIENTATION, 0, 0, -1); } -// Close the audio device for the current context, and destroys the context +// Close the audio device for all contexts void CloseAudioDevice(void) { - StopMusicStream(); // Stop music streaming and close current stream + for(int index=0; index= MAX_AUDIO_CONTEXTS) return NULL; + if(mixChannel >= MAX_MIX_CHANNELS) return NULL; if(!IsAudioDeviceReady()) InitAudioDevice(); - else StopMusicStream(); 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 int 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); + } + else // process short buffers + { + short *ptr = (short*)data; + alBufferData(buffer, mixc->alFormat, ptr, numberElements*sizeof(short), mixc->sampleRate); } - return 0; + alSourceQueueBuffers(mixc->alSource, 1, &buffer); + + return numberElements; } // fill buffer with zeros, returns number processed -static unsigned short FillAlBufferWithSilence(AudioContext_t *context, ALuint buffer) +static int 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 +376,42 @@ static void ResampleByteToFloat(char *chars, float *floats, unsigned short len) } } +// used to output raw audio streams, returns negative numbers on error +// if floating point is false the data size is 16bit short, otherwise it is float 32bit +RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint) +{ + int mixIndex; + for(mixIndex = 0; mixIndex < MAX_MIX_CHANNELS; mixIndex++) // find empty mix channel slot + { + if(mixChannelsActive_g[mixIndex] == NULL) break; + else if(mixIndex = MAX_MIX_CHANNELS - 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]); +} + +int BufferRawAudioContext(RawAudioContext ctx, void *data, int numberElements) +{ + int numBuffered = 0; + if(ctx >= 0) + { + MixChannel_t* mixc = mixChannelsActive_g[ctx]; + numBuffered = BufferMixChannel(mixc, data, numberElements); + } + return numBuffered; +} + + + //---------------------------------------------------------------------------------- @@ -767,205 +762,215 @@ void SetSoundPitch(Sound sound, float pitch) //---------------------------------------------------------------------------------- // Start music playing (open stream) -void PlayMusicStream(char *fileName) +// returns 0 on success +int PlayMusicStream(int musicIndex, char *fileName) { + int mixIndex; + + if(currentMusic[musicIndex].stream || currentMusic[musicIndex].chipctx) return 1; // error + + for(mixIndex = 0; mixIndex < MAX_MIX_CHANNELS; mixIndex++) // find empty mix channel slot + { + if(mixChannelsActive_g[mixIndex] == NULL) break; + else if(mixIndex = MAX_MIX_CHANNELS - 1) return 2; // error + } + if (strcmp(GetExtension(fileName),"ogg") == 0) { - // Stop current music, clean buffers, unload current stream - StopMusicStream(); - // Open audio stream - currentMusic.stream = stb_vorbis_open_filename(fileName, NULL, NULL); + currentMusic[musicIndex].stream = stb_vorbis_open_filename(fileName, NULL, NULL); - if (currentMusic.stream == NULL) + if (currentMusic[musicIndex].stream == NULL) { TraceLog(WARNING, "[%s] OGG audio file could not be opened", fileName); + return 3; // error } else { // Get file info - stb_vorbis_info info = stb_vorbis_get_info(currentMusic.stream); - - currentMusic.channels = info.channels; - currentMusic.sampleRate = info.sample_rate; + stb_vorbis_info info = stb_vorbis_get_info(currentMusic[musicIndex].stream); TraceLog(INFO, "[%s] Ogg sample rate: %i", fileName, info.sample_rate); TraceLog(INFO, "[%s] Ogg channels: %i", fileName, info.channels); TraceLog(DEBUG, "[%s] Temp memory required: %i", fileName, info.temp_memory_required); - if (info.channels == 2) currentMusic.format = AL_FORMAT_STEREO16; - else currentMusic.format = AL_FORMAT_MONO16; - - currentMusic.loop = true; // We loop by default - musicEnabled = true; - - // Create an audio source - alGenSources(1, ¤tMusic.source); // Generate pointer to audio source - - alSourcef(currentMusic.source, AL_PITCH, 1); - alSourcef(currentMusic.source, AL_GAIN, 1); - alSource3f(currentMusic.source, AL_POSITION, 0, 0, 0); - alSource3f(currentMusic.source, AL_VELOCITY, 0, 0, 0); - //alSourcei(currentMusic.source, AL_LOOPING, AL_TRUE); // ERROR: Buffers do not queue! - - // Generate two OpenAL buffers - alGenBuffers(2, currentMusic.buffers); - - // Fill buffers with music... - BufferMusicStream(currentMusic.buffers[0]); - BufferMusicStream(currentMusic.buffers[1]); - - // Queue buffers and start playing - alSourceQueueBuffers(currentMusic.source, 2, currentMusic.buffers); - alSourcePlay(currentMusic.source); - - // NOTE: Regularly, we must check if a buffer has been processed and refill it: UpdateMusicStream() + currentMusic[musicIndex].loop = true; // We loop by default + musicEnabled_g = true; + - currentMusic.totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic.stream) * currentMusic.channels; - currentMusic.totalLengthSeconds = stb_vorbis_stream_length_in_seconds(currentMusic.stream); + currentMusic[musicIndex].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[musicIndex].stream) * info.channels; + currentMusic[musicIndex].totalLengthSeconds = stb_vorbis_stream_length_in_seconds(currentMusic[musicIndex].stream); + + if (info.channels == 2){ + currentMusic[musicIndex].mixc = InitMixChannel(info.sample_rate, mixIndex, 2, false); + currentMusic[musicIndex].mixc->playing = true; + } + else{ + currentMusic[musicIndex].mixc = InitMixChannel(info.sample_rate, mixIndex, 1, false); + currentMusic[musicIndex].mixc->playing = true; + } + if(!currentMusic[musicIndex].mixc) return 4; // error } } else if (strcmp(GetExtension(fileName),"xm") == 0) { - // Stop current music, clean buffers, unload current stream - StopMusicStream(); - - // new song settings for xm chiptune - currentMusic.chipTune = true; - currentMusic.channels = 2; - currentMusic.sampleRate = 48000; - currentMusic.loop = true; - // only stereo is supported for xm - if(!jar_xm_create_context_from_file(¤tMusic.chipctx, currentMusic.sampleRate, fileName)) + if(!jar_xm_create_context_from_file(¤tMusic[musicIndex].chipctx, 48000, fileName)) { - currentMusic.format = AL_FORMAT_STEREO16; - jar_xm_set_max_loop_count(currentMusic.chipctx, 0); // infinite number of loops - currentMusic.totalSamplesLeft = jar_xm_get_remaining_samples(currentMusic.chipctx); - currentMusic.totalLengthSeconds = ((float)currentMusic.totalSamplesLeft) / ((float)currentMusic.sampleRate); - musicEnabled = true; + currentMusic[musicIndex].chipTune = true; + currentMusic[musicIndex].loop = true; + jar_xm_set_max_loop_count(currentMusic[musicIndex].chipctx, 0); // infinite number of loops + currentMusic[musicIndex].totalSamplesLeft = jar_xm_get_remaining_samples(currentMusic[musicIndex].chipctx); + currentMusic[musicIndex].totalLengthSeconds = ((float)currentMusic[musicIndex].totalSamplesLeft) / 48000.f; + musicEnabled_g = true; - TraceLog(INFO, "[%s] XM number of samples: %i", fileName, currentMusic.totalSamplesLeft); - TraceLog(INFO, "[%s] XM track length: %11.6f sec", fileName, currentMusic.totalLengthSeconds); + 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); - // Set up OpenAL - alGenSources(1, ¤tMusic.source); - alSourcef(currentMusic.source, AL_PITCH, 1); - alSourcef(currentMusic.source, AL_GAIN, 1); - alSource3f(currentMusic.source, AL_POSITION, 0, 0, 0); - alSource3f(currentMusic.source, AL_VELOCITY, 0, 0, 0); - alGenBuffers(2, currentMusic.buffers); - BufferMusicStream(currentMusic.buffers[0]); - BufferMusicStream(currentMusic.buffers[1]); - alSourceQueueBuffers(currentMusic.source, 2, currentMusic.buffers); - alSourcePlay(currentMusic.source); - - // NOTE: Regularly, we must check if a buffer has been processed and refill it: UpdateMusicStream() + currentMusic[musicIndex].mixc = InitMixChannel(48000, mixIndex, 2, false); + if(!currentMusic[musicIndex].mixc) return 5; // error + currentMusic[musicIndex].mixc->playing = true; } - else TraceLog(WARNING, "[%s] XM file could not be opened", fileName); + else + { + TraceLog(WARNING, "[%s] XM file could not be opened", fileName); + return 6; // error + } + } + else + { + TraceLog(WARNING, "[%s] Music extension not recognized, it can't be loaded", fileName); + return 7; // error } - else TraceLog(WARNING, "[%s] Music extension not recognized, it can't be loaded", fileName); + return 0; // normal return } -// Stop music playing (close stream) -void StopMusicStream(void) +// Stop music playing for individual music index of currentMusic array (close stream) +void StopMusicStream(int index) { - if (musicEnabled) + if (index < MAX_MUSIC_STREAMS && currentMusic[index].mixc) { - alSourceStop(currentMusic.source); - EmptyMusicStream(); // Empty music buffers - alDeleteSources(1, ¤tMusic.source); - alDeleteBuffers(2, currentMusic.buffers); + CloseMixChannel(currentMusic[index].mixc); - if (currentMusic.chipTune) + if (currentMusic[index].chipTune) { - jar_xm_free_context(currentMusic.chipctx); + jar_xm_free_context(currentMusic[index].chipctx); } else { - stb_vorbis_close(currentMusic.stream); + stb_vorbis_close(currentMusic[index].stream); + } + + if(!getMusicStreamCount()) musicEnabled_g = false; + if(currentMusic[index].stream || currentMusic[index].chipctx) + { + currentMusic[index].stream = NULL; + currentMusic[index].chipctx = NULL; } } +} - musicEnabled = false; +//get number of music channels active at this time, this does not mean they are playing +int getMusicStreamCount(void) +{ + int musicCount = 0; + for(int musicIndex = 0; musicIndex < MAX_MUSIC_STREAMS; musicIndex++) // find empty music slot + if(currentMusic[musicIndex].stream != NULL || currentMusic[musicIndex].chipTune) musicCount++; + + return musicCount; } // Pause music playing -void PauseMusicStream(void) +void PauseMusicStream(int index) { // Pause music stream if music available! - if (musicEnabled) + if (index < MAX_MUSIC_STREAMS && currentMusic[index].mixc && musicEnabled_g) { TraceLog(INFO, "Pausing music stream"); - alSourcePause(currentMusic.source); - musicEnabled = false; + alSourcePause(currentMusic[index].mixc->alSource); + currentMusic[index].mixc->playing = false; } } // Resume music playing -void ResumeMusicStream(void) +void ResumeMusicStream(int index) { // Resume music playing... if music available! ALenum state; - alGetSourcei(currentMusic.source, AL_SOURCE_STATE, &state); - - if (state == AL_PAUSED) - { - TraceLog(INFO, "Resuming music stream"); - alSourcePlay(currentMusic.source); - musicEnabled = true; + 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].mixc->alSource); + currentMusic[index].mixc->playing = true; + } } } -// Check if music is playing -bool IsMusicPlaying(void) +// Check if any music is playing +bool IsMusicPlaying(int index) { bool playing = false; ALint state; - - alGetSourcei(currentMusic.source, AL_SOURCE_STATE, &state); - if (state == AL_PLAYING) playing = true; + + if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc){ + alGetSourcei(currentMusic[index].mixc->alSource, AL_SOURCE_STATE, &state); + if (state == AL_PLAYING) playing = true; + } return playing; } // Set volume for music -void SetMusicVolume(float volume) +void SetMusicVolume(int index, float volume) { - alSourcef(currentMusic.source, 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].mixc){ + alSourcef(currentMusic[index].mixc->alSource, AL_PITCH, pitch); + } } // Get current music time length (in seconds) -float GetMusicTimeLength(void) +float GetMusicTimeLength(int index) { float totalSeconds; - if (currentMusic.chipTune) + if (currentMusic[index].chipTune) { - totalSeconds = currentMusic.totalLengthSeconds; + totalSeconds = currentMusic[index].totalLengthSeconds; } else { - totalSeconds = stb_vorbis_stream_length_in_seconds(currentMusic.stream); + totalSeconds = stb_vorbis_stream_length_in_seconds(currentMusic[index].stream); } return totalSeconds; } // Get current music time played (in seconds) -float GetMusicTimePlayed(void) +float GetMusicTimePlayed(int index) { float secondsPlayed; - if (currentMusic.chipTune) + if(index < MAX_MUSIC_STREAMS && currentMusic[index].mixc) { - uint64_t samples; - jar_xm_get_position(currentMusic.chipctx, NULL, NULL, NULL, &samples); - secondsPlayed = (float)samples / (currentMusic.sampleRate * currentMusic.channels); // Not sure if this is the correct value - } - else - { - int totalSamples = stb_vorbis_stream_length_in_samples(currentMusic.stream) * currentMusic.channels; - int samplesPlayed = totalSamples - currentMusic.totalSamplesLeft; - secondsPlayed = (float)samplesPlayed / (currentMusic.sampleRate * currentMusic.channels); + if (currentMusic[index].chipTune) + { + uint64_t samples; + jar_xm_get_position(currentMusic[index].chipctx, NULL, NULL, NULL, &samples); + 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].mixc->channels; + int samplesPlayed = totalSamples - currentMusic[index].totalSamplesLeft; + secondsPlayed = (float)samplesPlayed / (currentMusic[index].mixc->sampleRate * currentMusic[index].mixc->channels); + } } @@ -977,116 +982,118 @@ float GetMusicTimePlayed(void) //---------------------------------------------------------------------------------- // Fill music buffers with new data from music stream -static bool BufferMusicStream(ALuint buffer) +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 bytes) - int streamedBytes = 0; // samples of data obtained, channels are not included in calculation + 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 (musicEnabled) + + 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.chipTune) // There is no end of stream for xmfiles, once the end is reached zeros are generated for non looped chiptunes. - { - int readlen = MUSIC_BUFFER_SIZE_SHORT / 2; - jar_xm_generate_samples_16bit(currentMusic.chipctx, pcm, readlen); // reads 2*readlen shorts and moves them to buffer+size memory location - size += readlen * currentMusic.channels; // Not sure if this is what it needs - } + if(currentMusic[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT) + size = MUSIC_BUFFER_SIZE_SHORT / 2; else + size = currentMusic[index].totalSamplesLeft / 2; + + for(int x=0; x 0) size += (streamedBytes*currentMusic.channels); - else break; + active = false; + break; } } - TraceLog(DEBUG, "Streaming music data to buffer. Bytes streamed: %i", size); - } - - if (size > 0) - { - alBufferData(buffer, currentMusic.format, pcm, size*sizeof(short), currentMusic.sampleRate); - currentMusic.totalSamplesLeft -= size; - - if(currentMusic.totalSamplesLeft <= 0) active = false; // end if no more samples left } else { - active = false; - TraceLog(WARNING, "No more data obtained from stream"); + if(currentMusic[index].totalSamplesLeft >= MUSIC_BUFFER_SIZE_SHORT) + size = MUSIC_BUFFER_SIZE_SHORT; + else + size = currentMusic[index].totalSamplesLeft; + + 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; + } + } } return active; } // Empty music buffers -static void EmptyMusicStream(void) +static void EmptyMusicStream(int index) { ALuint buffer = 0; int queued = 0; - alGetSourcei(currentMusic.source, AL_BUFFERS_QUEUED, &queued); + alGetSourcei(currentMusic[index].mixc->alSource, AL_BUFFERS_QUEUED, &queued); while (queued > 0) { - alSourceUnqueueBuffers(currentMusic.source, 1, &buffer); + alSourceUnqueueBuffers(currentMusic[index].mixc->alSource, 1, &buffer); queued--; } } -// Update (re-fill) music buffers if data already processed -void UpdateMusicStream(void) +//determine if a music stream is ready to be written to +static int IsMusicStreamReadyForBuffering(int index) { - ALuint buffer = 0; ALint processed = 0; - bool active = true; + alGetSourcei(currentMusic[index].mixc->alSource, AL_BUFFERS_PROCESSED, &processed); + return processed; +} - if (musicEnabled) +// Update (re-fill) music buffers if data already processed +void UpdateMusicStream(int index) +{ + ALenum state; + bool active = true; + int numBuffers = IsMusicStreamReadyForBuffering(index); + + if (currentMusic[index].mixc->playing && index < MAX_MUSIC_STREAMS && musicEnabled_g && currentMusic[index].mixc && numBuffers) { - // Get the number of already processed buffers (if any) - alGetSourcei(currentMusic.source, AL_BUFFERS_PROCESSED, &processed); - - while (processed > 0) + active = BufferMusicStream(index, numBuffers); + + if (!active && currentMusic[index].loop) { - // Recover processed buffer for refill - alSourceUnqueueBuffers(currentMusic.source, 1, &buffer); - - // Refill buffer - active = BufferMusicStream(buffer); - - // If no more data to stream, restart music (if loop) - if ((!active) && (currentMusic.loop)) + if (currentMusic[index].chipTune) { - if(currentMusic.chipTune) - { - currentMusic.totalSamplesLeft = currentMusic.totalLengthSeconds * currentMusic.sampleRate; - } - else - { - stb_vorbis_seek_start(currentMusic.stream); - currentMusic.totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic.stream)*currentMusic.channels; - } - active = BufferMusicStream(buffer); + currentMusic[index].totalSamplesLeft = currentMusic[index].totalLengthSeconds * 48000; } - - // Add refilled buffer to queue again... don't let the music stop! - alSourceQueueBuffers(currentMusic.source, 1, &buffer); - - if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data..."); - - processed--; + else + { + stb_vorbis_seek_start(currentMusic[index].stream); + currentMusic[index].totalSamplesLeft = stb_vorbis_stream_length_in_samples(currentMusic[index].stream) * currentMusic[index].mixc->channels; + } + active = true; } + - ALenum state; - alGetSourcei(currentMusic.source, AL_SOURCE_STATE, &state); + if (alGetError() != AL_NO_ERROR) TraceLog(WARNING, "Error buffering data..."); + + alGetSourcei(currentMusic[index].mixc->alSource, AL_SOURCE_STATE, &state); - if ((state != AL_PLAYING) && active) alSourcePlay(currentMusic.source); + if (state != AL_PLAYING && active) alSourcePlay(currentMusic[index].mixc->alSource); - if (!active) StopMusicStream(); + if (!active) StopMusicStream(index); + } + else + return; + } // Load WAV file into Wave structure diff --git a/src/audio.h b/src/audio.h index afd881b7..1140a60a 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) @@ -100,15 +90,24 @@ bool IsSoundPlaying(Sound sound); // Check if a so void SetSoundVolume(Sound sound, float volume); // Set volume for a sound (1.0 is max level) void SetSoundPitch(Sound sound, float pitch); // Set pitch for a sound (1.0 is base level) -void PlayMusicStream(char *fileName); // Start music playing (open stream) -void UpdateMusicStream(void); // Updates buffers for music streaming -void StopMusicStream(void); // Stop music playing (close stream) -void PauseMusicStream(void); // Pause music playing -void ResumeMusicStream(void); // Resume playing paused music -bool IsMusicPlaying(void); // Check if music is playing -void SetMusicVolume(float volume); // Set volume for music (1.0 is max level) -float GetMusicTimeLength(void); // Get music time length (in seconds) -float GetMusicTimePlayed(void); // Get current music time played (in seconds) +int PlayMusicStream(int musicIndex, char *fileName); // Start music playing (open stream) +void UpdateMusicStream(int index); // Updates buffers for music streaming +void StopMusicStream(int index); // Stop music playing (close stream) +void PauseMusicStream(int index); // Pause music playing +void ResumeMusicStream(int index); // Resume playing paused music +bool IsMusicPlaying(int index); // Check if music is playing +void SetMusicVolume(int index, float volume); // Set volume for music (1.0 is max level) +float GetMusicTimeLength(int index); // Get music time length (in seconds) +float GetMusicTimePlayed(int index); // Get current music time played (in seconds) +int getMusicStreamCount(void); +void SetMusicPitch(int index, float pitch); + +// used to output raw audio streams, returns negative numbers on error +// if floating point is false the data size is 16bit short, otherwise it is float 32bit +RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint); + +void CloseRawAudioContext(RawAudioContext ctx); +int BufferRawAudioContext(RawAudioContext ctx, void *data, int numberElements); // returns number of elements buffered #ifdef __cplusplus } diff --git a/src/easings.h b/src/easings.h index e1e5465a..a8178f4a 100644 --- a/src/easings.h +++ b/src/easings.h @@ -18,11 +18,11 @@ * float speed = 1.f; * float currentTime = 0.f; * float currentPos[2] = {0,0}; -* float newPos[2] = {1,1}; -* float tempPosition[2] = currentPos;//x,y positions -* while(currentPos[0] < newPos[0]) -* currentPos[0] = EaseSineIn(currentTime, tempPosition[0], tempPosition[0]-newPos[0], speed); -* currentPos[1] = EaseSineIn(currentTime, tempPosition[1], tempPosition[1]-newPos[0], speed); +* float finalPos[2] = {1,1}; +* float startPosition[2] = currentPos;//x,y positions +* while(currentPos[0] < finalPos[0]) +* currentPos[0] = EaseSineIn(currentTime, startPosition[0], startPosition[0]-finalPos[0], speed); +* currentPos[1] = EaseSineIn(currentTime, startPosition[1], startPosition[1]-finalPos[0], speed); * currentTime += diffTime(); * * A port of Robert Penner's easing equations to C (http://robertpenner.com/easing/) diff --git a/src/raylib.h b/src/raylib.h index 911fd8b5..986dc7bf 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) @@ -894,15 +884,24 @@ bool IsSoundPlaying(Sound sound); // Check if a so void SetSoundVolume(Sound sound, float volume); // Set volume for a sound (1.0 is max level) void SetSoundPitch(Sound sound, float pitch); // Set pitch for a sound (1.0 is base level) -void PlayMusicStream(char *fileName); // Start music playing (open stream) -void UpdateMusicStream(void); // Updates buffers for music streaming -void StopMusicStream(void); // Stop music playing (close stream) -void PauseMusicStream(void); // Pause music playing -void ResumeMusicStream(void); // Resume playing paused music -bool IsMusicPlaying(void); // Check if music is playing -void SetMusicVolume(float volume); // Set volume for music (1.0 is max level) -float GetMusicTimeLength(void); // Get current music time length (in seconds) -float GetMusicTimePlayed(void); // Get current music time played (in seconds) +int PlayMusicStream(int musicIndex, char *fileName); // Start music playing (open stream) +void UpdateMusicStream(int index); // Updates buffers for music streaming +void StopMusicStream(int index); // Stop music playing (close stream) +void PauseMusicStream(int index); // Pause music playing +void ResumeMusicStream(int index); // Resume playing paused music +bool IsMusicPlaying(int index); // Check if music is playing +void SetMusicVolume(int index, float volume); // Set volume for music (1.0 is max level) +float GetMusicTimeLength(int index); // Get current music time length (in seconds) +float GetMusicTimePlayed(int index); // Get current music time played (in seconds) +int getMusicStreamCount(void); +void SetMusicPitch(int index, float pitch); + +// used to output raw audio streams, returns negative numbers on error +// if floating point is false the data size is 16bit short, otherwise it is float 32bit +RawAudioContext InitRawAudioContext(int sampleRate, int channels, bool floatingPoint); + +void CloseRawAudioContext(RawAudioContext ctx); +int BufferRawAudioContext(RawAudioContext ctx, void *data, int numberElements); // returns number of elements buffered #ifdef __cplusplus } diff --git a/src/windows_compile.bat b/src/windows_compile.bat new file mode 100644 index 00000000..f1d0fb29 --- /dev/null +++ b/src/windows_compile.bat @@ -0,0 +1,2 @@ +set PATH=C:\raylib\MinGW\bin;%PATH% +mingw32-make \ No newline at end of file