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Old 15th December 2018, 07:32   #14741  |  Link
mkver
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Originally Posted by asarian View Post
Thing is, I'd love to use ffmpeg directly, but I can't extract the Atmos stream with tsMuxer first (as it doesn't recognize it).
But you can use ffmpeg directly. A Bluray contains a slightly modified version of a transport stream that ffmpeg can handle. And if you build ffmpeg with libbluray, it can even parse playlists.
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Old 15th December 2018, 08:00   #14742  |  Link
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But you can use ffmpeg directly. A Bluray contains a slightly modified version of a transport stream that ffmpeg can handle. And if you build ffmpeg with libbluray, it can even parse playlists.
Thx.

Hmm, turns out ffmpeg (4.1) yielded the same errors. I can't say I'm hearing any distortion, but still.
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Old 15th December 2018, 08:44   #14743  |  Link
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This means that either something is wrong with your source or that something is wrong with ffmpeg's decoder. The best place to continue would probably be ffmpeg's bugtracker.
What happens if you use the new truehd_core bitstream filter (designed to drop atmos extensions; usage: ffmpeg -i <inputfile> -vn -sn -c:a copy -bsf:a truehd_core -f <outputfile>)?
If the errors always happen at the same place, you could upload just the few seconds of the track for an issue. (Btw: Does this Bluray use seamess branching and is the error at a branchpoint?)
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Old 15th December 2018, 10:05   #14744  |  Link
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Originally Posted by mkver View Post
This means that either something is wrong with your source or that something is wrong with ffmpeg's decoder. The best place to continue would probably be ffmpeg's bugtracker.
What happens if you use the new truehd_core bitstream filter (designed to drop atmos extensions; usage: ffmpeg -i <inputfile> -vn -sn -c:a copy -bsf:a truehd_core -f <outputfile>)?

Thx for the tip! To what can I output the TrueHD stream, though? FFmpeg won't let me output it to .wav (probably because wav is 6 channels only, iirc). I will need a stream I can downmix with eac3to (-down6) again.
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Old 15th December 2018, 10:50   #14745  |  Link
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The reason that ffmpeg won't let you mux the truehd stream into wav is because wav is a container for uncompressed PCM data, not for truehd data. You can e.g. mux it into Matroska. Just use a file with a "mka" extension and ffmpeg will know that you want to mux into Matroska.

Also keep in mind that if you intend to open a ticket in ffmpeg's bugtracker, you need to provide the truehd track with the atmos extension (or at least a snippet of it that allows to reproduce the issue).
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Old 15th December 2018, 11:02   #14746  |  Link
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Quote:
Originally Posted by asarian View Post
Same here
Same to what?

Quote:
eac3to.exe valerian.m2ts 2: c:\video\temp.pcm -down6
MKV, 1 video track, 2 audio tracks, 3 subtitle tracks, 2:17:25, 24p /1.001
1: h265/HEVC, 2160p24 /1.001 (16:9), 10 bits
2: TrueHD (Atmos), English, 7.1 channels, 48kHz, 2419ms
"TrueHD Atmos 7.1"
3: AC3 EX, English, 5.1 channels, 640kbps, 48kHz, 2419ms
"AC-3 5.1-EX"
...
[truehd @ 026336c0] Lossless check failed - expected 00, calculated 78.
Is a m2ts or a mkv?

Please don't use .pcm, use .w64 or .wav instead.

Some messages can be ignored.

Quote:
, but I can't extract the Atmos stream with tsMuxer first (as it doesn't recognize it).
tsMuxeR is a muxer, to extract use eac3to or ffmpeg or mkvextract if it is a mkv.

Quote:
Originally Posted by asarian View Post
FFmpeg won't let me output it to .wav (probably because wav is 6 channels only, iirc). I will need a stream I can downmix with eac3to (-down6) again.
wav can have more than 6 channels, but can't be greater than 4 GB (limit for strict wav format) use .w64

If: eac3to valerian.m2ts 2: c:\video\temp.w64

and: ffmpeg -i valerian.m2ts -vn -acodec pcm_s24le c:\video\temp.w64

produce the same messages and output you can ignore them or assume is a corrupt input.

After that you can downmix the w64 (recommended with MeGUI or BeHappy).
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Last edited by tebasuna51; 15th December 2018 at 11:09. Reason: typo
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Old 15th December 2018, 14:40   #14747  |  Link
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Originally Posted by tebasuna51 View Post

Is a m2ts or a mkv?
Good catch. Too used to .m2ts files. This test actually was on the mkverge .mkv version of the .m2ts (video and audio only), on my media server. You got me thinking, though, that maybe it got corrupted somehow. I just loaded up the disc again, but it makes no difference: about the same amount of errors.

Quote:
Please don't use .pcm, use .w64 or .wav instead.
What's wrong with .pcm?! I always convert every 7.1 source to .pcm (with -down6). .pcm is supposed to be lossless, right?

Quote:
If: eac3to valerian.m2ts 2: c:\video\temp.w64

and: ffmpeg -i valerian.m2ts -vn -acodec pcm_s24le c:\video\temp.w64

produce the same messages and output you can ignore them or assume is a corrupt input.

After that you can downmix the w64 (recommended with MeGUI or BeHappy).
Hmm, that's sad: the latest ffmpeg also produces the same errors. So, maybe I should file a small bug report with them; but, in all honesty, I can't hear any errors at all; so maybe I'll just forget about it.
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Old 15th December 2018, 21:35   #14748  |  Link
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Raw PCM has no header. No information which sample rate, which bit depth, which number format, how many channels, which channel layout ... no details, only a byte stream of sample data. You will have to tell a reading application every attribute, or it won't be able to interpret the byte stream.

WAV has a header but is limited to 32 bit size fields (max. 4 GB); W64 expands this limit.

FLAC is losslessly compressed.
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Old 16th December 2018, 01:36   #14749  |  Link
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Raw PCM has no header. No information which sample rate, which bit depth, which number format, how many channels, which channel layout ... no details, only a byte stream of sample data. You will have to tell a reading application every attribute, or it won't be able to interpret the byte stream.

WAV has a header but is limited to 32 bit size fields (max. 4 GB); W64 expands this limit.

FLAC is losslessly compressed.
Thanks. Thing is, .mkv is just an intermediate stage for me: eventually everything gets muxed to an .m2ts container again. So, it kinda needs to stay LPCM (for which I then use the good old Pcm2Tsmu, to get tsMuxer to accept it).
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Old 16th December 2018, 11:46   #14750  |  Link
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Originally Posted by asarian View Post
...then use the good old Pcm2Tsmu, to get tsMuxer to accept it.
Thanks, I write Pcm2Tsmu for that, but last tsMuxeR versions accept already .w64 like pcm input greater 4 GB.

Pcm2Tsmu is old and obsolete now.
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Old 16th December 2018, 12:09   #14751  |  Link
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Thanks, I write Pcm2Tsmu for that, but last tsMuxeR versions accept already .w64 like pcm input greater 4 GB.

Pcm2Tsmu is old and obsolete now.
Hehe, didn't even realize you wrote Pcm2Tsmu. Thanks for that! It's been an instrumental tool!

And yeah, even a 5.1 LPCM stream is nearly 7G. So, good tip on using .w64 format.
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Old 21st December 2018, 17:33   #14752  |  Link
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I have a bunch of MKVs with FLAC audio (varying from FLAC 1.0, 2.0 and 7.1) that I want to convert to DTS-HD MA.

I thought i could attempt to do this with eac3to alongside the arcsoft, surcode or DTS-HD Master Suite software. I'm running Windows 10 FYI.

Has anyone had good experience with latest windows trying to get this done? I run eac3to (using the UsEac3to GUI) and select the functionality to convert Track #2: FLAC to DTS from within eac3to.

I always end up getting the error: "Pressing the Surcode "Encode" button didn't seem to work..." I tried making the file names 1 character long and on a very short file path (i.e. F:/Movies/a.mkv) but still get the same error, on either 1.0.21 or 1.0.29 Surcode version.

I thought...well how about I just convert the FLAC to wavs (I tried it on a mono FLAC giving me just one center .wav file) then run it through Surcode...well Surcode converts that center mono file into a five channel DTS track (which I can successfully mux with the video but I feel like this is wrong...).

So lastly, I wanted to try with the actual DTS-HD Master Suite software, which seems like the most viable. I tried converting the single center channel .wav file but the encode just seems frozen in my queue as pending...


So any idea what would be the best way to tackle my goal of converting FLAC 1.0/2.0/7.1 from an MKV into DTS-HD MA 1.0/2.0/7.1 (respectively) in an MKV?

I saw this which seemed to indicate people don't reccommend DTS-HD Master Suite...
https://forum.doom9.org/showthread.php?t=158611

Also Arcsoft 6 doesn't even work on Window 10 anymore. I used this work around from another post that, when I run eac3to test, it detects Arcsoft dtsdecoder dll to be working (idk if it actually is though...).
https://forum.doom9.org/showthread.php?t=154070
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Old 22nd December 2018, 03:16   #14753  |  Link
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1) ArcSoft is a decoder, not a encoder. And is not necesary anymore because the free libdcadec.dll can do the decode of DTS-MA.

2) Surcode is a DTS encoder than can run with eac3to but can't produce DTS-MA, only standard DTS 768 or 1536 Kb/s. Forget it to recode FLAC.

3) If you want DTS-MA you need DTS-HD Master Suite and you only need eac3to to decode the FLAC's to monowav's:

eac3to input.flac output.wavs

Or in UsEac3to selecting 'wavs' in Output format
After that you can use DTS-HD Master Suite with the monowavs input:

Stream type DTS-HD Master Audio
Channel Layout: from 1.0 to 7.1
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Old 31st December 2018, 00:24   #14754  |  Link
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Can I add something to "stdout.wav | LAME" to make it show progress? Otherwise any option of LAME for verbosity is ignored and no progress shows. E.g. with QAAC it shows fine.
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Old 18th January 2019, 16:09   #14755  |  Link
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Ran into this problem: HEVC multi-part demuxing? Any chance HEVC demuxing will become a reality with eac3to?

Thanks.
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Old 18th January 2019, 16:46   #14756  |  Link
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Can you just join them with simple
copy /b 800.m2ts+801.m2ts+802.m2ts+803.m2ts+804.m2ts combined.m2ts
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Old 18th January 2019, 19:02   #14757  |  Link
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Can you just join them with simple
copy /b 800.m2ts+801.m2ts+802.m2ts+803.m2ts+804.m2ts combined.m2ts
Never mind my stupidity. Turns out you CAN demux the HEVC stream, just not to .mkv yet.
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Old 7th February 2019, 05:33   #14758  |  Link
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Yes, is a workaround that works fine, unfortunately you lose the 7.1 data.
Your workaround was to add -core to -demux when demuxing E-AC3 streams which caused eac3to to abort extraction due to "Applying (E-)AC3 delay failed" error.

The audio streams I wanted were not the problem E-AC3 ones, so I extracted just that stream => X.thd+ac3.

Then I ran eac3to [path] 1) -demux -core and let it extract all the streams. The X.thd+ac3 produced was binary identical to the one above.

As you suggested, I would have imagined using -core would lose something, but it doesn't seem to have here, do you have any explanation?
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Old 7th February 2019, 10:07   #14759  |  Link
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...As you suggested, I would have imagined using -core would lose something, but it doesn't seem to have here, do you have any explanation?
The -core eac3to parameter is intended to extract the core part (limited to 5.1) of a single audio track, before only with DTS-HD but now also EAC3 7.1.

A thd+ac3 is not a single audio track but 2 independent tracks interleaved.
To extract each track you can use:

eac3to X.thd+ac3 X.thd X.ac3

The -core parameter is ignored here.
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Old 10th February 2019, 01:45   #14760  |  Link
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Can I remap a 4.1 audio to bring the second voice channel up front when playing in stereo? Software players can mix voice to both channels but my TV can't.
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