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Old 17th May 2024, 13:27   #1  |  Link
PixelPirate
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Eac3to Keeps giving me an Error

For some reason when I try and convert DTS 5.1 core to AC-3 5.1 core, eac3to gives me the follwing error:

libDcaDec output changed from 6 channels, 48kHz to 8 channels, 48kHz.
Aborted at file position 1307836416.

Anyone know why this happens?
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Old 17th May 2024, 19:09   #2  |  Link
tebasuna51
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If you put the parameter -core in the command line I don't know what happen. Maybe a corrupt source?

Try with ffmpeg with a more updated libDcaDec.
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Old 17th May 2024, 20:56   #3  |  Link
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Quote:
Originally Posted by tebasuna51 View Post
If you put the parameter -core in the command line I don't know what happen. Maybe a corrupt source?

Try with ffmpeg with a more updated libDcaDec.
If I leave the DTS-MA track which has 8 channels untouched, then eac3to works. But sometimes when I reduce to core, eac3to does not work. I'm not sure why this happens.

When I try with FFmpeg, it works. I think I get a similar warning like eac3to, but it does not stop. It continues and works.

Last edited by PixelPirate; 17th May 2024 at 20:58.
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Old 18th May 2024, 08:49   #4  |  Link
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If ffmpeg show the same warning maybe there are a problem in the source DTS-MA, if you have one with that problem and you can upload it send me it to check.
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Old 18th May 2024, 20:05   #5  |  Link
PixelPirate
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Quote:
Originally Posted by tebasuna51 View Post
If ffmpeg show the same warning maybe there are a problem in the source DTS-MA, if you have one with that problem and you can upload it send me it to check.
This is the error eac3to gives me when the DTS core is used:

eac3to v3.48
command line: "eac3to.exe" "input.mka" "Output.ac3" -core
------------------------------------------------------------------------------
Running in fast mode
Keeping dialnorm
MKA, 1 audio track
1: DTS, English, 5.1 channels, 1509kbps, 48kHz, dialnorm: 0dB
Track 1 is used for destination file "Output.ac3".
[a01] Extracting audio track number 1...
[a01] Decoding with libDcaDec DTS Decoder...
[a01] Remapping channels...
[a01] Encoding AC3 <640kbps> with libAften...
[a01] Creating file "Output.ac3"...
[a01] libDcaDec output changed from 6 channels, 48kHz to 8 channels, 48kHz. <ERROR>
Aborted at file position 68157440. <ERROR>

Here is the sample:
https://mega.nz/file/EqA3nSSK#WTVIAM...Pizw4F7d6Wa34k
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Old 18th May 2024, 22:30   #6  |  Link
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The stream contains a lossless extension starting from that position or about 6 minutes. "She's not responding. Ok. Here we go." You can cut the stream at that point with a hex editor and extract the core from the second segment, then join the first part with the core. Or you can corrupt the lossless extension by replacing 64 58 20 25 -> FF FF FF FF. It won't play in some players like old Foobar, but eac3to will continue processing with a notification that the track isn't clean.

Usually you can't change channels or sampling rate when joining streams.
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Old 18th May 2024, 23:08   #7  |  Link
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Like j7n say your sample is a mix of 6 minutes of standard dts 5.1 and after 5 minutes of DTS-MA 7.1

The recoded to ac3 with ffmpeg is correct (never use the Aften encoder from eac3to).
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Old 19th May 2024, 04:31   #8  |  Link
PixelPirate
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Quote:
Originally Posted by tebasuna51 View Post
Like j7n say your sample is a mix of 6 minutes of standard dts 5.1 and after 5 minutes of DTS-MA 7.1

The recoded to ac3 with ffmpeg is correct (never use the Aften encoder from eac3to).
But if the source has 8 chaneels (7.1) instead of 6 channels (5.1) then what does the following FFmpeg command line do?

FFmpeg -i input.mkv -c:a:0 ac3 -b:a:0 640k -ac:a:0 6 "Output.ac3"

Does it discard the 2 extra channels, or does it downmix the 2 extra channels into SL and SR like eac3to? If it discards the 2 extra channels, then what is the FFmpeg command line to downmix the 2 extra channels into SL and SR?
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Old 19th May 2024, 08:19   #9  |  Link
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Quote:
Originally Posted by PixelPirate View Post
... or does it downmix the 2 extra channels into SL and SR like eac3to?
Yes, the dowmix is automatic, even if you don't put any extra parameter in the command line.

The ffmpeg message:

[af#0:0 @ 0000026ffad50880] Reconfiguring filter graph because audio parameters changed to 48000 Hz, 7.1, s32p

show the automatic change.
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Last edited by tebasuna51; 19th May 2024 at 08:24.
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Old 19th May 2024, 12:06   #10  |  Link
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tebasuna51,

Could you please tell me which one gives the best AC-3 5.1 audio output when downmixing from DTS-HD MA, or do you think it makes no difference?

1. Reduce to core with MKVToolNix GUI, then downmix to AC-3 5.1 core with either eac3to or FFmpeg.
2. Leave the DTS-HD MA 8-channel audio stream untounched, then downmix to AC-3 5.1 core with either eac3to or FFmpeg.
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Old 20th May 2024, 08:58   #11  |  Link
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In this thread, read it to know for what, I recommend:
Code:
ffmpeg.exe -i "dts-ma.dts" -filter_complex "asplit [f][s]; [f] pan=3.1|c0=c0|c1=c1|c2=c2|c3=c3 [r]; [s] pan=stereo|c0=0.5*c4+0.5*c6|c1=0.5*c5+0.5*c7, compand=attacks=0:decays=0:points=-90/-84|-8/-2|-6/-1|-0/-0.1, aformat=channel_layouts=stereo [d]; [r][d] amerge [a]" -map "[a]" -c:a ac3 -center_mixlev 0.707 -b:a 640k "output.ac3"
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