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author | J. Alexander Treuman <jat@spatialrift.net> | 2007-05-22 15:02:25 +0000 |
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committer | J. Alexander Treuman <jat@spatialrift.net> | 2007-05-22 15:02:25 +0000 |
commit | f2850a66f03accdaad8e36f590e2f1cb5bfe0036 (patch) | |
tree | a1d477a46df37d723d9a4ef3246d2a46b7167efd | |
parent | 92ff09881a5fa92c25bb768a7c58e341e0964f2f (diff) | |
download | mpd-f2850a66f03accdaad8e36f590e2f1cb5bfe0036.tar.gz mpd-f2850a66f03accdaad8e36f590e2f1cb5bfe0036.tar.xz mpd-f2850a66f03accdaad8e36f590e2f1cb5bfe0036.zip |
Reindenting
git-svn-id: https://svn.musicpd.org/mpd/trunk@6199 09075e82-0dd4-0310-85a5-a0d7c8717e4f
-rw-r--r-- | src/pcm_utils.c | 22 |
1 files changed, 12 insertions, 10 deletions
diff --git a/src/pcm_utils.c b/src/pcm_utils.c index 54003c0ab..20825d4e8 100644 --- a/src/pcm_utils.c +++ b/src/pcm_utils.c @@ -32,7 +32,7 @@ #endif void pcm_volumeChange(char *buffer, int bufferSize, AudioFormat * format, - int volume) + int volume) { mpd_sint32 temp32; mpd_sint8 *buffer8 = (mpd_sint8 *) buffer; @@ -83,8 +83,8 @@ void pcm_volumeChange(char *buffer, int bufferSize, AudioFormat * format, } static void pcm_add(char *buffer1, char *buffer2, size_t bufferSize1, - size_t bufferSize2, int vol1, int vol2, - AudioFormat * format) + size_t bufferSize2, int vol1, int vol2, + AudioFormat * format) { mpd_sint32 temp32; mpd_sint8 *buffer8_1 = (mpd_sint8 *) buffer1; @@ -139,7 +139,7 @@ static void pcm_add(char *buffer1, char *buffer2, size_t bufferSize1, } void pcm_mix(char *buffer1, char *buffer2, size_t bufferSize1, - size_t bufferSize2, AudioFormat * format, float portion1) + size_t bufferSize2, AudioFormat * format, float portion1) { int vol1; float s = sin(M_PI_2 * portion1); @@ -153,7 +153,8 @@ void pcm_mix(char *buffer1, char *buffer2, size_t bufferSize1, } #ifdef HAVE_LIBSAMPLERATE -static int pcm_getSamplerateConverter(void) { +static int pcm_getSamplerateConverter(void) +{ const char *conf, *test; int convalgo = SRC_SINC_FASTEST; int newalgo; @@ -185,8 +186,9 @@ static int pcm_getSamplerateConverter(void) { #endif /* outFormat bits must be 16 and channels must be 2! */ -void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer, size_t - inSize, AudioFormat * outFormat, char *outBuffer) +void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer, + size_t inSize, AudioFormat * outFormat, + char *outBuffer) { static char *bitConvBuffer; static int bitConvBufferLength; @@ -376,8 +378,8 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer, size_t } size_t pcm_sizeOfOutputBufferForAudioFormatConversion(AudioFormat * inFormat, - size_t inSize, - AudioFormat * outFormat) + size_t inSize, + AudioFormat * outFormat) { const int shift = sizeof(mpd_sint16) * outFormat->channels; size_t outSize = inSize; @@ -404,7 +406,7 @@ size_t pcm_sizeOfOutputBufferForAudioFormatConversion(AudioFormat * inFormat, } } - outSize /= shift; + outSize /= shift; outSize = floor(0.5 + (double)outSize * ((double)outFormat->sampleRate / (double)inFormat->sampleRate)); outSize *= shift; |