eac3to - audio conversion tool

[Snowknight26]

Try something like this instead:
eac3to input.ts 2: stdout.wav | aften -b 320 - output.ac3

[jasonwc]

I found odd behavior with eac3to 3.22 and the Slumdog Millionaire US AVC DTS-HD MA Blu-Ray. I converted the DTS-HD MA 24 bit track to a 16 bit FLAC with eac3to 3.22 and got a FLAC with a bitrate of 1331 Kbps. I used MKVMerge to create a MKV, and found that the video stalled on playback using ffdshow (libavcodec) to decode the FLAC. I had no such issue with madFLAC or the internal MPC-HC decoder. However, I was able to replicate the behavior in MPC-HC and Mediaporal using the ffdshow FLAC decoder.

I thought the issue might be corruption in the outputted FLAC because the video stream without audio played fine. Therefore, I demuxed the BD again and got an identically sized FLAC. Same problem.

I decided to redo the remux with eac3to 3.19 and identical settings. The FLAC now had a bitrate of 1393 Kbps. After remuxing with MKVMerge, I discovered the problem was gone. The remuxed video played back perfectly without stalling upon MPC-HC startup. I'm not sure if this is an ffdshow or eac3to problem, but it's weird that the FLAC bitrate changed, and the problem was fixed. The FLAC track was identical after both demuxes with 3.22, and all three demuxes used the same version of the FLAC encoder (1.2.1).

eac3to 3.22 log:

eac3to v3.22
command line: "C:\Users\Jason\Downloads\eac3to\eac3to.exe" "F:\In Process\Slumdog Millionaire 2008 Blu-ray 1080p AVC DTS-HD 5.1\" 1) 2: "D:\In Process\video.2.mkv" -seekToIFrames 3: "D:\In Process\audio.english.3.flac" -down16 4: "D:\In Process\audio.english.4.ac3" 5: "D:\In Process\audio.english.5.ac3" 1: "D:\In Process\chapters.txt" 7: "D:\In Process\subtitle.english.7.sup" -log="L:\Movies\Slumdog Millionaire 2008 1080p BluRay AVC FLAC-Remux\Slumdog Millionaire 2008 1080p BluRay AVC FLAC-Remux.log.txt" -progressnumbers
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 3 subtitle tracks, 2:00:38, 24p /1.001
1: Chapters, 28 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
4: AC3, English, 2.0 channels, 224kbps, 48kHz, dialnorm: -27dB
5: AC3, English, 2.0 channels, 224kbps, 48kHz, dialnorm: -27dB
6: AC3, French, 5.1 channels, 448kbps, 48kHz, dialnorm: -27dB
7: Subtitle (PGS), English
8: Subtitle (PGS), French
9: Subtitle (PGS), Spanish
Creating file "D:\In Process\chapters.txt"...
[a04] Extracting audio track number 4...
[a05] Extracting audio track number 5...
[s07] Extracting subtitle track number 7...
[a05] Removing AC3 dialog normalization...
[v02] Extracting video track number 2...
[a04] Removing AC3 dialog normalization...
[a03] Extracting audio track number 3...
[v02] Muxing video to Matroska...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Reducing depth from 24 to 16 bits...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "D:\In Process\audio.english.3.flac"...
[a05] Creating file "D:\In Process\audio.english.5.ac3"...
[a04] Creating file "D:\In Process\audio.english.4.ac3"...
[s07] Creating file "D:\In Process\subtitle.english.7.sup"...
[a03] The original audio track has a constant bit depth of 24 bits.
[a03] The processed audio track has a constant bit depth of 16 bits.
Added fps value (24 /1.001) to MKV header.
Video track 2 contains 173544 frames.
Subtitle track 7 contains 1385 captions.
eac3to processing took 17 minutes, 37 seconds.
Done.

eac3to 3.19 log:

eac3to v3.19
command line: "C:\Users\Jason\Downloads\eac3to\eac3to.exe" "F:\In Process\Slumdog Millionaire 2008 Blu-ray 1080p AVC DTS-HD 5.1\" 1) 2: "D:\In Process\video.2.mkv" -seekToIFrames 3: "D:\In Process\audio.english.3.flac" -down16 4: "D:\In Process\audio.english.4.ac3" 5: "D:\In Process\audio.english.5.ac3" 1: "D:\In Process\chapters.txt" 7: "D:\In Process\subtitle.english.7.sup" -log="K:\Movies\Slumdog Millionaire 2008 1080p BluRay AVC FLAC-Remux\Slumdog Millionaire 2008 1080p BluRay AVC FLAC-Remux.log.txt" -progressnumbers
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 3 subtitle tracks, 2:00:38, 24p /1.001
1: Chapters, 28 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
4: AC3, English, 2.0 channels, 224kbps, 48kHz, dialnorm: -27dB
5: AC3, English, 2.0 channels, 224kbps, 48kHz, dialnorm: -27dB
6: AC3, French, 5.1 channels, 448kbps, 48kHz, dialnorm: -27dB
7: Subtitle (PGS), English
8: Subtitle (PGS), French
9: Subtitle (PGS), Spanish
Creating file "D:\In Process\chapters.txt"...
[s07] Extracting subtitle track number 7...
[v02] Extracting video track number 2...
[v02] Muxing video to Matroska...
[a05] Extracting audio track number 5...
[a05] Removing AC3 dialog normalization...
[a04] Extracting audio track number 4...
[a04] Removing AC3 dialog normalization...
[a03] Extracting audio track number 3...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Reducing depth from 24 to 16 bits...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "D:\In Process\audio.english.3.flac"...
[a04] Creating file "D:\In Process\audio.english.4.ac3"...
[a05] Creating file "D:\In Process\audio.english.5.ac3"...
[s07] Creating file "D:\In Process\subtitle.english.7.sup"...
[a03] The original audio track has a constant bit depth of 24 bits.
[a03] The processed audio track has a constant bit depth of 16 bits.
Added fps value (24 /1.001) to MKV header.
Video track 2 contains 173544 frames.
Subtitle track 7 contains 1385 captions.
eac3to processing took 17 minutes, 57 seconds.
Done.

[Laurent]

I noticed yesterday a similar problem with many macroblocks in video if I demux video in MKV file rather than a H.264 file. As my process is a 2 steps process using first eac3to and then TsMuxer to produce a M2TS file, I cannot be sure if the problem with video in MKV container is due to eac3to or TsMuxer.

[Midzuki]

@ jasonwc:

if neither MPC-HC's decoder nor madFlac complained, but ffdshow stumbled, then the problem is in ffdshow, not in eac3to, IMHO.
BTW, you didn't mention what version of ffdshow you're using.

{
Anyway, and JMNSHO again, ffdshow became even-buggier with the advent of the "DXVA-era". :devil: And I don't intend to talk again with people who believe certain *design flaws* "must be preserved" for "historical reasons". :rolleyes:
}

[jasonwc]

@ jasonwc:

if neither MPC-HC's decoder nor madFlac complained, but ffdshow stumbled, then the problem is in ffdshow, not in eac3to, IMHO.
BTW, you didn't mention what version of ffdshow you're using.

{
Anyway, and JMNSHO again, ffdshow became even-buggier with the advent of the "DXVA-era". :devil: And I don't intend to talk again with people who believe certain *design flaws* "must be preserved" for "historical reasons". :rolleyes:
}

I used two recent SVN builds - the problem persists with the latest build - 3488, released yesterday. I am using the software decoder, specifically ffmpeg-mt.

[tebasuna51]

something like this ?

eac3to.exe input.ts 2:output.stdout | aften.exe -readtoeof 1 - aften.ac3

if yes, this doesn't seem to work...
Try something like this instead:
eac3to input.ts 2: stdout.wav | aften -b 320 - output.ac3
Don´t forget the -readtoeof 1 because a wav 5.0 24 bits can be > 4GB.
eac3to.exe input.ts 2: stdout.wav | Aften.exe -readtoeof 1 - aften.ac3
The default aften bitrate for 5.0 is 448 Kb/s.

[Midzuki]

The default aften bitrate for 5.0 is 448 Kb/s.

Probably he was thinking of NeroAacEnc when he wrote the Aften command-line... :D

[tebasuna51]

No, the parameter -b 320 is correct. And maybe recommended because:
2: E-AC3, French, 5.0 channels, 256kbps, 48kHz, dialnorm: -27dB, -2140ms
We can select between ...,192,224,256,320,384,448,512,576,640 Kb/s.

[asarian]

Using eac3to, any way I can downconvert a ~25GB 192 Khz (!) LPCM track (Akira Blu-Ray) to something more human, like, say, 48 Khz?

[nurbs]

Yes you can do that. See the manual or the first post as to how.

[Thunderbolt8]

whats the best solution again to deal with 7.1 DTS-HD MA tracks which are described as "strange setup" ? I want to convert the track to flac, so what are my options here?

is this information here still correct? http://forum.doom9.org/showthread.php?p=1361493#post1361493 using sonic to convert it to 5.1 makes most sense atm?

[Laurent]

I noticed yesterday a similar problem with many macroblocks in video if I demux video in MKV file rather than a H.264 file. As my process is a 2 steps process using first eac3to and then TsMuxer to produce a M2TS file, I cannot be sure if the problem with video in MKV container is due to eac3to or TsMuxer.

I come back to this problem with more explanations and tests. In fact, the problem is not new, I just verified and I have the same strange behaviour with eac3to v3.21 and eac3to v3.18.

Here is my scenario:
1 - eac3to to demux video in MKV container and convert audio to AC3
2 - TsMuxer to mux audio and video in a M2TS container
3 - playback of the M2TS file with the PS3

Video is H.264. When the file is decoded by the PS3, I have many macroblocks, and the result is the same whatever the version of eac3to (3.18, 3.21 and 3.22).
If at step 1, instead of demuxing video in MKV container, I demux the video in a H.264 file, then the playback with the PS3 is fine without macroblocks.

Note that all these files (M2TS and MKV) are played correctly on the PC.

So the problem is relative to the usage of MKV container, either when produced by eac3to, or when used as input by TsMuxer. And the problem is noticeable only when the final file is played with the PS3.

Now that I discovered this problem, I will of course avoid using MKV container.

[b66pak]

eac3to use haali media splitter (http://haali.su/mkv/) for demuxing...try other version (11-1-2009 (http://www.videohelp.com/download/MatroskaSplitter110109.exe))

[Snowknight26]

No it doesnt.

[Laurent]

eac3to use haali media splitter (http://haali.su/mkv/) for demuxing...try other version (11-1-2009 (http://www.videohelp.com/download/MatroskaSplitter110109.exe))

This is the version I am using.

[asarian]

I come back to this problem with more explanations and tests. In fact, the problem is not new, I just verified and I have the same strange behaviour with eac3to v3.21 and eac3to v3.18.

Here is my scenario:
1 - eac3to to demux video in MKV container and convert audio to AC3
2 - TsMuxer to mux audio and video in a M2TS container
3 - playback of the M2TS file with the PS3

Video is H.264. When the file is decoded by the PS3, I have many macroblocks, and the result is the same whatever the version of eac3to (3.18, 3.21 and 3.22).
If at step 1, instead of demuxing video in MKV container, I demux the video in a H.264 file, then the playback with the PS3 is fine without macroblocks.

Note that all these files (M2TS and MKV) are played correctly on the PC.

So the problem is relative to the usage of MKV container, either when produced by eac3to, or when used as input by TsMuxer. And the problem is noticeable only when the final file is played with the PS3.

Now that I discovered this problem, I will of course avoid using MKV container.
Hmm, you wouldn't be using an older version of CoreAVC, would ya? What you describe is a known issue with it. Just make sure you let ffdshow handle H264, and you should be fine. I stopped using CoreAVC for this precise reason.

[Laurent]

Hmm, you wouldn't be using an older version of CoreAVC, would ya? What you describe is a known issue with it. Just make sure you let ffdshow handle H264, and you should be fine. I stopped using CoreAVC for this precise reason.

But the playback problem is with the PlayStation, not the PC. And when demuxing with eac3to, I don't think a decoder is involved.
By the way, CoreAVC is not installed on my PC.

[GoodzMastaJ]

I've extracted the TrueHD track from Evangelion 1.11 blu-ray using eac3to. ffdshow reports it as having channels as shown in the screenshot below.

http://stuff.damagedgoodz.net/forumposts/7chtruehd.png

Note the presence of the side channels and absence of an LFE channel.
In actuality, there are no side channels but there is an LFE and a rear center channel which I suppose would make this a 6.1 track (rather than the 7.0 I'm getting from ffdshow). Using the mixer, I found the LFE channel is playing from what ffdshow thinks is Back Left (all that comes out of BackL is rumbling).

I'm not sure where the bug is, if it's in eac3to or ffdshow. Any ideas on how to pinpoint it? I made a similar post on the ffdshow-tryouts forum to stir up some suggestions there as well.

Here is a short sample that has a pretty good demonstration.
http://www.mediafire.com/?mmzydemji5m

[Killroy™]

Using eac3to, any way I can downconvert a ~25GB 192 Khz (!) LPCM track (Akira Blu-Ray) to something more human, like, say, 48 Khz?

OK... I'll be the first to ask... Why would you want to take such a beautiful audio track and throw away 2/3 of the data?

[Atak_Snajpera]

the same reason why 24bit is converted to 16bit. Human hearing has limitations.

[dansrfe]

Currently, are there any speaker systems/receiver's that actually support 192Khz? Now THAT would be the 9th Wonder of the World.

[Killroy™]

Currently, are there any speaker systems/receiver's that actually support 192Khz? Now THAT would be the 9th Wonder of the World.

Most modern AVR's support 192Khz with no problems.

[Midzuki]

Currently, are there any speaker systems/receiver's that actually support 192Khz?

Digital: In the world of today, 192kHz audio should already be something quite trivial. :)

Analog: In the world of today, speakers and audio amplifiers that support up to 96kHz still are very rare, and (IMHO) somewhat pointless...

I'm not sure where the bug is, if it's in eac3to or ffdshow.

Both eac3to and ffdshow use libavcodec for TrueHD decoding,
however I daresay eac3to "is less unreliable than" ffdshow. :devil:
I'd recommend converting the original 6.1 audio to a 6.1 .WAV
(assuming disk space is not an issue...)

Evangelion 1.11

I didn't see it, and I didn't like it. :D

(Yes man, quite often the remakes are quite inferior to the original versions.)

[dansrfe]

You know this might be a n00b question but I still don't understand exactly what a digital speaker is and what connectors are used with it. In a normal home theater setup where, for example, a htpc is outputting digital S/PDIF audio via HDMI (coming from the sound card or mobo sound module built-in to the graphics card), then is transmitted to the receiver all through a single HDMI cable, then then the receiver processes and decodes the audio if bitstreaming. But after that doesn't the receiver send a analog signal to the actual speakers? I'm totally confused about this. What exactly is a digital speaker and what is an analog speaker? What sort of connectors does a digital speaker have?

[Midzuki]

Thanks to some "oddities" of the English language, and also thanks to the marketing people :devil:, the word "speakers" became somewhat... undefined. :D From what I've read so far, it's never very clear when the word loudspeaker refers to the (so-called) "driver" itself, or to the "box" (enclosure) which contains one or more "drivers"... :confused: Anyway, usually the expression "digital speakers" applies to a set of "powered" drivers which includes a Digital2Analog converter. ACTUAL digital speakers are an entirely-different beast, and there is an article on Wikipedia (http://en.wikipedia.org/wiki/Digital_speakers) about them. HTH.

[Killroy™]

You know this might be a n00b question but I still don't understand exactly what a digital speaker is and what connectors are used with it. In a normal home theater setup where, for example, a htpc is outputting digital S/PDIF audio via HDMI (coming from the sound card or mobo sound module built-in to the graphics card), then is transmitted to the receiver all through a single HDMI cable, then then the receiver processes and decodes the audio if bitstreaming. But after that doesn't the receiver send a analog signal to the actual speakers? I'm totally confused about this. What exactly is a digital speaker and what is an analog speaker? What sort of connectors does a digital speaker have?

By "digital" we usually mean HDMI connectors. LPCM or bitstream of the audio source. "Analog" usually refers to the old multi-channel RCA-plugs used in older AVR's. Audio from the HTPC or STB comes from the analog output (HTPC) or the multi-channel analog outputs of the STB's. Since only very expensive HTPC analog cards could output 192Khz (properly) it is unsure if you will get a un-molested 192Khz output.

No such issues with the HDMI outputs.

The signal from the AVR to the speakers is always the same (analog) but can be degraded with lower quality wire. But even modest 24GA wires should have no problems with 192Khz at very short throws. Always use 12GA wire for optimum quality.

[dansrfe]

ah. I had this idea that digital speakers were some obscure, advanced, audio professional type stuff.

[Xorp]

Decoding this 6.1 DTS-MA track with eac3to 3.22 and Arcsoft 1.1.0.7 results in distortion: http://stfcc.org/misc/7channels.dtshd

[tebasuna51]

Decoding this 6.1 DTS-MA track with eac3to 3.22 and Arcsoft 1.1.0.7 results in distortion: http://stfcc.org/misc/7channels.dtshd
Read this post (http://forum.doom9.org/showthread.php?p=1409645#post1409645) and next (the channelmapping is solved with eac3to v3.21)

Test also my sample in the post using Arcsoft 1.1.0.0.

[tebasuna51]

I've extracted the TrueHD track from Evangelion 1.11 blu-ray using eac3to. ffdshow reports it as having channels as shown in the screenshot below.

Note the presence of the side channels and absence of an LFE channel.
In actuality, there are no side channels but there is an LFE and a rear center channel which I suppose would make this a 6.1 track (rather than the 7.0 I'm getting from ffdshow). Using the mixer, I found the LFE channel is playing from what ffdshow thinks is Back Left (all that comes out of BackL is rumbling).

I'm not sure where the bug is, if it's in eac3to or ffdshow. Any ideas on how to pinpoint it? I made a similar post on the ffdshow-tryouts forum to stir up some suggestions there as well.
Seems a ffdshow problem because eac3to decode the TrueHD fine to a wav 6.1 with MaskChannel 0x070F (FL FR FC LFE BC SL SR).
Try play the wav file.

[b66pak]

WARNING...the new mkvtoolnix 4.1.0 "header removal compression" (http://www.bunkus.org/videotools/mkvtoolnix/faq.html#header_removal_compression) make impossible decoding/demuxing for eac3to from mkv...

eac3to v3.22
command line: eac3to "mkvtoolnix.4.1.0.mkv" -demux
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 1 subtitle track, 0:01:16, 24p /1.001
1: h264/AVC, 1280x536 23.975p (160:67)
2: DTS, 5.1 channels, 48kHz
3: Subtitle (SRT)
[v01] The video bitstream is encoded in a non-standard framerate. <WARNING>
[v01] The video bitstream framerate field doesn't match the container framerate. <WARNING>
Bitstream parsing for track 2 failed. <WARNING>
Demuxing this track may still produce correct results - or not. <WARNING>
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[a02] Creating file "mkvtoolnix.4.1.0 - 2 - DTS, 5.1 channels, 48kHz.dts"...
[v01] Creating file "mkvtoolnix.4.1.0 - 1 - h264, 1280x536 23.975p.h264"...
[s03] Extracting subtitle track number 3...
[s03] Creating file "mkvtoolnix.4.1.0 - 3 - Subtitle (SRT).srt"...
Video track 1 contains 1819 frames.
eac3to processing took 2 seconds.
Done.


eac3to v3.22
command line: eac3to "mkvtoolnix.4.1.0 - 2 - DTS, 5.1 channels, 48kHz.dts" mkvtoolnix.4.1.0.wav
The format of the source file could not be detected.

_

[dansrfe]

Can anyone please post a simple and effective guide to getting Nero 7 to work with eac3to. I have the software and have bought it with the Nero Blu-ray / HD DVD plugin but I do not want to install it just for a few plugins. I prefer extracting/using only what is necessary for eac3to to utilize nero 7 for AC3 decoding. Another thing I do not understand about AC3 encoding with eac3to and libav is that why does it say "Reducing bit depth from 64bits to 24 bits" while processing?

[XhmikosR]

You can use Nero Lite/Micro. http://forum.doom9.org/showthread.php?p=1398214&highlight=lite#post1398214

If you want to experiment, you can manually find the needed files to get it to work.

[ACrowley]

Can anyone please post a simple and effective guide to getting Nero 7 to work with eac3to. I have the software and have bought it with the Nero Blu-ray / HD DVD plugin but I do not want to install it just for a few plugins. I prefer extracting/using only what is necessary for eac3to to utilize nero 7 for AC3 decoding. Another thing I do not understand about AC3 encoding with eac3to and libav is that why does it say "Reducing bit depth from 64bits to 24 bits" while processing?

I wouldnt use NERO AC3/EAC3 Decoder
http://forum.doom9.org/showthread.php?p=1404212#post1404212
Looks like DRC isnt disabled properly

Better use libav..works without Problems. And i dont think you hear big a Quality Difference generally between Nero and libavcodec on AC3/EAC3

"Reducing bit depth from 64bits to 24 bits" while processing means that the libanvcodec internal processing runs at 64Bit and eac3to reduces the output to 24bit. Its not a Problem ,its just the Process

[shogo_kawada]

Hi, I tried to post these questions in the newbies forum but I wasn't lucky, so I'll try again here, which maybe would have been the right place to begin with.

I'm willing to rip my HD-DVDs to multi-audio MKVs. I have an AV-receiver which can decode HD-audio, and the files will be played on a Dune Base HD, which, AFAIK, CAN bitstream HD audio from MKVs, so I don't need any kind of encoding nor transcoding of the tracks. I just need to demux them. So here are my doubts:

1. About dialog normalization. Should I keep it or not? If not, why is it there (in the disc) in the first place? I'd like the MKVs to sound the closest to the actual disc playback I can get.

2. About delay. I read in a guide that eac3to doesn't fix delay when simply demuxing. Is that true? I tried some demuxing and the log said delay WAS removed. Did I do anything wrong or is it supposed to happen even with simple demux?

3. Just to be sure: to simple demux I just have to put *.eac3, *.dtshd, *.thd as name for the target track, am I right?

Ok, that's it, thanks for your help :)

[nibus]

Can eac3to demux DTS Express audio tracks?

[TinTime]

1. About dialog normalization. Should I keep it or not? If not, why is it there (in the disc) in the first place? I'd like the MKVs to sound the closest to the actual disc playback I can get.

I'd keep it but this is just a matter of personal preference. It only affects the volume of the decoded audio.

2. About delay. I read in a guide that eac3to doesn't fix delay when simply demuxing. Is that true? I tried some demuxing and the log said delay WAS removed. Did I do anything wrong or is it supposed to happen even with simple demux?

It will fix the delay for everything except for TrueHD tracks where the delay required will be added to the output filename.

3. Just to be sure: to simple demux I just have to put *.eac3, *.dtshd, *.thd as name for the target track, am I right?

Yes, or you can just use the -demux switch.

[mini-moose]

sorry if this was discussed before but I can't find it.

I'm trying to convert a 24.000fps DTS to 23.976fps.

when using -slowdown eac3to assumes my source is 25.000 thus the slowdown is by 4% which is not what I'm aiming for.

I tried to define the fps of the DTS by creating a copy using the -24.000 switch. then ran the slowdown again and it still treated it as 25.000.

-changeto didn't seem to work either:

"C:\eac3to\eac3to.exe" C:\one.dts two.dts -changeTo24.000
Was asked to modify source to 24.000, but the original FPS value is unknown.
Please specify the original FPS value (e.g. option "-23.976").

"C:\eac3to\eac3to.exe" C:\one.dts two.dts -24.000 -changeTo24.000
"C:\eac3to\eac3to.exe" C:\one.dts two.dts -24.000 -changeTo23.976
both got me a "Please specify the source and dest files first and then the options."

I guess I'm doing something wrong. Hopefully someone can help :)

[TinTime]

"C:\eac3to\eac3to.exe" C:\one.dts two.dts -24.000 -changeTo23.976

I can't see anything wrong with this. Does it work if you put quotes around the filenames?

Or how about two steps?
"C:\eac3to\eac3to.exe" C:\one.dts two.wav -24.000 -changeTo23.976
"C:\eac3to\eac3to.exe" two.wav two.dts
Which step fails?

[mini-moose]

[QUOTE] TinTime
I can't see anything wrong with this. Does it work if you put quotes around the filenames?

Or how about two steps?
Code:

"C:\eac3to\eac3to.exe" C:\one.dts two.wav -24.000 -changeTo23.976
"C:\eac3to\eac3to.exe" two.wav two.dts


thanks. I will give it a try. but would that even do the slowdown I need if it works ? I figured I might have to "tag" the original dts somehow as 24fps so -slowdown will perform a slowdown from 24 tp 23.976 and from 25.

[mini-moose]

TinTime:
I can't see anything wrong with this. Does it work if you put quotes around the filenames?

Or how about two steps?
Code:

"C:\eac3to\eac3to.exe" C:\one.dts two.wav -24.000 -changeTo23.976
"C:\eac3to\eac3to.exe" two.wav two.dts


thanks. I will give it a try. but would that even do the slowdown I need if it works ? I figured I might have to "tag" the original dts somehow as 24fps so -slowdown will perform a slowdown from 24 tp 23.976 and from 25.

[tebasuna51]

.... but would that even do the slowdown I need if it works ? I figured I might have to "tag" the original dts somehow as 24fps so -slowdown will perform a slowdown from 24 tp 23.976 and from 25.
An audio file (dts, ac3, ...) don't have tags about the video fps, only have a duration. And aply a -24.000 -changeTo23.974 means only modify the duration with a coefficient of 1.001

With audio files -slowdown means modify the duration with a coefficient of (1001/960) always.

Use:
"C:\eac3to\eac3to.exe" "C:\one.dts" "C:\two.ac3" -24.000 -changeTo23.976
or
"C:\eac3to\eac3to.exe" "C:\one.dts" "C:\two.wavs" -24.000 -changeTo23.976
and use a external dts encoder.

[shogo_kawada]

I'd keep it but this is just a matter of personal preference. It only affects the volume of the decoded audio.
Ok, that's what I was thinking too, but just for the sake of knowledge: is there a reason for dialog normalization to be there (in the disc) in the first place? Why do most of people remove it? Why do studios put it there?

It will fix the delay for everything except for TrueHD tracks where the delay required will be added to the output filename.
Ok, I just tried to demux some DD+ tracks and indeed the delays were fixed, but the log informed me that a remaning delay of -8 ms couldn't be fixed. To fix the last unfixed gap, can I just add that exact delay in the track properties in MKVMerge?


Yes, or you can just use the -demux switch.
It's surely much simpler this way since I can avoid the very long command line. Just a doubt: will the video track be handled properly this way? I read somewhere that it would be better to demux video tracks to mkv for eac3to to properly deal with framerate, pulldown flags, and stuff like this (which I actually don't really get, but still...).

Thanks a lot for your help ;)

[TinTime]

Ok, that's what I was thinking too, but just for the sake of knowledge: is there a reason for dialog normalization to be there (in the disc) in the first place? Why do most of people remove it? Why do studios put it there?

The reason it's there is to make all audio from different sources sound roughly the same volume. The problem is that it's not universally used. When you ask "why do studios put it there" it might be more reasonable to ask "why do some studios not use it".


Ok, I just tried to demux some DD+ tracks and indeed the delays were fixed, but the log informed me that a remaning delay of -8 ms couldn't be fixed. To fix the last unfixed gap, can I just add that exact delay in the track properties in MKVMerge?

You can if you think you can hear an 8ms delay! Different audio types come with different frame sizes which are the minimum chunks you can edit with. EAC3 is (I think) the same as AC3 in that it has a 32ms frame size. So if the original audio is 40ms out of sync then the closest you can get to fixing the delay is to take 32ms off, leaving a delay of 8ms that can't be fixed.


It's surely much simpler this way since I can avoid the very long command line. Just a doubt: will the video track be handled properly this way? I read somewhere that it would be better to demux video tracks to mkv for eac3to to properly deal with framerate, pulldown flags, and stuff like this (which I actually don't really get, but still...).

I've got a feeling that there used to be a problem with mkvmerge and raw AVC streams, or something like that, but there isn't any more. I always demux my video with eac3to and then mux it to mkv with mkvmerge and I don't have any problems though.

[shogo_kawada]

The reason it's there is to make all audio from different sources sound roughly the same volume. The problem is that it's not universally used. When you ask "why do studios put it there" it might be more reasonable to ask "why do some studios not use it".

Ok so it's something which DO have some reason to be there and for us to keep when demuxing. That's the answer I was looking for, -keepdialnorm it is!


You can if you think you can hear an 8ms delay! Different audio types come with different frame sizes which are the minimum chunks you can edit with. EAC3 is (I think) the same as AC3 in that it has a 32ms frame size. So if the original audio is 40ms out of sync then the closest you can get to fixing the delay is to take 32ms off, leaving a delay of 8ms that can't be fixed.

Eheh you're right, I don't think I can hear an 8ms delay. BTW thanks again for the explanation, and if I get it right that means that I'll never end with a delay over 31 ms, which I think won't be an issue either.


I've got a feeling that there used to be a problem with mkvmerge and raw AVC streams, or something like that, but there isn't any more. I always demux my video with eac3to and then mux it to mkv with mkvmerge and I don't have any problems though.
Ok, so I got nothing left to worry :). It's time to start some conversion :thanks:

[SomeJoe]

Eheh you're right, I don't think I can hear an 8ms delay. BTW thanks again for the explanation, and if I get it right that means that I'll never end with a delay over 31 ms, which I think won't be an issue either.

Actually, maximum possible unfixable offset is 16 msec because if the offset in one direction is over 16 msec, then remove/add one frame will give 32-(original offset) msec offset in the other direction.

So if audio is 20 msec behind video, then remove one audio frame and now audio is 12 msec ahead of video.

[shogo_kawada]

Actually, maximum possible unfixable offset is 16 msec because if the offset in one direction is over 16 msec, then remove/add one frame will give 32-(original offset) msec offset in the other direction.

So if audio is 20 msec behind video, then remove one audio frame and now audio is 12 msec ahead of video.
That's even better, awesome! Definetely no need to worry about the audio offset then.

[Thunderbolt8]

WARNING...the new mkvtoolnix 4.1.0 "header removal compression" (http://www.bunkus.org/videotools/mkvtoolnix/faq.html#header_removal_compression) make impossible decoding/demuxing for eac3to from mkv...


_is this still a problem with eac3to (or for players)?

[tebasuna51]

is this still a problem with eac3to (or for players)?
Is a problem for eac3to (at least DTS tracks, AC3 seems ok).

PC players tested without problems: mpc-hc(Haali), vlc and KMP

Standalone player tested without problems: Xtreamer

BTW, I don't know how can compress AC3 headers. There are different values in headers: CRC's, RF protection, DRC values, ...

[dbone1026]

Madshi,

I have a Blu Ray disc of a tv show (Supernatural Season 1, Disc 1). The disc has the first 6 episodes of the tv show. When I run the disc through eac3to (using Clown_BD and/or Ripbot) eac3to only picks up 3 playlists (the 1st playlist is of all episodes, the 2nd playlist is of episode 1, and the 3rd playlist is of episode 2). Episode 3-6 don't show anywhere. If I use MakeMKV I can see all episodes listed out individually to select. Is there anything you can think of that would be causing this issue? I can just copy the individual episodes which are each m2ts files from the disc to my harddrive and then from there run through ripbot so no big deal, but thought it worth asking. Below is my eac3to log, let me know if I can provide anything else.

eac3to v3.22
command line: "C:\Program Files\Clown_BD_v0.76\eac3to\eac3to.exe" "F:" -progressnumbers -LOG="C:\Users\DAMIAN\Documents\Clown_BD\LOGS\eac3to_PASS1_LOG.LOG"
------------------------------------------------------------------------------
1) 00100.mpls, 4:20:33
[34+35+36+37+38+39].m2ts
- Chapters, 41 chapters
- VC-1, 1080p24 /1.001 (16:9)
- AC3, English, multi-channel, 48kHz
- AC3, French, stereo, 48kHz
- AC3, German, stereo, 48kHz
- AC3, Spanish, stereo, 48kHz
- AC3, Portuguese, stereo, 48kHz
- AC3, Japanese, stereo, 48kHz
- AC3, Japanese, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz

2) 00401.mpls, 00034.m2ts, 0:46:25
- Chapters, 6 chapters
- VC-1, 1080p24 /1.001 (16:9)
- AC3, English, multi-channel, 48kHz
- AC3, French, stereo, 48kHz
- AC3, German, stereo, 48kHz
- AC3, Spanish, stereo, 48kHz
- AC3, Portuguese, stereo, 48kHz
- AC3, Japanese, stereo, 48kHz
- AC3, Japanese, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz

3) 00402.mpls, 00037.m2ts, 0:42:11
- Chapters, 7 chapters
- VC-1, 1080p24 /1.001 (16:9)
- AC3, English, multi-channel, 48kHz
- AC3, French, stereo, 48kHz
- AC3, German, stereo, 48kHz
- AC3, Spanish, stereo, 48kHz
- AC3, Portuguese, stereo, 48kHz
- AC3, Japanese, stereo, 48kHz
- AC3, Japanese, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz
- AC3, English, stereo, 48kHz