eac3to - audio conversion tool

[mrr19121970]

I don't know. Why do people keep asking me questions about tsMuxeR?

We're hoping that one day we can do something like this:

eac3to.exe d:\ 1) 2: movie.h264 3: english.ac3 --> movie.m2ts

:beg: :beg:

[piratburner]

We're hoping that one day we can do something like this:

eac3to.exe d:\ 1) 2: movie.h264 3: english.ac3 --> movie.m2ts

:beg: :beg:

That's should be something :cool:

[Thunderbolt8]

just ask the tsmuxer guys to fix their bugs and you should be fine

edit: yeah, and ask them to implement lossless -> ac3 & dts decoding :P

[alc0re]

Dialnorm can be set to any value between 0 and 31. According to the AC3 specification both 0 and 31 means: No dialnorm processing. Now any dialnorm processing *lowers* the volume of the audio track. That means removing the dialnorm (which is what eac3to is doing) should result in *higher* volume. Currently eac3to sets dialnorm to 0. Unfortunetely dialnorm set to 1 means lowering volume a lot. So incorrectly working decoders might think that a dialnorm value of 0 means even lower volume than dialnorm 1. But the documentation clearly states that a dialnorm value of 0 shall be treated as "no dialnorm processing" (which means max volume). And all the PC AC3 decoders correctly see value 0 as "dialnorm processing deactivated".


IMHO the decoders in the PS3 and Panasonic are not working correctly. Or maybe my AC3 specification is outdated? Anyway, the documentation clearly says that dialnorm 0 is "reserved". So I think it's not really good that eac3to uses it. That means I'll change it to 31 in the next build. I think that should fix the problem you're seeing. However, I believe to remember that some Sony Blu-Rays had a dialnorm value of 0, too. Well, anyway...

So is there a command in eac3to to do the reverse that the dialog normalization does? As in re-apply the normalization? Say an audio track had -27 dialog normalization. If I process the .ac3 track with +27db switch (that's already had the dialog normalization removed) will that reverse the process?

Does dialog normalization only decrease the volume of the center channel? Or is it a flat volume decrease across all channels? I ask because then the +db switch might work to accomplish my goal.

Let's say you tell me the +db command will accomplish exactly what I'm trying to accomplish...do I need any of the 3rd party encoders? Because up to this point I have not needed them since I'm just extracting the audio not converting it. I'm assuming there's no conversion done with you apply a +/- audio gain, but I may be wrong.

[n0mag!c]

Dialnorm can be set to any value between 0 and 31. According to the AC3 specification both 0 and 31 means: No dialnorm processing. Now any dialnorm processing *lowers* the volume of the audio track. That means removing the dialnorm (which is what eac3to is doing) should result in *higher* volume. Currently eac3to sets dialnorm to 0. Unfortunetely dialnorm set to 1 means lowering volume a lot. So incorrectly working decoders might think that a dialnorm value of 0 means even lower volume than dialnorm 1. But the documentation clearly states that a dialnorm value of 0 shall be treated as "no dialnorm processing" (which means max volume). And all the PC AC3 decoders correctly see value 0 as "dialnorm processing deactivated".


IMHO the decoders in the PS3 and Panasonic are not working correctly. Or maybe my AC3 specification is outdated? Anyway, the documentation clearly says that dialnorm 0 is "reserved". So I think it's not really good that eac3to uses it. That means I'll change it to 31 in the next build. I think that should fix the problem you're seeing. However, I believe to remember that some Sony Blu-Rays had a dialnorm value of 0, too. Well, anyway...

Please don't be hurry to change it while only one man states the issue!
I'm using PS3 and frequently removing dialog normalization with ea3to with no such issue.

alc0re
It's must certainly be your setup.
Please check up your system options in player/receiver which affecting level/compression - "dynamic range compression", "night mode".
Does your player transmit bitstream to amplifier (via HDMI? optic?) Or player decodes stream itself?
I guess the first, because when you're 1.5 FFing, player switches transmition from bitstream to LPCM.

[Atak_Snajpera]

Wake up man! Download 2.84 !!!

[jonathonsunshine]

Howdy, I just went up from 2.6 (I think) to the latest version (as of Dec 22nd, 2008) 2.84 and it doesn't seem to be correcting gaps/overlays in truehd streams anymore.

It will fix them in a standard AC3 stream but not in the TrueHD stream. It doesn't even attempt to do its "2nd pass", it just acts as if there was nothing wrong in the 1st place and that all had gone perfectly.

Does anyone know where I can get older versions of eac3to ? Is this a known problem now ? What is the latest version that doesn't have this bug ? Or am I simply stuffing something up ?

Also, Atak_Snajpera, in the post immediatly before this one, is referring to this post, cept that I deleted it and reposted because I am indeed already using 2.84 (I posted 2.83 previously)

[Easy123]

I think I got a bug with v2.84.

I have a DTS Audio track here which is about 1h 59min long and it´s 23,976fps. I want to transcode it to PAL so I use the following command: eac3to "source.dts" "taget.wavs" -speedup -2pass . 2pass because it detects clipping. At the End the Soundtrack is about 1h 50min long, but it should be about 1h 54min in Length. Can it be that during the second pass, the framerate conversion gets aplied for a second time? Meaning eac3to speeds up the 1h 54min (which is correct Runtime) for a second time?


DTS, 5.1 channels, 1:59:41, 24 bits, 768kbps, 48khz
Decoding with ArcSoft DTS Decoder...
Changing FPS from 23.976 to 25.000...
Writing WAV...
Creating file "F:\HB2-deutsch.wavs.pass1.wav"...
Clipping detected, a 2nd pass will be necessary.
Caution: The WAV file is bigger than 4GB.
Some WAV readers might not be able to handle this file correctly.
Starting 2nd pass...
Reading WAV...
Changing FPS from 23.976 to 25.000...
Reducing depth from 64 to 24 bits...
Writing WAVs...
Creating file "F:\HB2-deutsch.SL.wav"...
Creating file "F:\HB2-deutsch.L.wav"...
Creating file "F:\HB2-deutsch.C.wav"...
Creating file "F:\HB2-deutsch.R.wav"...
Creating file "F:\HB2-deutsch.LFE.wav"...
Creating file "F:\HB2-deutsch.SR.wav"...
The original audio track has a constant bit depth of 64 bits.
The processed audio track has a constant bit depth of 24 bits.
eac3to processing took 50 minutes, 22 seconds.
Done.

[73ChargerFan]

I ask because when I extract an ac3 (dolby digital) track with dialog normalization, and let eac3to remove the dialog normalization, when I play my final encoded avchd structured dvd9 in either my bluray player or my PS3 the audio is really really low and I have to crank up my receiver's volume almost all the way up to hear anything.

I have the same issue, with MPC-HC (all versions for the past year.) I set my Pioneer VSX-1015TX to a volume of about -20 for almost everything, including HD-DVDs & music from foobar & windows sounds with volume level half way up. Standard def videos and dvrms files also play fine at that volume.

For remuxed blu-ray videos , I have to boost it to at least -15, and often to -5. (dts & ac3 bitstreamed over spdif.)

My understanding is that dialnorm is signal to the receiver to lower the volume. So, removing it would make it louder. But all my bd derived videos are louder.

I don't get it. Do audio drivers / media players / audio codecs ever add a dialnorm signal where there isn't one?

[wolfbane5]

I recently started using eac3to but I have a question. Everything works great for my purposes, however eac3to thinks that surcode and mkvtoolnix aren't installed when they actually are. It somehow found that haali was installed (don't know how it found that). I'm wondering if there's a way to tell eac3to that surcode and toolnix are installed? Right now, once I have the mono .wav streams, I have to physically load them all into surcode, create the dts and then go back to eac3to to remove the padding from the dts. If eac3to knows where surcode is, then I can bypass the extra work and get an unpadded dts easily.

[alc0re]

Please don't be hurry to change it while only one man states the issue!
I'm using PS3 and frequently removing dialog normalization with ea3to with no such issue.

alc0re
It's must certainly be your setup.
Please check up your system options in player/receiver which affecting level/compression - "dynamic range compression", "night mode".
Does your player transmit bitstream to amplifier (via HDMI? optic?) Or player decodes stream itself?
I guess the first, because when you're 1.5 FFing, player switches transmition from bitstream to LPCM.

n0magic :

While I appreciate your feedback, I'm not the only one with the issue. 73ChargerFan posted after you that he is having the same issue. If madshi changes the code to make it 31 instead of 0 and that fixes the issue, it shouldn't effect those of you that are not seeing the issue, since as he stated both a level of 0 and 31 are "supposed" to mean no dialog normalization. So honestly I'm not sure why you are opposed to him changing that. Besides, he stated that perhaps using 31 is better since 0 is reserved. Perhaps some receivers process this correctly and others don't since 0 is supposed to be reserved.

Like 73ChargerFan, I am not seeing any issues on any other type of audio thrown at my receiver from any other source other than ac3 audio processed with eac3to with dialog normalization removed. My receiver's settings are fine, I've triple checked them. Both my PS3 and Bluray player are bitstreaming to my receiver via optical.

madshi I'd appreciate if I could still get an answer to my question about reversing the dialog normalization using -/+db...(not rushing...just making sure you see my previous reply/question to your reply)

[tebasuna51]

madshi I'd appreciate if I could still get an answer to my question about reversing the dialog normalization using -/+db...(not rushing...just making sure you see my previous reply/question to your reply)

No, you can't use this method.
You need wait until madshi change the DialNorm to 31 (-31 dB, instead 0) in next release.

[Thunderbolt8]

madshi, what exactly does "Detected PTS break, increasing PTS by 33.4ms..." mean? that the video stream gets speed up at that position by 33.4ms? Im asking because I had one example now in which I try to sync an external audio track, so this means for that track that I have to delay the track at this position by -32ms (1 ac3 frame)? if so, then it would be useful if eac3to could output the exact timestamp of that action, otherwise it can only be estimated by watching the progress bar all the time eac3to is processing that stream.

[williewonton]

madshi

The problem reported previously appears to be that eac3to borks the file. By using TsmuxeR in the first instance to remux to m2ts (from the playlist and dropping unwanted streams) a playable file is created. Using Eac3to to demux that file and then remux in TsmuxeR generates a playable file. So perhaps Eac3to is not handling the extraction of data from the original Blu-ray structure somehow (only my guess)

[yfed]

Hello,
I need help obtaining 24bit/192kHz 5.1 FLAC file from BD of Trondheim Soloists "Divertimenti"

Here's my cmd:

C:\!tools\audio\eac3to>eac3to "J:\_LINDBERG\BDMV\STREAM\00015.m2ts" 1:"G:\chapters.txt" 3:"G:\multi.flac"
M2TS, 1 video track, 2 audio tracks, 1:08:58
1: Chapters, 11 chapters
2: MPEG2, 1080p24 (16:9)
3: RAW/PCM, English, 5.1 channels, 24 bits, 192khz
4: RAW/PCM, English, 2.0 channels, 24 bits, 192khz
Creating file "G:\chapters.txt"...
[a03] Extracting audio track number 3...
[a03] Reading RAW/PCM...
[a03] Swapping endian...
[a03] Remapping channels...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "G:\multi.flac"...
[a03] The original audio track has a constant bit depth of 24 bits.
Video track 2 contains 99312 frames.
eac3to processing took 45 minutes, 21 seconds.
Done.

However, the FLAC file is corrupted, when I try to seek towards the middle of it in foobar2000 - it says "unable to seek.. the file is corrupted". Maybe there are some FileSystem / FLAC limitations that eac3to ignores? Please, help

PS
flac 5.1 size: 6.46Gb
pcm 5.1 size: 13.3Gb

[madshi]

Please don't be hurry to change it while only one man states the issue!
Don't worry. The change I'm planning to do should have no effect at all on proper decoders. And it might help fix bad decoders.

Howdy, I just went up from 2.6 (I think) to the latest version (as of Dec 22nd, 2008) 2.84 and it doesn't seem to be correcting gaps/overlays in truehd streams anymore.

It will fix them in a standard AC3 stream but not in the TrueHD stream. It doesn't even attempt to do its "2nd pass", it just acts as if there was nothing wrong in the 1st place and that all had gone perfectly.
eac3to was never able to fix overlaps/gaps in TrueHD streams, if you just demux them to "*.thd" or "*.thd+ac3". However, eac3to can fix the overlaps/gaps if you transcode to another format (e.g. WAV or FLAC).

If you think there's a problem with eac3to v2.84, please post your full eac3to log here (in [ code] [ / code] blocks, please, without the spaces).

I think I got a bug with v2.84.

I have a DTS Audio track here which is about 1h 59min long and it´s 23,976fps. I want to transcode it to PAL so I use the following command: eac3to "source.dts" "taget.wavs" -speedup -2pass . 2pass because it detects clipping. At the End the Soundtrack is about 1h 50min long, but it should be about 1h 54min in Length. Can it be that during the second pass, the framerate conversion gets aplied for a second time? Meaning eac3to speeds up the 1h 54min (which is correct Runtime) for a second time?
Yes, your suspicion seems to be true. I think this only occurs when using the "-2pass" option, though. Just try without it and everything should probably be fine. You never have to use the "-2pass" option, eac3to will do 2 passes in any case where it's necessary, regardless of whether you used the "-2pass" option or not. The only sense of that option is to tell eac3to that probably 2 passes are necessary. In that situation eac3to does some things differently to speed up the processing a bit. It's only a performance optimization for situations where 2 passes are necessary. But it's never necessary to use that option...

madshi, what exactly does "Detected PTS break, increasing PTS by 33.4ms..." mean? that the video stream gets speed up at that position by 33.4ms? Im asking because I had one example now in which I try to sync an external audio track, so this means for that track that I have to delay the track at this position by -32ms (1 ac3 frame)? if so, then it would be useful if eac3to could output the exact timestamp of that action, otherwise it can only be estimated by watching the progress bar all the time eac3to is processing that stream.
There's nothing you have to do. This "detected PTS break" simply means that the timestamps in the source file are not continuous, but there's a jump in the timestamps somewhere. Now whenever there's such a jump in the timestamps, eac3to of course has to handle that somehow. It does so by assuming that the first video frame after the timestamp jump is supposed to be played exactly one frame after the last video frame before the timestamp jump. That's all this log message says. So again, you don't have to do anything. However, this message indicates that either the source is corrupt or that it was cut (e.g. to remove advertising). It's quite possible that you won't notice the problem and everything is just fine.

Hmmmm... You're right, though, in that I should list the runtime at which the problem occurred. At least you can then check that runtime to see whether there are any specific problems around this runtime. So I'll add that to the log output...

I recently started using eac3to but I have a question. Everything works great for my purposes, however eac3to thinks that surcode and mkvtoolnix aren't installed when they actually are. It somehow found that haali was installed (don't know how it found that). I'm wondering if there's a way to tell eac3to that surcode and toolnix are installed?
eac3to gets the mkvtoolnix path from one of these registry values:

HKEY_CURRENT_USER\Software\mkvmergeGUI\GUI\installation_path
HKEY_LOCAL_MACHINE\Software\Microsoft\Windows\CurrentVersion\Uninstall\MKVtoolnix\UninstallString
HKEY_LOCAL_MACHINE\Software\Microsoft\Windows\CurrentVersion\Uninstall\MKVtoolnix\DisplayIcon

If you have properly installed mkvtoolnix, all of these should be set. Surcode is located by checking these registry values:

HKEY_LOCAL_MACHINE\Software\Minnetonka Audio Software\SurCode DVD-DTS\Home
HKEY_LOCAL_MACHINE\Software\Minnetonka Audio Software\SurCode DVD DTS\Home

Again, if you have properly installed Surcode, these registry values should be set.

The problem reported previously appears to be that eac3to borks the file. By using TsmuxeR in the first instance to remux to m2ts (from the playlist and dropping unwanted streams) a playable file is created. Using Eac3to to demux that file and then remux in TsmuxeR generates a playable file. So perhaps Eac3to is not handling the extraction of data from the original Blu-ray structure somehow (only my guess)
I've checked back. Please read these posts:

http://forum.doom9.org/showpost.php?p=1173568&postcount=5854
http://forum.doom9.org/showpost.php?p=1176784&postcount=5954
http://forum.doom9.org/showpost.php?p=1176805&postcount=5956

So, if you have the same problem, according to the posts above it's not caused by eac3to. Just try a different playlist. If you still think eac3to is responsible for the problem, then please manually play the separate m2ts parts and check whether audio is alright if you watch them separately. If that is the case, please check where the audio corruption begins. Is it at the join point of two m2ts files? If so, please send me the last 50MB of the first m2ts files and the first 50MB of the 2nd m2ts file. Thanks!

But again, if these posts above are correct, then eac3to is innocent and the problem is on the disc.

Hello,
I need help obtaining 24bit/192kHz 5.1 FLAC file from BD of Trondheim Soloists "Divertimenti"

Here's my cmd:

However, the FLAC file is corrupted, when I try to seek towards the middle of it in foobar2000 - it says "unable to seek.. the file is corrupted". Maybe there are some FileSystem / FLAC limitations that eac3to ignores? Please, help

PS
flac 5.1 size: 6.46Gb
pcm 5.1 size: 13.3Gb
Does the eac3to log indicate any problems? Please try playing the FLAC file with madFlac. It's quite possible that it's a bug in foobar2000 and not in eac3to. If the problem doesn't occur with madFlac then it's probably a bug in foobar2000 and you should report it to the foobar2000 support/developers. If the problem also occurs with madFlac then I'll look into it.

[n0mag!c]

If madshi changes the code to make it 31 instead of 0 and that fixes the issue, it shouldn't effect those of you that are not seeing the issue, since as he stated both a level of 0 and 31 are "supposed" to mean no dialog normalization. Besides, he stated that perhaps using 31 is better since 0 is reserved. Perhaps some receivers process this correctly and others don't since 0 is supposed to be reserved.
Ok, let's wait for changes, wish it will help you!

Like 73ChargerFan, I am not seeing any issues on any other type of audio thrown at my receiver from any other source other than ac3 audio processed with eac3to with dialog normalization removed. My receiver's settings are fine, I've triple checked them. Both my PS3 and Bluray player are bitstreaming to my receiver via optical.
By the way, I'm bitstreaming via HDMI. And sometimes I switch to LPCM output from PS3. My receiver is Denon 2308.

[mrr19121970]

@madshi

When I demux, I'm only ever interested in the Movie, English & German audios.

nice to have #1 would be:
eac3to d:\ 1) -demux *h264* *english* *german*

so to only see the tracks that have those wildcards in.




nice to have #2 would be:
eac3to d:\ 1) 2: Movie.* 3: English.ac3 4: German.ac3

currently when you do this you get the fog-horn can't convert video. I don't want to convert it, just have it extracted as MPG2, VC1 or H264 as eac3to sees it (makes my BAT easier).



Thanks for looking & Happy Christmas.

[mrr19121970]

I just tried (Atonement HD-DVD):

eac3to q:\

and it only gave me:

1) INTRO_MAIN.EVO+MAIN_LOOP.EVO, 0:02:22



no main feature etc. it worked with earlier versions for sure (eg eac3to.v2.80.exe gives me 11 playlists).

[madshi]

I just tried (Atonement HD-DVD):

eac3to q:\

and it only gave me:

1) INTRO_MAIN.EVO+MAIN_LOOP.EVO, 0:02:22



no main feature etc. it worked with earlier versions for sure (eg eac3to.v2.80.exe gives me 11 playlists).
Can you please upload the XPL files? Should be just a few KBs...

[mrr19121970]

OK, I mailled it to you, as my attachment needs approval.

[Thunderbolt8]

upload it somewhere else, by the time it gets approved christmas is most likely already over :D

[wolfbane5]

Surcode is located by checking these registry values:

HKEY_LOCAL_MACHINE\Software\Minnetonka Audio Software\SurCode DVD-DTS\Home
HKEY_LOCAL_MACHINE\Software\Minnetonka Audio Software\SurCode DVD DTS\Home

Again, if you have properly installed Surcode, these registry values should be set.

I looked under Minnetonka Audio Software and found 2 folders: SurCode DVD DTS and SurCode Dolby Digital Premiere.

DVD DTS had the following: (Default) REG_SZ (value not set)
Dolby Digital had the following: (Default) REG_SZ

In that case, I'm assuming there should be a folder labeled SurCode DVD-DTS? and that SurCode DVD-DTS and SurCode DVD DTS are both missing a Home folder?

[zeropc]

i just downloaded the free nero aac codec pack, because i need the decoder to decode aac for ac3 encoding. so what do i now to use the codecs with eac3to?

thanks for help :)

[Chumbo]

i just downloaded the free nero aac codec pack, because i need the decoder to decode aac for ac3 encoding. so what do i now to use the codecs with eac3to?

thanks for help :)
I think you can just use piping to do what you need. Tebasuna has a post from a week or 2 back, I think, that shows how to use eac3to with an external encoder so search back to his posts.

[EDIT] Here's the post: tebasuna51 knows all (http://forum.doom9.org/showthread.php?p=1223011#post1223011) ;)

[asarian]

Please don't be hurry to change it while only one man states the issue!
I'm using PS3 and frequently removing dialog normalization with ea3to with no such issue.
I was just gonna say the same thing: I always remove dialog normalizatiion and never have an issue with it on my PS3; so let's not fix it if it ain't broken. :)

[alc0re]

I was just gonna say the same thing: I always remove dialog normalizatiion and never have an issue with it on my PS3; so let's not fix it if it ain't broken. :)

Read the post.

He said the fix he's going to implement won't affect the users that aren't having an issue. Stop saying not to fix it if there are users having issues with it. Just cause the issue doesn't effect you doesn't mean it isn't effecting others. Again, the fix he's planning will not effect you or the others that aren't currently having an issue with it, but it probably will help us that are having the issue, so why bother asking him not to fix it...

[tebasuna51]

I was just gonna say the same thing: I always remove dialog normalizatiion and never have an issue with it on my PS3; so let's not fix it if it ain't broken. :)

Don't worry, with DialNorm = 31 all is ok.
I always use 31 instead 0 and work fine.

[rebkell]

Don't worry, with DialNorm = 31 all is ok.
I always use 31 instead 0 and work fine.

If I'm understanding this conversation, after the change, by setting dialnorm = 0, it would be the same as the default is now, won't it?

[zeropc]

I think you can just use piping to do what you need. Tebasuna has a post from a week or 2 back, I think, that shows how to use eac3to with an external encoder so search back to his posts.

[EDIT] Here's the post: tebasuna51 knows all (http://forum.doom9.org/showthread.php?p=1223011#post1223011) ;)

unfortunately this doesn't help me cause i wanna decode aac and not encode.

[Thunderbolt8]

eac3to lists the 7.1 DTS-HD MA track of the nordic black hawk down edition with "(strange setup)" and "CAUTION: Decoding this track with ArcSoft results in low volume" , so heres a sample in case you are interested.

http://www.sendspace.com/file/n675tw

eac3to v2.84
command line: eac3to moviepath 1) 2: X:\hawk.mkv 4: X:\hawk.flac 4: X:\hawk.dtsma
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 4 subtitle tracks, 2:23:43
1: Chapters, 12 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48khz
4: DTS Master Audio, English, 7.1 (strange setup) channels, 16 bits, 48khz
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48khz)
5: Subtitle (PGS), Swedish
6: Subtitle (PGS), Danish
7: Subtitle (PGS), Norwegian
8: Subtitle (PGS), Finnish
CAUTION: Decoding this track with ArcSoft results in low volume.
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[a04] Extracting audio track number 4...
[v02] Muxing video to Matroska...
[a04] Decoding with ArcSoft DTS Decoder...
[a04] Encoding FLAC with libFlac...
[a04] Creating file "X:\hawk.dtsma"...
[a04] Creating file "X:\hawk.flac"...
[a04] The last DTS frame is incomplete and thus gets skipped.
[a04] The original audio track has a constant bit depth of 16 bits.
[a04] The last DTS frame is incomplete and thus gets skipped.
Added fps value to MKV header.
Video track 2 contains 206735 frames.
eac3to processing took 52 minutes, 6 seconds.
Done.


so I guess arcsoft does not process this track very well, is the channel order wrong? what would be if i used sonic instead, would these 5.1 channels then be the same 5.1 channels of a regular 5.1 track or would there be some mixing up? does 5.1 have the rear or side positions and are these 5.1 positions the same in 7.1 (when tracks are decoded with eac3to & played with madflac)?

[nautilus7]

unfortunately this doesn't help me cause i wanna decode aac and not encode.

Don't you have Nero 7 installed?

Merry Christmas to all!!

[rickardk]

eac3to lists the 7.1 DTS-HD MA track of the nordic black hawk down edition with "(strange setup)" and "CAUTION: Decoding this track with ArcSoft results in low volume" , so heres a sample in case you are interested.

http://www.sendspace.com/file/n675tw

eac3to v2.84
command line: eac3to moviepath 1) 2: X:\hawk.mkv 4: X:\hawk.flac 4: X:\hawk.dtsma
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 4 subtitle tracks, 2:23:43
1: Chapters, 12 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48khz
4: DTS Master Audio, English, 7.1 (strange setup) channels, 16 bits, 48khz
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48khz)
5: Subtitle (PGS), Swedish
6: Subtitle (PGS), Danish
7: Subtitle (PGS), Norwegian
8: Subtitle (PGS), Finnish
CAUTION: Decoding this track with ArcSoft results in low volume.
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[a04] Extracting audio track number 4...
[v02] Muxing video to Matroska...
[a04] Decoding with ArcSoft DTS Decoder...
[a04] Encoding FLAC with libFlac...
[a04] Creating file "X:\hawk.dtsma"...
[a04] Creating file "X:\hawk.flac"...
[a04] The last DTS frame is incomplete and thus gets skipped.
[a04] The original audio track has a constant bit depth of 16 bits.
[a04] The last DTS frame is incomplete and thus gets skipped.
Added fps value to MKV header.
Video track 2 contains 206735 frames.
eac3to processing took 52 minutes, 6 seconds.
Done.


so I guess arcsoft does not process this track very well, is the channel order wrong? what would be if i used sonic instead, would these 5.1 channels then be the same 5.1 channels of a regular 5.1 track or would there be some mixing up? does 5.1 have the rear or side positions and are these 5.1 positions the same in 7.1 (when tracks are decoded with eac3to & played with madflac)?

Alos tried this one and got som kind of static noise in the by eac3to created FLAC. I guess the ArcSoft decoder does something wrong ...

[Chumbo]

unfortunately this doesn't help me cause i wanna decode aac and not encode.
Ah, sorry, you're right. I miss-read your post. Ugh...

[Chumbo]

...Merry Christmas to all!!
Merry Christmas to you as well! :)
*
/ \
/o \
/ o \
/_ o _\
| |

[Thunderbolt8]

Alos tried this one and got som kind of static noise in the by eac3to created FLAC.same here :S

[Thunderbolt8]

got a problem with a H264 50i cap, when I try to slow it down to 23.976fps the audiotrack length increases to >5h (duration of video seems to be fine). might be a problem, because the track is rcognized by eac3to as 2.0, while its supposed to be 5.1. Already tried to fix it with delaycut, as Tebasuna once said with that flag that indicates a channel switch, but delaycut didn't report such a thing for this track here.

http://www.sendspace.com/file/iq4hgv

eac3to v2.84
command line: eac3to movie.ts G:\movie.mkv -slowdown
------------------------------------------------------------------------------
TS, 1 video track, 1 audio track, 1:50:05
1: h264/AVC, 1080i50 (16:9)
2: AC3, 2.0 channels, 384kbps, 48khz, dialnorm: -27dB, -732ms
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[a02] Removing AC3 dialog normalization...
[a02] Decoding with DirectShow (Nero Audio Decoder 2)...
[v01] Muxing video to Matroska...
[a02] DirectShow reports 2.0 channels, 24 bits, 48khz
[a02] Applying RAW/PCM delay...
[a02] Changing FPS from 25.000 to 23.976...
[a02] Encoding AC3 <448kbps> with libAften...
[a02] Creating file "G:\movie - 2 - AC3, 2.0 channels, 384kbps, 48khz.ac3"...
[a02] The last (E-)AC3 frame is incomplete and thus gets skipped.
Added fps value to MKV header.
Video track 1 contains 330251 frames.
eac3to processing took 23 minutes, 52 seconds.
Done.

[alc0re]

If I'm understanding this conversation, after the change, by setting dialnorm = 0, it would be the same as the default is now, won't it?

Yes. But as he's stated, 31 is the same as using 0, so the new default will be the same except it will help us with decoders that don't process a value of 0 correctly.

[rebkell]

Yes. But as he's stated, 31 is the same as using 0, so the new default will be the same except it will help us with decoders that don't process a value of 0 correctly.

I understand that, I was just pointing out for the ones that are worried about the change, that if they set the value to 0, it will be identical to what they have now.

[odin24]

Deleted.

[itsancho]

strange, but minor bug in 2.84...
eac3to "H:\Hellboy (2004) Blu-ray AVC PCM"
1) 00030.mpls, 00032.m2ts, 2:12:29
- h264/AVC, 1080p24 /1.001 (16:9)
- AC3, English, multi-channel, 48khz
- RAW/PCM, English, multi-channel, 48khz
- AC3, French, multi-channel, 48khz
- AC3, German, multi-channel, 48khz
- RAW/PCM, German, multi-channel, 48khz
- AC3, English, stereo, 48khz

2) 00066.mpls, 2:23:08
[30+31+38+39+40+64].m2ts
- MPEG2, 480i60 /1.001 (16:9)
- AC3, English, stereo, 48khz eac3to "H:\Hellboy (2004) Blu-ray AVC PCM" 1)
M2TS, 1 video track, 1 audio track, 5 subtitle tracks, 2:23:08
1: Chapters, 7 chapters
2: MPEG2, 480i60 /1.001 (16:9)
3: AC3 Surround, English, 2.0 channels, 192kbps, 48khz
4: Subtitle (PGS), English
5: Subtitle (PGS), French
6: Subtitle (PGS), German
7: Subtitle (PGS), Dutch
8: Subtitle (PGS), Korean

eac3to "H:\Hellboy (2004) Blu-ray AVC PCM" 2)
M2TS, 1 video track, 6 audio tracks, 21 subtitle tracks, 2:12:29
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: AC3, English, 5.1 channels, 448kbps, 48khz
4: RAW/PCM, English, 5.1 channels, 16 bits, 48khz
5: AC3, French, 5.1 channels, 448kbps, 48khz, dialnorm: -30dB
6: AC3, German, 5.1 channels, 448kbps, 48khz, dialnorm: -30dB
7: RAW/PCM, German, 5.1 channels, 16 bits, 48khz
8: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -26dB
9: Subtitle (PGS), English
...
29: Subtitle (PGS), Korean Merry Christmas everyone! And Thank You madshi!

[tebasuna51]

got a problem with a H264 50i cap, when I try to slow it down to 23.976fps the audiotrack length increases to >5h (duration of video seems to be fine). might be a problem, because the track is rcognized by eac3to as 2.0, while its supposed to be 5.1. Already tried to fix it with delaycut, as Tebasuna once said with that flag that indicates a channel switch, but delaycut didn't report such a thing for this track here.

The ac3 extracted with eac3to from your sample with:
eac3to v2.84
command line: "D:\eac3to.exe" "D:\breakfastsplit.m2ts" zz.ac3
------------------------------------------------------------------------------
TS, 1 video track, 1 audio track, 0:00:25
1: h264/AVC, 1080i50 (16:9)
2: AC3, 2.0 channels, 384kbps, 48khz, dialnorm: -27dB, -732ms
Track 2 is used for destination file "zz.ac3".
[a02] Extracting audio track number 2...
[a02] Removing AC3 dialog normalization...
[a02] Applying (E-)AC3 delay...
[a02] Creating file "zz_e.ac3"...
[a02] The last (E-)AC3 frame is incomplete and thus gets skipped.
Video track 1 contains 1234 frames.
eac3to processing took 1 second.
Done.

have 6 frames 2.0 and after change to 5.1. Delaycut log:
[Input info]
Bitrate=384
Actual rate=384.000000
Sampling Frec=48000
TotalFrames=752
Bytesperframe=1536.0000
Filesize=1155072
FrameDuration= 32.0000
Framespersecond= 31.2500
Duration=00:00:24.064
Channels mode=2/0: L+R
LFE=LFE: Not present
[Target info]
StartFrame=0
EndFrame=751
NotFixedDelay= 0.0000
Duration=00:00:24.064
====== PROCESSING LOG ======================
Time 00:00:00.192; Frame#= 7. Some basic parameters changed between Frame #1 and this frame
Number of written frames = 752
Number of Errors= 1

[Snowknight26]

I'm still a bit confused about changing channel mapping. With the -0,1,2,3,4,5 option, how do I know which number corresponds to which channel? Or better yet, how would I switch the LFE and Back Left channels using that switch (assuming I'm dealing with 5.1 content)?

[tebasuna51]

With -0,1,2,4,3,5 the change is LFE <-> BackLeft

[dorati]

@madshi:
I have a question to the Subtitle (PGS):
Is it possible to implement a switch, to extract only forced suptitles in the SUP-File?

Sometimes the forced-suptitles are one extra stream - this is no problem!
But Sometimes the forced-subtitles in the normal suptitle stream -(.
SupRip has a switch for forced subtitle and can extract only the forced to srt.
But I will have the graphic-forced-subtiteles to remux in the m2ts-container.

Merry X-Mas

[Thunderbolt8]

The ac3 extracted with eac3to from your sample with:
eac3to v2.84
command line: "D:\eac3to.exe" "D:\breakfastsplit.m2ts" zz.ac3
------------------------------------------------------------------------------
TS, 1 video track, 1 audio track, 0:00:25
1: h264/AVC, 1080i50 (16:9)
2: AC3, 2.0 channels, 384kbps, 48khz, dialnorm: -27dB, -732ms
Track 2 is used for destination file "zz.ac3".
[a02] Extracting audio track number 2...
[a02] Removing AC3 dialog normalization...
[a02] Applying (E-)AC3 delay...
[a02] Creating file "zz_e.ac3"...
[a02] The last (E-)AC3 frame is incomplete and thus gets skipped.
Video track 1 contains 1234 frames.
eac3to processing took 1 second.
Done.

have 6 frames 2.0 and after change to 5.1. Delaycut log:
as I said the line 'Time 00:00:00.192; Frame#= 7. Some basic parameters changed between Frame #1 and this frame' didnt appear in my case, there was no problem delaycut reported at all for me with this track :S
will try again

EDIT: SORRY, I NAMED THE FILE EXTENSION OF THE SAMPLE WRONG, ITS .TS NOT .M2TS HERE

edit²: got it working now, tried to process the file which was already slowed and in that case he didnt find that line with delaycut. when I tried to process the demuxed, non-slowed ac3 track then it worked.

[ggking7]

Should I use -keepPulldown with my 120Hz TV so the de-judder mechanism can do its thing?

[odin24]

Could someone please help me out, I'm a little uncertain about the description of the -logdts output. In the case below, does the "Core+Xch" mean the HD data is applied to the entire 6.1 channels, or just the extra Cs channel?

Thanks,


+ DTS-Core
- frameSize 2012
- DTS-ES +
- channelNo 6
- lfe 1
- channelDescr 6.1
- samplingRate 48000
- bitDepth 24
- bitrate 1509000
- dialNorm 4
- extAudio XCh
- samplesPerFrame 512
- copyHistory 1
+ DTS-HD
- fullSize 2040
- headerSize 28
- refClockCode 1/48000
- frameDurationCode 1
- activeMasks [1], [[1]]
+ Asset [0]
- fullSize 2012
- headerSize 10
- corePackets Core+XCh
- extSubStrPackets XBR
- bitResolution 24
- maxSampleRate 48000
- totalNumChannels 7
- activeSpeakers C L R Ls Rs LFE Cs ($1f)

[Atak_Snajpera]

Should I use -keepPulldown with my 120Hz TV so the de-judder mechanism can do its thing?

120hz / 23.976 = ~ 5
so the answer is NO

[Snowknight26]

With -0,1,2,4,3,5 the change is LFE <-> BackLeft

So going by what you said, the following is correct?:
0 = Left
1 = Right
2 = Center
3 = LFE
4 = Back Left
5 = Back Right

Also, can you change the channel order for 7.1 channel audio with eac3to?

Edit: It seems that when I change the channel order (-0,1,2,4,3,5), the resulting file still has the same channel order when I check with ffdshow.