diff -r 78a00bc68913 -r 767d3c4153a1 misc/libtremor/tremor/dsp.c
--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/misc/libtremor/tremor/dsp.c	Fri Sep 21 00:50:04 2012 +0200
@@ -0,0 +1,298 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis 'TREMOR' CODEC SOURCE CODE.   *
+ *                                                                  *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis 'TREMOR' SOURCE CODE IS (C) COPYRIGHT 1994-2003    *
+ * BY THE Xiph.Org FOUNDATION http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: PCM data vector blocking, windowing and dis/reassembly
+
+ ********************************************************************/
+
+#include <stdlib.h> 
+#include "ogg.h"
+#include "mdct.h"
+#include "ivorbiscodec.h"
+#include "codec_internal.h"
+#include "misc.h"
+#include "window_lookup.h"
+
+int vorbis_dsp_restart(vorbis_dsp_state *v){
+  if(!v)return -1;
+  {
+    vorbis_info *vi=v->vi;
+    codec_setup_info *ci;
+    
+    if(!vi)return -1;
+    ci=vi->codec_setup;
+    if(!ci)return -1;
+    
+    v->out_end=-1;
+    v->out_begin=-1;
+
+    v->granulepos=-1;
+    v->sequence=-1;
+    v->sample_count=-1;
+  }
+  return 0;
+}
+
+vorbis_dsp_state *vorbis_dsp_create(vorbis_info *vi){
+  int i;
+
+  vorbis_dsp_state *v=_ogg_calloc(1,sizeof(*v));
+  codec_setup_info *ci=(codec_setup_info *)vi->codec_setup;
+
+  v->vi=vi;
+  
+  v->work=(ogg_int32_t **)_ogg_malloc(vi->channels*sizeof(*v->work));
+  v->mdctright=(ogg_int32_t **)_ogg_malloc(vi->channels*sizeof(*v->mdctright));
+  for(i=0;i<vi->channels;i++){
+    v->work[i]=(ogg_int32_t *)_ogg_calloc(1,(ci->blocksizes[1]>>1)*
+					  sizeof(*v->work[i]));
+    v->mdctright[i]=(ogg_int32_t *)_ogg_calloc(1,(ci->blocksizes[1]>>2)*
+					       sizeof(*v->mdctright[i]));
+  }
+
+  v->lW=0; /* previous window size */
+  v->W=0;  /* current window size */
+
+  vorbis_dsp_restart(v);
+  return v;
+}
+
+void vorbis_dsp_destroy(vorbis_dsp_state *v){
+  int i;
+  if(v){
+    vorbis_info *vi=v->vi;
+
+    if(v->work){
+      for(i=0;i<vi->channels;i++)
+        if(v->work[i])_ogg_free(v->work[i]);
+      _ogg_free(v->work);
+    }
+    if(v->mdctright){
+      for(i=0;i<vi->channels;i++)
+        if(v->mdctright[i])_ogg_free(v->mdctright[i]);
+      _ogg_free(v->mdctright);
+    }
+
+    _ogg_free(v);
+  }
+}
+
+static LOOKUP_T *_vorbis_window(int left){
+  switch(left){
+  case 32:
+    return vwin64;
+  case 64:
+    return vwin128;
+  case 128:
+    return vwin256;
+  case 256:
+    return vwin512;
+  case 512:
+    return vwin1024;
+  case 1024:
+    return vwin2048;
+  case 2048:
+    return vwin4096;
+#ifndef LIMIT_TO_64kHz
+  case 4096:
+    return vwin8192;
+#endif
+  default:
+    return(0);
+  }
+}
+
+/* pcm==0 indicates we just want the pending samples, no more */
+int vorbis_dsp_pcmout(vorbis_dsp_state *v,ogg_int16_t *pcm,int samples){
+  vorbis_info *vi=v->vi;
+  codec_setup_info *ci=(codec_setup_info *)vi->codec_setup;
+  if(v->out_begin>-1 && v->out_begin<v->out_end){
+    int n=v->out_end-v->out_begin;
+    if(pcm){
+      int i;
+      if(n>samples)n=samples;
+      for(i=0;i<vi->channels;i++)
+	mdct_unroll_lap(ci->blocksizes[0],ci->blocksizes[1],
+			v->lW,v->W,v->work[i],v->mdctright[i],
+			_vorbis_window(ci->blocksizes[0]>>1),
+			_vorbis_window(ci->blocksizes[1]>>1),
+			pcm+i,vi->channels,
+			v->out_begin,v->out_begin+n);
+    }
+    return(n);
+  }
+  return(0);
+}
+
+int vorbis_dsp_read(vorbis_dsp_state *v,int s){
+  if(s && v->out_begin+s>v->out_end)return(OV_EINVAL);
+  v->out_begin+=s;
+  return(0);
+}
+
+long vorbis_packet_blocksize(vorbis_info *vi,ogg_packet *op){
+  codec_setup_info     *ci=(codec_setup_info *)vi->codec_setup;
+  oggpack_buffer       opb;
+  int                  mode;
+  int modebits=0;
+  int v=ci->modes;
+ 
+  oggpack_readinit(&opb,op->packet);
+
+  /* Check the packet type */
+  if(oggpack_read(&opb,1)!=0){
+    /* Oops.  This is not an audio data packet */
+    return(OV_ENOTAUDIO);
+  }
+
+  while(v>1){
+    modebits++;
+    v>>=1;
+  }
+
+  /* read our mode and pre/post windowsize */
+  mode=oggpack_read(&opb,modebits);
+  if(mode==-1)return(OV_EBADPACKET);
+  return(ci->blocksizes[ci->mode_param[mode].blockflag]);
+}
+
+
+static int ilog(ogg_uint32_t v){
+  int ret=0;
+  if(v)--v;
+  while(v){
+    ret++;
+    v>>=1;
+  }
+  return(ret);
+}
+
+int vorbis_dsp_synthesis(vorbis_dsp_state *vd,ogg_packet *op,int decodep){
+  vorbis_info          *vi=vd->vi;
+  codec_setup_info     *ci=(codec_setup_info *)vi->codec_setup;
+  int                   mode,i;
+
+  oggpack_readinit(&vd->opb,op->packet);
+
+  /* Check the packet type */
+  if(oggpack_read(&vd->opb,1)!=0){
+    /* Oops.  This is not an audio data packet */
+    return OV_ENOTAUDIO ;
+  }
+
+  /* read our mode and pre/post windowsize */
+  mode=oggpack_read(&vd->opb,ilog(ci->modes));
+  if(mode==-1 || mode>=ci->modes) return OV_EBADPACKET;
+
+  /* shift information we still need from last window */
+  vd->lW=vd->W;
+  vd->W=ci->mode_param[mode].blockflag;
+  for(i=0;i<vi->channels;i++)
+    mdct_shift_right(ci->blocksizes[vd->lW],vd->work[i],vd->mdctright[i]);
+  
+  if(vd->W){
+    int temp;
+    oggpack_read(&vd->opb,1);
+    temp=oggpack_read(&vd->opb,1);
+    if(temp==-1) return OV_EBADPACKET;
+  }
+  
+  /* packet decode and portions of synthesis that rely on only this block */
+  if(decodep){
+    mapping_inverse(vd,ci->map_param+ci->mode_param[mode].mapping);
+
+    if(vd->out_begin==-1){
+      vd->out_begin=0;
+      vd->out_end=0;
+    }else{
+      vd->out_begin=0;
+      vd->out_end=ci->blocksizes[vd->lW]/4+ci->blocksizes[vd->W]/4;
+    }
+  }
+
+  /* track the frame number... This is for convenience, but also
+     making sure our last packet doesn't end with added padding.
+     
+     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 we're out of sequence, dump granpos tracking until we sync back up */
+  if(vd->sequence==-1 || vd->sequence+1 != op->packetno-3){
+    /* out of sequence; lose count */
+    vd->granulepos=-1;
+    vd->sample_count=-1;
+  }
+  
+  vd->sequence=op->packetno;
+  vd->sequence=vd->sequence-3;
+  
+  if(vd->sample_count==-1){
+    vd->sample_count=0;
+  }else{
+    vd->sample_count+=
+      ci->blocksizes[vd->lW]/4+ci->blocksizes[vd->W]/4;
+  }
+  
+  if(vd->granulepos==-1){
+    if(op->granulepos!=-1){ /* only set if we have a
+			       position to set to */
+      
+      vd->granulepos=op->granulepos;
+      
+      /* is this a short page? */
+      if(vd->sample_count>vd->granulepos){
+	/* corner case; if this is both the first and last audio page,
+	   then spec says the end is cut, not beginning */
+	if(op->e_o_s){
+	  /* trim the end */
+	  /* no preceeding 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 coun t, not short count */
+	  
+	  vd->out_end-=vd->sample_count-vd->granulepos;
+	}else{
+	  /* trim the beginning */
+	  vd->out_begin+=vd->sample_count-vd->granulepos;
+	  if(vd->out_begin>vd->out_end)
+	    vd->out_begin=vd->out_end;
+	}
+	
+      }
+      
+    }
+  }else{
+    vd->granulepos+=
+      ci->blocksizes[vd->lW]/4+ci->blocksizes[vd->W]/4;
+    if(op->granulepos!=-1 && vd->granulepos!=op->granulepos){
+      
+      if(vd->granulepos>op->granulepos){
+	long extra=vd->granulepos-op->granulepos;
+	
+	if(extra)
+	  if(op->e_o_s){
+	    /* partial last frame.  Strip the extra samples off */
+	    vd->out_end-=extra;
+	  } /* 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 } */
+      vd->granulepos=op->granulepos;
+    }
+  }
+
+  return(0);
+}