diff options
author | Andreas Claesson <andreas.claesson@gmail.com> | 2005-05-24 15:06:23 +0000 |
---|---|---|
committer | Andreas Claesson <andreas.claesson@gmail.com> | 2005-05-24 15:06:23 +0000 |
commit | 29cc42bf9781f2407cc7ccfe801329b07434e50b (patch) | |
tree | 7a7b61381e788880c03b99325cb7448d5b7f88cd /src/replayGain.c | |
parent | 422410c46de81cec8872a9b91d07a7c9eca96b82 (diff) | |
download | mpd-29cc42bf9781f2407cc7ccfe801329b07434e50b.tar.gz mpd-29cc42bf9781f2407cc7ccfe801329b07434e50b.tar.xz mpd-29cc42bf9781f2407cc7ccfe801329b07434e50b.zip |
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.c | 108 |
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; + } + } |