Add mpd-indent.sh

Indent the entire tree, hopefully we can keep
it indented.

git-svn-id: https://svn.musicpd.org/mpd/trunk@4410 09075e82-0dd4-0310-85a5-a0d7c8717e4f

1 files changed, 124 insertions, 115 deletions

diff --git a/src/pcm_utils.c b/src/pcm_utils.c
index 74846387b..0210c290f 100644
--- a/src/pcm_utils.c
+++ b/src/pcm_utils.c
@@ -16,7 +16,6 @@
  * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
  */
 
-
 #include "pcm_utils.h"
 
 #include "mpd_types.h"
@@ -26,134 +25,143 @@
 #include <math.h>
 #include <assert.h>
 
-void pcm_volumeChange(char * buffer, int bufferSize, AudioFormat * format,
-		int volume)
+void pcm_volumeChange(char *buffer, int bufferSize, AudioFormat * format,
+		      int volume)
 {
 	mpd_sint32 temp32;
-	mpd_sint8 * buffer8 = (mpd_sint8 *)buffer;
-	mpd_sint16 * buffer16 = (mpd_sint16 *)buffer;
+	mpd_sint8 *buffer8 = (mpd_sint8 *) buffer;
+	mpd_sint16 *buffer16 = (mpd_sint16 *) buffer;
+
+	if (volume >= 1000)
+		return;
 
-	if(volume>=1000) return;
-	
-	if(volume<=0) {
-		memset(buffer,0,bufferSize);
+	if (volume <= 0) {
+		memset(buffer, 0, bufferSize);
 		return;
 	}
 
-	switch(format->bits) {
+	switch (format->bits) {
 	case 16:
-		while(bufferSize>0) {
+		while (bufferSize > 0) {
 			temp32 = *buffer16;
-			temp32*= volume;
-			temp32/=1000;
-			*buffer16 = temp32>32767 ? 32767 : 
-					(temp32<-32768 ? -32768 : temp32);
+			temp32 *= volume;
+			temp32 /= 1000;
+			*buffer16 = temp32 > 32767 ? 32767 :
+			    (temp32 < -32768 ? -32768 : temp32);
 			buffer16++;
-			bufferSize-=2;
+			bufferSize -= 2;
 		}
 		break;
 	case 8:
-		while(bufferSize>0) {
+		while (bufferSize > 0) {
 			temp32 = *buffer8;
-			temp32*= volume;
-			temp32/=1000;
-			*buffer8 = temp32>127 ? 127 : 
-					(temp32<-128 ? -128 : temp32);
+			temp32 *= volume;
+			temp32 /= 1000;
+			*buffer8 = temp32 > 127 ? 127 :
+			    (temp32 < -128 ? -128 : temp32);
 			buffer8++;
 			bufferSize--;
 		}
 		break;
 	default:
 		ERROR("%i bits not supported by pcm_volumeChange!\n",
-				format->bits);
+		      format->bits);
 		exit(EXIT_FAILURE);
 	}
 }
 
-static void pcm_add(char * buffer1, char * buffer2, size_t bufferSize1, 
-		size_t bufferSize2, int vol1, int vol2, AudioFormat * format)
+static void pcm_add(char *buffer1, char *buffer2, size_t bufferSize1,
+		    size_t bufferSize2, int vol1, int vol2,
+		    AudioFormat * format)
 {
 	mpd_sint32 temp32;
-	mpd_sint8 * buffer8_1 = (mpd_sint8 *)buffer1;
-	mpd_sint8 * buffer8_2 = (mpd_sint8 *)buffer2;
-	mpd_sint16 * buffer16_1 = (mpd_sint16 *)buffer1;
-	mpd_sint16 * buffer16_2 = (mpd_sint16 *)buffer2;
+	mpd_sint8 *buffer8_1 = (mpd_sint8 *) buffer1;
+	mpd_sint8 *buffer8_2 = (mpd_sint8 *) buffer2;
+	mpd_sint16 *buffer16_1 = (mpd_sint16 *) buffer1;
+	mpd_sint16 *buffer16_2 = (mpd_sint16 *) buffer2;
 
-	switch(format->bits) {
+	switch (format->bits) {
 	case 16:
-		while(bufferSize1>0 && bufferSize2>0) {
-			temp32 = (vol1*(*buffer16_1)+vol2*(*buffer16_2))/1000;
-			*buffer16_1 = temp32>32767 ? 32767 : 
-					(temp32<-32768 ? -32768 : temp32);
+		while (bufferSize1 > 0 && bufferSize2 > 0) {
+			temp32 =
+			    (vol1 * (*buffer16_1) +
+			     vol2 * (*buffer16_2)) / 1000;
+			*buffer16_1 =
+			    temp32 > 32767 ? 32767 : (temp32 <
+						      -32768 ? -32768 : temp32);
 			buffer16_1++;
 			buffer16_2++;
-			bufferSize1-=2;
-			bufferSize2-=2;
+			bufferSize1 -= 2;
+			bufferSize2 -= 2;
 		}
-		if(bufferSize2>0) memcpy(buffer16_1,buffer16_2,bufferSize2);
+		if (bufferSize2 > 0)
+			memcpy(buffer16_1, buffer16_2, bufferSize2);
 		break;
 	case 8:
-		while(bufferSize1>0 && bufferSize2>0) {
-			temp32 = (vol1*(*buffer8_1)+vol2*(*buffer8_2))/1000;
-			*buffer8_1 = temp32>127 ? 127 : 
-					(temp32<-128 ? -128 : temp32);
+		while (bufferSize1 > 0 && bufferSize2 > 0) {
+			temp32 =
+			    (vol1 * (*buffer8_1) + vol2 * (*buffer8_2)) / 1000;
+			*buffer8_1 =
+			    temp32 > 127 ? 127 : (temp32 <
+						  -128 ? -128 : temp32);
 			buffer8_1++;
 			buffer8_2++;
 			bufferSize1--;
 			bufferSize2--;
 		}
-		if(bufferSize2>0) memcpy(buffer8_1,buffer8_2,bufferSize2);
+		if (bufferSize2 > 0)
+			memcpy(buffer8_1, buffer8_2, bufferSize2);
 		break;
 	default:
-		ERROR("%i bits not supported by pcm_add!\n",format->bits);
+		ERROR("%i bits not supported by pcm_add!\n", format->bits);
 		exit(EXIT_FAILURE);
 	}
 }
 
-void pcm_mix(char * buffer1, char * buffer2, size_t bufferSize1, 
-		size_t bufferSize2, AudioFormat * format, float portion1)
+void pcm_mix(char *buffer1, char *buffer2, size_t bufferSize1,
+	     size_t bufferSize2, AudioFormat * format, float portion1)
 {
 	int vol1;
-	float s = sin(M_PI_2*portion1);
-	s*=s;
-	
-	vol1 = s*1000+0.5;
-	vol1 = vol1>1000 ? 1000 : ( vol1<0 ? 0 : vol1 );
+	float s = sin(M_PI_2 * portion1);
+	s *= s;
 
-	pcm_add(buffer1,buffer2,bufferSize1,bufferSize2,vol1,1000-vol1,format);
-}
+	vol1 = s * 1000 + 0.5;
+	vol1 = vol1 > 1000 ? 1000 : (vol1 < 0 ? 0 : vol1);
 
+	pcm_add(buffer1, buffer2, bufferSize1, bufferSize2, vol1, 1000 - vol1,
+		format);
+}
 
 /* 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 = NULL;
+	static char *bitConvBuffer = NULL;
 	static int bitConvBufferLength = 0;
-	static char * channelConvBuffer = NULL;
+	static char *channelConvBuffer = NULL;
 	static int channelConvBufferLength = 0;
-	char * dataChannelConv;
+	char *dataChannelConv;
 	int dataChannelLen;
-	char * dataBitConv;
+	char *dataBitConv;
 	int dataBitLen;
 
-	assert(outFormat->bits==16);
-	assert(outFormat->channels==2 || outFormat->channels==1);
+	assert(outFormat->bits == 16);
+	assert(outFormat->channels == 2 || outFormat->channels == 1);
 
 	/* converts */
-	switch(inFormat->bits) {
+	switch (inFormat->bits) {
 	case 8:
 		dataBitLen = inSize << 1;
-		if(dataBitLen > bitConvBufferLength) {
+		if (dataBitLen > bitConvBufferLength) {
 			bitConvBuffer = realloc(bitConvBuffer, dataBitLen);
 			bitConvBufferLength = dataBitLen;
 		}
 		dataBitConv = bitConvBuffer;
 		{
-			mpd_sint8 * in = (mpd_sint8 *)inBuffer;
-			mpd_sint16 * out = (mpd_sint16 *)dataBitConv;
+			mpd_sint8 *in = (mpd_sint8 *) inBuffer;
+			mpd_sint16 *out = (mpd_sint16 *) dataBitConv;
 			int i;
-			for(i=0; i<inSize; i++) {
+			for (i = 0; i < inSize; i++) {
 				*out++ = (*in++) << 8;
 			}
 		}
@@ -170,96 +178,96 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
 	}
 
 	/* converts only between 16 bit audio between mono and stereo */
-	if(inFormat->channels == outFormat->channels)
-	{
+	if (inFormat->channels == outFormat->channels) {
 		dataChannelConv = dataBitConv;
 		dataChannelLen = dataBitLen;
-	}
-	else {
-		switch(inFormat->channels) {
-		/* convert from 1 -> 2 channels */
+	} else {
+		switch (inFormat->channels) {
+			/* convert from 1 -> 2 channels */
 		case 1:
 			dataChannelLen = (dataBitLen >> 1) << 2;
-			if(dataChannelLen > channelConvBufferLength) {
+			if (dataChannelLen > channelConvBufferLength) {
 				channelConvBuffer = realloc(channelConvBuffer,
-						dataChannelLen);
+							    dataChannelLen);
 				channelConvBufferLength = dataChannelLen;
 			}
 			dataChannelConv = channelConvBuffer;
 			{
-				mpd_sint16 * in = (mpd_sint16 *)dataBitConv;
-				mpd_sint16 * out = (mpd_sint16 *)dataChannelConv;
+				mpd_sint16 *in = (mpd_sint16 *) dataBitConv;
+				mpd_sint16 *out =
+				    (mpd_sint16 *) dataChannelConv;
 				int i, inSamples = dataBitLen >> 1;
-				for(i=0;i<inSamples;i++) {
+				for (i = 0; i < inSamples; i++) {
 					*out++ = *in;
 					*out++ = *in++;
 				}
 			}
 			break;
-		/* convert from 2 -> 1 channels */
+			/* convert from 2 -> 1 channels */
 		case 2:
 			dataChannelLen = dataBitLen >> 1;
-			if(dataChannelLen > channelConvBufferLength) {
+			if (dataChannelLen > channelConvBufferLength) {
 				channelConvBuffer = realloc(channelConvBuffer,
-						dataChannelLen);
+							    dataChannelLen);
 				channelConvBufferLength = dataChannelLen;
 			}
 			dataChannelConv = channelConvBuffer;
 			{
-				mpd_sint16 * in = (mpd_sint16 *)dataBitConv;
-				mpd_sint16 * out = (mpd_sint16 *)dataChannelConv;
+				mpd_sint16 *in = (mpd_sint16 *) dataBitConv;
+				mpd_sint16 *out =
+				    (mpd_sint16 *) dataChannelConv;
 				int i, inSamples = dataBitLen >> 2;
-				for(i=0;i<inSamples;i++) {
-					*out = (*in++)/2;
-					*out++ += (*in++)/2;
+				for (i = 0; i < inSamples; i++) {
+					*out = (*in++) / 2;
+					*out++ += (*in++) / 2;
 				}
 			}
 			break;
 		default:
-			ERROR("only 1 or 2 channels are supported for conversion!\n");
+			ERROR
+			    ("only 1 or 2 channels are supported for conversion!\n");
 			exit(EXIT_FAILURE);
 		}
 	}
 
-	if(inFormat->sampleRate == outFormat->sampleRate) {
-		memcpy(outBuffer,dataChannelConv,dataChannelLen);
-	}
-	else {
+	if (inFormat->sampleRate == outFormat->sampleRate) {
+		memcpy(outBuffer, dataChannelConv, dataChannelLen);
+	} else {
 		/* only works if outFormat is 16-bit stereo! */
 		/* resampling code blatantly ripped from ESD */
 		mpd_uint32 rd_dat = 0;
 		mpd_uint32 wr_dat = 0;
 		mpd_sint16 lsample, rsample;
-		mpd_sint16 * out = (mpd_sint16 *)outBuffer;
-		mpd_sint16 * in = (mpd_sint16 *)dataChannelConv;
-		const int shift = sizeof(mpd_sint16)*outFormat->channels;
-		mpd_uint32 nlen = ((( dataChannelLen / shift) * 
-				(mpd_uint32)(outFormat->sampleRate)) /
-				inFormat->sampleRate);
+		mpd_sint16 *out = (mpd_sint16 *) outBuffer;
+		mpd_sint16 *in = (mpd_sint16 *) dataChannelConv;
+		const int shift = sizeof(mpd_sint16) * outFormat->channels;
+		mpd_uint32 nlen = (((dataChannelLen / shift) *
+				    (mpd_uint32) (outFormat->sampleRate)) /
+				   inFormat->sampleRate);
 		nlen *= outFormat->channels;
 
-		switch(outFormat->channels) {
+		switch (outFormat->channels) {
 		case 1:
-			while( wr_dat < nlen) {
-				rd_dat = wr_dat * inFormat->sampleRate / 
-				outFormat->sampleRate;
+			while (wr_dat < nlen) {
+				rd_dat = wr_dat * inFormat->sampleRate /
+				    outFormat->sampleRate;
 
-				lsample = in[ rd_dat++ ];
+				lsample = in[rd_dat++];
 
-				out[ wr_dat++ ] = lsample;
+				out[wr_dat++] = lsample;
 			}
 			break;
 		case 2:
-			while( wr_dat < nlen) {
-				rd_dat = wr_dat * inFormat->sampleRate / 
-				outFormat->sampleRate;
+			while (wr_dat < nlen) {
+				rd_dat = wr_dat * inFormat->sampleRate /
+				    outFormat->sampleRate;
 				rd_dat &= ~1;
 
-				lsample = in[ rd_dat++ ];
-				rsample = in[ rd_dat++ ];
+				lsample = in[rd_dat++];
+				rsample = in[rd_dat++];
 
-				out[ wr_dat++ ] = lsample;
-				out[ wr_dat++ ] = rsample;
+				out[wr_dat++] = lsample;
+				out[wr_dat++] = rsample;
 			}
 			break;
 		}
@@ -269,12 +277,13 @@ 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;
+	const int shift = sizeof(mpd_sint16) * outFormat->channels;
 	size_t outSize = inSize;
 
-	switch(inFormat->bits) {
+	switch (inFormat->bits) {
 	case 8:
 		outSize = outSize << 1;
 		break;
@@ -285,8 +294,8 @@ size_t pcm_sizeOfOutputBufferForAudioFormatConversion(AudioFormat * inFormat,
 		exit(EXIT_FAILURE);
 	}
 
-	if(inFormat->channels != outFormat->channels) {
-		switch(inFormat->channels) {
+	if (inFormat->channels != outFormat->channels) {
+		switch (inFormat->channels) {
 		case 1:
 			outSize = (outSize >> 1) << 2;
 			break;
@@ -295,9 +304,9 @@ size_t pcm_sizeOfOutputBufferForAudioFormatConversion(AudioFormat * inFormat,
 			break;
 		}
 	}
-	
-	outSize = (((outSize / shift) * (mpd_uint32)(outFormat->sampleRate)) /
-				inFormat->sampleRate);
+
+	outSize = (((outSize / shift) * (mpd_uint32) (outFormat->sampleRate)) /
+		   inFormat->sampleRate);
 
 	outSize *= shift;