view libmpcodecs/ad_dvdpcm.c @ 22104:5ecca25377e0

Support for AOPLAY_FINAL_CHUNK in ao_sdl.c

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

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

static ad_info_t info = 
{
	"Uncompressed DVD/VOB LPCM audio decoder",
	"dvdpcm",
	"Nick Kurshev",
	"A'rpi",
	""
};

LIBAD_EXTERN(dvdpcm)

static int init(sh_audio_t *sh)
{
/* DVD PCM Audio:*/
    sh->i_bps = 0;
    if(sh->codecdata_len==3){
	// we have LPCM header:
	unsigned char h=sh->codecdata[1];
	sh->channels=1+(h&7);
	switch((h>>4)&3){
	case 0: sh->samplerate=48000;break;
	case 1: sh->samplerate=96000;break;
	case 2: sh->samplerate=44100;break;
	case 3: sh->samplerate=32000;break;
	}
	switch ((h >> 6) & 3) {
	  case 0:
	    sh->sample_format = AF_FORMAT_S16_BE;
	    sh->samplesize = 2;
	    break;
	  case 1:
	    mp_msg(MSGT_DECAUDIO, MSGL_INFO, MSGTR_SamplesWanted);
	    sh->i_bps = sh->channels * sh->samplerate * 5 / 2;
	  case 2: 
	    sh->sample_format = AF_FORMAT_S24_BE;
	    sh->samplesize = 3;
	    break;
	  default:
	    sh->sample_format = AF_FORMAT_S16_BE;
	    sh->samplesize = 2;
	}
    } else {
	// use defaults:
	sh->channels=2;
	sh->samplerate=48000;
	sh->sample_format = AF_FORMAT_S16_BE;
	sh->samplesize = 2;
    }
    if (!sh->i_bps)
    sh->i_bps = sh->samplesize * sh->channels * sh->samplerate;
    return 1;
}

static int preinit(sh_audio_t *sh)
{
  sh->audio_out_minsize=2048;
  return 1;
}

static void uninit(sh_audio_t *sh)
{
}

static int control(sh_audio_t *sh,int cmd,void* arg, ...)
{
  int skip;
    switch(cmd)
    {
      case ADCTRL_SKIP_FRAME:
	skip=sh->i_bps/16;
	skip=skip&(~3);
	demux_read_data(sh->ds,NULL,skip);
	return CONTROL_TRUE;
    }
  return CONTROL_UNKNOWN;
}

static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
  int j,len;
  if (sh_audio->samplesize == 3) {
    if (((sh_audio->codecdata[1] >> 6) & 3) == 1) {
      // 20 bit
      // not sure if the "& 0xf0" and "<< 4" are the right way around
      // can somebody clarify?
      for (j = 0; j < minlen; j += 12) {
        char tmp[10];
        len = demux_read_data(sh_audio->ds, tmp, 10);
        if (len < 10) break;
        // first sample
        buf[j + 0] = tmp[0];
        buf[j + 1] = tmp[1];
        buf[j + 2] = tmp[8] & 0xf0;
        // second sample
        buf[j + 3] = tmp[2];
        buf[j + 4] = tmp[3];
        buf[j + 5] = tmp[8] << 4;
        // third sample
        buf[j + 6] = tmp[4];
        buf[j + 7] = tmp[5];
        buf[j + 8] = tmp[9] & 0xf0;
        // fourth sample
        buf[j + 9] = tmp[6];
        buf[j + 10] = tmp[7];
        buf[j + 11] = tmp[9] << 4;
      }
      len = j;
    } else {
      // 24 bit
      for (j = 0; j < minlen; j += 12) {
        char tmp[12];
        len = demux_read_data(sh_audio->ds, tmp, 12);
        if (len < 12) break;
        // first sample
        buf[j + 0] = tmp[0];
        buf[j + 1] = tmp[1];
        buf[j + 2] = tmp[8];
        // second sample
        buf[j + 3] = tmp[2];
        buf[j + 4] = tmp[3];
        buf[j + 5] = tmp[9];
        // third sample
        buf[j + 6] = tmp[4];
        buf[j + 7] = tmp[5];
        buf[j + 8] = tmp[10];
        // fourth sample
        buf[j + 9] = tmp[6];
        buf[j + 10] = tmp[7];
        buf[j + 11] = tmp[11];
      }
      len = j;
    }
  } else 
  len=demux_read_data(sh_audio->ds,buf,(minlen+3)&(~3));
  return len;
}