diff --git a/src/raudio.c b/src/raudio.c index 6f7e9a72f..b3874dbe1 100644 --- a/src/raudio.c +++ b/src/raudio.c @@ -410,9 +410,18 @@ static AudioData AUDIO = { // Global AUDIO context // Module specific Functions Declaration //---------------------------------------------------------------------------------- static void OnLog(void *pUserData, ma_uint32 level, const char *pMessage); + +// Reads audio data from an AudioBuffer object in internal/device formats +static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer, void *framesOut, ma_uint32 frameCount); +static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, float *framesOut, ma_uint32 frameCount); + static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount); static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, AudioBuffer *buffer); +static bool IsAudioBufferPlayingInLockedState(AudioBuffer *buffer); +static void StopAudioBufferInLockedState(AudioBuffer *buffer); +static void UpdateAudioStreamInLockedState(AudioStream stream, const void *data, int frameCount); + #if defined(RAUDIO_STANDALONE) static bool IsFileExtension(const char *fileName, const char *ext); // Check file extension static const char *GetFileExtension(const char *fileName); // Get pointer to extension for a filename string (includes the dot: .png) @@ -431,10 +440,8 @@ static bool SaveFileText(const char *fileName, char *text); // Save text AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage); void UnloadAudioBuffer(AudioBuffer *buffer); -bool IsAudioBufferPlayingInLockedState(AudioBuffer *buffer); bool IsAudioBufferPlaying(AudioBuffer *buffer); void PlayAudioBuffer(AudioBuffer *buffer); -void StopAudioBufferInLockedState(AudioBuffer *buffer); void StopAudioBuffer(AudioBuffer *buffer); void PauseAudioBuffer(AudioBuffer *buffer); void ResumeAudioBuffer(AudioBuffer *buffer); @@ -444,10 +451,6 @@ void SetAudioBufferPan(AudioBuffer *buffer, float pan); void TrackAudioBuffer(AudioBuffer *buffer); void UntrackAudioBuffer(AudioBuffer *buffer); -//---------------------------------------------------------------------------------- -// AudioStream management functions declaration -//---------------------------------------------------------------------------------- -void UpdateAudioStreamInLockedState(AudioStream stream, const void *data, int frameCount); //---------------------------------------------------------------------------------- // Module Functions Definition - Audio Device initialization and Closing @@ -467,12 +470,12 @@ void InitAudioDevice(void) } // Init audio device - // NOTE: Using the default device. Format is floating point because it simplifies mixing. + // 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 AUDIO.System.device. + config.playback.pDeviceID = NULL; // NULL for the default playback AUDIO.System.device config.playback.format = AUDIO_DEVICE_FORMAT; config.playback.channels = AUDIO_DEVICE_CHANNELS; - config.capture.pDeviceID = NULL; // NULL for the default capture AUDIO.System.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 = AUDIO_DEVICE_SAMPLE_RATE; @@ -488,7 +491,7 @@ void InitAudioDevice(void) } // Mixing happens on a separate 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. + // want to look at something a bit smarter later on to keep everything real-time, if that's necessary if (ma_mutex_init(&AUDIO.System.lock) != MA_SUCCESS) { TRACELOG(LOG_WARNING, "AUDIO: Failed to create mutex for mixing"); @@ -498,7 +501,7 @@ void InitAudioDevice(void) } // 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. + // while there's at least one sound being played result = ma_device_start(&AUDIO.System.device); if (result != MA_SUCCESS) { @@ -626,17 +629,7 @@ void UnloadAudioBuffer(AudioBuffer *buffer) } } -// Check if an audio buffer is playing, assuming the audio system mutex has been locked. -bool IsAudioBufferPlayingInLockedState(AudioBuffer *buffer) -{ - bool result = false; - - if (buffer != NULL) result = (buffer->playing && !buffer->paused); - - return result; -} - -// Check if an audio buffer is playing from a program state without lock. +// Check if an audio buffer is playing from a program state without lock bool IsAudioBufferPlaying(AudioBuffer *buffer) { bool result = false; @@ -647,8 +640,8 @@ bool IsAudioBufferPlaying(AudioBuffer *buffer) } // Play an audio buffer -// NOTE: Buffer is restarted to the start. -// Use PauseAudioBuffer() and ResumeAudioBuffer() if the playback position should be maintained. +// NOTE: Buffer is restarted to the start +// Use PauseAudioBuffer() and ResumeAudioBuffer() if the playback position should be maintained void PlayAudioBuffer(AudioBuffer *buffer) { if (buffer != NULL) @@ -661,24 +654,7 @@ void PlayAudioBuffer(AudioBuffer *buffer) } } -// Stop an audio buffer, assuming the audio system mutex has been locked. -void StopAudioBufferInLockedState(AudioBuffer *buffer) -{ - if (buffer != NULL) - { - if (IsAudioBufferPlayingInLockedState(buffer)) - { - buffer->playing = false; - buffer->paused = false; - buffer->frameCursorPos = 0; - buffer->framesProcessed = 0; - buffer->isSubBufferProcessed[0] = true; - buffer->isSubBufferProcessed[1] = true; - } - } -} - -// Stop an audio buffer from a program state without lock. +// Stop an audio buffer from a program state without lock void StopAudioBuffer(AudioBuffer *buffer) { ma_mutex_lock(&AUDIO.System.lock); @@ -725,7 +701,7 @@ void SetAudioBufferPitch(AudioBuffer *buffer, float pitch) if ((buffer != NULL) && (pitch > 0.0f)) { ma_mutex_lock(&AUDIO.System.lock); - // Pitching is just an adjustment of the sample rate. + // Pitching is just an adjustment of the sample rate // Note that this changes the duration of the sound: // - higher pitches will make the sound faster // - lower pitches make it slower @@ -951,15 +927,15 @@ Sound LoadSoundFromWave(Wave wave) if (wave.data != NULL) { - // When using miniaudio we need to do our own mixing. + // 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 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. + // 1) Convert the whole sound in one go at load time (here) + // 2) Convert the audio data in chunks at mixing time // - // First option has been selected, format conversion is done on the loading stage. - // The downside is that it uses more memory if the original sound is u8 or s16. + // First option has been selected, format conversion is done on the loading stage + // The downside is that it uses more memory if the original sound is u8 or s16 ma_format formatIn = ((wave.sampleSize == 8)? ma_format_u8 : ((wave.sampleSize == 16)? ma_format_s16 : ma_format_f32)); ma_uint32 frameCountIn = wave.frameCount; @@ -2167,55 +2143,6 @@ void UpdateAudioStream(AudioStream stream, const void *data, int frameCount) ma_mutex_unlock(&AUDIO.System.lock); } -void UpdateAudioStreamInLockedState(AudioStream stream, const void *data, int frameCount) -{ - if (stream.buffer != NULL) - { - if (stream.buffer->isSubBufferProcessed[0] || stream.buffer->isSubBufferProcessed[1]) - { - ma_uint32 subBufferToUpdate = 0; - - if (stream.buffer->isSubBufferProcessed[0] && stream.buffer->isSubBufferProcessed[1]) - { - // Both buffers are available for updating. - // Update the first one and make sure the cursor is moved back to the front. - subBufferToUpdate = 0; - stream.buffer->frameCursorPos = 0; - } - else - { - // Just update whichever sub-buffer is processed. - subBufferToUpdate = (stream.buffer->isSubBufferProcessed[0])? 0 : 1; - } - - ma_uint32 subBufferSizeInFrames = stream.buffer->sizeInFrames/2; - unsigned char *subBuffer = stream.buffer->data + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate); - - // Total frames processed in buffer is always the complete size, filled with 0 if required - stream.buffer->framesProcessed += subBufferSizeInFrames; - - // Does this API expect a whole buffer to be updated in one go? - // Assuming so, but if not will need to change this logic. - if (subBufferSizeInFrames >= (ma_uint32)frameCount) - { - ma_uint32 framesToWrite = (ma_uint32)frameCount; - - ma_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8); - memcpy(subBuffer, data, bytesToWrite); - - // Any leftover frames should be filled with zeros. - ma_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite; - - if (leftoverFrameCount > 0) memset(subBuffer + bytesToWrite, 0, leftoverFrameCount*stream.channels*(stream.sampleSize/8)); - - stream.buffer->isSubBufferProcessed[subBufferToUpdate] = false; - } - else TRACELOG(LOG_WARNING, "STREAM: Attempting to write too many frames to buffer"); - } - else TRACELOG(LOG_WARNING, "STREAM: Buffer not available for updating"); - } -} - // Check if any audio stream buffers requires refill bool IsAudioStreamProcessed(AudioStream stream) { @@ -2246,7 +2173,7 @@ void ResumeAudioStream(AudioStream stream) ResumeAudioBuffer(stream.buffer); } -// Check if audio stream is playing. +// Check if audio stream is playing bool IsAudioStreamPlaying(AudioStream stream) { return IsAudioBufferPlaying(stream.buffer); @@ -2293,9 +2220,9 @@ void SetAudioStreamCallback(AudioStream stream, AudioCallback callback) } } -// Add processor to audio stream. Contrary to buffers, the order of processors is important. +// Add processor to audio stream. Contrary to buffers, the order of processors is important // The new processor must be added at the end. As there aren't supposed to be a lot of processors attached to -// a given stream, we iterate through the list to find the end. That way we don't need a pointer to the last element. +// a given stream, we iterate through the list to find the end. That way we don't need a pointer to the last element void AttachAudioStreamProcessor(AudioStream stream, AudioCallback process) { ma_mutex_lock(&AUDIO.System.lock); @@ -2410,7 +2337,7 @@ static void OnLog(void *pUserData, ma_uint32 level, const char *pMessage) TRACELOG(LOG_WARNING, "miniaudio: %s", pMessage); // All log messages from miniaudio are errors } -// Reads audio data from an AudioBuffer object in internal format. +// Reads audio data from an AudioBuffer object in internal format static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer, void *framesOut, ma_uint32 frameCount) { // Using audio buffer callback @@ -2498,20 +2425,20 @@ static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer, // For static buffers we can fill the remaining frames with silence for safety, but we don't want // to report those frames as "read". The reason for this is that the caller uses the return value - // to know whether a non-looping sound has finished playback. + // to know whether a non-looping sound has finished playback if (audioBuffer->usage != AUDIO_BUFFER_USAGE_STATIC) framesRead += totalFramesRemaining; } return framesRead; } -// Reads audio data from an AudioBuffer object in device format. Returned data will be in a format appropriate for mixing. +// Reads audio data from an AudioBuffer object in device format, returned data will be in a format appropriate for mixing static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, float *framesOut, ma_uint32 frameCount) { // What's going on here is that we're continuously converting data from the AudioBuffer's internal format to the mixing format, which // should be defined by the output format of the data converter. We do this until frameCount frames have been output. The important // detail to remember here is that we never, ever attempt to read more input data than is required for the specified number of output - // frames. This can be achieved with ma_data_converter_get_required_input_frame_count(). + // frames. This can be achieved with ma_data_converter_get_required_input_frame_count() ma_uint8 inputBuffer[4096] = { 0 }; ma_uint32 inputBufferFrameCap = sizeof(inputBuffer)/ma_get_bytes_per_frame(audioBuffer->converter.formatIn, audioBuffer->converter.channelsIn); @@ -2693,6 +2620,83 @@ static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 fr } } +// Check if an audio buffer is playing, assuming the audio system mutex has been locked +static bool IsAudioBufferPlayingInLockedState(AudioBuffer *buffer) +{ + bool result = false; + + if (buffer != NULL) result = (buffer->playing && !buffer->paused); + + return result; +} + +// Stop an audio buffer, assuming the audio system mutex has been locked +static void StopAudioBufferInLockedState(AudioBuffer *buffer) +{ + if (buffer != NULL) + { + if (IsAudioBufferPlayingInLockedState(buffer)) + { + buffer->playing = false; + buffer->paused = false; + buffer->frameCursorPos = 0; + buffer->framesProcessed = 0; + buffer->isSubBufferProcessed[0] = true; + buffer->isSubBufferProcessed[1] = true; + } + } +} + +// Update audio stream, assuming the audio system mutex has been locked +static void UpdateAudioStreamInLockedState(AudioStream stream, const void *data, int frameCount) +{ + if (stream.buffer != NULL) + { + if (stream.buffer->isSubBufferProcessed[0] || stream.buffer->isSubBufferProcessed[1]) + { + ma_uint32 subBufferToUpdate = 0; + + if (stream.buffer->isSubBufferProcessed[0] && stream.buffer->isSubBufferProcessed[1]) + { + // Both buffers are available for updating + // Update the first one and make sure the cursor is moved back to the front + subBufferToUpdate = 0; + stream.buffer->frameCursorPos = 0; + } + else + { + // Just update whichever sub-buffer is processed + subBufferToUpdate = (stream.buffer->isSubBufferProcessed[0])? 0 : 1; + } + + ma_uint32 subBufferSizeInFrames = stream.buffer->sizeInFrames/2; + unsigned char *subBuffer = stream.buffer->data + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate); + + // Total frames processed in buffer is always the complete size, filled with 0 if required + stream.buffer->framesProcessed += subBufferSizeInFrames; + + // Does this API expect a whole buffer to be updated in one go? + // Assuming so, but if not will need to change this logic + if (subBufferSizeInFrames >= (ma_uint32)frameCount) + { + ma_uint32 framesToWrite = (ma_uint32)frameCount; + + ma_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8); + memcpy(subBuffer, data, bytesToWrite); + + // Any leftover frames should be filled with zeros + ma_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite; + + if (leftoverFrameCount > 0) memset(subBuffer + bytesToWrite, 0, leftoverFrameCount*stream.channels*(stream.sampleSize/8)); + + stream.buffer->isSubBufferProcessed[subBufferToUpdate] = false; + } + else TRACELOG(LOG_WARNING, "STREAM: Attempting to write too many frames to buffer"); + } + else TRACELOG(LOG_WARNING, "STREAM: Buffer not available for updating"); + } +} + // Some required functions for audio standalone module version #if defined(RAUDIO_STANDALONE) // Check file extension