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authorAndreas Claesson <andreas.claesson@gmail.com>2005-05-24 15:06:23 +0000
committerAndreas Claesson <andreas.claesson@gmail.com>2005-05-24 15:06:23 +0000
commit29cc42bf9781f2407cc7ccfe801329b07434e50b (patch)
tree7a7b61381e788880c03b99325cb7448d5b7f88cd /src/replayGain.c
parent422410c46de81cec8872a9b91d07a7c9eca96b82 (diff)
downloadmpd-29cc42bf9781f2407cc7ccfe801329b07434e50b.tar.gz
mpd-29cc42bf9781f2407cc7ccfe801329b07434e50b.tar.xz
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Adding modifications for internal 1-3-28 bits integer samples. Only mp3 and mpc of the input plugins are modified. Resampling is NOT working!!
git-svn-id: https://svn.musicpd.org/mpd/branches/ancl@3281 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Diffstat (limited to '')
-rw-r--r--src/replayGain.c108
1 files changed, 31 insertions, 77 deletions
diff --git a/src/replayGain.c b/src/replayGain.c
index f052c68bc..aac057668 100644
--- a/src/replayGain.c
+++ b/src/replayGain.c
@@ -107,20 +107,19 @@ void freeReplayGainInfo(ReplayGainInfo * info) {
void doReplayGain(ReplayGainInfo * info, char * buffer, int bufferSize,
AudioFormat * format)
{
- float * bufferFloat = (float *)buffer;
mpd_sint32 * buffer32 = (mpd_sint32 *)buffer;
- mpd_sint16 * buffer16 = (mpd_sint16 *)buffer;
- mpd_sint8 * buffer8 = (mpd_sint8 *)buffer;
- float scale;
- mpd_sint32 iScale;
+ int samples;
+ int shift;
+ int iScale;
+ if(format->bits!=32 || format->channels!=2 ) {
+ ERROR("Only 32 bit stereo is supported for doReplayGain!\n");
+ exit(EXIT_FAILURE);
+ return;
+ }
+
if(replayGainState == REPLAYGAIN_OFF || !info) return;
-/* DEBUG */
- if(bufferSize % (format->channels * 4))
- ERROR("doReplayGain: bufferSize=%i not multipel of %i\n",
- bufferSize, format->channels);
-/* /DEBUG */
if(info->scale < 0) {
switch(replayGainState) {
case REPLAYGAIN_TRACK:
@@ -133,79 +132,34 @@ void doReplayGain(ReplayGainInfo * info, char * buffer, int bufferSize,
break;
}
}
-
-#ifdef MPD_FIXED_POINT
- if(format->bits!=16 || format.channels!=2 || format->floatSamples) {
- ERROR("Only 16 bit stereo is supported in fixed point mode!\n");
- exit(EXIT_FAILURE);
- }
- /* If the volume change is very small then we hardly here the
- * difference anyway, and if the change is positiv then clipping
- * may occur. We don't want that. */
- if(info->scale > 0.99) return;
+ samples = bufferSize >> 2;
+ iScale = info->scale * 256;
+ shift = 8;
- iScale = scale * 32768.0; /* << 15 */
- while(bufferSize) {
- sample32 = (mpd_sint32)(*buffer16) * iScale;
- /* no need to check boundaries - we only lower the volume*/
- /* It would be good to add some kind of dither here... TODO?! */
- *buffer16 = (sample32 >> 15);
- bufferSize -= 2;
+ /* handle negative or zero scale */
+ if(iScale<=0) {
+ memset(buffer,0,bufferSize);
+ return;
}
- return;
-#else
-
- scale = info->scale;
- if(format->floatSamples) {
- if(format->bits==32) {
- while(bufferSize) {
- *bufferFloat *= scale;
- bufferFloat++;
- bufferSize-=4;
- }
- return;
- }
- else {
- ERROR("%i bit float not supported by doReplaygain!\n",
- format->bits);
- exit(EXIT_FAILURE);
- }
+ /* lower shift value as much as possible */
+ while(!(iScale & 1) && shift) {
+ iScale >>= 1;
+ shift--;
}
- switch(format->bits) {
- case 32:
- while(bufferSize) {
- double sample = (double)(*buffer32) * scale;
- if(sample>2147483647.0) *buffer32 = 2147483647;
- else if(sample<-2147483647.0) *buffer32 = -2147483647;
- else *buffer32 = rintf(sample);
- *buffer32++;
- bufferSize-=4;
- }
- break;
- case 16:
- while(bufferSize){
- float sample = *buffer16 * scale;
- *buffer16 = sample>32767.0 ? 32767 :
- (sample<-32768.0 ? -32768 : rintf(sample));
- buffer16++;
- bufferSize-=2;
- }
- break;
- case 8:
- while(bufferSize){
- float sample = *buffer8 * scale;
- *buffer8 = sample>127.0 ? 127 :
- (sample<-128.0 ? -128 : rintf(sample));
- buffer8++;
- bufferSize--;
- }
- break;
- default:
- ERROR("%i bits not supported by doReplaygain!\n", format->bits);
+ /* change samples */
+ /* no check for overflow needed - replaygain peak info prevent
+ * clipping and we have 3 headroom bits in our 32 bit samples */
+ if(iScale == 1) {
+ while(samples--)
+ *buffer32++ = *buffer32 >> shift;
}
-#endif
+ else {
+ while(samples--)
+ *buffer32++ = (*buffer32 >> shift) * iScale;
+ }
+
}