view libmpcodecs/ad_libdv.c @ 16946:47c5e9846cd3

ultra simple&slow pp filter, yes yet another spp like filter :) this one does actually compress&decompress the video at various shifts with lavc while the other spp filters are doing optimized intra only filtering limitations: mpeg4 is hardcoded, all options too, pretty trivial to change though, even filtering with non dct codecs like snow could be tried ... the qscale/qp is only taken fron the first MB of each image and then used for the whole image (would needs some small changes to lavc to let the user set the qscales for the mbs themselfs but iam to lazy ...) this needs ALOT of cpu time and memory especially at uspp=8 ...

#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 <[e-mail]>",
	"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