view libmpcodecs/ad_libdv.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.

#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <sys/types.h>
#include <unistd.h>
#include <math.h>

#include "config.h"
#include "mp_msg.h"
#include "help_mp.h"

#include "img_format.h"

#include <libdv/dv.h>
#include <libdv/dv_types.h>

#include "stream/stream.h"
#include "libmpdemux/demuxer.h"
#include "libmpdemux/stheader.h"

#include "ad_internal.h"

static ad_info_t info =
{
	"Raw DV Audio Decoder",
	"libdv",
	"Alexander Neundorf <[e-mail]>",
	"http://libdv.sf.net",
	""
};

LIBAD_EXTERN(libdv)

// defined in vd_libdv.c:
dv_decoder_t*  init_global_rawdv_decoder(void);

static int preinit(sh_audio_t *sh_audio)
{
  sh_audio->audio_out_minsize=4*DV_AUDIO_MAX_SAMPLES*2;
  return 1;
}

static int16_t *audioBuffers[4]={NULL,NULL,NULL,NULL};

static int init(sh_audio_t *sh)
{
  int i;
  WAVEFORMATEX *h=sh->wf;

  if(!h) return 0;
   
  sh->i_bps=h->nAvgBytesPerSec;
  sh->channels=h->nChannels;
  sh->samplerate=h->nSamplesPerSec;
  sh->samplesize=(h->wBitsPerSample+7)/8;

  sh->context=init_global_rawdv_decoder();

  for (i=0; i < 4; i++)
    audioBuffers[i] = malloc(2*DV_AUDIO_MAX_SAMPLES);

  return 1;
}

static void uninit(sh_audio_t *sh_audio)
{
  int i;
  for (i=0; i < 4; i++)
    free(audioBuffers[i]);
}

static int control(sh_audio_t *sh,int cmd,void* arg, ...)
{
    // TODO!!!
  return CONTROL_UNKNOWN;
}

static int decode_audio(sh_audio_t *audio, unsigned char *buf, int minlen, int maxlen)
{
   int len=0;
   dv_decoder_t* decoder=audio->context;  //global_rawdv_decoder;
   unsigned char* dv_audio_frame=NULL;
   int xx=ds_get_packet(audio->ds,&dv_audio_frame);
   if(xx<=0 || !dv_audio_frame) return 0; // EOF?

   dv_parse_header(decoder, dv_audio_frame);
   
   if(xx!=decoder->frame_size)
       mp_msg(MSGT_GLOBAL,MSGL_WARN,MSGTR_MPCODECS_AudioFramesizeDiffers,
           xx, decoder->frame_size);

   if (dv_decode_full_audio(decoder, dv_audio_frame,(int16_t**) audioBuffers))
   {
      /* Interleave the audio into a single buffer */
      int i=0;
      int16_t *bufP=(int16_t*)buf;
      
//      printf("samples=%d/%d  chans=%d  mem=%d  \n",decoder->audio->samples_this_frame,DV_AUDIO_MAX_SAMPLES,
//          decoder->audio->num_channels, decoder->audio->samples_this_frame*decoder->audio->num_channels*2);

//   return (44100/30)*4;

      for (i=0; i < decoder->audio->samples_this_frame; i++)
      {
         int ch;
         for (ch=0; ch < decoder->audio->num_channels; ch++)
            bufP[len++] = audioBuffers[ch][i];
      }
   }
   return len*2;
}