view libmpcodecs/ad_dvdpcm.c @ 17283:4fea216307d2

Partial support for QuickTime sound atom version 2. This doesn't add support for parsing the sound atom itself, but does recognize the different offset at which the ESDS atom starts. Also, this patch supports "3" in the channels field, which indicates 6-channel (5.1) audio. For more information, see this mail: From: Corey Hickey <[e-mail]> To: [e-mail] Date: Wed, 28 Dec 2005 23:42:46 -0800 Subject: [PATCH] (partially) support QuickTime sound atom version 2

#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;
}