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author | wm4 <wm4@nowhere> | 2012-11-05 17:02:04 +0100 |
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committer | wm4 <wm4@nowhere> | 2012-11-12 20:06:14 +0100 |
commit | d4bdd0473d6f43132257c9fb3848d829755167a3 (patch) | |
tree | 8021c2f7da1841393c8c832105e20cd527826d6c /audio/decode/ad_dvdpcm.c | |
parent | bd48deba77bd5582c5829d6fe73a7d2571088aba (diff) | |
download | mpv-d4bdd0473d6f43132257c9fb3848d829755167a3.tar.bz2 mpv-d4bdd0473d6f43132257c9fb3848d829755167a3.tar.xz |
Rename directories, move files (step 1 of 2) (does not compile)
Tis drops the silly lib prefixes, and attempts to organize the tree in
a more logical way. Make the top-level directory less cluttered as
well.
Renames the following directories:
libaf -> audio/filter
libao2 -> audio/out
libvo -> video/out
libmpdemux -> demux
Split libmpcodecs:
vf* -> video/filter
vd*, dec_video.* -> video/decode
mp_image*, img_format*, ... -> video/
ad*, dec_audio.* -> audio/decode
libaf/format.* is moved to audio/ - this is similar to how mp_image.*
is located in video/.
Move most top-level .c/.h files to core. (talloc.c/.h is left on top-
level, because it's external.) Park some of the more annoying files
in compat/. Some of these are relicts from the time mplayer used
ffmpeg internals.
sub/ is not split, because it's too much of a mess (subtitle code is
mixed with OSD display and rendering).
Maybe the organization of core is not ideal: it mixes playback core
(like mplayer.c) and utility helpers (like bstr.c/h). Should the need
arise, the playback core will be moved somewhere else, while core
contains all helper and common code.
Diffstat (limited to 'audio/decode/ad_dvdpcm.c')
-rw-r--r-- | audio/decode/ad_dvdpcm.c | 162 |
1 files changed, 162 insertions, 0 deletions
diff --git a/audio/decode/ad_dvdpcm.c b/audio/decode/ad_dvdpcm.c new file mode 100644 index 0000000000..41f6a1426d --- /dev/null +++ b/audio/decode/ad_dvdpcm.c @@ -0,0 +1,162 @@ +/* + * 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 "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) { + case 0: + sh->sample_format = AF_FORMAT_S16_BE; + sh->samplesize = 2; + break; + case 1: + mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Samples of this format are needed to improve support. Please contact the developers.\n"); + 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; +} |