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mpv/libmpcodecs/ad_dvdpcm.c
wm4 c8154630bf ad_dvdpcm: add back PCM decoder for DVD
This is needed by demux_mpg (and possibly by demux_ts) for PCM playback.
The decoder does the mapping from MPEG headers to the actual PCM format,
and also unpacks sample data for 20/24 bit formats.
2012-09-18 21:08:14 +02:00

163 lines
4.1 KiB
C

/*
* 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 "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_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Samples of this format are needed to improve support. Please contact the developers.\n");
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;
}