Mercurial > mplayer.hg
view mp3lib/layer1.c @ 24900:9079c9745ff9
A/V sync: take audio filter buffers into account
Substract the delay caused by filter buffering when calculating
currently playing audio position. This matters for af_scaletempo which
buffers significant and varying amounts of data. For other current
filters the effect is normally insignificant.
Instead of the old time-based filter delay field (which was ignored)
this version stores the per-filter delay in units of bytes input read
without corresponding output. This allows the current scaletempo
behavior where other filters before and after it can see the same
nominal samplerate even though the real duration of the data varies;
in this case the other filters can not know the delay they're causing
in terms of real time.
author | uau |
---|---|
date | Thu, 01 Nov 2007 06:52:50 +0000 |
parents | 1b1fdac4a68c |
children | 0f1b5b68af32 |
line wrap: on
line source
/* * Mpeg Layer-1 audio decoder * -------------------------- * copyright (c) 1995 by Michael Hipp, All rights reserved. See also 'README' * near unoptimzed ... * * may have a few bugs after last optimization ... * */ /* * Modified for use with MPlayer, for details see the changelog at * http://svn.mplayerhq.hu/mplayer/trunk/ * $Id$ * * The above-mentioned README file has the following to say about licensing: * * COPYING: you may use this source under LGPL terms! */ //#include "mpg123.h" static void I_step_one(unsigned int balloc[], unsigned int scale_index[2][SBLIMIT],struct frame *fr) { unsigned int *ba=balloc; unsigned int *sca = (unsigned int *) scale_index; if(fr->stereo == 2) { int i; int jsbound = fr->jsbound; for (i=0;i<jsbound;i++) { *ba++ = getbits(4); *ba++ = getbits(4); } for (i=jsbound;i<SBLIMIT;i++) *ba++ = getbits(4); ba = balloc; for (i=0;i<jsbound;i++) { if ((*ba++)) *sca++ = getbits(6); if ((*ba++)) *sca++ = getbits(6); } for (i=jsbound;i<SBLIMIT;i++) if ((*ba++)) { *sca++ = getbits(6); *sca++ = getbits(6); } } else { int i; for (i=0;i<SBLIMIT;i++) *ba++ = getbits(4); ba = balloc; for (i=0;i<SBLIMIT;i++) if ((*ba++)) *sca++ = getbits(6); } } static void I_step_two(real fraction[2][SBLIMIT],unsigned int balloc[2*SBLIMIT], unsigned int scale_index[2][SBLIMIT],struct frame *fr) { int i,n; int smpb[2*SBLIMIT]; /* values: 0-65535 */ int *sample; register unsigned int *ba; register unsigned int *sca = (unsigned int *) scale_index; if(fr->stereo == 2) { int jsbound = fr->jsbound; register real *f0 = fraction[0]; register real *f1 = fraction[1]; ba = balloc; for (sample=smpb,i=0;i<jsbound;i++) { if ((n = *ba++)) *sample++ = getbits(n+1); if ((n = *ba++)) *sample++ = getbits(n+1); } for (i=jsbound;i<SBLIMIT;i++) if ((n = *ba++)) *sample++ = getbits(n+1); ba = balloc; for (sample=smpb,i=0;i<jsbound;i++) { if((n=*ba++)) *f0++ = (real) ( ((-1)<<n) + (*sample++) + 1) * muls[n+1][*sca++]; else *f0++ = 0.0; if((n=*ba++)) *f1++ = (real) ( ((-1)<<n) + (*sample++) + 1) * muls[n+1][*sca++]; else *f1++ = 0.0; } for (i=jsbound;i<SBLIMIT;i++) { if ((n=*ba++)) { real samp = ( ((-1)<<n) + (*sample++) + 1); *f0++ = samp * muls[n+1][*sca++]; *f1++ = samp * muls[n+1][*sca++]; } else *f0++ = *f1++ = 0.0; } for(i=fr->down_sample_sblimit;i<32;i++) fraction[0][i] = fraction[1][i] = 0.0; } else { register real *f0 = fraction[0]; ba = balloc; for (sample=smpb,i=0;i<SBLIMIT;i++) if ((n = *ba++)) *sample++ = getbits(n+1); ba = balloc; for (sample=smpb,i=0;i<SBLIMIT;i++) { if((n=*ba++)) *f0++ = (real) ( ((-1)<<n) + (*sample++) + 1) * muls[n+1][*sca++]; else *f0++ = 0.0; } for(i=fr->down_sample_sblimit;i<32;i++) fraction[0][i] = 0.0; } } static int do_layer1(struct frame *fr,int single) { int clip=0; int i,stereo = fr->stereo; unsigned int balloc[2*SBLIMIT]; unsigned int scale_index[2][SBLIMIT]; DECLARE_ALIGNED(16, real, fraction[2][SBLIMIT]); // int single = fr->single; // printf("do_layer1(0x%02X 0x%02X 0x%02X 0x%02X 0x%02X 0x%02X 0x%02X 0x%02X )\n", // wordpointer[0],wordpointer[1],wordpointer[2],wordpointer[3],wordpointer[4],wordpointer[5],wordpointer[6],wordpointer[7]); fr->jsbound = (fr->mode == MPG_MD_JOINT_STEREO) ? (fr->mode_ext<<2)+4 : 32; if(stereo == 1 || single == 3) single = 0; I_step_one(balloc,scale_index,fr); for (i=0;i<SCALE_BLOCK;i++) { I_step_two(fraction,balloc,scale_index,fr); if(single >= 0) { clip += (fr->synth_mono)( (real *) fraction[single],pcm_sample,&pcm_point); } else { int p1 = pcm_point; clip += (fr->synth)( (real *) fraction[0],0,pcm_sample,&p1); clip += (fr->synth)( (real *) fraction[1],1,pcm_sample,&pcm_point); } } return clip; }