mirror of
https://github.com/mpv-player/mpv
synced 2024-12-23 15:22:09 +00:00
00323c06e2
Remove the help/ subdirectory, configure code to create toplevel help_mp.h, and all the '#include "help_mp.h"' lines from .c files.
163 lines
4.1 KiB
C
163 lines
4.1 KiB
C
/*
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* This file is part of MPlayer.
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*
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* MPlayer is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* MPlayer is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with MPlayer; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <unistd.h>
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#include "config.h"
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#include "mp_msg.h"
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#include "ad_internal.h"
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static const ad_info_t info =
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{
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"Uncompressed DVD/VOB LPCM audio decoder",
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"dvdpcm",
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"Nick Kurshev",
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"A'rpi",
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""
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};
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LIBAD_EXTERN(dvdpcm)
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static int init(sh_audio_t *sh)
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{
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/* DVD PCM Audio:*/
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sh->i_bps = 0;
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if(sh->codecdata_len==3){
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// we have LPCM header:
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unsigned char h=sh->codecdata[1];
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sh->channels=1+(h&7);
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switch((h>>4)&3){
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case 0: sh->samplerate=48000;break;
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case 1: sh->samplerate=96000;break;
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case 2: sh->samplerate=44100;break;
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case 3: sh->samplerate=32000;break;
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}
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switch ((h >> 6) & 3) {
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case 0:
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sh->sample_format = AF_FORMAT_S16_BE;
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sh->samplesize = 2;
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break;
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case 1:
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mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Samples of this format are needed to improve support. Please contact the developers.\n");
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sh->i_bps = sh->channels * sh->samplerate * 5 / 2;
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case 2:
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sh->sample_format = AF_FORMAT_S24_BE;
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sh->samplesize = 3;
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break;
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default:
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sh->sample_format = AF_FORMAT_S16_BE;
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sh->samplesize = 2;
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}
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} else {
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// use defaults:
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sh->channels=2;
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sh->samplerate=48000;
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sh->sample_format = AF_FORMAT_S16_BE;
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sh->samplesize = 2;
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}
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if (!sh->i_bps)
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sh->i_bps = sh->samplesize * sh->channels * sh->samplerate;
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return 1;
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}
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static int preinit(sh_audio_t *sh)
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{
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sh->audio_out_minsize=2048;
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return 1;
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}
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static void uninit(sh_audio_t *sh)
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{
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}
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static int control(sh_audio_t *sh,int cmd,void* arg, ...)
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{
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int skip;
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switch(cmd)
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{
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case ADCTRL_SKIP_FRAME:
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skip=sh->i_bps/16;
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skip=skip&(~3);
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demux_read_data(sh->ds,NULL,skip);
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return CONTROL_TRUE;
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}
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return CONTROL_UNKNOWN;
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}
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static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
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{
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int j,len;
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if (sh_audio->samplesize == 3) {
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if (((sh_audio->codecdata[1] >> 6) & 3) == 1) {
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// 20 bit
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// not sure if the "& 0xf0" and "<< 4" are the right way around
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// can somebody clarify?
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for (j = 0; j < minlen; j += 12) {
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char tmp[10];
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len = demux_read_data(sh_audio->ds, tmp, 10);
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if (len < 10) break;
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// first sample
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buf[j + 0] = tmp[0];
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buf[j + 1] = tmp[1];
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buf[j + 2] = tmp[8] & 0xf0;
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// second sample
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buf[j + 3] = tmp[2];
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buf[j + 4] = tmp[3];
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buf[j + 5] = tmp[8] << 4;
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// third sample
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buf[j + 6] = tmp[4];
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buf[j + 7] = tmp[5];
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buf[j + 8] = tmp[9] & 0xf0;
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// fourth sample
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buf[j + 9] = tmp[6];
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buf[j + 10] = tmp[7];
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buf[j + 11] = tmp[9] << 4;
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}
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len = j;
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} else {
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// 24 bit
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for (j = 0; j < minlen; j += 12) {
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char tmp[12];
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len = demux_read_data(sh_audio->ds, tmp, 12);
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if (len < 12) break;
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// first sample
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buf[j + 0] = tmp[0];
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buf[j + 1] = tmp[1];
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buf[j + 2] = tmp[8];
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// second sample
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buf[j + 3] = tmp[2];
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buf[j + 4] = tmp[3];
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buf[j + 5] = tmp[9];
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// third sample
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buf[j + 6] = tmp[4];
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buf[j + 7] = tmp[5];
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buf[j + 8] = tmp[10];
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// fourth sample
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buf[j + 9] = tmp[6];
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buf[j + 10] = tmp[7];
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buf[j + 11] = tmp[11];
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}
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len = j;
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}
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} else
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len=demux_read_data(sh_audio->ds,buf,(minlen+3)&(~3));
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return len;
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}
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