After importing ac3 in AA3 the file is truncated

[MicoMaco]

I demuxed mkv file with tsMuxer 1.10.6 in order to extract ac3 file. It's a 48 kHz 640 kbps ac3 file. When importing this demuxed ac3 file into Adobe Audition 3 I get this error message:

"end of file found too early, data will be truncated".

The duration of the source ac3 file was about 137 min in duration, after importing it into Adobe Audition 3 I end up with only 13 mins of file. Any ideas why this is happening? If someone knows how to overcome this issue please let me know; since I'm inexperienced I'd appreciate the details.

Thank you in advance!

MicoMaco

[tebasuna51]

Use mkvextract or eac3to (put the log if there are problems) to extract ac3 from mkv.

[setarip_old]

@tebqasuna51

Hi!

Your post here implies that "tsMuxeR" should not be used for extracting an .AC3 file from an .MKV.

Be advised that I have done this, without error, hundreds of times...

[tebasuna51]

Your post here implies that "tsMuxeR" should not be used for extracting an .AC3 file from an .MKV.

Be advised that I have done this, without error, hundreds of times
Maybe, but while TsMuxer can't extract wav files without errors I can't trust in TsMuxer like demuxer.

If you extract an ac3 from a mkv with TsMuxer you lost any delay for this track. If you use MkvExtract you can extract the Timecodes to know if there are delay. If you use eac3to the delay is inserted, like ac3 silence, to the extracted ac3 and DialNorm is cancelled (you can disable this).

BTW, use different soft to extract can confirm if is a corrupt original mkv file or/and can obtain a error message with some clue about the problem.

[MicoMaco]

I extracted ac3 from mkv file with both, tsMuxer and mkvExtract. The extracted files play fine (both), their duration match the original movie length plus the files match in size, yet when I try to import them in AA3 the files get truncated. At extracting there were no error messages created, only that the files were extracted successfully. Could it be maybe that AA3 can't properly handle ac3 files encoded at 640 kbps (anything above 448 kbps)? For this reason I'll try to create an ac3 (5.1 640 kbps 48 kHz) file from some stereo source with approximately same duration and than try importing it in AA3. Will see what happens.

[Ghitulescu]

Maybe there were transmission errors (usually reflected as nulls(0) in the body of the file), and while demuxers work (as they don't look inside) other programs may have problems.

What is the source of your MKV?

[b66pak]

@MicoMaco check it with delaycut and post the log!

http://forum.doom9.org/showthread.php?s=&threadid=71545
_

[MicoMaco]

@MicoMaco check it with delaycut and post the log!

http://forum.doom9.org/showthread.php?s=&threadid=71545
_


Hmm... I just opened the file with delaycut and processed it (didn't change any settings, they were:

Cut file - unchecked
Delay/Original length - unchecked
CRC errors - Silence)

After processing this is the only log I could see:

====== INPUT FILE INFO ========================
File is ac3
Bitrate (kbit/s) 640
Act rate (kbit/s) 640.000
File size (bytes) 643768320
Channels mode 3/2: L+C+R+SL+SR
Sampling Frec 48000
Low Frec Effects LFE: Present
Duration 02:14:07.104
Frame length (ms) 32.000000
Frames/second 31.250000
Num of frames 251472
Bytes per Frame 2560.0000
Size % Framesize 0
CRC present: YES
=============================================
====== TARGET FILE INFO ======================
Start Frame 0
End Frame 251471
Num of Frames 251472
Duration 02:14:07.104
NotFixedDelay 0.0000
=============================================
====== PROCESSING LOG ======================


I'm still testing custom made ac3 files with different bitrates, so far the ones exceeding 448 kbps (in my experiment I created one 640 kbps 5.1 48 kHz and one 512 kbps 5.1 48 kHz file) are truncated at importing in Adobe Audition 3. I have to create just one file with bitrate 448 kbps and than we'll see, maybe the problem isn't the file itself but AA3 which can't import ac3 files with bitrates exceeding 448 kbps.

to be continued...

[MicoMaco]

OK, I think I pinpointed the problem. As long as the file is ac3 file it doesn't pose a problem for importing into Adobe Audition 3. It doesn't matter if its bitrate is 640 kbps, 512 kbps or 448 kbps. What matters unfortunately is its duration, not the size of the file but its duration. I was playing with hex editor in VirtualDub and extracting different sizes of audibly the same file but encoded at different bitrates (meaning it was the same soundtrack but encoded at 640, 512 and 448 kbps bitrates). So I created dozens of files with different sizes and different durations, of course they were also encoded at different bitrates. No matter what was the size of the file or encoded bitrate, the files could be properly imported into Adobe Audition 3 if their duration was equal/under 2:04:11,443, however as soon as their duration matched/exceeded 2:04:31,104 the files were truncated and only small part of the file was imported.

Now when I pinpointed the problem maybe you could give me some advice how to overcome this nuisance at importing files into Adobe Audition 3. I've already found a way around, but it includes time consuming solution since I have to use delaycut.exe to chop original file into 2 parts which can than be imported into AA3...

[b66pak]

what do you want to do in AA3?
_

[MicoMaco]

what do you want to do in AA3?
_

It's simple basically, since DVD players can play only 448 kbps ac3 I need to downconvert them to 448 kbps. I use AA3 to create 6 mono wav files to be imported into sony vegas where I make a DVD standard ac3 file. I tried other methods than using AA3/sony vegas combo but files were just bad. So I ended up with sony vegas which uses original Dolby Digital encoder.

[tebasuna51]

Use eac3to command line with this syntax:

eac3to "ORIGINAL.mkv" 2: mono.wavs

Where:
"ORIGINAL.mkv" can be something like "d:\path\file name.mkv"
2: is the track number you want extract (first audio) or 3: if is second audio ...
and you obtain the 6 monowavs you want (use 'wavs' to obtain mono wavs)

Use HdBrStreamExtractor (GUI for eac3to) if you don't want use command line.

[MicoMaco]

Thank you very much tebasuna51, this info really helped!

[b66pak]

also you can do:


eac3to my640kb.ac3 mynew448kb.ac3 -448

or

eac3to my640kb.ac3 mynew384kb.ac3 -384
_

[MicoMaco]

also you can do:


eac3to my640kb.ac3 mynew448kb.ac3 -448

or

eac3to my640kb.ac3 mynew384kb.ac3 -384
_


Thx for the the advice, will try that too as soon as possible. However I'm a bit worried about that, I already tried only the downsample option in eac3to (in gui) but when cmd window opened and the process started there was an error message that this type of conversion cannot be done. I'm not sure but have a feeling that my source is 24 bit and that this is causing problems. Will try the direct cmd option you're suggesting.

[tebasuna51]

I was thinking you want use a certified ac3 encoder (Vegas) instead the free Aften.
Of course you can use the direct reencode like b66pak say. Or directly:
eac3to "ORIGINAL.mkv" 2: new.ac3 -448

Your source is ac3 5.1, 48 KHz, 640 Kb/s then isn't 24 bit per sample because:
bits / sample = (bits/second) / (channels x samples/second)
640000 / (6 x 48000 ) = 2.22

When use lossy compression the samples are translated from time domain to frequency domain and you can't restore the original precission. Only lossless compression can do it.

[MicoMaco]

@ tebasuna51

I still want and will use a certified ac3 encoder but it's always good to know all the options. I tried to reencode those 640 kbps ac3 files before but had no luck if I was using eac3to GUI. Today 640-->448 reencoding using a cmd line ended without any problem.Thx for the 24 bit per sample enlightening.

Do you still remember advising me to use HdBrStreamExtractor to properly extract the 5.1 ac3 file as 6 wavs? Created wavs were 24 bit per sample :) even though original ac3 is 16 bit. It's no problem for me cause I can easily downsample it, but still it's weird.

[tebasuna51]

... Created wavs were 24 bit per sample :) even though original ac3 is 16 bit...
I say you ac3 isn't 16 bit.
Use at least 24 bit for decoded ac3/dts, don't downsample feed the new encoder with 24 bit int or 32 bit float.

[MicoMaco]

I say you ac3 isn't 16 bit.
Use at least 24 bit for decoded ac3/dts, don't downsample feed the new encoder with 24 bit int or 32 bit float.

Huh... I didn't know that :thanks: . However, just out of curiosity I did exactly what you're suggesting and ac3 encoding ended perfectly :). Thx for stearing me into the right direction.

P.S. About this 16 bit and ac3. I got confused about this cause every time I was importing 5.1 ac3 into Adobe Audition 3 it said that it was 16 bit stereo file. I knew it was wrong about the stereo part but thought at least 16 bit was correct ;)

[ka81]

OK, I think I pinpointed the problem. As long as the file is ac3 file it doesn't pose a problem for importing into Adobe Audition 3. It doesn't matter if its bitrate is 640 kbps, 512 kbps or 448 kbps. What matters unfortunately is its duration, not the size of the file but its duration. I was playing with hex editor in VirtualDub and extracting different sizes of audibly the same file but encoded at different bitrates (meaning it was the same soundtrack but encoded at 640, 512 and 448 kbps bitrates). So I created dozens of files with different sizes and different durations, of course they were also encoded at different bitrates. No matter what was the size of the file or encoded bitrate, the files could be properly imported into Adobe Audition 3 if their duration was equal/under 2:04:11,443, however as soon as their duration matched/exceeded 2:04:31,104 the files were truncated and only small part of the file was imported.

Now when I pinpointed the problem maybe you could give me some advice how to overcome this nuisance at importing files into Adobe Audition 3. I've already found a way around, but it includes time consuming solution since I have to use delaycut.exe to chop original file into 2 parts which can than be imported into AA3...

same problem.
audio-files with duration 2:07:XX are not importing into Adobe Audition.


Have you found the solution??

[Overdrive80]

What version of AA are you using?? Its recommended version CC 2014.

[ka81]

What version of AA are you using?? Its recommended version CC 2014.
what is it?
where to look at version number?

[Overdrive80]

Google answer: here (https://www.google.es/search?q=adobe+audition+cc+2014&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:es-ES:official&client=firefox-a&channel=nts&gfe_rd=cr&ei=uVEgVIi0BvOs8weWhYHoDg)

[ka81]

Google answer: here (https://www.google.es/search?q=adobe+audition+cc+2014&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:es-ES:official&client=firefox-a&channel=nts&gfe_rd=cr&ei=uVEgVIi0BvOs8weWhYHoDg)ohh... stupid me.. ))

AA = Adobe Audition

i got installed Adobe Audition 3.0.1 build 8347.0

[tebasuna51]

Like Adobe Audition is a commercial soft you must report your problem in http://www.adobe.com

The Micomaco problem was solved with eac3to, but I don't know for what you need import your audio.

Please inform (MediaInfo) about your input audio file and what do you want make.

[ka81]

Please inform (MediaInfo) about your input audio file and what do you want make.
for example, ANY type of mediaformat files with duration 2:07:22

ONLY wav-files with that duration are importing ok!!!

[tebasuna51]

When AA3 import other files than WAV the first operation is decompress the file to PCM samples (WAV).
The audio editors only work with PCM samples, then the problem is in the decoders inside AA3 and we can't do anything with this, report the problem to Adobe.

The workaround is decompress your audio to WAV with external decoders before load in AA3.
There are free audio decoders for many audio formats (ffmpeg, eac3to, ...).