From b8cd9480877ba0bcf78190f88f05478f2c1be2e4 Mon Sep 17 00:00:00 2001 From: Ray Date: Wed, 29 Jan 2020 23:58:37 +0100 Subject: [PATCH] [raudio] REDESIGNED: Implement global context --- src/raudio.c | 231 +++++++++++++++++++++++++++------------------------ 1 file changed, 122 insertions(+), 109 deletions(-) diff --git a/src/raudio.c b/src/raudio.c index ed2eef181..f86f249a4 100644 --- a/src/raudio.c +++ b/src/raudio.c @@ -4,11 +4,11 @@ * * FEATURES: * - Manage audio device (init/close) +* - Manage raw audio context +* - Manage mixing channels * - Load and unload audio files * - Format wave data (sample rate, size, channels) * - Play/Stop/Pause/Resume loaded audio -* - Manage mixing channels -* - Manage raw audio context * * CONFIGURATION: * @@ -124,7 +124,9 @@ // After some math, considering a sampleRate of 48000, a buffer refill rate of 1/60 seconds and a // standard double-buffering system, a 4096 samples buffer has been chosen, it should be enough // In case of music-stalls, just increase this number -#define AUDIO_BUFFER_SIZE 4096 // PCM data samples (i.e. 16bit, Mono: 8Kb) +#if !defined(AUDIO_BUFFER_SIZE) + #define AUDIO_BUFFER_SIZE 4096 // PCM data samples (i.e. 16bit, Mono: 8Kb) +#endif //---------------------------------------------------------------------------------- // Types and Structures Definition @@ -191,11 +193,11 @@ void TraceLog(int msgType, const char *text, ...); // Show trace lo #define MAX_AUDIO_BUFFER_POOL_CHANNELS 16 +// NOTE: Different logic is used when feeding data to the playback device +// depending on whether or not data is streamed (Music vs Sound) typedef enum { AUDIO_BUFFER_USAGE_STATIC = 0, AUDIO_BUFFER_USAGE_STREAM } AudioBufferUsage; // Audio buffer structure -// NOTE: Slightly different logic is used when feeding data to the -// playback device depending on whether or not data is streamed struct rAudioBuffer { ma_pcm_converter dsp; // PCM data converter @@ -208,33 +210,42 @@ struct rAudioBuffer { int usage; // Audio buffer usage mode: STATIC or STREAM bool isSubBufferProcessed[2]; // SubBuffer processed (virtual double buffer) + unsigned int sizeInFrames; // Total buffer size in frames unsigned int frameCursorPos; // Frame cursor position - unsigned int bufferSizeInFrames; // Total buffer size in frames unsigned int totalFramesProcessed; // Total frames processed in this buffer (required for play timming) - unsigned char *buffer; // Data buffer, on music stream keeps filling + unsigned char *data; // Data buffer, on music stream keeps filling rAudioBuffer *next; // Next audio buffer on the list rAudioBuffer *prev; // Previous audio buffer on the list }; -#define AudioBuffer rAudioBuffer // HACK: To avoid CoreAudio (macOS) symbol collision +#define AudioBuffer rAudioBuffer // HACK: To avoid CoreAudio (macOS) symbol collision -// Audio buffers are tracked in a linked list -static AudioBuffer *firstAudioBuffer = NULL; // Pointer to first AudioBuffer in the list -static AudioBuffer *lastAudioBuffer = NULL; // Pointer to last AudioBuffer in the list +// Audio data context +typedef struct AudioData { + struct { + ma_context context; // miniaudio context data + ma_device device; // miniaudio device + ma_mutex lock; // miniaudio mutex lock + bool isReady; // Check if audio device is ready + float masterVolume; // Master volume (multiplied on output mixing) + } System; + struct { + AudioBuffer *first; // Pointer to first AudioBuffer in the list + AudioBuffer *last; // Pointer to last AudioBuffer in the list + } Buffer; + struct { + AudioBuffer *pool[MAX_AUDIO_BUFFER_POOL_CHANNELS]; // Multichannel AudioBuffer pointers pool + unsigned int poolCounter; // AudioBuffer pointers pool counter + unsigned int channels[MAX_AUDIO_BUFFER_POOL_CHANNELS]; // AudioBuffer pool channels + } MultiChannel; +} AudioData; -// miniaudio global variables -static ma_context context; // miniaudio context data -static ma_device device; // miniaudio device -static ma_mutex audioLock; // miniaudio mutex lock -static bool isAudioInitialized = false; // Check if audio device is initialized -static float masterVolume = 1.0f; // Master volume (multiplied on output mixing) - -// Multi channel playback global variables -static AudioBuffer *audioBufferPool[MAX_AUDIO_BUFFER_POOL_CHANNELS] = { 0 }; // Multichannel AudioBuffer pointers pool -static unsigned int audioBufferPoolCounter = 0; // AudioBuffer pointers pool counter -static unsigned int audioBufferPoolChannels[MAX_AUDIO_BUFFER_POOL_CHANNELS] = { 0 }; // AudioBuffer pool channels +//---------------------------------------------------------------------------------- +// Global Variables Definition +//---------------------------------------------------------------------------------- +static AudioData AUDIO = { 0 }; // miniaudio functions declaration static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message); @@ -244,7 +255,7 @@ static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 fr // AudioBuffer management functions declaration // NOTE: Those functions are not exposed by raylib... for the moment -AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 bufferSizeInFrames, int usage); +AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage); void CloseAudioBuffer(AudioBuffer *buffer); bool IsAudioBufferPlaying(AudioBuffer *buffer); void PlayAudioBuffer(AudioBuffer *buffer); @@ -281,9 +292,9 @@ static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const // Using a mutex here for thread-safety which makes things not real-time // This is unlikely to be necessary for this project, but may want to consider how you might want to avoid this - ma_mutex_lock(&audioLock); + ma_mutex_lock(&AUDIO.System.lock); { - for (AudioBuffer *audioBuffer = firstAudioBuffer; audioBuffer != NULL; audioBuffer = audioBuffer->next) + for (AudioBuffer *audioBuffer = AUDIO.Buffer.first; audioBuffer != NULL; audioBuffer = audioBuffer->next) { // Ignore stopped or paused sounds if (!audioBuffer->playing || audioBuffer->paused) continue; @@ -316,7 +327,7 @@ static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const ma_uint32 framesJustRead = (ma_uint32)ma_pcm_converter_read(&audioBuffer->dsp, tempBuffer, framesToReadRightNow); if (framesJustRead > 0) { - float *framesOut = (float *)pFramesOut + (framesRead*device.playback.channels); + float *framesOut = (float *)pFramesOut + (framesRead*AUDIO.System.device.playback.channels); float *framesIn = tempBuffer; MixAudioFrames(framesOut, framesIn, framesJustRead, audioBuffer->volume); @@ -356,7 +367,7 @@ static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const } } - ma_mutex_unlock(&audioLock); + ma_mutex_unlock(&AUDIO.System.lock); } // DSP read from audio buffer callback function @@ -364,7 +375,7 @@ static ma_uint32 OnAudioBufferDSPRead(ma_pcm_converter *pDSP, void *pFramesOut, { AudioBuffer *audioBuffer = (AudioBuffer *)pUserData; - ma_uint32 subBufferSizeInFrames = (audioBuffer->bufferSizeInFrames > 1)? audioBuffer->bufferSizeInFrames/2 : audioBuffer->bufferSizeInFrames; + ma_uint32 subBufferSizeInFrames = (audioBuffer->sizeInFrames > 1)? audioBuffer->sizeInFrames/2 : audioBuffer->sizeInFrames; ma_uint32 currentSubBufferIndex = audioBuffer->frameCursorPos/subBufferSizeInFrames; if (currentSubBufferIndex > 1) @@ -404,7 +415,7 @@ static ma_uint32 OnAudioBufferDSPRead(ma_pcm_converter *pDSP, void *pFramesOut, ma_uint32 framesRemainingInOutputBuffer; if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC) { - framesRemainingInOutputBuffer = audioBuffer->bufferSizeInFrames - audioBuffer->frameCursorPos; + framesRemainingInOutputBuffer = audioBuffer->sizeInFrames - audioBuffer->frameCursorPos; } else { @@ -415,8 +426,8 @@ static ma_uint32 OnAudioBufferDSPRead(ma_pcm_converter *pDSP, void *pFramesOut, ma_uint32 framesToRead = totalFramesRemaining; if (framesToRead > framesRemainingInOutputBuffer) framesToRead = framesRemainingInOutputBuffer; - memcpy((unsigned char *)pFramesOut + (framesRead*frameSizeInBytes), audioBuffer->buffer + (audioBuffer->frameCursorPos*frameSizeInBytes), framesToRead*frameSizeInBytes); - audioBuffer->frameCursorPos = (audioBuffer->frameCursorPos + framesToRead)%audioBuffer->bufferSizeInFrames; + memcpy((unsigned char *)pFramesOut + (framesRead*frameSizeInBytes), audioBuffer->data + (audioBuffer->frameCursorPos*frameSizeInBytes), framesToRead*frameSizeInBytes); + audioBuffer->frameCursorPos = (audioBuffer->frameCursorPos + framesToRead)%audioBuffer->sizeInFrames; framesRead += framesToRead; // If we've read to the end of the buffer, mark it as processed @@ -457,12 +468,12 @@ static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 fr { for (ma_uint32 iFrame = 0; iFrame < frameCount; ++iFrame) { - for (ma_uint32 iChannel = 0; iChannel < device.playback.channels; ++iChannel) + for (ma_uint32 iChannel = 0; iChannel < AUDIO.System.device.playback.channels; ++iChannel) { - float *frameOut = framesOut + (iFrame*device.playback.channels); - const float *frameIn = framesIn + (iFrame*device.playback.channels); + float *frameOut = framesOut + (iFrame*AUDIO.System.device.playback.channels); + const float *frameIn = framesIn + (iFrame*AUDIO.System.device.playback.channels); - frameOut[iChannel] += (frameIn[iChannel]*masterVolume*localVolume); + frameOut[iChannel] += (frameIn[iChannel]*AUDIO.System.masterVolume*localVolume); } } } @@ -473,7 +484,7 @@ static void InitAudioBufferPool() // Dummy buffers for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) { - audioBufferPool[i] = InitAudioBuffer(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, 0, AUDIO_BUFFER_USAGE_STATIC); + AUDIO.MultiChannel.pool[i] = InitAudioBuffer(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, 0, AUDIO_BUFFER_USAGE_STATIC); } } @@ -482,8 +493,8 @@ static void CloseAudioBufferPool() { for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) { - RL_FREE(audioBufferPool[i]->buffer); - RL_FREE(audioBufferPool[i]); + RL_FREE(AUDIO.MultiChannel.pool[i]->data); + RL_FREE(AUDIO.MultiChannel.pool[i]); } } @@ -493,11 +504,14 @@ static void CloseAudioBufferPool() // Initialize audio device void InitAudioDevice(void) { + // TODO: Load AUDIO context memory dynamically? + AUDIO.System.masterVolume = 1.0f; + // Init audio context - ma_context_config contextConfig = ma_context_config_init(); - contextConfig.logCallback = OnLog; + ma_context_config ctxConfig = ma_context_config_init(); + ctxConfig.logCallback = OnLog; - ma_result result = ma_context_init(NULL, 0, &contextConfig, &context); + ma_result result = ma_context_init(NULL, 0, &ctxConfig, &AUDIO.System.context); if (result != MA_SUCCESS) { TraceLog(LOG_ERROR, "Failed to initialize audio context"); @@ -507,78 +521,78 @@ void InitAudioDevice(void) // Init audio device // NOTE: Using the default device. Format is floating point because it simplifies mixing. ma_device_config config = ma_device_config_init(ma_device_type_playback); - config.playback.pDeviceID = NULL; // NULL for the default playback device. + config.playback.pDeviceID = NULL; // NULL for the default playback AUDIO.System.device. config.playback.format = DEVICE_FORMAT; config.playback.channels = DEVICE_CHANNELS; - config.capture.pDeviceID = NULL; // NULL for the default capture device. + config.capture.pDeviceID = NULL; // NULL for the default capture AUDIO.System.device. config.capture.format = ma_format_s16; config.capture.channels = 1; config.sampleRate = DEVICE_SAMPLE_RATE; config.dataCallback = OnSendAudioDataToDevice; config.pUserData = NULL; - result = ma_device_init(&context, &config, &device); + result = ma_device_init(&AUDIO.System.context, &config, &AUDIO.System.device); if (result != MA_SUCCESS) { - TraceLog(LOG_ERROR, "Failed to initialize audio playback device"); - ma_context_uninit(&context); + TraceLog(LOG_ERROR, "Failed to initialize audio playback AUDIO.System.device"); + ma_context_uninit(&AUDIO.System.context); return; } // Keep the device running the whole time. May want to consider doing something a bit smarter and only have the device running // while there's at least one sound being played. - result = ma_device_start(&device); + result = ma_device_start(&AUDIO.System.device); if (result != MA_SUCCESS) { - TraceLog(LOG_ERROR, "Failed to start audio playback device"); - ma_device_uninit(&device); - ma_context_uninit(&context); + TraceLog(LOG_ERROR, "Failed to start audio playback AUDIO.System.device"); + ma_device_uninit(&AUDIO.System.device); + ma_context_uninit(&AUDIO.System.context); return; } // Mixing happens on a seperate thread which means we need to synchronize. I'm using a mutex here to make things simple, but may // want to look at something a bit smarter later on to keep everything real-time, if that's necessary. - if (ma_mutex_init(&context, &audioLock) != MA_SUCCESS) + if (ma_mutex_init(&AUDIO.System.context, &AUDIO.System.lock) != MA_SUCCESS) { TraceLog(LOG_ERROR, "Failed to create mutex for audio mixing"); - ma_device_uninit(&device); - ma_context_uninit(&context); + ma_device_uninit(&AUDIO.System.device); + ma_context_uninit(&AUDIO.System.context); return; } TraceLog(LOG_INFO, "Audio device initialized successfully"); - TraceLog(LOG_INFO, "Audio backend: miniaudio / %s", ma_get_backend_name(context.backend)); - TraceLog(LOG_INFO, "Audio format: %s -> %s", ma_get_format_name(device.playback.format), ma_get_format_name(device.playback.internalFormat)); - TraceLog(LOG_INFO, "Audio channels: %d -> %d", device.playback.channels, device.playback.internalChannels); - TraceLog(LOG_INFO, "Audio sample rate: %d -> %d", device.sampleRate, device.playback.internalSampleRate); - TraceLog(LOG_INFO, "Audio buffer size: %d", device.playback.internalBufferSizeInFrames); + TraceLog(LOG_INFO, "Audio backend: miniaudio / %s", ma_get_backend_name(AUDIO.System.context.backend)); + TraceLog(LOG_INFO, "Audio format: %s -> %s", ma_get_format_name(AUDIO.System.device.playback.format), ma_get_format_name(AUDIO.System.device.playback.internalFormat)); + TraceLog(LOG_INFO, "Audio channels: %d -> %d", AUDIO.System.device.playback.channels, AUDIO.System.device.playback.internalChannels); + TraceLog(LOG_INFO, "Audio sample rate: %d -> %d", AUDIO.System.device.sampleRate, AUDIO.System.device.playback.internalSampleRate); + TraceLog(LOG_INFO, "Audio buffer size: %d", AUDIO.System.device.playback.internalBufferSizeInFrames); InitAudioBufferPool(); TraceLog(LOG_INFO, "Audio multichannel pool size: %i", MAX_AUDIO_BUFFER_POOL_CHANNELS); - isAudioInitialized = true; + AUDIO.System.isReady = true; } // Close the audio device for all contexts void CloseAudioDevice(void) { - if (isAudioInitialized) + if (AUDIO.System.isReady) { - ma_mutex_uninit(&audioLock); - ma_device_uninit(&device); - ma_context_uninit(&context); + ma_mutex_uninit(&AUDIO.System.lock); + ma_device_uninit(&AUDIO.System.device); + ma_context_uninit(&AUDIO.System.context); CloseAudioBufferPool(); - TraceLog(LOG_INFO, "Audio device closed successfully"); + TraceLog(LOG_INFO, "Audio AUDIO.System.device closed successfully"); } - else TraceLog(LOG_WARNING, "Could not close audio device because it is not currently initialized"); + else TraceLog(LOG_WARNING, "Could not close audio AUDIO.System.device because it is not currently initialized"); } // Check if device has been initialized successfully bool IsAudioDeviceReady(void) { - return isAudioInitialized; + return AUDIO.System.isReady; } // Set master volume (listener) @@ -587,7 +601,7 @@ void SetMasterVolume(float volume) if (volume < 0.0f) volume = 0.0f; else if (volume > 1.0f) volume = 1.0f; - masterVolume = volume; + AUDIO.System.masterVolume = volume; } //---------------------------------------------------------------------------------- @@ -595,7 +609,7 @@ void SetMasterVolume(float volume) //---------------------------------------------------------------------------------- // Initialize a new audio buffer (filled with silence) -AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 bufferSizeInFrames, int usage) +AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage) { AudioBuffer *audioBuffer = (AudioBuffer *)RL_CALLOC(1, sizeof(AudioBuffer)); @@ -605,7 +619,7 @@ AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sam return NULL; } - audioBuffer->buffer = RL_CALLOC(bufferSizeInFrames*channels*ma_get_bytes_per_sample(format), 1); + audioBuffer->data = RL_CALLOC(sizeInFrames*channels*ma_get_bytes_per_sample(format), 1); // Audio data runs through a format converter ma_pcm_converter_config dspConfig; @@ -637,7 +651,7 @@ AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sam audioBuffer->looping = false; audioBuffer->usage = usage; audioBuffer->frameCursorPos = 0; - audioBuffer->bufferSizeInFrames = bufferSizeInFrames; + audioBuffer->sizeInFrames = sizeInFrames; // Buffers should be marked as processed by default so that a call to // UpdateAudioStream() immediately after initialization works correctly @@ -656,7 +670,7 @@ void CloseAudioBuffer(AudioBuffer *buffer) if (buffer != NULL) { UntrackAudioBuffer(buffer); - RL_FREE(buffer->buffer); + RL_FREE(buffer->data); RL_FREE(buffer); } else TraceLog(LOG_ERROR, "CloseAudioBuffer() : No audio buffer"); @@ -748,35 +762,35 @@ void SetAudioBufferPitch(AudioBuffer *buffer, float pitch) // Track audio buffer to linked list next position void TrackAudioBuffer(AudioBuffer *buffer) { - ma_mutex_lock(&audioLock); + ma_mutex_lock(&AUDIO.System.lock); { - if (firstAudioBuffer == NULL) firstAudioBuffer = buffer; + if (AUDIO.Buffer.first == NULL) AUDIO.Buffer.first = buffer; else { - lastAudioBuffer->next = buffer; - buffer->prev = lastAudioBuffer; + AUDIO.Buffer.last->next = buffer; + buffer->prev = AUDIO.Buffer.last; } - lastAudioBuffer = buffer; + AUDIO.Buffer.last = buffer; } - ma_mutex_unlock(&audioLock); + ma_mutex_unlock(&AUDIO.System.lock); } // Untrack audio buffer from linked list void UntrackAudioBuffer(AudioBuffer *buffer) { - ma_mutex_lock(&audioLock); + ma_mutex_lock(&AUDIO.System.lock); { - if (buffer->prev == NULL) firstAudioBuffer = buffer->next; + if (buffer->prev == NULL) AUDIO.Buffer.first = buffer->next; else buffer->prev->next = buffer->next; - if (buffer->next == NULL) lastAudioBuffer = buffer->prev; + if (buffer->next == NULL) AUDIO.Buffer.last = buffer->prev; else buffer->next->prev = buffer->prev; buffer->prev = NULL; buffer->next = NULL; } - ma_mutex_unlock(&audioLock); + ma_mutex_unlock(&AUDIO.System.lock); } //---------------------------------------------------------------------------------- @@ -829,7 +843,7 @@ Sound LoadSoundFromWave(Wave wave) { // When using miniaudio we need to do our own mixing. // To simplify this we need convert the format of each sound to be consistent with - // the format used to open the playback device. We can do this two ways: + // the format used to open the playback AUDIO.System.device. We can do this two ways: // // 1) Convert the whole sound in one go at load time (here). // 2) Convert the audio data in chunks at mixing time. @@ -845,7 +859,7 @@ Sound LoadSoundFromWave(Wave wave) AudioBuffer *audioBuffer = InitAudioBuffer(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, frameCount, AUDIO_BUFFER_USAGE_STATIC); if (audioBuffer == NULL) TraceLog(LOG_WARNING, "LoadSoundFromWave() : Failed to create audio buffer"); - frameCount = (ma_uint32)ma_convert_frames(audioBuffer->buffer, audioBuffer->dsp.formatConverterIn.config.formatIn, audioBuffer->dsp.formatConverterIn.config.channels, audioBuffer->dsp.src.config.sampleRateIn, wave.data, formatIn, wave.channels, wave.sampleRate, frameCountIn); + frameCount = (ma_uint32)ma_convert_frames(audioBuffer->data, audioBuffer->dsp.formatConverterIn.config.formatIn, audioBuffer->dsp.formatConverterIn.config.channels, audioBuffer->dsp.src.config.sampleRateIn, wave.data, formatIn, wave.channels, wave.sampleRate, frameCountIn); if (frameCount == 0) TraceLog(LOG_WARNING, "LoadSoundFromWave() : Format conversion failed"); sound.sampleCount = frameCount*DEVICE_CHANNELS; @@ -884,7 +898,7 @@ void UpdateSound(Sound sound, const void *data, int samplesCount) StopAudioBuffer(audioBuffer); // TODO: May want to lock/unlock this since this data buffer is read at mixing time - memcpy(audioBuffer->buffer, data, samplesCount*audioBuffer->dsp.formatConverterIn.config.channels*ma_get_bytes_per_sample(audioBuffer->dsp.formatConverterIn.config.formatIn)); + memcpy(audioBuffer->data, data, samplesCount*audioBuffer->dsp.formatConverterIn.config.channels*ma_get_bytes_per_sample(audioBuffer->dsp.formatConverterIn.config.formatIn)); } else TraceLog(LOG_ERROR, "UpdateSound() : Invalid sound - no audio buffer"); } @@ -973,13 +987,13 @@ void PlaySoundMulti(Sound sound) // find the first non playing pool entry for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) { - if (audioBufferPoolChannels[i] > oldAge) + if (AUDIO.MultiChannel.channels[i] > oldAge) { - oldAge = audioBufferPoolChannels[i]; + oldAge = AUDIO.MultiChannel.channels[i]; oldIndex = i; } - if (!IsAudioBufferPlaying(audioBufferPool[i])) + if (!IsAudioBufferPlaying(AUDIO.MultiChannel.pool[i])) { index = i; break; @@ -989,7 +1003,7 @@ void PlaySoundMulti(Sound sound) // If no none playing pool members can be index choose the oldest if (index == -1) { - TraceLog(LOG_WARNING,"pool age %i ended a sound early no room in buffer pool", audioBufferPoolCounter); + TraceLog(LOG_WARNING,"pool age %i ended a sound early no room in buffer pool", AUDIO.MultiChannel.poolCounter); if (oldIndex == -1) { @@ -1002,32 +1016,32 @@ void PlaySoundMulti(Sound sound) index = oldIndex; // Just in case... - StopAudioBuffer(audioBufferPool[index]); + StopAudioBuffer(AUDIO.MultiChannel.pool[index]); } // Experimentally mutex lock doesn't seem to be needed this makes sense - // as audioBufferPool[index] isn't playing and the only stuff we're copying + // as AUDIO.MultiChannel.pool[index] isn't playing and the only stuff we're copying // shouldn't be changing... - audioBufferPoolChannels[index] = audioBufferPoolCounter; - audioBufferPoolCounter++; + AUDIO.MultiChannel.channels[index] = AUDIO.MultiChannel.poolCounter; + AUDIO.MultiChannel.poolCounter++; - audioBufferPool[index]->volume = sound.stream.buffer->volume; - audioBufferPool[index]->pitch = sound.stream.buffer->pitch; - audioBufferPool[index]->looping = sound.stream.buffer->looping; - audioBufferPool[index]->usage = sound.stream.buffer->usage; - audioBufferPool[index]->isSubBufferProcessed[0] = false; - audioBufferPool[index]->isSubBufferProcessed[1] = false; - audioBufferPool[index]->bufferSizeInFrames = sound.stream.buffer->bufferSizeInFrames; - audioBufferPool[index]->buffer = sound.stream.buffer->buffer; + AUDIO.MultiChannel.pool[index]->volume = sound.stream.buffer->volume; + AUDIO.MultiChannel.pool[index]->pitch = sound.stream.buffer->pitch; + AUDIO.MultiChannel.pool[index]->looping = sound.stream.buffer->looping; + AUDIO.MultiChannel.pool[index]->usage = sound.stream.buffer->usage; + AUDIO.MultiChannel.pool[index]->isSubBufferProcessed[0] = false; + AUDIO.MultiChannel.pool[index]->isSubBufferProcessed[1] = false; + AUDIO.MultiChannel.pool[index]->sizeInFrames = sound.stream.buffer->sizeInFrames; + AUDIO.MultiChannel.pool[index]->data = sound.stream.buffer->data; - PlayAudioBuffer(audioBufferPool[index]); + PlayAudioBuffer(AUDIO.MultiChannel.pool[index]); } // Stop any sound played with PlaySoundMulti() void StopSoundMulti(void) { - for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) StopAudioBuffer(audioBufferPool[i]); + for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) StopAudioBuffer(AUDIO.MultiChannel.pool[i]); } // Get number of sounds playing in the multichannel buffer pool @@ -1037,7 +1051,7 @@ int GetSoundsPlaying(void) for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) { - if (IsAudioBufferPlaying(audioBufferPool[i])) counter++; + if (IsAudioBufferPlaying(AUDIO.MultiChannel.pool[i])) counter++; } return counter; @@ -1243,7 +1257,7 @@ Music LoadMusicStream(const char *fileName) int result = jar_xm_create_context_from_file(&ctxXm, 48000, fileName); - if (result == 0) // XM context created successfully + if (result == 0) // XM AUDIO.System.context created successfully { music.ctxType = MUSIC_MODULE_XM; jar_xm_set_max_loop_count(ctxXm, 0); // Set infinite number of loops @@ -1374,7 +1388,6 @@ void StopMusicStream(Music music) { StopAudioStream(music.stream); - // Restart music context switch (music.ctxType) { #if defined(SUPPORT_FILEFORMAT_OGG) @@ -1401,7 +1414,7 @@ void UpdateMusicStream(Music music) { bool streamEnding = false; - unsigned int subBufferSizeInFrames = music.stream.buffer->bufferSizeInFrames/2; + unsigned int subBufferSizeInFrames = music.stream.buffer->sizeInFrames/2; // NOTE: Using dynamic allocation because it could require more than 16KB void *pcm = RL_CALLOC(subBufferSizeInFrames*music.stream.channels*music.stream.sampleSize/8, 1); @@ -1559,7 +1572,7 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un ma_format formatIn = ((stream.sampleSize == 8)? ma_format_u8 : ((stream.sampleSize == 16)? ma_format_s16 : ma_format_f32)); // The size of a streaming buffer must be at least double the size of a period - unsigned int periodSize = device.playback.internalBufferSizeInFrames/device.playback.internalPeriods; + unsigned int periodSize = AUDIO.System.device.playback.internalBufferSizeInFrames/AUDIO.System.device.playback.internalPeriods; unsigned int subBufferSize = AUDIO_BUFFER_SIZE; if (subBufferSize < periodSize) subBufferSize = periodSize; @@ -1610,8 +1623,8 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount) subBufferToUpdate = (audioBuffer->isSubBufferProcessed[0])? 0 : 1; } - ma_uint32 subBufferSizeInFrames = audioBuffer->bufferSizeInFrames/2; - unsigned char *subBuffer = audioBuffer->buffer + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate); + ma_uint32 subBufferSizeInFrames = audioBuffer->sizeInFrames/2; + unsigned char *subBuffer = audioBuffer->data + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate); // TODO: Get total frames processed on this buffer... DOES NOT WORK. audioBuffer->totalFramesProcessed += subBufferSizeInFrames;