Reindenting
git-svn-id: https://svn.musicpd.org/mpd/trunk@6199 09075e82-0dd4-0310-85a5-a0d7c8717e4f
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@ -32,7 +32,7 @@
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#endif
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void pcm_volumeChange(char *buffer, int bufferSize, AudioFormat * format,
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int volume)
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int volume)
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{
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mpd_sint32 temp32;
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mpd_sint8 *buffer8 = (mpd_sint8 *) buffer;
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@ -83,8 +83,8 @@ void pcm_volumeChange(char *buffer, int bufferSize, AudioFormat * format,
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}
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static void pcm_add(char *buffer1, char *buffer2, size_t bufferSize1,
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size_t bufferSize2, int vol1, int vol2,
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AudioFormat * format)
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size_t bufferSize2, int vol1, int vol2,
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AudioFormat * format)
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{
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mpd_sint32 temp32;
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mpd_sint8 *buffer8_1 = (mpd_sint8 *) buffer1;
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@ -139,7 +139,7 @@ static void pcm_add(char *buffer1, char *buffer2, size_t bufferSize1,
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}
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void pcm_mix(char *buffer1, char *buffer2, size_t bufferSize1,
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size_t bufferSize2, AudioFormat * format, float portion1)
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size_t bufferSize2, AudioFormat * format, float portion1)
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{
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int vol1;
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float s = sin(M_PI_2 * portion1);
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@ -153,7 +153,8 @@ void pcm_mix(char *buffer1, char *buffer2, size_t bufferSize1,
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}
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#ifdef HAVE_LIBSAMPLERATE
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static int pcm_getSamplerateConverter(void) {
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static int pcm_getSamplerateConverter(void)
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{
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const char *conf, *test;
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int convalgo = SRC_SINC_FASTEST;
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int newalgo;
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@ -185,8 +186,9 @@ static int pcm_getSamplerateConverter(void) {
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#endif
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/* outFormat bits must be 16 and channels must be 2! */
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void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer, size_t
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inSize, AudioFormat * outFormat, char *outBuffer)
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void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer,
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size_t inSize, AudioFormat * outFormat,
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char *outBuffer)
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{
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static char *bitConvBuffer;
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static int bitConvBufferLength;
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@ -376,8 +378,8 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char *inBuffer, size_t
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}
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size_t pcm_sizeOfOutputBufferForAudioFormatConversion(AudioFormat * inFormat,
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size_t inSize,
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AudioFormat * outFormat)
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size_t inSize,
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AudioFormat * outFormat)
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{
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const int shift = sizeof(mpd_sint16) * outFormat->channels;
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size_t outSize = inSize;
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@ -404,7 +406,7 @@ size_t pcm_sizeOfOutputBufferForAudioFormatConversion(AudioFormat * inFormat,
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}
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}
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outSize /= shift;
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outSize /= shift;
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outSize = floor(0.5 + (double)outSize *
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((double)outFormat->sampleRate / (double)inFormat->sampleRate));
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outSize *= shift;
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