ADDED: QOA music streaming (with auxiliar lib)
Some format tweaks
This commit is contained in:
parent
d3f5bc4043
commit
68ee0bb8dd
2 changed files with 357 additions and 78 deletions
157
src/raudio.c
157
src/raudio.c
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@ -230,6 +230,7 @@ typedef struct tagBITMAPINFOHEADER {
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#define QOA_IMPLEMENTATION
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#include "external/qoa.h" // QOA loading and saving functions
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#include "external/qoaplay.c" // QOA stream playing helper functions
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#endif
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#if defined(SUPPORT_FILEFORMAT_FLAC)
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@ -287,21 +288,6 @@ typedef struct tagBITMAPINFOHEADER {
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//----------------------------------------------------------------------------------
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// Types and Structures Definition
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//----------------------------------------------------------------------------------
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// Music context type
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// NOTE: Depends on data structure provided by the library
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// in charge of reading the different file types
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typedef enum {
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MUSIC_AUDIO_NONE = 0, // No audio context loaded
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MUSIC_AUDIO_WAV, // WAV audio context
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MUSIC_AUDIO_OGG, // OGG audio context
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MUSIC_AUDIO_FLAC, // FLAC audio context
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MUSIC_AUDIO_MP3, // MP3 audio context
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MUSIC_AUDIO_QOA, // QOA audio context
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MUSIC_MODULE_XM, // XM module audio context
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MUSIC_MODULE_MOD // MOD module audio context
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} MusicContextType;
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#if defined(RAUDIO_STANDALONE)
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// Trace log level
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// NOTE: Organized by priority level
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@ -317,6 +303,20 @@ typedef enum {
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} TraceLogLevel;
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#endif
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// Music context type
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// NOTE: Depends on data structure provided by the library
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// in charge of reading the different file types
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typedef enum {
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MUSIC_AUDIO_NONE = 0, // No audio context loaded
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MUSIC_AUDIO_WAV, // WAV audio context
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MUSIC_AUDIO_OGG, // OGG audio context
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MUSIC_AUDIO_FLAC, // FLAC audio context
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MUSIC_AUDIO_MP3, // MP3 audio context
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MUSIC_AUDIO_QOA, // QOA audio context
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MUSIC_MODULE_XM, // XM module audio context
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MUSIC_MODULE_MOD // MOD module audio context
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} MusicContextType;
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// NOTE: Different logic is used when feeding data to the playback device
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// depending on whether data is streamed (Music vs Sound)
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typedef enum {
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@ -1322,7 +1322,7 @@ void UnloadWaveSamples(float *samples)
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}
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//----------------------------------------------------------------------------------
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// Module Functions Definition - Music loading and stream playing (.OGG)
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// Module Functions Definition - Music loading and stream playing
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//----------------------------------------------------------------------------------
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// Load music stream from file
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@ -1395,21 +1395,16 @@ Music LoadMusicStream(const char *fileName)
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#if defined(SUPPORT_FILEFORMAT_QOA)
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else if (IsFileExtension(fileName, ".qoa"))
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{
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qoa_desc *ctxQoa = RL_CALLOC(1, sizeof(qoa_desc));
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// TODO: QOA stream support: Init context from file
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int result = 0;
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qoaplay_desc *ctxQoa = qoaplay_open(fileName);
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music.ctxType = MUSIC_AUDIO_QOA;
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music.ctxData = ctxQoa;
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if (result > 0)
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if (ctxQoa->file != NULL)
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{
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music.stream = LoadAudioStream(ctxQoa->samplerate, 16, ctxQoa->channels);
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// TODO: Read next frame(s) from QOA stream
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//music.frameCount = qoa_decode_frame(const unsigned char *bytes, unsigned int size, ctxQoa, short *sample_data, unsigned int *frame_len);
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// NOTE: We are loading samples are 32bit float normalized data, so,
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// we configure the output audio stream to also use float 32bit
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music.stream = LoadAudioStream(ctxQoa->info.samplerate, 32, ctxQoa->info.channels);
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music.frameCount = ctxQoa->info.samples;
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music.looping = true; // Looping enabled by default
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musicLoaded = true;
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}
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@ -1594,21 +1589,16 @@ Music LoadMusicStreamFromMemory(const char *fileType, const unsigned char *data,
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#if defined(SUPPORT_FILEFORMAT_QOA)
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else if (strcmp(fileType, ".qoa") == 0)
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{
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qoa_desc *ctxQoa = RL_CALLOC(1, sizeof(qoa_desc));
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// TODO: Init QOA context data
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int result = 0;
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qoaplay_desc *ctxQoa = qoaplay_open_memory(data, dataSize);
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music.ctxType = MUSIC_AUDIO_QOA;
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music.ctxData = ctxQoa;
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if (result > 0)
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if ((ctxQoa->file_data != NULL) && (ctxQoa->file_data_size != 0))
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{
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music.stream = LoadAudioStream(ctxQoa->samplerate, 16, ctxQoa->channels);
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// TODO: Read next frame(s) from QOA stream
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//music.frameCount = qoa_decode_frame(const unsigned char *bytes, unsigned int size, ctxQoa, short *sample_data, unsigned int *frame_len);
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// NOTE: We are loading samples are 32bit float normalized data, so,
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// we configure the output audio stream to also use float 32bit
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music.stream = LoadAudioStream(ctxQoa->info.samplerate, 32, ctxQoa->info.channels);
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music.frameCount = ctxQoa->info.samples;
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music.looping = true; // Looping enabled by default
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musicLoaded = true;
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}
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@ -1697,27 +1687,27 @@ Music LoadMusicStreamFromMemory(const char *fileType, const unsigned char *data,
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if (!musicLoaded)
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{
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if (false) { }
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#if defined(SUPPORT_FILEFORMAT_WAV)
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#if defined(SUPPORT_FILEFORMAT_WAV)
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else if (music.ctxType == MUSIC_AUDIO_WAV) drwav_uninit((drwav *)music.ctxData);
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#endif
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#if defined(SUPPORT_FILEFORMAT_OGG)
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#endif
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#if defined(SUPPORT_FILEFORMAT_OGG)
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else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData);
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#endif
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#if defined(SUPPORT_FILEFORMAT_MP3)
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#endif
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#if defined(SUPPORT_FILEFORMAT_MP3)
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else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); }
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#endif
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#if defined(SUPPORT_FILEFORMAT_QOA)
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else if (music.ctxType == MUSIC_AUDIO_QOA) { /*TODO: Release QOA context*/ RL_FREE(music.ctxData); }
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#endif
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#if defined(SUPPORT_FILEFORMAT_FLAC)
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#endif
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#if defined(SUPPORT_FILEFORMAT_QOA)
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else if (music.ctxType == MUSIC_AUDIO_QOA) qoaplay_close((qoaplay_desc *)music.ctxData);
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#endif
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#if defined(SUPPORT_FILEFORMAT_FLAC)
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else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL);
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#endif
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#if defined(SUPPORT_FILEFORMAT_XM)
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#endif
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#if defined(SUPPORT_FILEFORMAT_XM)
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else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData);
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#endif
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#if defined(SUPPORT_FILEFORMAT_MOD)
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#endif
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#if defined(SUPPORT_FILEFORMAT_MOD)
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else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); }
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#endif
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#endif
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music.ctxData = NULL;
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TRACELOG(LOG_WARNING, "FILEIO: Music data could not be loaded");
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@ -1763,7 +1753,7 @@ void UnloadMusicStream(Music music)
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else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); }
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#endif
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#if defined(SUPPORT_FILEFORMAT_QOA)
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else if (music.ctxType == MUSIC_AUDIO_QOA) { /*TODO: Release QOA context*/ RL_FREE(music.ctxData); }
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else if (music.ctxType == MUSIC_AUDIO_QOA) qoaplay_close((qoaplay_desc *)music.ctxData);
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#endif
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#if defined(SUPPORT_FILEFORMAT_FLAC)
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else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL);
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@ -1821,7 +1811,7 @@ void StopMusicStream(Music music)
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case MUSIC_AUDIO_MP3: drmp3_seek_to_start_of_stream((drmp3 *)music.ctxData); break;
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#endif
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#if defined(SUPPORT_FILEFORMAT_QOA)
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case MUSIC_AUDIO_QOA: /*TODO: Restart QOA context to beginning*/ break;
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case MUSIC_AUDIO_QOA: qoaplay_rewind((qoaplay_desc *)music.ctxData); break;
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#endif
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#if defined(SUPPORT_FILEFORMAT_FLAC)
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case MUSIC_AUDIO_FLAC: drflac__seek_to_first_frame((drflac *)music.ctxData); break;
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@ -1856,7 +1846,7 @@ void SeekMusicStream(Music music, float position)
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case MUSIC_AUDIO_MP3: drmp3_seek_to_pcm_frame((drmp3 *)music.ctxData, positionInFrames); break;
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#endif
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#if defined(SUPPORT_FILEFORMAT_QOA)
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case MUSIC_AUDIO_QOA: /*TODO: Seek to specific QOA frame*/ break;
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case MUSIC_AUDIO_QOA: qoaplay_seek_frame((qoaplay_desc *)music.ctxData, positionInFrames); break;
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#endif
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#if defined(SUPPORT_FILEFORMAT_FLAC)
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case MUSIC_AUDIO_FLAC: drflac_seek_to_pcm_frame((drflac *)music.ctxData, positionInFrames); break;
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@ -1892,11 +1882,13 @@ void UpdateMusicStream(Music music)
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unsigned int framesLeft = music.frameCount - music.stream.buffer->framesProcessed; // Frames left to be processed
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unsigned int framesToStream = 0; // Total frames to be streamed
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if ((framesLeft >= subBufferSizeInFrames) || music.looping) framesToStream = subBufferSizeInFrames;
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else framesToStream = framesLeft;
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int frameCountStillNeeded = framesToStream;
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int frameCountRedTotal = 0;
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int frameCountReadTotal = 0;
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switch (music.ctxType)
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{
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#if defined(SUPPORT_FILEFORMAT_WAV)
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@ -1906,8 +1898,8 @@ void UpdateMusicStream(Music music)
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{
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while (true)
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{
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int frameCountRed = (int)drwav_read_pcm_frames_s16((drwav *)music.ctxData, frameCountStillNeeded, (short *)((char *)AUDIO.System.pcmBuffer + frameCountRedTotal*frameSize));
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frameCountRedTotal += frameCountRed;
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int frameCountRed = (int)drwav_read_pcm_frames_s16((drwav *)music.ctxData, frameCountStillNeeded, (short *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize));
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frameCountReadTotal += frameCountRed;
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frameCountStillNeeded -= frameCountRed;
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if (frameCountStillNeeded == 0) break;
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else drwav_seek_to_first_pcm_frame((drwav *)music.ctxData);
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@ -1917,8 +1909,8 @@ void UpdateMusicStream(Music music)
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{
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while (true)
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{
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int frameCountRed = (int)drwav_read_pcm_frames_f32((drwav *)music.ctxData, frameCountStillNeeded, (float *)((char *)AUDIO.System.pcmBuffer + frameCountRedTotal*frameSize));
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frameCountRedTotal += frameCountRed;
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int frameCountRed = (int)drwav_read_pcm_frames_f32((drwav *)music.ctxData, frameCountStillNeeded, (float *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize));
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frameCountReadTotal += frameCountRed;
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frameCountStillNeeded -= frameCountRed;
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if (frameCountStillNeeded == 0) break;
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else drwav_seek_to_first_pcm_frame((drwav *)music.ctxData);
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@ -1931,8 +1923,8 @@ void UpdateMusicStream(Music music)
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{
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while (true)
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{
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int frameCountRed = stb_vorbis_get_samples_short_interleaved((stb_vorbis *)music.ctxData, music.stream.channels, (short *)((char *)AUDIO.System.pcmBuffer + frameCountRedTotal*frameSize), frameCountStillNeeded*music.stream.channels);
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frameCountRedTotal += frameCountRed;
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int frameCountRed = stb_vorbis_get_samples_short_interleaved((stb_vorbis *)music.ctxData, music.stream.channels, (short *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize), frameCountStillNeeded*music.stream.channels);
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frameCountReadTotal += frameCountRed;
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frameCountStillNeeded -= frameCountRed;
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if (frameCountStillNeeded == 0) break;
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else stb_vorbis_seek_start((stb_vorbis *)music.ctxData);
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@ -1944,9 +1936,9 @@ void UpdateMusicStream(Music music)
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{
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while (true)
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{
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int frameCountRed = (int)drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, frameCountStillNeeded, (float *)((char *)AUDIO.System.pcmBuffer + frameCountRedTotal*frameSize));
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frameCountRedTotal += frameCountRed;
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frameCountStillNeeded -= frameCountRed;
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int frameCountRead = (int)drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, frameCountStillNeeded, (float *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize));
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frameCountReadTotal += frameCountRead;
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frameCountStillNeeded -= frameCountRead;
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if (frameCountStillNeeded == 0) break;
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else drmp3_seek_to_start_of_stream((drmp3 *)music.ctxData);
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}
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@ -1955,7 +1947,18 @@ void UpdateMusicStream(Music music)
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#if defined(SUPPORT_FILEFORMAT_QOA)
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case MUSIC_AUDIO_QOA:
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{
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// TODO: Read QOA required framecount to fill buffer to keep music playing
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unsigned int frameCountRead = qoaplay_decode((qoaplay_desc *)music.ctxData, (float *)AUDIO.System.pcmBuffer, framesToStream);
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frameCountReadTotal += frameCountRead;
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/*
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while (true)
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{
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int frameCountRead = (int)qoaplay_decode((qoaplay_desc *)music.ctxData, (float *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize), frameCountStillNeeded);
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frameCountReadTotal += frameCountRead;
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frameCountStillNeeded -= frameCountRead;
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if (frameCountStillNeeded == 0) break;
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else qoaplay_rewind((qoaplay_desc *)music.ctxData);
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}
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*/
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} break;
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#endif
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#if defined(SUPPORT_FILEFORMAT_FLAC)
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@ -1964,8 +1967,8 @@ void UpdateMusicStream(Music music)
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while (true)
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{
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int frameCountRed = drflac_read_pcm_frames_s16((drflac *)music.ctxData, frameCountStillNeeded, (short *)((char *)AUDIO.System.pcmBuffer + frameCountRedTotal*frameSize));
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frameCountRedTotal += frameCountRed;
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frameCountStillNeeded -= frameCountRed;
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frameCountReadTotal += frameCountRead;
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frameCountStillNeeded -= frameCountRead;
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if (frameCountStillNeeded == 0) break;
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else drflac__seek_to_first_frame((drflac *)music.ctxData);
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}
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@ -1978,7 +1981,6 @@ void UpdateMusicStream(Music music)
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if (AUDIO_DEVICE_FORMAT == ma_format_f32) jar_xm_generate_samples((jar_xm_context_t *)music.ctxData, (float *)AUDIO.System.pcmBuffer, framesToStream);
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else if (AUDIO_DEVICE_FORMAT == ma_format_s16) jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)AUDIO.System.pcmBuffer, framesToStream);
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else if (AUDIO_DEVICE_FORMAT == ma_format_u8) jar_xm_generate_samples_8bit((jar_xm_context_t *)music.ctxData, (char *)AUDIO.System.pcmBuffer, framesToStream);
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//jar_xm_reset((jar_xm_context_t *)music.ctxData);
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} break;
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@ -1988,7 +1990,6 @@ void UpdateMusicStream(Music music)
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{
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// NOTE: 3rd parameter (nbsample) specify the number of stereo 16bits samples you want, so sampleCount/2
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jar_mod_fillbuffer((jar_mod_context_t *)music.ctxData, (short *)AUDIO.System.pcmBuffer, framesToStream, 0);
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//jar_mod_seek_start((jar_mod_context_t *)music.ctxData);
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} break;
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@ -2056,7 +2057,7 @@ float GetMusicTimePlayed(Music music)
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float secondsPlayed = 0.0f;
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if (music.stream.buffer != NULL)
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{
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#if defined(SUPPORT_FILEFORMAT_XM)
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#if defined(SUPPORT_FILEFORMAT_XM)
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if (music.ctxType == MUSIC_MODULE_XM)
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{
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uint64_t framesPlayed = 0;
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@ -2065,7 +2066,7 @@ float GetMusicTimePlayed(Music music)
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secondsPlayed = (float)framesPlayed/music.stream.sampleRate;
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}
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else
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#endif
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#endif
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{
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//ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
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int framesProcessed = (int)music.stream.buffer->framesProcessed;
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@ -2115,7 +2116,7 @@ AudioStream LoadAudioStream(unsigned int sampleRate, unsigned int sampleSize, un
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}
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// Checks if an audio stream is ready
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RLAPI bool IsAudioStreamReady(AudioStream stream)
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bool IsAudioStreamReady(AudioStream stream)
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{
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return ((stream.buffer != NULL) && // Validate stream buffer
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(stream.sampleRate > 0) && // Validate sample rate is supported
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@ -2277,6 +2278,7 @@ void AttachAudioStreamProcessor(AudioStream stream, AudioCallback process)
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ma_mutex_unlock(&AUDIO.System.lock);
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}
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// Remove processor from audio stream
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void DetachAudioStreamProcessor(AudioStream stream, AudioCallback process)
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{
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ma_mutex_lock(&AUDIO.System.lock);
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@ -2304,9 +2306,8 @@ void DetachAudioStreamProcessor(AudioStream stream, AudioCallback process)
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}
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// Add processor to audio pipeline. Order of processors is important
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// Works the same way as {Attach,Detach}AudioStreamProcessor functions, except
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// these two work on the already mixed output just before sending it to the
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// sound hardware.
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// Works the same way as {Attach,Detach}AudioStreamProcessor() functions, except
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// these two work on the already mixed output just before sending it to the sound hardware
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void AttachAudioMixedProcessor(AudioCallback process)
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{
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ma_mutex_lock(&AUDIO.System.lock);
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@ -2330,6 +2331,7 @@ void AttachAudioMixedProcessor(AudioCallback process)
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ma_mutex_unlock(&AUDIO.System.lock);
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}
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// Remove processor from audio pipeline
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void DetachAudioMixedProcessor(AudioCallback process)
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{
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ma_mutex_lock(&AUDIO.System.lock);
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@ -2508,7 +2510,6 @@ static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, f
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return totalOutputFramesProcessed;
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}
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// Sending audio data to device callback function
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// This function will be called when miniaudio needs more data
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// NOTE: All the mixing takes place here
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