Mercurial > mplayer.hg
view libmpcodecs/ad_dvdpcm.c @ 24892:80180dc13565
Change decode_audio() interface
Rewrite decode_audio to better deal with filters that handle input in
large blocks. It now always places output in sh_audio->a_out_buffer
(which was always given as a parameter before) and reallocates the
buffer if needed. After the changes filters can return arbitrarily
large blocks of data without some of it being lost. The new version
also allows simplifying some code.
author | uau |
---|---|
date | Thu, 01 Nov 2007 06:52:19 +0000 |
parents | 815f03b7cee5 |
children | 0f1b5b68af32 |
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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; }