1048 lines
30 KiB
C
1048 lines
30 KiB
C
/********************************************************************
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* *
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* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
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* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
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* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
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* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
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* *
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* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2015 *
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* by the Xiph.Org Foundation http://www.xiph.org/ *
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* *
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********************************************************************
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function: PCM data vector blocking, windowing and dis/reassembly
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last mod: $Id: block.c 19457 2015-03-03 00:15:29Z giles $
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Handle windowing, overlap-add, etc of the PCM vectors. This is made
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more amusing by Vorbis' current two allowed block sizes.
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********************************************************************/
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <ogg/ogg.h>
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#include "vorbis/codec.h"
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#include "codec_internal.h"
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#include "window.h"
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#include "mdct.h"
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#include "lpc.h"
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#include "registry.h"
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#include "misc.h"
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/* pcm accumulator examples (not exhaustive):
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<-------------- lW ---------------->
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<--------------- W ---------------->
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: .....|..... _______________ |
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: .''' | '''_--- | |\ |
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:.....''' |_____--- '''......| | \_______|
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:.................|__________________|_______|__|______|
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|<------ Sl ------>| > Sr < |endW
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|beginSl |endSl | |endSr
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|beginW |endlW |beginSr
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|< lW >|
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<--------------- W ---------------->
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| | .. ______________ |
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| | ' `/ | ---_ |
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|___.'___/`. | ---_____|
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|_______|__|_______|_________________|
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| >|Sl|< |<------ Sr ----->|endW
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| | |endSl |beginSr |endSr
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|beginW | |endlW
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mult[0] |beginSl mult[n]
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<-------------- lW ----------------->
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|<--W-->|
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: .............. ___ | |
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: .''' |`/ \ | |
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:.....''' |/`....\|...|
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:.........................|___|___|___|
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|Sl |Sr |endW
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| | |endSr
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| |beginSr
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| |endSl
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|beginSl
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|beginW
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*/
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/* block abstraction setup *********************************************/
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#ifndef WORD_ALIGN
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#define WORD_ALIGN 8
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#endif
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int vorbis_block_init(vorbis_dsp_state *v, vorbis_block *vb){
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int i;
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memset(vb,0,sizeof(*vb));
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vb->vd=v;
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vb->localalloc=0;
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vb->localstore=NULL;
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if(v->analysisp){
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vorbis_block_internal *vbi=
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vb->internal=_ogg_calloc(1,sizeof(vorbis_block_internal));
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vbi->ampmax=-9999;
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for(i=0;i<PACKETBLOBS;i++){
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if(i==PACKETBLOBS/2){
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vbi->packetblob[i]=&vb->opb;
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}else{
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vbi->packetblob[i]=
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_ogg_calloc(1,sizeof(oggpack_buffer));
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}
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oggpack_writeinit(vbi->packetblob[i]);
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}
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}
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return(0);
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}
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void *_vorbis_block_alloc(vorbis_block *vb,long bytes){
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bytes=(bytes+(WORD_ALIGN-1)) & ~(WORD_ALIGN-1);
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if(bytes+vb->localtop>vb->localalloc){
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/* can't just _ogg_realloc... there are outstanding pointers */
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if(vb->localstore){
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struct alloc_chain *link=_ogg_malloc(sizeof(*link));
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vb->totaluse+=vb->localtop;
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link->next=vb->reap;
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link->ptr=vb->localstore;
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vb->reap=link;
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}
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/* highly conservative */
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vb->localalloc=bytes;
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vb->localstore=_ogg_malloc(vb->localalloc);
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vb->localtop=0;
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}
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{
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void *ret=(void *)(((char *)vb->localstore)+vb->localtop);
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vb->localtop+=bytes;
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return ret;
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}
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}
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/* reap the chain, pull the ripcord */
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void _vorbis_block_ripcord(vorbis_block *vb){
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/* reap the chain */
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struct alloc_chain *reap=vb->reap;
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while(reap){
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struct alloc_chain *next=reap->next;
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_ogg_free(reap->ptr);
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memset(reap,0,sizeof(*reap));
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_ogg_free(reap);
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reap=next;
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}
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/* consolidate storage */
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if(vb->totaluse){
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vb->localstore=_ogg_realloc(vb->localstore,vb->totaluse+vb->localalloc);
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vb->localalloc+=vb->totaluse;
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vb->totaluse=0;
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}
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/* pull the ripcord */
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vb->localtop=0;
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vb->reap=NULL;
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}
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int vorbis_block_clear(vorbis_block *vb){
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int i;
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vorbis_block_internal *vbi=vb->internal;
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_vorbis_block_ripcord(vb);
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if(vb->localstore)_ogg_free(vb->localstore);
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if(vbi){
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for(i=0;i<PACKETBLOBS;i++){
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oggpack_writeclear(vbi->packetblob[i]);
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if(i!=PACKETBLOBS/2)_ogg_free(vbi->packetblob[i]);
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}
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_ogg_free(vbi);
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}
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memset(vb,0,sizeof(*vb));
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return(0);
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}
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/* Analysis side code, but directly related to blocking. Thus it's
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here and not in analysis.c (which is for analysis transforms only).
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The init is here because some of it is shared */
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static int _vds_shared_init(vorbis_dsp_state *v,vorbis_info *vi,int encp){
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int i;
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codec_setup_info *ci=vi->codec_setup;
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private_state *b=NULL;
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int hs;
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if(ci==NULL||
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ci->modes<=0||
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ci->blocksizes[0]<64||
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ci->blocksizes[1]<ci->blocksizes[0]){
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return 1;
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}
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hs=ci->halfrate_flag;
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memset(v,0,sizeof(*v));
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b=v->backend_state=_ogg_calloc(1,sizeof(*b));
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v->vi=vi;
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b->modebits=ov_ilog(ci->modes-1);
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b->transform[0]=_ogg_calloc(VI_TRANSFORMB,sizeof(*b->transform[0]));
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b->transform[1]=_ogg_calloc(VI_TRANSFORMB,sizeof(*b->transform[1]));
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/* MDCT is tranform 0 */
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b->transform[0][0]=_ogg_calloc(1,sizeof(mdct_lookup));
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b->transform[1][0]=_ogg_calloc(1,sizeof(mdct_lookup));
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mdct_init(b->transform[0][0],ci->blocksizes[0]>>hs);
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mdct_init(b->transform[1][0],ci->blocksizes[1]>>hs);
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/* Vorbis I uses only window type 0 */
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/* note that the correct computation below is technically:
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b->window[0]=ov_ilog(ci->blocksizes[0]-1)-6;
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b->window[1]=ov_ilog(ci->blocksizes[1]-1)-6;
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but since blocksizes are always powers of two,
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the below is equivalent.
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*/
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b->window[0]=ov_ilog(ci->blocksizes[0])-7;
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b->window[1]=ov_ilog(ci->blocksizes[1])-7;
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if(encp){ /* encode/decode differ here */
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/* analysis always needs an fft */
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drft_init(&b->fft_look[0],ci->blocksizes[0]);
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drft_init(&b->fft_look[1],ci->blocksizes[1]);
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/* finish the codebooks */
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if(!ci->fullbooks){
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ci->fullbooks=_ogg_calloc(ci->books,sizeof(*ci->fullbooks));
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for(i=0;i<ci->books;i++)
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vorbis_book_init_encode(ci->fullbooks+i,ci->book_param[i]);
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}
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b->psy=_ogg_calloc(ci->psys,sizeof(*b->psy));
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for(i=0;i<ci->psys;i++){
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_vp_psy_init(b->psy+i,
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ci->psy_param[i],
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&ci->psy_g_param,
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ci->blocksizes[ci->psy_param[i]->blockflag]/2,
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vi->rate);
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}
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v->analysisp=1;
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}else{
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/* finish the codebooks */
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if(!ci->fullbooks){
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ci->fullbooks=_ogg_calloc(ci->books,sizeof(*ci->fullbooks));
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for(i=0;i<ci->books;i++){
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if(ci->book_param[i]==NULL)
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goto abort_books;
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if(vorbis_book_init_decode(ci->fullbooks+i,ci->book_param[i]))
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goto abort_books;
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/* decode codebooks are now standalone after init */
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vorbis_staticbook_destroy(ci->book_param[i]);
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ci->book_param[i]=NULL;
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}
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}
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}
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/* initialize the storage vectors. blocksize[1] is small for encode,
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but the correct size for decode */
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v->pcm_storage=ci->blocksizes[1];
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v->pcm=_ogg_malloc(vi->channels*sizeof(*v->pcm));
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v->pcmret=_ogg_malloc(vi->channels*sizeof(*v->pcmret));
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{
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int i;
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for(i=0;i<vi->channels;i++)
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v->pcm[i]=_ogg_calloc(v->pcm_storage,sizeof(*v->pcm[i]));
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}
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/* all 1 (large block) or 0 (small block) */
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/* explicitly set for the sake of clarity */
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v->lW=0; /* previous window size */
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v->W=0; /* current window size */
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/* all vector indexes */
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v->centerW=ci->blocksizes[1]/2;
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v->pcm_current=v->centerW;
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/* initialize all the backend lookups */
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b->flr=_ogg_calloc(ci->floors,sizeof(*b->flr));
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b->residue=_ogg_calloc(ci->residues,sizeof(*b->residue));
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for(i=0;i<ci->floors;i++)
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b->flr[i]=_floor_P[ci->floor_type[i]]->
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look(v,ci->floor_param[i]);
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for(i=0;i<ci->residues;i++)
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b->residue[i]=_residue_P[ci->residue_type[i]]->
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look(v,ci->residue_param[i]);
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return 0;
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abort_books:
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for(i=0;i<ci->books;i++){
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if(ci->book_param[i]!=NULL){
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vorbis_staticbook_destroy(ci->book_param[i]);
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ci->book_param[i]=NULL;
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}
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}
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vorbis_dsp_clear(v);
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return -1;
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}
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/* arbitrary settings and spec-mandated numbers get filled in here */
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int vorbis_analysis_init(vorbis_dsp_state *v,vorbis_info *vi){
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private_state *b=NULL;
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if(_vds_shared_init(v,vi,1))return 1;
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b=v->backend_state;
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b->psy_g_look=_vp_global_look(vi);
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/* Initialize the envelope state storage */
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b->ve=_ogg_calloc(1,sizeof(*b->ve));
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_ve_envelope_init(b->ve,vi);
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vorbis_bitrate_init(vi,&b->bms);
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/* compressed audio packets start after the headers
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with sequence number 3 */
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v->sequence=3;
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return(0);
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}
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void vorbis_dsp_clear(vorbis_dsp_state *v){
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int i;
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if(v){
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vorbis_info *vi=v->vi;
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codec_setup_info *ci=(vi?vi->codec_setup:NULL);
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private_state *b=v->backend_state;
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if(b){
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if(b->ve){
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_ve_envelope_clear(b->ve);
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_ogg_free(b->ve);
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}
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if(b->transform[0]){
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mdct_clear(b->transform[0][0]);
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_ogg_free(b->transform[0][0]);
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_ogg_free(b->transform[0]);
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}
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if(b->transform[1]){
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mdct_clear(b->transform[1][0]);
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_ogg_free(b->transform[1][0]);
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_ogg_free(b->transform[1]);
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}
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if(b->flr){
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if(ci)
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for(i=0;i<ci->floors;i++)
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_floor_P[ci->floor_type[i]]->
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free_look(b->flr[i]);
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_ogg_free(b->flr);
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}
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if(b->residue){
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if(ci)
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for(i=0;i<ci->residues;i++)
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_residue_P[ci->residue_type[i]]->
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free_look(b->residue[i]);
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_ogg_free(b->residue);
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}
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if(b->psy){
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if(ci)
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for(i=0;i<ci->psys;i++)
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_vp_psy_clear(b->psy+i);
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_ogg_free(b->psy);
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}
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if(b->psy_g_look)_vp_global_free(b->psy_g_look);
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vorbis_bitrate_clear(&b->bms);
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drft_clear(&b->fft_look[0]);
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drft_clear(&b->fft_look[1]);
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}
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if(v->pcm){
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if(vi)
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for(i=0;i<vi->channels;i++)
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if(v->pcm[i])_ogg_free(v->pcm[i]);
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_ogg_free(v->pcm);
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if(v->pcmret)_ogg_free(v->pcmret);
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}
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if(b){
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/* free header, header1, header2 */
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if(b->header)_ogg_free(b->header);
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if(b->header1)_ogg_free(b->header1);
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if(b->header2)_ogg_free(b->header2);
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_ogg_free(b);
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}
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memset(v,0,sizeof(*v));
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}
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}
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float **vorbis_analysis_buffer(vorbis_dsp_state *v, int vals){
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int i;
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vorbis_info *vi=v->vi;
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private_state *b=v->backend_state;
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/* free header, header1, header2 */
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if(b->header)_ogg_free(b->header);b->header=NULL;
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if(b->header1)_ogg_free(b->header1);b->header1=NULL;
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if(b->header2)_ogg_free(b->header2);b->header2=NULL;
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/* Do we have enough storage space for the requested buffer? If not,
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expand the PCM (and envelope) storage */
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if(v->pcm_current+vals>=v->pcm_storage){
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v->pcm_storage=v->pcm_current+vals*2;
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for(i=0;i<vi->channels;i++){
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v->pcm[i]=_ogg_realloc(v->pcm[i],v->pcm_storage*sizeof(*v->pcm[i]));
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}
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}
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for(i=0;i<vi->channels;i++)
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v->pcmret[i]=v->pcm[i]+v->pcm_current;
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return(v->pcmret);
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}
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static void _preextrapolate_helper(vorbis_dsp_state *v){
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int i;
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int order=16;
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float *lpc=alloca(order*sizeof(*lpc));
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float *work=alloca(v->pcm_current*sizeof(*work));
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long j;
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v->preextrapolate=1;
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if(v->pcm_current-v->centerW>order*2){ /* safety */
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for(i=0;i<v->vi->channels;i++){
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/* need to run the extrapolation in reverse! */
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for(j=0;j<v->pcm_current;j++)
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work[j]=v->pcm[i][v->pcm_current-j-1];
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/* prime as above */
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vorbis_lpc_from_data(work,lpc,v->pcm_current-v->centerW,order);
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#if 0
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if(v->vi->channels==2){
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if(i==0)
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_analysis_output("predataL",0,work,v->pcm_current-v->centerW,0,0,0);
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else
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_analysis_output("predataR",0,work,v->pcm_current-v->centerW,0,0,0);
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}else{
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_analysis_output("predata",0,work,v->pcm_current-v->centerW,0,0,0);
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}
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#endif
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/* run the predictor filter */
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vorbis_lpc_predict(lpc,work+v->pcm_current-v->centerW-order,
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order,
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work+v->pcm_current-v->centerW,
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v->centerW);
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for(j=0;j<v->pcm_current;j++)
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v->pcm[i][v->pcm_current-j-1]=work[j];
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}
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}
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}
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/* call with val<=0 to set eof */
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int vorbis_analysis_wrote(vorbis_dsp_state *v, int vals){
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vorbis_info *vi=v->vi;
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codec_setup_info *ci=vi->codec_setup;
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if(vals<=0){
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int order=32;
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int i;
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float *lpc=alloca(order*sizeof(*lpc));
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/* if it wasn't done earlier (very short sample) */
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if(!v->preextrapolate)
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_preextrapolate_helper(v);
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/* We're encoding the end of the stream. Just make sure we have
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[at least] a few full blocks of zeroes at the end. */
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/* actually, we don't want zeroes; that could drop a large
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amplitude off a cliff, creating spread spectrum noise that will
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suck to encode. Extrapolate for the sake of cleanliness. */
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vorbis_analysis_buffer(v,ci->blocksizes[1]*3);
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v->eofflag=v->pcm_current;
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v->pcm_current+=ci->blocksizes[1]*3;
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for(i=0;i<vi->channels;i++){
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if(v->eofflag>order*2){
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/* extrapolate with LPC to fill in */
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long n;
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/* make a predictor filter */
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n=v->eofflag;
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if(n>ci->blocksizes[1])n=ci->blocksizes[1];
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vorbis_lpc_from_data(v->pcm[i]+v->eofflag-n,lpc,n,order);
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|
|
|
/* run the predictor filter */
|
|
vorbis_lpc_predict(lpc,v->pcm[i]+v->eofflag-order,order,
|
|
v->pcm[i]+v->eofflag,v->pcm_current-v->eofflag);
|
|
}else{
|
|
/* not enough data to extrapolate (unlikely to happen due to
|
|
guarding the overlap, but bulletproof in case that
|
|
assumtion goes away). zeroes will do. */
|
|
memset(v->pcm[i]+v->eofflag,0,
|
|
(v->pcm_current-v->eofflag)*sizeof(*v->pcm[i]));
|
|
|
|
}
|
|
}
|
|
}else{
|
|
|
|
if(v->pcm_current+vals>v->pcm_storage)
|
|
return(OV_EINVAL);
|
|
|
|
v->pcm_current+=vals;
|
|
|
|
/* we may want to reverse extrapolate the beginning of a stream
|
|
too... in case we're beginning on a cliff! */
|
|
/* clumsy, but simple. It only runs once, so simple is good. */
|
|
if(!v->preextrapolate && v->pcm_current-v->centerW>ci->blocksizes[1])
|
|
_preextrapolate_helper(v);
|
|
|
|
}
|
|
return(0);
|
|
}
|
|
|
|
/* do the deltas, envelope shaping, pre-echo and determine the size of
|
|
the next block on which to continue analysis */
|
|
int vorbis_analysis_blockout(vorbis_dsp_state *v,vorbis_block *vb){
|
|
int i;
|
|
vorbis_info *vi=v->vi;
|
|
codec_setup_info *ci=vi->codec_setup;
|
|
private_state *b=v->backend_state;
|
|
vorbis_look_psy_global *g=b->psy_g_look;
|
|
long beginW=v->centerW-ci->blocksizes[v->W]/2,centerNext;
|
|
vorbis_block_internal *vbi=(vorbis_block_internal *)vb->internal;
|
|
|
|
/* check to see if we're started... */
|
|
if(!v->preextrapolate)return(0);
|
|
|
|
/* check to see if we're done... */
|
|
if(v->eofflag==-1)return(0);
|
|
|
|
/* By our invariant, we have lW, W and centerW set. Search for
|
|
the next boundary so we can determine nW (the next window size)
|
|
which lets us compute the shape of the current block's window */
|
|
|
|
/* we do an envelope search even on a single blocksize; we may still
|
|
be throwing more bits at impulses, and envelope search handles
|
|
marking impulses too. */
|
|
{
|
|
long bp=_ve_envelope_search(v);
|
|
if(bp==-1){
|
|
|
|
if(v->eofflag==0)return(0); /* not enough data currently to search for a
|
|
full long block */
|
|
v->nW=0;
|
|
}else{
|
|
|
|
if(ci->blocksizes[0]==ci->blocksizes[1])
|
|
v->nW=0;
|
|
else
|
|
v->nW=bp;
|
|
}
|
|
}
|
|
|
|
centerNext=v->centerW+ci->blocksizes[v->W]/4+ci->blocksizes[v->nW]/4;
|
|
|
|
{
|
|
/* center of next block + next block maximum right side. */
|
|
|
|
long blockbound=centerNext+ci->blocksizes[v->nW]/2;
|
|
if(v->pcm_current<blockbound)return(0); /* not enough data yet;
|
|
although this check is
|
|
less strict that the
|
|
_ve_envelope_search,
|
|
the search is not run
|
|
if we only use one
|
|
block size */
|
|
|
|
|
|
}
|
|
|
|
/* fill in the block. Note that for a short window, lW and nW are *short*
|
|
regardless of actual settings in the stream */
|
|
|
|
_vorbis_block_ripcord(vb);
|
|
vb->lW=v->lW;
|
|
vb->W=v->W;
|
|
vb->nW=v->nW;
|
|
|
|
if(v->W){
|
|
if(!v->lW || !v->nW){
|
|
vbi->blocktype=BLOCKTYPE_TRANSITION;
|
|
/*fprintf(stderr,"-");*/
|
|
}else{
|
|
vbi->blocktype=BLOCKTYPE_LONG;
|
|
/*fprintf(stderr,"_");*/
|
|
}
|
|
}else{
|
|
if(_ve_envelope_mark(v)){
|
|
vbi->blocktype=BLOCKTYPE_IMPULSE;
|
|
/*fprintf(stderr,"|");*/
|
|
|
|
}else{
|
|
vbi->blocktype=BLOCKTYPE_PADDING;
|
|
/*fprintf(stderr,".");*/
|
|
|
|
}
|
|
}
|
|
|
|
vb->vd=v;
|
|
vb->sequence=v->sequence++;
|
|
vb->granulepos=v->granulepos;
|
|
vb->pcmend=ci->blocksizes[v->W];
|
|
|
|
/* copy the vectors; this uses the local storage in vb */
|
|
|
|
/* this tracks 'strongest peak' for later psychoacoustics */
|
|
/* moved to the global psy state; clean this mess up */
|
|
if(vbi->ampmax>g->ampmax)g->ampmax=vbi->ampmax;
|
|
g->ampmax=_vp_ampmax_decay(g->ampmax,v);
|
|
vbi->ampmax=g->ampmax;
|
|
|
|
vb->pcm=_vorbis_block_alloc(vb,sizeof(*vb->pcm)*vi->channels);
|
|
vbi->pcmdelay=_vorbis_block_alloc(vb,sizeof(*vbi->pcmdelay)*vi->channels);
|
|
for(i=0;i<vi->channels;i++){
|
|
vbi->pcmdelay[i]=
|
|
_vorbis_block_alloc(vb,(vb->pcmend+beginW)*sizeof(*vbi->pcmdelay[i]));
|
|
memcpy(vbi->pcmdelay[i],v->pcm[i],(vb->pcmend+beginW)*sizeof(*vbi->pcmdelay[i]));
|
|
vb->pcm[i]=vbi->pcmdelay[i]+beginW;
|
|
|
|
/* before we added the delay
|
|
vb->pcm[i]=_vorbis_block_alloc(vb,vb->pcmend*sizeof(*vb->pcm[i]));
|
|
memcpy(vb->pcm[i],v->pcm[i]+beginW,ci->blocksizes[v->W]*sizeof(*vb->pcm[i]));
|
|
*/
|
|
|
|
}
|
|
|
|
/* handle eof detection: eof==0 means that we've not yet received EOF
|
|
eof>0 marks the last 'real' sample in pcm[]
|
|
eof<0 'no more to do'; doesn't get here */
|
|
|
|
if(v->eofflag){
|
|
if(v->centerW>=v->eofflag){
|
|
v->eofflag=-1;
|
|
vb->eofflag=1;
|
|
return(1);
|
|
}
|
|
}
|
|
|
|
/* advance storage vectors and clean up */
|
|
{
|
|
int new_centerNext=ci->blocksizes[1]/2;
|
|
int movementW=centerNext-new_centerNext;
|
|
|
|
if(movementW>0){
|
|
|
|
_ve_envelope_shift(b->ve,movementW);
|
|
v->pcm_current-=movementW;
|
|
|
|
for(i=0;i<vi->channels;i++)
|
|
memmove(v->pcm[i],v->pcm[i]+movementW,
|
|
v->pcm_current*sizeof(*v->pcm[i]));
|
|
|
|
|
|
v->lW=v->W;
|
|
v->W=v->nW;
|
|
v->centerW=new_centerNext;
|
|
|
|
if(v->eofflag){
|
|
v->eofflag-=movementW;
|
|
if(v->eofflag<=0)v->eofflag=-1;
|
|
/* do not add padding to end of stream! */
|
|
if(v->centerW>=v->eofflag){
|
|
v->granulepos+=movementW-(v->centerW-v->eofflag);
|
|
}else{
|
|
v->granulepos+=movementW;
|
|
}
|
|
}else{
|
|
v->granulepos+=movementW;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* done */
|
|
return(1);
|
|
}
|
|
|
|
int vorbis_synthesis_restart(vorbis_dsp_state *v){
|
|
vorbis_info *vi=v->vi;
|
|
codec_setup_info *ci;
|
|
int hs;
|
|
|
|
if(!v->backend_state)return -1;
|
|
if(!vi)return -1;
|
|
ci=vi->codec_setup;
|
|
if(!ci)return -1;
|
|
hs=ci->halfrate_flag;
|
|
|
|
v->centerW=ci->blocksizes[1]>>(hs+1);
|
|
v->pcm_current=v->centerW>>hs;
|
|
|
|
v->pcm_returned=-1;
|
|
v->granulepos=-1;
|
|
v->sequence=-1;
|
|
v->eofflag=0;
|
|
((private_state *)(v->backend_state))->sample_count=-1;
|
|
|
|
return(0);
|
|
}
|
|
|
|
int vorbis_synthesis_init(vorbis_dsp_state *v,vorbis_info *vi){
|
|
if(_vds_shared_init(v,vi,0)){
|
|
vorbis_dsp_clear(v);
|
|
return 1;
|
|
}
|
|
vorbis_synthesis_restart(v);
|
|
return 0;
|
|
}
|
|
|
|
/* Unlike in analysis, the window is only partially applied for each
|
|
block. The time domain envelope is not yet handled at the point of
|
|
calling (as it relies on the previous block). */
|
|
|
|
int vorbis_synthesis_blockin(vorbis_dsp_state *v,vorbis_block *vb){
|
|
vorbis_info *vi=v->vi;
|
|
codec_setup_info *ci=vi->codec_setup;
|
|
private_state *b=v->backend_state;
|
|
int hs=ci->halfrate_flag;
|
|
int i,j;
|
|
|
|
if(!vb)return(OV_EINVAL);
|
|
if(v->pcm_current>v->pcm_returned && v->pcm_returned!=-1)return(OV_EINVAL);
|
|
|
|
v->lW=v->W;
|
|
v->W=vb->W;
|
|
v->nW=-1;
|
|
|
|
if((v->sequence==-1)||
|
|
(v->sequence+1 != vb->sequence)){
|
|
v->granulepos=-1; /* out of sequence; lose count */
|
|
b->sample_count=-1;
|
|
}
|
|
|
|
v->sequence=vb->sequence;
|
|
|
|
if(vb->pcm){ /* no pcm to process if vorbis_synthesis_trackonly
|
|
was called on block */
|
|
int n=ci->blocksizes[v->W]>>(hs+1);
|
|
int n0=ci->blocksizes[0]>>(hs+1);
|
|
int n1=ci->blocksizes[1]>>(hs+1);
|
|
|
|
int thisCenter;
|
|
int prevCenter;
|
|
|
|
v->glue_bits+=vb->glue_bits;
|
|
v->time_bits+=vb->time_bits;
|
|
v->floor_bits+=vb->floor_bits;
|
|
v->res_bits+=vb->res_bits;
|
|
|
|
if(v->centerW){
|
|
thisCenter=n1;
|
|
prevCenter=0;
|
|
}else{
|
|
thisCenter=0;
|
|
prevCenter=n1;
|
|
}
|
|
|
|
/* v->pcm is now used like a two-stage double buffer. We don't want
|
|
to have to constantly shift *or* adjust memory usage. Don't
|
|
accept a new block until the old is shifted out */
|
|
|
|
for(j=0;j<vi->channels;j++){
|
|
/* the overlap/add section */
|
|
if(v->lW){
|
|
if(v->W){
|
|
/* large/large */
|
|
const float *w=_vorbis_window_get(b->window[1]-hs);
|
|
float *pcm=v->pcm[j]+prevCenter;
|
|
float *p=vb->pcm[j];
|
|
for(i=0;i<n1;i++)
|
|
pcm[i]=pcm[i]*w[n1-i-1] + p[i]*w[i];
|
|
}else{
|
|
/* large/small */
|
|
const float *w=_vorbis_window_get(b->window[0]-hs);
|
|
float *pcm=v->pcm[j]+prevCenter+n1/2-n0/2;
|
|
float *p=vb->pcm[j];
|
|
for(i=0;i<n0;i++)
|
|
pcm[i]=pcm[i]*w[n0-i-1] +p[i]*w[i];
|
|
}
|
|
}else{
|
|
if(v->W){
|
|
/* small/large */
|
|
const float *w=_vorbis_window_get(b->window[0]-hs);
|
|
float *pcm=v->pcm[j]+prevCenter;
|
|
float *p=vb->pcm[j]+n1/2-n0/2;
|
|
for(i=0;i<n0;i++)
|
|
pcm[i]=pcm[i]*w[n0-i-1] +p[i]*w[i];
|
|
for(;i<n1/2+n0/2;i++)
|
|
pcm[i]=p[i];
|
|
}else{
|
|
/* small/small */
|
|
const float *w=_vorbis_window_get(b->window[0]-hs);
|
|
float *pcm=v->pcm[j]+prevCenter;
|
|
float *p=vb->pcm[j];
|
|
for(i=0;i<n0;i++)
|
|
pcm[i]=pcm[i]*w[n0-i-1] +p[i]*w[i];
|
|
}
|
|
}
|
|
|
|
/* the copy section */
|
|
{
|
|
float *pcm=v->pcm[j]+thisCenter;
|
|
float *p=vb->pcm[j]+n;
|
|
for(i=0;i<n;i++)
|
|
pcm[i]=p[i];
|
|
}
|
|
}
|
|
|
|
if(v->centerW)
|
|
v->centerW=0;
|
|
else
|
|
v->centerW=n1;
|
|
|
|
/* deal with initial packet state; we do this using the explicit
|
|
pcm_returned==-1 flag otherwise we're sensitive to first block
|
|
being short or long */
|
|
|
|
if(v->pcm_returned==-1){
|
|
v->pcm_returned=thisCenter;
|
|
v->pcm_current=thisCenter;
|
|
}else{
|
|
v->pcm_returned=prevCenter;
|
|
v->pcm_current=prevCenter+
|
|
((ci->blocksizes[v->lW]/4+
|
|
ci->blocksizes[v->W]/4)>>hs);
|
|
}
|
|
|
|
}
|
|
|
|
/* track the frame number... This is for convenience, but also
|
|
making sure our last packet doesn't end with added padding. If
|
|
the last packet is partial, the number of samples we'll have to
|
|
return will be past the vb->granulepos.
|
|
|
|
This is not foolproof! It will be confused if we begin
|
|
decoding at the last page after a seek or hole. In that case,
|
|
we don't have a starting point to judge where the last frame
|
|
is. For this reason, vorbisfile will always try to make sure
|
|
it reads the last two marked pages in proper sequence */
|
|
|
|
if(b->sample_count==-1){
|
|
b->sample_count=0;
|
|
}else{
|
|
b->sample_count+=ci->blocksizes[v->lW]/4+ci->blocksizes[v->W]/4;
|
|
}
|
|
|
|
if(v->granulepos==-1){
|
|
if(vb->granulepos!=-1){ /* only set if we have a position to set to */
|
|
|
|
v->granulepos=vb->granulepos;
|
|
|
|
/* is this a short page? */
|
|
if(b->sample_count>v->granulepos){
|
|
/* corner case; if this is both the first and last audio page,
|
|
then spec says the end is cut, not beginning */
|
|
long extra=b->sample_count-vb->granulepos;
|
|
|
|
/* we use ogg_int64_t for granule positions because a
|
|
uint64 isn't universally available. Unfortunately,
|
|
that means granposes can be 'negative' and result in
|
|
extra being negative */
|
|
if(extra<0)
|
|
extra=0;
|
|
|
|
if(vb->eofflag){
|
|
/* trim the end */
|
|
/* no preceding granulepos; assume we started at zero (we'd
|
|
have to in a short single-page stream) */
|
|
/* granulepos could be -1 due to a seek, but that would result
|
|
in a long count, not short count */
|
|
|
|
/* Guard against corrupt/malicious frames that set EOP and
|
|
a backdated granpos; don't rewind more samples than we
|
|
actually have */
|
|
if(extra > (v->pcm_current - v->pcm_returned)<<hs)
|
|
extra = (v->pcm_current - v->pcm_returned)<<hs;
|
|
|
|
v->pcm_current-=extra>>hs;
|
|
}else{
|
|
/* trim the beginning */
|
|
v->pcm_returned+=extra>>hs;
|
|
if(v->pcm_returned>v->pcm_current)
|
|
v->pcm_returned=v->pcm_current;
|
|
}
|
|
|
|
}
|
|
|
|
}
|
|
}else{
|
|
v->granulepos+=ci->blocksizes[v->lW]/4+ci->blocksizes[v->W]/4;
|
|
if(vb->granulepos!=-1 && v->granulepos!=vb->granulepos){
|
|
|
|
if(v->granulepos>vb->granulepos){
|
|
long extra=v->granulepos-vb->granulepos;
|
|
|
|
if(extra)
|
|
if(vb->eofflag){
|
|
/* partial last frame. Strip the extra samples off */
|
|
|
|
/* Guard against corrupt/malicious frames that set EOP and
|
|
a backdated granpos; don't rewind more samples than we
|
|
actually have */
|
|
if(extra > (v->pcm_current - v->pcm_returned)<<hs)
|
|
extra = (v->pcm_current - v->pcm_returned)<<hs;
|
|
|
|
/* we use ogg_int64_t for granule positions because a
|
|
uint64 isn't universally available. Unfortunately,
|
|
that means granposes can be 'negative' and result in
|
|
extra being negative */
|
|
if(extra<0)
|
|
extra=0;
|
|
|
|
v->pcm_current-=extra>>hs;
|
|
} /* else {Shouldn't happen *unless* the bitstream is out of
|
|
spec. Either way, believe the bitstream } */
|
|
} /* else {Shouldn't happen *unless* the bitstream is out of
|
|
spec. Either way, believe the bitstream } */
|
|
v->granulepos=vb->granulepos;
|
|
}
|
|
}
|
|
|
|
/* Update, cleanup */
|
|
|
|
if(vb->eofflag)v->eofflag=1;
|
|
return(0);
|
|
|
|
}
|
|
|
|
/* pcm==NULL indicates we just want the pending samples, no more */
|
|
int vorbis_synthesis_pcmout(vorbis_dsp_state *v,float ***pcm){
|
|
vorbis_info *vi=v->vi;
|
|
|
|
if(v->pcm_returned>-1 && v->pcm_returned<v->pcm_current){
|
|
if(pcm){
|
|
int i;
|
|
for(i=0;i<vi->channels;i++)
|
|
v->pcmret[i]=v->pcm[i]+v->pcm_returned;
|
|
*pcm=v->pcmret;
|
|
}
|
|
return(v->pcm_current-v->pcm_returned);
|
|
}
|
|
return(0);
|
|
}
|
|
|
|
int vorbis_synthesis_read(vorbis_dsp_state *v,int n){
|
|
if(n && v->pcm_returned+n>v->pcm_current)return(OV_EINVAL);
|
|
v->pcm_returned+=n;
|
|
return(0);
|
|
}
|
|
|
|
/* intended for use with a specific vorbisfile feature; we want access
|
|
to the [usually synthetic/postextrapolated] buffer and lapping at
|
|
the end of a decode cycle, specifically, a half-short-block worth.
|
|
This funtion works like pcmout above, except it will also expose
|
|
this implicit buffer data not normally decoded. */
|
|
int vorbis_synthesis_lapout(vorbis_dsp_state *v,float ***pcm){
|
|
vorbis_info *vi=v->vi;
|
|
codec_setup_info *ci=vi->codec_setup;
|
|
int hs=ci->halfrate_flag;
|
|
|
|
int n=ci->blocksizes[v->W]>>(hs+1);
|
|
int n0=ci->blocksizes[0]>>(hs+1);
|
|
int n1=ci->blocksizes[1]>>(hs+1);
|
|
int i,j;
|
|
|
|
if(v->pcm_returned<0)return 0;
|
|
|
|
/* our returned data ends at pcm_returned; because the synthesis pcm
|
|
buffer is a two-fragment ring, that means our data block may be
|
|
fragmented by buffering, wrapping or a short block not filling
|
|
out a buffer. To simplify things, we unfragment if it's at all
|
|
possibly needed. Otherwise, we'd need to call lapout more than
|
|
once as well as hold additional dsp state. Opt for
|
|
simplicity. */
|
|
|
|
/* centerW was advanced by blockin; it would be the center of the
|
|
*next* block */
|
|
if(v->centerW==n1){
|
|
/* the data buffer wraps; swap the halves */
|
|
/* slow, sure, small */
|
|
for(j=0;j<vi->channels;j++){
|
|
float *p=v->pcm[j];
|
|
for(i=0;i<n1;i++){
|
|
float temp=p[i];
|
|
p[i]=p[i+n1];
|
|
p[i+n1]=temp;
|
|
}
|
|
}
|
|
|
|
v->pcm_current-=n1;
|
|
v->pcm_returned-=n1;
|
|
v->centerW=0;
|
|
}
|
|
|
|
/* solidify buffer into contiguous space */
|
|
if((v->lW^v->W)==1){
|
|
/* long/short or short/long */
|
|
for(j=0;j<vi->channels;j++){
|
|
float *s=v->pcm[j];
|
|
float *d=v->pcm[j]+(n1-n0)/2;
|
|
for(i=(n1+n0)/2-1;i>=0;--i)
|
|
d[i]=s[i];
|
|
}
|
|
v->pcm_returned+=(n1-n0)/2;
|
|
v->pcm_current+=(n1-n0)/2;
|
|
}else{
|
|
if(v->lW==0){
|
|
/* short/short */
|
|
for(j=0;j<vi->channels;j++){
|
|
float *s=v->pcm[j];
|
|
float *d=v->pcm[j]+n1-n0;
|
|
for(i=n0-1;i>=0;--i)
|
|
d[i]=s[i];
|
|
}
|
|
v->pcm_returned+=n1-n0;
|
|
v->pcm_current+=n1-n0;
|
|
}
|
|
}
|
|
|
|
if(pcm){
|
|
int i;
|
|
for(i=0;i<vi->channels;i++)
|
|
v->pcmret[i]=v->pcm[i]+v->pcm_returned;
|
|
*pcm=v->pcmret;
|
|
}
|
|
|
|
return(n1+n-v->pcm_returned);
|
|
|
|
}
|
|
|
|
const float *vorbis_window(vorbis_dsp_state *v,int W){
|
|
vorbis_info *vi=v->vi;
|
|
codec_setup_info *ci=vi->codec_setup;
|
|
int hs=ci->halfrate_flag;
|
|
private_state *b=v->backend_state;
|
|
|
|
if(b->window[W]-1<0)return NULL;
|
|
return _vorbis_window_get(b->window[W]-hs);
|
|
}
|