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authorAvuton Olrich <avuton@gmail.com>2006-07-20 16:02:40 +0000
committerAvuton Olrich <avuton@gmail.com>2006-07-20 16:02:40 +0000
commit29a25b9933b32800f58dd73d5d1fc21993071c92 (patch)
tree4f456a6f8e44d42acc289c35534ea3e59c0aa96f /src/pcm_utils.c
parent099f0e103f7591eef81183292d704b3a77a99018 (diff)
downloadmpd-29a25b9933b32800f58dd73d5d1fc21993071c92.tar.gz
mpd-29a25b9933b32800f58dd73d5d1fc21993071c92.tar.xz
mpd-29a25b9933b32800f58dd73d5d1fc21993071c92.zip
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
Diffstat (limited to 'src/pcm_utils.c')
-rw-r--r--src/pcm_utils.c239
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;