view libmpcodecs/ad_libdv.c @ 11330:a974c00c779d

Removed temporary .cpp file used during the Matroska test. Updated the libebml and libmatroska requirements to at least v0.6.0 for both. There have been changes in the lacing code, and users WILL come and complain why mplayer, linked against older versions, will have issues playing newer files.
author mosu
date Thu, 30 Oct 2003 14:57:06 +0000
parents 28677d779205
children bcd805923554
line wrap: on
line source

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

#include "config.h"

#ifdef HAVE_LIBDV095

#include "img_format.h"

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

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

#include "ad_internal.h"

static ad_info_t info =
{
	"Raw DV Audio Decoder",
	"libdv",
	"Alexander Neundorf <neundorf@kde.org>",
	"http://libdv.sf.net",
	""
};

LIBAD_EXTERN(libdv)

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

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)
       printf("warning! audio framesize differs! read=%d  hdr=%d  \n",
           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;
}

#endif