Mercurial > mplayer.hg
view libmpcodecs/ad_dvdpcm.c @ 36963:e539d330c7be
Remove unnecessary bounds checks in Win32 GUI.
The checks that the rendered potmeter button
doesn't exceed the bounds is not necessary as
the item value is already limited within the
range of 0 to 100.
Patch by Hans-Dieter Kosch, hdkosch kabelbw de.
author | ib |
---|---|
date | Mon, 24 Mar 2014 12:52:01 +0000 |
parents | e648bb154916 |
children |
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/* * 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) { default: 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; /* Fallthrough, 20 bit will be output as 24 bit */ case 2: sh->sample_format = AF_FORMAT_S24_BE; sh->samplesize = 3; break; } } 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; }