diff options
| author | Ray <[email protected]> | 2019-02-22 13:13:11 +0100 |
|---|---|---|
| committer | Ray <[email protected]> | 2019-02-22 13:13:11 +0100 |
| commit | 374811c440302701496bfb474ce5861c951c5884 (patch) | |
| tree | f81022aea0804a9b21fbcf44d702b376172de936 /src/raudio.c | |
| parent | 8382ab9ada2ecccd9a28702af193d242a8f16a6a (diff) | |
| download | raylib-374811c440302701496bfb474ce5861c951c5884.tar.gz raylib-374811c440302701496bfb474ce5861c951c5884.zip | |
Change ternary operator formatting
Diffstat (limited to 'src/raudio.c')
| -rw-r--r-- | src/raudio.c | 22 |
1 files changed, 11 insertions, 11 deletions
diff --git a/src/raudio.c b/src/raudio.c index 4f3e9220..451d3bc3 100644 --- a/src/raudio.c +++ b/src/raudio.c @@ -834,7 +834,7 @@ Sound LoadSoundFromWave(Wave wave) // // I have decided on the first option because it offloads work required for the format conversion to the to the loading stage. // The downside to this is that it uses more memory if the original sound is u8 or s16. - mal_format formatIn = ((wave.sampleSize == 8) ? mal_format_u8 : ((wave.sampleSize == 16) ? mal_format_s16 : mal_format_f32)); + mal_format formatIn = ((wave.sampleSize == 8)? mal_format_u8 : ((wave.sampleSize == 16)? mal_format_s16 : mal_format_f32)); mal_uint32 frameCountIn = wave.sampleCount/wave.channels; mal_uint32 frameCount = (mal_uint32)mal_convert_frames(NULL, DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, NULL, formatIn, wave.channels, wave.sampleRate, frameCountIn); @@ -946,7 +946,7 @@ void ExportWaveAsCode(Wave wave, const char *fileName) // Write byte data as hexadecimal text fprintf(txtFile, "static unsigned char %s_DATA[%i] = { ", varFileName, dataSize); - for (int i = 0; i < dataSize - 1; i++) fprintf(txtFile, ((i%BYTES_TEXT_PER_LINE == 0) ? "0x%x,\n" : "0x%x, "), ((unsigned char *)wave.data)[i]); + for (int i = 0; i < dataSize - 1; i++) fprintf(txtFile, ((i%BYTES_TEXT_PER_LINE == 0)? "0x%x,\n" : "0x%x, "), ((unsigned char *)wave.data)[i]); fprintf(txtFile, "0x%x };\n", ((unsigned char *)wave.data)[dataSize - 1]); fclose(txtFile); @@ -997,8 +997,8 @@ void SetSoundPitch(Sound sound, float pitch) // Convert wave data to desired format void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels) { - mal_format formatIn = ((wave->sampleSize == 8) ? mal_format_u8 : ((wave->sampleSize == 16) ? mal_format_s16 : mal_format_f32)); - mal_format formatOut = (( sampleSize == 8) ? mal_format_u8 : (( sampleSize == 16) ? mal_format_s16 : mal_format_f32)); + mal_format formatIn = ((wave->sampleSize == 8)? mal_format_u8 : ((wave->sampleSize == 16)? mal_format_s16 : mal_format_f32)); + mal_format formatOut = (( sampleSize == 8)? mal_format_u8 : (( sampleSize == 16)? mal_format_s16 : mal_format_f32)); mal_uint32 frameCountIn = wave->sampleCount; // Is wave->sampleCount actually the frame count? That terminology needs to change, if so. @@ -1511,7 +1511,7 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un stream.channels = 1; // Fallback to mono channel } - mal_format formatIn = ((stream.sampleSize == 8) ? mal_format_u8 : ((stream.sampleSize == 16) ? mal_format_s16 : mal_format_f32)); + mal_format formatIn = ((stream.sampleSize == 8)? mal_format_u8 : ((stream.sampleSize == 16)? mal_format_s16 : mal_format_f32)); // The size of a streaming buffer must be at least double the size of a period. unsigned int periodSize = device.bufferSizeInFrames/device.periods; @@ -1528,7 +1528,7 @@ AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, un audioBuffer->looping = true; // Always loop for streaming buffers. stream.audioBuffer = audioBuffer; - TraceLog(LOG_INFO, "[AUD ID %i] Audio stream loaded successfully (%i Hz, %i bit, %s)", stream.source, stream.sampleRate, stream.sampleSize, (stream.channels == 1) ? "Mono" : "Stereo"); + TraceLog(LOG_INFO, "[AUD ID %i] Audio stream loaded successfully (%i Hz, %i bit, %s)", stream.source, stream.sampleRate, stream.sampleSize, (stream.channels == 1)? "Mono" : "Stereo"); return stream; } @@ -1566,7 +1566,7 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount) else { // Just update whichever sub-buffer is processed. - subBufferToUpdate = (audioBuffer->isSubBufferProcessed[0]) ? 0 : 1; + subBufferToUpdate = (audioBuffer->isSubBufferProcessed[0])? 0 : 1; } mal_uint32 subBufferSizeInFrames = audioBuffer->bufferSizeInFrames/2; @@ -1769,7 +1769,7 @@ static Wave LoadWAV(const char *fileName) // NOTE: subChunkSize comes in bytes, we need to translate it to number of samples wave.sampleCount = (wavData.subChunkSize/(wave.sampleSize/8))/wave.channels; - TraceLog(LOG_INFO, "[%s] WAV file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo"); + TraceLog(LOG_INFO, "[%s] WAV file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo"); } } } @@ -1890,7 +1890,7 @@ static Wave LoadOGG(const char *fileName) TraceLog(LOG_DEBUG, "[%s] Samples obtained: %i", fileName, numSamplesOgg); - TraceLog(LOG_INFO, "[%s] OGG file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo"); + TraceLog(LOG_INFO, "[%s] OGG file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo"); stb_vorbis_close(oggFile); } @@ -1917,7 +1917,7 @@ static Wave LoadFLAC(const char *fileName) if (wave.channels > 2) TraceLog(LOG_WARNING, "[%s] FLAC channels number (%i) not supported", fileName, wave.channels); if (wave.data == NULL) TraceLog(LOG_WARNING, "[%s] FLAC data could not be loaded", fileName); - else TraceLog(LOG_INFO, "[%s] FLAC file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo"); + else TraceLog(LOG_INFO, "[%s] FLAC file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo"); return wave; } @@ -1944,7 +1944,7 @@ static Wave LoadMP3(const char *fileName) if (wave.channels > 2) TraceLog(LOG_WARNING, "[%s] MP3 channels number (%i) not supported", fileName, wave.channels); if (wave.data == NULL) TraceLog(LOG_WARNING, "[%s] MP3 data could not be loaded", fileName); - else TraceLog(LOG_INFO, "[%s] MP3 file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1) ? "Mono" : "Stereo"); + else TraceLog(LOG_INFO, "[%s] MP3 file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo"); return wave; } |
