view libmpcodecs/ad_libvorbis.c @ 22616:09dc129234a0

Matroska seeking fixes If a relative seek forward went past the last index position the Matroska demuxer did not seek to any index position. It did however set the mkv_d->skip_to_timecode variable which meant that the next fill_buffer() call would read from the current position until the target position (probably the end of the file). Fix this by changing the code to seek to the last index position if that is between the current and target positions. Also change backwards relative seek to accept an exactly matching index position (<= vs <) and reorganize the seeking conditionals to allow making the above change without turning the code into a complete mess.

#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <stdarg.h>
#include <math.h>

#include "config.h"
#include "ad_internal.h"

static ad_info_t info = 
{
	"Ogg/Vorbis audio decoder",
	"libvorbis",
	"Felix Buenemann, A'rpi",
	"libvorbis",
	""
};

LIBAD_EXTERN(libvorbis)

#ifdef TREMOR
#include <tremor/ivorbiscodec.h>
#else
#include <vorbis/codec.h>
#endif

// This struct is also defined in demux_ogg.c => common header ?
typedef struct ov_struct_st {
  vorbis_info      vi; /* struct that stores all the static vorbis bitstream
			  settings */
  vorbis_comment   vc; /* struct that stores all the bitstream user comments */
  vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
  vorbis_block     vb; /* local working space for packet->PCM decode */
  float            rg_scale; /* replaygain scale */
#ifdef TREMOR
  int              rg_scale_int;
#endif
} ov_struct_t;

static int read_vorbis_comment( char* ptr, const char* comment, const char* format, ... ) {
  va_list va;
  int clen, ret;

  va_start( va, format );
  clen = strlen( comment );
  ret = strncasecmp( ptr, comment, clen) == 0 ? vsscanf( ptr+clen, format, va ) : 0;
  va_end( va );

  return ret;
}

static int preinit(sh_audio_t *sh)
{
  sh->audio_out_minsize=1024*4; // 1024 samples/frame
  return 1;
}

static int init(sh_audio_t *sh)
{
  unsigned int offset, i, length, hsizes[3];
  void *headers[3];
  unsigned char* extradata;
  ogg_packet op;
  vorbis_comment vc;
  struct ov_struct_st *ov;
#define ERROR() { \
    vorbis_comment_clear(&vc); \
    vorbis_info_clear(&ov->vi); \
    free(ov); \
    return 0; \
  }

  /// Init the decoder with the 3 header packets
  ov = malloc(sizeof(struct ov_struct_st));
  vorbis_info_init(&ov->vi);
  vorbis_comment_init(&vc);

  if(! sh->wf) {
    mp_msg(MSGT_DECAUDIO,MSGL_ERR,"ad_vorbis, extradata seems to be absent! exit\n");
    ERROR();
  }

  if(! sh->wf->cbSize) {
    mp_msg(MSGT_DECAUDIO,MSGL_ERR,"ad_vorbis, extradata seems to be absent!, exit\n");
    ERROR();
  }

  mp_msg(MSGT_DECAUDIO,MSGL_V,"ad_vorbis, extradata seems is %d bytes long\n", sh->wf->cbSize);
  extradata = (char*) (sh->wf+1);
  if(!extradata) {
    mp_msg(MSGT_DECAUDIO,MSGL_ERR,"ad_vorbis, extradata seems to be NULL!, exit\n");
    ERROR();
  }

  if(*extradata != 2) {
    mp_msg (MSGT_DEMUX, MSGL_WARN, "ad_vorbis: Vorbis track does not contain valid headers.\n");
    ERROR();
  }

  offset = 1;
  for (i=0; i < 2; i++) {
    length = 0;
    while ((extradata[offset] == (unsigned char) 0xFF) && length < sh->wf->cbSize) {
      length += 255;
      offset++;
    }
    if(offset >= (sh->wf->cbSize - 1)) {
      mp_msg (MSGT_DEMUX, MSGL_WARN, "ad_vorbis: Vorbis track does not contain valid headers.\n");
      ERROR();
    }
    length += extradata[offset];
    offset++;
    mp_msg (MSGT_DEMUX, MSGL_V, "ad_vorbis, offset: %u, length: %u\n", offset, length);
    hsizes[i] = length;
  }

  headers[0] = &extradata[offset];
  headers[1] = &extradata[offset + hsizes[0]];
  headers[2] = &extradata[offset + hsizes[0] + hsizes[1]];
  hsizes[2] = sh->wf->cbSize - offset - hsizes[0] - hsizes[1];
  mp_msg (MSGT_DEMUX, MSGL_V, "ad_vorbis, header sizes: %d %d %d\n", hsizes[0], hsizes[1], hsizes[2]);

  for(i=0; i<3; i++) {
    op.bytes = hsizes[i];
    op.packet = headers[i];
    op.b_o_s  = (i == 0);
    if(vorbis_synthesis_headerin(&ov->vi,&vc,&op) <0) {
      mp_msg(MSGT_DECAUDIO,MSGL_ERR,"OggVorbis: header n. %d broken! len=%ld\n", i, op.bytes);
      ERROR();
    }
    if(i == 2) {
      float rg_gain=0.f, rg_peak=0.f;
    char **ptr=vc.user_comments;
    while(*ptr){
      mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbisComment: %s\n",*ptr);
      /* replaygain */
      read_vorbis_comment( *ptr, "replaygain_album_gain=", "%f", &rg_gain );
      read_vorbis_comment( *ptr, "rg_audiophile=", "%f", &rg_gain );
      if( !rg_gain ) {
	read_vorbis_comment( *ptr, "replaygain_track_gain=", "%f", &rg_gain );
	read_vorbis_comment( *ptr, "rg_radio=", "%f", &rg_gain );
      }
      read_vorbis_comment( *ptr, "replaygain_album_peak=", "%f", &rg_peak );
      if( !rg_peak ) {
	read_vorbis_comment( *ptr, "replaygain_track_peak=", "%f", &rg_peak );
	read_vorbis_comment( *ptr, "rg_peak=", "%f", &rg_peak );
      }
      ++ptr;
    }
    /* replaygain: scale */
    if(!rg_gain)
      ov->rg_scale = 1.f; /* just in case pow() isn't standard-conformant */
    else
      ov->rg_scale = pow(10.f, rg_gain/20);
    /* replaygain: anticlip */
    if(ov->rg_scale * rg_peak > 1.f)
      ov->rg_scale = 1.f / rg_peak;
    /* replaygain: security */
    if(ov->rg_scale > 15.) 
      ov->rg_scale = 15.;
#ifdef TREMOR
    ov->rg_scale_int = (int)(ov->rg_scale*64.f);
#endif
    mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Bitstream is %d channel%s, %dHz, %dbit/s %cBR\n",(int)ov->vi.channels,ov->vi.channels>1?"s":"",(int)ov->vi.rate,(int)ov->vi.bitrate_nominal,
	(ov->vi.bitrate_lower!=ov->vi.bitrate_nominal)||(ov->vi.bitrate_upper!=ov->vi.bitrate_nominal)?'V':'C');
    if(rg_gain || rg_peak)
      mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Gain = %+.2f dB, Peak = %.4f, Scale = %.2f\n", rg_gain, rg_peak, ov->rg_scale);
    mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Encoded by: %s\n",vc.vendor);
    }
  }

  vorbis_comment_clear(&vc);

//  printf("lower=%d  upper=%d  \n",(int)ov->vi.bitrate_lower,(int)ov->vi.bitrate_upper);

  // Setup the decoder
  sh->channels=ov->vi.channels; 
  sh->samplerate=ov->vi.rate;
  sh->samplesize=2;
  // assume 128kbit if bitrate not specified in the header
  sh->i_bps=((ov->vi.bitrate_nominal>0) ? ov->vi.bitrate_nominal : 128000)/8;
  sh->context = ov;

  /// Finish the decoder init
  vorbis_synthesis_init(&ov->vd,&ov->vi);
  vorbis_block_init(&ov->vd,&ov->vb);
  mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Init OK!\n");

  return 1;
}

static void uninit(sh_audio_t *sh)
{
  struct ov_struct_st *ov = sh->context;
  vorbis_dsp_clear(&ov->vd);
  vorbis_block_clear(&ov->vb);
  vorbis_info_clear(&ov->vi);
  free(ov);
}

static int control(sh_audio_t *sh,int cmd,void* arg, ...)
{
    switch(cmd)
    {
#if 0      
      case ADCTRL_RESYNC_STREAM:
	  return CONTROL_TRUE;
      case ADCTRL_SKIP_FRAME:
	  return CONTROL_TRUE;
#endif
    }
  return CONTROL_UNKNOWN;
}

static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen)
{
        int len = 0;
        int samples;
#ifdef TREMOR
        ogg_int32_t **pcm;
#else
        float **pcm;
#endif
        float scale;
        struct ov_struct_st *ov = sh->context;
	while(len < minlen) {
	  while((samples=vorbis_synthesis_pcmout(&ov->vd,&pcm))<=0){
	    ogg_packet op;
	    double pts;
	    memset(&op,0,sizeof(op)); //op.b_o_s = op.e_o_s = 0;
	    op.bytes = ds_get_packet_pts(sh->ds,&op.packet, &pts);
	    if(op.bytes<=0) break;
	    if (pts != MP_NOPTS_VALUE) {
		sh->pts = pts;
		sh->pts_bytes = 0;
	    }
	    if(vorbis_synthesis(&ov->vb,&op)==0) /* test for success! */
	      vorbis_synthesis_blockin(&ov->vd,&ov->vb);
	  }
	  if(samples<=0) break; // error/EOF
	  while(samples>0){
	    int i,j;
	    int clipflag=0;
	    int convsize=(maxlen-len)/(2*ov->vi.channels); // max size!
	    int bout=((samples<convsize)?samples:convsize);
	  
	    if(bout<=0) break; // no buffer space

	    /* convert floats to 16 bit signed ints (host order) and
	       interleave */
#ifdef TREMOR
           if (ov->rg_scale_int == 64) {
	    for(i=0;i<ov->vi.channels;i++){
	      ogg_int16_t *convbuffer=(ogg_int16_t *)(&buf[len]);
	      ogg_int16_t *ptr=convbuffer+i;
	      ogg_int32_t  *mono=pcm[i];
	      for(j=0;j<bout;j++){
		int val=mono[j]>>9;
		/* might as well guard against clipping */
		if(val>32767){
		  val=32767;
		  clipflag=1;
		}
		if(val<-32768){
		  val=-32768;
		  clipflag=1;
		}
		*ptr=val;
		ptr+=ov->vi.channels;
	      }
	    }
	   } else
#endif /* TREMOR */
	   {
#ifndef TREMOR
            scale = 32767.f * ov->rg_scale;
#endif
	    for(i=0;i<ov->vi.channels;i++){
	      ogg_int16_t *convbuffer=(ogg_int16_t *)(&buf[len]);
	      ogg_int16_t *ptr=convbuffer+i;
#ifdef TREMOR
	      ogg_int32_t  *mono=pcm[i];
	      for(j=0;j<bout;j++){
		int val=(mono[j]*ov->rg_scale_int)>>(9+6);
#else
	      float  *mono=pcm[i];
	      for(j=0;j<bout;j++){
		int val=mono[j]*scale;
		/* might as well guard against clipping */
		if(val>32767){
		  val=32767;
		  clipflag=1;
		}
		if(val<-32768){
		  val=-32768;
		  clipflag=1;
		}
#endif    /* TREMOR */
		*ptr=val;
		ptr+=ov->vi.channels;
	      }
	    }
	   }
		
	    if(clipflag)
	      mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"Clipping in frame %ld\n",(long)(ov->vd.sequence));
	    len+=2*ov->vi.channels*bout;
	    sh->pts_bytes += 2*ov->vi.channels*bout;
	    mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[decoded: %d / %d ]\n",bout,samples);
	    samples-=bout;
	    vorbis_synthesis_read(&ov->vd,bout); /* tell libvorbis how
						    many samples we
						    actually consumed */
	  } //while(samples>0)
//          if (!samples) break; // why? how?
	}



  return len;
}