eac3to - audio conversion tool

[Thunderbolt8]

You got my explanation the wrong way. If you demux video, you *do* need to apply a delay in this specific case.no I got it right, but what I wanted to point out that as far as I can say now it does look quite good / best without that delay, so atm it seems to be the opposite.
but I will have a look on more episodes more closely and then compare different scenes to see if this will be the same there. maybe those will show different.

[The_Keymaker]

Fellow Forum members,

The latest version (v1.49) of EAC3toGUI can be found here:

http://www.sendspace.com/file/xilu5v

Changes and features in this version include:

- Fixed small bug that prevented text in Command Line Preview window from "wrapping".

As usual, remember to use the settings menu option to tell EAC3toGUI where the eac3to executable
is located.

Please report any problems or feature requests.

Regards,
The_Keymaker

[Thunderbolt8]

hm why are DTS-HD HiRes track are patched from 16 to 24-bit?
the band of brothers tracks are all said to be 16-bit files at the beginning of eac3to command line, but then it patches the bitdepth to 24-bits and the filesize increases :S

[shanghai2004]

I posted this before in the audio thread, but was advised I should post here.

I try to convert the audio from the Eagles Farewell 1 Tour into flac, but no luck so far.

I uses EVOdemux to get the LPCM track. Track should be 48k, 2 channel, 24 bits. I changed the extension into RAW. eac3to is testing for RAW/PCM, but gives up.

E:\cod>eac3to.exe CONCERT_PT1_2.raw CONCERT_PT1_2.flac
This might be a RAW/PCM file. Trying to figure out the details.
This will probably take a while. Please be patient...
This doesn't seem to be a RAW/PCM file.
The format of the source file could not be detected.

Anything I can try?

[shambles]

something i just thought of.. do all the hd dvd/blu-ray sound formats have the lfe channel attenuated by 10db? for ac3, eac3, truehd, dts and dts-hd the nero/sonic decoders should handle the 10db boost, right? but what about the lpcm tracks?

the most obvious way to test would be with discs that have both lossless and pcm tracks i suppose, but i don't have any such discs myself..

[XolocoTuxmaster]

Hey guys, does anyone of you know how can I use Nero 7 or Sonic Scenarist BDA to encode a PCM track to TrueHD or DTS-HD MA?

and what about ffmpeg?

[Chumbo]

I posted this before in the audio thread, but was advised I should post here.

I try to convert the audio from the Eagles Farewell 1 Tour into flac, but no luck so far.

I uses EVOdemux to get the LPCM track. Track should be 48k, 2 channel, 24 bits. I changed the extension into RAW. eac3to is testing for RAW/PCM, but gives up.

E:\cod>eac3to.exe CONCERT_PT1_2.raw CONCERT_PT1_2.flac
This might be a RAW/PCM file. Trying to figure out the details.
This will probably take a while. Please be patient...
This doesn't seem to be a RAW/PCM file.
The format of the source file could not be detected.

Anything I can try?
Did you try using the extension pcm instead of raw?

[ACrowley]

Hey guys, does anyone of you know how can I use Nero 7 or Sonic Scenarist BDA to encode a PCM track to TrueHD or DTS-HD MA?

and what about ffmpeg?

Ofcourse You cant encode DTS HD MAS or TrueHD with any of these Tools and ofcourse not with ffmpeg

-DTS HD MAS Suite can encode all DTS Audio Formats
http://www.dtsonline.com/pro-audio/mas.php
~ 2000$

-Dolby Media Producer can encode all Dolby Media Formats
http://www.dolby.com/professional/pro_audio_engineering/DMP_01.html
~ 11000$

[XolocoTuxmaster]

I thought DTS HD MA Suite was included on scenarist like it's in cinevision

So the TrueHD and DTS HD MA support on these programs is just for decoding/muxing... :(:(

[ACrowley]

I thought DTS HD MA Suite was included on scenarist like it's in cinevision

So the TrueHD and DTS HD MA support on these programs is just for decoding/muxing... :(:(


DTSHD MAS Encoder is a optional Extra for the in latest Cinevision Full Version ( all Codec Version)
http://www.sonic.com/products/Professional/CineVision/faqs.aspx
Q: What about audio encoding? Does CineVision include any audio codecs?
A: Sonic CineVision packages that include all video codecs include the DTS-HD Master Audio Suite encoder software from DTS, giving users the flexibility to encode all DTS codecs on a separate station to the CineVision video encoding application. The DTS-HD Master Audio Suite is an optional extra for some CineVision systems, please consult your Sonic representative for pricing information.

But hey..do not search for it in pirated Cinevsison 2.02 :)
Its not included!
And theres no TrueHD Encoder in Cinevision.You can not TrueHD with it..only with the Dolby Suite

What you mean is general Codec Support in Scenarist/Cinevsison. That means you can open/use DTSHD/THD/DDP Files.

When you want to by the DTSHD MAS Encoder ,better do not get Cinevision. You need min. 70.609,20 €
There are no Consumer/freeware Encoder, as i told you.

[shanghai2004]

Did you try using the extension pcm instead of raw?

Thanks for the reply!

Yes tried PCM as extension, same result.

I'm starting to wonder if the demux process has something wrong
or there is something special about 24 bit PCM on HD-DVD? :confused:

Maybe the best way to trouble shoot is to generate a known
good 24 bit 2 channel LPCM track and try to feed that into eac3to.

[XolocoTuxmaster]

DTSHD MAS Encoder is a optional Extra for the in latest Cinevision Full Version ( all Codec Version)
http://www.sonic.com/products/Professional/CineVision/faqs.aspx
Q: What about audio encoding? Does CineVision include any audio codecs?
A: Sonic CineVision packages that include all video codecs include the DTS-HD Master Audio Suite encoder software from DTS, giving users the flexibility to encode all DTS codecs on a separate station to the CineVision video encoding application. The DTS-HD Master Audio Suite is an optional extra for some CineVision systems, please consult your Sonic representative for pricing information.

But hey..do not search for it in pirated Cinevsison 2.02 :)
Its not included!
And theres no TrueHD Encoder in Cinevision.You can not TrueHD with it..only with the Dolby Suite

What you mean is general Codec Support in Scenarist/Cinevsison. That means you can open/use DTSHD/THD/DDP Files.

When you want to by the DTSHD MAS Encoder ,better do not get Cinevision. You need min. 70.609,20 €
There are no Consumer/freeware Encoder, as i told you.

Ok thanks

[Creator1]

Thanks to madshi for updating this excellent tool. Seems like WAV/RAW is more supported now since last time I checked.

I have a question that someone here might be able to answer (maybe madshi with all your experience with the sound formats?).

I cannot seem to be able to import a DD+ or TrueHD track that comes from an HD-DVD into Scenarist BDA? Scenarist says it cannot find some kind of code that should be there for it to recognize the file as a DD+ or trueHD track? I can post the detailed message tonight when I am at home.

Anybody knows if there is a difference between the DD+ and truehd formats between HD-DVD and Blu-Ray?

I have no problems importing DTS-HD or DTS-HD MA audio files from HD-DVD to scenarist BDA. I only have the problem with DD+ and TrueHD.

Of course, I am trying to find a solution without recompressing or without going to LPCM.

Thanks in advance for any help.

[nautilus7]

What does it have to do with eac3to?

[Creator1]

What does it have to do with eac3to?

This was not too helpful but very well, I made a new thread for my question.

[madshi]

Fellow Forum members,

The latest version (v1.49) of EAC3toGUI can be found here:

http://www.sendspace.com/file/xilu5v

Changes and features in this version include:

- Fixed small bug that prevented text in Command Line Preview window from "wrapping".

As usual, remember to use the settings menu option to tell EAC3toGUI where the eac3to executable
is located.
Thanks! I've uploaded it to my server.

One little suggestion: Would you consider adding WM_DROPFILES support? That would allow users to drag & drop source files onto EAC3toGUI. Also you could allow users to drag & drop "eac3to.exe" on your main form. That would tell you in which path eac3to.exe is located.

[madshi]

hm why are DTS-HD HiRes track are patched from 16 to 24-bit?
the band of brothers tracks are all said to be 16-bit files at the beginning of eac3to command line, but then it patches the bitdepth to 24-bits and the filesize increases :S
The decoder always calculates 24bit for DTS-HD Hi-Res tracks. Sonic normally dithers down to 16bit internally. If you want to have it that way you can use the undocumented switch "-dontPatchDts". That will disable to 16bit->24bit patching. Alternatively you could also ask eac3to to dither down to 16bit. I don't know whether Sonic's or my down dithering has a better quality.

Which is your target format? FLAC? How about dithering down to 18bit then?

[madshi]

I posted this before in the audio thread, but was advised I should post here.

I try to convert the audio from the Eagles Farewell 1 Tour into flac, but no luck so far.

I uses EVOdemux to get the LPCM track. Track should be 48k, 2 channel, 24 bits. I changed the extension into RAW. eac3to is testing for RAW/PCM, but gives up.

E:\cod>eac3to.exe CONCERT_PT1_2.raw CONCERT_PT1_2.flac
This might be a RAW/PCM file. Trying to figure out the details.
This will probably take a while. Please be patient...
This doesn't seem to be a RAW/PCM file.
The format of the source file could not be detected.

Anything I can try?
Yes, you can try the next eac3to version. More about that later...

[madshi]

something i just thought of.. do all the hd dvd/blu-ray sound formats have the lfe channel attenuated by 10db? for ac3, eac3, truehd, dts and dts-hd the nero/sonic decoders should handle the 10db boost, right? but what about the lpcm tracks?
The receiver should apply the 10db boost when it receives multichannel audio over either analog connection or as PCM over HDMI.

[madshi]

eac3to v2.09 released

http://madshi.net/eac3to.zip

* EVO demuxing added with proper delays for all audio tracks
* EVO file joining/rebuilding added
* automated EVO video remuxing (Matroska) added
* automated rewriting of Matroska timestamps to 24p via mkvtoolnix added
* multiple operations on the source file can now be run at the same time
* switch "-test" tests all external DirectShow filters and tools
* latest ffmpeg/libav TrueHD and E-AC3 decoder patches included
* latest libAften build included
* libav TrueHD decoder is now the default decoder for TrueHD/MLP
* support for libav DTS decoding added
* fixed a whole lot of bugs (and might have added a few new ones)
Let me stress one new feature so that it won't go unnoticed:

full EVO demuxing/remuxing support added

It's a bit difficult to explain all the new features. Let me simply give you a few examples to get you going:

Example 1:
This one looks really simple. Basically it encodes the source PCM file to both AC3 and FLAC - at the same time! This saves time because the source file only needs to be read once:
eac3to source.pcm dest.ac3 dest.flac

Example 2:
This example demuxes the first PCM, TrueHD or DTS Master Audio track that is stored in the EVO container and transcodes it to FLAC:
eac3to source.evo dest.flac

Example 3:
This example demuxes the first video track and all audio tracks of the whole HD DVD movie:
eac3to FEATURE_1.EVO+FEATURE_2.EVO -demux

Example 4:
Now it gets more complicated. First let's list the contents of the EVO source:
eac3to FEATURE_1.EVO+FEATURE_2.EVO
eac3to analyzes the source files and outputs something like this:
EVO/VOB, 2 video tracks, 5 audio tracks, 1:55:18
1: Joined EVO/VOB file
2: VC-1
3: VC-1
4: TrueHD, 5.1 channels, 48khz, dialnorm: -24dB
5: E-AC3, 5.1 channels, 384kbit/s, 48khz, dialnorm: -27dB, -17ms
6: E-AC3, 5.1 channels, 384kbit/s, 48khz, dialnorm: -27dB, -17ms
7: E-AC3, 2.0 channels, 192kbit/s, 48khz, dialnorm: -27dB, -17ms
8: E-AC3, 2.0 channels, 192kbit/s, 48khz, dialnorm: -27dB, -4ms
As you can see, all video and audio tracks are listed. Now instead of using the "-demux" or "-auto" options, which do everything automatically, you can manually decide what to do. The following command muxes the primary VC-1 video track to MKV and transcodes the TrueHD track to FLAC:
eac3to FEATURE_1.EVO+FEATURE_2.EVO 2: video.mkv 4: english.flac

Example 5:
Same EVO files as in Example 4. Now the following command will simply join the EVO files (same functionality as "rebuild" in EvoDemux).
eac3to FEATURE_1.EVO+FEATURE_2.EVO joined.evo

Example 5:
The following command line muxes the primary video track to MKV and rewrites the timestamps to 23.976. Furthermore all AC3, E-AC3, DTS and DTS-HD Hi-Res tracks are demuxed. And all PCM, TrueHD and DTS-HD Master Audio tracks are automatically converted to FLAC:
eac3to FEATURE_1.EVO+FEATURE_2.EVO movie.mkv

The whole EVO demuxing/remuxing functionality should automatically handle all video and audio delay correctly. So the final video and audio files should all be in perfect sync. There are 2 limitations, though:
(1) Delaying of bitstream formats (AC3, E-AC3, DTS) still isn't supported. So the necessary delay value is added to the file name. You can do the delaying by using delaycut.
(2) I think I got all the delay stuff right, but we'll only find out with LOTS of testing.

Generally because there was so much functionality added in this version please expect new bugs to show up. Especially the new EVO features will need a lot of testing...

[idbirch2]

Wow! Nice work madshi! This will give The_Keymaker something to do!

[bmnot]

Ok, so how do I use this Surcode software in eac3to?

I did
eac3to source.thd destination.dts -1536
TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
Removing dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
Getting "Nero Audio Decoder 2" instance failed.

I don't have Nero or Sonic installed, only Surcode 1.0.29. Seems like eac3to is trying to use Nero. How do I tell it to use Surcode?

[nautilus7]

You need nero to decode the trueHD track. That's the error message about.

Alternately you can download the new version of eac3to and give exactly the same command. It will use the free libav decoder.

[nautilus7]

Madshi, you rock man!!! :thanks:

This is 4 tools to 1: eac3to v2.08 + evodemux + h264tsto + offsetpts = eac3to v2.09
You definitely need a name change now.

automated rewriting of Matroska timestamps to 24p via mkvtoolnix addedDo you mean 24 or 23,976? I don't really understand why most HD DVDs are 24p but all HD DVD encodes around the net are 23,976. :D


EDIT1: You forgot to add support for Blu-ray (.m2ts) de/re-muxing. :p
EDIT2: I think libaften.dll is version R715, right? Latest is R723 since yesterday.

[sparknburn]

Ive used this awesome little app before but had to reinstall Windows and now I'm getting:

D:\BATTLESTAR_S1_D4\ep5>"D:\hd\eacto\eac3to.exe" "D:\BATTLESTAR_S1_D4\ep5\audio.
ddp" "D:\BATTLESTAR_S1_D4\ep5\audio.ac3" -sonic
E-AC3, 5.1 channels, 0:43:46, 384kbit/s, 48khz, dialnorm: -27dB
Removing dialog normalization...
Decoding with DirectShow (Sonic Audio Decoder)...
Getting "Sonic Audio Decoder" instance failed.

I have the Sonic HD DVD Decoder installed but it refuses to cooperate. Any ideas?

EDIT:

I'm trying the latest version with the new features.

EDIT 2:

Alrighty, love the new version! Thanks! I was able to convert the first audio track (TrueHD) to ac3 and then convert the video the an 1080 MKV file. I threw all that into TMPEnc to make a 720p HD DivX version of BSG. Yay!

[Snowknight26]

eac3to.exe PEVOB_1.EVO+PEVOB_2.EVO
The format of the source file could not be detected.

The Phantom of the Opera HD DVD, and yes, using 2.09.

[jruggle]

EDIT2: I think libaften.dll is version R715, right? Latest is R723 since yesterday.
Keep in mind that every commit to aften-svn does not affect libaften or affect encoding in any way. Cosmetic changes, changes to documentation, the pcm decoder, commandline program, build system, etc...
The only commit to affect libaften since r715 is the very recent r724.

[bmnot]

Alternately you can download the new version of eac3to and give exactly the same command. It will use the free libav decoder.

I did and it still says it's looking for Nero.

[Rectal Prolapse]

madshi, if I had a harem I would send over two of my imaginary girls right now! :)

[superx]

Madshi great job, only issue I have it with the AVC files from HD-dvd they add all those extra flags and 3:2 pulldown crap, so the sync is never right for me. I tried it on transformers.

I don't know if you know anything about AVC on HD-dvd but if there is a way we can get that fixed. for the audio sync issue, even though I know its the video that is the problem.

[moshmothma]

EDIT1: You forgot to add support for Blu-ray (.m2ts) de/re-muxing. :p


Madshi, you are the man!! Good work. Have you thought about m2ts as input (output)? What about ts as input and output? Thanks

[madshi]

Ok, so how do I use this Surcode software in eac3to?

I did
eac3to source.thd destination.dts -1536
TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
Removing dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
Getting "Nero Audio Decoder 2" instance failed.

I don't have Nero or Sonic installed, only Surcode 1.0.29. Seems like eac3to is trying to use Nero. How do I tell it to use Surcode?
You need to differ between *en*coding and *de*coding. You want to decode TrueHD and encode the result to DTS. eac3to will automatically be using Surcode for encoding. But before it can do that you first need to succeed in decoding the TrueHD track. For that purpose you can use the Nero decoder or the libav decoder. The older versions of eac3to used the Nero decoder by default, the new version is using libav by default now. You can force libav to be used by adding the "-libav" parameter.

[madshi]

Madshi, you rock man!!! :thanks:
Thanks... :)

This is 4 tools to 1: eac3to v2.08 + evodemux + h264tsto + offsetpts = eac3to v2.09
True. Although EvoDemux and h264tsto still can do some things that eac3to cannot do.

You definitely need a name change now.
Well, that's really difficult, see the past discussions about that.

Do you mean 24 or 23,976? I don't really understand why most HD DVDs are 24p but all HD DVD encodes around the net are 23,976. :D
Most people say 24p, but actually what they mean is 23.976. Well, it's confusing because I believe real cinema is really 24.000 while most HD DVDs and Blu-Rays are 23.976. Anyway, when I say 24p I mean 23.976. So eac3to is rewriting timestamps to 23.976. With 24.000 audio sync would be lost.

[madshi]

Ive used this awesome little app before but had to reinstall Windows and now I'm getting:

D:\BATTLESTAR_S1_D4\ep5>"D:\hd\eacto\eac3to.exe" "D:\BATTLESTAR_S1_D4\ep5\audio.
ddp" "D:\BATTLESTAR_S1_D4\ep5\audio.ac3" -sonic
E-AC3, 5.1 channels, 0:43:46, 384kbit/s, 48khz, dialnorm: -27dB
Removing dialog normalization...
Decoding with DirectShow (Sonic Audio Decoder)...
Getting "Sonic Audio Decoder" instance failed.

I have the Sonic HD DVD Decoder installed but it refuses to cooperate. Any ideas?
Works for me. Please try "eac3to -test". That will check whether the Sonic Audio Decoder generally works or not. If it doesn't work, the older eac3to builds should fail to work, too.

Alrighty, love the new version! Thanks! I was able to convert the first audio track (TrueHD) to ac3 and then convert the video the an 1080 MKV file. I threw all that into TMPEnc to make a 720p HD DivX version of BSG. Yay!
:)

[madshi]

The Phantom of the Opera HD DVD, and yes, using 2.09.
Hmmmm... I remember that Phantom was a problem for me back when I tried remuxing it to MKV. Maybe there's something strange about that movie? Anyway, can you upload a sample, maybe 50MB? But before you upload, please check whether you can reproduce the problem with the sample, or else it won't help. Thanks!

[madshi]

Keep in mind that every commit to aften-svn does not affect libaften or affect encoding in any way. Cosmetic changes, changes to documentation, the pcm decoder, commandline program, build system, etc...
The only commit to affect libaften since r715 is the very recent r724.
Ah, good to know. Are you planning to add further enhancements to improve audio quality? :) I wish I had better equipment here to compare the quality...

[madshi]

I did and it still says it's looking for Nero.
Please run "eac3to" without any parameters and see which version it is reporting. Maybe your browser cache still gave you the old version although you redownloaded. The latest version is v2.09. You can still use the old version, but you need to specify the "-libav" switch then, if you don't have Nero installed.

[madshi]

Madshi great job, only issue I have it with the AVC files from HD-dvd they add all those extra flags and 3:2 pulldown crap, so the sync is never right for me. I tried it on transformers.

I don't know if you know anything about AVC on HD-dvd but if there is a way we can get that fixed. for the audio sync issue, even though I know its the video that is the problem.
I've had no trouble at all with Transformers. Well, I didn't remux it with the latest eac3to version (eac3to was not up to the task at the time I remuxed Transformers), but the methods I used were the same. Try Transformers with the latest eac3to. I think it should work just fine. The key might be that the timestamps are rewritten. Also the h264 decoder might play a certain role. I'm using Cyberlink's h264 decoder with hardware acceleration.

[madshi]

Madshi, you are the man!! Good work.
Thnx!

Have you thought about m2ts as input (output)? What about ts as input and output?
Obviously now that I added EVO input support, m2ts/ts input support sounds like a reasonable next step and of course I'd like to have that feature for my own needs, too. So yes, it's probably going to happen sooner or later. Rather later, though. I want to get EVO support stable and reliable first.

m2ts/ts output support is probably not going to happen. Personally, MKV is my preferred container.

[madshi]

eac3to v2.10 released

http://madshi.net/eac3to.zip

* fixed crash which occurred when doing "EVO/VOB -> Surcode DTS encoding"
* "eac3to source.evo movie.mkv" syntax replaces "-auto" option
* "eac3to 1.evo+2.evo movie.evo" syntax supported now for simple EVO/VOB joining

[nautilus7]

True. Although EvoDemux and h264tsto still can do some things that eac3to cannot do.You mean that evodemux can read xpl and vti files, right? Beyond that is anything else that evodemux does and eac3to doesn't?

Well, that's really difficult, see the past discussions about that.I know it's difficult to find a name. Even more difficult than writing the program... :)

Most people say 24p, but actually what they mean is 23.976. Well, it's confusing because I believe real cinema is really 24.000 while most HD DVDs and Blu-Rays are 23.976. Anyway, when I say 24p I mean 23.976. So eac3to is rewriting timestamps to 23.976. With 24.000 audio sync would be lost.Your explanation confuses me even more: When i sync subtitles for an hd dvd i use 24 fps, but when it's for .mkv i change the fps to 23,976. Don't know why but it works...

[madshi]

You mean that evodemux can read xpl and vti files, right? Beyond that is anything else that evodemux does and eac3to doesn't?
Here's what EvoDemux can do what eac3to cannot do:

(1) rebuilding with only some selected video/audio/subtitle tracks
(2) handling of subtitles
(3) reading of xpl and vti information
(4) reading video track details (resolution, fps etc)

I might add reading of xpl and video track details sooner or later. Not sure about subtitles. eac3to will probably never support (1).

I know it's difficult to find a name. Even more difficult than writing the program... :)
It's one problem to find a good name. And another problem that the name "eac3to" is well known now. Changing a name of a well known tool can lead to confusion.

Your explanation confuses me even more: When i sync subtitles for an hd dvd i use 24 fps, but when it's for .mkv i change the fps to 23,976. Don't know why but it works...
Don't know why you need to sync subtitles with 24fps. Maybe the tool you're using for syncing automatically changes 24fps to 23.976 internally? I've no idea. It should be 23.976 everywhere.

[nautilus7]

Don't know why you need to sync subtitles with 24fps. Maybe the tool you're using for syncing automatically changes 24fps to 23.976 internally? I've no idea. It should be 23.976 everywhere.
I add subtitles for my HD DVDs with srt2xas. It's a normal 24 fps .srt sub that is converted in .xas format (time is not changed during srt --> xas conversion). The program also adds a few lines in the xpl and then the sub is selectable during HD DVD playback (with u key in powerdvd for example). That's all! You have to believe me. :p

[madshi]

I add subtitles for my HD DVDs with srt2xas. It's a normal 24 fps .srt sub
Where does that file originally come from? From a NTSC DVD? In that case it never was 24.000, it always was 23.976 respectively 29.97.

[nautilus7]

Usually, i take the subs from a pal dvd and i apply a 25fps--> 24fps conversion using subtitle workshop or subtitle creator.

[madshi]

Usually, i take the subs from a pal dvd and i apply a 25fps--> 24fps conversion using subtitle workshop or subtitle creator.
Hmmmm... You should be using 23.976 there. Maybe with the next movie you can try both 24.000 and 23.976 and see which fits better. I have also sometimes done the same PAL -> NTSC conversion for subtitles and I believe to remember that I always used 23.976 and had no problems with that.

[nautilus7]

I am making pan's labyrinth hd dvd subs right now. And i made subs for a .x264 (.mkv) encode yesterday. Iused 23,976 for the encode and i am going to use 24 for the hd dvd. :p I took the original subs from the R2 pal dvd and i convert them to 24 and 23,976 respectively.

I think we should stop now, because we are out of topic. :eek:

[bmnot]

Please run "eac3to" without any parameters and see which version it is reporting. Maybe your browser cache still gave you the old version although you redownloaded. The latest version is v2.09. You can still use the old version, but you need to specify the "-libav" switch then, if you don't have Nero installed.

Good call on the browser cache. Never expect that problem with Firefox. Got the new one and everything's fine now.

Just 2 more stupid questions and I'll be set.
-Is there a quality difference between between Nero and libav? I don't like the Ahead company, I'd rather use community software whenever possible.
-When running it says "Removing dialog normalization..." which is good, but it does not mention the status of DRC, if it's being removed also or not.

That's for your help!

[nautilus7]

TrueHD is a lossless format. Therefore, nero decoder and libav decoder produce bit identical results.
It's not the same where it comes to e-ac3 decoding though. Nero is better there.

DRC and Dialog Norm. are removed by both decoders.

[rack04]

"The Sonic Audio Decoder doesn't decode TrueHD properly"

Anyone know why I'm getting this error?

http://i11.photobucket.com/albums/a199/rack04/eac3to.jpg

[madshi]

-Is there a quality difference between between Nero and libav?
See nautilus7's reply.

-When running it says "Removing dialog normalization..." which is good, but it does not mention the status of DRC, if it's being removed also or not.
Both Nero's TrueHD decoder and the libav TrueHD decoder don't apply DRC, anyway, so there's nothing eac3to needs to do. It's different for E-AC3.

[madshi]

"The Sonic Audio Decoder doesn't decode TrueHD properly"

Anyone know why I'm getting this error?
You're getting this error because the Sonic Audio Decoder doesn't decode TrueHD properly... :p

No, seriously. The Sonic Audio Decoder adds noise/distortion if you decode TrueHD to full 5.1. Strange enough everything's fine if you decode to 2.0 only. Anyway, because of this problem I've decided that eac3to will not allow using the Sonic Decoder for TrueHD. Please use libav instead which should give you perfect results.

[moshmothma]

Madshi, I used eac3to 2.09 to combine my 300 evos to one file and encode the truehd audio to flac and then mux to mkv. Everything worked fine accept the flac never got muxed. The operation ended without errors and gracefully. I manually muxed the file using mkvtoolnix and everything was cool.

This is the syntax I used:

C:\downloads\eac3to>eac3to M:\HDDVD\300_HDDVD\HVDVD_TS\feature_300NDOM6LF1VC1_HD
1.EVO+M:\HDDVD\300_HDDVD\HVDVD_TS\feature_300NDOM6LF1VC1_HD1_Divide.EVO 2: 300.mkv 5: truehd.flac -nero

I used -nero cause libav chocked on the 1st go around. Please let me know if there is anything I am missing.

Also, can I mux wav with mkv instead of flac? Thanks

[rack04]

You're getting this error because the Sonic Audio Decoder doesn't decode TrueHD properly... :p

No, seriously. The Sonic Audio Decoder adds noise/distortion if you decode TrueHD to full 5.1. Strange enough everything's fine if you decode to 2.0 only. Anyway, because of this problem I've decided that eac3to will not allow using the Sonic Decoder for TrueHD. Please use libav instead which should give you perfect results.

Thanks. Do using libav I would just leave the force filter to default?

[madshi]

Everything worked fine accept the flac never got muxed.
It's not intended to. Theoretically I could add options to mux audio files to the MKV file, but the usage of eac3to is already complicated enough for my taste. Adding further options for muxing would make things extremely complicated, I fear. Because of that eac3to just muxes the video to MKV and leaves audio muxing to you. This way you can at least choose exactly which audio tracks you want to have muxed. Maybe you also want to mux additional audio tracks from DVD? Or subtitle tracks? eac3to cannot handle all that.

I used -nero cause libav chocked on the 1st go around.
Generally, if you find any situation where the libav decoder chokes, *please* try to make a little sample. Because the libav TrueHD developer can only fix bugs he gets a sample for. If you have a sample, just let me know. I'm in contact with the libav TrueHD developer and can forward any samples to him. Let's make the libav TrueHD decoder stable! All we have to do is to provide the developer with samples whenever a problem occurs with his decoder...

Also, can I mux wav with mkv instead of flac? Thanks
Never tried that yet, but I think you can. Of course file size will grow noticably.

[madshi]

Thanks. Do using libav I would just leave the force filter to default?
Yep, that's right.

[shambles]

Example 5:
The following command line muxes the primary video track to MKV and rewrites the timestamps to 23.976. Furthermore all AC3, E-AC3, DTS and DTS-HD Hi-Res tracks are demuxed. And all PCM, TrueHD and DTS-HD Master Audio tracks are automatically converted to FLAC:
eac3to FEATURE_1.EVO+FEATURE_2.EVO movie.mkv

the created flac tracks are not muxed into the mkv with this example though, right? would be nice if you could create the flac + remux the video/flac track into a matroska container with just a single command line.

edit: hmh, answered while i was typing, nevermind. although, the less times you need to (re)mux something, the less discspace needed/less wear on the harddrives/less annoyance..

other than that, it's all awesome :D m2ts support would also be amazing if it would mean you wouldn't have to concatenate before remuxing (especially amazing for the seamless branching movies)

[madshi]

although, the less times you need to (re)mux something, the less discspace needed/less wear on the harddrives/less annoyance..
Let's first check if the automatic delay calculation of eac3to works correctly before even thinking about auto muxing of audio tracks into MKV. Auto muxing wouldn't make much sense if the delay calculation doesn't work reliably... ;)

m2ts support would also be amazing if it would mean you wouldn't have to concatenate before remuxing (especially amazing for the seamless branching movies)
eac3to can handle multiple EVO source files without you having to concatenate them before. I expect the same behaviour for m2ts files.

[nautilus7]

Because of that eac3to just muxes the video to MKV and leaves audio muxing to you. This way you can at least choose exactly which audio tracks you want to have muxed. Maybe you also want to mux additional audio tracks from DVD? Or subtitle tracks? eac3to cannot handle all that.

In addition, maybe someone want to specify languages for each audio/subtitle track he muxes. So it's good that eac3to doesn't mux the files.

[moshmothma]

In addition, maybe someone want to specify languages for each audio/subtitle track he muxes. So it's good that eac3to doesn't mux the files.

Just make it an option then and not requirement. Madshi, please consider auto muxing. Thanks again

[moshmothma]

Generally, if you find any situation where the libav decoder chokes, *please* try to make a little sample. Because the libav TrueHD developer can only fix bugs he gets a sample for. If you have a sample, just let me know. I'm in contact with the libav TrueHD developer and can forward any samples to him. Let's make the libav TrueHD decoder stable! All we have to do is to provide the developer with samples whenever a problem occurs with his decoder...


Doh!! Ok, will do in the future. How do I make a sample? Just demux and use and split program? Thanks again

[nautilus7]

Demux the track and open it with a hex editor like HxD.
Cut a piece which contains the part that gives you the problem and then upload.

[nautilus7]

@ madshi

If you have time, please add the processing time display feature we discussed some days before.

Thanks.

[madshi]

Just make it an option then and not requirement. Madshi, please consider auto muxing. Thanks again
Maybe later. But first I need to know whether delay calculation is correct or not. So guys, please test remuxing to MKV with eac3to with some movies and let me know if audio seems to be in sync or not. Thanks.

[madshi]

If you have time, please add the processing time display feature we discussed some days before.
Forgot about that. I'll try to remember...

[nautilus7]

OK, thanks!

I just tested 24 bit trueHD decoding with Inside man HD DVD. I got bit identical flac files with both libav and nero decoder.

I understand that you need us to test the delay... Any particular movie? :p
I 'll just have to find one that needs a delay though, right?

[Thunderbolt8]

The decoder always calculates 24bit for DTS-HD Hi-Res tracks. Sonic normally dithers down to 16bit internally. If you want to have it that way you can use the undocumented switch "-dontPatchDts". That will disable to 16bit->24bit patching. Alternatively you could also ask eac3to to dither down to 16bit. I don't know whether Sonic's or my down dithering has a better quality.

Which is your target format? FLAC? How about dithering down to 18bit then?
hm dithering down was related to lower overall quality, right? then I better stick to the 24-bit version just to be sure.

nice additions btw in the new version! did you also plan to make eac3to .m2ts compatible in the future ? :P

[madshi]

I understand that you need us to test the delay... Any particular movie? :p
I 'll just have to find one that needs a delay though, right?
Yeah, any movies where eac3to shows some bigger delay values in the track listing. But also those with small delay values might be interesting to test.

Please remember that eac3to can not (yet) apply delay on bitstream audio (E-AC3, DTS). You'll need to use delaycut for those. But eac3to should write the needed delay into the filename of the demuxed audio track.

[madshi]

hm dithering down was related to lower overall quality, right? then I better stick to the 24-bit version just to be sure.
Yes, dithering down reduces audio quality. However, most experts seem to agree that more than 20bit doesn't have any benefit. So dithering down to 20bit should (at least in theory) not harm.

nice additions btw in the new version! did you also plan to make eac3to .m2ts compatible in the future ? :P
You're about the twenty-seventh person asking for that. Please see my earlier replies about this.

[Thunderbolt8]

ok, just read it now .m2ts later then, first making this one stable:
some questions though:

-so the offsetpts function is included as well now, for example when remuxing studio canal HD DVD we dont need to do that with the main .evo files manually any more?

-what about the fps rate, will it remain at 23.976 or change it to 24000/1001 ? might be useful to have another -option command for the other one. I guess you might merge eac3to with h264tsto sooner or later anyway, so in case of TV broadcasts 23.976 should be useful and for HD DVDs 24000/1001


btw. that -83ms for the audio track of the band of brothers remuxed proved to be right at the end. in some scenes this very fine difference could be spotted.

[madshi]

-so the offsetpts function is included as well now, for example when remuxing studio canal HD DVD we dont need to do that with the main .evo files manually any more?
Correct.

-what about the fps rate, will it remain at 23.976 or change it to 24000/1001 ? might be useful to have another -option command for the other one. I guess you might merge eac3to with h264tsto sooner or later anyway, so in case of TV broadcasts 23.976 should be useful and for HD DVDs 24000/1001
I'll change it to 24000/1001. Should work for both HD DVDs/Blu-Rays and for TV broadcasts.

btw. that -83ms for the audio track of the band of brothers remuxed proved to be right at the end. in some scenes this very fine difference could be spotted.
That's good!

[Thunderbolt8]

hm better please leave it at 23.976 for broadcasts, because sometimes the audio tracks get replaced with those coming from DVDs and those are afaik exact 23.976 values (at least I once tested it with a star wars broadcast and found some scenes were 23.976 looked better over 23.9760239 (rest scenes were more like neutral)). or at least please add a switch for that case

[shambles]

ntsc = 30000/1001, with the video 24000/1001 when ivtc'd. i don't think any source should ever be 23.976 instead of 24000/1001..

[Thunderbolt8]

are you 100% sure about that, also in cases of DVDs? is there actually a possibility to have proof for this, some technical sheets or such?

[shambles]

http://en.wikipedia.org/wiki/NTSC :)

[nautilus7]

How do i check whether a truehd track is 16, 20 or 24 bits?

I am not sure if the truehd track from inside man hd dvd is 24 bit. The flac i got from it claims to be 24, but size is only 1,88 GB for 2 a hour movie, whether the prestige's flac track (which is definately 24 bits) is 2,79 GB.

[Thunderbolt8]

cant see them talking there explicitly NTSC being 24000/1001 fps. they use 23.976 fps too often there, even though in case this might only be used as abbreviation to 24000/1001. still not enough to convince its really 24000/1001 all the time

[Snowknight26]

Madshi, here is a sample from The Phantom of the Opera:
http://www.stfcc.org/misc/PEVOB_1.EVO

[MuteyM]

Generally, if you find any situation where the libav decoder chokes, *please* try to make a little sample. Because the libav TrueHD developer can only fix bugs he gets a sample for. If you have a sample, just let me know. I'm in contact with the libav TrueHD developer and can forward any samples to him. Let's make the libav TrueHD decoder stable! All we have to do is to provide the developer with samples whenever a problem occurs with his decoder...

Hi Madshi, love your app, and I've got what I think is a libav problem for you to puzzle over... On the NIN HD DVD, every single demuxed TrueHD track gives the same error message, right at the very end of decoding:
eac3to.exe EVOB010.thd blah.wav -libav
TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
Writing WAV...
Removing dialog normalization...
Creating/writing file "D:\blah.24bit.wav"...
This audio track contains more than 16 bit of information.
-------------------------------------------------------------------------------[
mlp @ 68A442E0]End of stream indicated
[mlp @ 68A442E0]Substream 1 parity check failed
[mlp @ 68A442E0]Substream 1 checksum failed
[mlp @ 68A442E0]Substream 1 length mismatch.
Done.

Up to version 2.08, this error was ignored by eac3to and the resulting output file could be used without problems. But starting with version 2.09, the output file gets erased upon error! If it's not a big deal, I'd like to request you revert back to the old behaviour, for cases where libav chokes or just for general debugging of errors.

I'm 99% sure it's not a decrypting or demuxing problem, because I get the same results with demux.exe, EvobDemux, and eac3to 2.10's demux-to-wav functionality, and it happens with all EVOs on the disc. But the resulting output file always sounds perfect.

So I'm thinking it's a bug in libav (or maybe a defect on my disc?) Anyway, you can download a sample .thd file from here:
http://www.sendspace.com/file/udpvza

Once again, thanks for such an amazing app!

[nautilus7]

I had the same error, but with matrix truehd. It's not really a big problem, just few millisecs that are missing at the end of the track.

I had made a sample and madshi said that he was going to forward it to the libav developer.

Someone forgot to do it... :devil: (just kidding)

[moshmothma]

I had the same error, but with matrix truehd. It's not really a big problem, just few millisecs that are missing at the end of the track.

I had made a sample and madshi said that he was going to forward it to the libav developer.

Someone forgot to do it... :devil: (just kidding)

Madshi, that's the error I had with truehd from 300. Just FYI

[bmnot]

Is it possible to make 24-bit legacy DTS tracks from 24-bit DD+/TrueHD tracks?

[scarbrtj]

I recently tried my first DTS-HD --> DTS conversion. Went well insofaras got a perfect DTS stream, playable by itself. I muxed this DTS with the video AVC elementary stream using mkvmerge GUI.

If I mux just the AVC stream to .mkv, or just play it back by itself, plays perfectly and smoothly. I am using WMP 11 for playback.

BUT... if I mux the DTS and AVC together, I get "problems." I am sending the bitstream out by SPDIF. First I tried ffdshow. The file overall plays, but every few seconds there's just a slight "jump" or hiccup. It is mild, but noticeable enough to be bothersome. Perhaps this is the audio and video trying to stay in sync? Then I tried AC3 filter for my SPDIF output of the DTS. This allows the file to play smoothly, but about 10 minutes in, the audio and video begin to noticeable desync. By 1 hour in, they're really out of sync (a few seconds).

If I convert the DTS-HD into AC3 and mux that with my 1920x1080 AVC file, I get very smooth playback and no out-of-sync whatsoever.

I'd love of course to start using DTS's higher bitrates, but I can't seem to conquer this. I have tried multiple iterations.

Thoughts?

[Chumbo]

I recently tried my first DTS-HD --> DTS conversion. Went well insofaras got a perfect DTS stream, playable by itself. I muxed this DTS with the video AVC elementary stream using mkvmerge GUI.

If I mux just the AVC stream to .mkv, or just play it back by itself, plays perfectly and smoothly. I am using WMP 11 for playback.

BUT... if I mux the DTS and AVC together, I get "problems." I am sending the bitstream out by SPDIF. First I tried ffdshow. The file overall plays, but every few seconds there's just a slight "jump" or hiccup. It is mild, but noticeable enough to be bothersome. Perhaps this is the audio and video trying to stay in sync? Then I tried AC3 filter for my SPDIF output of the DTS. This allows the file to play smoothly, but about 10 minutes in, the audio and video begin to noticeable desync. By 1 hour in, they're really out of sync (a few seconds).

If I convert the DTS-HD into AC3 and mux that with my 1920x1080 AVC file, I get very smooth playback and no out-of-sync whatsoever.

I'd love of course to start using DTS's higher bitrates, but I can't seem to conquer this. I have tried multiple iterations.

Thoughts?
You must use a timecodes file on the video stream in mkvmerge with the correct fps. Read the mkvmerge help as it has a detailed explanation of this feature.

[Chumbo]

@madshi,
I just wanted to say thanks for the new version. Man you packed a lot of stuff into it. Very nice. I want to contribute any findings, so here goes.

I have a short test pcm file that converts just fine to both ac3 and dts, see output below. However, when you run it to just get the info on the file, the program crashes, i.e., "eac3to audio.pcm" crashes.

Output of eac3tov2 audio.pcm audio.ac3 -384 -16 -littleThis might be a RAW/PCM file. Trying to figure out the details.
This will probably take a while. Please be patient...
---The RAW/PCM file seems to be little endian.
---The suggested endian (little) should be correct.
---The RAW/PCM file seems to have a bitdepth of 16 bits.
---The suggested depth of 16 bits should be correct.
---The RAW/PCM file seems to have 6 channels.
---RAW/PCM, 5.1 channels, 0:04:19, 16 bits, 48khz
---Reading RAW/PCM...
Encoding AC3...
Creating/writing file "audio.ac3"...
-------------------------------------------------------------------------------
Done.

Output of eac3tov2 audio.pcm audio.dts -16 -littleThis might be a RAW/PCM file. Trying to figure out the details.
This will probably take a while. Please be patient...
---The RAW/PCM file seems to be little endian.
---The suggested endian (little) should be correct.
---The RAW/PCM file seems to have a bitdepth of 16 bits.
---The suggested depth of 16 bits should be correct.
---The RAW/PCM file seems to have 6 channels.
---RAW/PCM, 5.1 channels, 0:04:19, 16 bits, 48khz
---Reading RAW/PCM...
Writing WAVs...
Creating/writing file "audio.L.wav"...
Creating/writing file "audio.R.wav"...
Creating/writing file "audio.C.wav"...
Creating/writing file "audio.LFE.wav"...
Creating/writing file "audio.SL.wav"...
Creating/writing file "audio.SR.wav"...
-------------------------------------------------------------------------------
Found Surcode DTS Encoder version 1.0.23.0.
Surcode encoding successfully started. Please wait...
Closing Surcode...
Done.
I'll send the debug info separately. Thank you.

[scarbrtj]

You must use a timecodes file on the video stream in mkvmerge with the correct fps. Read the mkvmerge help as it has a detailed explanation of this feature.

I did, I think; I thought of this (see below). As I said, the AVC file if muxed into .mkv by itself plays back flawlessly. Also, playback is flawless if I mux AVC with AC3 (generated from DTS-HD). But... the .mkv of an AVC and DTS (actually it is DTS-ES 6.1 stream) plays back jumpy (ffdshow audio decode to SPDIF) or out-of-sync (AC3filter to SPDIF).

The timecodes file I used was:

# timecode format v1
Assume 23.976

Should this have produced a good DTS/AVC mkv merge? I did not use this to merge AVC and AC3 and the mkv plays back great (but did have to specify a 23.976 framerate in mkvmerge GUI).

[Chumbo]

I did, I think; I thought of this (see below). As I said, the AVC file if muxed into .mkv by itself plays back flawlessly. Also, playback is flawless if I mux AVC with AC3 (generated from DTS-HD). But... the .mkv of an AVC and DTS (actually it is DTS-ES 6.1 stream) plays back jumpy (ffdshow audio decode to SPDIF) or out-of-sync (AC3filter to SPDIF).

The timecodes file I used was:

# timecode format v1
Assume 23.976

Should this have produced a good DTS/AVC mkv merge? I did not use this to merge AVC and AC3 and the mkv plays back great (but did have to specify a 23.976 framerate in mkvmerge GUI).
I'm sorry, you did mention that muxing ac3 works fine and I read right past it. My bad. Your timecode stuff is correct, so I'm not sure why it's not syncing unless it's an issue with whatever splitter you're using.

I, personally, avoid WMP at all costs. ;) Try using MPC and use its built-in source filters: DTS/AC3, Matroska and MPEG PS/TS/PVA. I normally use the Haali splitter, but at times, with DTS, I have to switch to MPC's internal filters to get smooth play back. Good luck. :)

[Snowknight26]

I have a short test pcm file that converts just fine to both ac3 and dts, see output below. However, when you run it to just get the info on the file, the program crashes, i.e., "eac3to audio.pcm" crashes.
I can confirm this. Happens on several pcm streams I have.

[itsancho]

Hi Madshi, love your app, and I've got what I think is a libav problem for you to puzzle over... On the NIN HD DVD, every single demuxed TrueHD track gives the same error message, right at the very end of decoding:
eac3to.exe EVOB010.thd blah.wav -libav
TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
Writing WAV...
Removing dialog normalization...
Creating/writing file "D:\blah.24bit.wav"...
This audio track contains more than 16 bit of information.
-------------------------------------------------------------------------------[
mlp @ 68A442E0]End of stream indicated
[mlp @ 68A442E0]Substream 1 parity check failed
[mlp @ 68A442E0]Substream 1 checksum failed
[mlp @ 68A442E0]Substream 1 length mismatch.
Done.

Up to version 2.08, this error was ignored by eac3to and the resulting output file could be used without problems. But starting with version 2.09, the output file gets erased upon error! If it's not a big deal, I'd like to request you revert back to the old behaviour, for cases where libav chokes or just for general debugging of errors.

I'm 99% sure it's not a decrypting or demuxing problem, because I get the same results with demux.exe, EvobDemux, and eac3to 2.10's demux-to-wav functionality, and it happens with all EVOs on the disc. But the resulting output file always sounds perfect.

So I'm thinking it's a bug in libav (or maybe a defect on my disc?) Anyway, you can download a sample .thd file from here:
http://www.sendspace.com/file/udpvza

Once again, thanks for such an amazing app!

absolutely the same problem with "TMNT" and when i forced to use nero - everything its OK...

[shambles]

How do i check whether a truehd track is 16, 20 or 24 bits?

I am not sure if the truehd track from inside man hd dvd is 24 bit. The flac i got from it claims to be 24, but size is only 1,88 GB for 2 a hour movie, whether the prestige's flac track (which is definately 24 bits) is 2,79 GB.

the bitrate is a good indicator.. 16bit is usually less than 1500kbps, 20bit > 2000kbps, 24bit > 3000kbps

[madshi]

hm better please leave it at 23.976 for broadcasts, because sometimes the audio tracks get replaced with those coming from DVDs and those are afaik exact 23.976 values (at least I once tested it with a star wars broadcast and found some scenes were 23.976 looked better over 23.9760239 (rest scenes were more like neutral)). or at least please add a switch for that case
I don't believe you can see a difference of 7ms - and that's all there is between 23.976 and 23.9760239 at the end of a two hour movie. FWIW, I believe 24/1.001 is "more correct" than 23.976.

[madshi]

How do i check whether a truehd track is 16, 20 or 24 bits?

I am not sure if the truehd track from inside man hd dvd is 24 bit. The flac i got from it claims to be 24, but size is only 1,88 GB for 2 a hour movie, whether the prestige's flac track (which is definately 24 bits) is 2,79 GB.
We had a similar effect with Pirates of the Caribbean 1. I checked that movie and there were some very short sequences of the PCM track which were 24bit while the majority of the track was 16bit. I think the same thing is very likely to be true with the inside man track.

[madshi]

Madshi, here is a sample from The Phantom of the Opera:
http://www.stfcc.org/misc/PEVOB_1.EVO
Thanks! Will have a look.

[madshi]

On the NIN HD DVD, every single demuxed TrueHD track gives the same error message, right at the very end of decoding:
eac3to.exe EVOB010.thd blah.wav -libav
TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
Writing WAV...
Removing dialog normalization...
Creating/writing file "D:\blah.24bit.wav"...
This audio track contains more than 16 bit of information.
-------------------------------------------------------------------------------[
mlp @ 68A442E0]End of stream indicated
[mlp @ 68A442E0]Substream 1 parity check failed
[mlp @ 68A442E0]Substream 1 checksum failed
[mlp @ 68A442E0]Substream 1 length mismatch.
Done.

Up to version 2.08, this error was ignored by eac3to and the resulting output file could be used without problems. But starting with version 2.09, the output file gets erased upon error! If it's not a big deal, I'd like to request you revert back to the old behaviour, for cases where libav chokes or just for general debugging of errors.

I'm 99% sure it's not a decrypting or demuxing problem, because I get the same results with demux.exe, EvobDemux, and eac3to 2.10's demux-to-wav functionality, and it happens with all EVOs on the disc. But the resulting output file always sounds perfect.

So I'm thinking it's a bug in libav (or maybe a defect on my disc?) Anyway, you can download a sample .thd file from here:
http://www.sendspace.com/file/udpvza
Yeah, it seems to be a bug in the libav decoder. My guess is that the decoder believes that the truehd stream is done and finished and then surprisingly there's more truehd data coming in. And the decoder doesn't seem to like that. Should be easy to fix, though. Give the decoder developer a few days. His replies sometimes take a few days, but he always comes back with a fix. You can use the Nero decoder in the meanwhile ("-nero" switch).

Someone forgot to do it... :devil: (just kidding)
No, I didn't... ;)

[madshi]

Is it possible to make 24-bit legacy DTS tracks from 24-bit DD+/TrueHD tracks?
Sure. Just do "eac3to source.thd dest.dts" or "eac3to source.eac3 dest.dts". Or directly from the EVO source: "eac3to feature_1.evo+feature_2.evo 2: dest.dts".

However, you need to have the commercial (and quite expensive) Surcode DTS encoder installed!

[madshi]

I recently tried my first DTS-HD --> DTS conversion. Went well insofaras got a perfect DTS stream, playable by itself. I muxed this DTS with the video AVC elementary stream using mkvmerge GUI.

If I mux just the AVC stream to .mkv, or just play it back by itself, plays perfectly and smoothly. I am using WMP 11 for playback.

BUT... if I mux the DTS and AVC together, I get "problems." I am sending the bitstream out by SPDIF. First I tried ffdshow. The file overall plays, but every few seconds there's just a slight "jump" or hiccup. It is mild, but noticeable enough to be bothersome. Perhaps this is the audio and video trying to stay in sync? Then I tried AC3 filter for my SPDIF output of the DTS. This allows the file to play smoothly, but about 10 minutes in, the audio and video begin to noticeable desync. By 1 hour in, they're really out of sync (a few seconds).

If I convert the DTS-HD into AC3 and mux that with my 1920x1080 AVC file, I get very smooth playback and no out-of-sync whatsoever.

I'd love of course to start using DTS's higher bitrates, but I can't seem to conquer this. I have tried multiple iterations.

Thoughts?
My first guess would be that the muxer doesn't like the DTS-HD core. There's a small difference to "normal" DTS files. Normal DTS files usually have 2013 bytes per DTS frame while the DTS core from a DTS-HD track only has 2012 bytes per DTS frame. I don't know it behaves this way. And it gets even stranger: If you want to keep audio sync, the DTS parser needs to behave as if the frames were 2013 bytes long!! Extremely strange and kind of annoying. Personally, I'm not muxing the DTS files into the MKV file. Instead I'm keeping them external. The MPC HC can play them as external files. Earlier in this thread you'll find a modified source filter which plays these DTS core tracks with correct audio sync.

[nautilus7]

We had a similar effect with Pirates of the Caribbean 1. I checked that movie and there were some very short sequences of the PCM track which were 24bit while the majority of the track was 16bit. I think the same thing is very likely to be true with the inside man track.
That's possible...
I tried the -check16bit switch, but it told me thats the track contains more than 16 bit of information. But how many exactly?

How did the studio manage to make a track with variable bit depth?

[nautilus7]

We had a similar effect with Pirates of the Caribbean 1. I checked that movie and there were some very short sequences of the PCM track which were 24bit while the majority of the track was 16bit. I think the same thing is very likely to be true with the inside man track.
That's possible...
I tried the -check16bit switch, but it told me that the track contains more than 16 bit of information. But how many exactly, none knows...

How did the studio manage to make a track with variable bit depth?

[shambles]

i think it's just constant 20bit.. the track the nero decoder had problems with before was inside man, and madshi suspected it could be 18 or 20bit.. i dithered some other 24bit tracks down to 20bit and they compressed to very similar bitrates

[nautilus7]

Can flac have bit depths of 20 bits? If yes why eac3to said "writing 24 bit flac"...

[madshi]

Madshi, here is a sample from The Phantom of the Opera:
http://www.stfcc.org/misc/PEVOB_1.EVO
This EVO file seems to be majorly fucked up. The EVO structure is corrupt, so seems to be the audio data. Haven't even looked at the video data. Please rerip the movie and try again.

[madshi]

That's possible...
I tried the -check16bit switch, but it told me that the track contains more than 16 bit of information. But how many exactly, none knows...

How did the studio manage to make a track with variable bit depth?
Every TrueHD track has a bitdepth of 24bit. *EVERY ONE*. Every TrueHD decoder always outputs 24bit. However, here comes the big key: Sometimes only 16bit of those 24bit are filled with real audio data and the rest is just zeroed out. With Pirates 1 most of the track had only 16bit of the 24bit filled with data, while the remaining 8bit were always zeroes. But there were a few (very few) parts of the audio track where suddenly those 8bit had data in them.

With Inside Man, if it's a 20bit track (which I don't know), the upper 20bit are filled with real data while the lower 4bit are always filled with zeroes.

[madshi]

I have a short test pcm file that converts just fine to both ac3 and dts, see output below. However, when you run it to just get the info on the file, the program crashes, i.e., "eac3to audio.pcm" crashes.
Will be fixed in the next build. Thanks for reporting.

[nautilus7]

Every TrueHD track has a bitdepth of 24bit. *EVERY ONE*. Every TrueHD decoder always outputs 24bit. However, here comes the big key: Sometimes only 16bit of those 24bit are filled with real audio data and the rest is just zeroed out. With Pirates 1 most of the track had only 16bit of the 24bit filled with data, while the remaining 8bit were always zeroes. But there were a few (very few) parts of the audio track where suddenly those 8bit had data in them.

With Inside Man, if it's a 20bit track (which I don't know), the upper 20bit are filled with real data while the lower 4bit are always filled with zeroes.I decoded the truehd track to wavs. Then i dithered down to 22 and 20 bits. All of them were different, at least at the parts i looked in. Maybe i was unlucky and checked the wrong parts. I 'll check again if i get time.

[shambles]

you'd need to truncate, not dither, the bit depth to compare like that.. but afaik there are no apps that can truncate/convert without dithering to other than the usual 8/16/24/32 bits

[Sephiroth0000]

Madshi need help with EAC3TO please!

I have noticed that when I EVODEMUX my HD DVD it tells me with the audio how many seconds its going to be in or out by. With one certain movie the time is +83ms and when I attempt to do it via the GUI of EAC3TO (yes I have all the nesscary files) it saids its not reconised and it fails.....help!?

[nautilus7]

you'd need to truncate, not dither, the bit depth to compare like that.. but afaik there are no apps that can truncate/convert without dithering to other than the usual 8/16/24/32 bits
Sorry, wasn't aware of that.

[madshi]

you'd need to truncate, not dither, the bit depth to compare like that.. but afaik there are no apps that can truncate/convert without dithering to other than the usual 8/16/24/32 bits
That's correct. @nautilus7, "dithering" involves adding a certain amount of random noise to an audio track. So obviously after dithering you can't compare audio tracks, anymore. You cannot even compare the audio tracks, if you do the same operation twice. Because of the random noise the result is different everytime.

[madshi]

Madshi need help with EAC3TO please!

I have noticed that when I EVODEMUX my HD DVD it tells me with the audio how many seconds its going to be in or out by. With one certain movie the time is +83ms and when I attempt to do it via the GUI of EAC3TO (yes I have all the nesscary files) it saids its not reconised and it fails.....help!?
"its" not recognized? What do you mean with "its"? The source EVO file(s)? Or the audio file you demuxed with EvoDemux? Please post the eac3to output here and give me a few more details about what exactly you did.

[shanghai2004]

Just downloaded V2.10 of the program. Impressive!
Now I can directly feed it my EVO files.

But...

Still cannot extract LPCM audio from the Eagles HD-DVD.

E:\codec_mpa>eac3to c:\hddvd\hvdvd_ts\concert1.evo test.flac
EVO/VOB, 1 video track, 2 audio tracks, 1:26:36
1: VC-1
2: RAW/PCM, 2.0 channels, 16 bits, 48khz
3: DTS Master Audio, 5.1 channels, 24 bits, 48khz, dialnorm: -4dB
Track 2 is used for destination file "test.flac".
Demuxing 1st audio track...
Reading RAW/PCM...
Swapping endian...
Encoding FLAC...
Creating/writing file "test.flac"...
Done.

Seems good, but playing back test.flac file sounds like sound is played too slow (low pitch) and with static (digital noise).

EVO plays correctly in PowerDVD...

Helpfull if I supply a sample of the EVO file?

[madshi]

Helpfull if I supply a sample of the EVO file?
Yes, that'd be helpful! But only if the problem can be reproduced with the sample, too. Thanks!

[Sephiroth0000]

"its" not recognized? What do you mean with "its"? The source EVO file(s)? Or the audio file you demuxed with EvoDemux? Please post the eac3to output here and give me a few more details about what exactly you did.

DEMUXING HD DVD and using extensions ddp. for the audio. Taking DDP file and putting into EAC3TO GUI and putting output as wav. extension and then attempting to set audio delay to +83ms. Click on CONVERT and then it fails with the line *83 not reconised (cannot remember exactly as not at computer)

[madshi]

DEMUXING HD DVD and using extensions ddp. for the audio. Taking DDP file and putting into EAC3TO GUI and putting output as wav. extension and then attempting to set audio delay to +83ms. Click on CONVERT and then it fails with the line *83 not reconised (cannot remember exactly as not at computer)
I need the exact eac3to output text. Can't help without that.

[Chumbo]

I wanted to report another issue I just ran into. I'm not sure if what I attempted is correct or not, but basically I wanted to demux the video into an mkv and the specific audio to dts from TrueHD. The process successfully created the mkv and created the WAV files, but never created the resulting DTS. Below is the full output from the process. Note that there was no crash, just what's indicated below. Unfortunately, the created WAV files were deleted. It would be nice to keep those if the process fails. :) eac3tov2 FEATURE_1.EVO+FEATURE_2.EVO 2: e:\media\video\movie.mkv 4: movie.dts

EVO/VOB, 1 video track, 4 audio tracks, 2:15:09
1: Joined EVO/VOB file
2: h264/AVC
3: E-AC3, 5.1 channels, 768kbit/s, 48khz, dialnorm: -27dB, -84ms
4: TrueHD, 5.1 channels, 48khz, dialnorm: -24dB, -84ms
5: E-AC3, 5.1 channels, 448kbit/s, 48khz, dialnorm: -27dB, -84ms
6: E-AC3, 5.1 channels, 448kbit/s, 48khz, dialnorm: -27dB, -84ms
Demuxing 2nd audio track...
Removing dialog normalization...
Writing WAVs...
Creating/writing file "movie.L.wav"...
Creating/writing file "movie.R.wav"...
Creating/writing file "movie.LFE.wav"...
Creating/writing file "movie.SL.wav"...
Creating/writing file "movie.C.wav"...
Creating/writing file "movie.SR.wav"...
Muxing video to Matroska...
-------------------------------------------------------------------------------
[mlp @ 68A4D2E0]End of stream indicated
[mlp @ 68A4D2E0]Substream 1 parity check failed
[mlp @ 68A4D2E0]Substream 1 checksum failed
[mlp @ 68A4D2E0]Substream 1 length mismatch.
The libav decoder reported an error while decoding.
Waiting for DirectShow decoder thread to finish. Please wait...

[EDIT] I know this was reported earlier and I'll use the -nero for now. Sorry for the repost. This was with v2.10 btw.

[XolocoTuxmaster]

Thanks for the new update, it's good to know that ffmpeg/libav is really reliable.

What about a linux version now? XD

At least, can you tell me where I can found the patch and how to apply/use it in the linux version of ffmpeg (or libav?)?

[scarbrtj]

My first guess would be that the muxer doesn't like the DTS-HD core. There's a small difference to "normal" DTS files. Normal DTS files usually have 2013 bytes per DTS frame while the DTS core from a DTS-HD track only has 2012 bytes per DTS frame. I don't know it behaves this way. And it gets even stranger: If you want to keep audio sync, the DTS parser needs to behave as if the frames were 2013 bytes long!! Extremely strange and kind of annoying. Personally, I'm not muxing the DTS files into the MKV file. Instead I'm keeping them external. The MPC HC can play them as external files. Earlier in this thread you'll find a modified source filter which plays these DTS core tracks with correct audio sync.

As Mr. Spock would say: "Fascinating."

Unfortunately, I must mux, for I must use Windows Media Center :)

When you talk of this filter... do you mean it will play the DTS core file properly separately, or play it properly muxed? And is the filter dtsac3source.ax?

[scarbrtj]

My first guess would be that the muxer doesn't like the DTS-HD core. There's a small difference to "normal" DTS files. Normal DTS files usually have 2013 bytes per DTS frame while the DTS core from a DTS-HD track only has 2012 bytes per DTS frame. I don't know it behaves this way. And it gets even stranger: If you want to keep audio sync, the DTS parser needs to behave as if the frames were 2013 bytes long!! Extremely strange and kind of annoying. Personally, I'm not muxing the DTS files into the MKV file. Instead I'm keeping them external. The MPC HC can play them as external files. Earlier in this thread you'll find a modified source filter which plays these DTS core tracks with correct audio sync.

Oooh! Posting twice in a row. So embarassing. BUT... might one hypothesize (to avoid the "2012/2013" bug) that if one went:

DTS-HD --> mono wavs --> SurCode DTS encode --> mux AVC and DTS

versus:

DTS-HD --> DTS core --> mux AVC and DTS

that the former would be more ideal than the latter? To get a "normal" DTS track?

[madshi]

Thanks for the new update, it's good to know that ffmpeg/libav is really reliable.
Well, not fully yet, but we're working on it. It seems there's one last annoying bug, but I hope it'll be sorted out soon.

What about a linux version now? XD
No chance in hell, sorry.

[madshi]

When you talk of this filter... do you mean it will play the DTS core file properly separately, or play it properly muxed? And is the filter dtsac3source.ax?
The filter only works for separate files and yes, I mean the modified dtsac3source.ax.

might one hypothesize (to avoid the "2012/2013" bug) that if one went:

DTS-HD --> mono wavs --> SurCode DTS encode --> mux AVC and DTS

versus:

DTS-HD --> DTS core --> mux AVC and DTS

that the former would be more ideal than the latter? To get a "normal" DTS track?
Yes, transcoding the DTS-HD track to something else should help. You can use the Surcode DTS encoder or you could also use FLAC (maybe dithered down to 18 bits?).

[nautilus7]

madshi, i'm sorry. I don't have any hd dvd with a lossless track that needs a delay.
Actually, i remember adding a delay to letters from iwo jima hd dvd when i was making a flac some weeks ago, so i tried that movie.
But, unfortunately, that was due the eac3to bug that caused truehd tracks to be out of sync (no delay needed), so nothing helpful to report.

But i want to ask something...

I typed:

eac3to feature1.evo+feature2.evo 4: output1.flac 4: output2.ac3

where track 4 is a truehd audio track. The processing completed fine, but the "decoding truehd track" and "removing dialnorm" was displayed twice, i guess one for each output track (flac and ac3).
Do these actions actually happen twice?

[madshi]

The processing completed fine, but the "decoding truehd track" and "removing dialnorm" was displayed twice, i guess one for each output track (flac and ac3).
Do these actions actually happen twice?
Yes...

[nautilus7]

Well... is this necessary?


Oh, i forgot to mention it before. I used nero decoder, cause the libav produced the "famous" error again. As you said earlier, it must be the only flaw to the code.

[madshi]

Well... is this necessary?
Yes and no. Of course an intelligent human can easily see that double decoding is kind of superfluous. But trying to duplicate this with a computer program is not always that straightforward. Sure I could do it, but it would make the code a lot more complicated. So I implemented a simpler solution which sometimes results in double decoding. On the positive side, due to the simpler code there's less potential for bugs.

[nautilus7]

Ok, i understand.

Is there anything (except delay :p ) that would you like me to test?

[Thunderbolt8]

can you tell in how far codes or routines from other programs (for example evodemux or h264tsto or haali filters) were used regarding .evo joining & de- and remuxing? would be useful in case we get errors, so we might know which program(s) would produce a similar error in that case and dont have to try each other tool then.

[Chumbo]

Another one to report. I ran the -test switch to see it in action and note the results for the Nero Audio Decoder. Please let me know what other info I can provide:eac3tov2 feature_1.evo+feature_2.evo -test

Nero Audio Decoder (Nero 7 or older) is not working correctly
Sonic Audio Decoder (4.3.0.169) works fine
Haali Media Splitter (2007-11-18) is installed
Surcode DTS Encoder (1.0.23.0) is installed
MkvToolnix (v2.1.0) is installed
EVO/VOB, 2 video tracks, 4 audio tracks, 1:29:31
1: Joined EVO/VOB file
2: VC-1
3: VC-1
4: E-AC3, 5.1 channels, 384kbit/s, 48khz, dialnorm: -27dB, -8ms
5: TrueHD, 5.1 channels, 48khz, dialnorm: -24dB, -1ms
6: E-AC3, 2.0 channels, 192kbit/s, 48khz, dialnorm: -27dB, -8ms
7: E-AC3, 2.0 channels, 192kbit/s, 48khz, dialnorm: -27dB, -27ms
Using graphedt (renamed recode.exe), the Nero splitter only showed one video stream and 3 audio tracks. I was able to play the audio via the Nero Audio Decoder 2.

btw, EVOdemux confirms the two video streams and 4 audio streams, so I'm not sure why the Nero stuff is missing 1 video and the remaining audio stream.

Please let me know what else I can do to help with this one.

May I request a wish list item please. In addition to the video codec, i.e., "VC-1" above as an example, is there a way to provide a few more pieces of info? Something like "VC-1, 1920x1080, 23.976, pulldown" and pulldown would only be there if the video has the pulldown flag. If it's doable. :)

[EDIT] Well, 5 seconds after I posted this, I tried something else. I renamed eac3tov2.exe to recode.exe and reran the line above and what do you know?Nero Audio Decoder (Nero 7 or older) works fine
Sonic Audio Decoder (4.3.0.169) works fine
Haali Media Splitter (2007-11-18) is installed
Surcode DTS Encoder (1.0.23.0) is installed
MkvToolnix (v2.1.0) is installed
EVO/VOB, 2 video tracks, 4 audio tracks, 1:29:31
1: Joined EVO/VOB file
2: VC-1
3: VC-1
4: E-AC3, 5.1 channels, 384kbit/s, 48khz, dialnorm: -27dB, -8ms
5: TrueHD, 5.1 channels, 48khz, dialnorm: -24dB, -1ms
6: E-AC3, 2.0 channels, 192kbit/s, 48khz, dialnorm: -27dB, -8ms
7: E-AC3, 2.0 channels, 192kbit/s, 48khz, dialnorm: -27dB, -27ms

[shanghai2004]

Yes, that'd be helpful! But only if the problem can be reproduced with the sample, too. Thanks!

http://www.sendspace.com/file/i867cw

Its about 96MB. Plays fine in PowerDVD.
Extracting the LPCM audio track results in corrupted audio file.

Hope you have time to take a look. Thanks ahead!

[yonta]

Samples of 24bit Blu-ray LPCM which crash eac3to v2.10.

All files were demuxed with xport and they seem to be OK when converted to wav with sox/wavewizard.

24bit.2.0ch.Blu-ray.LPCM.sample1
http://www.sendspace.com/file/tnbe4v

24bit.2.0ch.Blu-ray.LPCM.sample2
http://www.sendspace.com/file/ett8j3

24bit.2.0ch.Blu-ray.LPCM.sample3
http://www.sendspace.com/file/5ncjkr

24bit.5.1ch.Blu-ray.LPCM
http://www.sendspace.com/file/64lsvl

Thank you madshi for your great effort!

[madshi]

Ok, i understand.

Is there anything (except delay :p ) that would you like me to test?
Nothing specific. Just let eac3to convert some movies for you and check whether everything works alright. That'd be helpful.

[madshi]

Well, 5 seconds after I posted this, I tried something else. I renamed eac3tov2.exe to recode.exe
Ehm, normally Nero's decoder only works if you rename the exe to "recode.exe". However, eac3to has implemented a hack around this by also allowing "eac3to.exe". Renaming "eac3to.exe" to "eac3tov2.exe" disables this hack. Please don't rename "eac3to.exe".

[madshi]

can you tell in how far codes or routines from other programs (for example evodemux or h264tsto or haali filters) were used regarding .evo joining & de- and remuxing? would be useful in case we get errors, so we might know which program(s) would produce a similar error in that case and dont have to try each other tool then.
Evo demuxing and joining is completely my own code. I got the idea from Ron's (drmpeg's) original code, but I've totally rewritten it. Timestamp fixing is based on my own OffsetPTS, but also rewritten. Evo remuxing is done on the fly by feeding the joined Evo directly into the "Haali Media Splitter (AR)" without writing the joined Evo file to harddisk first. Timestamp rewriting is done in an extra step by calling mkvtoolnix.

FWIW, I've yesterday converted three h264 HD DVD movies (Shooter, Transformers and Hunt for Red October) with the new eac3to and all three movies converted perfectly (apart from the one remaining libav TrueHD decoder bug). The final MKV files work great and also seeking works great. So I'm feeling safe to say that with the latest Haali filters remuxing h264 HD DVD movies works very well. So no need to demux the h264 video track to a raw file and mux it to MKV by dropping it into mkvtoolnix, anymore. eac3to uses the Haali filters instead which seem now up to the task, as far as I can say so far. I still have some problems with rewriting timestamps with Equilibrium, but I think that's not Haali's fault, but a bug in mkvtoolnix.

[madshi]

Samples of 24bit Blu-ray LPCM which crash eac3to v2.10.
Thank you for the samples. The crash should be fixed in the next eac3to build. However, how did you demux these samples? They look kind of corrupt to me. Did you use TsRemux or xport?

[madshi]

May I request a wish list item please. In addition to the video codec, i.e., "VC-1" above as an example, is there a way to provide a few more pieces of info? Something like "VC-1, 1920x1080, 23.976, pulldown" and pulldown would only be there if the video has the pulldown flag. If it's doable. :)
That's already planned for a future version.

[madshi]

http://www.sendspace.com/file/i867cw

Its about 96MB. Plays fine in PowerDVD.
Does PowerDVD play the PCM track or the DTS-HD track?

The PCM track seems to be corrupt. Haali's splitter doesn't even offer to demux it. Sonic's splitter crashes when trying to demux it. Nero's filter demuxes the track, but it's no valid PCM data. eac3to demuxes it, too, but it's also not valid PCM data.

Maybe EvoDemux corrupted the EVO while rebuilding? Please try eac3to on the original EVO files.

[shambles]

is eac3to supposed to apply the correct delay if you only encode the lossless track from the evos to flac (no video remuxing)? it doesn't seem apply any delay at all..

[nautilus7]

Did you use TsRemux or xport?

He mentions xport. :p

[madshi]

is eac3to supposed to apply the correct delay if you only encode the lossless track from the evos to flac (no video remuxing)? it doesn't seem apply any delay at all..
The delay should still be applied correctly. How does the eac3to output look like?

[madshi]

He mentions xport. :p
Oooops. Will recheck the samples.

[shambles]

E:\Program Files\eac3to>eac3to 101.evo f:\1.flac
EVO/VOB, 1 video track, 4 audio tracks, 1:11:52
1: VC-1
2: E-AC3, 5.1 channels, 448kbit/s, 48khz, dialnorm: -27dB, 2002ms
3: TrueHD, 5.1 channels, 48khz, dialnorm: -24dB, 2002ms
4: E-AC3, 5.1 channels, 448kbit/s, 48khz, dialnorm: -27dB, 2002ms
5: E-AC3, 5.1 channels, 448kbit/s, 48khz, dialnorm: -27dB, 2002ms
Track 3 is used for destination file "1.flac".
Demuxing 2nd audio track...
Removing dialog normalization...
Encoding FLAC...
Creating/writing file "f:\1.24bit.flac"...
This audio track contains only 16 bit of information.
The zero bytes were successfully removed.
Done.

also, the new eac3to version seems to report dialnorm -24db for all truehd tracks (both in evo and demuxed) while the old ones reported -27db (even for the same tracks)

[madshi]

3: TrueHD, 5.1 channels, 48khz, dialnorm: -24dB, 2002ms
Hmmmm... The delay really doesn't seem to be applied. So is the FLAC track out of sync by 2 seconds?

It would be very helpful, if you could upload a little sample of the EVO file. Maybe 50MB, if possible?

also, the new eac3to version seems to report dialnorm -24db for all truehd tracks (both in evo and demuxed) while the old ones reported -27db (even for the same tracks)
Ah - thanks for reporting! That's a bug, will be fixed in next build.

[shambles]

Hmmmm... The delay really doesn't seem to be applied. So is the FLAC track out of sync by 2 seconds?

It would be very helpful, if you could upload a little sample of the EVO file. Maybe 50MB, if possible?

yes, the flac is 2s oos

sample (http://rapidshare.com/files/78072796/1.evo.html)

[madshi]

yes, the flac is 2s oos

sample (http://rapidshare.com/files/78072796/1.evo.html)
Thank you. Bug confirmed and will be fixed in next build.

[Chumbo]

Ehm, normally Nero's decoder only works if you rename the exe to "recode.exe". However, eac3to has implemented a hack around this by also allowing "eac3to.exe". Renaming "eac3to.exe" to "eac3tov2.exe" disables this hack. Please don't rename "eac3to.exe".
Aha, so that's the ticket. Very interesting...:) Thank you.

[Chupacabras]

I have some movie in .ts format (video is mpeg2, audio is dts). eac3to doesn't want to work with extracted dts (stream extracted by mencoder).
It writes "The format of the source file could not be detected.":

X:\eac3to>eac3to.exe "X:\audio.dts" "X:\audio.ac3" -448
The format of the source file could not be detected.

What could be wrong?

[madshi]

I have some movie in .ts format (video is mpeg2, audio is dts). eac3to doesn't want to work with extracted dts (stream extracted by mencoder).
It writes "The format of the source file could not be detected.":

X:\eac3to>eac3to.exe "X:\audio.dts" "X:\audio.ac3" -448
The format of the source file could not be detected.

What could be wrong?
Possibly the DTS file is beginning in the middle of a DTS frame. eac3to only accepts DTS files which begin with a full DTS frame. Please try running the DTS file through delaycut. Afterwards eac3to should be able to handle the file just fine.

[madshi]

eac3to v2.11 released

http://madshi.net/eac3to.zip

* libav E-AC3 decoding is without DRC now
* libav AC3 decoding added (without DRC)
* libav E-AC3 and AC3 decoding hacked to return full 24 bit
* fixed: delay was not applied to lossless audio tracks
* fixed crash when parsing PCM files without doing any conversion
* TrueHD dialnorm was displayed incorrectly
* changed 23.976 to 24/1.001
* fixed some more minor bugs
Please do not use the "avcodec.dll" and "avutil-49.dll" that ship with eac3to with any other application. And please do not replace these dlls in eac3to, either, for the time being. The reason for this is that I have done some custom changes to the code to make the E-AC3 and AC3 decoders output 24 bit data instead of 16 bit. So the dlls shipping with eac3to are not compatible, anymore.

FWIW, the libav E-AC3 decoder is looking quite good now. There are situations where it fails to work (not all E-AC3 features are supported by the decoder yet). But as long as it works the output sounds good to me. The distortion I had with earlier versions of the decoder seem to be fully gone.

[nautilus7]

Nice. Thanks!

[Chumbo]

Thanks for the update madshi. I've run into an issue already that I wanted to report. In attempting to move the video into an mkv container, I get the following:eac3to "HVDVD_TS\feature_1.evo"+"HVDVD_TS\feature_2.evo" 2: "e:\media\movie.mkv" 3: "e:\media\movie.dts"

- EVO/VOB, 1 video track, 3 audio tracks, 2:07:40
1: Joined EVO/VOB file
2: h264/AVC
3: E-AC3, 5.1 channels, 1536kbit/s, 48khz, dialnorm: -27dB, -84ms
4: E-AC3, 5.1 channels, 768kbit/s, 48khz, dialnorm: -27dB, -84ms
5: E-AC3, 5.1 channels, 768kbit/s, 48khz, dialnorm: -27dB, -84ms
Extracting audio track number 3...
Muxing video to Matroska...
Removing dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
Disabling DRC for Nero (E-)AC3 decoding...
DirectShow reports 5.1 channels, 24 bits, 48khz
Applying delay...
Writing WAVs...
Creating/writing file "e:\media\movie.R.wav"...
Creating/writing file "e:\media\movie.L.wav"...
Creating/writing file "e:\media\movie.C.wav"...
Creating/writing file "e:\media\movie.SL.wav"...
Creating/writing file "e:\media\movie.LFE.wav"...
Creating/writing file "e:\media\movie.SR.wav"...
-The Haali Media Splitter didn't accept the input stream.
-The last (E-)AC3 frame is incomplete and thus gets skipped.
- Found Surcode DTS Encoder version 1.0.23.0.
Surcode encoding successfully started. Please wait...
Closing Surcode...
Error renaming MKV file.
MKV file was successfully created, but the timecodes were not rewritten.The dts file is created successfully, but I can't find the mkv file it says was created successfully even though there's an error. Ideas? Let me know if you want me to try anything specific.

[EDIT] I tried running it so it only goes into an mkv and used the -no24p to see the results and here's what I got:eac3to "HVDVD_TS\feature_1.evo"+"HVDVD_TS\feature_2.evo" 2: "e:\media\movie.mkv" -no24p

- EVO/VOB, 1 video track, 3 audio tracks, 2:07:40
1: Joined EVO/VOB file
2: h264/AVC
3: E-AC3, 5.1 channels, 1536kbit/s, 48khz, dialnorm: -27dB, -84ms
4: E-AC3, 5.1 channels, 768kbit/s, 48khz, dialnorm: -27dB, -84ms
5: E-AC3, 5.1 channels, 768kbit/s, 48khz, dialnorm: -27dB, -84ms
Muxing video to Matroska...
-The Haali Media Splitter didn't accept the input stream.
- eac3to processing took 1 second.
Done.

[madshi]

@Chumbo, the problem is this:

-The Haali Media Splitter didn't accept the input stream.
Because of this there most probably never was a MKV file. I'm wondering why eac3to tried to continue with the whole operation. It should have aborted with the above mentioned error message. It's probably a bug that the operation was not aborted.

Which movie is that? Did you already successfully convert some other movies to MKV with eac3to? Or was this your first try?

[Chumbo]

...
Which movie is that? Did you already successfully convert some other movies to MKV with eac3to? Or was this your first try?
This is Stardust. Yes, I've converted several others with the previous versions with no problem. I'll go back a version and try it and report back.

[idbirch2]

Regarding Phantom Of The Opera, I have also obtainted the BluRay version and that suffers from the same issues the HD-DVD did, I really don't think this is a bad rip to blame, more likely bad authoring.

The BluRay .m2ts stream suffers from the same video breakup/rainbow-ness as the HD-DVD unless it is played back with the Sonic decoders, in which case audio sync becomes a major pain in the ass if re-encoding/remuxing. Why can the Sonic decoders handle these seemingly damaged streams?

[Chumbo]

@Chumbo, the problem is this:


Because of this there most probably never was a MKV file. I'm wondering why eac3to tried to continue with the whole operation. It should have aborted with the above mentioned error message. It's probably a bug that the operation was not aborted.

Which movie is that? Did you already successfully convert some other movies to MKV with eac3to? Or was this your first try?
Okay, the one thing I left out was that I was using a small clip, i.e., using the -50MB switch to only process the first 50MB of the file(s). I've confirmed that this may be the source of this problem I'm having, because so far, it has not happened when I do a full process on the entire file or files.

The complete process for Stardust completed successfully.eac3to HVDVD_TS\feature_1.evo+HVDVD_TS\feature_2.evo 2: "e:\media\video\movie.mkv"

EVO/VOB, 1 video track, 3 audio tracks, 2:07:40
1: Joined EVO/VOB file
2: h264/AVC
3: E-AC3, 5.1 channels, 1536kbit/s, 48khz, dialnorm: -27dB, -84ms
4: E-AC3, 5.1 channels, 768kbit/s, 48khz, dialnorm: -27dB, -84ms
5: E-AC3, 5.1 channels, 768kbit/s, 48khz, dialnorm: -27dB, -84ms
Muxing video to Matroska...
Rewriting MKV timecodes by using "mkvmerge". Please wait...

mkvmerge v2.1.0 ('Another Place To Fall') built on Aug 19 2007 13:39:56
'e:\media\video\movie.old.mkv': Using the Matroska demultiplexer.
'e:\media\video\movie.old.mkv' track 1: Using the MPEG-4 part 10 (AVC) video output module.
Track 1 of 'e:\media\video\movie.old.mkv': Extracted the aspect ratio information from the MPEG-4 layer 10 (AVC) video data and set the display dimensions to 1920/1080.
The file 'e:\media\video\movie.mkv' has been opened for writing.
progress: 100%
The cue entries (the index) are being written...
Muxing took 2500 seconds.

eac3to processing took 1 hour, 46 minutes.
Timestamp rewriting took 41 minutes, 44 seconds.
Done.

[moshmothma]

eac3to v2.11 released

http://madshi.net/eac3to.zip

*
* fixed: delay was not applied to lossless audio tracks




Madshi, I remuxed the Matrix Reloaded (truehd) to mkv and flac. The audio is about -500ms off. I am using 2.11. Do I need to use a timecodes file when muxing to mkv? Thanks

[Chumbo]

Madshi, I remuxed the Matrix Reloaded (truehd) to mkv and flac. The audio is about -500ms off. I am using 2.11. Do I need to use a timecodes file when muxing to mkv? Thanks
Based on the log output of my post right before yours, I think the answer is no.

[madshi]

Okay, the one thing I left out was that I was using a small clip, i.e., using the -50MB switch to only process the first 50MB of the file(s). I've confirmed that this may be the source of this problem I'm having, because so far, it has not happened when I do a full process on the entire file or files.
So another h264 movie that got converted fine. That's good... :) Originally I was fearing to have a lot of problems with h264 movies.

[madshi]

Madshi, I remuxed the Matrix Reloaded (truehd) to mkv and flac. The audio is about -500ms off. I am using 2.11. Do I need to use a timecodes file when muxing to mkv? Thanks
Can you please run "eac3to feature_1.evo+feature_2.evo" just to get eac3to's summary of your movie files (without doing any conversion) and post the summary here? Thanks. Also please double check that you're really using v2.11. The internet brower cache sometimes gives you an old version when you believe you have downloaded the latest one...

[nautilus7]

I have matrix reloaded hd dvd. No delay is needed. Unless there are different versions of this disc...

[Thunderbolt8]

I can confirm that

PTM of first video frame = 00000DC7
PTM of last video frame = 15D0DB4F
VC-1 video stream 0 found!
First PTS = 00000DC7
Dolby TrueHD audio stream 1 found!
First PTS = 00000DC7

[madshi]

Thanks nautilus and Thunderbolt. And is the FLAC track in sync for both of you?

[Thunderbolt8]

didnt test it with the new version, it is just what I copied from ~20 pages back :P

but I just made a remux of the eyes wide shut HD DVD, one time using eac3to for remuxing & flac conversion and the other time with the 'old' way with evodemux rebuilding & separate audio (truehd) demuxing + transforming to flac. and in both cases both tracks, audio and video made in both ways had exact the same length, so I guess its fine. guess Ill stick with the eac3to way then, because it saves me quite some time.

[nautilus7]

Thanks nautilus and Thunderbolt. And is the FLAC track in sync for both of you?

I made a flac with an earlier version (2.03). Yes it was in sync.

[Pink Floyd]

Hi

Can the EAC3to tool be used to downmix DTS 5.1 tracks to normal 2 Channel stereo wav? I need to listen some DTS 5.1 Stuff on the move. I wonder if anyone has the need for stereo downmix.

From the release notes I see that the EAC3to tool uses the Cinemaster Video/ Audio Decoder for DTS Decoding. I messed around the registry keys for the Cinemaster Audio decoder and found that the following key controls the DTS Decoding Options:

HKEY_LOCAL_MACHINE\SOFTWARE\Sonic\CommonMPEGDecoders\4.2\AudioDecoder\DTSOuputOptions. The default for the DTSOuputOptions is 0xb hex. I this is changed to 0x0 hex then the decoded output for a 5.1 Channel DTS file is a 2 Channel downmixed file. I'm not certain whether this is a downmixed output of just the Front Left and Right Channels.

Note that by changing this key value the output can be controlled to 2, 3, 4, 5 and 5.1 Channels.

Regards

[madshi]

Can the EAC3to tool be used to downmix DTS 5.1 tracks to normal 2 Channel stereo wav? I need to listen some DTS 5.1 Stuff on the move. I wonder if anyone has the need for stereo downmix.

From the release notes I see that the EAC3to tool uses the Cinemaster Video/ Audio Decoder for DTS Decoding. I messed around the registry keys for the Cinemaster Audio decoder and found that the following key controls the DTS Decoding Options:

HKEY_LOCAL_MACHINE\SOFTWARE\Sonic\CommonMPEGDecoders\4.2\AudioDecoder\DTSOuputOptions. The default for the DTSOuputOptions is 0xb hex.
eac3to is already manipulating these registry values to force the Sonic decoder to output all channels and *not* to downmix. eac3to's purpose is to deliver as many channels as possible. Downmixing to 2 channels is currently not supported. I could probably add that, but I'd need to know how this is usually done technically. Does anybody know? Which channels are mixed in with which volume?

[rickardk]

Latest version works like a charm. Thanks for this great tool!
I have problems with two titles though:

Godfellas:
eac3to says that the two evo-files are of unknown format.
I can however demux video and audio with evodemux. But when I try to convert the eac3 audiotrack eac3to complains about the track being dirty.


Kingdom of heaven:
Joined the two m2ts files with copy /b. Then I used TSRemux to extract audio. Eac3to complains again about audiotrack being dirty. So I tried to demux the joined m2ts file with xport and it gives me an error halfway through. Is there any other way to join m2ts files before demuxing?

[madshi]

Godfellas:
eac3to says that the two evo-files are of unknown format.
I can however demux video and audio with evodemux. But when I try to convert the eac3 audiotrack eac3to complains about the track being dirty.
It seems that there really is something bad with those EVO files. eac3to is very strict. It doesn't allow dirty files (files with corruption or bit faults). Please try reripping.

Kingdom of heaven:
Joined the two m2ts files with copy /b. Then I used TSRemux to extract audio. Eac3to complains again about audiotrack being dirty. So I tried to demux the joined m2ts file with xport and it gives me an error halfway through. Is there any other way to join m2ts files before demuxing?
If demuxing audio is your only aim then you can use the latest xport version and demux audio separately for both m2ts files. Which audio track format do you want to demux? eac3to can join most formats for you. E.g. you can use "eac3to firstchunk.pcm+secondchunk.pcm destination.flac".

[rickardk]

It seems that there really is something bad with those EVO files. eac3to is very strict. It doesn't allow dirty files (files with corruption or bit faults). Please try reripping.


If demuxing audio is your only aim then you can use the latest xport version and demux audio separately for both m2ts files. Which audio track format do you want to demux? eac3to can join most formats for you. E.g. you can use "eac3to firstchunk.pcm+secondchunk.pcm destination.flac".

Already reripped Godfellas. Same problem.

Video stream and decoded dts ma track converted to flac into a mkv container was the goal for Kingdom of Heaven.

Have encounterd problems before when joining m2ts files (from several titles). Sometimes the lenght of the joined m2ts file is reported wrong when played back in WMP or PowerDVD. Don't know if copy /b is the right way to join m2ts files.

[Thunderbolt8]

dont use tsremux any more, just join the files with copy /b and then direclty use xport for demuxing. should usually work

[madshi]

Already reripped Godfellas. Same problem.
Can you please upload a little sample? Try to create a sample of 50MB. Then please try whether eac3to does not recognize that 50MB sample, either. Only then please upload the sample. If eac3to does handle the sample, you may have to upload a bigger sample (up to 300MB).

[rickardk]

dont use tsremux any more, just join the files with copy /b and then direclty use xport for demuxing. should usually work


Tried that but it didnt work.
Im using tsremux to make ts files from the two m2ts right now. Sometime then the lenght of joined m2ts files are all wrong I us to make ts from the m2ts files and then join the ts files. Works sometimes...

Also reripped Kingdom of Heaven and I'm sure the rip is clean.

[rickardk]

Can you please upload a little sample? Try to create a sample of 50MB. Then please try whether eac3to does not recognize that 50MB sample, either. Only then please upload the sample. If eac3to does handle the sample, you may have to upload a bigger sample (up to 300MB).

Don't really know how to make a small sample evo...
The errors starts right away so I'm sure 50MB is enough.

Btw. I will really put eac3to to the test the next couple of weeks as I'm planning to rerip my entire collection of Blu-rays and HD DVDs (app 200 discs). Video untouched, will just keep best audio track (High BR audio to FLAC) mkv as container.

[Thunderbolt8]

Tried that but it didnt work.
Im using tsremux to make ts files from the two m2ts right now. Sometime then the lenght of joined m2ts files are all wrong I us to make ts from the m2ts files and then join the ts files. Works sometimes...

Also reripped Kingdom of Heaven and I'm sure the rip is clean.
the displayed length should be wrong yes, its only accurate up to the first point where another file was joined. nevertheless, demuxing should be possible and the audio then should only be off for 5ms(LPCM) - 32ms((E)AC3) for the first part.

btw. which audio track do you mean? LPCM or eac3 ? in case of eac3 you might need to use delaycut, before eac3to will accept the input

[rickardk]

the displayed length should be wrong yes, its only accurate up to the first point where another file was joined. nevertheless, demuxing should be possible and the audio then should only be off for 5ms(LPCM) - 32ms((E)AC3) for the first part.

btw. which audio track do you mean? LPCM or eac3 ? in case of eac3 you might need to use delaycut, before eac3to will accept the input

It's a DTS-HD Master Audio track...

[Thunderbolt8]

are you using xport v1.00?

if it still shouldnt work despite that send a sample to drmpeg and see what he will say regarding that.

[rickardk]

are you using xport v1.00?

if it still shouldnt work despite that send a sample to drmpeg and see what he will say regarding that.
Yes Im using 1.00. I will try one more time to rerip and then demux the two m2ts-files one by one.
If it doesnt work I'll send him a sample (don't really know what to sample though as the problem occurs in the joint of the two m2ts-files it seems). Will make a sample of Goodfellas for madshi later today.

Thanks for all help!

[madshi]

Don't really know how to make a small sample evo...
You can use a hexeditor for that. E.g. this one:

http://www.mh-nexus.de/

Btw. I will really put eac3to to the test the next couple of weeks as I'm planning to rerip my entire collection of Blu-rays and HD DVDs (app 200 discs). Video untouched, will just keep best audio track (High BR audio to FLAC) mkv as container.
If I was you I'd do the HD DVDs first. Blu-Ray remuxing is still a bit more difficult. When you're done with all your HD DVDs, maybe the software for Blu-Ray remuxing has improved (or not).

[moshmothma]

I made a flac with an earlier version (2.03). Yes it was in sync.

Ok, my mistake. Looks like the problem is related to my setup. Twice now on I have played mkvs created through eac3to and twice thought they were out of sync at one point and played later and they seemed fine. Playing reloaded, i had to apply an audio offset to get it to be in sync initially but playing again doesn't require it. I'll try and figure out what is wrong.
Now, if I can figure out why mkvs take so infernally long to open I'll be happy.
Thanks all (esp Madshi)

[drmpeg]

Don't really know how to make a small sample evo...
I have a few simple command line tools for cutting files to create samples.

http://www.w6rz.net/head.zip

Creates a file from the first x bytes of a file.

head <infile> <outfile> <length>

head movie.evo clip.evo 50000000

************************************************

http://www.w6rz.net/tail.zip

Creates a file from the last x bytes of a file.

tail <infile> <outfile> <length>

tail movie.evo clip.evo 50000000

************************************************

http://www.w6rz.net/clip.zip

Creates a file of x bytes starting at y bytes into the file.

clip <infile> <outfile> <start offset> <length>

clip movie.evo clip.evo 1000000000 50000000

Ron

[Thunderbolt8]

regarding that kingdom of heaven sound might thing, it might be useful to get samples from the end and beginning of those parts, where the other .m2ts files are joined at with copy /b.
so when for example "copy /b a + b + c" then samples from end of a, beginning of b, maybe also end of b, beginning of c and also maybe beginning of a (just as general info).
if you really should have problems with xport regarding (the sound of) this movie, then just send him those samples, probably he can come up with a solution.


another thing, got a question to the conversion of dts-hd ma tracks to flac: the rebuilt .evo with this track only has a size of 1,92GB (16-bit; ~2h30min movie length) for 'the pianist'. the converted flac has a size of 978mb in the end, thats actually reduction of more than 50%. Ive already seen this movie and therefore know that especially later in the movie there are lots of quite scenes with much silence, but could it still be that way that the size shrinks down that much?

[tebasuna51]

Downmixing to 2 channels is currently not supported. I could probably add that, but I'd need to know how this is usually done technically. Does anybody know? Which channels are mixed in with which volume?

Here (http://forum.doom9.org/showthread.php?p=1005866#post1005866) there are some ideas. The recommended downmix is the Dolby ProLogic II

[shanghai2004]

Does PowerDVD play the PCM track or the DTS-HD track?

The PCM track seems to be corrupt. Haali's splitter doesn't even offer to demux it. Sonic's splitter crashes when trying to demux it. Nero's filter demuxes the track, but it's no valid PCM data. eac3to demuxes it, too, but it's also not valid PCM data.

Maybe EvoDemux corrupted the EVO while rebuilding? Please try eac3to on the original EVO files.

Thanks for trying!

New sample from original disk (disk has no AACS), HxD used to cut 80MB sample. Tried 2.11 on this sample with same result.

http://www.sendspace.com/file/zcz2a6

Seems LPCM on HD-DVD does something special as I'm not able to get the LPCM audio track with any program so far.

[calinb]

<snip>
Seems LPCM on HD-DVD does something special as I'm not able to get the LPCM audio track with any program so far.
Yeah--I can't get LPCM to work with eac3to either. Here's an audio-only sample that I made with EVOdemux. eac3to didn't work on the original (video+audio+subs) .evo either.

This sample plays fine with Nero 7 Showtime.

EDIT:

I took drmpeg's idea and used Linux "head" utility to trim all after the first 10MB:

http://www.sendspace.com/file/q1hjgy

Be advised that the first 9 seconds or so is silence.

[madshi]

another thing, got a question to the conversion of dts-hd ma tracks to flac: the rebuilt .evo with this track only has a size of 1,92GB (16-bit; ~2h30min movie length) for 'the pianist'. the converted flac has a size of 978mb in the end, thats actually reduction of more than 50%. Ive already seen this movie and therefore know that especially later in the movie there are lots of quite scenes with much silence, but could it still be that way that the size shrinks down that much?
It's a bit hard to say. However, DTS MA has the "problem" that there's always a 1.5Mbps core that eats bandwidth/space. And on top of that comes the lossless data. With lossless compression without such a core with a very silent / dialog oriented movie you can easily get lower than 1.5Mbps. I have to say that 1.92GB -> 978MB sounds quite extreme, but I guess it's possible. Does the FLAC file have the correct length and does it seems to play fine? If so, it's most probably correct.

[madshi]

Here (http://forum.doom9.org/showthread.php?p=1005866#post1005866) there are some ideas. The recommended downmix is the Dolby ProLogic II
Thank you!!

[Thunderbolt8]

It's a bit hard to say. However, DTS MA has the "problem" that there's always a 1.5Mbps core that eats bandwidth/space. And on top of that comes the lossless data. With lossless compression without such a core with a very silent / dialog oriented movie you can easily get lower than 1.5Mbps. I have to say that 1.92GB -> 978MB sounds quite extreme, but I guess it's possible. Does the FLAC file have the correct length and does it seems to play fine? If so, it's most probably correct.
it has the correct length and also play fine, so then I guess im lucky :P

[rickardk]

I have a few simple command line tools for cutting files to create samples.

http://www.w6rz.net/head.zip

Creates a file from the first x bytes of a file.

head <infile> <outfile> <length>

head movie.evo clip.evo 50000000

************************************************

http://www.w6rz.net/tail.zip

Creates a file from the last x bytes of a file.

tail <infile> <outfile> <length>

tail movie.evo clip.evo 50000000

************************************************

http://www.w6rz.net/clip.zip

Creates a file of x bytes starting at y bytes into the file.

clip <infile> <outfile> <start offset> <length>

clip movie.evo clip.evo 1000000000 50000000

Ron


Thanks!
Samples of Godfellas and Kingdom of Heaven are coming today, not enough time for this yestersday.


Thunderbolt8
Ripping The Pianist right now. I will report filesize of flac when done. You used Sonic CinePlayer Decoder v4.3.0 right?

[Thunderbolt8]

I used eac3to :P
i installed all these codecs back then when they were still needed for eac3to, dont know if they are also needed with the new versions. I just used the normal command line without any switch.

[nautilus7]

For dts (all kinds) sonic is the default-best quality decoder. So... if you have it installed, you used it.

You can type:

eac3to -test

to test the decoders/filters/splitters.

[rickardk]

it has the correct length and also play fine, so then I guess im lucky :P

The DTS-HD MA track from The Pianist is 1.87 GB (EU release). The FLAC created with eac3to is 978 MB (1*025*558*426 byte). Length 2h28m.

[Thunderbolt8]

almost same for me, length of mine is 1.025.559.456 Bytes :P

[calinb]

10MB LPCM audio sample:

http://www.sendspace.com/file/q1hjgy

I trimmed all but the start of the large sample with the Linux "head" utility. This sample plays in Nero Showtime but eac3to fails to produce a playable file, as also reported by shanghai2004. The first 9 seconds is silence.

[Chupacabras]

Possibly the DTS file is beginning in the middle of a DTS frame. eac3to only accepts DTS files which begin with a full DTS frame. Please try running the DTS file through delaycut. Afterwards eac3to should be able to handle the file just fine.
Thanks for advice.
delaycut processed source file, but showed many errors:
...
Time 00:00:02.110; Frame#= 199. Unsynchronized frame...SKIPPED 68 bytes. Found new synch word
Time 00:00:02.132; Frame#= 201. Unsynchronized frame...SKIPPED 68 bytes. Found new synch word
Too Many Errors. Stop Logging.
Number of written frames = 112737
Number of Errors= 112743

Final (fixed) file is half the original size (original is 571MB, fixed is 216MB). Is that normal?

Then I tryed eac3to with this fixed audio, it wrote this:
X:\eac3to>eac3to.exe "X:\audio_fixed.dts" "X:\audio_fixed.ac3" -448 -sonic
DTS, 5.1 channels, 0:20:07, 24 bits, 1536kbit/s, 48khz
Decoding with DirectShow (Sonic Audio Decoder)...
The DirectShow audio decoder didn't accept the input stream.

[nautilus7]

Your dts file is f@cked up. Try demuxing the file again, maybe using a different program.

[Chupacabras]

I tryed these:
mencoder
projectx (demuxes only video)
dgindex (throws error)
tsremux (throws error)

I don't know what else could I try...
(unfortunately I don't have original disc, file was downloaded)

[nautilus7]

I am not sure, but i think xport can demux .ts files. Try it.

[TripleH]

A little off-topic - something strange about Nero Audio Decoder 2 and Blu-ray Disc Dolby TrueHD audio tracks.

I built the following graph with graphedit:

File source (BD m2ts file) ---> Nero Splitter ---> Nero Audio Decoder 2 ---> Renderer.

The graph plays fine, but when I go to the Nero Audio Decoder 2 settings it reports that the bitrate is 640kbps (so I guess it basically gets the DD core).

Is this a mistake or is it really plays the DD core instead of TrueHD audio ?

Thanks.

Edit: I think the problem is even worse.

I took a look at two BD movies with Dolby TrueHD track using TSRemux and it recognized all of the audio streams, including the TrueHD track.

But in Nero splitter and Haali it only shows 4 audio tracks, where there are five actually, so this makes me think that the problem is the splitter.

Which splitter can recognize Dolby TrueHD tracks ?

[nautilus7]

Without really know how to help you... What do want to do?

[Thunderbolt8]

I am not sure, but i think xport can demux .ts files. Try it.
yes, it definately can. just leave out the -h option, which is for blu-ray .m2ts files

[madshi]

10MB LPCM audio sample:

http://www.sendspace.com/file/q1hjgy

I trimmed all but the start of the large sample with the Linux "head" utility. This sample plays in Nero Showtime but eac3to fails to produce a playable file, as also reported by shanghai2004. The first 9 seconds is silence.
Thanks. LPCM demuxing will be fixed in the next build. I found out that the LPCM headers differ between EVO and VOB. That's why demuxing didn't work: The current eac3to version expects VOB LPCM headers! Doh...

[madshi]

Thanks for advice.
delaycut processed source file, but showed many errors:

Final (fixed) file is half the original size (original is 571MB, fixed is 216MB). Is that normal?
Ah well, delaycut probably just removed all the DTS-HD information. Delaycut doesn't fully support DTS-HD. But still, decoding should work now.

Then I tryed eac3to with this fixed audio, it wrote this:
X:\eac3to>eac3to.exe "X:\audio_fixed.dts" "X:\audio_fixed.ac3" -448 -sonic
DTS, 5.1 channels, 0:20:07, 24 bits, 1536kbit/s, 48khz
Decoding with DirectShow (Sonic Audio Decoder)...
The DirectShow audio decoder didn't accept the input stream.
Try "eac3to -test" to check whether the Sonic decoder is working correctly on your PC.

[madshi]

unfortunately I don't have original disc, file was downloaded
In that case I can't help you further since Doom9 is not about pirating but about free rights with the content you legally own. See forum rule 6.

[madshi]

A little off-topic - something strange about Nero Audio Decoder 2 and Blu-ray Disc Dolby TrueHD audio tracks.

I built the following graph with graphedit:

File source (BD m2ts file) ---> Nero Splitter ---> Nero Audio Decoder 2 ---> Renderer.

The graph plays fine, but when I go to the Nero Audio Decoder 2 settings it reports that the bitrate is 640kbps (so I guess it basically gets the DD core).

Is this a mistake or is it really plays the DD core instead of TrueHD audio ?

Thanks.

Edit: I think the problem is even worse.

I took a look at two BD movies with Dolby TrueHD track using TSRemux and it recognized all of the audio streams, including the TrueHD track.

But in Nero splitter and Haali it only shows 4 audio tracks, where there are five actually, so this makes me think that the problem is the splitter.

Which splitter can recognize Dolby TrueHD tracks ?
I'm not sure, maybe the Sonic one. Anyway, that doesn't really belong into this thread. BTW, the Nero Audio Decoder doesn't like Blu-Ray TrueHD streams (because of the interweaved AC3 frames). So even if you found a splitter which exports TrueHD you can still not easily decode it with Nero. Except by using eac3to, of course.

[Beastie Boy]

Wow, this tool has evolved! Many thanks Madshi for your work on this. It is very much appreciated.

Since the latest update, is it still worth buying the Nero plugin? I have bought a copy of Nero 7 with the intention of adding the plugin, but it seems as though I needn't bother.
What is your recommendation?

Cheers, Beastie.

[nautilus7]

You "need" nero/plugin if you care about best quality in e-ac3/ac3 decoding, because nero decoder is a reference one and gives max quality.

If not, the free ffmpeg decoder is almost mature and can be used instead. The choice is up to you, i think.

[Beastie Boy]

Thanks for the reply. Nero it is then as quality is the most important thing for me.

Cheers, Beastie.

[TripleH]

I'm not sure, maybe the Sonic one. Anyway, that doesn't really belong into this thread. BTW, the Nero Audio Decoder doesn't like Blu-Ray TrueHD streams (because of the interweaved AC3 frames). So even if you found a splitter which exports TrueHD you can still not easily decode it with Nero. Except by using eac3to, of course.

OK. I asked that question because I want to avoid from re-encoding the audio.

Is there any quality loss when going from HD Audio (Nero decoder for DDP/TrueHD and Sonic for DTS-HA MA) to FLAC ?

Thanks.

[nautilus7]

OK. I asked that question because I want to avoid from re-encoding the audio.

Is there any quality loss when going from HD Audio (Nero decoder for DDP/TrueHD and Sonic for DTS-HA MA) to FLAC ?

Thanks.You are a bit confused... :confused:
eac3to can do exactly what you need.

Dolby TrueHD and DTS-HD Master Audio are lossless formats. Flac is lossless too. So any conversion between these 3 formats is without quality loss.
Dolby E-AC3, AC3, DTS and DTS-HD High Resolution are lossy formats. Decode any of them and encode to a different format will result in quality loss.

But what eac3to does, is to minimize the quality loss by using proper decoders (freeware and not freeware) and various "modifications" so they output the best audio quality.

It seems that you 've never read the 1st post of this thread. Do it and i am sure your questions will disappear.

[n_response]

hello !

I have a "DTS-HD MA 7.1" audio track demux from blue ray dvd.

I want to convent to ac3

I already install nero 7.

but when I use these comaned line

It shows that:



E:\AUDIO>eac3to 01.dtshd ac3.ac3 -640
DTS Hi-Res, 5.1 channels, 4:49:45, 24 bits, 1676kbit/s, 48khz
Decoding DTS-HD track to raw. Please wait...
Find sync word: 7ffe8001
Find sync extension: 3f
The file size of the raw file doesn't seem to fit.
The expected file size for 16 bit is 9.32 GB.
The expected file size for 24 bit is 13.98 GB.
The real file size is 4.38 GB.




And it was fiald!~~

Waiting for the question! And Marry Cristmas!~~

[nautilus7]

I asked you in the other thread about sonic..
You didn't answer if you have sonic decoders installed and if they are working fine.

Read the 1st post and stop telling that you have nero installed. Nero is not needed in your case.

And you 're waiting for the answer. The question is been made by you.

[madshi]

The file size of the raw file doesn't seem to fit.
Pleae update your eac3to version. That error message you've posted is from eac3to v1.x. We're already at v2.x now.

[madshi]

eac3to v2.12 released

http://madshi.net/eac3to.zip

thanks to Ron/drmpeg for all his help

* video resolution, framerate and mode (progressive/interlaced) are displayed
* rewriting timestamps should now always write the correct framerate
* after a full EVO/VOB processing the number of video frames is shown
* EVO 16 bit and 24 bit LPCM demuxing supported now (need samples for 20 bit)
* (E-)AC3 bitstream can be delayed now (similar to delaycut)
* DTS bitstream can be delayed now (similar to delaycut)
* DTS-HD High-Res and Master Audio bitstream can be delayed now
* when demuxing bitstream audio tracks from EVO delay is automatically applied
* some little bugs fixed

[nautilus7]

Pleae update your eac3to version. That error message you've posted is from eac3to v1.x. We're already at v2.x now.Oh my God!!! :angry:

I couldn't imagine that!

[nautilus7]

eac3to v2.12 released

http://madshi.net/eac3to.zip

thanks to Ron/drmpeg for all his help

* video resolution, framerate and mode (progressive/interlaced) are displayed
* rewriting timestamps should now always write the correct framerate
* after a full EVO/VOB processing the number of video frames is shown
* EVO 16 bit and 24 bit LPCM demuxing supported now (need samples for 20 bit)
* (E-)AC3 bitstream can be delayed now (similar to delaycut)
* DTS bitstream can be delayed now (similar to delaycut)
* DTS-HD High-Res and Master Audio bitstream can be delayed now
* when demuxing bitstream audio tracks from EVO delay is automatically applied
* some little bugs fixed

Merry Cristmas and a happy new year to you too Madshi. :D

[Thunderbolt8]

looks great, thanks you two :thanks:

does that mean that eac3to already delays those eac, dts etc. files now automatically (if needed), as with the all the others e.g. truehd, or do we have to do that manually, when taking it from a HD DVD source for example? (am asking because you wrote 'can do now' and not 'does automatically' or something like that)

edit: I dont know exactly what bitstream audios are, but I guess that are those eac3, dts track I was talking about, correct? so according to your update it can do/does both now, automatically if needed and manually, when we feed it with such a track?

[nautilus7]

I dont know exactly what bitstream audios are, but I guess that are those eac3, dts track I was talking about, correct? so according to your update it can do/does both now, automatically if needed and manually, when we feed it with such a track?
You 're right!

[rickardk]

Been remuxing HD DVDs all dayt today with not a single error using eac3to. Thanks again for this tool. You have done a fantastic job!

[madshi]

Merry Cristmas and a happy new year to you too Madshi. :D
Thanks. Same to you and all other Doom9 hotties! :D

[madshi]

does that mean that eac3to already delays those eac, dts etc. files now automatically (if needed), as with the all the others e.g. truehd, or do we have to do that manually, when taking it from a HD DVD source for example?
Basically eac3to can now apply any delay on any audio track. The only exception is if the source is TrueHD and if you want to keep the audio as TrueHD. That's the only situation where eac3to cannot apply a delay. But as soon as you recode TrueHD to anything else, again eac3to can apply a delay.

am asking because you wrote 'can do now' and not 'does automatically' or something like that
If you feed eac3to EVO files you don't need to care about audio delays, anymore. (Except if you want to keep the TrueHD tracks as external TrueHD files).

Of course that's what I *hope*. It does need to be tested...

[madshi]

Been remuxing HD DVDs all dayt today with not a single error using eac3to. Thanks again for this tool. You have done a fantastic job!
Thank you. I'm glad it works fine for you. Once you're done with all your HD DVDs, could you please post a movie list here? Also have you had a chance to check whether the audio sync is always correct?

[Thunderbolt8]

ive tested it with the pianist and bourne identity so far (only main track though). and both, dts-hd ma (pianist) and eac3 (bourne), were fine (though in both cases no delay had to be applied anyway)

[Snowknight26]

If you feed eac3to EVO files you don't need to care about audio delays, anymore. (Except if you want to keep the TrueHD tracks as external TrueHD files).

Of course that's what I *hope*. It does need to be tested...

So lets say that audio track 1 (DTS) has a delay or 1001ms. If you demux that audio track, and want to mux it with the video track into an mkv container, you don't need to add the 1001ms delay because eac3to automatically applied it when it demuxed it?

[n_response]

thanks to madshi & nautilus7

I use the last version. It's Done.

Have a nice day~~~~

[Biggiesized]

Hello, I'm having trouble converting a .dtshd 5.1 and 7.1 48/24 track that I have. I have the Sonic Audio Decoder (4.3.0.169) installed but I always get an error when I run the command line.

"The format of the source file could not be detected."

Any idea why this is happening? I can play the file fine in Windows Media Player. I can upload the track (it's short--about 2:00) if anyone wants to play with it.

By the way, I just downloaded the latest version of this program today. I'm using the GUI, but I could just as easily use the command line version (since I have that too).

[nautilus7]

So lets say that audio track 1 (DTS) has a delay or 1001ms. If you demux that audio track, and want to mux it with the video track into an mkv container, you don't need to add the 1001ms delay because eac3to automatically applied it when it demuxed it?

Yes, that's right. Unless there's a bug in the new version. :p
But you have a very big delay value, so you'll be able to notice if something wrong.

[nautilus7]

Hello, I'm having trouble converting a .dtshd 5.1 and 7.1 48/24 track that I have. I have the Sonic Audio Decoder (4.3.0.169) installed but I always get an error when I run the command line.

"The format of the source file could not be detected."

Any idea why this is happening? I can play the file fine in Windows Media Player. I can upload the track (it's short--about 2:00) if anyone wants to play with it.

By the way, I just downloaded the latest version of this program today. I'm using the GUI, but I could just as easily use the command line version (since I have that too).Are you sure it's a dts-hd file? What are you trying to do? I think it's better to test again using the cli.
Anyway, you can upload the track if it's small.

[Biggiesized]

Yes, I'm positive it's a DTS-HD file--otherwise it wouldn't have had a .dtshd extension and play correctly.

I'm uploading the 5.1 mix since it's only 20 MB. I'll upload the 7.1 longer mix later if necessary(it's about 90 MB).

http://www.sendspace.com/file/f7bw5d

EDIT: I'm trying to transcode it to .flac or .wav or even .ac3 if I can get it to work.

[nautilus7]

Your file is a dts-hd track that has only the dts core with no extensions in. This means it's a normal dts file.
Except from the "garbage" it has in the beginning and the end. That's why eac3to can't recognize it.

Change the extension to .dts and use latest delaycut (v1.3.0.0) to fix the track (remove the garbage). Then it 'll work. I guess the same applies to the 7.1 track you have.

[rickardk]

Thank you. I'm glad it works fine for you. Once you're done with all your HD DVDs, could you please post a movie list here? Also have you had a chance to check whether the audio sync is always correct?

Yes I will post a list...done with 25 titles so far (using two computers) without a single problem (Goodfellas is the exeption, but I guess the disc is corrupt. Will try a fresh disc on thursday).

I'm doing a fast check on audio sync (5 minutes from start and 15 minutes from end). The TrueHD track on Training Day was out of sync. BUT it's out of sync when played back in PowerDVD also. So I guess it's a bad authoring.

I hope to go through the whole collection before the end of the year.

Then I will move on to the Blu-ray titles. I don't really know how to handle audio delay on movies spread over multiple m2ts files yet. But I guess that's a headache to handle later on.


One thing that's bother me though is if I should use 23.976 as timecode or 24000/1001.

I can't get perfect smooth playback on some titles (don't know if the timecode provided when muxing the mkvs will make any diffrence). And I'm not sure if it's going to influence audio sync at all (tiny diffrence),

My TV (Pioneer LX608) accepts 24p. And the combination of this TV and a nvidia 8600gts (Vista 32) I can output 23, 24, 25, 29, 50, 59, 60.

When using 59 (I guess it's 59.97Hz) the audio falls behind (don't know if it's related to the 3:2 pulldown).
Choosing 23 (don't know how exact this is but I guess 23.976Hz) results in perfect audio sync but some titles stutters sometimes.

Does the timecode I set when remuxing affect this in anyway?
Right now I'm starting to believe that 23.976 gives the best result (over 24000/1001) but I'm not sure if it's just my imagination.

I know it's the wrong place for this discussion. But it would be great to hear if someone know how the timecode affects audio sync AND smooth video playback.

[rickardk]

Your file is a dts-hd track that has only the dts core with no extensions in. This means it's a normal dts file.
Except from the "garbage" it has in the beginning and the end. That's why eac3to can't recognize it.

Change the extension to .dts and use latest delaycut (v1.3.0.0) to fix the track (remove the garbage). Then it 'll work. I guess the same applies to the 7.1 track you have.

Where do I find delaycut v1.3.0.0? Can just find v1.2.1.2.

[nautilus7]

I have to ask this, sorry...
Does such thing as 23,976 really exist? Where does it come from? I believe it's 24000/1001(23,9760XXXXX) that is called 23,976 in sort. Nothing more. But maybe i just said a nonsense.

[nautilus7]

Where do I find delaycut v1.3.0.0? Can just find v1.2.1.2.
v1.3.0.0 is made by madshi (not the original author of delaycut), thus it can be found on this forum, in delaycut thread (use search).

EDIT: here's the link http://madshi.net/delaycut.rar

[rickardk]

I have to ask this, sorry...
Does such thing as 23,976 really exist? Where does it come from? I believe it's 24000/1001(23,9760XXXXX) that is called 23,976 in sort. Nothing more. But maybe i just said a nonsense.

Don't know. 24000/1001 is the standard for what most titles are mastered at I think. But I guess it's a rounding thing. And that's why I ask. If it will make any diffrence at all on smooth video playback and audio sync.

[nautilus7]

Exactly. So when 23,976 is used instead of 24000/1001 is wrong. Of course the difference is tiny, but there shouldn't be a debate whether the one or the other should be chosen.

[rickardk]

Exactly. So when 23,976 is used instead of 24000/1001 is wrong. Of course the difference is tiny, but there shouldn't be a debate whether the one or the other should be chosen.

Exactly what I think. But why does some titles playback perfect when timecode are set to 23.976 but NOT when set to 24000/1001. That's what I can't accept. One example is The Last Samurai HD DVD.
I guess I have a bad understanding in how the frames are handled...

[nautilus7]

You mean .mkv? Don't have a clue. Maybe an audio problem.
I have this hd dvd, maybe i try something tomorrow.

[rickardk]

You mean .mkv? Don't have a clue. Maybe an audio problem.
I have this hd dvd, maybe i try something tomorrow.

Yes mkv... Syriana is another title with the same problem (?).

[madshi]

One thing that's bother me though is if I should use 23.976 as timecode or 24000/1001.
The differerence is extremely small. But the more "correct" value is 24000/1001 and that's what the latest eac3to version is using.

I can't get perfect smooth playback on some titles
But audio sync is correct? This could be a software setup related problem.

My TV (Pioneer LX608) accepts 24p. And the combination of this TV and a nvidia 8600gts (Vista 32) I can output 23, 24, 25, 29, 50, 59, 60.
23 and 59 sound strange to me. It's strange to name "23.976" as "23". But who knows... Maybe PowerStrip would be worth a try for you?

Choosing 23 (don't know how exact this is but I guess 23.976Hz) results in perfect audio sync but some titles stutters sometimes.
You mean the video stutters sometimes? Is this with MPEG2, VC-1 or h264 (or with all three)? Maybe your PC is not fast enough to handle some of those movies?

Does the timecode I set when remuxing affect this in anyway?
You set a timecode when remuxing? How/where?

[buzzqw]

just playing with this tools.. great work!

just to add to wisth list
output to stdout (for using with neroaac/lame/oggenc...)
parsing of ts files

thanks!

BHH

[nautilus7]

I was about to ask for mp3, aac, vorbis support.

I know this tools focuses on HD DVD/Blu-ray audio, but these formats are useful to convert the audio commentaries to.

[madshi]

Which format do these tools expect via stdout?

[nautilus7]

I guess the q is for buzzqw. Don't know what stdout is anyway. :D

[rickardk]

But audio sync is correct? This could be a software setup related problem.

Yes it's in sync. I really think it's a frame rate/refresh rate missmatch problem.


23 and 59 sound strange to me. It's strange to name "23.976" as "23". But who knows... Maybe PowerStrip would be worth a try for you?

Powerstrip does not work well with the new line of nvidia cards. Don't know why they are naming the refresh rates like that.


You mean the video stutters sometimes? Is this with MPEG2, VC-1 or h264 (or with all three)? Maybe your PC is not fast enough to handle some of those movies?

It looks like video drops some frames (or not sync them with the refresh rate). The EVR renderer says no frames are dropped though. It's a very very subtle stutter. I can't spot it when I feed my display with 59.97 (the 3:2 pulldown mask small problems). I have a Q6600 at 3GHz. Should be enough. Yes I have seen this with all three codecs. The strange thing is that by setting a less exact (23.976) timecode thoose titles will playback perfect.

The example mentioned above The Last Samurai will have small micro stutters like once every minute when played back with the timecode for the mkv set to 24000/1001 (or if I play back the disc in PowerDVD). But when I rewrite the mkv with the timecode set to the less exact 23.976, the mkv plays perfect smooth all through the movie.

Don't know how the renderer is working but it may be correcting something to match audio and video sync when the timecode is set to 24000/1001.

To make it even more strange. NO title (yet) is experience any problem (the other way around) by using the less exact timecode.
So it may be better to have eac3to use 23.976 by default.

If someone (who like me have a 24p capable TV or projector and a graphic card that can output 24p) can test this it would be great!!

Done so much testing on this tonight that my eyes are bleeding...

You set a timecode when remuxing? How/where?
Don't know how to set 24000/1001 with a tmecode file in mkvmerge (can just set decimal values...any ideas?).

But to be able to compare I let eac3to set the timecode to 24000/1001 (don't know how you do this). Then I make a copy and run it through mkvmerge with a timecode file loaded with assume 23.976.

In mkvinfo the default duration for video frame shows the exact same value for mkvs with 24000/1001 and 23.976. I assumed that this would tell me that they should playback in the exact same way. But I guess I REALLY don't understand how the frames are handled.

For testing of Blu-rays I compare playback of the original m2ts with a mkv with timecode set to 23.976.

[madshi]

I guess the q is for buzzqw. Don't know what stdout is anyway. :D
Yeah, the q was for buzzqw. I could allow eac3to to output audio data through "stdout". If another program is written to accept data through "stdin" eac3to could pass audio data to the other program without having to write the data to a temporary file first.

[Thunderbolt8]

Don't know how to set 24000/1001 with a timecode file in mkvmerge (can just set decimal values...any ideas?).yes, you can only set decimal values, and not the "/". so what works best is calc, then calculate 24000/1001 and just copy & paste the result into that .txt file (and exchange the comma for a dot!)

[madshi]

It looks like video drops some frames (or not sync them with the refresh rate). The EVR renderer says no frames are dropped though. It's a very very subtle stutter. I can't spot it when I feed my display with 59.97 (the 3:2 pulldown mask small problems). I have a Q6600 at 3GHz. Should be enough. Yes I have seen this with all three codecs.
If you've seen it with all 3 codecs then it's most probably not caused by a too slow PC.

The strange thing is that by setting a less exact (23.976) timecode thoose titles will playback perfect.

The example mentioned above The Last Samurai will have small micro stutters like once every minute when played back with the timecode for the mkv set to 24000/1001 (or if I play back the disc in PowerDVD). But when I rewrite the mkv with the timecode set to the less exact 23.976, the mkv plays perfect smooth all through the movie.

Don't know how the renderer is working but it may be correcting something to match audio and video sync when the timecode is set to 24000/1001.

To make it even more strange. NO title (yet) is experience any problem (the other way around) by using the less exact timecode.
So it may be better to have eac3to use 23.976 by default.
Well, I'm not so sure about that. Obviously your graphics card is set to 23.976 and not to 24/1.001. Now I don't know why that is the case. Maybe your display is reporting that it wants 23.976 instead of 24/1.001. Or maybe your graphics card isn't able to output 24/1.001, but only 23.976. Either way it could be related to your specific display. Another person with different hardware could have an experience which is exactly the opposite of yours. One thing is for sure: 24/1.001 is more correct than 23.976 (although the difference is really small). If that wasn't the case I'd change back to 23.976 right now. But since 24/1.001 is the more correct value, I don't really like the idea of going back to 23.976 because of one single experience report, only. No offense! I do appreciate the feedback very much.

If someone (who like me have a 24p capable TV or projector and a graphic card that can output 24p) can test this it would be great!!
Good call! We do need testers here...

[rickardk]

I agree to 100%. But I'm starting to think that 3 decimals may be a maximum of what is used of modern graphic cards. I don't really have a clue, just speculations. But my display (TV) is the latest from Pioneer and should handle 24p perfect.
Actuallt I know it does, because I tested with a stand alone Pioneer Blu-ray a month ago with perfect result.

But what does that mean. I have searched the net all day for spec on what refresh rates modern 24p capable TVs and projectors expects. And that refresh rate that stand alone players actually outputs.

Also if someone know and could explain how the frames are handled by the renderer it would be great. Where does the rounding and calculation on how long to show each frame take placè? Does it take the refresh rate output into tte equation (matching)? I know that the use of reclock was essential before we had EVR.

As I'm going to´remux my whole collection it would be great to do it "right".

[madshi]

Ok, now if it's one of those new Kuro Pioneers then I might reconsider, since one of those may end up in my home, too, sooner or later... :D

[rickardk]

Ok, now if it's one of those new Kuro Pioneers then I might reconsider, since one of those may end up in my home, too, sooner or later... :D

It's a 60 inch 1080p Pioneer LX608 (G8 KURO)...
I'm about to test with my ATI2600 now...If it's related to the graphic card. I hope not because PQ is better on my nvidia 8600.

[buzzqw]

@madshi

Which format do these tools expect via stdout?

varius.. from raw pcm (aften) to WAV file, from mono to multi channels, sample rate can be 8000 to 48000., from 8 to 32 floating bit...
oggenc can parse
OggEnc input files must currently be 32, 24, 16, or 8 bit PCM WAV, AIFF, or AIFF/C files, or 32 bit IEEE floating point WAV. Files may be mono or stereo (or more channels) and any sample rate.
neroaacenc
The file must be in Microsoft WAV format and contain PCM data.
for enc_aacplus can be wav or raw

a messy.. .. anyway 16 bit wav file is accepted by all
if sample rate, bit depth, wav or raw pcm and channels can be specified, any external encoder will accept

about ts files (both mpeg and avc) ?

thanks for your interest

BHH

[tebasuna51]

Which format do these tools expect via stdout?

Same than wav files with standard header and:

Lame (.mp3): only mono or stereo and int samples 16, 24 or 32 bits (don't support float)

NeroAacEnc (.mp4): support 5.1 and any bitdepth 16, 24, 32 int or 32 float. Support big files > 4GB (with -ignorelength) if the value in field RiffLength is 36 + DataLength.

OggEnc2 (.ogg): same as NeroAacEnc (support for big files not tested)


EDIT: Hello buzzqw, we have crosspost. And yes Enc_aacplus to use CT aac encoder (from winamp) only work with 16 bit int.

[MuteyM]

Yeah, it seems to be a bug in the libav decoder. My guess is that the decoder believes that the truehd stream is done and finished and then surprisingly there's more truehd data coming in. And the decoder doesn't seem to like that. Should be easy to fix, though. Give the decoder developer a few days. His replies sometimes take a few days, but he always comes back with a fix.

Hi Madshi, since it might take awhile to get libav updated, any chance you can update eac3to to not delete the output file upon libav error?