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Old 20th December 2007, 00:53   #2081  |  Link
moshmothma
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Originally Posted by nautilus7 View Post
I had the same error, but with matrix truehd. It's not really a big problem, just few millisecs that are missing at the end of the track.

I had made a sample and madshi said that he was going to forward it to the libav developer.

Someone forgot to do it... (just kidding)
Madshi, that's the error I had with truehd from 300. Just FYI
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Old 20th December 2007, 01:49   #2082  |  Link
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Is it possible to make 24-bit legacy DTS tracks from 24-bit DD+/TrueHD tracks?
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Old 20th December 2007, 03:24   #2083  |  Link
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I recently tried my first DTS-HD --> DTS conversion. Went well insofaras got a perfect DTS stream, playable by itself. I muxed this DTS with the video AVC elementary stream using mkvmerge GUI.

If I mux just the AVC stream to .mkv, or just play it back by itself, plays perfectly and smoothly. I am using WMP 11 for playback.

BUT... if I mux the DTS and AVC together, I get "problems." I am sending the bitstream out by SPDIF. First I tried ffdshow. The file overall plays, but every few seconds there's just a slight "jump" or hiccup. It is mild, but noticeable enough to be bothersome. Perhaps this is the audio and video trying to stay in sync? Then I tried AC3 filter for my SPDIF output of the DTS. This allows the file to play smoothly, but about 10 minutes in, the audio and video begin to noticeable desync. By 1 hour in, they're really out of sync (a few seconds).

If I convert the DTS-HD into AC3 and mux that with my 1920x1080 AVC file, I get very smooth playback and no out-of-sync whatsoever.

I'd love of course to start using DTS's higher bitrates, but I can't seem to conquer this. I have tried multiple iterations.

Thoughts?
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Old 20th December 2007, 03:49   #2084  |  Link
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Quote:
Originally Posted by scarbrtj View Post
I recently tried my first DTS-HD --> DTS conversion. Went well insofaras got a perfect DTS stream, playable by itself. I muxed this DTS with the video AVC elementary stream using mkvmerge GUI.

If I mux just the AVC stream to .mkv, or just play it back by itself, plays perfectly and smoothly. I am using WMP 11 for playback.

BUT... if I mux the DTS and AVC together, I get "problems." I am sending the bitstream out by SPDIF. First I tried ffdshow. The file overall plays, but every few seconds there's just a slight "jump" or hiccup. It is mild, but noticeable enough to be bothersome. Perhaps this is the audio and video trying to stay in sync? Then I tried AC3 filter for my SPDIF output of the DTS. This allows the file to play smoothly, but about 10 minutes in, the audio and video begin to noticeable desync. By 1 hour in, they're really out of sync (a few seconds).

If I convert the DTS-HD into AC3 and mux that with my 1920x1080 AVC file, I get very smooth playback and no out-of-sync whatsoever.

I'd love of course to start using DTS's higher bitrates, but I can't seem to conquer this. I have tried multiple iterations.

Thoughts?
You must use a timecodes file on the video stream in mkvmerge with the correct fps. Read the mkvmerge help as it has a detailed explanation of this feature.
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Old 20th December 2007, 04:04   #2085  |  Link
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@madshi,
I just wanted to say thanks for the new version. Man you packed a lot of stuff into it. Very nice. I want to contribute any findings, so here goes.

I have a short test pcm file that converts just fine to both ac3 and dts, see output below. However, when you run it to just get the info on the file, the program crashes, i.e., "eac3to audio.pcm" crashes.

Output of eac3tov2 audio.pcm audio.ac3 -384 -16 -little
Code:
This might be a RAW/PCM file. Trying to figure out the details.
This will probably take a while. Please be patient...
---The RAW/PCM file seems to be little endian.
---The suggested endian (little) should be correct.
---The RAW/PCM file seems to have a bitdepth of 16 bits.
---The suggested depth of 16 bits should be correct.
---The RAW/PCM file seems to have 6 channels.
---RAW/PCM, 5.1 channels, 0:04:19, 16 bits, 48khz
---Reading RAW/PCM...
Encoding AC3...
Creating/writing file "audio.ac3"...
-------------------------------------------------------------------------------
Done.
Output of eac3tov2 audio.pcm audio.dts -16 -little
Code:
This might be a RAW/PCM file. Trying to figure out the details.
This will probably take a while. Please be patient...
---The RAW/PCM file seems to be little endian.
---The suggested endian (little) should be correct.
---The RAW/PCM file seems to have a bitdepth of 16 bits.
---The suggested depth of 16 bits should be correct.
---The RAW/PCM file seems to have 6 channels.
---RAW/PCM, 5.1 channels, 0:04:19, 16 bits, 48khz
---Reading RAW/PCM...
Writing WAVs...
Creating/writing file "audio.L.wav"...
Creating/writing file "audio.R.wav"...
Creating/writing file "audio.C.wav"...
Creating/writing file "audio.LFE.wav"...
Creating/writing file "audio.SL.wav"...
Creating/writing file "audio.SR.wav"...
-------------------------------------------------------------------------------
Found Surcode DTS Encoder version 1.0.23.0.
Surcode encoding successfully started. Please wait...
Closing Surcode...
Done.
I'll send the debug info separately. Thank you.
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Old 20th December 2007, 04:35   #2086  |  Link
scarbrtj
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Quote:
Originally Posted by Chumbo View Post
You must use a timecodes file on the video stream in mkvmerge with the correct fps. Read the mkvmerge help as it has a detailed explanation of this feature.
I did, I think; I thought of this (see below). As I said, the AVC file if muxed into .mkv by itself plays back flawlessly. Also, playback is flawless if I mux AVC with AC3 (generated from DTS-HD). But... the .mkv of an AVC and DTS (actually it is DTS-ES 6.1 stream) plays back jumpy (ffdshow audio decode to SPDIF) or out-of-sync (AC3filter to SPDIF).

The timecodes file I used was:

# timecode format v1
Assume 23.976

Should this have produced a good DTS/AVC mkv merge? I did not use this to merge AVC and AC3 and the mkv plays back great (but did have to specify a 23.976 framerate in mkvmerge GUI).
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Old 20th December 2007, 04:50   #2087  |  Link
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Quote:
Originally Posted by scarbrtj View Post
I did, I think; I thought of this (see below). As I said, the AVC file if muxed into .mkv by itself plays back flawlessly. Also, playback is flawless if I mux AVC with AC3 (generated from DTS-HD). But... the .mkv of an AVC and DTS (actually it is DTS-ES 6.1 stream) plays back jumpy (ffdshow audio decode to SPDIF) or out-of-sync (AC3filter to SPDIF).

The timecodes file I used was:

# timecode format v1
Assume 23.976

Should this have produced a good DTS/AVC mkv merge? I did not use this to merge AVC and AC3 and the mkv plays back great (but did have to specify a 23.976 framerate in mkvmerge GUI).
I'm sorry, you did mention that muxing ac3 works fine and I read right past it. My bad. Your timecode stuff is correct, so I'm not sure why it's not syncing unless it's an issue with whatever splitter you're using.

I, personally, avoid WMP at all costs. Try using MPC and use its built-in source filters: DTS/AC3, Matroska and MPEG PS/TS/PVA. I normally use the Haali splitter, but at times, with DTS, I have to switch to MPC's internal filters to get smooth play back. Good luck.
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Old 20th December 2007, 06:34   #2088  |  Link
Snowknight26
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Quote:
Originally Posted by Chumbo View Post
I have a short test pcm file that converts just fine to both ac3 and dts, see output below. However, when you run it to just get the info on the file, the program crashes, i.e., "eac3to audio.pcm" crashes.
I can confirm this. Happens on several pcm streams I have.
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Old 20th December 2007, 07:12   #2089  |  Link
itsancho
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Quote:
Originally Posted by MuteyM View Post
Hi Madshi, love your app, and I've got what I think is a libav problem for you to puzzle over... On the NIN HD DVD, every single demuxed TrueHD track gives the same error message, right at the very end of decoding:
Code:
eac3to.exe EVOB010.thd blah.wav -libav
TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
Writing WAV...
Removing dialog normalization...
Creating/writing file "D:\blah.24bit.wav"...
This audio track contains more than 16 bit of information.
-------------------------------------------------------------------------------[
mlp @ 68A442E0]End of stream indicated
[mlp @ 68A442E0]Substream 1 parity check failed
[mlp @ 68A442E0]Substream 1 checksum failed
[mlp @ 68A442E0]Substream 1 length mismatch.
Done.
Up to version 2.08, this error was ignored by eac3to and the resulting output file could be used without problems. But starting with version 2.09, the output file gets erased upon error! If it's not a big deal, I'd like to request you revert back to the old behaviour, for cases where libav chokes or just for general debugging of errors.

I'm 99% sure it's not a decrypting or demuxing problem, because I get the same results with demux.exe, EvobDemux, and eac3to 2.10's demux-to-wav functionality, and it happens with all EVOs on the disc. But the resulting output file always sounds perfect.

So I'm thinking it's a bug in libav (or maybe a defect on my disc?) Anyway, you can download a sample .thd file from here:
http://www.sendspace.com/file/udpvza

Once again, thanks for such an amazing app!
absolutely the same problem with "TMNT" and when i forced to use nero - everything its OK...
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Old 20th December 2007, 08:44   #2090  |  Link
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Quote:
Originally Posted by nautilus7 View Post
How do i check whether a truehd track is 16, 20 or 24 bits?

I am not sure if the truehd track from inside man hd dvd is 24 bit. The flac i got from it claims to be 24, but size is only 1,88 GB for 2 a hour movie, whether the prestige's flac track (which is definately 24 bits) is 2,79 GB.
the bitrate is a good indicator.. 16bit is usually less than 1500kbps, 20bit > 2000kbps, 24bit > 3000kbps
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Old 20th December 2007, 09:53   #2091  |  Link
madshi
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Originally Posted by Thunderbolt8 View Post
hm better please leave it at 23.976 for broadcasts, because sometimes the audio tracks get replaced with those coming from DVDs and those are afaik exact 23.976 values (at least I once tested it with a star wars broadcast and found some scenes were 23.976 looked better over 23.9760239 (rest scenes were more like neutral)). or at least please add a switch for that case
I don't believe you can see a difference of 7ms - and that's all there is between 23.976 and 23.9760239 at the end of a two hour movie. FWIW, I believe 24/1.001 is "more correct" than 23.976.
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Old 20th December 2007, 09:55   #2092  |  Link
madshi
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Quote:
Originally Posted by nautilus7 View Post
How do i check whether a truehd track is 16, 20 or 24 bits?

I am not sure if the truehd track from inside man hd dvd is 24 bit. The flac i got from it claims to be 24, but size is only 1,88 GB for 2 a hour movie, whether the prestige's flac track (which is definately 24 bits) is 2,79 GB.
We had a similar effect with Pirates of the Caribbean 1. I checked that movie and there were some very short sequences of the PCM track which were 24bit while the majority of the track was 16bit. I think the same thing is very likely to be true with the inside man track.
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Old 20th December 2007, 09:57   #2093  |  Link
madshi
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Originally Posted by Snowknight26 View Post
Madshi, here is a sample from The Phantom of the Opera:
http://www.stfcc.org/misc/PEVOB_1.EVO
Thanks! Will have a look.
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Old 20th December 2007, 10:08   #2094  |  Link
madshi
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Quote:
Originally Posted by MuteyM View Post
On the NIN HD DVD, every single demuxed TrueHD track gives the same error message, right at the very end of decoding:
Code:
eac3to.exe EVOB010.thd blah.wav -libav
TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
Writing WAV...
Removing dialog normalization...
Creating/writing file "D:\blah.24bit.wav"...
This audio track contains more than 16 bit of information.
-------------------------------------------------------------------------------[
mlp @ 68A442E0]End of stream indicated
[mlp @ 68A442E0]Substream 1 parity check failed
[mlp @ 68A442E0]Substream 1 checksum failed
[mlp @ 68A442E0]Substream 1 length mismatch.
Done.
Up to version 2.08, this error was ignored by eac3to and the resulting output file could be used without problems. But starting with version 2.09, the output file gets erased upon error! If it's not a big deal, I'd like to request you revert back to the old behaviour, for cases where libav chokes or just for general debugging of errors.

I'm 99% sure it's not a decrypting or demuxing problem, because I get the same results with demux.exe, EvobDemux, and eac3to 2.10's demux-to-wav functionality, and it happens with all EVOs on the disc. But the resulting output file always sounds perfect.

So I'm thinking it's a bug in libav (or maybe a defect on my disc?) Anyway, you can download a sample .thd file from here:
http://www.sendspace.com/file/udpvza
Yeah, it seems to be a bug in the libav decoder. My guess is that the decoder believes that the truehd stream is done and finished and then surprisingly there's more truehd data coming in. And the decoder doesn't seem to like that. Should be easy to fix, though. Give the decoder developer a few days. His replies sometimes take a few days, but he always comes back with a fix. You can use the Nero decoder in the meanwhile ("-nero" switch).

Quote:
Originally Posted by nautilus7 View Post
Someone forgot to do it... (just kidding)
No, I didn't...
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Old 20th December 2007, 10:10   #2095  |  Link
madshi
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Originally Posted by bmnot View Post
Is it possible to make 24-bit legacy DTS tracks from 24-bit DD+/TrueHD tracks?
Sure. Just do "eac3to source.thd dest.dts" or "eac3to source.eac3 dest.dts". Or directly from the EVO source: "eac3to feature_1.evo+feature_2.evo 2: dest.dts".

However, you need to have the commercial (and quite expensive) Surcode DTS encoder installed!
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Old 20th December 2007, 10:14   #2096  |  Link
madshi
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Quote:
Originally Posted by scarbrtj View Post
I recently tried my first DTS-HD --> DTS conversion. Went well insofaras got a perfect DTS stream, playable by itself. I muxed this DTS with the video AVC elementary stream using mkvmerge GUI.

If I mux just the AVC stream to .mkv, or just play it back by itself, plays perfectly and smoothly. I am using WMP 11 for playback.

BUT... if I mux the DTS and AVC together, I get "problems." I am sending the bitstream out by SPDIF. First I tried ffdshow. The file overall plays, but every few seconds there's just a slight "jump" or hiccup. It is mild, but noticeable enough to be bothersome. Perhaps this is the audio and video trying to stay in sync? Then I tried AC3 filter for my SPDIF output of the DTS. This allows the file to play smoothly, but about 10 minutes in, the audio and video begin to noticeable desync. By 1 hour in, they're really out of sync (a few seconds).

If I convert the DTS-HD into AC3 and mux that with my 1920x1080 AVC file, I get very smooth playback and no out-of-sync whatsoever.

I'd love of course to start using DTS's higher bitrates, but I can't seem to conquer this. I have tried multiple iterations.

Thoughts?
My first guess would be that the muxer doesn't like the DTS-HD core. There's a small difference to "normal" DTS files. Normal DTS files usually have 2013 bytes per DTS frame while the DTS core from a DTS-HD track only has 2012 bytes per DTS frame. I don't know it behaves this way. And it gets even stranger: If you want to keep audio sync, the DTS parser needs to behave as if the frames were 2013 bytes long!! Extremely strange and kind of annoying. Personally, I'm not muxing the DTS files into the MKV file. Instead I'm keeping them external. The MPC HC can play them as external files. Earlier in this thread you'll find a modified source filter which plays these DTS core tracks with correct audio sync.
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Old 20th December 2007, 11:04   #2097  |  Link
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Quote:
Originally Posted by madshi View Post
We had a similar effect with Pirates of the Caribbean 1. I checked that movie and there were some very short sequences of the PCM track which were 24bit while the majority of the track was 16bit. I think the same thing is very likely to be true with the inside man track.
That's possible...
I tried the -check16bit switch, but it told me thats the track contains more than 16 bit of information. But how many exactly?

How did the studio manage to make a track with variable bit depth?
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Old 20th December 2007, 11:15   #2098  |  Link
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Quote:
Originally Posted by madshi View Post
We had a similar effect with Pirates of the Caribbean 1. I checked that movie and there were some very short sequences of the PCM track which were 24bit while the majority of the track was 16bit. I think the same thing is very likely to be true with the inside man track.
That's possible...
I tried the -check16bit switch, but it told me that the track contains more than 16 bit of information. But how many exactly, none knows...

How did the studio manage to make a track with variable bit depth?
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Old 20th December 2007, 11:18   #2099  |  Link
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i think it's just constant 20bit.. the track the nero decoder had problems with before was inside man, and madshi suspected it could be 18 or 20bit.. i dithered some other 24bit tracks down to 20bit and they compressed to very similar bitrates
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Old 20th December 2007, 11:33   #2100  |  Link
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Can flac have bit depths of 20 bits? If yes why eac3to said "writing 24 bit flac"...

Last edited by nautilus7; 20th December 2007 at 11:35.
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