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Old 26th June 2010, 22:03   #10221  |  Link
DaFees
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Quote:
Originally Posted by tebasuna51 View Post
@DaFees
Please put the eac3to log.
I wish I could, but I'm not sure if I am getting a correct log. I ran multiavchd and that didn't provide a useful log. I did try running eac3to via commandline and it did give me a log, but I do not think it was helpful:

Code:
eac3to v3.21
command line: C:\multiavchd\tools\eac3to\eac3to.exe  "D:\BD Wip\00001.track_4352.ac3" "D:\BD Wip\00001.wav"
------------------------------------------------------------------------------
TrueHD/AC3, 5.1 channels, 48kHz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB)
Extracting TrueHD stream...
Removing TrueHD dialog normalization...
Decoding with libav/ffmpeg...
Writing WAV...
Creating file "D:\BD Wip\00001.wav"...
The original audio track has a constant bit depth of 16 bits.
Caution: The WAV file is bigger than 4GB.  <WARNING>
Some WAV readers might not be able to handle this file correctly.  <WARNING>
Superfluous zero bytes detected, will be stripped in 2nd pass.
Starting 2nd pass...
Reading WAV...
Stripping zero bytes...
Writing WAV...
Creating file "D:\BD Wip\00001.wav"...
Caution: The WAV file is bigger than 2GB.  <WARNING>
Some WAV readers might not be able to handle this file correctly.  <WARNING>
eac3to processing took 19 minutes, 51 seconds.
Done.
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Old 26th June 2010, 22:43   #10222  |  Link
Atak_Snajpera
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Try this
C:\multiavchd\tools\eac3to\eac3to.exe "D:\BD Wip\00001.track_4352.ac3" "D:\BD Wip\00001.w64"
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Old 27th June 2010, 00:12   #10223  |  Link
tebasuna51
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@DaFees
Of course the log help to know the problem.

eac3to works fine and decode the TrueHD without problems, I don't know what is the problem with multiavchd.
Try with the w64 format like Atak_Snajpera say you.
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Old 27th June 2010, 00:48   #10224  |  Link
DaFees
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Quote:
Originally Posted by Atak_Snajpera View Post
Try this
C:\multiavchd\tools\eac3to\eac3to.exe "D:\BD Wip\00001.track_4352.ac3" "D:\BD Wip\00001.w64"
Well I tried this and I even remuxed with tsmuxer and when I tried to play the final video I got no sound and stuttery video, again.

Still though eac3to still told me all was well:

Code:
eac3to v3.21
command line: C:\multiavchd\tools\eac3to\eac3to.exe  "D:\BD Wip\00001.track_4352.ac3" "D:\BD Wip\00001.w64"
------------------------------------------------------------------------------
TrueHD/AC3, 5.1 channels, 48kHz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48kHz, dialnorm: -27dB)
Extracting TrueHD stream...
Removing TrueHD dialog normalization...
Decoding with libav/ffmpeg...
Writing W64...
Creating file "D:\BD Wip\00001.w64"...
The original audio track has a constant bit depth of 16 bits.
Superfluous zero bytes detected, will be stripped in 2nd pass.
Starting 2nd pass...
Reading W64...
Stripping zero bytes...
Writing W64...
Creating file "D:\BD Wip\00001.w64"...
eac3to processing took 19 minutes, 6 seconds.
Done.
I should add that multiavchd will play the error sound that comes with eac3to every time it refuses to convert the audio.

Last edited by DaFees; 27th June 2010 at 00:51.
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Old 27th June 2010, 11:22   #10225  |  Link
tebasuna51
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Play D:\BD Wip\00001.wav to listen the audio, if play ok put your problem in multiavchd thread because eac3to seems work fine.
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Old 27th June 2010, 12:17   #10226  |  Link
DaFees
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Ok well I just checked and eac3to is working fine, because the .wav file played fine on my PC and yes I will post in the multiavchd thread for help, but that still doesn't totally resolve the issue. Ok, so multiavchd won't work. Well then why won't my converted audio stream when remuxed with the video play on my PS3?

I've tried the conversion two ways: Using multiavchd and tsmuxer gui to demux the audio + eac3to to convert to lpcm + tsmuxer gui again to remux the converted audio with the video.
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Old 27th June 2010, 12:31   #10227  |  Link
madshi
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Quote:
Originally Posted by Momber View Post
The mapping of 3.0 soundtracks converted to flac is incorrect. The center channel ends up in the left surround.
Can I have a short sample, please?

Quote:
Originally Posted by Momber View Post
If that is the case, perhaps it would be a good idea for eac3to to simply decline the encoding of 3.0 files to flac for the time being.
No, I will not limit eac3to just because decoders don't behave correctly. Instead the decoders should be fixed.

Quote:
Originally Posted by deank View Post
A feature request for two options:

-cutF (cut at frame)
-cutT (cut at HH:MM:SS.MLS) (or any variation using time format)
Are you talking about audio or video? Cutting how much? Have you tried the "-edit" option?

Quote:
Originally Posted by AnryV View Post
eac3to don't remove the dialog normalization with Arcsoft decoder (1.1.0.1)
Can I have a DTS-HD sample with dialnorm, please? Don't know if I can fix this. Maybe yes, maybe no. Editing the DTS-HD headers is very complicated.

Quote:
Originally Posted by Snowknight26 View Post
Reducing the bitdepth from 24-bit to 16-bit seems to cause tons of static. Tested on a 5.1 channel 24-bit WAV file. Should reproduce easily.
Quote:
Originally Posted by Xorp View Post
I'm getting that too and also on a normal 16-bit 7.1 TrueHD to FLAC conversion.
Sorry about that. One of the v3.21 fixes caused this. Will be fixed in the next build.

Quote:
Originally Posted by Blue_MiSfit View Post
I'm seeing odd behavior with Battleship Potemkin
Another Blu-Ray I don't have. Would it be possible to upload the first small m2ts file and the first maybe 10MB of the movie m2ts file, together with the PLAYLIST and CLIPINF folders?

Quote:
Originally Posted by `Orum View Post
First, an option to disable saving the internal logging of the output to a file
Is the creation of the log file such a big problem? After all, you can simply delete it again!

Quote:
Originally Posted by `Orum View Post
Secondly, a way to silence the completion "uncorking" sound.
Quote:
Originally Posted by `Orum View Post
yeah I know you can do that, but I find them useful at times...just thinking a command line flag would be best.
Nope, sorry. I don't like the idea of adding tons of funny options that must be documented/explained somewhere. That makes eac3to usage only more complicated. BTW, when writing batch files you can use stuff like "if exist success.wav goto :someMark". That should fix all "order" problems.

Quote:
Originally Posted by DaFees View Post
Ok well I just checked and eac3to is working fine, because the .wav file played fine on my PC and yes I will post in the multiavchd thread for help, but that still doesn't totally resolve the issue. Ok, so multiavchd won't work. Well then why won't my converted audio stream when remuxed with the video play on my PS3?
So there's a bug with multiavchd and another bug with tsmuxer. Why are you asking for help about these tools in the eac3to thread? As you said yourself, eac3to seems to be working fine.
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Old 27th June 2010, 12:35   #10228  |  Link
madshi
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eac3to v3.22 released

http://madshi.net/eac3to.zip

Code:
* fixed: reducing bitdepth from >16bit to 16bit resulted in static (v3.21)
* updated to latest ffmpeg/libav version
* libav MP1/MP2/MP3 decoders now decode to floating point
* libav E-AC3 decoder now supports spectral extension
The updated ffmpeg/libav libraries are quite a big change. It would be nice if you guys could do some libav decoder testing with various codecs (AC3, E-AC3, DTS, MP1, MP2, MP3, MLP, TrueHD) to make sure that things still work as intended.
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Old 27th June 2010, 16:23   #10229  |  Link
deathlord
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Quote:
Originally Posted by deathlord View Post
Hi

I have just encoutered a nasty problem using eac3to 3.17.
When converting 5.1 48kHz dts files to 5.1 flac, eac3to just stopped without error, leaving an unfinished file and no log. It happen only with one file out of ten.
Even worse, rerunning the identical command line once more, the problem does not appear anymore.
This has happend twice in the last couple of days.

Any ideas?
Unfortunately I was unable to reproduce this problem. I could run two instances of eac3to using arcsoft at the same time.

Probably there really was (is) something wrong with my PC...

Another thing:

How can I silence the last 3 seconds, say, of a flac file? Is there a way to use the -edit function to do this without manually entering "runtime minus 3s"?

I would like to systematically silence the annoying sound at the end of tv shows.

Cheers
deathlord
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Old 27th June 2010, 18:30   #10230  |  Link
AnryV
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Quote:
Originally Posted by madshi View Post
Can I have a DTS-HD sample with dialnorm, please? Don't know if I can fix this. Maybe yes, maybe no. Editing the DTS-HD headers is very complicated.
.....
Quote:
Originally Posted by AnryV View Post
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Old 27th June 2010, 22:19   #10231  |  Link
RainyDog
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Eac3to, or likely the ArcSoft decoder, doesn't seem to like my one channel DTS file. I want to convert it to 192kbps ac3 as one channel DTS seems like a massive waste of space. Is there anyway I can force eac3to to use a different DTS decoder or any other program I could try? Cheers in advance for any responses, the log is below.

Quote:
eac3to v3.22
command line: "C:\Program Files (x86)\eac3to\eac3to.exe" "S:\Temp\RipBot264temp\job2\2_audio_Japanese.dts" "S:\Temp\Profound.ac3" -192
------------------------------------------------------------------------------
DTS, 1.0 channels, 2:53:28, 24 bits, 768kbps, 48kHz
Decoding with ArcSoft DTS Decoder...
The ArcSoft DTS Decoder reported an error while decoding. <ERROR>
Aborted at file position 262144. <ERROR>
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Old 27th June 2010, 22:58   #10232  |  Link
hubblec4
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Quote:
Originally Posted by RainyDog View Post
Eac3to, or likely the ArcSoft decoder, doesn't seem to like my one channel DTS file. I want to convert it to 192kbps ac3 as one channel DTS seems like a massive waste of space. Is there anyway I can force eac3to to use a different DTS decoder or any other program I could try? Cheers in advance for any responses, the log is below.
In this case i use Megui to encode the dts-file.
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Old 28th June 2010, 03:15   #10233  |  Link
Snowknight26
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Quote:
Originally Posted by RainyDog View Post
Is there anyway I can force eac3to to use a different DTS decoder or any other program I could try? Cheers in advance for any responses, the log is below.
Check the first post, check the readme, check by running eac3to without any parameters. Kind of hard to miss.

Quote:
Originally Posted by hubblec4 View Post
In this case i use Megui to encode the dts-file.
He's encoding AC3, not DTS, so that wouldn't help.
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Old 28th June 2010, 03:36   #10234  |  Link
Midzuki
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The (new) avcodec.dll is working correctly now
( finally! ),
and is much faster
than the outdated one:

Code:
[C:\AUDIOEXZ\LAB]
=>eac3to input447.dts output447.wav -libav -down16
DTS, 2.0 channels, 0:01:17, 24 bits, 447kbps, 48kHz
Decoding with libav/ffmpeg...
Reducing depth from 64 to 16 bits...
Writing WAV...
Creating file "output447.wav"...
Clipping detected, a 2nd pass will be necessary.  <WARNING>
Starting 2nd pass...
Decoding with libav/ffmpeg...
Reducing depth from 64 to 16 bits...
Writing WAV...
Applying -0,13dB gain...
Creating file "output447.wav"...
eac3to processing took N ;-) seconds.
Done.
a lot, madshi!
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Old 28th June 2010, 07:19   #10235  |  Link
hubblec4
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Quote:
Originally Posted by Snowknight26 View Post


He's encoding AC3, not DTS, so that wouldn't help.
ok, i mean i use Megui to decode the dts-file and then encode to ac3.
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Old 28th June 2010, 13:10   #10236  |  Link
ACrowley
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@madshi

When you anyhow take a look into the DTS-HD Header because of the DialNorm Issue, can you add a Routine into eac3to to restore/strip the DTS-HD Header ?

..new libavcodec is faster on AC3 and EAC3, Great! EAC3/AC3/THD looks good with libav so far i can see

Last edited by ACrowley; 28th June 2010 at 14:44.
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Old 28th June 2010, 16:22   #10237  |  Link
Jeff Flowerday
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Quote:
Originally Posted by madshi View Post
eac3to v3.22 released

http://madshi.net/eac3to.zip

Code:
* fixed: reducing bitdepth from >16bit to 16bit resulted in static (v3.21)
* updated to latest ffmpeg/libav version
* libav MP1/MP2/MP3 decoders now decode to floating point
* libav E-AC3 decoder now supports spectral extension
The updated ffmpeg/libav libraries are quite a big change. It would be nice if you guys could do some libav decoder testing with various codecs (AC3, E-AC3, DTS, MP1, MP2, MP3, MLP, TrueHD) to make sure that things still work as intended.
I was having a problem with static from the TrueHD track in "Pale Rider", with 3.21. But all was fine with 3.22

Thanks for the fix!
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Old 28th June 2010, 18:53   #10238  |  Link
Laurent
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Quote:
Originally Posted by madshi View Post
eac3to v3.22 released

http://madshi.net/eac3to.zip

Code:
* fixed: reducing bitdepth from >16bit to 16bit resulted in static (v3.21)
* updated to latest ffmpeg/libav version
* libav MP1/MP2/MP3 decoders now decode to floating point
* libav E-AC3 decoder now supports spectral extension
The updated ffmpeg/libav libraries are quite a big change. It would be nice if you guys could do some libav decoder testing with various codecs (AC3, E-AC3, DTS, MP1, MP2, MP3, MLP, TrueHD) to make sure that things still work as intended.
Not yet tested but I can tell you that support of E-AC3 spectral extension in eac3to is a major enhancement for all people living in France.
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Old 28th June 2010, 20:43   #10239  |  Link
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Quote:
Originally Posted by madshi View Post
eac3to v3.22 released

* libav E-AC3 decoder now supports spectral extension[/code]
The updated ffmpeg/libav libraries are quite a big change. It would be nice if you guys could do some libav decoder testing with various codecs (AC3, E-AC3, DTS, MP1, MP2, MP3, MLP, TrueHD) to make sure that things still work as intended.
Decoding/Transcoding E-AC3 streams within spectral extension don't seem to work properly for me using libav decoder...

1/
Code:
eac3to input.ts 2:audio.ac3
Quote:
TS, 1 video track, 2 audio tracks, 2 subtitle tracks, 0:00:13, 50i
1: h264/AVC, 1440x1080 50i (16:9)
2: E-AC3, French, 5.0 channels, 256kbps, 48kHz, dialnorm: -27dB, -2140ms
3: E-AC3, qaa, 2.0 channels, 128kbps, 48kHz, dialnorm: -23dB, -2132ms
4: Subtitle (DVB), French
5: Subtitle (DVB), French
a02 The Nero decoder doesn't seem to work, will use libav instead.
a02 Extracting audio track number 2...
a02 Removing E-AC3 dialog normalization...
a02 Decoding with libav/ffmpeg...
a02 Remapping channels...
a02 Applying RAW/PCM delay...
a02 The AC3 encoder received a non-supported data format (float, 5, 64, -).
Aborted at file position 1048576.
Same thing for encoding directly from eac3 :
2 /
Code:
eac3 input.eac3 output.ac3
Quote:
E-AC3, 5.0 channels, 0:00:11, 256kbps, 48kHz
The Nero decoder doesn't seem to work, will use libav instead.
Decoding with libav/ffmpeg...
Remapping channels...
The AC3 encoder received a non-supported data format (float, 5, 64, -).
Aborted at file position 262144.
>> sample <<

All is fine using FFmpeg w/ ffac3 or piping aften...

Last edited by Kurtnoise; 28th June 2010 at 20:46.
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Old 28th June 2010, 21:03   #10240  |  Link
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Quote:
Decoding/Transcoding E-AC3 streams within spectral extension don't seem to work properly for me using libav decoder...
this work perfect for me:

Code:
eac3to input.ts 2:audio.wav
the result:

Code:
General
Complete name                    : F:\test_eac3_spectral.ts.wav
Format                           : Wave
File size                        : 7.36 MiB
Duration                         : 10s 724ms
Overall bit rate                 : 5 760 Kbps

Audio
Format                           : PCM
Format settings, Endianness      : Little
Format settings, Sign            : Unsigned
Codec ID                         : 00001000-0000-0100-8000-00AA00389B71
Codec ID/Hint                    : Microsoft
Duration                         : 10s 724ms
Bit rate mode                    : Constant
Bit rate                         : 5 760 Kbps
Channel(s)                       : 5 channels
Channel positions                : Front: L C R, Side: L R
Sampling rate                    : 48.0 KHz
Stream size                      : 7.36 MiB (100%)
the error in encoding to .ac3 is related to 5.0 channels...
_
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