view libmpcodecs/ad_dvdpcm.c @ 34175:a345e7162d0a

Move TranslateFilename() to util/string.c. Now that the Win32 GUI uses symbolic constants for its messages, the code of TranslateFilename() both GUIs use is almost identical. So, share the code.
author ib
date Wed, 26 Oct 2011 15:14:06 +0000
parents cc27da5d7286
children e648bb154916
line wrap: on
line source

/*
 * This file is part of MPlayer.
 *
 * MPlayer is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * MPlayer is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License along
 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
 */

#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 const 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;
}