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mpv/audio/decode/ad_dvdpcm.c
wm4 4873b32c59 Rename directories, move files (step 2 of 2)
Finish renaming directories and moving files. Adjust all include
statements to make the previous commit compile.

The two commits are separate, because git is bad at tracking renames
and content changes at the same time.

Also take this as an opportunity to remove the separation between
"common" and "mplayer" sources in the Makefile. ("common" used to be
shared between mplayer and mencoder.)
2012-11-12 20:08:18 +01:00

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 "core/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;
}