diff options
| author | Ray <[email protected]> | 2021-03-22 20:41:33 +0100 |
|---|---|---|
| committer | Ray <[email protected]> | 2021-03-22 20:41:33 +0100 |
| commit | 24dae29a0344823e3a8306f0685664f1580a7751 (patch) | |
| tree | e360864d1ab8cf228623238d42e52d48900173e3 /src/raudio.c | |
| parent | 2c0a5339482cda97690917d4922f5b672e67c231 (diff) | |
| download | raylib-24dae29a0344823e3a8306f0685664f1580a7751.tar.gz raylib-24dae29a0344823e3a8306f0685664f1580a7751.zip | |
Review latest PR and some formatting
Diffstat (limited to 'src/raudio.c')
| -rw-r--r-- | src/raudio.c | 27 |
1 files changed, 14 insertions, 13 deletions
diff --git a/src/raudio.c b/src/raudio.c index 75cd6960..a15ca71e 100644 --- a/src/raudio.c +++ b/src/raudio.c @@ -1313,7 +1313,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char* data, int d else if (TextIsEqual(fileExtLower, ".wav")) { drwav *ctxWav = RL_CALLOC(1, sizeof(drwav)); - + bool success = drwav_init_memory(ctxWav, (const void*)data, dataSize, NULL); music.ctxType = MUSIC_AUDIO_WAV; @@ -1419,7 +1419,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char* data, int d // copy data to allocated memory for default UnloadMusicStream unsigned char *newData = RL_MALLOC(dataSize); - int it = dataSize / sizeof(unsigned char); + int it = dataSize/sizeof(unsigned char); for (int i = 0; i < it; i++){ newData[i] = data[i]; } @@ -1437,7 +1437,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char* data, int d music.ctxType = MUSIC_MODULE_MOD; // NOTE: Only stereo is supported for MOD - music.stream = InitAudioStream(AUDIO.System.device.sampleRate, 16, 2); + music.stream = InitAudioStream(AUDIO.System.device.sampleRate, 16, 2); music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod)*2; // 2 channels music.looping = true; // Looping enabled by default musicLoaded = true; @@ -1464,7 +1464,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char* data, int d #if defined(SUPPORT_FILEFORMAT_OGG) else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData); #endif - #if defined(SUPPORT_FILEFORMAT_XM) + #if defined(SUPPORT_FILEFORMAT_XM) else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData); #endif #if defined(SUPPORT_FILEFORMAT_MOD) @@ -1657,8 +1657,8 @@ void UpdateMusicStream(Music music) if ((music.ctxType == MUSIC_MODULE_XM) || music.ctxType == MUSIC_MODULE_MOD) { - if (samplesCount > 1) sampleLeft -= samplesCount / 2; - else sampleLeft -= samplesCount; + if (samplesCount > 1) sampleLeft -= samplesCount/2; + else sampleLeft -= samplesCount; } else sampleLeft -= samplesCount; @@ -1718,14 +1718,14 @@ float GetMusicTimeLength(Music music) float GetMusicTimePlayed(Music music) { #if defined(SUPPORT_FILEFORMAT_XM) - if (music.ctxType == MUSIC_MODULE_XM) - { + if (music.ctxType == MUSIC_MODULE_XM) + { uint64_t samples = 0; jar_xm_get_position(music.ctxData, NULL, NULL, NULL, &samples); samples = samples % (music.sampleCount); - return (float)(samples) / (music.stream.sampleRate * music.stream.channels); - } + return (float)(samples)/(music.stream.sampleRate*music.stream.channels); + } #endif float secondsPlayed = 0.0f; if (music.stream.buffer != NULL) @@ -1890,10 +1890,11 @@ int GetAudioStreamBufferSizeDefault() { // if the buffer is not set, compute one that would give us a buffer good enough for a decent frame rate if (AUDIO.Buffer.defaultSize == 0) - AUDIO.Buffer.defaultSize = AUDIO.System.device.sampleRate / 30; + AUDIO.Buffer.defaultSize = AUDIO.System.device.sampleRate/30; - return AUDIO.Buffer.defaultSize; + return AUDIO.Buffer.defaultSize; } + //---------------------------------------------------------------------------------- // Module specific Functions Definition //---------------------------------------------------------------------------------- @@ -2014,7 +2015,7 @@ static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, f inputFramesToProcessThisIteration = inputBufferFrameCap; } - float *runningFramesOut = framesOut + (totalOutputFramesProcessed * audioBuffer->converter.config.channelsOut); + float *runningFramesOut = framesOut + (totalOutputFramesProcessed*audioBuffer->converter.config.channelsOut); /* At this point we can convert the data to our mixing format. */ ma_uint64 inputFramesProcessedThisIteration = ReadAudioBufferFramesInInternalFormat(audioBuffer, inputBuffer, (ma_uint32)inputFramesToProcessThisIteration); /* Safe cast. */ |
