[Guybrush]

audio seems desynched for mov files. at least one,
http://movies.apple.com/movies/fox/avp/a...e_m320.mov
but it seems slightly with others too

i noticed this too, but original mplayer does this as well.
m480 files always play correct tho.
it's probably because of ffmpeg's svq3 decoding, because if i decode the video with quicktimeessentials.qtx it doenst go out of sync.

[ezar2003]

in codecs.conf have:

audiocodec flac
info "free lossless audio codec"
status untested
comment "using libmpflac or libflac"
format 0x43614c66
driver flac
dll "libmpflac"

audiocodec ffflac
info "ffmpeg flac audio decoder"
status working
format 0x43614c66
driver ffmpeg
dll "flac"

remove the first part.
only you let this:

audiocodec ffflac
info "ffmpeg flac audio decoder"
status working
format 0x43614c66
driver ffmpeg
dll "flac"

the "libmpflac" doesn't need.

[ezar2003]

@guybrush: for dts-wav decoder is need this code.

add in demux_audio.c (warning: i add [ i] (some spaces) because it detected as italic)

   demuxer->movi_end = s->end_pos;
//    printf("wav: %x .. %x\n",(int)demuxer->movi_start,(int)demuxer->movi_end);

   // check if it contains dts audio
   if((w->wformattag == 0x01) && (w->nchannels == 2) && (w->nsamplespersec == 44100)) {
unsigned char buf[16384]; // vlc uses 16384*4 (4 dts frames)
unsigned int i;
stream_read(s, buf, sizeof(buf));
for (i = 0; i < sizeof(buf); i += 2) {
   // dts, 14 bit, le
   if((buf[ i] == 0xff) && (buf[i+1] == 0x1f) && (buf[i+2] == 0x00) &&
      (buf[i+3] == 0xe8) && ((buf[i+4] & 0xfe) == 0xf0) && (buf[i+5] == 0x07)) {
sh_audio->format = 0x2001;
mp_msg(msgt_demux,msgl_v,"[demux_audio] dts audio in wav, 14 bit, le\n");
break;
   }
   // dts, 14 bit, be
   if((buf[ i] == 0x1f) && (buf[i+1] == 0xff) && (buf[i+2] == 0xe8) &&
      (buf[i+3] == 0x00) && (buf[i+4] == 0x07) && ((buf[i+5] & 0xfe) == 0xf0)) {
sh_audio->format = 0x2001;
mp_msg(msgt_demux,msgl_v,"[demux_audio] dts audio in wav, 14 bit, be\n");
break;
   }
   // dts, 16 bit, be
   if((buf[ i] == 0x7f) && (buf[i+1] == 0xfe) && (buf[i+2] == 0x80) &&
      (buf[i+3] == 0x01)) {
sh_audio->format = 0x2001;
mp_msg(msgt_demux,msgl_v,"[demux_audio] dts audio in wav, 16 bit, be\n");
break;
   }
   // dts, 16 bit, le
   if((buf[ i] == 0xfe) && (buf[i+1] == 0x7f) && (buf[i+2] == 0x01) &&
      (buf[i+3] == 0x80)) {
sh_audio->format = 0x2001;
mp_msg(msgt_demux,msgl_v,"[demux_audio] dts audio in wav, 16 bit, le\n");
break;
   }
}
if (sh_audio->format == 0x2001)
   mp_msg(msgt_demux,msgl_dbg2,"[demux_audio] dts sync offset = %u\n", i);

   }
   stream_seek(s,demuxer->movi_start);
 } break;

you can download sources from here: mplayer_source_2004_08_20
and precompiled mplayer: mplayer_20040820

[pike]

[i]

[mvoosten]

dts wav's play really nice! though downmixed to stereo i really like it! thanks guys! can't wait for mutichannel support!!!!

[ezar2003]

@mvoosten: if you have a dts decoder, it decode in multichannel, if sound is multichannel and 48khz. you need last xbmc:

 - 20-08-2004 changed: passthrough of ac3/dts is always used unless samplerate != 48khz or output to all channels is enabled, and the source is stereo.

if in info button you see: 'dts passhrough' your decoder decode sound. no see channels in info!!!

@jmarshall: isn't it?

[jmarshall]

dts .wav files will not pass through directly as they are not 48khz. they will:

1. be decoded (using lib-dts)
2. be optionally upsampled to 48khz
3. be output as either ac3 or pcm - not sure, as i haven't tested it.

[chadoe]

a few dts (.wav) samples can be found here: http://www.sr.se/multikanal/english/e_index.stm

[ezar2003]

i test it now!
thank you charly!
the only file that i have is 'sound test - starwars' :kickass:

[ezar2003]

ok.
i test with dts (14 mb)

21-08-2004 21:09:30 debug   msg:mplayer_open_file(smb://xxxxxxx;xxxxxxx:[email protected]/storage/samples/dts/surroundtest_011212.wav)
21-08-2004 21:09:30 info stating file smb://xxxxxxx;xxxxxxx:[email protected]/storage/samples/dts/surroundtest_011212.wav.conf
21-08-2004 21:09:30 debug   msg:get_path('surroundtest_011212.wav.conf') -> 'q:\mplayer\surroundtest_011212.wav.conf'
21-08-2004 21:09:30 info stating file q:\mplayer\surroundtest_011212.wav.conf
21-08-2004 21:09:30 debug   msg:
21-08-2004 21:09:30 debug   msg:playing smb://xxxxxxx;xxxxxxx:[email protected]/storage/samples/dts/surroundtest_011212.wav.
21-08-2004 21:09:30 debug   msg:auto open z:\subtitle
21-08-2004 21:09:30 debug   msg:filename for url is now smb://xxxxxxx;xxxxxxx:[email protected]/storage/samples/dts/surroundtest_011212.wav
21-08-2004 21:09:30 debug   msg:protocolmb
21-08-2004 21:09:30 debug   msg:use standard file for protocolmb
21-08-2004 21:09:30 debug   msg:winsock2 init: 0
21-08-2004 21:09:30 debug   msg:[file] file size is 22347820 bytes
21-08-2004 21:09:30 debug   msgtream: [file] smb://xxxxxxx;xxxxxxx:[email protected]/storage/samples/dts/surroundtest_011212.wav
21-08-2004 21:09:30 debug   msgtream: description: file
21-08-2004 21:09:30 debug   msgtream: author: albeu
21-08-2004 21:09:30 debug   msgtream: comment: based on the code from (probably arpi)
21-08-2004 21:09:30 debug   msg:cache_pre_init: 0 [0] 0  pre:0  eof:0
...
21-08-2004 21:09:30 debug   msg:==> found audio stream: 0
21-08-2004 21:09:30 debug   msg:format tag: 1 (0x1)
21-08-2004 21:09:30 debug   msg:channels: 2
21-08-2004 21:09:30 debug   msgamplerate: 44100
21-08-2004 21:09:30 debug   msg:avg byte/sec: 176400
21-08-2004 21:09:30 debug   msg:block align: 4
21-08-2004 21:09:30 debug   msg:bits/sample: 16
21-08-2004 21:09:30 debug   msg:cbsize: 0
21-08-2004 21:09:30 debug   msg:[demux_audio] dts audio in wav, 14 bit, le

detected dts packet but it decode by software

21-08-2004 21:09:30 debug   msg:demux_audio: audio data 0x2c - 0x0
21-08-2004 21:09:30 debug   msg:audio file detected.
21-08-2004 21:09:30 debug   msg:==========================================================================
21-08-2004 21:09:30 debug   msg:opening audio decoder: [ffmpeg] ffmpeg/libavcodec audio decoders
21-08-2004 21:09:30 debug   msg:dec_audio: allocating 131072 + 65536 = 196608 bytes for output buffer.
21-08-2004 21:09:30 debug   msg:ffmpeg's libavcodec audio codec
21-08-2004 21:09:30 debug   msg:info: libavcodec init ok!
21-08-2004 21:09:31 debug   msg:audio: 44100 hz, 2 ch, 16 bit (0x10), ratio: 176400->176400 (1411.2 kbit)
[b]21-08-2004 21:09:31 debug   msgelected audio codec: [ffdts] afm:ffmpeg (dts)[b]
21-08-2004 21:09:31 debug   msg:==========================================================================
21-08-2004 21:09:31 debug   msg:checking audio filter chain for 44100hz/2ch/16bit -> 44100hz/2ch/16bit...
21-08-2004 21:09:31 debug   msg:[libaf] adding filter volume
21-08-2004 21:09:31 debug   msg:af_pre: af format: 2 bps, 2 ch, 44100 hz, little endian signed int
21-08-2004 21:09:31 debug   msg:af_pre: 44100hz 2ch signed 16-bit (little-endian)
21-08-2004 21:09:31 debug   msg:ao: [win32] 44100hz 2ch signed 16-bit (little-endian) (2 bps)
21-08-2004 21:09:31 debug   msg:ao: description: windows waveout audio output
ao: author: saschasommer <[email protected]>
21-08-2004 21:09:31 debug   msg:building audio filter chain for 44100hz/2ch/16bit -> 44100hz/2ch/16bit...
21-08-2004 21:09:31 debug   msg:video: no video
21-08-2004 21:09:31 debug   msg:freeing 0 unused video chunks
21-08-2004 21:09:31 debug   msgtarting playback...
21-08-2004 21:09:31 debug   msg:mplayer_open_file(smb://xxxxxxx;xxxxxxx:[email protected]/storage/samples/dts/surroundtest_011212.wav) done 1.0000

nothing to do?
can it:
a) detect that is dts
b) upsample to 48khz
c) passthrougt to decoder

[ravemax]

and precompiled mplayer: mplayer_20040820

works perfect with all kickassanime releases - all previous mplayer.dlls crashed.

[Guybrush]

do you also know why it crashed?

[baerbel]

@ravemax did you test gundam-seed: gundam seed 02 by kaa?

on my system, at time index 9.22 i get massive slowdowns and image/sound out of sync, for a couple of seconds. this is, when sound is turned on (had to add format 0xff to the libfaad section of codec conf of a 08-20 build). mplayer.dll is mplayer_20040820 from ezar2003.

i tested an older mplayer.dll from ezar2003 too, which had the same problem (08-16).

there is no slowdown, if the sound isn't decoded.

@guybrush: i never had crashes with older dlls. just didn't have sound for those 'aac in avi' files.

cu

[ezar2003]

@baerbel: i update, if you will to test it, you can download @ mplayer_20040822

[baerbel]

ezar2003, i tested your new dll (with my and your codec.conf), but the droped frames are still present.

i checked that file with bsplayer and mplayer (1.0pre5) for windows, and there are no droped frames.

cu