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authorAvuton Olrich <avuton@gmail.com>2006-07-14 20:08:35 +0000
committerAvuton Olrich <avuton@gmail.com>2006-07-14 20:08:35 +0000
commita8938c91ba2f65ad9c1c33b2d8a6ee252ff6974a (patch)
treed87e581eb410738c022013fcfd96490ceec823e2
parent2a4a75d9c72ef7b695c6e70af16daf841ec96848 (diff)
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[CLEANUP] Remove unused function
Cleanup whitespace->tabs git-svn-id: https://svn.musicpd.org/mpd/trunk@4338 09075e82-0dd4-0310-85a5-a0d7c8717e4f
-rw-r--r--src/pcm_utils.c63
-rw-r--r--src/pcm_utils.h2
2 files changed, 24 insertions, 41 deletions
diff --git a/src/pcm_utils.c b/src/pcm_utils.c
index c7bcaa208..74846387b 100644
--- a/src/pcm_utils.c
+++ b/src/pcm_utils.c
@@ -26,21 +26,6 @@
#include <math.h>
#include <assert.h>
-void pcm_changeBufferEndianness(char * buffer, int bufferSize, int bits) {
- char temp;
-
- switch(bits) {
- case 16:
- while(bufferSize) {
- temp = *buffer;
- *buffer = *(buffer+1);
- *(buffer+1) = temp;
- bufferSize-=2;
- }
- break;
- }
-}
-
void pcm_volumeChange(char * buffer, int bufferSize, AudioFormat * format,
int volume)
{
@@ -85,7 +70,7 @@ void pcm_volumeChange(char * buffer, int bufferSize, AudioFormat * format,
}
}
-void pcm_add(char * buffer1, char * buffer2, size_t bufferSize1,
+static void pcm_add(char * buffer1, char * buffer2, size_t bufferSize1,
size_t bufferSize2, int vol1, int vol2, AudioFormat * format)
{
mpd_sint32 temp32;
@@ -242,40 +227,40 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
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)) /
+ 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);
nlen *= 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;
- rd_dat &= ~1;
+ 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;
}
}
diff --git a/src/pcm_utils.h b/src/pcm_utils.h
index a366c2a2d..32f5444aa 100644
--- a/src/pcm_utils.h
+++ b/src/pcm_utils.h
@@ -25,8 +25,6 @@
#include <stdlib.h>
-void pcm_changeBufferEndianness(char * buffer, int bufferSize, int bits);
-
void pcm_volumeChange(char * buffer, int bufferSize, AudioFormat * format,
int volume);