Mercurial > mplayer.hg
view libmpcodecs/ad_dvdpcm.c @ 30309:ddd809228aef
More format support in fmt-conversion.c: split RGB32 and BGR32
into ARGB, BGRA, ABGR, RGBA.
author | reimar |
---|---|
date | Sun, 17 Jan 2010 14:40:44 +0000 |
parents | 0f1b5b68af32 |
children | bbb6ebec87a0 |
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#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 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; }