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authorWarren Dukes <warren.dukes@gmail.com>2004-10-23 01:04:58 +0000
committerWarren Dukes <warren.dukes@gmail.com>2004-10-23 01:04:58 +0000
commit12597322a2c4a1271249eb66192bfb6f0e1294e5 (patch)
tree201e8b5d6ded1df66e19951df617214369bc4acc /src/pcm_utils.c
parent413bf61ea1c3beaee98197e8d759ac2375ce51ef (diff)
downloadmpd-12597322a2c4a1271249eb66192bfb6f0e1294e5.tar.gz
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configure shout encoding quality and audio format
git-svn-id: https://svn.musicpd.org/mpd/trunk@2307 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Diffstat (limited to '')
-rw-r--r--src/pcm_utils.c75
1 files changed, 49 insertions, 26 deletions
diff --git a/src/pcm_utils.c b/src/pcm_utils.c
index e9beeeead..b3191a399 100644
--- a/src/pcm_utils.c
+++ b/src/pcm_utils.c
@@ -153,7 +153,7 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
int dataBitLen;
assert(outFormat->bits==16);
- assert(outFormat->channels==2);
+ assert(outFormat->channels==2 || outFormat->channels==1);
/* converts */
switch(inFormat->bits) {
@@ -185,32 +185,55 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
}
/* converts only between 16 bit audio between mono and stereo */
- switch(inFormat->channels) {
- case 1:
- dataChannelLen = (dataBitLen >> 1) << 2;
- if(dataChannelLen > channelConvBufferLength) {
- channelConvBuffer = realloc(channelConvBuffer,
- dataChannelLen);
- channelConvBufferLength = dataChannelLen;
- }
- dataChannelConv = channelConvBuffer;
- {
- mpd_sint16 * in = (mpd_sint16 *)dataBitConv;
- mpd_sint16 * out = (mpd_sint16 *)dataChannelConv;
- int i, inSamples = dataBitLen >> 1;
- for(i=0;i<inSamples;i++) {
- *out++ = *in;
- *out++ = *in++;
- }
- }
- break;
- case 2:
+ if(inFormat->channels == outFormat->channels)
+ {
dataChannelConv = dataBitConv;
dataChannelLen = dataBitLen;
- break;
- default:
- ERROR("only 1 or 2 channels are supported for conversion!\n");
- exit(EXIT_FAILURE);
+ }
+ else {
+ switch(inFormat->channels) {
+ /* convert from 1 -> 2 channels */
+ case 1:
+ dataChannelLen = (dataBitLen >> 1) << 2;
+ if(dataChannelLen > channelConvBufferLength) {
+ channelConvBuffer = realloc(channelConvBuffer,
+ dataChannelLen);
+ channelConvBufferLength = dataChannelLen;
+ }
+ dataChannelConv = channelConvBuffer;
+ {
+ mpd_sint16 * in = (mpd_sint16 *)dataBitConv;
+ mpd_sint16 * out = (mpd_sint16 *)dataChannelConv;
+ int i, inSamples = dataBitLen >> 1;
+ for(i=0;i<inSamples;i++) {
+ *out++ = *in;
+ *out++ = *in++;
+ }
+ }
+ break;
+ /* convert from 2 -> 1 channels */
+ case 2:
+ dataChannelLen = dataBitLen >> 1;
+ if(dataChannelLen > channelConvBufferLength) {
+ channelConvBuffer = realloc(channelConvBuffer,
+ dataChannelLen);
+ channelConvBufferLength = dataChannelLen;
+ }
+ dataChannelConv = channelConvBuffer;
+ {
+ 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;
+ }
+ }
+ break;
+ default:
+ ERROR("only 1 or 2 channels are supported for conversion!\n");
+ exit(EXIT_FAILURE);
+ }
}
if(inFormat->sampleRate == outFormat->sampleRate) {
@@ -247,7 +270,7 @@ void pcm_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
}
size_t pcm_sizeOfOutputBufferForAudioFormatConversion(AudioFormat * inFormat,
- char * inBuffer, size_t inSize, AudioFormat * outFormat)
+ size_t inSize, AudioFormat * outFormat)
{
const int shift = sizeof(mpd_sint16);
size_t outSize = inSize;