Mercurial > mplayer.hg
changeset 13595:f67d20c01332
Support for 24 bit and 20 bit LPCM (simple and slow :-( )
author | reimar |
---|---|
date | Sat, 09 Oct 2004 20:04:27 +0000 |
parents | acfe017b6195 |
children | 3a2e1e6ae834 |
files | help/help_mp-en.h libmpcodecs/ad_dvdpcm.c |
diffstat | 2 files changed, 62 insertions(+), 0 deletions(-) [+] |
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--- a/help/help_mp-en.h Sat Oct 09 13:13:28 2004 +0000 +++ b/help/help_mp-en.h Sat Oct 09 20:04:27 2004 +0000 @@ -53,6 +53,8 @@ "\n"; #endif +#define MSGTR_SamplesWanted "Samples of this format are needed to improve support. Please contact the developers.\n" + // ========================= MPlayer messages =========================== // mplayer.c:
--- a/libmpcodecs/ad_dvdpcm.c Sat Oct 09 13:13:28 2004 +0000 +++ b/libmpcodecs/ad_dvdpcm.c Sat Oct 09 20:04:27 2004 +0000 @@ -3,6 +3,8 @@ #include <unistd.h> #include "config.h" +#include "mp_msg.h" +#include "help_mp.h" #include "ad_internal.h" #include "../libaf/af_format.h" @@ -20,6 +22,7 @@ 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]; @@ -35,6 +38,9 @@ sh->sample_format = AFMT_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 = AFMT_AF_FLAGS | AF_FORMAT_I | AF_FORMAT_BE | AF_FORMAT_US; @@ -51,6 +57,7 @@ sh->sample_format = AFMT_S16_BE; sh->samplesize = 2; } + if (!sh->i_bps) sh->i_bps = sh->samplesize * sh->channels * sh->samplerate; return 1; } @@ -82,6 +89,59 @@ 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; }