eac3to - audio conversion tool

[mrr19121970]

I tried it 5 times again, and not once have I had the same problem. It must have been a fluke.

Thanks for looking though.

[mic64]

Thanks for the idea, but I get:


$ ls cdrom
AACS BDMV CERTIFICATE
$ wine /home/me/eac3to cdrom "1)"
The format of the source file could not be detected.


I should have mentioned that 'wine eac3to videofolder "1)"' works perfectly after the disc is decrypted and ripped to the hard disk.

you can´t just use the command like this:
eac3to /mnt/cdrom ....

you have to start with the letter that wine assigned for "My Computer".
it should look like this:
eac3to z:/mnt/cdrom/
then you can append the title.
eac3to z:/mnt/cdrom "1)"

[hubblec4]

OffTopic:

You know that there is an option to delete doubled Entries automaticly !

Thats an good working Workaround for that case ... not always perfect but an good option.

sorry but i don't know. can you tell me how and which programm you use??

hubble

[madshi]

eac3to v3.05 released

http://madshi.net/eac3to.zip

* warning is shown if h264 video bitstream contains "full range" flag
* h264 video bitstream "full range" flag is automatically removed
* you can disable removal of the "full range" flag by doing "-keepFullRange"
* added reader for external DVD, HD DVD and Blu-Ray SUP files
* external SUP files can be delayed now
* number of HD DVD and DVD subtitles in SUP track is counted and displayed
* number of forced and non-forced Blu-Ray subtitles in SUP track is displayed
* "-check" option now also works for demuxed audio, video and subtitle tracks
* when reading from physical disk drive, 2KB (instead of 1MB) blocks are read
* improved automatic skipping over damaged first 5MB of TS/m2ts files
* fixed: resampling and Surcode encoding didn't work in one step
* fixed: TRP detection crashed
* fixed: track listing sometimes contained tracks without description
* fixed: h264 with missing framerate in 1st sequence header made eac3to crash
* fixed: some AC3WAV files were not detected correctly
* fixed: video frame count was not displayed when 2nd pass was executed

[Espadon]

Hello Madshi,

Thanks for your time and your application.

I have been using eac3 version 3.04, trying to convert a normal DTS track to another DTS track (just needed to resample the track to PAL format), but has been experiencing a problem with it.
Here is the message i got after all the wav files were created....

"Encoding DTS <1536kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.23.0.
Surcode says/asks: "At least one valid source file must be specified to encode.".
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode..."

I tried using version 1.0.23
I'm trying right now with version 1.0.21....... is this a known issue?

Thanks

[Thunderbolt8]

thanks!

can you comment on what the difference will be regarding the change to 2KB blocks from 1MB when reading from HDD? I am thinking to get myself another HD to create raid 0 in order to speed up eac3to processing and especially because I hope it will take away some load from the disk. when its reading from the same drive windows is installed on as well, then I hardly can do something else until its finished (especially the more bitrate the streams have and the more tracks are processed at the same time). would raid 0 be able to change that, or would it only speed up the whole process, but the HDD load thing would remain as now?

[madshi]

I have been using eac3 version 3.04, trying to convert a normal DTS track to another DTS track (just needed to resample the track to PAL format), but has been experiencing a problem with it.
Here is the message i got after all the wav files were created....

"Encoding DTS <1536kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.23.0.
Surcode says/asks: "At least one valid source file must be specified to encode.".
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode..."
Surcode sometimes has problems with funny file names. Try renaming the DTS file to something very simple, e.g. "test.dts". Maybe that helps...

can you comment on what the difference will be regarding the change to 2KB blocks from 1MB when reading from HDD?
I found that reading physical discs is much faster when reading 2KB blocks instead of 1MB blocks. Don't ask me why, I don't really understand it. However, reading files from harddisk is about twice as fast with 1MB blocks compared to 2KB blocks. So v3.05 checks which drive the source file is read from. If it's a CDROM/DVD/Blu-Ray/HD DVD drive, 2KB blocks are used. Otherwise 1MB blocks are used.

I am thinking to get myself another HD to create raid 0 in order to speed up eac3to processing and especially because I hope it will take away some load from the disk. when its reading from the same drive windows is installed on as well, then I hardly can do something else until its finished (especially the more bitrate the streams have and the more tracks are processed at the same time). would raid 0 be able to change that, or would it only speed up the whole process, but the HDD load thing would remain as now?
I'd suggest not using raid, but just using a second separate HDD for video processing purposes. Not using the HDD where Windows is installed on should help a lot.

[DaMacFunkin]

Sorry Bit of a quickie question here: do you still need to have nero 7 installed to use this tool to decode DD+ streams to wavs? Thanks.

[Thunderbolt8]

I'd suggest not using raid, but just using a second separate HDD for video processing purposes. Not using the HDD where Windows is installed on should help a lot.
hm the way I have it now is that the partition windows is on is the reading partition, but the output is placed on another drive. but still, the reading alone already delays the system quite much.
I thought it might be more effective that way to seperate the reading and writing process to 2 different drives, as its faster to copy stuff from one HD to another than just to different folders on the same drive or also to mux with mkvmerge to another drive than the one you are reading from. so eac3to doesnt behave in the same way?

[idbirch2]

Great, thanks for the useful update. Being alerted to forced English subs at the time of demuxing will save a lot of time messing around with SupRead/Rip.

[bigdog660]

...* when reading from physical disk drive, 2KB (instead of 1MB) blocks are read...
* fixed: video frame count was not displayed when 2nd pass was executed

Thanks so much for this!

[Chumbo]

eac3to v3.05 released

http://madshi.net/eac3to.zip

* fixed: video frame count was not displayed when 2nd pass was executed
Thank you for all the updates and this one in particular. :)

[nautilus7]

eac3to v3.05 releasedThanks for the weekly update!

[rack04]

How will the forced subtitles be shown? I have Apocalypto BD and it's doesn't show any subtitles being forced.

[odin24]

Great, thanks for the useful update. Being alerted to forced English subs at the time of demuxing will save a lot of time messing around with SupRead/Rip.

+1 :thanks:

[bkman]

Hi madshi,

Excuse me if this is already an option and I've somehow missed it, but can you please add an option to disable the second pass upon clipping being detected?

Eac3to already takes long enough to convert audio of decent length without doing two passes, and I don't always care about some minor clipping artefacts.

[madshi]

hm the way I have it now is that the partition windows is on is the reading partition, but the output is placed on another drive. but still, the reading alone already delays the system quite much.
I thought it might be more effective that way to seperate the reading and writing process to 2 different drives, as its faster to copy stuff from one HD to another than just to different folders on the same drive or also to mux with mkvmerge to another drive than the one you are reading from. so eac3to doesnt behave in the same way?
I don't know, you can test it for yourself. I think performance will be better with eac3to, too, but it might still slow the OS down. Try reading+writing from/to the non-OS harddisk. That might take a longer time, but I think the OS should be more responsive that way.

You can also try the "-lowPriority" switch. That might also help OS responsiveness (on the cost of longer processing time). This switch should especially well on Vista and Windows 7. It should also help on XP and older OSs, but not as much as with Vista.

Sorry Bit of a quickie question here: do you still need to have nero 7 installed to use this tool to decode DD+ streams to wavs? Thanks.
See first post of this thread.

Excuse me if this is already an option and I've somehow missed it, but can you please add an option to disable the second pass upon clipping being detected?

Eac3to already takes long enough to convert audio of decent length without doing two passes, and I don't always care about some minor clipping artefacts.
I'd suggest to use e.g. "-3db" instead. That will lower volume a bit, which usually gets rid of clipping. As a result a 2nd pass will in most cases not be necessary, anymore. So this basically does what you are asking for, and has the additional bonus of getting rid of clipping. Of course it comes at the cost of generally lowered volume, whenever you use this switch...

How will the forced subtitles be shown? I have Apocalypto BD and it's doesn't show any subtitles being forced.
Here are some possible reports posted by eac3to:

[s07] Counted 23 forced subtitles.
[s08] Counted 167 subtitles.
[s09] Counted 158 normal and 5 forced subtitles.

This information is only posted after full processing has run through! It is not possible for eac3to to report this without reading the complete source files. So don't expect this information in the track listing...

[madshi]

Question:

Does anybody have a 6.1 or 7.1 speaker setup and a Dolby Digital EX capable AVR and the possibility to send 6.1 or 7.1 PCM via HDMI to the receiver and would be willing to do some tests for me? If so, please PM me (don't post in this thread). Thanks!

[jwdaigle]

I am having troubles getting the arcsoft decoders to work with eac3to. When I run eac3to -test, I get "ArcSoft DTS Decoder doesn't seem to be installed". Im running 3.04 of eac3to, and just purchased TMT (the download gave me .126). The codecs do work with Graphstudio (using render media file), so that proves they are registered.

So how does eac3to decide if TMT is installed? Does it have anything to do with these 2 lines from a Process Monitor output (the referenced file does not exist in my fresh .126 retail installation):

5:20:13.4001806 PM eac3to.exe 5220 QueryOpen C:\Program Files\Common Files\ArcSoft\Bin\RPAWINET.DLL FAST IO DISALLOWED
5:20:13.4003641 PM eac3to.exe 5220 CreateFile C:\Program Files\Common Files\ArcSoft\Bin\RPAWINET.DLL NAME NOT FOUND Desired Access: Read Attributes, Disposition: Open, Options: Open Reparse Point, Attributes: n/a, ShareMode: Read, Write, Delete, AllocationSize: n/a

Is the existence of RPAWINET.DLL in the arcsoft bin directory the test to see if it is installed?

I have already tried adding Program files\common files\arcsoft\bin to the path before running eac3to -test. I have also tried copying the 4 mag*.dll files into the eac3to folder from arcsoft. As I mentioned earlier, the codecs are registered (I had to manually register them using regsvr32). I didnt register all of them (only the ones that seemed to be needed for video and audio playback), but will try that later.

Any help greatly appreciated.

[mbcd]

Hi Madshi

I didn`t tried new eac3to, but is it possible to "mark" subtitles which have forced subs inside by filename ?

I would like to see if thes special track is ONLY forced, or it is normal subs with forced inside.

Could be nice to know sometimes. Could you add this if it make sence to you?

BTW:

My Virus-Scanner shows now some funny Warning:

Madshi wants to take controll of all our Computers :sly: :angry: :devil: :p

Regards

[Sylver]

Is it possible to use Neroaacenc.exe and use a bitrate ? (instead of quality)
And profile HEv2 ?

And when i try to convert DTS-MA to AAC (without ArcSoft/Sonic), i have the message : "The libav DTS decoder doesn't decode the full DTS-HD information.". How will be my aac file ? Correct or not ?

Thanks :)

[Greif]

yeah use search to find the post showing how to pass via stdout to neroaacenc using whatever parameters you want

[tebasuna51]

@madshi
Please can you solve this 'bug' :cool:
eac3to 6o321.wav dplII.wav -down2

WAV, 5.1 channels, 0:00:20, 16 bits, 4608kbps, 48khz
Can't downmix this channel configuration ($3f).
Internal error - unknown audio output format!

With mask channel $60f work but the output from FAAD decoder and OGGDEC decoder have WAVE_FORMAT_EXTENSIBLE header with mask channel $3f.

I know M$ prefer now $60f but say also to be backward compatible...

The two only free decoders with WAVE_FORMAT_EXTENSIBLE output are punished with this.

Or maybe you prefer support aac,m4a,ogg decode with libav?

[madshi]

I am having troubles getting the arcsoft decoders to work with eac3to.
There have been many problems getting them to work. Usually manually registering the filters and adding the path to the environment settings (and a reboot afterwards) has helped most people. Some people had success installing the trial version first and then upgrading it to the full version. Don't know what else to say right now.

I didn`t tried new eac3to, but is it possible to "mark" subtitles which have forced subs inside by filename ?

I would like to see if thes special track is ONLY forced, or it is normal subs with forced inside.
eac3to only knows this after processing has run through, but I guess I could rename the sup files as a last step. I'll think about it...

My Virus-Scanner shows now some funny Warning
False alarm. Please write a complaint to your anti virus company.

Is it possible to use Neroaacenc.exe and use a bitrate ? (instead of quality)
Yes. E.g. "eac3to source.whatever dest.aac -192".

And profile HEv2 ?
AFAIK, the Nero encoder automatically chooses which encoding methods to use, depending on the bitrate/quality. Anyway, eac3to currently doesn't let you specify whether HEv2 should be used or not. If you want full control over the Nero AAC encoder, you may want to use piping, see Greif's comment.

And when i try to convert DTS-MA to AAC (without ArcSoft/Sonic), i have the message : "The libav DTS decoder doesn't decode the full DTS-HD information.". How will be my aac file ? Correct or not ?
Correct yes, but slightly lower quality compared to what could be achieved by using a DTS-HD capable decoder.

Please can you solve this 'bug' :cool:
Will be fixed in the next build. This was supposed to work already, so I consider it a clear and simple bug.

Or maybe you prefer support aac,m4a,ogg decode with libav?
m4a and ogg containers are very difficult to support because eac3to doesn't even recognize their format. I'd have to write full mp4/ogg container parsers to make this work with the current eac3to code structure. I could add AAC decoding support with libav. But AFAIK the libav AAC decoder is not yet feature complete, so I'm afraid of adding compatability problems. I rather want to wait until libav's AAC decoder can decode all AAC formats before supporting it.

[Sylver]

Thanks for ur answers !

[mrr19121970]

I did some updates in the meantime for Clown_BD GUI:

Selecting Audio/Subtitle streams based on language or type:
http://usera.ImageCave.com/mrr19121970/GUI/New_Step3.jpg

Direct Burning of created BD structure (instead of creating a .ISO):
http://usera.ImageCave.com/mrr19121970/GUI/New_Step1-BURN.jpg

See here eacto, tsMuxer & ImgBurn Made Easy (http://forum.slysoft.com/showthread.php?p=166910) for more details.

[Blackadder]

Hello. I have a question. I tried to encode dts from an lpcm track, In the middle of the process i got an error. This is only the 2nd time that i've used this program so i don't have much experience with it. I have here the log file and hopefully someone can tell me what i have done wrong. By the way, i have installed Surcode dvd dts.

Thanks in advance !
This is the log file:

eac3to v3.05
command line: "F:\HD Appz\EAC3to\eac3to.exe" "N:\Sin City.m2ts" 2: "N:\SinC.dts" -1536
------------------------------------------------------------------------------
M2TS, 1 video track, 1 audio track, 1 subtitle track, 2:04:01, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)
2: RAW/PCM, 7.1 channels, 24 bits, 48khz
3: Subtitle (PGS)
[a02] The Surcode DTS encoder doesn't support back channels. Will downmix them.
[a02] Extracting audio track number 2...
[a02] Reading RAW/PCM...
[a02] Swapping endian...
[a02] Remapping channels...
[a02] Mixing surround channels...
[a02] Writing WAVs...
[a02] Creating file "N:\SinC.L.wav"...
[a02] Creating file "N:\SinC.R.wav"...
[a02] Creating file "N:\SinC.LFE.wav"...
[a02] Creating file "N:\SinC.SR.wav"...
[a02] Creating file "N:\SinC.C.wav"...
[a02] Creating file "N:\SinC.SL.wav"...
[a02] The original audio track has a constant bit depth of 24 bits.
[a02] The processed audio track has a constant bit depth of 24 bits.
Encoding DTS <1536kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.23.0.
Surcode says/asks: "At least one valid source file must be specified to encode.".
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...

[mikelebron]

Why isnt the ArcSoft TrueHD and ArcSoft (E-)AC3 supported but the Arcsoft DTS-HD is?

[lexor]

Why isnt the ArcSoft TrueHD and ArcSoft (E-)AC3 supported but the Arcsoft DTS-HD is?

Because it's a lot more work, and unlike DTS components there is no better substitute.

TrueHD - libav handles perfectly
AC3 - handled by Nero 7 (yes it's old and unavailable if you don't already have it... but libav is good enough here too)

On the other hand nothing handles the plethora of DTS formats quite as well as ArcSoft.

[mikelebron]

Makes sense... Question.. I noticed my MKVs that were remuxed from TrueHD, DTS-MA, or PCM to FLAC have a much lower volume... How can I correct this? Any tips?

[Snowknight26]

Any reason behind patching DTS source bit depth flag?

[jwdaigle]

I am posting this in the hopes that it helps someone else with a similar configuration to me. Before I got this working, eac3to -test would tell me that it doesnt like like the arcsoft dts decoder is installed.

I am running Vista32 Service Pack1 with no "codec packs" installed. My aim was to have the ability to play video files (usually in mkv files) with a minimum of crud added to my system.

Prerequisites:
- You have successfully installed Arcsoft TMT Retail. This info is not meant to help you pirate the Arcsoft codecs, but rather you bought the product and want to use the codecs outside of the TMT environment.
- You can play video files with the Arcsoft player (I did all my testing with .mkv files with 264 video inside). ie, the arcsoft retail product works on your machine.


If you cant do these 2 things, it is unlikely eac3to can use it.

I also installed the haali media splitter to be able to read mkv files.

Setup:
Really only two things are required.

- The dll files in C:\Program files\Common Files\Arcsoft\Bin need to be accessible in your path. Do this by going to control panel/user accounts. Select "Change My environment variables" from the left hand side. If there is already a Path variable in the top listbox, select it and click edit - if there isnt click on new. Type C:\Program Files\Common Files\Arcsoft\Bin in "variable value" and "Path" in the Variable name. Click ok/ok/ok.
- The Arcsoft audio codecs must be registered. I am not sure if I registered more than I needed to, but the ones I registered are: ASAudio.ax, ASAudioEffectHD.ax, ASAudioHD.ax, ASVid.ax, VideoRenderForEVR.ax, VideoRenderForVMR.ax. You can do this by running a command prompt as administrator (right click on command and select "Run as administrator") and typing regsvr32 "C:\Program Files\Arcsoft\TotalMedia Theater\Codec\each_file_in_my_list.ax"

You should get a "registration of ... was successful.

I then rebooted.

After this brief process, I re-ran eac3to -test (3.05 for me) and it now reported that it can see the arcsoft codecs.

I didnt need to copy any of the dlls around (thats what the path is for).

Success!

I also suggest that you get a copy of the (free) GraphStudio to more fully test that the arcsoft video/audio codecs are working well.

Hope this helps -

Joe

[Blue_MiSfit]

Hi Folks

*GOSH this thread is getting HUGE!!!

I've got a problematic source, which is a MPEG Program Stream with MPEG-2 video and MP2 audio at 192kbps. I want to transcode the MP2 to an AC3. Usual command results in this output:

I:\Rework\SD_HD\141588>eac3to RiseOfTheFootSoldier_178_2007.mpg 2:audio.ac3 -192

MPG, 1 video track, 1 audio track, 1:59:03
1: MPEG2, 1080i60 /1.001 (16:9)
2: MP2, 2.0 channels, 192kbps, 48khz
a02 Extracting audio track number 2...
a02 Decoding with libav/ffmpeg...
a02 Encoding AC3 <192kbps> with libAften...
a02 Creating file "audio.ac3"...
a02 Audio overlaps for 48ms at playtime 0:00:00.
a02 Audio overlaps for 48ms at playtime 0:00:00.
a02 Audio overlaps for 48ms at playtime 0:00:00.
a02 Starting 2nd pass...
a02 Creating file "audio.ac3"...
Video track 1 contains 214074 frames.
eac3to processing took 15 minutes, 1 second.
Done.


So, it looks like there are some gaps. Fine, there are no sync issues with the source or the resulting AC3. However, the resulting AC3 has some nasty clipping during loud portions! Any way to fix this? I also get the clipping if I decode the MP2 to WAV, so it seems like that's where it's introduced.

I'll be watching. Madishi, if you want the MP2 I can provide it!

~MiSfit

[Blue_MiSfit]

Okay, so some followup.

I can only get a clean decoded WAV if I decode to 32 bit float (using foobar2000). Unfortunately my DP600 only takes 24 bit WAV or BWF PCM input... :)

Trying more things...

~MiSfit

[Blue_MiSfit]

Finally, success! I had to demux to MP2 with eac3to, then use foobar to decode to full 32 bit float, then use eac3to to encode this to AC3. It detected the clipping, fixed it, and there was much rejoycing!


D:\Dolby\DolbyInput>eac3to footsoldier_32bit.wav footsoldier.ac3 -192
WAV, 2.0 channels, 1:59:03, 32 bits <float>, 3072kbps, 48khz
Reading WAV...
Encoding AC3 <192kbps> with libAften...
Creating file "footsoldier.ac3"...
Clipping detected, a 2nd pass will be necessary.
Original audio track: max 32 bits, average 30 bits, most common 29 bits.
Starting 2nd pass...
Reading WAV...
Encoding AC3 <192kbps> with libAften...
Creating file "footsoldier.ac3"...
The processed audio track has a constant bit depth of 64 bits.
eac3to processing took 2 minutes, 21 seconds.
Done.

D:\Dolby\DolbyInput>


Hooray!!!! This title has been bugging me for AGES!

The real question is - why couldn't eac3to detect this in the original MP2, and run a second pass to fix the clipping? This roundabout business wasn't much fun..

~MiSfit

[DrNein]

jwdaigle (Joe),

If only DTS decode is desired then the minimal files required are the filter (registered) ASAudioHD.ax, plus the libraries DtsDec.dll and dtsdecoderdll.dll. The others required in the path to allow using any filters are checkactivate.dll, MagCore.dll, MagPCMac.dll, MagUIEngine.dll, and MagUIInter.dll.

Do you find any of the other filters useful externally such as for playback with MPC-HC?

[ACrowley]

Makes sense... Question.. I noticed my MKVs that were remuxed from TrueHD, DTS-MA, or PCM to FLAC have a much lower volume... How can I correct this? Any tips?

You get exactly the same Volume when reencoding to FLAC

FLAC reencoding is lossless and no Volume/DynamicRangeCompression etc will be applied from FLAC encoder.

You Problem can only be caused by :

1. Youre using a Decoder in eac3to with Dynamic Range Compression or Dialnorm enabled.
-Which Decoder are you using in ea23to ?

2. Youre using a FLAC Decoder with some other Output Settings.
-Take Madflac !

3. Youre using a DTS/ TrueHD Decoder for Playback with some changed Volume Settings as like Normalization, Gain etc...
Which player/Decoder are you using for Playback

I think Nr 3 and or 2 is your Problem....

[madshi]

However, the resulting AC3 has some nasty clipping during loud portions! Any way to fix this? I also get the clipping if I decode the MP2 to WAV, so it seems like that's where it's introduced.

I'll be watching. Madishi, if you want the MP2 I can provide it!
Yes, please.

[Amefurashi]

Hello,

I was trying to convert a 6.1ch DTS-MA track in a AAC-MP4 one using eac3to 3.05, keeping the original channels. Log says:


eac3to.exe English.dtsma English.mp4 -quality=0.30
DTS Master Audio, 6.1 channels, 24 bits, 48khz
(core: DTS-ES, 6.1 channels, 24 bits, 1509kbps, 48khz)
NeroAacEnc doesn't support 6.1 encoding. Will double the 7th channel.
Doubling 7th channel...
Decoding with ArcSoft DTS Decoder...
Remapping channels...
Encoding AAC <0.30> with NeroAacEnc...


Commandline is the following:

eac3to.exe English.dtsma English.mp4 -quality=0.30

IIRC NeroAACEnc manages to encode streams up to 8 channels. Am I doing something wrong?

[shambles]

with the new version, in case of truehd tracks, eac3to always removes zero-bytes after encoding to flac.. isn't the bluray truehd spec limited to 24bit? so wouldn't it make more sense to always truncate to 24bit and then do the extra pass if it's only 16bits?

[madshi]

I was trying to convert a 6.1ch DTS-MA track in a AAC-MP4 one using eac3to 3.05, keeping the original channels.

IIRC NeroAACEnc manages to encode streams up to 8 channels. Am I doing something wrong?
AFAIK Nero can encode 5.1 and 7.1, but not 6.1. So eac3to converts your 6.1 track to 7.1 to make it encodable with Nero.

isn't the bluray truehd spec limited to 24bit?
Huh? TrueHD is *always* 24bit. Never lower, never higher. But it's possible that only some of those 24 bits are filled with actual data, with the remaining bits being zeroed out.

[shambles]

oh right. but then why is this happening:

eac3to v3.05
command line: eac3to E:\D181716D1V2 1) 3: f:\terror.flac
------------------------------------------------------------------------------
M2TS, 1 video track, 6 audio tracks, 2 subtitle tracks, 1:45:33, 24p /1.001
1: Chapters, 31 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB)
4: AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
5: AC3, Spanish, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
6: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
7: AC3, English, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB
8: AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
9: Subtitle (PGS), English
10: Subtitle (PGS), Spanish
[a03] Extracting audio track number 3...
[a03] Extracting TrueHD stream...
[a03] Removing TrueHD dialog normalization...
[a03] Decoding with libav/ffmpeg...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "f:\terror.flac"...
[a03] The original audio track has a constant bit depth of 24 bits.
[a03] Superfluous zero bytes detected, will be stripped in 2nd pass.
[a03] Starting 2nd pass...
[a03] Decoding FLAC...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "f:\terror.flac"...
Video track 2 contains 151849 frames.
eac3to processing took 26 minutes, 35 seconds.
Done.

[bigdog660]

I had the chance to demux 3 BD movies last night, and I have attached their logs.

eac3to v3.05
command line: "D:\HDDVDTL\eac3to.exe" "E:\" 1) 2: "D:\DISASTER_MOVIE\feature.mkv" 1: "D:\DISASTER_MOVIE\chapters.txt" 4: "D:\DISASTER_MOVIE\feature.ac3" -core
------------------------------------------------------------------------------
M2TS, 2 video tracks, 2 audio tracks, 3 subtitle tracks, 1:28:28, 24p /1.001
1: Chapters, 17 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: h264/AVC, 480p24 /1.001 (20:11)
4: DTS Master Audio, English, 7.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
5: AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
6: Subtitle (PGS), English
7: Subtitle (PGS), English
8: Subtitle (PGS), Spanish
Creating file "D:\DISASTER_MOVIE\chapters.txt"...
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[a04] Extracting DTS core...
[v02] Muxing video to Matroska...
[a04] Decoding with ArcSoft DTS Decoder...
[a04] Encoding AC3 <640kbps> with libAften...
[a04] Creating file "D:\DISASTER_MOVIE\feature.ac3"...
[a04] The original audio track has a constant bit depth of 24 bits.
Added fps value to MKV header.
Video track 2 contains 127258 frames.
Video track 3 contains 127258 frames.
eac3to processing took 1 hour, 44 minutes.
Done.

eac3to v3.05
command line: "D:\HDDVDTL\eac3to.exe" "E:\" 1) 2: "D:\PRIDE_AND_GLORY\feature.mkv" -stripPulldown 1: "D:\PRIDE_AND_GLORY\chapters.txt" 3: "D:\PRIDE_AND_GLORY\feature.ac3"
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 2 subtitle tracks, 2:10:09, 24p /1.001
1: Chapters, 33 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
4: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB)
5: Subtitle (PGS), English
6: Subtitle (PGS), Spanish
Creating file "D:\PRIDE_AND_GLORY\chapters.txt"...
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a03] Removing AC3 dialog normalization...
[v02] Muxing video to Matroska...
[a03] Creating file "D:\PRIDE_AND_GLORY\feature.ac3"...
Added fps value to MKV header.
Video track 2 contains 187239 frames.
eac3to processing took 1 hour, 3 minutes.
Done.

eac3to v3.05
command line: "D:\HDDVDTL\eac3to.exe" "E:\" 1) 2: "D:\THE_LOVE_GURU\feature.mkv" 1: "D:\THE_LOVE_GURU\chapters.txt" 3: "D:\THE_LOVE_GURU\feature.ac3" -core
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 5 subtitle tracks, 1:26:41, 24p /1.001
1: Chapters, 20 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB)
4: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
5: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
6: Subtitle (PGS), English
7: Subtitle (PGS), English
8: Subtitle (PGS), French
9: Subtitle (PGS), Spanish
10: Subtitle (PGS), Portuguese
Creating file "D:\THE_LOVE_GURU\chapters.txt"...
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a03] Extracting AC3 stream...
[v02] Muxing video to Matroska...
[a03] Removing AC3 dialog normalization...
[a03] Creating file "D:\THE_LOVE_GURU\feature.ac3"...
Added fps value to MKV header.
Video track 2 contains 124704 frames.
eac3to processing took 1 hour, 22 minutes.
Done.

As you can see the processing took anywhere from 1:03:00 to 1:44:00 which is much better than v3.04, but not as good a v2.87. As previously discussed via email, v2.87 performance was fast enough demux a 1:45 hour movie in 27 minutes.

What size blocks did you use in v2.87? Was it 4K blocks? Maybe it's better to go back to that size or even try 8K blocks?

One of the areas v3.05 really shines is reading disc structure ["E:\" and "E:\" 1)]! Much faster than v2.87 or v3.04.

As always, you work is really appreciated!:thanks:

[Amefurashi]

AFAIK Nero can encode 5.1 and 7.1, but not 6.1. So eac3to converts your 6.1 track to 7.1 to make it encodable with Nero.

I didn't know that. :(

Changelog of version 2.51 says:

Release Date: Jul 1, 2008
* added (undocum.) option "-no7doubling" to disable 6.1 -> 7.1 channel doubling

This option doesn't exist anymore: may I ask what did it "technically" do?


BTW thanks for this awesome tool, keep up the good work! ;)

[nautilus7]

oh right. but then why is this happening:This is exactly what madshi explained to you, and because i can't tell in a better way, read it again from him.

[Blue_MiSfit]

@Madshi:

The clipping MP2 is uploading. I re-verified that both MPG -> MP2 -> AC3 and MPG -> AC3 both produce clipping, but MPG -> MP2 -> 32float WAV (with Foobar) > AC3 (with eac3to) works fine.

I'll update this post with the link when it's done uploading.


http://www.megaupload.com/?d=NNYU3RBX


~MiSfit

[shambles]

This is exactly what madshi explained to you, and because i can't tell in a better way, read it again from him.

but the log shows that the track has a constant bit depth of 24bits, so why is eac3to detecting zero-bytes and performing the 2nd pass? this is happening with all truehd tracks, i've done like 15 of them in the last couple days

[lexor]

but the log shows that the track has a constant bit depth of 24bits, so why is eac3to detecting zero-bytes and performing the 2nd pass? this is happening with all truehd tracks, i've done like 15 of them in the last couple days

Consider numbers: 1 , 01, 001, 0001 etc.

They are all the same, they are all 1. Prefixing a number with zeroes does not alter the value. TrueHD tracks are padded to 24 bits. So if source was 16bit PCM it would be padded to 24bits with zeroes by the TrueHD encoder. What eac3to is doing is detecting that padding and removes it on the second pass.

[Blue_MiSfit]

I ran into another MP2 source that had the same clipping issue today, and the same fix worked. If you need it, let me know.

~MiSfit

[shambles]

Consider numbers: 1 , 01, 001, 0001 etc.

They are all the same, they are all 1. Prefixing a number with zeroes does not alter the value. TrueHD tracks are padded to 24 bits. So if source was 16bit PCM it would be padded to 24bits with zeroes by the TrueHD encoder. What eac3to is doing is detecting that padding and removes it on the second pass.

there's no padding! the resultant flac tracks are 24bit. i know eac3to removes zero-bytes if there's only 16bits of valid information but these are all tracks that have full 24 valid bits

[lchiu7]

Been trying to use a program tonmt which is a wrapper to transform BD and HD-DVD titles into a format that can be played by the Popcorn Hour. A critical component is eac3to. I am new to this so bear with me. Running tonmt on the HD-DVD title Shooter, it seems to fail on eac3to and since there are no java errors I wonder it it's a problem with either my configuration (Vista 32 SP1) or with eac3to (or something else). The HD-DVD was ripped to my local HD with AnyDVD. No errors occurred in the rip and all previous steps including merging the EVO and pulldown seemed to complete okay

This is my eac3to status log

eac3to (v3.05) is up to date
Nero Audio Decoder (Nero 7) works fine
ArcSoft DTS Decoder doesn't seem to be installed
http://www.arcsoft.com/products/totalmediatheatre
Sonic Audio Decoder (3.5.0.0) doesn't seem to be installed
Haali Matroska Muxer (2007-06-03) is installed
There's a new version (2009-01-11) available
http://haali.net/mkv
Nero AAC Encoder could not be located
http://www.nero.com/eng/nero-aac-codec.html
Copy NeroAacEnc.exe to the eac3to or to the Windows folder.
Surcode DTS Encoder doesn't seem to be installed
http://www.surcode.com
MkvToolnix (2.2.0.0, 2008-03-04) is installed
There's a new release version (2.4.1.0) available
http://www.bunkus.org/videotools/mkvtoolnix
There's a new beta version (2.4.2.0, 2009-01-18) available
http://www.bunkus.org/videotools/mkvtoolnix/win32/pre

However when eac3to runs on the eac3 file created by tonmt the following error occurs

eac3to v3.05
command line: C:\apps\ToNMT_5.1.0\eac3to\eac3to.exe C:\Temp\HDDVD_c0.eac3 C:\Temp\HDDVD.AC3
------------------------------------------------------------------------------
E-AC3, 5.1 channels, 2:05:44, 1536kbps, 48khz, dialnorm: -27dB
Disabling DRC for Nero (E-)AC3 decoding...
Removing E-AC3 dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
Aborted at file position 1448429568. <ERROR>

Out of interest tried running an earlier version of eac3to. Not sure which version since -test doesn't provide version information but dated Jan 2008

This runs to completion but produces no output file

No idea what to do next? Any assistance appreciated

Thanks

[lexor]

there's no padding! the resultant flac tracks are 24bit. i know eac3to removes zero-bytes if there's only 16bits of valid information but these are all tracks that have full 24 valid bits
You should've mentioned that, seeing how the log does not.

If you did, it would've sounded like a bug report and not like you asking for someone to explain to you the meaning of those lines.

[shambles]

i should've phrased the first post better, yes, but the log in the second post does clearly show that the audio track has a constant bit depth of 24 bits

[Sharc]

When I use -speedup it seems that eac3to always assumes 23.976fps for the audio source.
How can I speedup 24.0000 fps (film) => 25 fps?

[lexor]

i should've phrased the first post better, yes, but the log in the second post does clearly show that the audio track has a constant bit depth of 24 bits

huh? no it does not. It shows that original track (TrueHD, 3: selection) has 24bits. Which as madshi explained and I elaborated it always would. The log says nothing about output bitdepth of the flac file.

[shambles]

when eac3to reports "original audio track has" this or that bit depth after processing an audio track, it shows the result of the bit depth analysation. it's always the number of valid bits.

for example, a 16bit truehd track:

eac3to v3.05
command line: eac3to G:\BDROM 1) 3: f:\amelie.flac
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 4 subtitle tracks, 2:01:35, 24p /1.001
1: Chapters, 12 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, French, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 640kbps, 48khz)
4: RAW/PCM, Japanese, 2.0 channels, 16 bits, 48khz
5: AC3, French, 2.0 channels, 192kbps, 48khz
6: Subtitle (PGS), Japanese
7: Subtitle (PGS), French
8: Subtitle (PGS), Japanese
9: Subtitle (PGS), Japanese
[a03] Extracting audio track number 3...
[a03] Extracting TrueHD stream...
[a03] Decoding with libav/ffmpeg...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "f:\amelie.flac"...
[a03] The original audio track has a constant bit depth of 16 bits.
[a03] Superfluous zero bytes detected, will be stripped in 2nd pass.
[a03] Starting 2nd pass...
[a03] Decoding FLAC...
[a03] Reducing depth from 24 to 16 bits...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "f:\amelie.flac"...
[a03] The processed audio track has a constant bit depth of 16 bits.
Video track 2 contains 174912 frames.
eac3to processing took 20 minutes, 49 seconds.
Done.

[lexor]

hmm, does sound like something is wrong, so we'll have to wait for madshi to look into it. It could still be the case that the second pass is unrelated to the bitdepth stuff (maybe it's one of those branching or other special structures on the disk) that require second pass to get the timings right.

so it could be just bug in logging.

[madshi]

oh right. but then why is this happening:

[a03] The original audio track has a constant bit depth of 24 bits.
[a03] Superfluous zero bytes detected, will be stripped in 2nd pass.
Yeah, that is weird. The track can either have a constant bit depth of 24 bits or there can be zero bytes, but not both at the same time. Will have to check that...

I had the chance to demux 3 BD movies last night, and I have attached their logs.

As you can see the processing took anywhere from 1:03:00 to 1:44:00 which is much better than v3.04, but not as good a v2.87. As previously discussed via email, v2.87 performance was fast enough demux a 1:45 hour movie in 27 minutes.

What size blocks did you use in v2.87? Was it 4K blocks? Maybe it's better to go back to that size or even try 8K blocks?
I've tested the best speed with my specific Blu-Ray drive and it was with 2K blocks. v2.87 used 64K blocks, IIRC, but it performed noticably worse on my PC. So maybe the optimal block read size depends on the drive model? Uh, that's ugly. Will have to think about how to solve that.

Have you done that other test I asked you to do: Namely ripping the movie to harddisk first and then comparing v2.87 vs. v3.05 when running on the harddisk folder? I think v3.05 should be noticably faster in that situation...

This option doesn't exist anymore: may I ask what did it "technically" do?
Why would you want to know what an old option did, which doesn't even exist, anymore?

The clipping MP2 is uploading. I re-verified that both MPG -> MP2 -> AC3 and MPG -> AC3 both produce clipping, but MPG -> MP2 -> 32float WAV (with Foobar) > AC3 (with eac3to) works fine.
Thanks, I'll check this out.

Been trying to use a program tonmt which is a wrapper to transform BD and HD-DVD titles into a format that can be played by the Popcorn Hour. A critical component is eac3to. I am new to this so bear with me. Running tonmt on the HD-DVD title Shooter, it seems to fail on eac3to and since there are no java errors I wonder it it's a problem with either my configuration (Vista 32 SP1) or with eac3to (or something else). The HD-DVD was ripped to my local HD with AnyDVD. No errors occurred in the rip and all previous steps including merging the EVO and pulldown seemed to complete okay

Out of interest tried running an earlier version of eac3to. Not sure which version since -test doesn't provide version information but dated Jan 2008

This runs to completion but produces no output file

No idea what to do next? Any assistance appreciated
Please try using "-libav" to use the libav decoder instead of Nero. Maybe that helps? How big is that E-AC3 track? Can you reproduce the problem with a small sample of the track? If so, it would be nice if you could upload it for me to test...

When I use -speedup it seems that eac3to always assumes 23.976fps for the audio source.
How can I speedup 24.0000 fps (film) => 25 fps?
When the audio is in a container eac3to automatically applies 24.000 -> 25.000 speedup. When the audio is demuxed, eac3to cannot possibly know which FPS the audio has, so it assumes the usual case of 23.976 -> 25.000. You can tell eac3to which FPS the demuxed audio file has by using the parameter "-24.000". So the command line would be: "eac3to source.whatever dest.whatever -24.000 -speedup".

[Sharc]

All clear now. Thanks madshi.

[nautilus7]

TrueHD sample (http://www.sendspace.com/file/3vfsp7) that shows same behavior. Maybe it happens with all real 24 bit tracks...

eac3to aaa.thd aaa.flac
TrueHD, 5.1 channels, 48khz
Decoding with libav/ffmpeg...
Encoding FLAC with libFlac...
Creating file "aaa.flac"...
The original audio track has a constant bit depth of 24 bits.
Superfluous zero bytes detected, will be stripped in 2nd pass.
Starting 2nd pass...
Decoding FLAC...
Encoding FLAC with libFlac...
Creating file "aaa.flac"...
eac3to processing took 9 seconds.
Done.

eac3to aaa.flac
FLAC, 5.1 channels, 0:00:36, 24 bits, 2051kbps, 48khz

[Thunderbolt8]

might be only a display bug. the trueHD track for the zodiac BD is 3,43 GB, and the constant 24-bit track with then supposed to be superflous zero bytes still has 3,23 GB of size. should be less in case they were really stripped. mediainfo also tells VALID_BITS: 24 for that track.

[shambles]

i've ripped ~30 truehd tracks to flac in the last couple days and it always does the 2nd pass whether the tracks have 24 valid bits or not. with 24bit tracks it doesn't seem to remove anything tho, but it does make the processing take more time obviously.

[nautilus7]

Can you name some movies to confirm whether they are true 24bit? For example, the sample i posted above is from Planet Terror.

Just curious...

EDIT: I can confirm about Godfather 1,2,3. I don't remember/know about the others. So it seems it's a general bug.

[shambles]

planet terror yes, and lately i also ripped godfather 1&2 (part3 probably also had 24bit track but didn't rip that yet), dexter season 1 (all 12 eps), spider-man 3 and lucky number slevin usa release and they all have full 24bit truehd tracks

[bigdog660]

Have you done that other test I asked you to do: Namely ripping the movie to harddisk first and then comparing v2.87 vs. v3.05 when running on the harddisk folder? I think v3.05 should be noticably faster in that situation...

I did that test for v3.04: "After dumping the ISO to the HDD using AnyDVD HD, and loading the ISO using Virtual CloneDrive, eac3to 3.04 only took 15 minutes and 56 seconds! And that was reading and writing to the same drive (D:\)! Also the command eac3to J:\ and eac3to J:\ 1) were extremely fast executing."

I will now do the same test for v2.87 and v3.05 plus I will give you data for a rip from the BD drive just for additional info.

[73ChargerFan]

I've tested the best speed with my specific Blu-Ray drive and it was with 2K blocks. v2.87 used 64K blocks, IIRC, but it performed noticably worse on my PC. So maybe the optimal block read size depends on the drive model? Uh, that's ugly. Will have to think about how to solve that.

I can think of two ways:

1 - post a utility that users can run to benchmark the read speed of their bd or bd/hd-dvd drive, and which when run will email to a dedicated address the results, including drive model & os. Evaluate and include in the program.

2 - include the benchmark routine in eac3to, it can be run by a command line switch and store the value in the registry or in an .ini file. When eac3to runs, it will use a default value unless it the .ini file is present, in which case it loads the value from there.

[kypec]

I can think of two ways:

1 - post a utility that users can run to benchmark the read speed of their bd or bd/hd-dvd drive, and which when run will email to a dedicated address the results, including drive model & os. Evaluate and include in the program.

2 - include the benchmark routine in eac3to, it can be run by a command line switch and store the value in the registry or in an .ini file. When eac3to runs, it will use a default value unless it the .ini file is present, in which case it loads the value from there.:eek:
Sounds like too sophisticated solution for such sleek CLI application to me.
Why not just introduce another option switch -buffer with variable parameter for size in kB, something like -buffer2 / -buffer16 / -buffer64 / -buffer256. Any experienced user can run few tests first in order to benchmark his BD/HDD drive and will use the best option always later on. In case of missing -buffer? switch defaults will be used of course though they may perform worse on user's system. That's the beauty of CLI which should not be mangled by INI or REGISTRY writing whatsoever.

[73ChargerFan]

What is wrong with an ini file? He's already got a directory full of DLLs and a PlugIns directory.

[madshi]

Guys, you don't really need to discuss how I should implement something. :) I'll make up my own solution, anyway...

[kypec]

What is wrong with an ini file? He's already got a directory full of DLLs and a PlugIns directory.
1. All those DLLs and Plugins are to be READ ONLY
2. Update process is much easier -> just overwrite target files with the new ones and you're done, there's no need to keep INI file and watch for its compatibility across future versions of eac3to
3. Implementation of INI parser into existing source code means much more work than adding one more CLI switch
4. Unless there is significant amount of configuration options needed for application then it's better to stay away from INI/registry altogether
5. GUI add-ons can easily adopt any new CLI option and keep it along with many other settings in their own INI files if necessary

[lchiu7]

Please try using "-libav" to use the libav decoder instead of Nero. Maybe that helps? How big is that E-AC3 track? Can you reproduce the problem with a small sample of the track? If so, it would be nice if you could upload it for me to test...
..p".

-libav works fine so it must be my Nero 7 installation. Since everybody seems to be having no problems I don't think it's a bug in eac3to. The eac3 file is 1.2G. Happy to upload it for testing if you can tell me what tool I could use to cut it down to size

[edit]

Just realised that it dies a fair way into the eac3 file so would be hard to provide a piece to test. Since a member on another BBS ripped the same title with no problems and -libav works, I am going to put this done to a bad Nero 7 installation rather than any inherent problem with eac3to (which is a great tool BTW). Thanks for looking

[bigdog660]

Hello Madshi! Finally, here are my test results for all three versions.

Source: Hellboy (2:12:29)

Rip from HDD Folder C: to HDD D: using v2.87

eac3to v2.87
command line: "D:\HDDVDTL\eac3to.exe" "C:\TEMP\HELLBOY\BDMV\STREAM\00090.m2ts" 2: "D:\HELLBOY\feature.mkv" 1: "D:\HELLBOY\chapters.txt" 4: "D:\HELLBOY\feature.ac3"
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 7 subtitle tracks, 2:12:29, 24p /1.001
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, 5.1 channels, 16 bits, 48khz
4: AC3, 5.1 channels, 640kbps, 48khz
5: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -29dB
6: AC3 Surround, 2.0 channels, 192kbps, 48khz, dialnorm: -26dB
7: Subtitle (PGS)
8: Subtitle (PGS)
9: Subtitle (PGS)
10: Subtitle (PGS)
11: Subtitle (PGS)
12: Subtitle (PGS)
13: Subtitle (PGS)
Creating file "D:\HELLBOY\chapters.txt"...
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[v02] Muxing video to Matroska...
[a04] Creating file "D:\HELLBOY\feature.ac3"...
Added fps value to MKV header.
Video track 2 contains 190583 frames.
eac3to processing took 39 minutes, 18 seconds.
Done.

Rip from Virtual Drive K: to HDD D: using v2.87

eac3to v2.87
command line: "D:\HDDVDTL\eac3to.exe" "K:\" 1) 2: "D:\HELLBOY\feature.mkv" 1: "D:\HELLBOY\chapters.txt" 4: "D:\HELLBOY\feature.ac3"
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 7 subtitle tracks, 2:12:29
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, English, 5.1 channels, 16 bits, 48khz
4: AC3, English, 5.1 channels, 640kbps, 48khz
5: AC3, Thai, 5.1 channels, 640kbps, 48khz, dialnorm: -29dB
6: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -26dB
7: Subtitle (PGS), English
8: Subtitle (PGS), English
9: Subtitle (PGS), Chinese
10: Subtitle (PGS), Chinese
11: Subtitle (PGS), Portuguese
12: Subtitle (PGS), Spanish
13: Subtitle (PGS), Thai
Creating file "D:\HELLBOY\chapters.txt"...
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[v02] Muxing video to Matroska...
[a04] Creating file "D:\HELLBOY\feature.ac3"...
Added fps value to MKV header.
Video track 2 contains 190583 frames.
eac3to processing took 37 minutes, 13 seconds.
Done.


Rip from BD Drive E: to HDD D: using v2.87

eac3to v2.87
command line: "D:\HDDVDTL\eac3to.exe" "E:\" 1) 2: "D:\HELLBOY\New Folder\feature.mkv" 1: "D:\HELLBOY\New Folder\chapters.txt" 4: "D:\HELLBOY\New Folder\feature.ac3"
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 7 subtitle tracks, 2:12:29
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, English, 5.1 channels, 16 bits, 48khz
4: AC3, English, 5.1 channels, 640kbps, 48khz
5: AC3, Thai, 5.1 channels, 640kbps, 48khz, dialnorm: -29dB
6: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -26dB
7: Subtitle (PGS), English
8: Subtitle (PGS), English
9: Subtitle (PGS), Chinese
10: Subtitle (PGS), Chinese
11: Subtitle (PGS), Portuguese
12: Subtitle (PGS), Spanish
13: Subtitle (PGS), Thai
Creating file "D:\HELLBOY\New Folder\chapters.txt"...
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[v02] Muxing video to Matroska...
[a04] Creating file "D:\HELLBOY\New Folder\feature.ac3"...
Added fps value to MKV header.
Video track 2 contains 190583 frames.
eac3to processing took 58 minutes, 33 seconds.
Done.

Rip from HDD Folder C: to HDD D: using v3.04

eac3to v3.04
command line: "D:\HDDVDTL\eac3to.exe" "C:\TEMP\HELLBOY\BDMV\STREAM\00090.m2ts" 2: "D:\HELLBOY\feature.mkv" 1: "D:\HELLBOY\chapters.txt" 4: "D:\HELLBOY\feature.ac3"
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 7 subtitle tracks, 2:12:29, 24p /1.001
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, 5.1 channels, 16 bits, 48khz
4: AC3, 5.1 channels, 640kbps, 48khz
5: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -29dB
6: AC3 Surround, 2.0 channels, 192kbps, 48khz, dialnorm: -26dB
7: Subtitle (PGS)
8: Subtitle (PGS)
9: Subtitle (PGS)
10: Subtitle (PGS)
11: Subtitle (PGS)
12: Subtitle (PGS)
13: Subtitle (PGS)
Creating file "D:\HELLBOY\chapters.txt"...
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[v02] Muxing video to Matroska...
[a04] Creating file "D:\HELLBOY\feature.ac3"...
Added fps value to MKV header.
Video track 2 contains 190583 frames.
eac3to processing took 22 minutes, 52 seconds.
Done.

Rip from Virtual Drive K: to HDD D: using v3.04

eac3to v3.04
command line: "D:\HDDVDTL\eac3to.exe" "K:\" 1) 2: "D:\HELLBOY\feature.mkv" 1: "D:\HELLBOY\chapters.txt" 4: "D:\HELLBOY\feature.ac3"
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 7 subtitle tracks, 2:12:29
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, English, 5.1 channels, 16 bits, 48khz
4: AC3, English, 5.1 channels, 640kbps, 48khz
5: AC3, Thai, 5.1 channels, 640kbps, 48khz, dialnorm: -29dB
6: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -26dB
7: Subtitle (PGS), English
8: Subtitle (PGS), English
9: Subtitle (PGS), Chinese
10: Subtitle (PGS), Chinese
11: Subtitle (PGS), Portuguese
12: Subtitle (PGS), Spanish
13: Subtitle (PGS), Thai
Creating file "D:\HELLBOY\chapters.txt"...
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[v02] Muxing video to Matroska...
[a04] Creating file "D:\HELLBOY\feature.ac3"...
Added fps value to MKV header.
Video track 2 contains 190583 frames.
eac3to processing took 20 minutes, 56 seconds.
Done.


Rip from BD Drive E: to HDD D: using v3.04

eac3to v3.04
command line: "D:\HDDVDTL\eac3to.exe" "E:\" 1) 2: "D:\HELLBOY\New Folder\feature.mkv" 1: "D:\HELLBOY\New Folder\chapters.txt" 4: "D:\HELLBOY\New Folder\feature.ac3"
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 7 subtitle tracks, 2:12:29
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, English, 5.1 channels, 16 bits, 48khz
4: AC3, English, 5.1 channels, 640kbps, 48khz
5: AC3, Thai, 5.1 channels, 640kbps, 48khz, dialnorm: -29dB
6: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -26dB
7: Subtitle (PGS), English
8: Subtitle (PGS), English
9: Subtitle (PGS), Chinese
10: Subtitle (PGS), Chinese
11: Subtitle (PGS), Portuguese
12: Subtitle (PGS), Spanish
13: Subtitle (PGS), Thai
Creating file "D:\HELLBOY\New Folder\chapters.txt"...
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[v02] Muxing video to Matroska...
[a04] Creating file "D:\HELLBOY\New Folder\feature.ac3"...
Added fps value to MKV header.
Video track 2 contains 190583 frames.
eac3to processing took 4 hours, 15 minutes.
Done.

Rip from HDD Folder C: to HDD D: using v3.05

eac3to v3.05
command line: "D:\HDDVDTL\eac3to.exe" "C:\TEMP\HELLBOY\BDMV\STREAM\00090.m2ts" 2: "D:\HELLBOY\feature.mkv" 1: "D:\HELLBOY\chapters.txt" 4: "D:\HELLBOY\feature.ac3"
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 7 subtitle tracks, 2:12:29, 24p /1.001
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, 5.1 channels, 16 bits, 48khz
4: AC3, 5.1 channels, 640kbps, 48khz
5: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -29dB
6: AC3 Surround, 2.0 channels, 192kbps, 48khz, dialnorm: -26dB
7: Subtitle (PGS)
8: Subtitle (PGS)
9: Subtitle (PGS)
10: Subtitle (PGS)
11: Subtitle (PGS)
12: Subtitle (PGS)
13: Subtitle (PGS)
Creating file "D:\HELLBOY\chapters.txt"...
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[v02] Muxing video to Matroska...
[a04] Creating file "D:\HELLBOY\feature.ac3"...
Added fps value to MKV header.
Video track 2 contains 190583 frames.
eac3to processing took 24 minutes, 46 seconds.
Done.

Rip from Virtual Drive K: to HDD D: using v3.05

eac3to v3.05
command line: "D:\HDDVDTL\eac3to.exe" "K:\" 1) 2: "D:\HELLBOY\feature.mkv" 1: "D:\HELLBOY\chapters.txt" 4: "D:\HELLBOY\feature.ac3"
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 7 subtitle tracks, 2:12:29
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, English, 5.1 channels, 16 bits, 48khz
4: AC3, English, 5.1 channels, 640kbps, 48khz
5: AC3, Thai, 5.1 channels, 640kbps, 48khz, dialnorm: -29dB
6: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -26dB
7: Subtitle (PGS), English
8: Subtitle (PGS), English
9: Subtitle (PGS), Chinese
10: Subtitle (PGS), Chinese
11: Subtitle (PGS), Portuguese
12: Subtitle (PGS), Spanish
13: Subtitle (PGS), Thai
Creating file "D:\HELLBOY\chapters.txt"...
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[v02] Muxing video to Matroska...
[a04] Creating file "D:\HELLBOY\feature.ac3"...
Added fps value to MKV header.
Video track 2 contains 190583 frames.
eac3to processing took 50 minutes, 17 seconds.
Done.

Rip from BD Drive E: to HDD D: using v3.05

eac3to v3.05
command line: "D:\HDDVDTL\eac3to.exe" "E:\" 1) 2: "D:\HELLBOY\New Folder\feature.mkv" 1: "D:\HELLBOY\New Folder\chapters.txt" 4: "D:\HELLBOY\New Folder\feature.ac3"
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 7 subtitle tracks, 2:12:29
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, English, 5.1 channels, 16 bits, 48khz
4: AC3, English, 5.1 channels, 640kbps, 48khz
5: AC3, Thai, 5.1 channels, 640kbps, 48khz, dialnorm: -29dB
6: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -26dB
7: Subtitle (PGS), English
8: Subtitle (PGS), English
9: Subtitle (PGS), Chinese
10: Subtitle (PGS), Chinese
11: Subtitle (PGS), Portuguese
12: Subtitle (PGS), Spanish
13: Subtitle (PGS), Thai
Creating file "D:\HELLBOY\New Folder\chapters.txt"...
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[v02] Muxing video to Matroska...
[a04] Creating file "D:\HELLBOY\New Folder\feature.ac3"...
Added fps value to MKV header.
Video track 2 contains 190583 frames.
eac3to processing took 1 hour, 51 minutes.
Done.

I am still running the test on v3.04, but I can tell you right now it's results are going to be the slowest of the 3 by far.:devil:

As you can see, 2.87 beats 3.04 and 3.05 in every way. The computer was not under load during any of these tests, and you can see the command is the same for each test.

As for the controversy to add an INI file, add setting in the registry, or add an option for the command line, my vote is for the INI file. Basically Madshi could provide guide lines to give us an idea of the value to put in the INI file, and it would be up to us users to tweak the value for our system. This also eliminates the need for ac3to to write to it's directory (for those worried about that). Registry setting are more difficult to change and there's always the odd change of a less experienced user messing his/her registry up. A command line option could work, but you have to remember to set it each time, and then those of us who use GUI's will have to wait for the option to be supported. I still use yr_eac3to_more and it hasn't been updated for ages. Anyway, that's my 2 cents on that subject.

Madshi, as always, I appreciate everything you do.:thanks:

Edit: Folder rip shows 3.04/3.05 does better than 2.87, but BD drive performance with 3.04/3.05 needs some work.

[madshi]

@bigdog660, thanks for testing! You're doing:

(1) direct rip from BD drive.
(2) rip from virtual BD.

The one thing I'm really interested in is:

(3) rip from folder structure on harddisk.

I guess (3) might be more or less similar to (2), but I'm not sure. Probably (3) is the fastest of all options.

Edit: Actually (3) should be noticably different compared to (2) because eac3to v3.05 will see the virtual BD drive as a physical drive and thus use 2KB read blocks, while for (3) it will use 1MB read blocks instead.

[bigdog660]

The one thing I'm really interested in is:

(3) rip from folder structure on harddisk.

Ah, you want "eac3to 00090.m2ts 1: chapters.txt 2: movie.mkv 4: movie.ac3"?

I will add those tests to my previous post for all three versions.

[DrNein]

A problem applying large delay values to AC3 (at least) was introduced with 3.04 and remains with 3.05 (3.03 is okay). For example, if applying a value of +9472ms, eac3to might report "A remaining delay of +11712ms could not be fixed).

[madshi]

Thanks bigdog660,

so that means:

(1) v3.0x is much faster (up to 2x) than v2.87 when handling folders/files on harddisk.
(2) Using 2KB block reads slows up read access to both your physical and virtual drive compared to the 64K block reads used by v2.87.
(3) Using 1MB block reads extremely slows up read access to your physical drive, but noticably improves read access to your virtual drive, compared to v2.87.
(4) My physical drive shows much better performance with 2KB block reads than with 64K block reads.

Yippieh-yey...

[madshi]

A problem applying large delay values to AC3 (at least) was introduced with 3.04 and remains with 3.05 (3.03 is okay). For example, if applying a value of +9472ms, eac3to might report "A remaining delay of +11712ms could not be fixed).
Already found and fixed that problem earlier today. It's just a cosmetical issue. The delay was applied correctly, it's only the log output which is wrong.

[DrNein]

Already found and fixed that problem earlier today. It's just a cosmetical issue. The delay was applied correctly, it's only the log output which is wrong.

Coolsville :cool:

[mikelebron]

Any help on how I can fix this issue? The originating m2ts file works fine in players like TMT... I demuxed the m2ts file to see if this would fix the issue.. but as you can see when I try to convert it to flac it fails.. The original m2ts file was created by TSRemuxer..

eac3to v3.05
command line: eac3to audio1.thd+ac3 e:\audiott.flac
------------------------------------------------------------------------------
TrueHD/AC3, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 640kbps, 48khz)
Extracting TrueHD stream...
Decoding with libav/ffmpeg...
Encoding FLAC with libFlac...
Creating file "e:\audiott.flac"...
[libav] Substream 1 checksum failed <WARNING>
[libav] Lossless check failed - expected 16, calculated 9d <WARNING>
[libav] Substream 1 checksum failed <WARNING>
[libav] Lossless check failed - expected 14, calculated 94 <WARNING>
[libav] Substream 1 checksum failed <WARNING>
[libav] Invalid channel 3 specified as output from matrix <WARNING>
The libav decoder reported an error while decoding. <ERROR>
Aborted at file position 53215232. <ERROR>

[madshi]

Any help on how I can fix this issue? The originating m2ts file works fine in players like TMT... I demuxed the m2ts file to see if this would fix the issue.. but as you can see when I try to convert it to flac it fails.. The original m2ts file was created by TSRemuxer..

eac3to v3.05
command line: eac3to audio1.thd+ac3 e:\audiott.flac
------------------------------------------------------------------------------
TrueHD/AC3, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 640kbps, 48khz)
Extracting TrueHD stream...
Decoding with libav/ffmpeg...
Encoding FLAC with libFlac...
Creating file "e:\audiott.flac"...
[libav] Substream 1 checksum failed <WARNING>
[libav] Lossless check failed - expected 16, calculated 9d <WARNING>
[libav] Substream 1 checksum failed <WARNING>
[libav] Lossless check failed - expected 14, calculated 94 <WARNING>
[libav] Substream 1 checksum failed <WARNING>
[libav] Invalid channel 3 specified as output from matrix <WARNING>
The libav decoder reported an error while decoding. <ERROR>
Aborted at file position 53215232. <ERROR>
I'd guess that either tsMuxeR or TsRemux did some nasty things to the TrueHD track. I'd suggest reripping the original disc and decoding that with eac3to.

[madshi]

Hello Madshi! Finally, here are my test results for all three versions.
Could you please keep that Hellboy source, so that you can retest later with v3.06? That would be nice - thanks!

[bigdog660]

Could you please keep that Hellboy source, so that you can retest later with v3.06? That would be nice - thanks!

No problem... thanks!

[tebasuna51]

The clipping MP2 is uploading. I re-verified that both MPG -> MP2 -> AC3 and MPG -> AC3 both produce clipping, but MPG -> MP2 -> 32float WAV (with Foobar) > AC3 (with eac3to) works fine.

Here is a sample (12 sec) with the problem isolated (http://www.sendspace.com/file/5nm3yr), is decoded fine with Foobar2000, BeSweet, NicAudio and also ffmpeg, but using eac3to-libav there are a inversion at clipping points than cause audible cliks (see the png included)

I hope than help to fix the problem.

[honai]

I have both the US and German Blu-rays of WALL-E.

Surprisingly, eac3to shows a runtime of 1:38:25 (00095.mpls) for the US disc, but 1:32:43 (00119.mpls) for the German disc.

I have noticed that the US version consists basically of a single large M2TS, but the German version is split up into lots of smaller segments. eac3to also reports many overlaps for the German disc.

Since I don't believe that they cut 6 minutes of footage for the German market, is it possible that eac3to realizes the gaps incorrectly? (The resulting audio files for the German disc are in fact 1:33:35 long.)

[madshi]

I hope than help to fix the problem.
It will help, thanks!

I have both the US and German Blu-rays of WALL-E.

Surprisingly, eac3to shows a runtime of 1:38:25 (00095.mpls) for the US disc, but 1:32:43 (00119.mpls) for the German disc.
Strange. My "Englisch.flac" from the US disc is 1:38:11 long, so 1:32:43 is definitely too short. Does the eac3to title listing (which you get if you simply do "eac3to blurayDrive:") also list 1:32:43? If so, the overlap removal can not be responsible for the too short runtime, because the overlap removal is only post processing. Are there no other playlists with a bigger runtime on the German disc? Have you tried actually processing the disc? Maybe the runtime display is simply wrong and processing the disc will still work nevertheless?

Maybe you could upload the CLIPINF and PLAYLIST folders? Zipped it should be only a few KBs. Thanks!

P.S: What does playlist "00082.mpls" from the German Blu-Ray say? You can do "eac3to bluRayFolder\PLAYLIST\00082.mpls", if this playlist isn't offered by default.

[honai]

Thanks. Yes, I already checked out that possibility. For 00082.mpls it says 1:32:31 for the German disc. I have seen somewhere else that for this playlist the runtime is supposed to be 1:38:32, that's why I wondered if maybe eac3to is making a mistake.

I have also observed that for 00119.mpls (German) eac3to initially reported 1:32:43 runtime in the tracklisting, when selecting the title for demuxing it shows 1:33:33, but demuxed/processed runtime was 1:33:35, so it seems that those are guesses by eac3to anyway, right? I'm especially wondering about the discrepancy between the initial title listing and the listing when that title is selected.

And yes, 00119.mpls and 00120.mpls are the longest-running titles displayed.

Here's the requested archive:

http://www.sendspace.com/file/ufd3xb

Thanks for looking into this, much appreciated.

EDIT:

After verifying each file from the 00082.mpls I found that one of my M2TS was corrupt. Doh! Will try again, but I guess this fixes it. Sorry for wasting your time.

EDIT:

Could you please add a sanity check if one of the M2TS files referenced in the playlist has a size of 0 bytes or is missing altogether from HDD? I have noticed that in cases like above eac3to will silently ignore the missing/corrupt file. Thanks!

[Snowknight26]

eac3to doesn't seem to encode 32-bit FLACs, but works fine if I remove -down32, resulting in a 24-bit FLAC:

eac3to v3.05
command line: eac3to.exe "Z:\Encoding Tools\temp\audio\2fast2furious.eac3" "Z:\Encoding Tools\temp\audio\2fast2furious.flac" -down32
------------------------------------------------------------------------------
E-AC3, 5.1 channels, 1:47:36, 1536kbps, 48khz
The Nero decoder doesn't seem to work, will use libav instead.
Decoding with libav/ffmpeg...
Remapping channels...
Reducing depth from 64 to 32 bits...
The FLAC encoder received a non-supported data format.
Aborted at file position 262144.

Also, any reason why creating a silent AC3 (E-AC3?) frame would fail?
eac3to v3.05
command line: eac3to.exe "G:\hotfuzz" 1) 4: ..\temp\audio\hotfuzz.eac3
------------------------------------------------------------------------------
EVO, 1 video track, 6 audio tracks, 7 subtitle tracks, 2:00:52
4: E-AC3 EX, English, 5.1 channels, 1536kbps, 48khz, dialnorm: -27dB, 133ms
[a04] Extracting audio track number 4...
[a04] Removing E-AC3 dialog normalization...
[a04] Applying (E-)AC3 delay...
[a04] Creating silent AC3 frame failed. <WARNING>
[a04] Creating file "..\temp\audio\hotfuzz.eac3"...
Video track 3 contains 173880 frames.
eac3to processing took 3 minutes, 8 seconds.
Done.

[madshi]

Could you please add a sanity check if one of the M2TS files referenced in the playlist has a size of 0 bytes or is missing altogether from HDD? I have noticed that in cases like above eac3to will silently ignore the missing/corrupt file. Thanks!
Will add a warning...

eac3to doesn't seem to encode 32-bit FLACs, but works fine if I remove -down32, resulting in a 24-bit FLAC
FLAC does not support 32bit, highest bitdepth supported is 24bit. BTW, some experts believe that even 16bit is good enough, others disagree. But all experts agree that 24bit is more than enough. So there's no reason to go 32bit, anyway, unless you want to do further processing on the file...

Also, any reason why creating a silent AC3 (E-AC3?) frame would fail?
Silent frame creation only works for AC3 because Aften doesn't support E-AC3 encoding yet. It's a bug that eac3to even tries to encode a silent E-AC3 frame, but it's only a cosmetical problem.

[Snowknight26]

FLAC does not support 32bit, highest bitdepth supported is 24bit.

http://flac.sourceforge.net/faq.html#general__samples
FLAC supports linear PCM samples with a resolution between 4 and 32 bits per sample.

:\

So there's no reason to go 32bit, anyway, unless you want to do further processing on the file...

...or for Fun. I was just trying to see how big a 64-bit floating point FLAC file (from an EAC3 file) would be, but since it doesn't support floating point precision, I wen't for 32, then 24bit when that didn't work.

[Inspector.Gadget]

Hey Madshi, quick question:

I'm using eac3to through the HD-DVD/Blu-ray Streams Extractor to pull some audio off a concert Blu-ray. Currently, I'm muxing the audio (extracted as AC3, FLAC, and DTS) to MKV with MKVMerge, splitting the resulting .MKA file by chapter, and then demuxing the audio from each resulting .MKA segment. I'm doing this in order to split the audio tracks by song. Is there a faster way to do this, perhaps during extraction or transcoding? Thanks.

[madshi]

http://flac.sourceforge.net/faq.html#general__samples
Maybe the spec supports it, but I think the official encoder does not. See here:

http://www.hydrogenaudio.org/forums/lofiversion/index.php/t23613.html

It's possible that this post is outdated, though, not sure...

Is there a faster way to do this, perhaps during extraction or transcoding?
Right now it's not possible with eac3to, if that's what you're asking. Don't know what you can or can not do with other tools.

[Snowknight26]

Maybe the spec supports it, but I think the official encoder does not. See here:

http://www.hydrogenaudio.org/forums/lofiversion/index.php/t23613.html

It's possible that this post is outdated, though, not sure...

Oh well, not like it was an issue to begin with. :p

[Blue_MiSfit]

Hi Madshi,

Did you get a chance to peek at my problematic MP2? It's the one with clipping etc..

Thanks,
~MiSfit

[madshi]

Did you get a chance to peek at my problematic MP2? It's the one with clipping etc..
Yes. It decodes without clipping when using "-nero" or "-sonic", so the issue is related to libav decoding. I'm checked and libav returns the data to me with clipping in the data and I don't really see how I can detect and remove that. I think I'll have to update to the latest libav SVN version and maybe do some patches to get floating point from libav instead of integer samples. Then I might get it solved. However, this is not likely to happen this week. Maybe next. For now you can use Nero or Sonic for decoding...

[Snowknight26]

Madshi, if you are thinking about doing that (the patch for floating point support), you might wish to submit your changes to the SVN as well, for everyone's benefit.

[GZZ]

I think I found a small cosmetic error when parsing a playlist file:

Result:


d:\eac3to>eac3to "H:\BDMV\PLAYLIST\00000.mpls"
1) 00000.mpls, 0:50:10
[4+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+
5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+
5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5+5].m2ts
- h264/AVC, 1080p24 /1.001 (16:9)
- RAW/PCM, English, multi-channel, 48khz


Not sure it should say 5+5+5+5+5+5...... it should just be [4+5].m2ts

I have attached the playlist file here, for you to test it.

http://hbar.dk/files/00000.rar

[madshi]

I think I found a small cosmetic error when parsing a playlist file:

Not sure it should say 5+5+5+5+5+5...... it should just be [4+5].m2ts
Some playlists are really that stupid. I don't think it's a bug.

[madshi]

eac3to v3.06 released

http://madshi.net/eac3to.zip

* added MKV reading/parsing support
* added demux support for MKV (E-)AC3, DTS(-HD), AAC, MPx, FLAC and WAV tracks
* added demux support for MKV "modern style" MPEG2, VC-1 and h264/AVC tracks
* reading from (HD) DVD and Blu-Ray drives uses different reading APIs now
* empty tracks in TS/m2ts container are not listed, anymore
* for 24.000 fps video tracks a little warning is displayed now
* when demuxing subtitle files, the number of captions is added to the filename
* timestamp derived FPS is used for gap checking instead of video bitstream FPS
* fixed: 44.1khz AC3 encoding was still broken
* fixed: zero byte stripping pass was done for true 24bit TrueHD tracks
* fixed: downconverting WAV files with 0x3f channel mask didn't work
* fixed: log output "remaining delay [...]" was sometimes wrong for AC3 tracks
* fixed: silent frame creation was tried for E-AC3 although it can't work
Please note that MKV read support is not complete yet. E.g. subtitles and chapters can't be demuxed yet. Also some MKV files muxed with old versions of mkvtoolnix, gdsmux or other muxing tools may not work correctly. If you run into any trouble with audio/video tracks, just let me know.

Good news is that MKV video muxing and demuxing is now a transparent operation, if you use eac3to both for muxing + demuxing. In other words: If you mux a video track with eac3to and then demux it again, you'll get the exact same data back. No changes, nothing missing... *)

-------

*) exceptions:

(1) of course eac3to does some video bitstream manipulations like cropping 1088 to 1080, removing padding bytes etc, but these changes are done intentionally and are not caused by muxing/demuxing
(2) VC-1 muxing loses the very last video frame, don't know why

[d1g1ta7]

On the NBC HDTV muxes used to distribute programming to network affiliates in the US, NBC sends the surround audio via 3 stereo tracks - the first contains front left and front right audio, the second contains the center and lfe, the third contains rear left and rear right. Would it be possible to combine these tracks in eac3to, outputting as 5.1?

[Snowknight26]

C:\unzipped\eac3to>eac3to.exe "Z:\Movies\1408\1408.mkv"
MKV, 1 video track, 1 audio track, 16 subtitle tracks, 1:52:26, 24p /1.001
1: V_MPEG4/ISO/AVC, English, 1920x800p (12:5)
2: DTS, English, 5.1 channels, 16 bits, 1509kbps, 48khz
3: S_TEXT/ASS, Finnish, "Finnish"
4: S_TEXT/ASS, Finnish, "Finnish"
5: S_TEXT/ASS, Finnish, "Finnish"
6: S_TEXT/ASS, Finnish, "Finnish"
7: S_TEXT/ASS, Finnish, "Finnish"
8: S_TEXT/ASS, Finnish, "Finnish"
9: S_TEXT/ASS, Finnish, "Finnish"
10: S_TEXT/ASS, Finnish, "Finnish"
11: S_TEXT/ASS, Finnish, "Finnish"
12: S_TEXT/ASS, Finnish, "Finnish"
13: S_TEXT/ASS, Finnish, "Finnish"
14: S_TEXT/ASS, Finnish, "Finnish"
15: S_TEXT/ASS, Finnish, "Finnish"
16: S_TEXT/ASS, Finnish, "Finnish"
17: S_TEXT/ASS, Finnish, "Finnish"
18: S_TEXT/ASS, Finnish, "Finnish"
Only the last track is Finnish.

And then funny enough, the next mkv I had muxed says
C:\unzipped\eac3to>eac3to.exe "Z:\Movies\11-14\11-14.mkv"
MKV, 1 video track, 1 audio track, 1:25:43, 23.975p
1: V_MPEG4/ISO/AVC, English, 1080p (16:9)
2: DTS, English, 5.1 channels, 24 bits, 1509kbps, 48khz

And finally..
C:\unzipped\eac3to>eac3to.exe "Z:\Movies\The Third Man\the third man.mkv"
MKV, 1 video track, 4 audio tracks, 1:45:13, 24p /1.001
1: V_MPEG4/ISO/AVC, English, 1436x1080p (359:270)
2: FLAC, English, 1.0 channels, 1:45:13, 16 bits, 48khz
"FLAC 1.0 @ 202kbps"
3: A_VORBIS, English, 1.0 channels, 48khz
"Abridged recording of Graham GreeneÆs treatment, read by actor Richard Clarke"
4: A_VORBIS, English, 1.0 channels, 48khz
"Abridged recording of Graham GreeneÆs treatment, read by actor Richard Clarke"
5: A_VORBIS, English, 1.0 channels, 48khz
"Abridged recording of Graham GreeneÆs treatment, read by actor Richard Clarke"
Only the last track has that description. And that Æ is really a ´. Guess eac3to doesn't support unicode?

Samples can be provided if needed.

[rebkell]

timestamp derived FPS is used for gap checking instead of video bitstream FPS


Could you explain that one a little bit? I have captures that lose audio and it skips PTS and DTS timestamps, before correcting itself and the gap correction in eac3to didn't handle it correctly(which was totally understandable), just wondering how this change will work with those files.

[madshi]

On the NBC HDTV muxes used to distribute programming to network affiliates in the US, NBC sends the surround audio via 3 stereo tracks - the first contains front left and front right audio, the second contains the center and lfe, the third contains rear left and rear right. Would it be possible to combine these tracks in eac3to, outputting as 5.1?
Your wish to combine those 3 tracks to one would be hard to express via command line. So I think the best solution would be to demux all 3 tracks to "WAVs", that should give you 6 mono WAV files. Then you can combine those back together to one multichannel file. Currently eac3to can not combine multiple mono WAV files into one multichannel WAV, but there are other tools that can do that (e.g. wavewizard).

Only the last track is Finnish.

And then funny enough, the next mkv I had muxed says

Only the last track has that description.

Samples can be provided if needed.
Yes, please. Samples would be appreciated.

And that Æ is really a ´. Guess eac3to doesn't support unicode?
The command line itself doesn't support unicode, IIRC. eac3to does for file names, but not for some other things...

timestamp derived FPS is used for gap checking instead of video bitstream FPS


Could you explain that one a little bit? I have captures that lose audio and it skips PTS and DTS timestamps, before correcting itself and the gap correction in eac3to didn't handle it correctly(which was totally understandable), just wondering how this change will work with those files.
If there is a file which has different timestamps than the video bitstream framerate information suggests, older eac3to versions reported gaps/overlaps which weren't there. This shouldn't happen after this change, anymore. But I've no idea if that helps with the specific captures you have.

[Snowknight26]

Yes, please. Samples would be appreciated.
http://www.stfcc.org/misc/eac3to/11-14.mkv
http://www.stfcc.org/misc/eac3to/1408.mkv
http://www.stfcc.org/misc/eac3to/thirdman.mkv

The command line itself doesn't support unicode, IIRC. eac3to does for file names, but not for some other things...
It does when you do cmd /u.

Another issue(?):
G:\Encoding Tools\eac3to>eac3to.exe 24.s01e01.mkv
MKV, 1 video track, 1 audio track, 0:42:44, 24p /1.001
1: h264/AVC, English, 692x472 23.976p
2: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
v01 The video track has a non-standard framerate.
http://www.stfcc.org/misc/24.s01e01.mkv

[Jeff Flowerday]

Madshi:

Thanks for the mkv support, very much appreciated!

I have a wish list item:

With more and more concerts footage coming out on Blu-Ray, it would be real sweet if you could add a split by chapter option when converting the lossless audio to flac files. Of course it would only work when reading an actual Blu-Ray structure.

[madshi]

http://www.stfcc.org/misc/eac3to/11-14.mkv
http://www.stfcc.org/misc/eac3to/1408.mkv
http://www.stfcc.org/misc/eac3to/thirdman.mkv
Thanks. "23.975" was due to truncating instead of rounding. It was "23.9758...". The next build will round that to 23.976. All tracks having the same "description" will also be fixed in the next build. A bigger problem is that eac3to doesn't like the h264/AVC tracks in your MKVs. This is caused by the video stream itself not containing any sequence headers. Because of that eac3to just lists the MKV properties of the video track instead of parsing the video bitstream itself. You can see that by the tracks being listed as "V_MPEG4/ISO/AVC". All video and audio tracks beginning with "V_" or "A_" are Matroska names. If tracks are listed like that, eac3to was not able to fully understand/parse the video/audio bitstream. Will have to work on that for next week. That's probably the problem with MKV: There are so many "funny" files out there which are different from what eac3to itself creates. So it might take a while until all files are properly supported.

BTW, if anybody is interested, you can use the undocumented switch "-logmkv" to get a tree structure of the MKV file. It's quite similar to what "mkvinfo" outputs. That way you can see whether eac3to read your MKV file correctly...

Another issue(?)
The video in that MKV is encoded with 23.976/1.000fps. I consider that "non-standard". The correct framerate would be 24.000/1.001fps. That's why eac3to (correctly) posts a warning. You can fix the problem by using "eac3to 24.s01e01.mkv fixed.mkv -slowdown". That will patch the video bitstream from 23.976/1.000 to 24.000/1.001. I don't think you will notice a difference when watching, though. Would be just for "peace of mind".

I have a wish list item:

With more and more concerts footage coming out on Blu-Ray, it would be real sweet if you could add a split by chapter option when converting the lossless audio to flac files. Of course it would only work when reading an actual Blu-Ray structure.
Don't know, that's more difficult than it might seem. The eac3to infrastructure wasn't really built to split output files. Just imagine that eac3to finds out that the FLAC files need a second pass: eac3to would have to enumerate through all chapters again, reopen every of those separate FLAC files and redo them! That's a logistical nightmare...

Can't foobar2000 split FLAC files into chapters somehow? I thought I had seen something like that...

[Snowknight26]

A bigger problem is that eac3to doesn't like the h264/AVC tracks in your MKVs. This is caused by the video stream itself not containing any sequence headers. Because of that eac3to just lists the MKV properties of the video track instead of parsing the video bitstream itself. You can see that by the tracks being listed as "V_MPEG4/ISO/AVC". All video and audio tracks beginning with "V_" or "A_" are Matroska names. If tracks are listed like that, eac3to was not able to fully understand/parse the video/audio bitstream.

Strange. The only thing that they share in common is that they were all muxed with mkvmerge.

The video in that MKV is encoded with 23.976/1.000fps. I consider that "non-standard". The correct framerate would be 24.000/1.001fps. That's why eac3to (correctly) posts a warning. You can fix the problem by using "eac3to 24.s01e01.mkv fixed.mkv -slowdown". That will patch the video bitstream from 23.976/1.000 to 24.000/1.001.
Is that all it does? It won't keep the audio muxed into the mkv?

[Thunderbolt8]

thanks again for new, mkv demuxing & getting back the full video data is really useful!

[yesgrey]

eac3to v3.06 released
* added demux support for MKV "modern style" MPEG2, VC-1 and h264/AVC tracks

Thank you very much! It's great having mkv support!
:thanks:
I suppose eac3to adds the Aud's when demuxing h264 streams, right?

[nautilus7]

Thanks for the update. New features are always welcome.

[mikelebron]

I have a quad core.. is eac3to optimized for a quad core? Are there any settings that I could benefit from? Also want to say great program.. I appreciate your efforts!

[shambles]

awesome update, thankyouverymuch! :)

[mini-moose]

having an issue I think I read many have but I couldn't find any suggested solution other than installing nero (I dislike any nero after v6) :

eac3to v3.06
command line: eac3to.exe F:\00005.m2ts 2: C:\0005.ac3 -640
------------------------------------------------------------------------------
TrueHD/AC3, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB)
Extracting TrueHD stream...
Removing TrueHD dialog normalization...
Decoding with libav/ffmpeg...
Encoding AC3 <640kbps> with libAften...
Creating file "C:\0005.ac3"...
[libav] Lossless check failed - expected 2c, calculated 43 <WARNING>
[libav] Lossless check failed - expected 90, calculated a4 <WARNING>
[libav] End of stream indicated <WARNING>
The original audio track has a constant bit depth of 16 bits.
eac3to processing took 3 minutes, 51 seconds.
Done.

thanks

[rcjc]

Like with mini-moose's TrueHD track with an embedded 448kbps 5.1 AC3 track, is there any benefit in encoding that TrueHD track to 640kbps AC3 or 1536or768 DTS versus just extracting and using the embedded 448 AC3?

[Blue_MiSfit]

Madshi,

Thanks so much for looking into the clipping problem! It makes sense that it's a libavcodec problem. I look forward to an update :)

I can't adequately express how awesome this program is, and how even more awesome the quality of support and frequency up updates are!!

Thank you!!

-MiSfit

[yonta]

MKV, 1 video track, 1 audio track, 1 subtitle track, 0:00:11, 60i /1.001
"SPORTSILLUSTRAT", TNT in HD (TNTH)
1: MPEG2, English, 1080i60 /1.001 (16:9)
"MPEG2 1080i"
2: AC3, English, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
"DD5.1 448k"
3: Subtitle (SRT)

eac3to says there's a SRT subtitle in this file when there's no subtitle.
http://www.sendspace.com/file/nq923t



MKV, 1 video track, 1:30:18, 23.999p
1: h264/AVC, 1080p24 (16:9)
[v01] The video bitstream framerate field doesn't match the container framerate. <WARNING>
[v01] The video framerate is correct, but rather unusual. <WARNING>

This is the log on an exact 24.000fps file.
http://www.sendspace.com/file/p6c0ta

[madshi]

Strange. The only thing that they share in common is that they were all muxed with mkvmerge.
Which mkvmerge version? Older versions removed the sequence headers from the stream. Don't know if the latest version still does that.

Is that all it does? It won't keep the audio muxed into the mkv?
It will modify the video bitstream and mux it into a new MKV file. The audio track will be demuxed, you'll have to add it back into the new MKV file with mkvmerge.

I suppose eac3to adds the Aud's when demuxing h264 streams, right?
Yes. And I double checked to make sure that the AUDs which are added back in are 100% identical to what they were in the original file. The AUDs are not stored in the MKV file. But the h264 specification says clearly enough how AUDs have to look like, so I was able to reproduce them perfectly (at least with the h264 file I was testing with).

I have a quad core.. is eac3to optimized for a quad core? Are there any settings that I could benefit from?
eac3to uses multiple threads wherever it seems appropriate. If you do simple things using quad core doesn't help over using single core. But with CPU intensive tasks like e.g. resampling audio or decoding multiple audio tracks at the same time, eac3to uses multiple threads. So it should make use of your quad core quite well. Often the speed of your harddisk is the limiting factor, though...

having an issue I think I read many have but I couldn't find any suggested solution other than installing nero (I dislike any nero after v6) :

eac3to v3.06
command line: eac3to.exe F:\00005.m2ts 2: C:\0005.ac3 -640
------------------------------------------------------------------------------
TrueHD/AC3, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB)
Extracting TrueHD stream...
Removing TrueHD dialog normalization...
Decoding with libav/ffmpeg...
Encoding AC3 <640kbps> with libAften...
Creating file "C:\0005.ac3"...
[libav] Lossless check failed - expected 2c, calculated 43 <WARNING>
[libav] Lossless check failed - expected 90, calculated a4 <WARNING>
[libav] End of stream indicated <WARNING>
The original audio track has a constant bit depth of 16 bits.
eac3to processing took 3 minutes, 51 seconds.
Done.
It's not a problem, if it only occurs once or twice. Especially if it's directly before "end of stream". Every such warning means that maybe 0.8ms of your audio file might not be perfectly lossless. The other millions of milliseconds in the audio file are still lossless.

Like with mini-moose's TrueHD track with an embedded 448kbps 5.1 AC3 track, is there any benefit in encoding that TrueHD track to 640kbps AC3 or 1536or768 DTS versus just extracting and using the embedded 448 AC3?
I wouldn't use DTS 768kbps. Using 640kbps AC3 or 1536kbps DTS *may* produce slightly better results than the 448kbps AC3 core track. But I don't really know. It also depends on how good the encoders are you're using...

MKV, 1 video track, 1 audio track, 1 subtitle track, 0:00:11, 60i /1.001
"SPORTSILLUSTRAT", TNT in HD (TNTH)
1: MPEG2, English, 1080i60 /1.001 (16:9)
"MPEG2 1080i"
2: AC3, English, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
"DD5.1 448k"
3: Subtitle (SRT)

eac3to says there's a SRT subtitle in this file when there's no subtitle.
Actually there are closed captions contained in that MPEG2 video track, which eac3to can (theoretically) extract to a SRT file. So the track listing is correct. However, actually trying to extract the SRT doesn't seem to work in this particular case, so I'll have to look into that.

MKV, 1 video track, 1:30:18, 23.999p
1: h264/AVC, 1080p24 (16:9)
[v01] The video bitstream framerate field doesn't match the container framerate. <WARNING>
[v01] The video framerate is correct, but rather unusual. <WARNING>

This is the log on an exact 24.000fps file.
Will be fixed in the next build.

[GZZ]

I think I found a small cosmetic error when parsing a playlist file:

Not sure it should say 5+5+5+5+5+5...... it should just be [4+5].m2ts


Some playlists are really that stupid. I don't think it's a bug.

I notice that sometimes when trying to parse the associated stream files for the playlist can take longer time then a normal playlist that dosnt contain 5+5+5+5+5+5.... Is it because your program try to read all the files (the same file over and over again) until it has read it the number of times its shown. In this example it will be like 20 times. I think removing the console showing duplicated files from the list will be more readable for the user.

/GZZ

[ultratoto14]

Thanks madshi for the support of mkv as input.

[Guest]

A bigger problem is that eac3to doesn't like the h264/AVC tracks in your MKVs. This is caused by the video stream itself not containing any sequence headers. The SPS's and PPS's are stored in the CodecPrivate data. It's easy to extract them. When demuxing, you should extract them and inject them at the start of the demuxed stream.

Because of that eac3to just lists the MKV properties of the video track instead of parsing the video bitstream itself. You can see that by the tracks being listed as "V_MPEG4/ISO/AVC". All video and audio tracks beginning with "V_" or "A_" are Matroska names. If tracks are listed like that, eac3to was not able to fully understand/parse the video/audio bitstream. See above.

All the MKV files with AVC I've seen have "V_MPEG4/ISO/AVC" as the CodecID. Are you saying some files don't use that? What do they use?

[madshi]

The SPS's and PPS's are stored in the CodecPrivate data. It's easy to extract them. When demuxing, you should extract them and inject them at the start of the demuxed stream.
Yes, thanks. I'm planning to do that for the next release. But I'm thinking: Maybe it would be better to inject them not only at the start of the stream, but also in front of every I(DR) frame? Because I think in the original h264 stream normally sequence headers are stored in front of every I frame.

All the MKV files with AVC I've seen have "V_MPEG4/ISO/AVC" as the CodecID. Are you saying some files don't use that?
No no, you're right that MKV files usually use "V_MPEG4/ISO/AVC". I haven't seen any other codec ID than that. What I meant was that eac3to normally displays "h264/AVC", if it can properly parse such a video bitstream. The fact that eac3to displayed "V_MPEG4/ISO/AVC" means that it was not able to properly parse the bitstream. That's all I meant.

However, I do think that in theory it would be possible to have h264 MKV files which are not using "V_MPEG4/ISO/AVC", but "V_MS/VFW/FOURCC" instead, just as some VC-1 MKVs do, but with "avc1" in the CodecPrivate's BitmapInfoHeader. So eac3to also accepts "V_MS/VFW/FOURCC" with "avc1" as valid h264/AVC MKVs, too, just to be safe...

[Guest]

Yes, thanks. I'm planning to do that for the next release. But I'm thinking: Maybe it would be better to inject them not only at the start of the stream, but also in front of every I(DR) frame? Because I think in the original h264 stream normally sequence headers are stored in front of every I frame. It's not required because the activated SPS/PPS's remain in effect until a new one appears in the stream. But it would be a good idea to re-insert them if they are not present already, to allow for subsequent stream cutting, etc. You'll have to make a reliable way to detect IDRs, however.

However, I do think that in theory it would be possible to have h264 MKV files which are not using "V_MPEG4/ISO/AVC", but "V_MS/VFW/FOURCC" instead, just as some VC-1 MKVs do, but with "avc1" in the CodecPrivate's BitmapInfoHeader. So eac3to also accepts "V_MS/VFW/FOURCC" with "avc1" as valid h264/AVC MKVs, too, just to be safe... What is this "CodecPrivate's BitmapInfoHeader" you talk about? I've seen no documentation of such a thing and cannot find anything like that in the streams I have.

[Kurtnoise]

What is this "CodecPrivate's BitmapInfoHeader" you talk about? I've seen no documentation of such a thing and cannot find anything like that in the streams I have.
iirc, this comes from the vfw x264 files which uses AVI as container...

[madshi]

What is this "CodecPrivate's BitmapInfoHeader" you talk about? I've seen no documentation of such a thing and cannot find anything like that in the streams I have.
Which CodecID do you expect for VC-1 streams? I've seen both "V_VC1" and "V_MS/VFW/FOURCC". The latter CodecID can mean "VC-1", but it also can mean any other VfW codec. See here:

http://haali.net/mkv/codecs.pdf

That PDF shows that for the CodecID "V_MS/VFW/FOURCC" the CodecPrivate data contains a "BITMAPINFOHEADER" structure which is an official win32 structure. The field "dwCompression" of that structure contains the "FOURCC" which tells us which codec is actually contained in this track. You can find a list of valid FOURCC values here:

http://www.fourcc.org/
http://msdn.microsoft.com/de-de/library/ms867195(en-us).aspx#fourcccodes

Valid FOURCC values are e.g. "avc1", "H264", "X264" for h264 streams and "WVC1" for VC-1 streams. So if the MKV CodecID says "V_MS/VFW/FOURCC", we have to look at the FOURCC value stored in "((BITMAPINFOHEADER) CodecPrivate)->dwCompression".

[Jeff Flowerday]

Don't know, that's more difficult than it might seem. The eac3to infrastructure wasn't really built to split output files. Just imagine that eac3to finds out that the FLAC files need a second pass: eac3to would have to enumerate through all chapters again, reopen every of those separate FLAC files and redo them! That's a logistical nightmare...

Can't foobar2000 split FLAC files into chapters somehow? I thought I had seen something like that...

Understandable!

Yes foobar2000 can split based on cue sheet, it's just such a pain to manually create a cue sheet. The cue sheet isn't complex just the conversion of milliseconds to frames is a little time consuming.

Actually an option to output a cue sheet from eac3to would take all the pain away.

http://wiki.hydrogenaudio.org/index.php?title=Cuesheet

Performer, title and track names aren't really important since we need to tag the resultant flac anyway. "FEATURE" , "FEATURE ARTIST" and chapter numbers for track names would suffice.

If you don't see value in it, I'll right a program to convert your chapter files to cue sheets.

Thanks,
Jeff

[Guest]

Which CodecID do you expect for VC-1 streams? I've seen both "V_VC1" and "V_MS/VFW/FOURCC". I only have a few I made myself, so my experience isn't important yet.

The latter CodecID can mean "VC-1", but it also can mean any other VfW codec. See here:

http://haali.net/mkv/codecs.pdf

That PDF shows that for the CodecID "V_MS/VFW/FOURCC" the CodecPrivate data contains a "BITMAPINFOHEADER" structure which is an official win32 structure. The field "dwCompression" of that structure contains the "FOURCC" which tells us which codec is actually contained in this track. You can find a list of valid FOURCC values here: Ah! Thank you very much for the explanation.

[honai]

I'm trying to remux an MKV I created with an earlier version of eac3to which still contains pulldown information. This is the error eac3to throws:


eac3to v3.06
command line: eac3to "Ray 1080p VC-1 English.mkv" 1: video.mkv 2: english.dts
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 2:32:20, 24p /1.001
1: V_MS/VFW/FOURCC/WVC1, English, 1080p (16:9)
"Video 1080p VC-1"
2: DTS, English, 5.1 channels, 24 bits, 1510kbps, 48khz
"Audio English DTS 5.1 1536kbps"
[v01] Video tracks with unknown parameters can't be muxed to MKV. <ERROR>
[v01] Internal error - unknown video output format! <ERROR>


However, demuxing with -demux works and eac3to correctly identifies the video as VC-1.

Is this a bug or intentional misbehavior? :confused:

[pihug12]

eac3to v3.06
command line: eac3to ../test/fb-walle720.mkv -demux
------------------------------------------------------------------------------
MKV, 1 video track, 2 audio tracks, 2 subtitle tracks, 1:38:42, 23.975p
1: V_MPEG4/ISO/AVC, English, 1280x528p (80:33)
2: DTS-ES, English, 5.1 channels, 24 bits, 1509kbps, 48khz
"DTS"
3: AC3 EX, French, 5.1 channels, 640kbps, 48khz
"AC3 5.1"
4: S_TEXT/UTF8, French
5: S_TEXT/UTF8, French
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[a03] Extracting audio track number 3...
[v01] Creating file "..testfb-walle720 - 1 - V_MPEG4ISOAVC, English, 1280x528p.h264"...
[a02] Creating file "..testfb-walle720 - 2 - DTS-ES, English, 5.1 channels, 24 bits, 1509kbps, 48khz, 'DTS'.dts"...
[a03] Creating file "..testfb-walle720 - 3 - AC3 EX, French, 5.1 channels, 640kbps, 48khz, 'AC3 5.1'.ac3"...
eac3to processing took 6 minutes, 21 seconds.
Done.

The two subtitles files aren't demux. Bug ?

pihug12 :cool:

[Jeff Flowerday]

The two subtitles files aren't demux. Bug ?

pihug12 :cool:

http://forum.doom9.org/showthread.php?p=1244439#post1244439

[Snowknight26]

Madshi, though you say you plan on adding chapter and subtitle demuxing from mkvs, you didn't mention attachments. Any plans for those?

Also, doing anything with eac3to over a network share seems to be really slow. I'm remuxing an mkv and eac3to uses 0% CPU usage 90% of the time. It spikes up to about 10% for a fraction of a second... same with the network usage graph.

No no, you're right that MKV files usually use "V_MPEG4/ISO/AVC". I haven't seen any other codec ID than that. What I meant was that eac3to normally displays "h264/AVC", if it can properly parse such a video bitstream. The fact that eac3to displayed "V_MPEG4/ISO/AVC" means that it was not able to properly parse the bitstream. That's all I meant.

Wonder why most can't be parsed while a few odd ones (like this first one) can:
eac3to.exe "G:\Movies\Babylon A.D\Babylon.A.D.mkv"
MKV, 1 video track, 1 audio track, 1 subtitle track, 1:40:58, 24p /1.001
1: h264/AVC, English, 1920x820 24p /1.001
2: DTS, English, 5.1 channels, 24 bits, 1509kbps, 48khz

eac3to.exe G:\Movies\300\300.mkv
MKV, 1 video track, 1 audio track, 19 subtitle tracks, 1:56:33, 23.975p
1: V_MPEG4/ISO/AVC, English, 1920x800p (12:5)
2: DTS, English, 5.1 channels, 24 bits, 1510kbps, 48khz

On a different note, is patching the DTS to 24 bits necessary in this scenario?:
eac3to.exe "G:\Unsorted\Band of Brothers\BOB02" 3) 3: bandofbrothers.ep4.wav
EVO, 1 video track, 2 audio tracks, 2 subtitle tracks, 0:59:42
"MAIN_VIDEO_02"
1: Chapters, 12 chapters with names
2: h264/AVC, 1080p24 /1.001 (16:9) with pulldown flags
3: DTS Hi-Res, English, 5.1 channels, 16 bits, 2046kbps, 48khz
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48khz), -83ms
4: DTS Hi-Res, Japanese, 5.1 channels, 16 bits, 2046kbps, 48khz
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48khz), -83ms
5: Subtitle (VobSub), Japanese
6: Subtitle (VobSub), Japanese
a03 The ArcSoft and Sonic decoders don't seem to work, will use libav instead.
a03 The libav DTS decoder doesn't decode the full DTS-HD information.
a03 Extracting audio track number 3...
a03 Extracting DTS core...
a03 Patching bitdepth to 24 bits...
a03 Decoding with libav/ffmpeg...
a03 Remapping channels...
a03 Applying RAW/PCM delay...
a03 Reducing depth from 64 to 24 bits...
a03 Writing WAV...
a03 Creating file "bandofbrothers.ep4.wav"...
-------------------------------

And finally, a small observation. Analysing mkvs with FLAC audio tracks takes far longer than ones with AC3 or DTS. Weird how that happens.

Edit: I don't think there should be a dash between HD DVD in "(2) Nero HD-DVD / Blu-Ray plugin".

[tebasuna51]

...
On a different note, is patching the DTS to 24 bits necessary in this scenario?:
...
3: DTS Hi-Res, English, 5.1 channels, 16 bits, 2046kbps, 48khz
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48khz), -83ms
...
a03 The ArcSoft and Sonic decoders don't seem to work, will use libav instead.
a03 The libav DTS decoder doesn't decode the full DTS-HD information.
...
a03 Reducing depth from 64 to 24 bits...
...

Of course, only DTS MA have a exact bitdepth, DTS HR and standard DTS can't recover the exact bitdepth from the source then must be decoded to the precision equivalent to the internal data in frequency domain.

I suggest to madshi don't put the bitdepth of the source with standard DTS and DTS HR because this data is misunderstand (and irrelevant).

[Snowknight26]

What I was pointing out is that it doesn't matter if its being patched to 24 bits during conversion to WAV because its being decoded as 64 bit -> 24 bit WAV anyway.

[magic144]

Hi guys,
just wondering if there's any difference in this workflow (other than efficiency of passes/typing/processing involved) for a BD title with a VC-1 video stream...

Up until now, I've just been using:-

eac3to L: 1) -demux, followed by
eac3to video.vc1 video.mkv

to demux and then put/house the demuxed .vc1 'raw' output into a more useful .mkv container for graphedit to use for frameserving (via Haali Media Splitter, WMVideo Decoder DMO)

however, I'm curious to know if that would produce *exactly* the same video result (frame-for-frame) as, e.g. (assuming track 2 is the video in question):-
eac3to L: 1) 2: video.mkv

would the first 2-step method lose/discard any important information (frame-rate, etc) that the original container used to define the .vc1 stream?
I have read that a raw .vc1 stream is somewhat dependent on its container to fully specify it, so I'm worried that a 2-step process would be lossy in some way?

I do notice this in the output of the 2nd step (again, if using the initial 2-step process):-
VC-1, 1080p24 /1.001 (16:9)
Muxing video to Matroska...
Added fps value to MKV header.

...so the .vc1 file must have retained some of the original attribute information in some way? I'm just not knowledgeable enough to know if there's going to be any difference between the methods here!

cheers,
m

[ron spencer]

do I still need to use vc1conv when eac3to demuxes an HD DVD? I am converting some to bluray....I notice that the EAC3to dialog says "Removing VC1 pulldown". Does this mean I do not need to run vc1conv on the VC-1 video?

thanks

[Blue_MiSfit]

You don't need to. eac3to will remove the pulldown.

I'd suggest just ripping from the disc to an MKV, and then using that as a source for TSMuxeR.

~MiSfit

[odin24]

Is it possible that eac3to reports the wrong information in regards to 1080p or 1080i? I have a BD that says 1080i on the back... which is odd, however eac3to and tsMuxeR both report 1080p. I trust eac3to more than what's on the back cover.

[ron spencer]

thanks....If I have forced subs, can txmuxer set up a set of subtitles as forced?

[odin24]

thanks....If I have forced subs, can txmuxer set up a set of subtitles as forced?

Unfortunately no, but it can be done fairly easily. Follow these instructions (http://forum.doom9.org/showpost.php?p=1224654&postcount=152).

You need BDEdit (http://pel.hu/down/BDedit.exe) for this.

[Blue_MiSfit]

Here's another silly sample for you, Madshi (20mb cut of source AC3)

http://www.megaupload.com/?d=NQF6AX7M

This one is the 3.1ch 640kbps AC3 track from the BluRay release of Bubble. eac3to complains about encoding it down to a 2ch AC3:

J:\Bubble>eac3to Bubble.ac3 2ch.ac3 -down2 -192
AC3, 3.1 channels, 1:13:27, 640kbps, 48khz, dialnorm: -27dB
Disabling DRC for Nero (E-)AC3 decoding...
Removing AC3 dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
DirectShow reports 3.1 channels, 24 bits, 48khz
The AC3 encoder received a non-supported data format (pcm, 4, 24, -).
Aborted at file position 352593920.


How can I convert this bad boy down to 2ch, 192kbps Dolby Digital properly?

Thanks!

~MiSfit

[bigdog660]

Rip from BD Drive E: to HDD D: using v3.06

eac3to v3.06
command line: "D:\HDDVDTL\eac3to.exe" "E:\" 1) 2: "D:\HELLBOY\feature.mkv" 1: "D:\HELLBOY\chapters.txt" 4: "D:\HELLBOY\feature.ac3"
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 7 subtitle tracks, 2:12:29, 24p /1.001
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, English, 5.1 channels, 16 bits, 48khz
4: AC3, English, 5.1 channels, 640kbps, 48khz
5: AC3, Thai, 5.1 channels, 640kbps, 48khz, dialnorm: -29dB
6: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -26dB
7: Subtitle (PGS), English
8: Subtitle (PGS), English
9: Subtitle (PGS), Chinese
10: Subtitle (PGS), Chinese
11: Subtitle (PGS), Portuguese
12: Subtitle (PGS), Spanish
13: Subtitle (PGS), Thai
Creating file "D:\HELLBOY\chapters.txt"...
[a04] Extracting audio track number 4...
[v02] Extracting video track number 2...
[v02] Muxing video to Matroska...
[a04] Creating file "D:\HELLBOY\feature.ac3"...
Added fps value to MKV header.
Video track 2 contains 190583 frames.
eac3to processing took 36 minutes, 26 seconds.
Done.


Sweet! :D Ripping at about 3.64x on a 4x capable BD drive! Thanks for looking into this Madshi. This makes v3.06 the fastest by far.

I wonder if others have seen such an improvement?

:thanks: again!

[lchiu7]

I am having problems with eac3to finding and using the Nero 7 DD decoder. Initially -test would show it had found it but every time I would run a job to extract the DD track from a eac3 track, it would die in some random place in the file. I have had to use -libav to complete the processing.

Installed the latest update to Nero 7 7.11.10 and how when I run -test I receive the following message\

Nero Audio Decoder (Nero 7) is not working correctly
http://www.nero.com/eng/store-blu-ray.html

Nero works fine since I am able to use Nero Vision to playback a .ts file with h.264 video and DD 5.1 audio

Not entirely sure what to do now. If I continue to use -libav what sonic compromises, if any, am I having to make?

Thanks

[DrNein]

On this note, does anyone know if its possible with the ArcSoft DTS Decoder to do similar.

That is, install the full version but then remove all its filters except for the one that eacto needs. I don't mind getting ArcSoft TotalMedia Theatre but I fear all its filters will gunk up my other software like PowerDVD, ProShow Gold. Anyone have any experience with installing a Lite version or have a batch script that can unregister all the unneeded filters except for the one eacto needs?

Also curious, do people have scripts that load/unload filters based on tools that may need them? That way, they are loaded when needed and removed when task is completed.

I suppose that maybe eacto is coded to strictly build its graph manually to force its use of a specific filter. But I am not well versed in all this graph, DS Filters, codecs to know for sure. I just know that installing one program can hijack another programs use of its filters that it installed for a specific task. What a nightmare - DLL hell for codecs :eek:

Jom covered some of that here:
http://forum.doom9.org/showthread.php?p=1239786#post1239786

Also, see post #8037 and 8041
http://forum.doom9.org/showthread.php?p=1242555#post1242555

As for codec hell, that's what a good configurable player such as MPC-HC is for. ;)

Although others like PowerDVD will only use their own anyway.

[DrNein]

I am having problems with eac3to finding and using the Nero 7 DD decoder. Initially -test would show it had found it but every time I would run a job to extract the DD track from a eac3 track, it would die in some random place in the file. I have had to use -libav to complete the processing.

Installed the latest update to Nero 7 7.11.10 and how when I run -test I receive the following message\

Nero Audio Decoder (Nero 7) is not working correctly
http://www.nero.com/eng/store-blu-ray.html

Nero works fine since I am able to use Nero Vision to playback a .ts file with h.264 video and DD 5.1 audio

Not entirely sure what to do now. If I continue to use -libav what sonic compromises, if any, am I having to make?

Thanks

Also, see #7880 and on and particularly:
http://forum.doom9.org/showthread.php?p=1238209#post1238209

[tebasuna51]

This one is the 3.1ch 640kbps AC3 track from the BluRay release of Bubble. eac3to complains about encoding it down to a 2ch AC3:

How can I convert this bad boy down to 2ch, 192kbps Dolby Digital properly?
Select your preferred downmix
function Dmix3Stereo(clip a) # 3 Channels L,R,C
{
flr = GetChannel(a, 1, 2)
fcc = GetChannel(a, 3, 3)
return MixAudio(flr, fcc, 0.5858, 0.4142)
}
function Dmix4lStereo(clip a) # 4 Channels L,R,C + LFE
{
flr = GetChannel(a, 1, 2)
fcc = GetChannel(a, 3, 3)
lfe = GetChannel(a, 4, 4)
clf = MixAudio(fcc, lfe, 0.2929, 0.2929)
return MixAudio(flr, clf, 0.4142, 1.0)
}
NicAc3Source("Bubble.ac3")
Dmix3Stereo
#Dmix4lStereo
Normalize()
And encode with EncWAVtoAC3 v3.0b

[idbirch2]

I just demuxed Eight Below, purely to see the subtitle information and if the english SUPs contained any forced captions and there's a discrepency between the subtitle track numbers displayed at the beginning of the log and those used at the end to display forced caption info. Here's the log so you can see what I mean:

eac3to v3.06
command line: "C:\program files\audio\eac3to\eac3to.exe" "I:\0Stuff\Eight Below Blu-Ray" 1) -demux
------------------------------------------------------------------------------
M2TS, 1 video track, 10 audio tracks, 16 subtitle tracks, 2:00:24, 24p /1.001
1: Chapters, 15 chapters
2: MPEG2, 1080p24 /1.001 (16:9)
3: RAW/PCM, English, 5.1 channels, 16 bits, 48khz
4: AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
5: DTS, French, 5.1 channels, 24 bits, 1509kbps, 48khz, dialnorm: -4dB
6: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
7: DTS, Italian, 5.1 channels, 24 bits, 1509kbps, 48khz, dialnorm: -4dB
8: AC3, Italian, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
9: DTS, German, 5.1 channels, 24 bits, 1509kbps, 48khz, dialnorm: -4dB
10: AC3, German, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
11: DTS, Spanish, 5.1 channels, 24 bits, 1509kbps, 48khz, dialnorm: -4dB
12: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
13: Subtitle (PGS), English
14: Subtitle (PGS), English
15: Subtitle (PGS), French
16: Subtitle (PGS), Italian
17: Subtitle (PGS), German
18: Subtitle (PGS), Spanish
19: Subtitle (PGS), Dutch
20: Subtitle (PGS), Norwegian
21: Subtitle (PGS), Danish
22: Subtitle (PGS), Swedish
23: Subtitle (PGS), Finnish
24: Subtitle (PGS), Icelandic
25: Subtitle (PGS), French
26: Subtitle (PGS), Italian
27: Subtitle (PGS), German
28: Subtitle (PGS), Spanish
Creating file "00046 - Chapters.txt"...
[a11] Extracting audio track number 11...
[a07] Extracting audio track number 7...
[a09] Extracting audio track number 9...
[a10] Extracting audio track number 10...
[a10] Removing AC3 dialog normalization...
[s14] Extracting subtitle track number 14...
[s20] Extracting subtitle track number 20...
[a05] Extracting audio track number 5...
[s15] Extracting subtitle track number 15...
[s17] Extracting subtitle track number 17...
[a08] Extracting audio track number 8...
[s21] Extracting subtitle track number 21...
[s13] Extracting subtitle track number 13...
[s23] Extracting subtitle track number 23...
[s27] Extracting subtitle track number 27...
[s25] Extracting subtitle track number 25...
[a08] Removing AC3 dialog normalization...
[a07] Removing DTS dialog normalization...
[s16] Extracting subtitle track number 16...
[s19] Extracting subtitle track number 19...
[a03] Extracting audio track number 3...
[s22] Extracting subtitle track number 22...
[a11] Removing DTS dialog normalization...
[a09] Removing DTS dialog normalization...
[a06] Extracting audio track number 6...
[a06] Removing AC3 dialog normalization...
[s18] Extracting subtitle track number 18...
[s24] Extracting subtitle track number 24...
[a05] Removing DTS dialog normalization...
[a12] Extracting audio track number 12...
[a12] Removing AC3 dialog normalization...
[s28] Extracting subtitle track number 28...
[v02] Extracting video track number 2...
[a03] Reading RAW/PCM...
[s26] Extracting subtitle track number 26...
[a04] Extracting audio track number 4...
[a03] Swapping endian...
[a04] Removing AC3 dialog normalization...
[a03] Remapping channels...
[a03] Swapping endian...
[a03] Remapping channels...
[a03] Creating file "00046 - 3 - PCM, English, 5.1 channels, 16 bits, 48khz.pcm"...
[v02] Creating file "00046 - 2 - MPEG2, 1080p24.m2v"...
[a05] Creating file "00046 - 5 - DTS, French, 5.1 channels, 24 bits, 1509kbps, 48khz.dts"...
[a07] Creating file "00046 - 7 - DTS, Italian, 5.1 channels, 24 bits, 1509kbps, 48khz.dts"...
[a09] Creating file "00046 - 9 - DTS, German, 5.1 channels, 24 bits, 1509kbps, 48khz.dts"...
[a11] Creating file "00046 - 11 - DTS, Spanish, 5.1 channels, 24 bits, 1509kbps, 48khz.dts"...
[a04] Creating file "00046 - 4 - AC3, English, 5.1 channels, 640kbps, 48khz.ac3"...
[a06] Creating file "00046 - 6 - AC3, French, 5.1 channels, 640kbps, 48khz.ac3"...
[a08] Creating file "00046 - 8 - AC3, Italian, 5.1 channels, 640kbps, 48khz.ac3"...
[a10] Creating file "00046 - 10 - AC3, German, 5.1 channels, 640kbps, 48khz.ac3"...
[a12] Creating file "00046 - 12 - AC3, Spanish, 5.1 channels, 640kbps, 48khz.ac3"...
[s14] Creating file "00046 - 14 - Subtitle (PGS), English.sup"...
[s17] Creating file "00046 - 17 - Subtitle (PGS), German.sup"...
[s13] Creating file "00046 - 13 - Subtitle (PGS), English.sup"...
[s24] Creating file "00046 - 24 - Subtitle (PGS), Icelandic.sup"...
[s18] Creating file "00046 - 18 - Subtitle (PGS), Spanish.sup"...
[s23] Creating file "00046 - 23 - Subtitle (PGS), Finnish.sup"...
[s20] Creating file "00046 - 20 - Subtitle (PGS), Norwegian.sup"...
[s22] Creating file "00046 - 22 - Subtitle (PGS), Swedish.sup"...
[s21] Creating file "00046 - 21 - Subtitle (PGS), Danish.sup"...
[s16] Creating file "00046 - 16 - Subtitle (PGS), Italian.sup"...
[s15] Creating file "00046 - 15 - Subtitle (PGS), French.sup"...
[s19] Creating file "00046 - 19 - Subtitle (PGS), Dutch.sup"...
[a03] The original audio track has a constant bit depth of 16 bits.
[a04] Audio overlaps for 18ms at playtime 0:47:21. <WARNING>
[a04] Audio overlaps for 22ms at playtime 1:38:36. <WARNING>
[a05] Audio overlaps for 7ms at playtime 0:47:21. <WARNING>
[a06] Audio overlaps for 18ms at playtime 0:47:21. <WARNING>
[a06] Audio overlaps for 22ms at playtime 1:38:36. <WARNING>
[a07] Audio overlaps for 7ms at playtime 0:47:21. <WARNING>
[a08] Audio overlaps for 18ms at playtime 0:47:21. <WARNING>
[a08] Audio overlaps for 22ms at playtime 1:38:36. <WARNING>
[a09] Audio overlaps for 7ms at playtime 0:47:21. <WARNING>
[a10] Audio overlaps for 18ms at playtime 0:47:21. <WARNING>
[a10] Audio overlaps for 22ms at playtime 1:38:36. <WARNING>
[a11] Audio overlaps for 7ms at playtime 0:47:21. <WARNING>
[a12] Audio overlaps for 18ms at playtime 0:47:21. <WARNING>
[a12] Audio overlaps for 22ms at playtime 1:38:36. <WARNING>
[s25] Creating file "00046 - 25 - Subtitle (PGS), French.sup"...
[s26] Creating file "00046 - 26 - Subtitle (PGS), Italian.sup"...
[s27] Creating file "00046 - 27 - Subtitle (PGS), German.sup"...
[s28] Creating file "00046 - 28 - Subtitle (PGS), Spanish.sup"...
[a04] Starting 2nd pass...
[a04] Realizing (E-)AC3 gaps...
[a04] Creating file "00046 - 4 - AC3, English, 5.1 channels, 640kbps, 48khz.ac3"...
[a05] Starting 2nd pass...
[a05] Realizing DTS gaps...
[a05] Creating file "00046 - 5 - DTS, French, 5.1 channels, 24 bits, 1509kbps, 48khz.dts"...
[a06] Starting 2nd pass...
[a06] Realizing (E-)AC3 gaps...
[a06] Creating file "00046 - 6 - AC3, French, 5.1 channels, 640kbps, 48khz.ac3"...
[a07] Starting 2nd pass...
[a07] Realizing DTS gaps...
[a07] Creating file "00046 - 7 - DTS, Italian, 5.1 channels, 24 bits, 1509kbps, 48khz.dts"...
[a08] Starting 2nd pass...
[a08] Realizing (E-)AC3 gaps...
[a08] Creating file "00046 - 8 - AC3, Italian, 5.1 channels, 640kbps, 48khz.ac3"...
[a09] Starting 2nd pass...
[a09] Realizing DTS gaps...
[a09] Creating file "00046 - 9 - DTS, German, 5.1 channels, 24 bits, 1509kbps, 48khz.dts"...
[a10] Starting 2nd pass...
[a10] Realizing (E-)AC3 gaps...
[a10] Creating file "00046 - 10 - AC3, German, 5.1 channels, 640kbps, 48khz.ac3"...
[a11] Starting 2nd pass...
[a11] Realizing DTS gaps...
[a11] Creating file "00046 - 11 - DTS, Spanish, 5.1 channels, 24 bits, 1509kbps, 48khz.dts"...
[a12] Starting 2nd pass...
[a12] Realizing (E-)AC3 gaps...
[a12] Creating file "00046 - 12 - AC3, Spanish, 5.1 channels, 640kbps, 48khz.ac3"...
Video track 2 contains 173195 frames.
Subtitle track 14 contains 947 captions.
Subtitle track 15 contains 1053 captions.
Subtitle track 16 contains 864 normal and 24 forced captions.
Subtitle track 17 contains 892 normal and 27 forced captions.
Subtitle track 18 contains 943 normal and 29 forced captions.
Subtitle track 19 contains 949 normal and 31 forced captions.
Subtitle track 20 contains 694 normal and 21 forced captions.
Subtitle track 21 contains 947 normal and 27 forced captions.
Subtitle track 22 contains 945 normal and 30 forced captions.
Subtitle track 23 contains 952 normal and 29 forced captions.
Subtitle track 24 contains 923 normal and 31 forced captions.
Subtitle track 25 contains 889 normal and 20 forced captions.
Subtitle track 26 contains 23 forced captions.
Subtitle track 27 contains 25 forced captions.
Subtitle track 28 contains 27 forced captions.
Subtitle track 29 contains 29 forced captions.
eac3to processing took 23 minutes, 43 seconds.
Done.

As you can see, at the beginning of the log, the subs are listed as using tracks 13-28 and indeed, these are the numbers used in the filenames of the demuxed subs but right at the end of the log, subtitle tracks suddenly go from 14-29, is this a bug?

[pihug12]

I try to make a batch file to convert DTS sounds in MKV to AC3 640kbit/s.
mkdir temp
cd temp
..\eac3to.exe ..\%Input% -demux
..\eac3to.exe *.dts *.ac3 -640
..\eac3to.exe *.h264+*.ac3 %Output%
The demux works fine but then I get :
Source file "*.dts" not found.
Source file "*.h264" not found.

I think I cannot use the * character. How can I make that ?

___________

Why can't I demux a MKV file in an another output folder (ex: eac3to.exe video.mkv temp -demux) ?
C:\Documents and Settings\User\Desktop\eac3to>eac3to.exe video.mkv temp -demux
MKV, 1 video track, 2 audio tracks, 2 subtitle tracks, 1:38:42, 23.975p
1: V_MPEG4/ISO/AVC, English, 1280x528p (80:33)
2: DTS-ES, English, 5.1 channels, 24 bits, 1509kbps, 48khz
"DTS"
3: AC3 EX, French, 5.1 channels, 640kbps, 48khz
"AC3 5.1"
4: S_TEXT/UTF8, French
5: S_TEXT/UTF8, French
Track 2 is used for destination file "temp".
This audio conversion is not supported.

pihug12 :cool:

[Blue_MiSfit]

@Tebasuna51:

Thanks for your help! You're always the man with the downmixing scripts!

Quick question though. I currently have EncWAVtoAC3 2.0, and am trying to find 3.0. The official site seems to be down. Do you have a good mirror? For now, I will just use the handy-dandy SoundOut avisynth plugin, but I'd like to update EncWAVtoAC3.

Thanks again!
~MiSfit

[ACrowley]

@Tebasuna51:

Thanks for your help! You're always the man with the downmixing scripts!

Quick question though. I currently have EncWAVtoAC3 2.0, and am trying to find 3.0. The official site seems to be down. Do you have a good mirror? For now, I will just use the handy-dandy SoundOut avisynth plugin, but I'd like to update EncWAVtoAC3.

Thanks again!
~MiSfit

http://code.google.com/p/wavtoac3encoder/

[jj666]

I try to make a batch file to convert DTS sounds in MKV to AC3 640kbit/s.


link (http://audioconverter.heartware.dk/Tutorial).

Would suggest to try something similar to the above link. As that's obviously a HD source with re-encoded video, why didn't you convert the audio at the same time as encoding the movie?

-jj-

[laserfan]

Today I looked at my new Vantage Point BD and eac3to reported 51 chapters. This movie has 16 chapters so on a hunch went back to an earlier version of eac3to and sure enough, got 16 chapters. After trying subsequent versions I found that it broke going from v2.65 to v2.66:

eac3to v2.65
command line: eac3to h:\bdmv\stream\00011.m2ts
------------------------------------------------------------------------------
M2TS, 2 video tracks, 7 audio tracks, 15 subtitle tracks, 1:30:05
1: Chapters, 16 chapters *CORRECT*
2: h264/AVC, 1080p24 /1.001 (16:9)
3: h264/AVC, 480p24 /1.001 (20:11)
4: TrueHD/AC3, English, 5.1 channels, 48khz
5: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -30dB
6: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -30dB
7: AC3, Portuguese, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
8: AC3, Thai, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
9: AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
10: DTS Express, English, 2.0 channels, 24 bits, 192kbps, 48khz
11: Subtitle (PGS), English
12: Subtitle (PGS), English
13: Subtitle (PGS), French
14: Subtitle (PGS), Spanish
15: Subtitle (PGS), Portuguese
16: Subtitle (PGS), Chinese
17: Subtitle (PGS), Chinese
18: Subtitle (PGS), Korean
19: Subtitle (PGS), Thai
20: Subtitle (PGS), Spanish
21: Subtitle (PGS), Portuguese
22: Subtitle (PGS), Korean
23: Subtitle (PGS), Spanish
24: Subtitle (PGS), Portuguese
25: Subtitle (PGS), Englisheac3to v2.66
command line: eac3to h:\bdmv\stream\00011.m2ts
------------------------------------------------------------------------------
M2TS, 2 video tracks, 7 audio tracks, 15 subtitle tracks, 1:30:05
1: Chapters, 51 chapters *INCORRECT*
2: h264/AVC, 1080p24 /1.001 (16:9)
3: h264/AVC, 480p24 /1.001 (20:11)
4: TrueHD/AC3, English, 5.1 channels, 48khz
5: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -30dB
6: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -30dB
7: AC3, Portuguese, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
8: AC3, Thai, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
9: AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
10: DTS Express, English, 2.0 channels, 24 bits, 192kbps, 48khz
11: Subtitle (PGS), English
12: Subtitle (PGS), English
13: Subtitle (PGS), French
14: Subtitle (PGS), Spanish
15: Subtitle (PGS), Portuguese
16: Subtitle (PGS), Chinese
17: Subtitle (PGS), Chinese
18: Subtitle (PGS), Korean
19: Subtitle (PGS), Thai
20: Subtitle (PGS), Spanish
21: Subtitle (PGS), Portuguese
22: Subtitle (PGS), Korean
23: Subtitle (PGS), Spanish
24: Subtitle (PGS), Portuguese
25: Subtitle (PGS), English
A peek at the changelog shows the likely culprit:
v2.66
* changed eac3to to allow AAC encoding with 7.1 channels (for new Nero encoder)
* fixed AGM creation for files bigger than 4GB
* added support for Nero's new AAC Encoder download URL
* lowered volume of error/success sounds
* when there are 2 similar playlists the one with less chapters is ignored now
Broken all the way to 3.06. Let me know Madshi if I can help in any way to fix it.

[Snowknight26]

I try to make a batch file to convert DTS sounds in MKV to AC3 640kbit/s.

Assuming there is only 1 video and 1 DTS track:


for /f "delims=;" %%a in ('dir *.mkv /b /s') do {
eac3to %%a 1: %%a.video.mkv 2: %%a.ac3
mkvmerge -o %%a.DTS.mkv --language 1:eng --default-track 1:yes -d 1 -A -S %%a.video.mkv -a 0 -D -S %%a.ac3 --track-order 0:1,1:0
del %%a.video.mkv
del %%a.ac3
}
pause


Something like that at least.

[~bT~]

@ laserfan

have you tried the latest version??

[sidekick2]

@ laserfan

have you tried the latest version??

He said it's broken all the way thru 3.06 at the bottom of his post. He was using older versions to see where the breakage had occurred.

When I started reading his post myself, I thought, why the *heck* isn't he using a version from this century?!? :D

[laserfan]

And who knows, maybe it's the authoring that's wrong, not eac3to! I should not have assumed a Madshi bug when it comes to BD discs! :p

[madshi]

I'm trying to remux an MKV I created with an earlier version of eac3to which still contains pulldown information.

However, demuxing with -demux works and eac3to correctly identifies the video as VC-1.

Is this a bug or intentional misbehavior? :confused:
Support for such older VC-1 MKVs will come in a future version.

Madshi, though you say you plan on adding chapter and subtitle demuxing from mkvs, you didn't mention attachments. Any plans for those?
In the long run, yes. In theory it's really easy to extract MKV attachments. However, making the eac3to infrastructure understand that there are not only "video", "audio" and "subtitle" tracks, but also "attachments" is quite a lot of work. So it might take a while until I find the time to implement that. Don't know...

Also, doing anything with eac3to over a network share seems to be really slow. I'm remuxing an mkv and eac3to uses 0% CPU usage 90% of the time. It spikes up to about 10% for a fraction of a second... same with the network usage graph.
Anyone else having similar problems? It seems to work fine for me. Did older versions work better?

Wonder why most can't be parsed while a few odd ones (like this first one) can
Because some of them have sequence headers in the stream and some don't.

And finally, a small observation. Analysing mkvs with FLAC audio tracks takes far longer than ones with AC3 or DTS. Weird how that happens.
That's because eac3to actually decodes the FLAC track during analyzation. I guess I can optimize that...

I suggest to madshi don't put the bitdepth of the source with standard DTS and DTS HR because this data is misunderstand (and irrelevant).
Might be a good idea, at least to get rid of all these annoying "is it really necessary that eac3to patches it to 24bit?" questions. Will think about it.

Up until now, I've just been using:-

eac3to L: 1) -demux, followed by
eac3to video.vc1 video.mkv

to demux and then put/house the demuxed .vc1 'raw' output into a more useful .mkv container for graphedit to use for frameserving (via Haali Media Splitter, WMVideo Decoder DMO)

however, I'm curious to know if that would produce *exactly* the same video result (frame-for-frame) as, e.g. (assuming track 2 is the video in question):-
eac3to L: 1) 2: video.mkv

would the first 2-step method lose/discard any important information (frame-rate, etc) that the original container used to define the .vc1 stream?
I have read that a raw .vc1 stream is somewhat dependent on its container to fully specify it, so I'm worried that a 2-step process would be lossy in some way?
Both methods should produce similar results, as long as there are no gaps/overlaps in the video track (which is really rare). VC-1 does not depend on the container.

Is it possible that eac3to reports the wrong information in regards to 1080p or 1080i? I have a BD that says 1080i on the back... which is odd, however eac3to and tsMuxeR both report 1080p. I trust eac3to more than what's on the back cover.
eac3to reports two different things: In the track listing (e.g. if you do "eac3to BluRayFolder") eac3to shows what the Blu-Ray information records say without actually looking at the video data itself. This information *should* be reliable, more so than the back cover, but it could theoretically be wrong. If you actually let eac3to parse the movie file, it checks out the video bitstream. This should be even more reliable (unless there's a bug in eac3to). In your specific case it seems that the back cover is simply wrong.

Here's another silly sample for you, Madshi (20mb cut of source AC3)

This one is the 3.1ch 640kbps AC3 track from the BluRay release of Bubble. eac3to complains about encoding it down to a 2ch AC3
eac3to currently can't downmix funny channel configurations. It's on my to do list to add support for that, but it doesn't have top priority right now...

I just demuxed Eight Below, purely to see the subtitle information and if the english SUPs contained any forced captions and there's a discrepency between the subtitle track numbers displayed at the beginning of the log and those used at the end to display forced caption info.
Thanks, looks like a clear bug.

Why can't I demux a MKV file in an another output folder (ex: eac3to.exe video.mkv temp -demux) ?
You can't just invent your own command line commands. You'll have to live with what eac3to understands. What you could do is "eac3to video.mkv temp\video.*". That will demux into the MKV into a folder named "temp", if that's what you want.

Today I looked at my new Vantage Point BD and eac3to reported 51 chapters. This movie has 16 chapters so on a hunch went back to an earlier version of eac3to and sure enough, got 16 chapters. After trying subsequent versions I found that it broke going from v2.65 to v2.66
Can you please upload the CLIPINF and PLAYLIST folders? Zipped that should only be a few KBs.

[lchiu7]

See post #8037 and 8041
http://forum.doom9.org/showthread.php?p=1242555#post1242555

Sorry but can't see the relevance of those posts. I am having problems with the Nero codec, not Arcsoft.

A quick check of the DS Filter chain shows

"Nero Audio Decoder 2",{83E69A3D-56A5-4850-89C9-F6B6F003EB9B},00600000,"c:\program files\common files\ahead\dsfilter\neaudio2.ax"

Not entirely sure what else to check

And I am running Nero 7, not 8.

[laserfan]

...eac3to reported 51 chapters. This movie has 16 chapters so on a hunch went back to an earlier version of eac3to and sure enough, got 16 chapters. After trying subsequent versions I found that it broke going from v2.65 to v2.66...
Can you please upload the CLIPINF and PLAYLIST folders? Zipped that should only be a few KBs.Here you go--I think 00216 is the mpls of interest and the clpi prolly 00011. Advise if I can provide any further info!

[magic144]

Both methods should produce similar results, as long as there are no gaps/overlaps in the video track (which is really rare). VC-1 does not depend on the container.

thanks for the reply madshi,

are you saying that one of these 2 methods does NOT remove/compensate for gaps/overlaps - I thought eac3to would always remove gaps/overlaps when demuxing a full-disc-structure title... if this is not the case, which of these 2 methods doesn't invoke eac3to's gap removal...?

or is it simply the case that eac3to does not do ANY Video gap compensation (in any mode - i.e. it just does audio) - I guess if that was the case and that ever happened, eac3to would just flag it in an info/warning message? (never seen such a warning on any of my discs to date anyway) - is it something eac3to might handle in the future?

cheers again,
m

ps - incidentally, does anybody out there know of a way to accurately frameserve (in an AviSynth script) a raw .vc1 file without going to an intermediate .mkv container? I don't think the Haali source file splitter filter will take a .vc1 file as input if I try to build such a graph in graphedit.

[Thunderbolt8]

Here you go--I think 00216 is the mpls of interest and the clpi prolly 00011. Advise if I can provide any further info!better upload it externally, as getting approval can take ages.

[laserfan]

better upload it externally, as getting approval can take ages.Wonder what this is about; the last time I attached something here no approval was needed at all afaict.

[topsham]

I am trying to use the new aften build (762), but i get this error:

eac3to input.eac3 output.ac3
E-AC3, 5.1 channels, 1:42:01, 2046kbit/s, 48khz, dialnorm: -27dB
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
Encoding AC3...
invalid number of channels
Initialization of the AC3 encoder failed.


eac3to is latest (2.30) and this happens with all libaften builds (generic, SSE3 etc). It also happens when i use as input wav or raw or ac3 files. In other words always. But if i use aften.exe and encode to ac3 outside eac3to everything works fine.

Does it seem as a libaften or an eac3to problem?

I am just starting to use eac3to and have encountered this problem. Is there a fix please?

[DrNein]

Sorry but can't see the relevance of those posts. I am having problems with the Nero codec, not Arcsoft.

A quick check of the DS Filter chain shows

"Nero Audio Decoder 2",{83E69A3D-56A5-4850-89C9-F6B6F003EB9B},00600000,"c:\program files\common files\ahead\dsfilter\neaudio2.ax"

Not entirely sure what else to check

And I am running Nero 7, not 8.

Sorry, I mixed up two replies. That should have been:

See post #7880 and on and particularly:
http://forum.doom9.org/showthread.php?p=1238209#post1238209

[0xdeadbeef]

I wonder if eac3to considers a PAL speedup (-speedup) for the exported subtitles.
I exported video, audio and subtitles from a M2TS stream and applied the "-speedup" option to the audio tracks (23.976Hz -> 25Hz). Then I needed to convert the exported SUPs via SupRead (export options: HDDVD, 25Hz) and SubtitleCreator to SUB/IDX.
Unfortunately, audio and subtitles are getting out of sync (subtitles after audio), the more time passes by.
So I wonder: are the SUP time stamps not fixed correctly when using the "-speedup" options or does one of the other tools introduce the problem? Would it make sense to try 23.976fps in SupRead? Does this have an influence at all if converting from SUP (BD) to SUP (HDDVD)?

[mrr19121970]

I wonder if eac3to considers a PAL speedup (-speedup) for the exported subtitles.
I exported video, audio and subtitles from a M2TS stream and applied the "-speedup" option to the audio tracks (23.976Hz -> 25Hz). Then I needed to convert the exported SUPs via SupRead (export options: HDDVD, 25Hz) and SubtitleCreator to SUB/IDX.
Unfortunately, audio and subtitles are getting out of sync (subtitles after audio), the more time passes by.
So I wonder: are the SUP time stamps not fixed correctly when using the "-speedup" options or does one of the other tools introduce the problem? Would it make sense to try 23.976fps in SupRead? Does this have an influence at all if converting from SUP (BD) to SUP (HDDVD)?


I assume you are using tsMuxeR to build your streams back together ? You need to change the video frame rate to 25 and bind the subtitles to frame rate.

The GUI Clown_BD (http://clownbd.techxt.com) can do this in 1 option for you.

[madshi]

Here you go--I think 00216 is the mpls of interest and the clpi prolly 00011. Advise if I can provide any further info!
Thanks.

It seems that playlists 1 and 216 are identical - except that 216 has more chapters than 1. Don't know why they made the disc that way. There's a good reason, though, why eac3to chooses the playlist with more chapters. See here:

http://forum.doom9.org/showpost.php?p=1189107&postcount=6376

When there are two playlists with a different number of chapters, which should I use? There are good arguments either way...

are you saying that one of these 2 methods does NOT remove/compensate for gaps/overlaps - I thought eac3to would always remove gaps/overlaps when demuxing a full-disc-structure title... if this is not the case, which of these 2 methods doesn't invoke eac3to's gap removal...?

or is it simply the case that eac3to does not do ANY Video gap compensation (in any mode - i.e. it just does audio) - I guess if that was the case and that ever happened, eac3to would just flag it in an info/warning message? (never seen such a warning on any of my discs to date anyway) - is it something eac3to might handle in the future?
eac3to automatically fixes audio gaps/overlaps, but it does not automatically fix video gaps/overlaps. And the only way to fix video gaps/overlaps is to modify the timestamps of a container. So if you demux video, there's nothing eac3to can do to fix the gaps/overlaps. However, the next eac3to version will at least also complain about gaps/overlaps if you demux video (older versions didn't).

I am just starting to use eac3to and have encountered this problem. Is there a fix please?
Make sure you're using the latest eac3to version. If the problem still occurs then please post the eac3to log.

I wonder if eac3to considers a PAL speedup (-speedup) for the exported subtitles.
No. FPS changes for subtitles is not implemented yet. It's on my to do list, though...

[madshi]

eac3to v3.07 released

http://madshi.net/eac3to.zip

* added support for MKV video tracks without sequence headers in bitstream
* added support for old style MKV AAC tracks
* added support for various MKV "A_MS/ACM" audio formats
* added support for various MKV "V_MS/VFW/FOURCC" video formats
* added warning for tracks where bitstream parsing failed
* demuxing a video track now also complains about video gaps/overlaps
* the "-check" option now also complains about video gaps/overlaps
* optimized memory allocation
* fixed: adding subtitle caption count to filenames sometimes didn't work
* fixed: subtitle caption counts in log sometimes had wrong track numbers
* fixed: all non-supported MKV tracks shared the same description
* fixed: incorrect framerate mismatch complaint was shown for pulldown sources
* fixed: FLAC tracks in MKV files don't slow down detection, anymore
* fixed: source file detection read 300MB from every source file
MKV support for those audio and video codecs which are natively supported by eac3to should be pretty complete now. MKV tracks using other video/audio codecs may demux successfully (or not), but are not officially supported by eac3to.

If you have any MKV samples with eac3to supported audio/video codecs which are not properly handled by eac3to, please upload a small sample for me. Thanks!

MKV subtitles, chapters and attachments are still not supported at all. Maybe next week...

[Thunderbolt8]

thanks!

[0xdeadbeef]

Ok, I have another problem. I tried to convert the E-AC3 soundtracks of the European King King HD-DVD to AC3 and - while I was at it - apply a PAL speedup:

eac3to.exe h:\HVDVD_TS\FEATURE_1.EVO+h:\HVDVD_TS\FEATURE_2.EVO 2: d:\dvd\kk\Chapters.txt 3: d:\dvd\kk\kk.* 4: d:\dvd\kk\en.ac3 -libav -speedup -384 7: d:\dvd\kk\de.ac3 -libav -speedup -384 11: d:\dvd\kk\en.sup 14: d:\dvd\kk\de.sup 28: d:\dvd\kk\de_forced.sup

For the original EVOs EAC3TO reported an audio delay of 1001ms only for the English sound track.

HD-DVD King Kong

1: Joined EVO file
2: Chapters, 50 chapters with names
3: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
4: E-AC3, English, 5.1 channels, 1536kbps, 48khz, dialnorm: -27dB, 1001ms
5: E-AC3, French, 5.1 channels, 384kbps, 48khz, dialnorm: -27dB
6: E-AC3, Italian, 5.1 channels, 384kbps, 48khz, dialnorm: -27dB
"Italian"
7: E-AC3, German, 5.1 channels, 384kbps, 48khz, dialnorm: -27dB
8: E-AC3, Spanish, 5.1 channels, 384kbps, 48khz, dialnorm: -27dB
9: E-AC3, Japanese, 5.1 channels, 384kbps, 48khz, dialnorm: -27dB
[...]

However, after muxing the video and audio streams with TSMuxer, the audio is about 32 seconds (!) too early - and I'm not even sure if this is consistent. As converting the EVOs to M2TS with TSMuxer alone (no audio conversion) worked perfectly, I would assume that something goes wrong here with the audio conversion. I'm not quite sure what though. Any hints?

[honai]

Thanks madshi!

[madshi]

Ok, I have another problem. I tried to convert the E-AC3 soundtracks of the European King King HD-DVD to AC3 and - while I was at it - apply a PAL speedup
You forgot to speedup the video track.

[0xdeadbeef]

You forgot to speedup the video track.

Thanks for the hint. I'm pretty sure I read somehwere (3.06 command line help?) that the "-speedup" option was only valid for audio tracks.
But ok, obviously my fault. I must admit that I used a GUI for my M2TS conversions and used the command line the frist time for the EVOs.

[madshi]

Thanks for the hint. I'm pretty sure I read somehwere (3.06 command line help?) that the "-speedup" option was only valid for audio tracks.
It's not. How is a sped up audio track supposed to sync with a non-sped up video track? ;)

[laserfan]

It seems that playlists 1 and 216 are identical - except that 216 has more chapters than 1. Don't know why they made the disc that way. There's a good reason, though, why eac3to chooses the playlist with more chapters. See here:

http://forum.doom9.org/showpost.php?p=1189107&postcount=6376

When there are two playlists with a different number of chapters, which should I use? There are good arguments either way....But madshi, the new versions of eac3to don't in any way expose the correct chapter list. Here's all it shows me:D:\>eac3to h:
1) 00216.mpls, 00011.m2ts, 1:30:05
- h264/AVC, 1080p24 /1.001 (16:9)
- h264/AVC, 480p24 /1.001 (16:9)
- TrueHD, English, multi-channel, 48khz
- AC3, French, multi-channel, 48khz
- AC3, Spanish, multi-channel, 48khz
- AC3, Portuguese, multi-channel, 48khz
- AC3, Thai, multi-channel, 48khz
- AC3, English, stereo, 48khz
- DTS Express, English, stereo, 48khz

2) 00202.mpls, 00105.m2ts, 0:26:44
- MPEG2, 1080p24 /1.001 (16:9)
- AC3, English, stereo, 48khz

3) 00203.mpls, 00106.m2ts, 0:16:00
- MPEG2, 1080p24 /1.001 (16:9)
- AC3, English, stereo, 48khzBut then even if I force eac3to to look at 00001 it still gives me 51 chapters? Maybe it was/is another .mpls that gives 16 chapters?

D:\>eac3to h:\bdmv\playlist\00001.mpls
1) 00001.mpls, 00011.m2ts, 1:30:05
- h264/AVC, 1080p24 /1.001 (16:9)
- h264/AVC, 480p24 /1.001 (16:9)
- TrueHD, English, multi-channel, 48khz
- AC3, French, multi-channel, 48khz
- AC3, Spanish, multi-channel, 48khz
- AC3, Portuguese, multi-channel, 48khz
- AC3, Thai, multi-channel, 48khz
- AC3, English, stereo, 48khz
- DTS Express, English, stereo, 48khz

D:\>eac3to h:\bdmv\playlist\00001.mpls 1: chapter.txt
M2TS, 2 video tracks, 7 audio tracks, 15 subtitle tracks, 1:30:05, 24p /1.001
1: Chapters, 51 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: h264/AVC, 480p24 /1.001 (20:11)
4: TrueHD/AC3, English, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 640kbps, 48khz)
5: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -30dB
6: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -30dB
7: AC3, Portuguese, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
8: AC3, Thai, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
9: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
10: DTS Express, English, 2.0 channels, 24 bits, 192kbps, 48khz
11: Subtitle (PGS), English
12: Subtitle (PGS), English
13: Subtitle (PGS), French
14: Subtitle (PGS), Spanish
15: Subtitle (PGS), Portuguese
16: Subtitle (PGS), Chinese
17: Subtitle (PGS), Chinese
18: Subtitle (PGS), Korean
19: Subtitle (PGS), Thai
20: Subtitle (PGS), Spanish
21: Subtitle (PGS), Portuguese
22: Subtitle (PGS), Korean
23: Subtitle (PGS), Spanish
24: Subtitle (PGS), Portuguese
25: Subtitle (PGS), English
Creating file "chapter.txt"...
Done.Still puzzled why the older version of eac3to finds-and-picks 16 chapters correctly, but the newer doesn't expose 16 chapters anywhere? In that post you linked to, at least that movie showed the feature as being in multiple playlists; this one does not.

Well, I guess this command:

D:\>eac3to h:\bdmv\playlist\ xxxxx.mpls

is giving me the main movie playlist (& incorrect 51 chapters) every time, so apparently I've not used a proper eac3to command... :confused:

EDIT: Here are the very different results when using eac3to v2.65! Apparently 00217 is the correct mpls to choose!
D:\>"c:\program files (x86)\eac3told\eac3to.exe" h:\
1) 00217.mpls, 00011.m2ts, 1:30:05
- h264/AVC, 1080p24 /1.001 (16:9)
- h264/AVC, 480p24 /1.001 (16:9)
- TrueHD, English, multi-channel, 48khz
- AC3, French, multi-channel, 48khz
- AC3, Spanish, multi-channel, 48khz
- AC3, Portuguese, multi-channel, 48khz
- AC3, Thai, multi-channel, 48khz
- AC3, English, stereo, 48khz
- DTS Express, English, stereo, 48khz

2) 00216.mpls, 00011.m2ts, 1:30:05
- h264/AVC, 1080p24 /1.001 (16:9)
- h264/AVC, 480p24 /1.001 (16:9)
- TrueHD, English, multi-channel, 48khz
- AC3, French, multi-channel, 48khz
- AC3, Spanish, multi-channel, 48khz
- AC3, Portuguese, multi-channel, 48khz
- AC3, Thai, multi-channel, 48khz
- AC3, English, stereo, 48khz
- DTS Express, English, stereo, 48khz

3) 00202.mpls, 00105.m2ts, 0:26:44
- MPEG2, 1080p24 /1.001 (16:9)
- AC3, English, stereo, 48khz

4) 00203.mpls, 00106.m2ts, 0:16:00
- MPEG2, 1080p24 /1.001 (16:9)
- AC3, English, stereo, 48khz

D:\>"c:\program files (x86)\eac3told\eac3to.exe" h:\ 1)
M2TS, 2 video tracks, 7 audio tracks, 15 subtitle tracks, 1:30:05
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: h264/AVC, 480p24 /1.001 (20:11)
4: TrueHD/AC3, English, 5.1 channels, 48khz
5: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -30dB
6: AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -30dB
7: AC3, Portuguese, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
8: AC3, Thai, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
9: AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
10: DTS Express, English, 2.0 channels, 24 bits, 192kbps, 48khz
11: Subtitle (PGS), English
12: Subtitle (PGS), English
13: Subtitle (PGS), French
14: Subtitle (PGS), Spanish
15: Subtitle (PGS), Portuguese
16: Subtitle (PGS), Chinese
17: Subtitle (PGS), Chinese
18: Subtitle (PGS), Korean
19: Subtitle (PGS), Thai
20: Subtitle (PGS), Spanish
21: Subtitle (PGS), Portuguese
22: Subtitle (PGS), Korean
23: Subtitle (PGS), Spanish
24: Subtitle (PGS), Portuguese
25: Subtitle (PGS), English

[magic144]

eac3to automatically fixes audio gaps/overlaps, but it does not automatically fix video gaps/overlaps. And the only way to fix video gaps/overlaps is to modify the timestamps of a container. So if you demux video, there's nothing eac3to can do to fix the gaps/overlaps.

thanks again Madshi - and for the new version :-)

One more question then. Are you saying that doing something like this:-

eac3to L: 1) 2: video.mkv

rather than a blanket disc title demux would keep a VC1 stream in a container (.mkv, albeit a different one from its original .m2ts housing) and allow eac3 to fix video gaps/overlaps, and is that an existing feature or a future planned capability? If I remember rightly, you said the use of video gaps/overlaps is incredibly rare in source material - but you have seen it?

Cheers,
m

[0xdeadbeef]

It's not. How is a sped up audio track supposed to sync with a non-sped up video track? ;)

Well, indeed it doesn't seem to make much sense to have a seperate speedup option for each stream, either. I guess it's highly unlikely that someone wants to speed up just some streams. This also creates confusion regarding the chapter timestamps and the subtitles.
Why don't you just make it a global option?

[Snowknight26]

eac3to.exe "G:\Movies\The Third Man\The.Third.Man.mkv"
MKV, 1 video track, 4 audio tracks, 1:45:13, 24p /1.001
1: h264/AVC, English, 1436x1080 23.976p
2: FLAC, English, 1.0 channels, 1:45:13, 16 bits, 48khz
"FLAC 1.0 @ 202kbps"
3: VORBIS, English, 1.0 channels, 48khz
"Commentary w/ Steven Soderbergh and Tony Gilroy"
4: VORBIS, English, 1.0 channels, 48khz
"Commentary w/ film scholar Dana Polan"
5: VORBIS, English, 1.0 channels, 48khz
"Abridged recording of Graham GreeneÆs treatment, read by actor Richard Clark
e"
v01 The video bitstream is encoded in a non-standard framerate.
Bitstream parsing for tracks 3-5 failed.
Demuxing these tracks may still produce correct results - or not.
The video was encoded using 23.976fps, just like all my encodes I have done. Should I be specifying 24000/1001 instead? Or does it not matter because its only a cosmetic issue?
And a small suggestion. It might be better to say "Demuxing these tracks may or may not produce correct results." instead of what you currently have. At least thats what I think.

[DrNein]

Well, indeed it doesn't seem to make much sense to have a seperate speedup option for each stream, either. I guess it's highly unlikely that someone wants to speed up just some streams. This also creates confusion regarding the chapter timestamps and the subtitles.
Why don't you just make it a global option?

Applying speedup/slowdown per stream is useful when replacing audio tracks -and thus avoiding the extra processing time by applying it to the video only. Also, the original audio can be demuxed unmodified for preservation or additional manipulation.

[honai]

madshi,

would it require a lot of effort to include a simple video cutting mechanism, e.g. remove the first 2000ms of the video stream when demuxing and muxing from/to MKV?

[tvjunky]

Hi Madshi!
Did you have changed anything regarding Drive-Read-Speed in your last version? I used the following command to copy video and audio direct from my drives to a hdd-folder:

eac3to f: 1) 3: video.mkv 9: audio.flac (from my HD-DVD-Drive)

With version 3.06 it took ~45 min.
With version 3.07 it took 12 hours!!!

I have the same effect with my Blu-Ray-Drive. I stopped the test with my Blu-Ray-Drive after 4 hours.

Thanks in advance, S.

[KevinMcPool]

Hi,i have previously converted dtshd and truehd tracks to lpcm for my PCH A100 but have now gotten an A110.Is it possible to convert the lpcm tracks back to dtshd or truehd without any loss in quality???Thanks.

[tebasuna51]

Hi,i have previously converted dtshd and truehd tracks to lpcm for my PCH A100 but have now gotten an A110.Is it possible to convert the lpcm tracks back to dtshd or truehd without any loss in quality???Thanks.
Of course if you have the appropiate encoder (don't exist free encoders) but you don't need eac3to for this.

The most common free lossless encoder is FLAC, supported in mkv container.
You can demand multichannel FLAC support in PCH.

[KevinMcPool]

Of course if you have the appropiate encoder (don't exist free encoders) but you don't need eac3to for this.

The most common free lossless encoder is FLAC, supported in mkv container.
You can demand multichannel FLAC support in PCH.
Thanks for your reply but i dont quite understand.Can eac3to re-convert the lpcm back to dtshd or truehd for remuxing to .ts or .m2ts.Like you say there is no mutichannel support for FLAC over hdmi with the PCH

[jj666]

Thanks for your reply but i dont quite understand.Can eac3to re-convert the lpcm back to dtshd or truehd for remuxing to .ts or .m2ts.Like you say there is no mutichannel support for FLAC over hdmi with the PCH

No, it can't. You can only convert to AC3 (with EAC3TO) or DTS (with EAC3TO via Surcode) - both lossy so I don't think will achieve the quality you want...

As tebasuna said, what you are asking for cannot be achieved with free encoders.

Just re-rip the disk and don't convert the audio this time - TONMT should process TrueHD I believe (or keep the original .M2TS file if not seamless branching disk).

Cheers,

-jj-

[KevinMcPool]

No, it can't. You can only convert to AC3 (with EAC3TO) or DTS (with EAC3TO via Surcode) - both lossy so I don't think will achieve the quality you want...

As tebasuna said, what you are asking for cannot be achieved with free encoders.

Just re-rip the disk and don't convert the audio this time - TONMT should process TrueHD I believe (or keep the original .M2TS file if not seamless branching disk).

Cheers,

-jj-
Ok thanks JJ666.I'll re-rip the disks using tsmuxer on those with dtshd and tonmt(+ ?also ts4np) on those with truehd.

[hubblec4]

Hi Madshi

Your work is great and very well.


I found some old videos with sound.ogg
Is it possible to support them. It's not important but will be nice.
ogg is freeware, so i think its maybe easy to do this??!


hubble

[KevinMcPool]

I have a 5.1 set-up with a modern hdmi receiver that can decode Dtshd and truehd, a PCH A110 and i have a number of movies with 7.1 dtshd sound..would it be better to demux with eac3to and the sonic decoder to 5.1 dtshd or let my receiver matrix the 7.1 channels to my 5.1 setup???Thanks.

[lchiu7]

Sorry, I mixed up two replies. That should have been:

See post #7880 and on and particularly:
http://forum.doom9.org/showthread.php?p=1238209#post1238209

Thanks but I have the ax file and dll and both are registered. The fact that Nerovision can playback AC3 audio indicates they must be working so the problem is elsewhere

[topsham]

Make sure you're using the latest eac3to version. If the problem still occurs then please post the eac3to log.
The latest version fixed it for me. Thank you, much appreciated.

[tebasuna51]

There are a problem with 'stdout.wav' and automatic second pass for overflow.
When this occurs the output is duplicated but the corrected second part is out of sync (because the second header) with garbage result.

Seems the problem is solved when add -2pass or -normalize parameters. Now seems the first ouput is inhibited and only the second pass is really output.
I say 'seems' because a second problem: when use 'stdout' there aren't log and the user don't know what happen.

There are a method to cancel the automatic second pass for overflow?
Many times are little peaks (<0.1dB) due to imperfections in lossy codecs than can be assumed.

[bobberty]

Hi madshi, thank you for eac3to...it's such an awesome program.

I'd like to use eac3to as an archival tool. By this, I mean I'd like to take a BD or HDDVD and demux the main movie video stream and audio streams. I'd like to keep them in their original format, without processing or doing anything to the streams, ideally just a bit for bit extraction. Then I'd like to back up these demuxed files to DVD-R to save for future use. Since programs and containers and encoders change all the time, I'd like to keep these original files in their original state (or as close to their original state as possible).

Is just passing -demux to eac3to the best way to do this? Would I need to pass other options, like -keepDialnorm and -keepPulldown?

Also, for demuxing, does it make a difference if I use eac3to by itself without installing the Nero, Arcsoft, and Sonic filters? Or does having these filters installed give better demuxing results?

Thanks again for eac3to, it's great. Bob.

edit:

It would appear that it doesn't matter if you have the filters installed or not for demuxing. I demuxed a HDDVD with the filters and without, and the files came out the same. I compared their md5 checksums and everything was the same.

[bigdog660]

Hi Madshi!
Did you have changed anything regarding Drive-Read-Speed in your last version? I used the following command to copy video and audio direct from my drives to a hdd-folder:

eac3to f: 1) 3: video.mkv 9: audio.flac (from my HD-DVD-Drive)

With version 3.06 it took ~45 min.
With version 3.07 it took 12 hours!!!

I have the same effect with my Blu-Ray-Drive. I stopped the test with my Blu-Ray-Drive after 4 hours.

Thanks in advance, S.

Madshi,

I can confirm it is a lot slower using Hellboy as I had in previous tests. Been running for 20 minutes, but I only have three progress dashes in the command window. I'm not going to finish the test since it's obvious it will take hours.

P.S. Reading eac3to e: reads about as fast as it did in v3.06, but reading eac3to e: 1) is a little slower than v3.06.

TIA

[Snowknight26]

Just a couple of issues:
C:\unzipped\eac3to>eac3to.exe "C:\temp\i'm on a boat.mkv" 1: C:\temp\boat.h264 -
changeto29.970
MKV, 1 video track, 1 audio track, 0:03:09, 29.989p
1: h264/AVC, 720p29.989
2: AAC, 2.0 channels, 44.1khz
v01 The video bitstream is encoded in a non-standard framerate.
Was asked to modify track 1: to 29.970, but the original FPS is not supported.

C:\unzipped\eac3to>eac3to.exe "C:\temp\i'm on a boat.mkv" 1: C:\temp\boat.h264
MKV, 1 video track, 1 audio track, 0:03:09, 29.989p
1: h264/AVC, 720p29.989
2: AAC, 2.0 channels, 44.1khz
v01 The video bitstream is encoded in a non-standard framerate.
v01 Extracting video track number 1...
v01 Creating file "C:\temp\boat.h264"...
v01 Video overlaps for 1 frames at playtime 0:00:00.
v01 Video overlaps for 1 frames at playtime 0:00:01.
v01 Video overlaps for 1 frames at playtime 0:00:01.
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eac3to processing took 10 seconds.
Done.

Source file is this (http://www.youtube.com/watch?v=R7yfISlGLNU&fmt=22) remuxed from mp4 to mkv with mkvmerge (default settings).

[dburckh]

Madshi,

I can confirm it is a lot slower using Hellboy as I had in previous tests. Been running for 20 minutes, but I only have three progress dashes in the command window. I'm not going to finish the test since it's obvious it will take hours.

P.S. Reading eac3to e: reads about as fast as it did in v3.06, but reading eac3to e: 1) is a little slower than v3.06.

TIA
+Bump
3.0.7 seems to suffer from the same problem:
Setup: XP SP3; AnyDVD 6.5.2.2; eac3to 3.0.7; LG GGC-H20L; 300 (HD DVD)

[VonZippa]

While decoding a TrueHD audio track using eac3to, I received this error:

libav Lossless check failed - expected 0, calculated d6

A search through the forums revealed that this is likely a bug in the libav TrueHD decoder. I am providing a 10 MB sample which reproduces this bug.

http://rapidshare.com/files/196566632/shank73.thd.html

This sample is from the very end of the track, which is when the error occurs.

Thanks for the great work on eac3to... it's an awesome program and it's been an invaluable tool!

[bigdog660]

Quote:
Originally Posted by bigdog660
Madshi,

I can confirm it is a lot slower using Hellboy as I had in previous tests. Been running for 20 minutes, but I only have three progress dashes in the command window. I'm not going to finish the test since it's obvious it will take hours.

P.S. Reading eac3to e: reads about as fast as it did in v3.06, but reading eac3to e: 1) is a little slower than v3.06.

TIA

Quote:
Originally Posted by dburckh
+Bump
3.0.7 seems to suffer from the same problem:
Setup: XP SP3; AnyDVD 6.5.2.2; eac3to 3.0.7; LG GGC-H20L; 300 (HD DVD)

Just so there is no confusion: In my original post, I was comparing v3.07 to v3.06 where as v3.07 is much slower demuxing than v3.06. Thanks.

[ultratoto14]

Same here, stopped eac3to after 15 minutes in eac3to 1) with 3.07

Go back to 3.06 and 35 minutes to extract h264 + ac3 of Peter Pan's movie.

[mariusella]

Awesome, we've needed a separte thread for a while. Cant wait to play with some TrueHD tracks! Thanks for all the hard work Madshi

Yeah you are right

[peterjcat]

Same here, stopped eac3to after 15 minutes in eac3to 1) with 3.07

Go back to 3.06 and 35 minutes to extract h264 + ac3 of Peter Pan's movie.

Reading direct from disc eac3to 1) takes forever for me too. Am now ripping direct to HD and going from there, which is as fast as ever.

Recent eac3to was supposed to increase read speeds direct from disc, but I wonder if it works for all drives.

I am using a USB drive (LGC combo drive in an USB enclosure) which may make the difference, perhaps if everyone who's experiencing slowness can report what kind of drive they're using it might help Madshi out.

[ultratoto14]

The speed increase was there in 3.06, ripping and extracting in one pass is 35 minutes for me with a 4X liteon bd drive. But the speed increase is gone (far far away) in the 3.07. For a 40Gb disk, it couldn't be faster.

[madshi]

Actually an option to output a cue sheet from eac3to would take all the pain away.

http://wiki.hydrogenaudio.org/index.php?title=Cuesheet

Performer, title and track names aren't really important since we need to tag the resultant flac anyway. "FEATURE" , "FEATURE ARTIST" and chapter numbers for track names would suffice.

If you don't see value in it, I'll right a program to convert your chapter files to cue sheets.
I was already asked to offer an option to produce tsMuxeR compatible chapter files. But to be honest, I don't really like the idea to add one option for every tool out there which may use its own private chapter format... :(

But then even if I force eac3to to look at 00001 it still gives me 51 chapters?
Oh, that seems to be a bug, will have to check that...

Are you saying that doing something like this:-

eac3to L: 1) 2: video.mkv

rather than a blanket disc title demux would keep a VC1 stream in a container (.mkv, albeit a different one from its original .m2ts housing) and allow eac3 to fix video gaps/overlaps, and is that an existing feature or a future planned capability?
Yes. It's an existing feature.

If I remember rightly, you said the use of video gaps/overlaps is incredibly rare in source material - but you have seen it?
It's rare, but not unheard of. Especially broken broadcasts can have such problems.

Well, indeed it doesn't seem to make much sense to have a seperate speedup option for each stream, either. I guess it's highly unlikely that someone wants to speed up just some streams.
Actually it does make sense to only speed up some streams. E.g. you may want to store the original audio track as it is (for future use) and speed it up at the same time. If I made "-speedup" a global option, doing such a thing would not be possible, anymore...

The video was encoded using 23.976fps, just like all my encodes I have done. Should I be specifying 24000/1001 instead? Or does it not matter because its only a cosmetic issue?
It's mostly a cosmetic issue. However, all Blu-Rays I've seen are using 24000/1001 (or 24000/1000, but never 23976/1000). And all broadcasts I've seen use 60000/1001. So I do consider 23976/1000 "non-standard". But again: It's mostly a cosmetic issue...

would it require a lot of effort to include a simple video cutting mechanism, e.g. remove the first 2000ms of the video stream when demuxing and muxing from/to MKV?
"A lot of effort" is a relative term. It would be possible and doable, but it would cost some time, and I still have so many other things on my to do list. However, I could use some very limited cutting functionality myself for cleaning up the beginning of self-recorded broadcasts. So it might come sooner or later, but not too soon...

I found some old videos with sound.ogg
Is it possible to support them. It's not important but will be nice.
You mean Vorbis, right? Ogg is just a container. I will probably not add full support for Vorbis audio tracks. Eventually I might add support for decoding it through libav (if libav actually contains a Vorbis decoder, don't know). But I'm not sure yet...

There are a problem with 'stdout.wav' and automatic second pass for overflow.
Didn't think about that. Will check that out later...

There are a method to cancel the automatic second pass for overflow?
Many times are little peaks (<0.1dB) due to imperfections in lossy codecs than can be assumed.
You can use "-3db" or even "-1db" to avoid 2nd pass in most cases. Since clipping can only occur with floating point input, anyway, there should be no measurable loss in audio quality caused by the volume change.

I have a 5.1 set-up with a modern hdmi receiver that can decode Dtshd and truehd, a PCH A110 and i have a number of movies with 7.1 dtshd sound..would it be better to demux with eac3to and the sonic decoder to 5.1 dtshd or let my receiver matrix the 7.1 channels to my 5.1 setup???Thanks.
Let your receiver do the work.

I'd like to take a BD or HDDVD and demux the main movie video stream and audio streams. I'd like to keep them in their original format, without processing or doing anything to the streams, ideally just a bit for bit extraction.
Why extracting HD DVD video tracks bit for bit? I think it would make more sense to let eac3to remove the pulldown. If you let eac3to do that, you'll usually end up with fully Blu-Ray compatible streams. In the early days of the HD DVD / Blu-Ray war Microsoft had supplied the Dual-Format-Studios with a little tool which would convert their HD DVD style VC-1 tracks to Blu-Ray style VC-1. eac3to does basically the same thing. I see no sense in keeping the pulldown flags in the video stream...

Is just passing -demux to eac3to the best way to do this? Would I need to pass other options, like -keepDialnorm and -keepPulldown?
You can use these options, but I don't really recommend that. The manipulations eac3to does by default usually have their purpose. But if you insist, of course you can stop eac3to from "improving" the video/audio data.

Also, for demuxing, does it make a difference if I use eac3to by itself without installing the Nero, Arcsoft, and Sonic filters? Or does having these filters installed give better demuxing results?
No need for any of those filters if you just demux.

Just a couple of issues:
C:\unzipped\eac3to>eac3to.exe "C:\temp\i'm on a boat.mkv" 1: C:\temp\boat.h264 -
changeto29.970
MKV, 1 video track, 1 audio track, 0:03:09, 29.989p
1: h264/AVC, 720p29.989
2: AAC, 2.0 channels, 44.1khz
v01 The video bitstream is encoded in a non-standard framerate.
Was asked to modify track 1: to 29.970, but the original FPS is not supported.
Strange source. Will check that out later...

Edit: I can download that Youtube video as FLV, but not as MP4. How can I get that in MP4 format?

While decoding a TrueHD audio track using eac3to, I received this error:

A search through the forums revealed that this is likely a bug in the libav TrueHD decoder. I am providing a 10 MB sample which reproduces this bug.
Thanks for the sample. But your search should also have revealed that it's no problem if you just receive one of those libav warnings. That means that max 0.8ms of your audio track is not lossless (if at all). All the other millions of milliseconds are still perfect.

[madshi]

eac3to v3.08 released

http://madshi.net/eac3to.zip

* fixed: reading physical disc speed was abysmal (introduced in v3.07)
* fixed: read error from physical drive resulted in crash

[Rectal Prolapse]

madshi, just out of curiosity - do you use asynchronous I/O (ie. overlapped I/O) when reading and writing streams? I can easily saturate RAID0'd hard drives if I have 3 pending I/O requests going at the same time, using 32 megabyte buffers each request. Going from 50 MB/sec to 160 MB/sec is amazing! :)

[laserfan]

eac3to v3.08 released
An update during the workweek! Many thanks madshi! :)

[tvjunky]

eac3to v3.08 released

* fixed: reading physical disc speed was abysmal (introduced in v3.07)
* fixed: read error from physical drive resulted in crash

Thank you very, very much for this, madshi, and for all the time you spent in this great tool!!!

S.

[Snowknight26]

Edit: I can download that Youtube video as FLV, but not as MP4. How can I get that in MP4 format?

Fastest way is probably to do this:
Visit the URL (http://www.youtube.com/watch?v=R7yfISlGLNU&fmt=22) then type copy/paste this into the address bar:
javascript:document.location.href='http://www.youtube.com/get_video?fmt='+(isHDAvailable?'22':'18')+'&video_id='+swfArgs['video_id']+'&t='+swfArgs['t']
Should prompt you to download it.

[liquidator87]

May be a stupid question, but reading from bluray folder from hard disk is still supported? Cause I get "Error reading file ".\BDMV\STREAM\xxxxx.m2ts" with latest versions, but for example v3.01 works good

[madshi]

just out of curiosity - do you use asynchronous I/O (ie. overlapped I/O) when reading and writing streams?
No, but I'm reading/writing in different threads, at least when multiple tracks are demuxed.

I can easily saturate RAID0'd hard drives if I have 3 pending I/O requests going at the same time, using 32 megabyte buffers each request. Going from 50 MB/sec to 160 MB/sec is amazing! :)
How many MBs does the current eac3to build do when doing simple demuxing (no audio transcoding)?

May be a stupid question, but reading from bluray folder from hard disk is still supported? Cause I get "Error reading file ".\BDMV\STREAM\xxxxx.m2ts" with latest versions, but for example v3.01 works good
Should still work, and it seems to work for me. From which folder did you call eac3to and which command line did you use?

Fastest way is probably to do this
That works, thanks.

[Rectal Prolapse]

madshi, I actually haven't done any tests with recent builds of eac3to - sorry! I'll get around to it, one day... I do recall eac3to 2.xx was fairly decent for speeds.

For reference, I have 2 Samsung 1 TB Spinpoint F1 drives in a RAID0 configuration on a motherboard with Intel ICH8 SATA controller. File copies with Windows Explorer is dreadfully slow - 50 MB/sec, so it is pretty clear Windows XP doesn't do multithreaded or asynchronous I/O.
But Teracopy, which does asynchronous I/O, is roughly twice as fast.

[honai]

"A lot of effort" is a relative term. It would be possible and doable, but it would cost some time, and I still have so many other things on my to do list. However, I could use some very limited cutting functionality myself for cleaning up the beginning of self-recorded broadcasts. So it might come sooner or later, but not too soon...

Yes, that's my intention, too. :)

[yonta]

madshi, Can you take a look at this sample (http://www.sendspace.com/file/uc953c)?
It seems to be a 29.97p but eac3to doesn't seem to properly handle it.

[hubblec4]

@madshi

Vorbis (ogg) support. i mean only for decoding. encoding is not necessary.

hubble

[T800]

I've tried "The Getaway" and "Letters from Iwo Jima" HDDVD's but get the same error on both. I've tried eac3to 3.03, 3.07 and 3.08

"AC3 overflow in the thd ac3 joiner."

It always aborts very near the start

Am I doing something wrong?

http://i471.photobucket.com/albums/rr75/T800mc/Picture1-1.png

EDIT:
These are discs that have also failed using ToNMT.

[T800]

You decode Track 4 to THD+AC3, but you use an HD-DVD, so there is no AC3 inside THD. EAC3TO does also not show that it should be (in Track 4).

Try to use only "audio.thd" or (better) "audio.*", that should work. EAC3TO cannot mix AC3 into THD by itself.

How do I create a Blu-ray style THD track from the HDDVD THD track?

[mbcd]

Sorry, my fault ... eac3to can handle it (read about many times its not possible to do without an "special" encoder.

[madshi]

It seems to be a 29.97p but eac3to doesn't seem to properly handle it.
Interlaced VC-1 streams are currently not properly handled by eac3to. Fixing that is on my to do list.

I've tried "The Getaway" and "Letters from Iwo Jima" HDDVD's but get the same error on both. I've tried eac3to 3.03, 3.07 and 3.08

"AC3 overflow in the thd ac3 joiner."

It always aborts very near the start
It's good that it aborts very near the start, because this means that you can probably create a little sample for me, so that I can reproduce the problem on my PC... ;)

EAC3TO cannot mix AC3 into THD by itself.
WRONG.

How do I create a Blu-ray style THD track from the HDDVD THD track?
Just as T800 tried to do.

Thats nor possible until you habe an Authoring-Program für BLURAY or DOLBYDIGITAL-HD.

The only way is to store it as normal HD-DVD-THD without AC3-Core inside. But TSMuxer will not accept that if you`d like to use it there.

So there is no "official" way at this moment until you have one of those Authoring Programs.
Wrong again. eac3to can convert TrueHD tracks to TrueHD+AC3 just fine.

[Jeff Flowerday]

Interlaced VC-1 streams are currently not properly handled by eac3to. Fixing that is on my to do list.




Would that explain why I've never been able to get my 1080i VC1 AC/DC No Bull blu-ray muxed into a mkv/flac without sync issues on play back?

[T800]

It's good that it aborts very near the start, because this means that you can probably create a little sample for me, so that I can reproduce the problem on my PC... ;)



Should I just put the small TrueHD+AC3 files on Mediafile that were produced?

[Atak_Snajpera]

send him iwo1.evo sample (50mb for example) You can use DGsplit

[zn]

I have tried different Nero 7 versions (rebooting and using General-CleanTool after each uninstall)

7.2.7.0
7.5.1.1
7.5.9.0
7.7.5.1a
7.8.5.0
7.9.6.0
7.11.10.0

and eac3to always response with "Nero Audio Decoder (Nero 7) is not working correctly"

Special test - I choose Nero Audio Decoder as preffered in MPC and ...
if I am using mpc.exe to open MKV file with AC3 and MP3 track - only MP3 decoded by Nero Audio Decoder
if I rename mpc.exe to recode.exe and open the same file, then both type of tracks is decoded by Nero

And the most interesting thing, some time ago I have 7.11.10.0 installed and it was working for some time (eac3to), and then later it stopped working like this (renaming eac3to.exe to recode.exe don't help).

Any ideas?

[kypec]

Would moderators be so kind and make this thread sticky, please? This project tool is being heavily developed for quite long time, I think it deserves to stand above most others.

[T800]

It's good that it aborts very near the start, because this means that you can probably create a little sample for me, so that I can reproduce the problem on my PC... ;)


Hopefully I've done this right.

The Getaway Sample:
http://www.mediafire.com/?tm1zdtm3mtm

Iwo Jima Sample:
http://www.mediafire.com/?emod3nxdiwd

[madshi]

Would that explain why I've never been able to get my 1080i VC1 AC/DC No Bull blu-ray muxed into a mkv/flac without sync issues on play back?
Yes.

And the most interesting thing, some time ago I have 7.11.10.0 installed and it was working for some time (eac3to), and then later it stopped working like this (renaming eac3to.exe to recode.exe don't help).

Any ideas?
I don't really know why it's not working for you. You don't need to rename eac3to. eac3to works around that file name requirement.

Hopefully I've done this right.

The Getaway Sample:
http://www.mediafire.com/?tm1zdtm3mtm

Iwo Jima Sample:
http://www.mediafire.com/?emod3nxdiwd
You have done it right. But it seems that the samples are too small, cause I cannot reproduce the problem on my PC:

C:\Desktop>eac3to iwo1sample.EVO 3: test.thd+ac3
EVO, 1 video track, 2 audio tracks, 4 subtitle tracks, 0:00:43
1: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
2: E-AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
3: TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
4: Subtitle (DVD)
5: Subtitle (DVD)
6: Subtitle (DVD)
7: Subtitle (DVD)
a03 Extracting audio track number 3...
a03 Extracting audio track number 3...
a03 Removing TrueHD dialog normalization...
a03 Removing TrueHD dialog normalization...
a03 Decoding with libav/ffmpeg...
a03 Encoding AC3 <640kbps> with libAften...
a03 Creating file "test.thd+ac3"...
a03 The original audio track has a constant bit depth of 16 bits.
Video track 1 contains 1034 frames.
eac3to processing took 3 seconds.
Done.
You can easily try this yourself. Please try to cut a bigger samples, maybe 100MB, then check if you can reproduce the problem with the bigger sample. I can only fix the problem if I can reproduce the problem on my PC.

I don't need two different samples from 2 different movies. One sample from one movie is all I need, as long as I can reproduce the problem with that sample...

[T800]

You have done it right. But it seems that the samples are too small, cause I cannot reproduce the problem on my PC:

C:\Desktop>eac3to iwo1sample.EVO 3: test.thd+ac3
EVO, 1 video track, 2 audio tracks, 4 subtitle tracks, 0:00:43
1: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
2: E-AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
3: TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
4: Subtitle (DVD)
5: Subtitle (DVD)
6: Subtitle (DVD)
7: Subtitle (DVD)
a03 Extracting audio track number 3...
a03 Extracting audio track number 3...
a03 Removing TrueHD dialog normalization...
a03 Removing TrueHD dialog normalization...
a03 Decoding with libav/ffmpeg...
a03 Encoding AC3 <640kbps> with libAften...
a03 Creating file "test.thd+ac3"...
a03 The original audio track has a constant bit depth of 16 bits.
Video track 1 contains 1034 frames.
eac3to processing took 3 seconds.
Done.
You can easily try this yourself. Please try to cut a bigger samples, maybe 100MB, then check if you can reproduce the problem with the bigger sample. I can only fix the problem if I can reproduce the problem on my PC.

I don't need two different samples from 2 different movies. One sample from one movie is all I need, as long as I can reproduce the problem with that sample...

It must be something I'm doing wrong then, I just tried the sample I put on Mediafire and get this:

http://i471.photobucket.com/albums/rr75/T800mc/Picture1-2.png

[nautilus7]

I have tried different Nero 7 versions (rebooting and using General-CleanTool after each uninstall)

7.2.7.0
7.5.1.1
7.5.9.0
7.7.5.1a
7.8.5.0
7.9.6.0
7.11.10.0

and eac3to always response with "Nero Audio Decoder (Nero 7) is not working correctly"

Special test - I choose Nero Audio Decoder as preffered in MPC and ...
if I am using mpc.exe to open MKV file with AC3 and MP3 track - only MP3 decoded by Nero Audio Decoder
if I rename mpc.exe to recode.exe and open the same file, then both type of tracks is decoded by Nero

And the most interesting thing, some time ago I have 7.11.10.0 installed and it was working for some time (eac3to), and then later it stopped working like this (renaming eac3to.exe to recode.exe don't help).

Any ideas?You need Nero 7.8.5.0 or later installed + the HD DVD/Blu-ray plug in registered!

[madshi]

It must be something I'm doing wrong then, I just tried the sample I put on Mediafire and get this
Strange. Why does it work for me? Are you sure that you're using the latest eac3to version? Check "eac3to -test".

[T800]

Strange. Why does it work for me? Are you sure that you're using the latest eac3to version? Check "eac3to -test".

Here:
http://i471.photobucket.com/albums/rr75/T800mc/Picture1-3.png

[Jeff Flowerday]

Yes.



Do you have a sample to work with if not I can give you one?

[madshi]

It must be something I'm doing wrong then, I just tried the sample I put on Mediafire and get this:

http://i471.photobucket.com/albums/rr75/T800mc/Picture1-2.png
Can somebody else please try this? Can you reproduce the problem with the sample T800 uploaded?

@T800, you didn't replace any files in the eac3to folder, did you? E.g. libAften DLL or something like that? Please try deleting the whole eac3to folder and redownloading it, just to be safe... For me the samples you uploaded definitely convert just fine!

Do you have a sample to work with if not I can give you one?
I would have asked for a sample, if I needed one... ;)

[T800]

I haven't changed anything.

I will delete and download again on Monday (no access to Windows at home).

[TinTime]

Can somebody else please try this? Can you reproduce the problem with the sample T800 uploaded?

Got these results:
eac3to v3.08
command line: "D:\Vtemp\programs\eac3to\eac3to.exe" iwo1sample.evo 3: zi.thd+ac3
------------------------------------------------------------------------------
EVO, 1 video track, 2 audio tracks, 4 subtitle tracks, 0:00:43
1: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
2: E-AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
3: TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
4: Subtitle (DVD)
5: Subtitle (DVD)
6: Subtitle (DVD)
7: Subtitle (DVD)
[a03] Extracting audio track number 3...
[a03] Extracting audio track number 3...
[a03] Removing TrueHD dialog normalization...
[a03] Removing TrueHD dialog normalization...
[a03] Decoding with libav/ffmpeg...
[a03] Encoding AC3 <640kbps> with libAften...
[a03] Creating file "zi.thd+ac3"...
[a03] The original audio track has a constant bit depth of 16 bits.
Video track 1 contains 1034 frames.
eac3to processing took 3 seconds.
Done.

eac3to v3.08
command line: "D:\Vtemp\programs\eac3to\eac3to.exe" getawaysample.evo 2: zg.thd+ac3
------------------------------------------------------------------------------
EVO, 1 video track, 3 audio tracks, 3 subtitle tracks, 0:00:35
1: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
2: TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
3: E-AC3, 5.1 channels, 384kbps, 48khz, dialnorm: -27dB
4: E-AC3 Surround, 2.0 channels, 384kbps, 48khz, dialnorm: -27dB
5: Subtitle (DVD)
6: Subtitle (DVD)
7: Subtitle (DVD)
[a02] Extracting audio track number 2...
[a02] Extracting audio track number 2...
[a02] Removing TrueHD dialog normalization...
[a02] Removing TrueHD dialog normalization...
[a02] Decoding with libav/ffmpeg...
[a02] Encoding AC3 <640kbps> with libAften...
[a02] Creating file "zg.thd+ac3"...
[a02] AC3 overflow in the thd ac3 joiner. <ERROR>
Aborted at file position 55574528. <ERROR>

So in the first run Iwo Jima worked but The Getaway failed. However I ran both again and they both worked this time.

Gave it a third go and Iwo Jima failed but The Getaway worked.

Gave it a fourth go and Iwo Jima worked but The Getaway failed.

So an intermittent problem - good luck!

Also, as this is my first post in this thread, I want to say a big thank you for this tool. It's incredibly useful. :thanks:

[madshi]

So in the first run Iwo Jima worked but The Getaway failed. However I ran both again and they both worked this time.

Gave it a third go and Iwo Jima failed but The Getaway worked.

Gave it a fourth go and Iwo Jima worked but The Getaway failed.

So an intermittent problem - good luck!
Thanks, that helps. I'll increase some internal buffers for the next build, that might fix the problem (I hope).

[TinTime]

Ok. I'll hang on to the samples and give them a few runs through with 3.09 when you release it.

[T800]

Thanks, that helps. I'll increase some internal buffers for the next build, that might fix the problem (I hope).

I'm using Windows via Parallel Desktop on a mac if that makes any difference.

[zn]

Anyone can confirm that eac3to+nero7 combination works without Microsoft NET Framework 2 and 3?

[dat720]

madshi, not sure if you've noticed but the Nero AAC Codec link has changed on the Nero site, after a quick google i found:
http://www.nero.com/eng/downloads-nerodigital-nero-aac-codec.php

Where eac3to still reports:
http://www.nero.com/eng/nero-aac-codec.html

[Snowknight26]

Even with v3.08, eac3to seems to lot take a lot longer (5+ seconds) to start.

[zn]

madshi, not sure if you've noticed but the Nero AAC Codec link has changed on the Nero site, after a quick google i found:
http://www.nero.com/eng/downloads-nerodigital-nero-aac-codec.php

Where eac3to still reports:
http://www.nero.com/eng/nero-aac-codec.html

http://www.nero.com/eng/nero-aac-codec.html
-> changed to ->
http://www.nero.com/eng/technologies-aac-codec.html

[DrNein]

Anyone can confirm that eac3to+nero7 combination works without Microsoft NET Framework 2 and 3?

I am not aware of any NET dependency. They certainly work without version 3. I just have version 2 SP2 installed and that covers the few apps I use including ATI CCC. I have not come across any needing 3 or 3.5 yet.

[kypec]

Hello madshi, first I want to thank you for such an excellent tool.
I use it only for very simple decoding of AC3 (within VOB) to WAV, however, there is some bug in the output header of WAV file occuring.
According to this RIFF WAVE specification details (http://www.sonicspot.com/guide/wavefiles.html#fmt) the Compression code in Format chunk should be 1 (0x0001) for standard uncompressed PCM.
When I run eac3to something.vob something.wav -libav -down16 the resulting wav file has Compression code set to -1 (0xFFFE) which is wrong. All other attributes are just fine (Number of channels=2, Sample rate=48000, Average bytes per second=192000, Block align=4, Bits per sample=16). Consequently, some apps like normalize (http://savannah.nongnu.org/download/normalize/normalize-0.7.7-win32.zip) for instance refuse to work with such WAV files. Is this related to libav or is it your WAV output writer which produces such corrupted header values?

[Jeff Flowerday]

I am not aware of any NET dependency. They certainly work without version 3. I just have version 2 SP2 installed and that covers the few apps I use including ATI CCC. I have not come across any needing 3 or 3.5 yet.

.NET 3.0 and 3.5 both still use the CLR of .NET 2.0. So you can think of them as additions to 2.0

Yes this is confusing but . NET 2.0 is really that last base .Net version we've had. 3.0 and 3.5 being supersets on top of 2.0

That said I'm pretty sure madshi develops in a non managed code version of delphi so there is no dependency on the .NET Framework.

[madshi]

madshi, not sure if you've noticed but the Nero AAC Codec link has changed on the Nero site, after a quick google i found
I was already using the new link to check for new Nero AAC versions. I just output the wrong URL to the command line window. Will be fixed in the next build.

Even with v3.08, eac3to seems to lot take a lot longer (5+ seconds) to start.
Can you please post a few more details? Does it occur with all kinds of source files? Or just with physical Blu-Ray/HD DVD discs? Or [...]? What part of the processing exactly takes longer? Etc...

I use it only for very simple decoding of AC3 (within VOB) to WAV, however, there is some bug in the output header of WAV file occuring.
According to this RIFF WAVE specification details (http://www.sonicspot.com/guide/wavefiles.html#fmt) the Compression code in Format chunk should be 1 (0x0001) for standard uncompressed PCM.
That documentation is outdated. The WAV file created by eac3to is perfectly fine. It just uses a newer revision of the WAV file format, which is not supported by some applications. If you run into trouble you can use the eac3to "-simple" switch which will make eac3to output "old style" WAV files (with Compression code 0x0001). Please note that old style WAV files do not have a field to store the channel configuration of the channels. So for multichannel data, using the new file format revision is recommended. See also here:

http://www.microsoft.com/whdc/device/audio/multichaud.mspx

[kypec]

That documentation is outdated. The WAV file created by eac3to is perfectly fine. It just uses a newer revision of the WAV file format, which is not supported by some applications. If you run into trouble you can use the eac3to "-simple" switch which will make eac3to output "old style" WAV files (with Compression code 0x0001). Please note that old style WAV files do not have a field to store the channel configuration of the channels. So for multichannel data, using the new file format revision is recommended. See also here:http://www.microsoft.com/whdc/device/audio/multichaud.mspx
Thanks for your explanation, I noticed there are some Extra format bytes in the header present. It's also nice that there is -simple option switch prepared already. Unfortunately its existence is well hidden in the documentation (usage help). Could you please make it visible in further releases (and maybe also other undocumented switches which there possibly are)?:thanks:

:::EDIT:::
I do not want to sound rude, yet I still think that the Compression code should be set to 0x0001.This is standard value meaning Uncompressed PCM. It has nothing to do with number of channels, sample rate, bit depth of samples, or mapping of the channels in case of WAVE_FORMAT_EXTENSIBLE enhancements. Therefore I see no reason why you set it to 0xFFFE for standard uncompressed PCM Windows WAV file, it just doesn't make sense.
:::EDIT 2:::
Sorry, now I found that 0xFFFE designates that WAVE_FORMAT_EXTENSIBLE additions. Sorry for being such dumb ass.:o

[honai]

Can you please post a few more details? Does it occur with all kinds of source files? Or just with physical Blu-Ray/HD DVD discs? Or [...]? What part of the processing exactly takes longer? Etc...

I have also noticed that on certain inputs, e.g. demuxing a playlist from the German WALL-E from HDD (lots of small segments), eac3to seems to do nothing for a few seconds first, then outputs the stats and continues as usual. Seems like there is a pre-check at this moment that takes longer than expected therefore you didn't see the need to include a progress bar for this step ...

[Snowknight26]

Can you please post a few more details? Does it occur with all kinds of source files? Or just with physical Blu-Ray/HD DVD discs? Or [...]? What part of the processing exactly takes longer? Etc...

All files. I'm mostly running eac3to over a gigabit network, but it happens even when running it directly to/from an incredibly fast RAID. As soon as I type any command, eac3to sits there for several seconds before even showing the progress bar.

[DrNein]

You need Nero 7.8.5.0 or later installed + the HD DVD/Blu-ray plug in registered!

See: http://forum.doom9.org/showthread.php?p=1238209#post1238209

eac3to 3.06 and later apparently do not work with Nero Audio Decoder. Please take a look madshi.

[magic144]

Hi, I'm wondering if I've found something useful.
I was curious as to why eac3to reported 47 chapters on a BD title (X-Files 2) when in fact standalone-player playback (and PowerDVD 8) offers only 27 - I started to poke around in BDInfo and BDedit.

BDInfo (0.5.2) reported the correct number of chapters (27) for playlist 00004.MPLS (playlist 00002.MPLS is a monolithic 1-chapter playlist of the same content by the looks of it, even though eac3to reports 47 chapters for it too)

Upon closer inspection, BDedit reveals that playlist 00004.MPLS has 46 of what it calls PlayListMarks. However, of those marks, only 27 are identified as having type "Entry-Mark" and these seem to tie up with actual chapter points, whereas the other PlayListMark types, which BDedit calls "Link Points", do not/should not feature in a chapter list.

(by the way, I think eac3to has 47 chapters here because it has created 1 extra timestamp at the very end of the playlist times - also note that playlist 00002.mpls only has 1 Entry-Mark and 46 subsequent Link Points showing in BDedit, which also ties up with BDInfo's reporting of only 1 chapter for that playlist)

Do you think this is right madshi? And can you/should you therefore exclude these "link point" types from your chapter lists??

Thanks,
m

[laserfan]

...BDInfo (0.5.2) reported the correct number of chapters (27) for playlist 00004.MPLS ....BDedit reveals that playlist 00004.MPLS has 46 of what it calls PlayListMarks...27 are identified as having type "Entry-Mark" and these seem to tie up with actual chapter points...I looked to BDInfo for chapters on this one (those are accurate), but I can't correlate the "mark times" for entry-marks in BDedit to any actual chapter mark in the movie. So I dunno if these relate to chapters at all, or if maybe something's wrong with BDedit.

[magic144]

@laserfan,

the "Entry-Mark" "mark times" in BDedit are always shown relative to a particular PlayList item (PI),
so in order to work out the absolute time to which they refer, you need to know:-
a) the PlayList Item absolute start time as well as
b) the "mark time" (also taking the PI's "IN time" into account which always seems to be an offset of 2 seconds in this case)

for example, in playlist 00004.MPLS, PlayListMark 002 (the second actual entry-mark in the list, PlayListMark 000 being the first, and 001 being a link-point) is an "entry-mark" referencing PI 001 and having a "mark time" of 00:01:43.351
We know that PlayList Item (PI) 001 (00018.m2ts) starts at 00:02:47.417 (basically, the length of PI 000), with its usual IN time of 00:00:02.000

so PlayListMark 002 occurs at absolute time 00:01:43.351 - 00:00:02.000 + 00:02:47.417 = 00:04:28.768 (exactly as BDInfo shows for the start time of Chapter 2) - what eac3to refers to as Chapter 3

m

[laserfan]

Good job, sir... I had tried (and failed rather miserably) to find that relationship! :o

[magic144]

well like I said, I'm just inferring things from the outputs of these progs - hopefully I've found something that might improve madshi's chapter list output!

[madshi]

I have also noticed that on certain inputs, e.g. demuxing a playlist from the German WALL-E from HDD (lots of small segments), eac3to seems to do nothing for a few seconds first, then outputs the stats and continues as usual. Seems like there is a pre-check at this moment that takes longer than expected therefore you didn't see the need to include a progress bar for this step ...
Yeah, for seamless branching titles eac3to opens every m2ts file and searches for the first and last video timestamp etc. That is done before the first progress bar appears. I guess I could add just another progress bar for this initial file scan. It usually causes a noticable delay only for seamless branching titles.

All files. I'm mostly running eac3to over a gigabit network, but it happens even when running it directly to/from an incredibly fast RAID. As soon as I type any command, eac3to sits there for several seconds before even showing the progress bar.
I can't reproduce that. For me eac3to starts processing at once, except - see above.

eac3to 3.06 and later apparently do not work with Nero Audio Decoder. Please take a look madshi.
The Nero Audio Decoder works just fine for me in v3.08.

[madshi]

I was curious as to why eac3to reported 47 chapters on a BD title (X-Files 2) when in fact standalone-player playback (and PowerDVD 8) offers only 27 - I started to poke around in BDInfo and BDedit.

BDInfo (0.5.2) reported the correct number of chapters (27) for playlist 00004.MPLS (playlist 00002.MPLS is a monolithic 1-chapter playlist of the same content by the looks of it, even though eac3to reports 47 chapters for it too)

Upon closer inspection, BDedit reveals that playlist 00004.MPLS has 46 of what it calls PlayListMarks. However, of those marks, only 27 are identified as having type "Entry-Mark" and these seem to tie up with actual chapter points, whereas the other PlayListMark types, which BDedit calls "Link Points", do not/should not feature in a chapter list.

Do you think this is right madshi? And can you/should you therefore exclude these "link point" types from your chapter lists??
Actually I'm already removing "link points" from the chapter list - if there is an "entry mark" somewhere afterwards. However, if there are "link points" after the last "entry mark" I'm not deleting these "link points". The reason for that is that at least one Blu-Ray had "entry marks" for the first half runtime of the movie and "link points" for the second half runtime of the movie.

Are those additional "link points" in X-Files 2 behind the last "entry point"? If not, I'd like to look into the problem. In that case please zip and upload the PLAYLIST and CLIPINF folders. Should only be a few KBs zipped.

[honai]

I guess I could add just another progress bar for this initial file scan. It usually causes a noticable delay only for seamless branching titles.

I think it might be enough to just output a string, "Performing initial file scan. This could take a few moments. Please wait ...", then continue as usual.

[magic144]

Hi madshi,
the link points seem to be interspersed with entry marks, so no, they're not after the last one.
Here's the files you wanted to check into the issue further. Hope this helps.

m

http://www.mediafire.com/?mnmxlytnyny

[laserfan]

For fun I just looked at Groundhog Day BD, and while it uses 16 chapters, all of eac3to, BDinfo, and BDedit show 31 (!) this latter having 31 entry-marks (only), no link-points ie. it's not branching at all.

So I don't know what's going on with some of the new titles. :confused:

[madshi]

the link points seem to be interspersed with entry marks, so no, they're not after the last one.
Here's the files you wanted to check into the issue further.
Actually with playlist 1, there is one entry mark, and then 46 link points. There's no further entry mark after all those link points. So that makes 47 chapters for playlist 1.

[magic144]

@madshi
sure, but it's playlist 4 that seems to be the right one (listed as the second title), which eac3to also lists as having 47 chapters

[magic144]

@laserfan
do none of the playlists for Groundhog Day register in BDinfo as having only 16 chapters then :-(
if so, there's apparently more variables to this riddle...

[laserfan]

do none of the playlists for Groundhog Day register in BDinfo as having only 16 chapters then...Good question! There are three .mpls I can find for the main movie (00001, 00220, 00221). Predictably (I guess, given I think madshi said he looked for longest .mpls?) eac3to finds 00221.mpls, which is the only one of the three with the incorrect chapter list! The other two do indeed show 16 chapters!

One has to wonder if authoring houses are doing this deliberately to confuse us re-doers and backer-uppers! :p

[Atak_Snajpera]

One has to wonder if authoring houses are doing this deliberately to confuse us re-doers and backer-uppers!
Most likely :) This would explain why main movie is very often divided in many parts without any particular order and size.

[madshi]

sure, but it's playlist 4 that seems to be the right one
And how is eac3to supposed to know that playlist 4 is the right one instead of playlist 1?

which eac3to also lists as having 47 chapters
There's a bug which result in eac3to always listing the default playlist's chapters, even if you manually specify a playlist. This will be fixed in the next build.

[mrr19121970]

I'm trying to demux the blu-ray 'While She Was Out'. This is what ea3to tells me about the stream:

eac3to v3.08
command line: "E:\TVIX\eac3to\eac3to.exe" "R:\" 1) 1: "D:\DEMUX\New\Chapters_1.txt" 2: "D:\DEMUX\New\Video_2.*" -slowdown 4: "D:\DEMUX\New\Audio_4_English.AC3" -slowdown 3: "D:\DEMUX\New\Audio_3_German.AC3" -slowdown -LOG="D:\DEMUX\New\eac3to_PASS3_LOG.LOG"
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 1 subtitle track, 1:22:24, 50i
1: Chapters, 17 chapters
2: VC-1, 1080i50 (16:9)
3: DTS Master Audio, German, 5.1 channels, 16 bits, 48khz
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48khz)
4: DTS Master Audio, English, 5.1 channels, 16 bits, 48khz
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48khz)
5: Subtitle (PGS), German
Creating file "D:\DEMUX\New\Chapters_1.txt"...
[a04] The ArcSoft and Sonic decoders don't seem to work, will use libav instead.
[a04] The libav DTS decoder doesn't decode the full DTS-HD information. <WARNING>
[a03] The ArcSoft and Sonic decoders don't seem to work, will use libav instead.
[a03] The libav DTS decoder doesn't decode the full DTS-HD information. <WARNING>
[v02] Extracting video track number 2...
[v02] Writing new framerate "24fps /1.001" to bitstream.
[a03] Extracting audio track number 3...
[a03] Extracting DTS core...
[a03] Decoding with libav/ffmpeg...
[a03] Remapping channels...
[a03] Changing FPS from 25.000 to 23.976...
[a04] Extracting audio track number 4...
[a04] Extracting DTS core...
[a04] Decoding with libav/ffmpeg...
[a04] Remapping channels...
[a04] Changing FPS from 25.000 to 23.976...
[a03] Encoding AC3 <640kbps> with libAften...
[a04] Encoding AC3 <640kbps> with libAften...
[v02] Creating file "D:\DEMUX\New\Video_2.vc1"...
[a03] Creating file "D:\DEMUX\New\Audio_3_German.AC3"...
[a04] Creating file "D:\DEMUX\New\Audio_4_English.AC3"...
[a04] The original audio track has a constant bit depth of 64 bits.
[a04] The processed audio track has a constant bit depth of 64 bits.
[a03] The original audio track has a constant bit depth of 64 bits.
[a03] The processed audio track has a constant bit depth of 64 bits.
Video track 2 contains 123601 frames.
eac3to processing took 20 minutes, 43 seconds.
Done.


However the resulting VC1 stream is 48000/1001, even if I give '-stripPulldown' the resulting stream still has the interlaced flag. Is this normal ?


D:\DEMUX\While She Was Out _ 1>"E:\TVIX\eac3to\eac3to.exe" video_2.vc1
VC-1, 1080i48 /1.001 (16:9)

[madshi]

Just a couple of issues:
C:\unzipped\eac3to>eac3to.exe "C:\temp\i'm on a boat.mkv" 1: C:\temp\boat.h264 -
changeto29.970
MKV, 1 video track, 1 audio track, 0:03:09, 29.989p
1: h264/AVC, 720p29.989
2: AAC, 2.0 channels, 44.1khz
v01 The video bitstream is encoded in a non-standard framerate.
Was asked to modify track 1: to 29.970, but the original FPS is not supported.

C:\unzipped\eac3to>eac3to.exe "C:\temp\i'm on a boat.mkv" 1: C:\temp\boat.h264
MKV, 1 video track, 1 audio track, 0:03:09, 29.989p
1: h264/AVC, 720p29.989
2: AAC, 2.0 channels, 44.1khz
v01 The video bitstream is encoded in a non-standard framerate.
v01 Extracting video track number 1...
v01 Creating file "C:\temp\boat.h264"...
v01 Video overlaps for 1 frames at playtime 0:00:00.
v01 Video overlaps for 1 frames at playtime 0:00:01.
v01 Video overlaps for 1 frames at playtime 0:00:01.
v01 Video overlaps for 1 frames at playtime 0:00:02.
[...]
v01 Video overlaps for 1 frames at playtime 0:03:09.
v01 Video overlaps for 1 frames at playtime 0:03:09.
v01 Video overlaps for 1 frames at playtime 0:03:10.
eac3to processing took 10 seconds.
Done.
These problems will be fixed in the next build. The problems were caused by the really strange framerate, which is really "29.989fps". Even the video bitstream contains exactly this framerate.

I'm trying to demux the blu-ray 'While She Was Out'. This is what ea3to tells me about the stream:

eac3to v3.08
command line: "E:\TVIX\eac3to\eac3to.exe" "R:\" 1) 1: "D:\DEMUX\New\Chapters_1.txt" 2: "D:\DEMUX\New\Video_2.*" -slowdown 4: "D:\DEMUX\New\Audio_4_English.AC3" -slowdown 3: "D:\DEMUX\New\Audio_3_German.AC3" -slowdown -LOG="D:\DEMUX\New\eac3to_PASS3_LOG.LOG"
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 1 subtitle track, 1:22:24, 50i
1: Chapters, 17 chapters
2: VC-1, 1080i50 (16:9)
3: DTS Master Audio, German, 5.1 channels, 16 bits, 48khz
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48khz)
4: DTS Master Audio, English, 5.1 channels, 16 bits, 48khz
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48khz)
5: Subtitle (PGS), German
Creating file "D:\DEMUX\New\Chapters_1.txt"...
[a04] The ArcSoft and Sonic decoders don't seem to work, will use libav instead.
[a04] The libav DTS decoder doesn't decode the full DTS-HD information. <WARNING>
[a03] The ArcSoft and Sonic decoders don't seem to work, will use libav instead.
[a03] The libav DTS decoder doesn't decode the full DTS-HD information. <WARNING>
[v02] Extracting video track number 2...
[v02] Writing new framerate "24fps /1.001" to bitstream.
[a03] Extracting audio track number 3...
[a03] Extracting DTS core...
[a03] Decoding with libav/ffmpeg...
[a03] Remapping channels...
[a03] Changing FPS from 25.000 to 23.976...
[a04] Extracting audio track number 4...
[a04] Extracting DTS core...
[a04] Decoding with libav/ffmpeg...
[a04] Remapping channels...
[a04] Changing FPS from 25.000 to 23.976...
[a03] Encoding AC3 <640kbps> with libAften...
[a04] Encoding AC3 <640kbps> with libAften...
[v02] Creating file "D:\DEMUX\New\Video_2.vc1"...
[a03] Creating file "D:\DEMUX\New\Audio_3_German.AC3"...
[a04] Creating file "D:\DEMUX\New\Audio_4_English.AC3"...
[a04] The original audio track has a constant bit depth of 64 bits.
[a04] The processed audio track has a constant bit depth of 64 bits.
[a03] The original audio track has a constant bit depth of 64 bits.
[a03] The processed audio track has a constant bit depth of 64 bits.
Video track 2 contains 123601 frames.
eac3to processing took 20 minutes, 43 seconds.
Done.

However the resulting VC1 stream is 48000/1001, even if I give '-stripPulldown' the resulting stream still has the interlaced flag. Is this normal ?
eac3to can not make truely interlaced content progressive. Such a thing is not possible without fully reencoding the video track. eac3to can only remove the pulldown from an HD DVD style video track. Such HD DVD style tracks are progressively encoded, but contain additional pulldown flags. The Blu-Ray you have seems to be native interlaced. No way to fix that. However, playback of "1080i48 /1.001" should work fairly well, too.

AFAIK Blu-Ray does not support 1080p25 content. If you want PAL framerate, you have to use 1080i50. Don't ask me why the Blu-Ray companies have made such a stupid decision...

[madshi]

eac3to v3.09 released

http://madshi.net/eac3to.zip

* added support for MKV "SRT/UTF8", "SRT/ASCII", "ASS" and "SSA" subtitles
* increased some internal buffers to avoid AC3 overflow in the "thd ac3 joiner"
* fixed: frame counting didn't work for MKV video tracks
* fixed: video tracks FPS change was sometimes declined
* fixed: video tracks with "strange" FPS were sometimes handled incorrectly
* fixed: clipping removal 2nd pass was executed even for "stdout"
* fixed: "eac3to -test" displayed an outdated Nero download link
* fixed: specifying a specific playlist still used default playlist's chapters

[Atak_Snajpera]

AFAIK Blu-Ray does not support 1080p25 content. If you want PAL framerate, you have to use 1080i50. Don't ask me why the Blu-Ray companies have made such a stupid decision...
Officially you are right. However I believe that players which have AVCHD label should play those files without any problems. There are HD-camcorder like Canon HF10/100 (AVCHD structure) which can record in 25p. PS3 proves my theory and plays without any complaints :)

[mrr19121970]

the 1080i50 is a shop bought blu-ray. i guess it's tsMuxeR that's wrecking it, but thankfully a straight copy of the main stream onto my TVIX HDD is playable.

tsMuxeR GUI doesn't even reckognise the VC1 stream, and the CLI accepts it and chews it up. I'll try again with manually setting the framerate to see if that helps.

[Snowknight26]

Thanks madshi! The youtube video works now.

One thing I noticed, though, is that the language for subtitles is no longer displayed.

[magic144]

And how is eac3to supposed to know that playlist 4 is the right one instead of playlist 1?

There's a bug which result in eac3to always listing the default playlist's chapters, even if you manually specify a playlist. This will be fixed in the next build.

cheers for the fix madshi - just tested 3.09 on X-Files 2 and it now lists and extracts 27 chapters for title 2 - thanks!!

as to identifying which title/playlist is the "right" title, I wouldn't know how to figure that out programatically, I guess the easiest way for now is just manually, i.e. to try the disc in a standalone player or software player and see what the chapters are there and then see which title eac3to identifies that matches

presumably, to do this programatically, you'd have to do what a player does and navigate through the disc's playback logic, taking into account user menu choices and such-like along the way - no doubt far more than eac3to was ever envisioned as doing! - I think the way it is right now is perfectly fine!!

[laserfan]

eac3to v3.09 released

* fixed: specifying a specific playlist still used default playlist's chaptersI just tried this on Groundhog Day, and although it still selects the 00221.mpls playlist with (incorrect) 31 chapters, I can now direct it to 00001.mpls with (correct) 16 chapters shown!

Thanks madshi for the update! :)

[itsancho]

oops, no languages with 3.09 C:\Users\iTSAN>z\eac3to "H:\Lost (2004) (TV series) The Complete Fourth Season
2007) Blu-ray AVC PCM\Disc 1" 2)
M2TS, 1 video track, 7 audio tracks, 22 subtitle tracks, 0:43:09, 24p /1.001
1: Chapters, 8 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: AC3, 5.1 channels, 640kbps, 48khz
4: RAW/PCM, 5.1 channels, 16 bits, 48khz
5: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz
6: AC3, 5.1 channels, 640kbps, 48khz
7: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz
8: AC3, 5.1 channels, 640kbps, 48khz
9: AC3, 2.0 channels, 192kbps, 48khz, dialnorm: -30dB
10: Subtitle (PGS)
11: Subtitle (PGS)
12: Subtitle (PGS)
13: Subtitle (PGS)
14: Subtitle (PGS)
15: Subtitle (PGS)
16: Subtitle (PGS)
17: Subtitle (PGS)
18: Subtitle (PGS)
19: Subtitle (PGS)
20: Subtitle (PGS)
21: Subtitle (PGS)
22: Subtitle (PGS)
23: Subtitle (PGS)
24: Subtitle (PGS)
25: Subtitle (PGS)
26: Subtitle (PGS)
27: Subtitle (PGS)
28: Subtitle (PGS)
29: Subtitle (PGS)
30: Subtitle (PGS)
31: Subtitle (PGS)

[Thunderbolt8]

* fixed: frame counting didn't work for MKV video tracksdoes this refer to the frame adding of the video file after remuxing or was this only a display bug? would we have to redo some files because of that if fluent playback is possibly not given?

[DrNein]

The Nero Audio Decoder works just fine for me in v3.08.

Hmm, this is odd. I retested and it is "not working correctly" in versions 3.06 through 3.09 but with earlier versions it "works fine". The problem is not just in reporting either as it actually falls back to libav/ffmpeg.

[madshi]

does this refer to the frame adding of the video file after remuxing or was this only a display bug?
Just a display bug.

Hmm, this is odd. I retested and it is "not working correctly" in versions 3.06 through 3.09 but with earlier versions it "works fine". The problem is not just in reporting either as it actually falls back to libav/ffmpeg.
A while ago I tested the Nero decoder by feeding it an E-AC3 test frame. Then I thought about adding support for the Nero 9 AC3 decoder, which however can't decode E-AC3, anymore, but only AC3. So I changed the Nero decoder test to decode an AC3 test frame. But in the meanwhile I've decided that adding Nero 9 decoder support doesn't make much sense without E-AC3 support, so I recently switched back to testing E-AC3 decoding instead of AC3 decoding. I don't remember exactly in which version I changed that (it's not in the changelog, I think), but it could have been v3.06. So my best guess would be that AC3 decoding works on your PC, but E-AC3 decoding does not produce the expected results. Don't ask me why, though. As I said, on my PC it works just fine.

[madshi]

eac3to v3.10 released

http://madshi.net/eac3to.zip

* Blu-Ray title listing now includes chapter information
* fixed: v3.09 didn't show track languages for Blu-Rays

[Snowknight26]

* fixed: v3.09 didn't show track languages for Blu-Rays

Still doesn't show them for mkv files.

[madshi]

Still doesn't show them for mkv files.
Does for me. Does the MKV file you're testing actually have language information in it? :)

[Snowknight26]

Microsoft Windows [Version 6.0.6001]
Copyright (c) 2006 Microsoft Corporation. All rights reserved.

C:\unzipped\eac3to>eac3to.exe "Z:\Movies\Chungking Express\Chung Hing sam lam.mkv" -logmkv
EBML
DocTypeId: "matroska"
DocTypeVersion: 1
DocTypeReadVersion: 1
Segment
Meta Seek Information
Seek
SeekID: (binary, len: 4)
SeekPosition: 4099
Seek
SeekID: (binary, len: 4)
SeekPosition: 4249
Seek
SeekID: (binary, len: 4)
SeekPosition: 13110584910
Seek
SeekID: (binary, len: 4)
SeekPosition: 13110552269
Void: (binary, len: 4025)
Segment Information
TimecodeScale: 1000000
MuxingApp: "libebml v0.7.7 + libmatroska v0.8.1"
WritingApp: "mkvmerge v2.4.2 ('Oh My God') built on Jan 18 2009 17:30:28"
Duration: 1:42:56.160
DateUTC: 2/2/2009 3:48:18 AM
SegmentUID: (binary, len: 16)
Track
Track Entry
TrackNumber: 1
TrackUID: 1
TrackType: video
FlagEnabled: 1
FlagDefault: 1
FlagForced: 0
FlagLacing: 0
MinCache: 1
TrackTimecodeScale: 1
MaxBlockAdditionID: 0
CodecID: "V_MPEG4/ISO/AVC"
CodecDecodeAll: 1
CodecPrivate: (binary, len: 41)
DefaultDuration: 41708299
Language: "chi"
Video
PixelWidth: 1800
PixelHeight: 1080
FlagInterlaced: 0
DisplayWidth: 5
DisplayHeight: 3
Track Entry
TrackNumber: 2
TrackUID: 707616952
TrackType: audio
FlagEnabled: 1
FlagDefault: 1
FlagForced: 0
FlagLacing: 1
MinCache: 0
TrackTimecodeScale: 1
MaxBlockAdditionID: 0
CodecID: "A_DTS"
CodecDecodeAll: 1
Language: "chi"
Audio
SamplingFrequency: 48000
Channels: 6
Track Entry
TrackNumber: 3
TrackUID: 4182481851
TrackType: subtitle
FlagEnabled: 1
FlagDefault: 1
FlagForced: 0
FlagLacing: 0
MinCache: 0
TrackTimecodeScale: 1
MaxBlockAdditionID: 0
CodecID: "S_TEXT/ASS"
CodecDecodeAll: 1
CodecPrivate: (binary, len: 912)
Language: "eng"
Void: (binary, len: 1024)
Cluster
MKV, 1 video track, 1 audio track, 1 subtitle track, 1:42:56, 24p /1.001
1: h264/AVC, Chinese, 1800x1080 23.976p
2: DTS, Chinese, 5.1 channels, 24 bits, 1509kbps, 48khz
3: Subtitle (ASS)
v01 The video bitstream is encoded in a non-standard framerate.

C:\unzipped\eac3to>

It does.

[itsancho]

1st is 3.10, 2nd - 3.08 C:\Users\iTSAN>z\eac3to "L:\Firefly (2002) (TV series 2002-2003) The Complete Se
ries Blu-ray AVC dts-HD MA [107]\13 Heart of Gold.mkv"
MKV, 1 video track, 1 audio track, 2 subtitle tracks, 0:42:28, 24p /1.001
1: h264/AVC, English, 1080p24 /1.001 (16:9)
"AVC 18.9"
2: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
"dts-HD MA 5.1 24-Bit 3648"
3: Subtitle (SRT)
4: Subtitle (SRT)

C:\Users\iTSAN>zz\eac3to "L:\Firefly (2002) (TV series 2002-2003) The Complete S
eries Blu-ray AVC dts-HD MA [107]\13 Heart of Gold.mkv"
MKV, 1 video track, 1 audio track, 2 subtitle tracks, 0:42:28, 24p /1.001
1: h264/AVC, English, 1080p24 /1.001 (16:9)
"AVC 18.9"
2: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
"dts-HD MA 5.1 24-Bit 3648"
3: TEXT/UTF8, Bulgarian
4: TEXT/UTF8, English
Bitstream parsing for tracks 3 and 4 failed.
Demuxing these tracks may still produce correct results - or not.

[madshi]

eac3to.exe "Z:\Movies\Chungking Express\Chung Hing sam lam.mkv" -logmkv
Ouch. Proven wrong by my own debug switch... :o

Video and audio track languages did work, but subtitle track languages not.

[madshi]

eac3to v3.11

http://madshi.net/eac3to.zip

* fixed: MKV subtitle track language wasn't shown

<sigh> Hopefully this will be the last build for this week. Please don't find any more bugs, guys... :p

[itsancho]

thank u, madshi! great support, quick response! thanks again!

[DrNein]

A while ago I tested the Nero decoder by feeding it an E-AC3 test frame. Then I thought about adding support for the Nero 9 AC3 decoder, which however can't decode E-AC3, anymore, but only AC3. So I changed the Nero decoder test to decode an AC3 test frame. But in the meanwhile I've decided that adding Nero 9 decoder support doesn't make much sense without E-AC3 support, so I recently switched back to testing E-AC3 decoding instead of AC3 decoding. I don't remember exactly in which version I changed that (it's not in the changelog, I think), but it could have been v3.06. So my best guess would be that AC3 decoding works on your PC, but E-AC3 decoding does not produce the expected results. Don't ask me why, though. As I said, on my PC it works just fine.

Okay, I will not ask you why... but can I ask you what dependencies NeAudio2.ax has for E-AC3 decoding other than NeEacDec.dll? ;)

Even with the complete Audio Plugins, DSFilter, and Lib folders restored, it is still not working with eac3to 3.06-3.11. Perhaps another DSFilter ax needs registering or there is some other registry entry missing?

I see what you mean about that period where it behaved differently: 2.87 is the oldest version I still have which works on my system (so, presumably at least 2.87-3.05) while 2.80 does not.

[73ChargerFan]

All files. I'm mostly running eac3to over a gigabit network, but it happens even when running it directly to/from an incredibly fast RAID. As soon as I type any command, eac3to sits there for several seconds before even showing the progress bar.
Just a thought, but check to see if your antivirus is scanning the m2ts or other audio/video files.

I had to exclude file extensions, my media drives and the working directories I use for video conversions/muxing.

[odin24]

Does eac3to still detect forced subs on BDs? If so it didn't on Iron Man (North American).

Thanks,

[Snowknight26]

I had to exclude file extensions, my media drives and the working directories I use for video conversions/muxing.

I have nothing aside from vital OS processes/services running on the machine that exhibited the behavior. But oh well, it seems to have gone away for the most part with 3.11.

[kypec]

Hi madshi, I tried that -simple switch as you suggested. It works, however, I found out that the WAV files produced with that option are not bit-to-bit identical (data chunk wise). How is that possible? Here are my sample files (http://www.mediafire.com/?sharekey=8b5e96c28ff29ad7e62ea590dc5e5dbbe04e75f6e8ebb871), which I used for the test. I'd really appreciate if somebody else could test it also and post his report on different/identical AC3->WAV results.
Command lines invoked ->
Modern WAV output (WAVE_FORMAT_EXTENSIBLE) T:\eac3to\eac3to.exe "test.VOB" "R:\test.wav" -down16 -libav
Compatible WAV output (old style PCM header)T:\eac3to\eac3to.exe "test.VOB" "R:\simple.wav" -down16 -libav -simple
In the following pictures are highlighted just a few discrepancies between those two output files.
http://preview.shareapic.net/preview4/014695611.png (http://www.shareapic.net/content.php?id=14695611&owner=kypec) http://preview.shareapic.net/preview4/014695617.png (http://www.shareapic.net/content.php?id=14695617&owner=kypec)
In general, they differ a LOT, the samples do not match although they should because they were produced from the same source, by using the same tool and same decoder library. How come they're not identical is beyond my comprehension.:confused:
When I imagine all the guys who struggle to get LOSSLESS audio from their TrueHD/DTS-MA and so on while there seem to be weird issues even with ordinary stereo AC3...:eek:

[madshi]

Okay, I will not ask you why... but can I ask you what dependencies NeAudio2.ax has for E-AC3 decoding other than NeEacDec.dll? ;)

Even with the complete Audio Plugins, DSFilter, and Lib folders restored, it is still not working with eac3to 3.06-3.11. Perhaps another DSFilter ax needs registering or there is some other registry entry missing?
I don't really know. I installed full Nero 7, registered proper (actually bought by me) license codes for Nero 7 and for the Blu-Ray/HD DVD plugin, and Nero 7 decoding works just fine for me, without having to manually register any filters.

In the past people only had problems if they tried getting around installing full Nero 7 and/or if they tried to use stolen license keys. I can't - and don't want to - help doing either of that. Now of course I don't know if you have installed full Nero 7 and if you have proper license keys. If you haven't, please don't ask me for help. If you have, I'm sorry about the problem you have, but I don't really know how to help, since everything works just fine on my PC.

Does eac3to still detect forced subs on BDs? If so it didn't on Iron Man (North American).
I think it should still work. Not all Blu-Rays do have subs marked as "forced". Please also note that eac3to only knows after having processed the whole source file.

I tried that -simple switch as you suggested. It works, however, I found out that the WAV files produced with that option are not bit-to-bit identical (data chunk wise). How is that possible?
Because the "simple" header is shorter than the "extensible" one. So you simply cannot compare the files like you did. Your comparison program compared byte by byte without accounting for the different header sizes. Here's a test you can do:

Ask eac3to to convert the "-simple" wav file to extensible by doing "eac3to simple.wav full.wav". And ask eac3to to convert the extensible wav file to simple by doing "eac3to extensible.wav simplified.wav -simple". Then compare "simple.wav" with "simplified.wav" and compare "full.wav" with "extensible.wav" and you should notice that both comparisons are 100% bit identical.

When I imagine all the guys who struggle to get LOSSLESS audio from their TrueHD/DTS-MA and so on while there seem to be weird issues even with ordinary stereo AC3...:eek:
There are no weird issues. Your testing method is invalid.

[gigah72]

the languages are missing

3.08

c:\_sysprg\staxrip\eac3to>"c:\_sysprg\staxrip\eac3to\eac3to.exe" "C:\_data\_vd\_
Full\MATRIX_HDDVD" 1)
EVO, 2 video tracks, 8 audio tracks, 5 subtitle tracks, 2:16:17
"feature"
1: Joined EVO file
2: Chapters, 38 chapters with names
3: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
4: VC-1, 480p30 /1.001 (3:2), 66ms
5: E-AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
6: TrueHD, English, 5.1 channels, 48khz, dialnorm: -27dB
7: E-AC3, German, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
8: E-AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
"Commentary 1"
9: E-AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
"Commentary 2"
10: E-AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
"Commentary 3"
11: E-AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
"Commentary 4"
12: E-AC3, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, -3ms
13: Subtitle (DVD), English
14: Subtitle (DVD), German
15: Subtitle (DVD), German, "SDH"
16: Subtitle (DVD)
17: Subtitle (DVD)


3.11

c:\_sysprg\staxrip\eac3to>"c:\_sysprg\staxrip\eac3to\eac3to.exe" "C:\_data\_vd\_
Full\MATRIX_HDDVD" 1)
EVO, 2 video tracks, 8 audio tracks, 5 subtitle tracks, 2:16:17
1: Joined EVO file
2: Chapters, 38 chapters with names
3: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
4: VC-1, 480p30 /1.001 (3:2), 66ms
5: E-AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
6: TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
7: E-AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
8: E-AC3 Surround, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
9: E-AC3 Surround, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
10: E-AC3 Surround, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
11: E-AC3 Surround, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
12: E-AC3, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, -3ms
13: Subtitle (DVD)
14: Subtitle (DVD)
15: Subtitle (DVD)
16: Subtitle (DVD)
17: Subtitle (DVD)

[T800]

Can somebody else please try this? Can you reproduce the problem with the sample T800 uploaded?

@T800, you didn't replace any files in the eac3to folder, did you? E.g. libAften DLL or something like that? Please try deleting the whole eac3to folder and redownloading it, just to be safe... For me the samples you uploaded definitely convert just fine!


Just to let you know, v3.11 works fine for me now. :)

[kypec]

Because the "simple" header is shorter than the "extensible" one. So you simply cannot compare the files like you did. Your comparison program compared byte by byte without accounting for the different header sizes.
You apparently didn't look at my comparison results closely. I am aware of the difference in header part of the WAV files. I know that EXTENSIBLE header is longer (by 24 bytes precisely) and I have taken this difference into account for my binary comparison. The problem is that RAW SAMPLE DATA are different which I have clearly marked in the pictures.

Here's a test you can do:

Ask eac3to to convert the "-simple" wav file to extensible by doing "eac3to simple.wav full.wav". And ask eac3to to convert the extensible wav file to simple by doing "eac3to extensible.wav simplified.wav -simple". Then compare "simple.wav" with "simplified.wav" and compare "full.wav" with "extensible.wav" and you should notice that both comparisons are 100% bit identical.

There are no weird issues. Your testing method is invalid.
I did as you proposed and the findings were even more confusing now. I think there must be definitely something wrong with my system. I wrote a batch file in order to prove whether my suspicions are right:
T:\eac3to\eac3to.exe test.vob test1.wav -libav -down16
T:\eac3to\eac3to.exe test.vob test2.wav -libav -down16
T:\eac3to\eac3to.exe test.vob test3.wav -libav -down16
T:\eac3to\eac3to.exe test.vob test4.wav -libav -down16
T:\eac3to\eac3to.exe test.vob test5.wav -libav -down16
exit
Here comes the big surprise: NO two files of those 5 runs are 100% identical. You can download whole bunch of 5 outputs with MD5 checksums (http://www.mediafire.com/file/ayjwgngye1k/5tests.7z). Each of these 5 files is UNIQUE!:mad:
Can anybody else confirm eac3to produces consistent output with WAV files (using built-in libav only). You can use my test.VOB file (http://www.mediafire.com/?sharekey=8b5e96c28ff29ad7e62ea590dc5e5dbbe04e75f6e8ebb871) for your tests.
My system specs: Athlon 64 X2 4200+, Windows XP SP2 32-bit

Desperately waiting for any possible explanation of this abnormal behaviour...:thanks:

[honai]

It's not a bug, it's a feature.

madshi uses random noise dithering for downsampling (instead of a fixed noise pattern in previous 2.x or 1.x versions).

Actually, you have successfully proved that the random number generator of eac3to works correctly. :)

[kypec]

It's not a bug, it's a feature.

madshi uses random noise dithering for downsampling (instead of a fixed noise pattern in previous 2.x or 1.x versions).

Actually, you have successfully proved that the random number generator of eac3to works correctly. :)
Is there some switch to force eac3to NOT TO use random noise dithering? Or even better, to disable any dithering at all? I think basic mathematical rounding (64..24-bit -> 16-bit) would do just great for me. The problem is that documentation (on-screen usage help) is rather brief and lacking many undocumented options e.g. -simple. I wonder why madshi is hiding them from us?

I tried omit -down16 option as to avoid downconverting. Resulting WAVs (24-bit) are still not consistent. Isn't AC3 internally only 24-bit also? I thought lossy format means some info is lost during encoding, not generated randomly on each decoding.:confused:

[madshi]

the languages are missing

eac3to.exe "C:\_data\_vd\_
Full\MATRIX_HDDVD" 1)
Argh, now languages are working for Blu-Ray and MKV, but not for HD DVD. Will be fixed in the next build (probably next Sunday).

Is there some switch to force eac3to NOT TO use random noise dithering? Or even better, to disable any dithering at all? I think basic mathematical rounding (64..24-bit -> 16-bit) would do just great for me.
If you like audio artifacts, then yes, it would be great for you.

I tried omit -down16 option as to avoid downconverting. Resulting WAVs (24-bit) are still not consistent. Isn't AC3 internally only 24-bit also?
If you use libav for AC3 decoding, eac3to gets 32bit floating point audio data from libav. AC3 is not stored in integer samples, but it's stored in the time domain. If you want to get "consistent" (the name doesn't fit) decoding output, you can use the undocumented "-full" switch. That will give you floating point WAV files for libav AC3 decoding. Or you can use Nero or Sonic for AC3 decoding (if you have the decoders installed) which internally downsample to 24bit. Don't know which downsample algorithm they use, maybe they use a fixed noise pattern, in that case the output will be "consistent".

BTW, "consistent" is actually bad when talking about reducing audio bitdepth.

[buzzqw]

Hi madshi!

is possible to specify what track to extract ?

the -demux seems to extract all.. i would like to extract only audio, or subs, or specific stream/track

thanks!

BHH

[madshi]

is possible to specify what track to extract ?

the -demux seems to extract all.. i would like to extract only audio, or subs, or specific stream/track
Yes, of course. You have full control over what gets demuxed, transcoded, whatever. E.g. you can do:

eac3to source.m2ts 1: video.h264 2: English.ac3 3: French.ac3 8: English.sup
In order to find out which track has which track number, you need to do a track listing first by doing "eac3to source.m2ts".

This logic applies to all supported source formats.

[hubblec4]

@madshi

great thanks for the updates.

mkv-support is almost ready. i miss the support for subtitle vob.sub/idx.

can you realize that??

hubble

[buzzqw]

yup ! thansk madshi

(it's time to update the command line help... ;) )

BHH

[madshi]

mkv-support is almost ready. i miss the support for subtitle vob.sub/idx.

can you realize that??
It's on my to do list, along with chapters and attachments.

[kypec]

If you use libav for AC3 decoding, eac3to gets 32bit floating point audio data from libav. AC3 is not stored in integer samples, but it's stored in the time domain. If you want to get "consistent" (the name doesn't fit) decoding output, you can use the undocumented "-full" switch. That will give you floating point WAV files for libav AC3 decoding. Or you can use Nero or Sonic for AC3 decoding (if you have the decoders installed) which internally downsample to 24bit. Don't know which downsample algorithm they use, maybe they use a fixed noise pattern, in that case the output will be "consistent".

BTW, "consistent" is actually bad when talking about reducing audio bitdepth.
I see, after a bit of study in A/52 documentation about noise dither during decoding. Could you please elaborate why "consistency" (meaning to obtain identical results when decoding through same tools from the same source) is bad? I was so sure until today, that SAME BINARY SOURCE DATA->SAME PROCESSING ALGORITHM->SAME BINARY OUTPUT in the world of computing...well...I'll just have to live with that being not so true always:rolleyes:

[madshi]

I see, after a bit of study in A/52 documentation about noise dither during decoding. Could you please elaborate why "consistency" (meaning to obtain identical results when decoding through same tools from the same source) is bad? I was so sure until today, that SAME BINARY SOURCE DATA->SAME PROCESSING ALGORITHM->SAME BINARY OUTPUT in the world of computing...
The first law of audio processing is that when reducing bitdepth of digital audio data, you have to use dithering. Dithering means adding random noise to the data before rounding it down. The nature of "random" noise is that it's different every time. That's really all there is to it.

Ok, some algorithms use fixed noise patterns or calculate the noise the same way every time. With such algorithms you will get identical results if you repeat the same processing steps multiple times. But truely random noise is different every time. So being "inconsistent" is exactly the way to go - - but only when reducing audio bitdepth, of course.

[hubblec4]

It's on my to do list, along with chapters and attachments.

wow.

you are the best one.


hubble

[Snowknight26]

Couple things: time to upgrade to 3.11, -1536 is understood, and add -resampleTo48000.

[jj666]

...

Given the directory structure, that's a Rule #6 and shouldn't be discussed here.

-jj-

[Mixpower]

Given the directory structure, that's a Rule #6 and shouldn't be discussed here.

-jj-

Why ?
Because the ripped blu-ray disk is in my downloads dir ?

[jj666]

It's just some friendly advice, as the rules clearly state we don't discuss downloaded content. Once that is out of the way, it's a very friendly forum :-)

Cheers,

-jj-

[rica]

Dear mods.
Don't you think it is time to make this thread sticky?
Please check out replies and views in this thread and compare with the ex-threads which are already sticky.
Sorry in advance if i went too far.



_ _ _ _ _ _ ___

[DrNein]

I don't really know. I installed full Nero 7, registered proper (actually bought by me) license codes for Nero 7 and for the Blu-Ray/HD DVD plugin, and Nero 7 decoding works just fine for me, without having to manually register any filters.

Yeah, I was trying to be clever with a minimal install (Micro + any additional dll/ax restored). For now I have installed ShowTime (along with Burning ROM which I do want) and it "works fine". Of course there are superfluous filters and all that Nero Home & Indexing nonsense too which while offensive on priniciple I would not be obsessive about taking space (once disabled) except that I want to limit the backup image size to one DVD-5. I suppose I can instead just start eliminating unneeded files (move or archive) until something breaks ;)

Thanks again for your help and great proggy.

[mrr19121970]

Apparently the language on HD-DVD is no longer coming through with v3.11

http://forum.slysoft.com/showpost.php?p=176152&postcount=246

eac3to v3.11
command line: "C:\Program Files\eac3to\eac3to.exe" "E:\" 2) 2: "C:\temp\Evan Almighty\Chapters_2.txt" 3: "C:\temp\Evan Almighty\Video_3.*" 5: "C:\temp\Evan Almighty\Audio_5_Undetermined.AC3" -LOG="C:\temp\Evan Almighty\eac3to_PASS3_LOG.LOG"
------------------------------------------------------------------------------
EVO, 2 video tracks, 4 audio tracks, 2 subtitle tracks, 1:35:41
1: Joined EVO file
2: Chapters, 20 chapters with names
3: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
4: VC-1, 480p30 /1.001 (3:2)
5: E-AC3, 5.1 channels, 1536kbps, 48khz, dialnorm: -27dB
6: E-AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
7: AC3 Surround, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, -8ms
8: E-AC3, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, -27ms
9: Subtitle (DVD)
10: Subtitle (DVD)
Creating file "C:\temp\Evan Almighty\Chapters_2.txt"...
Disabling DRC for Nero (E-)AC3 decoding...
[v03] Extracting video track number 3...
[a05] Extracting audio track number 5...
[v03] Writing new framerate "24fps /1.001" to bitstream.
[a05] Removing E-AC3 dialog normalization...
[a05] Decoding with DirectShow (Nero Audio Decoder 2)...
[v03] Removing VC-1 pulldown...
[a05] DirectShow reports 5.1 channels, 24 bits, 48khz
[a05] Encoding AC3 <640kbps> with libAften...
[v03] Creating file "C:\temp\Evan Almighty\Video_3.vc1"...
[a05] Creating file "C:\temp\Evan Almighty\Audio_5_Undetermined.AC3"...
Video track 3 contains 137641 frames.
Video track 4 contains 162247 frames.
eac3to processing took 14 minutes, 4 seconds.
Done.

[Kurtnoise]

you should read previous posts instead of posting this...

[Snowknight26]

Is eac3to success.wav and eac3to error.wav supposed to fail detecting the file type? :p

[dorati]

@madshi:

Is it possible to generate the Filename of the Streamnumber with 2 Digits?? (00001 - 02 - VC-1, 1080p24.vc1 instead of 00001 - 2 - VC-1, 1080p24.vc1) to have the right order under windows?
Is it possible to skip the last Chapter, when the Last Chapter is = Movie-Length?
And is it possible to have one option to create the Chapters.txt in a format to direkt use in TsMuxer? ... Like this:


00:00:00.000
00:02:02.039
00:05:26.493
00:07:58.144
00:12:35.380
00:17:50.695
00:21:31.916
00:25:37.494
00:29:09.873
00:32:23.400
00:34:54.759
00:37:54.314
00:40:44.150
00:45:51.540
00:52:04.663
00:56:51.199
00:59:41.787
01:05:12.617
01:08:16.217
01:12:34.559
01:17:09.792
01:20:39.084
01:24:14.925
01:27:08.390


thx for this Great Tool !!!!

[laserfan]

@madshi:

Is it possible to skip the last Chapter, when the Last Chapter is = Movie-Length?
And is it possible to have one option to create the Chapters.txt in a format to direkt use in TsMuxer? ... Like this:


00:00:00.000
00:02:02.039
...Actually txMuxeR needs semicolon-delimited timecodes. You should check-out QPFile Generator discussed here (http://forum.doom9.org/showthread.php?p=1240718#post1240718) which creates --custom-chapters list as well as qpfile.txt, celltimes, etc.

[dorati]

But when paste into the window, the format I use above is right !

[laserfan]

Ah, without looking at your (pending) attachment let me guess--you are using the tsMuxeR GUI. Still, qpfile gen also outputs a timecodes file just like you want.

[mrr19121970]

And is it possible to have one option to create the Chapters.txt in a format to direkt use in TsMuxer? ... Like this:


you could use the GUI Clown_BD, it does what you're asking Madshi to do for you...

http://forum.slysoft.com/showthread.php?t=25818

[nwg]

you could use the GUI Clown_BD, it does what you're asking Madshi to do for you...

http://forum.slysoft.com/showthread.php?t=25818

That is very good software. I prefer using it to the TsMuxer GUI as it also reduces the number of steps when converting audio.

[chairlgmc]

Hello,

having been a silent watcher for several weeks now and still on a steep learning curve and having read posts hither and yon I need someone to confirm the following for me before I commit myself to command line hell.

I want to either extract the dts Core of a dts HD track and presumably get 24 bit/48khz (or 96khz on the rare occasions they exist) 5.1 soundtrack or convert DTS HD into DTS 24/96. The reason for this is that I want to continue to use my much loved AV32R processor whch will handle DTS 24/96 via spdif but has no hdmi input to transfer DTSHD. this is because it A sounds fantastic an B has very good room equalisation software that is not available via the 5.1 bypass.

Goals -

1 - have 24bit/48khz 5.1 dts output via spdif. (presume yes if core is encoded as such)
2 - have the above tagged as DTS 24/96 if necessary to kick the AV32 into life.(what i am unsure of)
3 - Core is ok if it will achieve 2 above. So far my tag has not lit up to say 24/96 - just vanilla DTS

6.1 and 7.1 not needed.

I know i can extract the core from a DTSHD BluRay - the confirmation i cannot find is whether it will identify as DTS 24/96 via spdif.

Will playback via MPC HC/ffdshow and auzentech digital out.

AV32R will decode a 24/96 stereo flac file (presumably output as linear pcm) no problem. Is fine with dts es too. spdif cant handle DTSHD but i am given to understand elsewhere that it can cope with 24/96.

Sorry to be a quality junkie and somewhat exacting.

Am loving my decoded HDCD's. Thank You.

Anthony

[TinTime]

The core can be 96/24, but only if the full track is DTS-MA with a sample rate of at least 96Khz. I haven't seen any yet.

So extracting the core with eac3to (assuming it gets the 96/24 stream and not the plain core?) is your best bet for DTS if you want it passed over spdif.

Do you know for sure that you have a blu-ray with DTS 96/24 in it's audio?

[GZZ]

I get the following warning in eac3to, not sure what it means and if I should be concerned about it:

The error happen on the Nordic version of Troy.


eac3to v3.11
command line: eac3to "H:\" 1) 1:"d:\Encoding\Troy\Chapters.txt" 2:"d:\Encoding\Troy\Video.mkv" 4:"d:\Encoding\Troy\Audio.dts" -768 17:"d:\Encoding\Troy\SubDK.sup" 18:"d:\Encoding\Troy\SubFI.sup" 19:"d:\Encoding\Troy\SubNO.sup" 21:"d:\Encoding\Troy\SubSWE.sup"
------------------------------------------------------------------------------
M2TS, 1 video track, 6 audio tracks, 17 subtitle tracks, 3:16:03, 24p /1.001
1: Chapters, 46 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: AC3, English, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
4: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB)
5: AC3, French, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
6: AC3, German, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
7: AC3, Italian, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
8: AC3, Spanish, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
9: Subtitle (PGS), English
10: Subtitle (PGS), French
11: Subtitle (PGS), German
12: Subtitle (PGS), German
13: Subtitle (PGS), Italian
14: Subtitle (PGS), Italian
15: Subtitle (PGS), Spanish
16: Subtitle (PGS), Dutch
17: Subtitle (PGS), Danish
18: Subtitle (PGS), Finnish
19: Subtitle (PGS), Norwegian
20: Subtitle (PGS), Portuguese
21: Subtitle (PGS), Swedish
22: Subtitle (PGS), French
23: Subtitle (PGS), German
24: Subtitle (PGS), Italian
25: Subtitle (PGS), Spanish
Creating file "d:\Encoding\Troy\Chapters.txt"...
[v02] Extracting video track number 2...
[s19] Extracting subtitle track number 19...
[s18] Extracting subtitle track number 18...
[a04] Extracting audio track number 4...
[a04] Extracting TrueHD stream...
[s17] Extracting subtitle track number 17...
[s21] Extracting subtitle track number 21...
[a04] Removing TrueHD dialog normalization...
[a04] Decoding with libav/ffmpeg...
[a04] Writing WAVs...
[v02] Muxing video to Matroska...
[a04] Creating file "d:\Encoding\Troy\Audio.L.wav"...
[a04] Creating file "d:\Encoding\Troy\Audio.C.wav"...
[a04] Creating file "d:\Encoding\Troy\Audio.SL.wav"...
[a04] Creating file "d:\Encoding\Troy\Audio.SR.wav"...
[a04] Creating file "d:\Encoding\Troy\Audio.LFE.wav"...
[a04] Creating file "d:\Encoding\Troy\Audio.R.wav"...
[s19] Creating file "d:\Encoding\Troy\SubNO.sup"...
[s18] Creating file "d:\Encoding\Troy\SubFI.sup"...
[s17] Creating file "d:\Encoding\Troy\SubDK.sup"...
[s21] Creating file "d:\Encoding\Troy\SubSWE.sup"...
[a04] [libav] End of stream indicated <WARNING>
[a04] [libav] Lossless check failed - expected 0, calculated 24 <WARNING>
[a04] The original audio track has a constant bit depth of 16 bits.
Encoding DTS <768kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.23.0.
Surcode encoding successfully started. Please wait...
Closing Surcode...
Added fps value to MKV header.
Video track 2 contains 282025 frames.
Subtitle track 17 contains 1394 captions.
Subtitle track 18 contains 1397 captions.
Subtitle track 19 contains 1390 captions.
Subtitle track 21 contains 1027 captions.
eac3to processing took 1 hour, 23 minutes.
Surcode encoding took 39 minutes, 48 seconds.
Done.


/GZZ

[jj666]

Nothing to worry about, if you check a few pages back, same issue was already explained by Madshi.

Cheers,

-jj-

[kurt]

D:\Movies>eac3to test.mkv
MKV, 1 video track, 1 audio track, 0:05:22, 24p
1: h264/AVC, 1280x544 23.999p
2: AC3, 5.1 channels, 640kbps, 48khz
v01 The video bitstream is encoded in a non-standard framerate.
v01 The video bitstream framerate field doesn't match the container framerate.

I got a problem with this file on the Popcorn Hour. Autoframerate isn't working, PCH uses 23,976 (or ,999) instead of 24,000 hz as output. I thought it might be a muxing problem.

How can I distinguish between container and bitstream framerate? which tools can change the framerate to 24.000?

I tried demuxing the elementary stream with eac3to and loading it in mkvmerge (while specifying fps to 24), but eac3to reports the same message of the resulting mkv as before....

[SIZE="1"]General
Complete name : D:\Movies\test.mkv
Format : Matroska
File size : 197 MiB
Duration : 5mn 22s
Overall bit rate : 5 122 Kbps
Encoded date : UTC 2009-02-19 17:21:37
Writing application : mkvmerge v2.4.2 ('Oh My God') built on Jan 18 2009 17:30:28
Writing library : libebml v0.7.7 + libmatroska v0.8.1

Video
ID : 1
Format : AVC
Format/Info : Advanced Video Codec
Format profile : High@L3.1
Format settings, CABAC : Yes
Format settings, ReFrames : 5 frames
Muxing mode : Container profile=Unknown@3.1
Codec ID : V_MPEG4/ISO/AVC
Duration : 5mn 22s
Bit rate : 4 271 Kbps
Nominal bit rate : 4 392 Kbps
Width : 1 280 pixels
Height : 544 pixels
Display aspect ratio : 2.35
Frame rate : 24.000 fps
Original frame rate : 23.999 fps
Resolution : 24 bits
Colorimetry : 4:2:0
Scan type : Progressive
Bits/(Pixel*Frame) : 0.256
Writing library : x264 core 64 r985 e256ea1



Edit: I found the solution. A remux in mkvmerge with a timecode file (24,000 fps) solved the autoframerate problem :)

[MuteyM]

@Jeff Flowerday and others who have requested the ability to split eac3to output by chapter for concerts and such, I have whipped up a little GUI to do just that!

After selecting the disc to split, it lets you choose which audio stream to use and which chapters you want to output. You can also edit the metadata and chapter names. eac3to is then called to demux the disc, and HDCR then splits it into individual FLAC files. WMA9 Pro 5.1 is also supported for output, for those that have a receiver that can process it (Windows Media Encoder must be installed for this to work.)

Currently HDCR only supports HD DVD, but Blu-Ray support is next on my to-do list.

Download it at:
http://www.megaupload.com/?d=45O7LO1S

Or see the official thread (http://forum.doom9.org/showthread.php?t=145140)

[Jeff Flowerday]

@Jeff Flowerday and others who have requested the ability to split eac3to output by chapter for concerts and such, I have whipped up a little GUI to do just that!

After selecting the disc to split, it lets you choose which audio stream to use and which chapters you want to output. You can also edit the metadata and chapter names. eac3to is then called to demux the disc, and HDCR then splits it into individual FLAC files. WMA9 Pro 5.1 is also supported for output, for those that have a receiver that can process it (Windows Media Encoder must be installed for this to work.)

Currently HDCR only supports HD DVD, but Blu-Ray support is next on my to-do list.

Download it at:
http://www.megaupload.com/?d=45O7LO1S

Or see the official thread (http://forum.doom9.org/showthread.php?t=145140)

Thanks, I already wrote myself a chapter to cue sheet converter so I'm fine using foobar2000.

[mrr19121970]

@Madshi

I though that eac3to had died on me:

eac3to.exe "D:\"

just the black DOS box, no progress bar or 'working on it message'. It turns out that Swiderwick Cronicles has 284 playlists, hence the delay:

CLIPINFO & PLAYLIST (http://clownbd.techxt.com/Downloads/Spiderwick Cronicles - BDMV.zip)

is there any chance of adding the '----' progress bar here too ?

Thanks.

[Kythe]

Madshi,
First of all, thank you for a superb product.

Second, I'm repeatedly getting a rather odd error that I haven't gotten in the past. The log file ends:

[a04] Extracting audio track number 4...
[a04] Extracting TrueHD stream...
[a04] Removing TrueHD dialog normalization...
[a04] Decoding with libav/ffmpeg...
[a04] Encoding FLAC with libFlac...
[a04] Creating file "c:\proc\tlb\tlb.flac"...
[a04] [libav] Lossless check failed - expected 0, calculated 84 <WARNING>
[a04] The original audio track has a constant bit depth of 16 bits.
[a04] Superfluous zero bytes detected, will be stripped in 2nd pass.
[a04] Starting 2nd pass...
[a04] Decoding FLAC...
[a04] Reducing depth from 24 to 16 bits...
[a04] Encoding FLAC with libFlac...
[a04] Creating file "c:\proc\tlb\tlb.flac"...
[a04] The FLAC decoder thread seems to hang. <ERROR>
Aborted at file position 1164103859. <ERROR>

This is an attempt to extract a TrueHD from a blu-ray disk image, and put it into a flac container.
What can cause this, and how might it be solved? Could there be limitations in the hardware (the memory may be a bit low on the processing computer)?

[gigah72]

i'm trying to get a hd 23.976 movie to 24.000fps. with speedup it becomes 25fps. how to get it to exact 24fps, so i can mux it into m2ts with 24fps?

[Snowknight26]

-changeTo24.000

[madshi]

Is eac3to success.wav and eac3to error.wav supposed to fail detecting the file type? :p
Oooops! :)

Is it possible to generate the Filename of the Streamnumber with 2 Digits?? (00001 - 02 - VC-1, 1080p24.vc1 instead of 00001 - 2 - VC-1, 1080p24.vc1) to have the right order under windows?
XP and newer OSs automatically sort the tracks correctly even with only 1 digit. Are you using an older OS than XP?

Is it possible to skip the last Chapter, when the Last Chapter is = Movie-Length?
Yes, that should be possible.

And is it possible to have one option to create the Chapters.txt in a format to direkt use in TsMuxer?
Possible, yes. But I don't want to do it. I don't like the idea of adding a multitude of different switches for a multitude of different programs out there. tsMuxeR has been requested, so has foobar2000. Maybe there are more. I'd have to add one switch for any such application. I don't like that idea...

1 - have 24bit/48khz 5.1 dts output via spdif. (presume yes if core is encoded as such)
2 - have the above tagged as DTS 24/96 if necessary to kick the AV32 into life.(what i am unsure of)
3 - Core is ok if it will achieve 2 above. So far my tag has not lit up to say 24/96 - just vanilla DTS
It doesn't make any sense to flag the DTS 96/24, if it isn't really encoded that way. Actually it would play incorrectly, if we did.

If the original DTS-HD track is 96khz then the core may be 96khz or 48khz. So extracting the core may be what you need - or not. Depending on how the studio encoded the track. eac3to lists both the core and DTS-HD track properties, so you always see whether the core is 96khz or not.

D:\Movies>eac3to test.mkv
MKV, 1 video track, 1 audio track, 0:05:22, 24p
1: h264/AVC, 1280x544 23.999p
2: AC3, 5.1 channels, 640kbps, 48khz
v01 The video bitstream is encoded in a non-standard framerate.
v01 The video bitstream framerate field doesn't match the container framerate.
[...]

Edit: I found the solution. A remux in mkvmerge with a timecode file (24,000 fps) solved the autoframerate problem :)
Maybe that solved the specific playback problem you had. But it will not modify the video bitstream from 23.999p to 24.000p. If you want to do that, you can use eac3to's "-changeTo24.000" option.

@Jeff Flowerday and others who have requested the ability to split eac3to output by chapter for concerts and such, I have whipped up a little GUI to do just that!

After selecting the disc to split, it lets you choose which audio stream to use and which chapters you want to output. You can also edit the metadata and chapter names. eac3to is then called to demux the disc, and HDCR then splits it into individual FLAC files. WMA9 Pro 5.1 is also supported for output, for those that have a receiver that can process it (Windows Media Encoder must be installed for this to work.)
Thanks!

I though that eac3to had died on me:

eac3to.exe "D:\"

just the black DOS box, no progress bar or 'working on it message'. It turns out that Swiderwick Cronicles has 284 playlists, hence the delay

CLIPINFO & PLAYLIST (http://clownbd.techxt.com/Downloads/Spiderwick Cronicles - BDMV.zip)

is there any chance of adding the '----' progress bar here too ?
Yeah, should be possible...

Second, I'm repeatedly getting a rather odd error that I haven't gotten in the past.
That's a strange error. To be honest, I've no idea what causes it. I suppose your harddisk has enough space left? And your CPU is not overheated?

Does anybody else have this problem?

[ron spencer]

is it possible to generate a chapter list like this:

00:01:00.111
00:02:03.115

etc. without the chapter names, etc.

This is useful for us converting to dvd as well.

Not sure if can be done or not...jsut asking.

[madshi]

is it possible to generate a chapter list like this:

00:01:00.111
00:02:03.115
Not to be harsh, but I've replied to the very same question just one post above yours. You may really want to at least browse through the last 3 pages before asking for a feature...

[kurt]

Maybe that solved the specific playback problem you had. But it will not modify the video bitstream from 23.999p to 24.000p. If you want to do that, you can use eac3to's "-changeTo24.000" option.

thx, will keep that in mind for the next strange muxed video... :)

[TinTime]

XP and newer OSs automatically sort the tracks correctly even with only 1 digit. Are you using an older OS than XP?

Only explorer sorts them logically for humans - I can imagine the calls MS support would get if it didn't! Batch commands such as DIR and FOR will sort them in the strictly correct order though (e.g. 10 before 9).

As far as I'm concerned I've got a step in my demux batch script to zero pad any single digit filenames after the eac3to demux so I'm not bothered either way about this.

[monohouse]

-----

[madshi]

is there a way or option to disable 2nd pass ?
Which kind of 2nd pass? For removing clipping? For fixing audio gaps/overlaps? Something else? I don't really see why you would want to skip any of that. IMHO it should all be done.

[monohouse]

-----

[six172]

Hello everyone, I have been lurking here for several weeks and trying to digest all the information you guys have given me.

Madshi, thanks so much for a wonderful tool this has helped out so much.

I have been able to backup several of my HD-DVD movies and move them into MKV for playback on my Western Digital My TV, which I love. All of the discs with DD+ I am able to convert over to DTS and they play fine with very few errors. Although I am running into a problem, I do have 5 or so discs with Dolby TrueHD on them and I want to get those moved over to my WD as well. I have tried converting the audio to RAW, FLAC, PCM, and LPCM with no success. Every time I do I just get a Dolby Digital sound not the TrueHD sound that I am looking for. I have a Pioneer VSX-92 that supports all of the formats and would be able to decode the audio if only I could get it sent over. I am not concerned with space, since it's only 5 movies so 50 gigs per movie is not a big deal. I just want to know if it can be done.

Currently I am backing the movies up with AnyDVDHD, then using EVOdemux to join the two EVO files, and remove the TrueHD track. I then open Eac3to and change the audio over to whatever format (PCM,LPCM,FLAC) then I use either TSmuxer to combine the files to make either an M2ts or a TS file, although I get an error in TSmuxer that says it "Can't detect stream type." I also use Ripbot 264 to create an MKV files, which like I said play fine, but it's not TrueHD.

So my question is really, has anyone successfully been able to convert the audio and video into a container like TS or MKV?

I appreciate your responses and I hope you will forgive my Newbiness.

[gigah72]

...my HD-DVD movies and move them into MKV for playback on my Western Digital My TV, which I love. All of the discs with DD+ I am able to convert over to DTS and they play fine with very few errors. Although I am running into a problem, I do have 5 or so discs with Dolby TrueHD on them and I want to get those moved over to my WD as well. I have tried converting the audio to RAW, FLAC, PCM, and LPCM with no success. Every time I do I just get a Dolby Digital sound not the TrueHD sound that I am looking for. I have a Pioneer VSX-92 that supports all of the formats and would be able to decode the audio if only I could get it sent over. I am not concerned with space, since it's only 5 movies so 50 gigs per movie is not a big deal. I just want to know if it can be done.



from WD i know only the HD TV: http://support.wdc.com/product/download.asp?groupid=1001&lang=en

it's HDMI 1.2, no passthrough of THD & and other new stuff, so the problem is not with eac3to or your pioneer.

[madshi]

unless it shouldn't, clipping removal changes the signal when in reality clipping only matters when you can actually hear it, if you can't then it isn't clipping (not to mention clipping removal is a subjective thing (when performed in a digital fashion), and usually belongs to the sound mastering department (meaning there are plugins [VST etc] which are better equipped and are more suited for this kind of work), but if you are changing the signal you will always hear the changes because the changes are digital (see that's the thing, im still not quite sure how you define eac3to to be, is it an mastering tool ? (because it does perform dither and such) or is it a converter ?, or you still haven't decided on borders for the program ?)
I'm sorry to say, but you don't really seem to know what you're talking about. There are 2 totally different situations:

(1) If you have floating point data where some samples have a peak which is out of spec, converting this audio data to PCM would result in clipping. Solving this problem is extremely easy: It's just lowering the volume of the whole track a little to make sure that no audio samples have too high peaks. This is an absolutely lossless process (if done in floating point). The signal is not changed (just the volume is changed), it is not a subjective thing, and nobody will hear the difference (except for the volume difference and the solved clipping artifacts, obviously). And no VST plugin can do that any better than eac3to. I would call this process "clipping prevention".

(2) If you have a PCM track which was converted without "clipping prevention", then some VST plugins can detect that and can try to undo the damage. This is a complicated and error prone process which is difficult to do and really changes the signal. I'd call this "clipping repair". eac3to doesn't even try to do that.

So again: There is no reason whatsoever to disable eac3to's clipping prevention system. It has no disadvantages (I know of). Except longer processing time, of course...

And dithering: If eac3to is supposed to output PCM tracks from any source then there's no other way than to use dithering. It's a necessary evil if you want to reduce audio bitdepth or if you want to convert floating point audio data to PCM. If I removed dithering, eac3to would lose half of its functionality.

the gap/overlap is sometimes wrong
Then send me a sample and I'll try to fix it.

[madshi]

@six172, you may want to ask on the tsMuxeR thread or on a "Western Digital My TV" thread, if there is any such. Can't help you out since this is not an eac3to related problem.

[mrr19121970]

@six172, you may want to ask on the tsMuxeR thread or on a "Western Digital My TV" thread, if there is any such. Can't help you out since this is not an eac3to related problem.

@sixt, you're doing a very long winded process with out dated software. drop EVOdemux, get tsMuxeR v1.8.18, wait for eac3to 3.12 (that fixes the HD-DVD stream language issue) and try again.

Please see here:
eacto, tsMuxer & ImgBurn Made Easy (http://forum.slysoft.com/showthread.php?t=25818)

for a very comfortable solution to what you're trying to acheive.

[halsboss]

Hi, a newbie seeking advice. Hoping to convert arbitrary HDTV .ac3 (could be any kbs, #channels) to 224k 2-channel .ac3 but I am not sure what the options do. So far, I have guessed
"C:\SOFTWARE\eac3to\eac3to.exe" "T80 3_2ch 448Kbps DELAY 0ms.ac3" "T80 3_2ch 448Kbps DELAY 0ms-eac3to.ac3" -224 -down2 -mixlfe -resampleTo48000 +6db
but am not sure what "-mixlfe" does. Or "-phaseshift". The ac3 always seems very quiet so I bumped it up by an arbitrary 6db.

Can someone please explain and confirm I'm doing the right thing ?

I gather DRC is a good thing since it lessens the difference between loud things such as bangs and normal speaking, so when increasing the volume so I can hear them speaking (I'm slightly deaf) it won't blow the family away when a bang occurs... is that right ? and if so, can we do DRC with eac3to ? (I only have the eac3to and no Nero etc).

[tebasuna51]

but am not sure what "-mixlfe" does. Or "-phaseshift". The ac3 always seems very quiet so I bumped it up by an arbitrary 6db.
-mixlfe is not recommended by Dolby because can produce artifacs, but you can make the two versions and compare.
-phaseshift is not recommended if source is ac3 because the surround channels are already phaseshifted, if you do other phaseshift the result was wrong.
-normalize is the parameter recommended after a downmix to obtain the max volume, instead the arbitrary +6dB

I gather DRC is a good thing since it lessens the difference between loud things such as bangs and normal speaking, so when increasing the volume so I can hear them speaking (I'm slightly deaf) it won't blow the family away when a bang occurs... is that right ? and if so, can we do DRC with eac3to ?
DRC is a good thing when play the audio, but use it to reencode is lose quality. If you play with PC you can use AC3Filter to add DRC on the fly. If you use receivers there are a 'night' mode to play.

Now eac3to don't decode with DRC activated and Aften can't do correct DRC management.

[GZZ]

I think there is a small bug in EAC3to way of parsing the mpls files and the m2ts files.

I did a test on Der Untergang (Nordic version)


just to see which mpls and m2ts files are used...

d:\eac3to>eac3to "H:\"
1) 00007.mpls, 00010.m2ts+00009.m2ts, 2:35:23
- Chapters, 12 chapters
- h264/AVC, 1080p24 (16:9)


------------------------------------

d:\eac3to>eac3to "H:\" 1)
M2TS, 1 video track, 2 audio tracks, 4 subtitle tracks, 2:35:23, 24p
1: Chapters, 12 chapters
2: h264/AVC, 1080p24 (16:9)
3: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
4: DTS Master Audio, 5.1 channels, 24 bits, 48khz, dialnorm: -4dB
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz, dialnorm: -4dB)
5: Subtitle (PGS)
6: Subtitle (PGS)
7: Subtitle (PGS)
8: Subtitle (PGS)
v02 The video framerate is correct, but rather unusual.

Here we dont have any names on the subtitles, there should be, because when I open the mpls files in text mode I see this:

MPLS0200 :
p 

2 
P00010M2TS

›üÀ
�\P .


 b   ‘deu °00009M2TS

›üÀœw³ Ž


 b
 �adeu

†adeu
 �dan

�nor
 �swe
 �swe   ‘deu $ 
00008M2TS 
›üÀ
œ ª 

›üÀÿÿ

Bóÿÿ

%iÿÿ

(*Õÿÿ

Gµ'ÿÿ

�.“ÿÿ


Þ6ûÿÿ

Ÿ3ÿÿ

SÙHÿÿ

;½/ÿÿ

†
‚ÿÿ

�>Íÿÿ

deu = Deutsch (German) (Audio)
dan = Danish (Subtitle)
nor = Norwegian (Subtitle)
swe = Swedish (Subtitle)
swe = Swedish (Subtitle) (Should have been Finnish, mastering error on the bluray, not eac3to fault)

But as you can see the mpls language/Subtitles are NOT always parsed and shown in eac3to. BUG!

---------------------------------------------
If I then try to parse the 2 m2ts files I also dont get any subtitle/language and I think its because eac3to only parse the first file begin 00010.m2ts which is a 1 sec long black screen video clip without audio/subtitles.

d:\eac3to>eac3to "H:\BDMV\STREAM\00010.m2ts"+"H:\BDMV\STREAM\00009.m2ts"
M2TS, 1 video track, 2 audio tracks, 4 subtitle tracks, 2:35:23, 24p
1: Chapters, 12 chapters
2: h264/AVC, 1080p24 (16:9)
3: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
4: DTS Master Audio, 5.1 channels, 24 bits, 48khz, dialnorm: -4dB
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz, dialnorm: -4dB)
5: Subtitle (PGS)
6: Subtitle (PGS)
7: Subtitle (PGS)
8: Subtitle (PGS)
v02 The video framerate is correct, but rather unusual.


---------------------------------------------
If I then skip the 00010.m2ts file and go for the 00009.m2ts which is the main video, it shows all the language and subtitles as indicated in the mpls file. GOOD!

d:\eac3to>eac3to "H:\BDMV\STREAM\00009.m2ts"
M2TS, 1 video track, 2 audio tracks, 4 subtitle tracks, 2:35:21, 24p
1: h264/AVC, 1080p24 (16:9)
2: AC3, German, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
3: DTS Master Audio, German, 5.1 channels, 24 bits, 48khz, dialnorm: -4dB
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz, dialnorm: -4dB)
4: Subtitle (PGS), Danish
5: Subtitle (PGS), Norwegian
6: Subtitle (PGS), Swedish
7: Subtitle (PGS), Swedish
v01 The video framerate is correct, but rather unusual.



There is also a problem with extracting video and audio from this using the mpls file. Because the audio goes out of sync because of the small 00010.m2ts black screen file. Giving you this when extracting:


d:\eac3to>eac3to "H:\" 1) 4:"D:\Audio.dts" -core
M2TS, 1 video track, 2 audio tracks, 4 subtitle tracks, 2:35:23, 24p
1: Chapters, 12 chapters
2: h264/AVC, 1080p24 (16:9)
3: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
4: DTS Master Audio, 5.1 channels, 24 bits, 48khz, dialnorm: -4dB
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz, dialnorm: -4dB)
5: Subtitle (PGS)
6: Subtitle (PGS)
7: Subtitle (PGS)
8: Subtitle (PGS)
v02 The video framerate is correct, but rather unusual.
a04 Extracting audio track number 4...
a04 0:00:00 Detected PTS break, increasing PTS by 41.7ms...
a04 Extracting DTS core...
a04 Removing DTS dialog normalization...
a04 Creating file "D:\Audio.dts"...


The audio is out of sync by 1-2 sec. The solution is to rip the files from the 00009.m2ts files (main movie) then the audio is in sync. Dont know if eac3to can make a work around for this.


Last a small request if possible. If you want to extract the chapters, is it then possible to do this without having eac3to to scan the video stream. As the chapters are located in the mpls file ?

[monohouse]

-----

[Chumbo]

yhe, you're right about 1) it makes alote of sense, I did know what I was talking about, but it was 2), dithering is right, but I didn't mention it for that reason.

so if three is no way to disable second pass, is the last version without 2nd pass still available ?
Here's the best place I know of that has older versions still available.

http://www.videohelp.com/tools/eac3to/old-versions#download

[mrr19121970]

@Madshi...

is this a bug (http://forum.doom9.org/showthread.php?t=125966) ? Processing a HD-DVD, giving an explicit name eac3to accepts it, writes it in the log:

command line: "C:\Program Files\eac3to\eac3to.exe" "E:\" 1) 2: "C:\Temp\Batman Begins\Chapters_2.txt" 3: "C:\Temp\Batman Begins\Video_3.*" 6: "C:\Temp\Batman Begins\Audio_6_Undetermined.THD+AC3" -LOG="C:\Temp\Batman Begins\eac3to_PASS3_LOG.LOG"


...

[a06] Creating file "C:\Temp\Batman Begins\Audio_6_Undetermined.THD+AC3"

but actually writes :

Audio_6_Undetermined DELAY 200ms.THD+AC3

[rica]

madshi,

can you compare these flac files created by eac3to?

http://www.sendspace.com/file/scv0kk

The method:
Extract mlp+decode to wav with DVD-Audio Explorer 2008.07.21 (Beta3) by MaximA and retransmit to flac via eac3to.
eac3to\eac3to C:\users\rica\desktop\tracs\E.wav C:\E\E_01.flac
------------------------------------------------------------------------------
WAV, 5.1 channels, 0:06:32, 24 bits, 13824kbps, 96khz
Reading WAV...
Encoding FLAC with libFlac...
Creating file "C:\E\E_01.flac"...
The original audio track has a constant bit depth of 24 bits.
eac3to processing took 1 minute, 30 seconds.
Done.
It is perfect.

_ _ _ _ _ _ _

http://www.sendspace.com/file/8snz93

The method:


Extract mlp directly with DVD_Audio Explorer and transmit to flac with eac3to.
eac3to\eac3to C:\users\rica\desktop\tracs\E_new.mlp C:\E\E_new.flac
------------------------------------------------------------------------------
MLP, 5.1 channels, 24 bits, 96khz
Decoding with libav/ffmpeg...
Encoding FLAC with libFlac...
Creating file "C:\E\E_new.flac"...
The original audio track has a constant bit depth of 24 bits.
eac3to processing took 2 minutes, 12 seconds.
Done.
It has problems; i think channel mapping issue.


_ _ _ _ _

EDIT: Here is the extracted original mlp (by method2):

http://www.sendspace.com/file/xj5698




_ _ _ _ _

[jamos]

Any progress with DTS-HD MA 7.1 support? N/M see I need the arcsoft decoder to get this..

[jfcarbel]

Added to Delphi Wiki page (http://delphi.wikia.com/wiki/Good_Quality_Applications_Built_With_Delphi#Lesser_Known_Delphi_Application_List) :)

[peterjcat]

but actually writes :

Audio_6_Undetermined DELAY 200ms.THD+AC3

It's not a bug, it's a feature! It's telling you, in a way that makes sure you don't forget it, that you need to delay that audio track 200ms when you are muxing it (eg using tsMuxeR). eac3to can't apply TrueHD delay itself so you have to delay the whole track when muxing, or it will be out of sync.

[peterjcat]

@sixt, you're doing a very long winded process with out dated software. drop EVOdemux, get tsMuxeR v1.8.18, wait for eac3to 3.12 (that fixes the HD-DVD stream language issue) and try again.

Please see here:
eacto, tsMuxer & ImgBurn Made Easy (http://forum.slysoft.com/showthread.php?t=25818)

for a very comfortable solution to what you're trying to acheive.

This is true, nobody uses EVOdemux anymore, eac3to will do everything faster and better. Keep in mind that you'll never get your WDTV to play TrueHD (or DTSMA) because, as mentioned, it doesn't have the right connections.

Your best bet would be to convert the TrueHD losslessly to an LPCM stream using eac3to (or one of the tools that use it, maybe the above, maybe ToNMT?). The WDTV should be able to support multichannel LPCM over HDMI. You say you've tried that but you don't say what exactly you've tried and what exactly's happening.

Of course, if you're not using eac3to, there are other places that should be able to help you better.

[Chumbo]

...Your best bet would be to convert the TrueHD losslessly to an LPCM stream...
I hear, no pun intended, that flac is a better lossless choice. Please correct me if I'm wrong.

[nwg]

It's not a bug, it's a feature! It's telling you, in a way that makes sure you don't forget it, that you need to delay that audio track 200ms when you are muxing it (eg using tsMuxeR). eac3to can't apply TrueHD delay itself so you have to delay the whole track when muxing, or it will be out of sync.

I have never added a delay and has been in sync everytime. As soon as tried adding the delay (one was 1001ms) the sync was way off.

[tebasuna51]

It has problems; i think channel mapping issue.

Yep, seems the remap needed is missing (or now libav is correct and don't need remap).
We have problems with 18,19 and 20 modes
# MLP Channel Assignments MS channels order libav remap
-- ------------------------ ------------------ -----------
18 Lf Rf Ls Rs (LFE) FL FR LF BL BR -0,1,4,2,3,5
19 Lf Rf Ls Rs (C) FL FR FC BL BR -0,1,4,2,3,5
20 Lf Rf Ls Rs (C LFE) FL FR FC LF BL BR -0,1,4,5,2,3

[jamos]

Any progress with DTS-HD MA 7.1 support? N/M see I need the arcsoft decoder to get this..

Ok got arcsoft decoder seemed to decode dts hd fine to pcm but when remuxed with tsmuxer channels are mixed. the right speaker is playing the center channel when playing pcm.

this may be a issue with PCM2tsmu not sure but i did use the -c 8 switch with it and tsmuxer does say 7.1.

[peterjcat]

I hear, no pun intended, that flac is a better lossless choice. Please correct me if I'm wrong.

FLAC is indeed a better (more efficient) choice IF you have a device that will play it back, such as a PC. The OP wanted to hear lossless audio on his WD TV and there's no way that's playing back multi-channel FLAC. However, it should play LPCM as there's no decoding involved. The recommendation was just for those circumstances.

[MuteyM]

@Jeff Flowerday and others who have requested the ability to split eac3to output by chapter for concerts and such, I have whipped up a little GUI to do just that!
...


Just a quick note that I've now added Blu-Ray support to HDConcertRipper. This is the last post I'll make about it in the eac3to thread, anyone interested can continue the discussion in the official thread (http://forum.doom9.org/showthread.php?t=145140).

[peterjcat]

I have never added a delay and has been in sync everytime. As soon as tried adding the delay (one was 1001ms) the sync was way off.

A delay specified in the filename like that, or just a delay reported when eac3to is running? eac3to fixes the latter delays, doesn't fix the former. And I guess sometimes it might put a delay in a filename when it doesn't need to, I'll let madshi speak to that, but believe me it's there for a reason.

[Snowknight26]

The OP wanted to hear lossless audio on his WD TV and there's no way that's playing back multi-channel FLAC.

Says it does here: http://www.wdc.com/en/products/products.asp?DriveID=572

[peterjcat]

Says it does here: http://www.wdc.com/en/products/products.asp?DriveID=572

No, it says it plays FLAC music. It doesn't support FLAC in movies, let alone multi-channel FLAC in movies, let alone decoding multi-channel FLAC in movies to LPCM for lossless output via HDMI.

[jamos]

Ok got arcsoft decoder seemed to decode dts hd fine to pcm but when remuxed with tsmuxer channels are mixed. the right speaker is playing the center channel when playing pcm.

this may be a issue with PCM2tsmu not sure but i did use the -c 8 switch with it and tsmuxer does say 7.1.

Hmm why does eac3to remap channels when using arcsoft decoder to get a pcm file? seems to be a bug in the channels for 7.1 as the center is playing out of right front.


eac3to v3.11
command line: "F:\bd\eac3to\eac3to.exe" "00001.m2ts" 3: "app.pcm"
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 3 subtitle tracks, 1:52:02, 24p /1.001
1: Chapters, 17 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 7.1 channels, 16 bits, 48khz
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48khz)
4: AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
5: AC3 Surround, English, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB
6: Subtitle (PGS), English
7: Subtitle (PGS), English
8: Subtitle (PGS), Spanish
[a03] Extracting audio track number 3...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Swapping endian...
[a03] Remapping channels...
[a03] Creating file "app.pcm"...
[a03] The original audio track has a constant bit depth of 16 bits.
Video track 2 contains 161161 frames.
eac3to processing took 16 minutes, 9 seconds.
Done.

then
Pcm2Tsmu.exe app.pcm output.pcm -i 16 -c 8

[Mark_A_W]

Hmm why does eac3to remap channels when using arcsoft decoder to get a pcm file? seems to be a bug in the channels for 7.1 as the center is playing out of right front.



Not for me - but I am converting to FLAC

As far as I can tell (as I have to downmix in madflac to 5.1 as I have a 5.1 soundcard/setup), the channels are correct.

For 7.1 soundtracks, I'm certainly not getting the centre coming out the right speaker.

[jamos]

Not for me - but I am converting to FLAC

As far as I can tell (as I have to downmix in madflac to 5.1 as I have a 5.1 soundcard/setup), the channels are correct.

For 7.1 soundtracks, I'm certainly not getting the centre coming out the right speaker.

ps3 has to use pcm when streaming does not support Flac afaik. are you using pcm2tsmu and sonic decoder also?

[yesgrey]

madshi,
does eac3to supports multiple mkv source files as it does with ts files?
For example eac3to source1.mkv+source2.mkv only gives the info of the first file, but if using .ts files gives the info of the sum of the files.

[madshi]

I think there is a small bug in EAC3to way of parsing the mpls files and the m2ts files.

I did a test on Der Untergang (Nordic version)

Here we dont have any names on the subtitles, there should be, because when I open the mpls files in text mode I see this [...]

But as you can see the mpls language/Subtitles are NOT always parsed and shown in eac3to. BUG!
eac3to reads the subtitle languages from the clipinfo file and not from the playlist file. It seems that the clipinfo file for 00009.m2ts does not contain the subtitle languages, for whatever strange reason...

There is also a problem with extracting video and audio from this using the mpls file. Because the audio goes out of sync because of the small 00010.m2ts black screen file.

The audio is out of sync by 1-2 sec. The solution is to rip the files from the 00009.m2ts files (main movie) then the audio is in sync. Dont know if eac3to can make a work around for this.
Would you mind uploading the CLIPINF + PLAYLIST folders together with 00010.m2ts and the first 30MB of 00009.m2ts? Having these files might help me fix the problem.

Last a small request if possible. If you want to extract the chapters, is it then possible to do this without having eac3to to scan the video stream. As the chapters are located in the mpls file ?
Technically no problem. But I don't know right now if I can easily fit that into the eac3to source code structure. Will have to check...

does eac3to supports multiple mkv source files as it does with ts files?
For example eac3to source1.mkv+source2.mkv only gives the info of the first file, but if using .ts files gives the info of the sum of the files.
This is not supported yet, at least not officially. Does it work at all (apart from the wrong runtime display, of course)? Haven't even tried myself yet.

Hmm why does eac3to remap channels when using arcsoft decoder to get a pcm file?
Because PCM channel ordering is different from WAV channel ordering.

seems to be a bug in the channels for 7.1 as the center is playing out of right front.
I can't reproduce any problems here. I even tried doing the same thing as you (eac3to + Pcm2Tsmu + tsMuxeR) and for me the channel order is always correct. I can only guess that there's something wrong in your playback chain...

The method:

Extract mlp directly with DVD_Audio Explorer and transmit to flac with eac3to.

It has problems; i think channel mapping issue.
Yep, seems the remap needed is missing (or now libav is correct and don't need remap).
We have problems with 18,19 and 20 modes
# MLP Channel Assignments MS channels order libav remap
-- ------------------------ ------------------ -----------
18 Lf Rf Ls Rs (LFE) FL FR LF BL BR -0,1,4,2,3,5
19 Lf Rf Ls Rs (C) FL FR FC BL BR -0,1,4,2,3,5
20 Lf Rf Ls Rs (C LFE) FL FR FC LF BL BR -0,1,4,5,2,3
Thank you. Bug found and fixed.

[yesgrey]

This is not supported yet, at least not officially. Does it work at all (apart from the wrong runtime display, of course)? Haven't even tried myself yet.
It works, but the results are not correct.

First, I have demuxed each file separatelly, muxed the raw streams with tsMuxer to .ts files, and then used eac3to to join the two .ts files to .mkv and .dts. It worked flawlessly, even correcting the audio overlapping.
Then I tryed with the two mkv files, but the result was not good. The mkv resulting file was exactly the same size of the file obtained with the .ts method, but I have compared both files and they were different. The dts audio file was only half, he considered the other audio part to be overlapping the first...

[madshi]

It works, but the results are not correct.

First, I have demuxed each file separatelly, muxed the raw streams with tsMuxer to .ts files, and then used eac3to to join the two .ts files to .mkv and .dts. It worked flawlessly, even correcting the audio overlapping.
Then I tryed with the two mkv files, but the result was not good. The mkv resulting file was exactly the same size of the file obtained with the .ts method, but I have compared both files and they were different. The dts audio file was only half, he considered the other audio part to be overlapping the first...
Ah, makes sense. For EVO and m2ts files, eac3to automatically adjust timecodes for multiple input files. For MKV it does not. So it's understandable that eac3to sees the audio as overlapping, because both MKV files have audio begin at runtime 0:00:00. I could only solve this properly if I actually implemented support for MKV file joining. But I'm not sure how big the demand is for such a feature? For TS and m2ts it's very common to have splitted files. But for MKV it should be rather rare?

[madshi]

eac3to v3.12 released

http://madshi.net/eac3to.zip

* fixed: track languages for HD DVD discs were not shown
* fixed: MLP channel order was wrong for some specific channel configurations
* fixed: "DirectShow reported 255 channels" happened sometimes
Haven't had time to work on eac3to last weekend, so here at least some bugfixes...

[jamos]

Because PCM channel ordering is different from WAV channel ordering.


I can't reproduce any problems here. I even tried doing the same thing as you (eac3to + Pcm2Tsmu + tsMuxeR) and for me the channel order is always correct. I can only guess that there's something wrong in your playback chain...


what version of tsmuxer did you use? Im wondering if its a bug with the 1.81 betas that have been recently released? I will try with 1.88. Is there a way I can cut a sample of the pcm audio file and send it to you? I only got this bug with a 7.1 source also..5.1 works fine. also could be sonic decoder too I am guessing.

[yesgrey]

I could only solve this properly if I actually implemented support for MKV file joining.
If it's too much work don't bother. The first method I used worked flawlessly, and it's just some more minutes of processing... maybe you should just indicate that multiple mkv source files are not supported.

[pihug12]

eac3to v3.12
command line: eac3to 1.h264+2.ac3+3.ac3 test.mkv
------------------------------------------------------------------------------
h264/AVC, 1280x528 23.975p
The video track has a non-standard framerate. <WARNING>
Muxing video to Matroska...
Added fps value to MKV header.
Video track contains 591 frames.
eac3to processing took 1 second.
Done.

eac3to doesn't seem to mux the audio and subtitles files in the MKV.
And two log files are made : "log.txt" and "test - Log.txt".

pihug12 :cool:

[madshi]

what version of tsmuxer did you use?
1.8.8b

Im wondering if its a bug with the 1.81 betas that have been recently released? I will try with 1.88. Is there a way I can cut a sample of the pcm audio file and send it to you? I only got this bug with a 7.1 source also..5.1 works fine. also could be sonic decoder too I am guessing.
I have no idea. But I'm not interested in doing any further bug hunting on my side because I don't think that this is a bug in eac3to. If you see any indication that eac3to is at fault, please let me know, though.

eac3to doesn't seem to mux the audio and subtitles files in the MKV.
And two log files are made : "log.txt" and "test - Log.txt".
eac3to only muxes video to MKV. And you should NOT do "some.h264+some.ac3"! File joining is only supported for files of the same type. If you do "some.h264+some.ac3", eac3to will mux the AC3 data as the last h264 video frame!

The "log.txt" file is created in the eac3to folder. The "outputFileName - Log.txt" is created in the target folder. This is by design.

[kurt]

new version (3.12) is ready to download? -test shows old one...

[pihug12]

eac3to only muxes video to MKV. And you should NOT do "some.h264+some.ac3"! File joining is only supported for files of the same type. If you do "some.h264+some.ac3", eac3to will mux the AC3 data as the last h264 video frame!
Did you plan a feature to mux audio tracks and subtitles files to MKV ?

The "log.txt" file is created in the eac3to folder. The "outputFileName - Log.txt" is created in the target folder. This is by design.
I think it's useless if output folder and eac3to folder are the same.

pihug12 :cool:

[Jeff Flowerday]

Getting the following from a particular blu-ray.

J:\>eac3to . 2)

This playlist contains m2ts parts with different track properties (2).

Here is the mpls file.

9502

[rica]

Thank you. Bug found and fixed.

Thanks a lot.
This seems better than the perfect:

http://www.sendspace.com/file/mnxf9g

Used method in transcoding:
mlp extracted by DVDAExplorer+transcoded to flac by eac3to v.3.12.

Used way in listening:
Bitperfect streaming via WASAPI over foobar2000 v. 0.9.3 beta 2 or Media Center 13 over analog outputs.

This was the perfect before 3.12: (ex-perfect)

http://www.sendspace.com/file/scv0kk

The method:
Extract mlp+decode to wav with DVD-Audio Explorer 2008.07.21 (Beta3) by MaximA and retransmit to flac via eac3to.
It is perfect.



_ _ _ _ _ _ _

[GZZ]

Would you mind uploading the CLIPINF + PLAYLIST folders together with 00010.m2ts and the first 30MB of 00009.m2ts? Having these files might help me fix the problem.


http://hbar.dk/Files/Der_Untergang_Nordic.rar

Splitted the m2ts file using tsmuxer into a 30 mb file.

Size file are 21 mb (best compression). I hope it works or else pm me. :)

[madshi]

new version (3.12) is ready to download? -test shows old one...
What do you mean? The download gets me v3.12. If you don't get that, try emptying your browser cache and redownload.

Did you plan a feature to mux audio tracks and subtitles files to MKV ?
No. At least not in the near future.

Getting the following from a particular blu-ray.

J:\>eac3to . 2)

This playlist contains m2ts parts with different track properties (2).

Here is the mpls file.
Attaching files to this forum takes a lot of time. It's better to upload them somewhere. But in this case the mpls file doesn't help, anyway. In most cases like this, the playlist contains one very short m2ts part which has different track types (e.g. AC3 instead of DTS-HD or SD video instead of HD video or something like that). Do a track listing for every m2ts part in the playlist (if there are not too many). Probably the very first or very last m2ts part is the problem. Try playing this m2ts part separately. Maybe you can simply delete it?

Thanks a lot.
This seems better than the perfect:
Don't know why. Maybe the DVD-Audio Explorer does some weird audio processing? I've no idea...

[rica]

Don't know why. Maybe the DVD-Audio Explorer does some weird audio processing? I've no idea...

I'm interested in the conclusions, don't wonder why. (This might be a decoder isssue, who knows :) )
Thanks again.

[jamos]

1.8.8b


I have no idea. But I'm not interested in doing any further bug hunting on my side because I don't think that this is a bug in eac3to. If you see any indication that eac3to is at fault, please let me know, though.





Ok no problem though I can not even get 1.88b to work properly with a 8 channel pcm file. will not play. I only get this with 8 channel dts.

[Blue_MiSfit]

@madshi:

Awhile back I saw clipping audio when transcoding MP2->AC3. You mentioned it was due to internal bit depth restrictions in libavcodec, which could probably be overcome with a more recent build of libavcodec. Did you ever integrate this?

Just curious ;)

Thanks again for your _AMAZING_ tool!!!

~MiSfit

[zeropc]

can someone please explain to me why the '"full range" flag should be removed or kept?

thanks

[VonZippa]

When I run "eac3to -test", one of the things I keep seeing is:

Haali Matroska Muxer (2009-01-10) is installed
There's a new version (2009-01-11) available
http://haali.net/mkv

The problem is, I have the latest download from that webpage installed, yet I keep getting the same message. I've tried completely uninstalling the old version and then installing, but it does the same thing. What am I missing?

[Snowknight26]

Nothing. The issue is with Haali's website.

[madshi]

Awhile back I saw clipping audio when transcoding MP2->AC3. You mentioned it was due to internal bit depth restrictions in libavcodec, which could probably be overcome with a more recent build of libavcodec. Did you ever integrate this?
Not yet.

can someone please explain to me why the '"full range" flag should be removed or kept?
This is a flag in the h264 stream which tells the decoder where black and white level are set in the source. Basically with the flag set, black and whites are at PC level. With the flag cleared they are at video level. It's not 100% correct what I'm saying, I think, but you get the picture. Now if you have a video stream where the flag is set incorrectly, and if you have a decoder which honors this flag, you'll either get washed out blacks/whites or you'll get clipped shadow/white detail.

Now virtually all movie and TV sources are encoded to video levels. So the flag should always be turned off. However, some broadcasters (especially some European HDTV broadcasters) have this flag accidently set. Now the sat/cable receiver decoders usually ignore this flag, so the problem doesn't show. But if you play such a stream on e.g. an NMT media player, you'll get wrong black/white level, because the NMT actually honors the flag.

eac3to checks if the flag is set and complains, because the thread is normally supposed to be not set. eac3to can remove it, and even does so by default. If you have a stream where the flag actually needs to be set, you can tell eac3to not to remove it. But I've yet to see such a stream...

[GZZ]

Did you get my file Madshi ?

/GZZ

[Atak_Snajpera]

Nothing. The issue is with Haali's website.
haali's website shows correct date because he lives in Russia (Different time zone!)

[laserfan]

haali's website shows correct date because he lives in Russia (Different time zone!)That's right--all one has to do to make that Haali error go away is to edit the file date to some time within 11-Jan.

Not sure how many different ways there are to do this but I used Attribute Changer iirc.

[TinTime]

madshi, I feel kind of picky reporting this but anyway, I think I've found a very, very minor bug. If I run...

eac3to.exe 1.eac3 1.dts -core

...then it causes eac3to to crash. No points for spotting the pointless switch I accidentally used! If I drop the -core it works fine of course, so it's not like there isn't a workaround (i.e. use the tool correctly :rolleyes:). I thought I'd better report it though.

I've tried it with a few eac3 files (all from HDDVD) and it always crashes, so you should be able to replicate it (unless it's a problem with my pc). Let me know though if you want a sample, bug report, etc.

[jmonier]

There appears to be a problem converting RAW/PCM 3.0 (Vicky Cristina Barcelona) to WAV. The center channel ends up being tagged as LFE. eac3to does correctly identify it (although PowerDVD identifies it as LPCM 3.0) but then doesn't seem to handle it as such.

Using -down2 also seems to have a similar problem and apparently throws away what it thinks is the LFE channel.

[SomeJoe]

madshi,

Thanks for your tool, it works incredibly well.

I'd like to ask (and you would seem to be the best person to answer this) -- why is it that Dolby Digital Plus (.eac3) tracks from HD-DVD cannot be muxed to Blu-Ray and played on a Blu-Ray player? Is there that much of a fundamental difference between the .eac3 used on HD-DVD and what should/could be used on BD?

I own several HD-DVDs and would like to (in some instances) use the HD-DVD .eac3 track directly on a BD back-up, but I've never been able to get it to work. I always have to change the audio to something else (DTS, PCM, or standard AC3).

[73ChargerFan]

That's right--all one has to do to make that Haali error go away is to edit the file date to some time within 11-Jan.
Or... eac3to is updated 50 times each year, and Haali's is only once, or maybe twice. An exception would work for the next 49 weeks.

[xkodi]

Don't know why. Maybe the DVD-Audio Explorer does some weird audio processing? I've no idea...

last time i checked "DVD-Audio Explorer" produces the same PCM output, i.e. bit by bit identical, to "DigiOnAudio2 Professional", which is reference DVD-Audio Player/Decoder/Encoder.

p.s. "DVD-Audio Explorer" also supports "SMART" (System Managed Audio Resource Technique) for DVD-Audio: http://patches.sonic.com/pdf/white-papers/wp_dvd_audio.pdf

[peterjcat]

madshi,

Thanks for your tool, it works incredibly well.

I'd like to ask (and you would seem to be the best person to answer this) -- why is it that Dolby Digital Plus (.eac3) tracks from HD-DVD cannot be muxed to Blu-Ray and played on a Blu-Ray player? Is there that much of a fundamental difference between the .eac3 used on HD-DVD and what should/could be used on BD?

I own several HD-DVDs and would like to (in some instances) use the HD-DVD .eac3 track directly on a BD back-up, but I've never been able to get it to work. I always have to change the audio to something else (DTS, PCM, or standard AC3).

DD+ on HD DVD and on Blu-ray have very little in common and it's unlikely you'll ever get it to work, as a Blu-ray player expects never to see HD DVD-style DD+ tracks and won't know what to make of them.

I wonder whether a dual-format player might have better luck, though? I don't know how separate the HD DVD/BD sides are to these things.

[SomeJoe]

DD+ on HD DVD and on Blu-ray have very little in common and it's unlikely you'll ever get it to work, as a Blu-ray player expects never to see HD DVD-style DD+ tracks and won't know what to make of them.

I wonder whether a dual-format player might have better luck, though? I don't know how separate the HD DVD/BD sides are to these things.

Well, yes, that's what I've read. But it doesn't seem to make a lot of sense. Why would DD+ be so fundamentally different on HD-DVD compared to Blu-Ray?

No other audio codec for the two formats is different. AC3 is interchangeable, as is all flavors of DTS, PCM, and even TrueHD. (The one small difference on TrueHD is that on HD-DVD, TrueHD can exist without a lossy AC3 core because HD-DVD has TrueHD as a mandatory-support codec, whereas on Blu-Ray, TrueHD is an optional-support codec, and therefore the lossy AC3 core is required).

Why would only DD+ be so fundamentally different that it would be the only audio codec that's not at all interchangeable?

I'm just looking for a bit more insight and explanation than "they're too different", because I'm not sure I can believe that.

Perhaps it's just a question that we don't know enough about the specifications of the format on Blu-Ray (I believe there's only one commercial Blu-Ray disc that uses DD+), or some other knowledge-block that would prevent us from currently being able to do it. But I highly doubt that the two incarnations of DD+ are just fundamentally different.

[nwg]

My Sony S350 BD player is supposed to do DD+. I have done HD DVD to Blu Ray and kept the DD+. I can't actually play DD+, but it comes out as standard DD ok. I can't imagine why it wouldn't come out as DD+ if I had the capability.

[~bT~]

what is the latest version of the ArcSoft DTS Decoder? and what is the recommended version if any, for use with eac3to?

[TruckChase!]

Latest version of dtsdecoder.dll is 1.1.0
Latest version of asaudiohd.ax is from 2/5/2008.

(I'm not sure what eac3to looks for)

Both come with the latest version of tmt, which can be updated for for free with any modern version. Unfortunately, it doesn't seem to work for some people. I still can't get it to work on Vista64; see my post @ the bottom of this thread:

http://forum.doom9.org/showthread.php?t=139946&page=2

[peterjcat]

Well, yes, that's what I've read. But it doesn't seem to make a lot of sense. Why would DD+ be so fundamentally different on HD-DVD compared to Blu-Ray?

No other audio codec for the two formats is different. AC3 is interchangeable, as is all flavors of DTS, PCM, and even TrueHD. (The one small difference on TrueHD is that on HD-DVD, TrueHD can exist without a lossy AC3 core because HD-DVD has TrueHD as a mandatory-support codec, whereas on Blu-Ray, TrueHD is an optional-support codec, and therefore the lossy AC3 core is required).

Why would only DD+ be so fundamentally different that it would be the only audio codec that's not at all interchangeable?

I'm just looking for a bit more insight and explanation than "they're too different", because I'm not sure I can believe that.

Perhaps it's just a question that we don't know enough about the specifications of the format on Blu-Ray (I believe there's only one commercial Blu-Ray disc that uses DD+), or some other knowledge-block that would prevent us from currently being able to do it. But I highly doubt that the two incarnations of DD+ are just fundamentally different.

Well, FWIW here's what Wikipedia has to say:

On HD DVD, DD+ is designated a mandatory audio-codec. An HD DVD movie may use DD+ as the primary (or only) audio track. An HD DVD player is required to support DD+ audio by decoding and outputting it to the player's output jacks. As stored on disc, the DD+ bitstream can carry for any number of audio-channels up to the maximum allowed, at any bitrate up to 3.0 Mbit/s.

On Blu-ray Disc, DD+ is an optional codec, and is deployed as an extension to a "core" AC-3 5.1 audiotrack. The AC-3 core is encoded at 640 kbit/s, carries 5 primary channels (and 1 LFE), and is independently playable as a movie audio track by any Blu-ray player. The DD+ extension bitstream is used on players that support it by replacing the rear channels in the 5.1 setup with higher fidelity versions, along with providing a possible channel extension to 6.1 or 7.1. The complete audio track is allowed a combined bitrate of 1.7 Mbit/s: 640 kbit/s for the AC-3 5.1 core, and 1 Mbit/s for the DD+ extension. During playback, both the core and extension bitstreams contribute to the final audio-output, according to rules embedded in the bitstream metadata.

So that sounds pretty different to me. On BD it's a core+extension model similar to DTS-HD but using a legacy DD core; on HD DVD it's a whole new codec with different compression and no core.

I think it's just another historical gift of the format war. Clearly the BD group wanted the studios to include legacy audio so the core+extension or interleaved codecs made sense, Toshiba had control over playback so didn't mind all-new codecs with internal decoding. Dolby developed a lossy codec for each format and called them both DD+.

I'm not saying that they've got nothing in common, but I don't think we can assume they can be easily interchanged.

That said, it's entirely possible that all or most DD+ decoders are in fact capable of decoding both kinds of DD+ but the tools we're using do something bad to DD+ like they have with TrueHD for so long. (eg tsMuxeR flags DD+ as AC3, which can't help). So I definitely think it's worth investigating that.

[Thunderbolt8]

what does the message mean which eac3to sometimes gives with some captures "this video track contains the (probably incorrect) 'full range' flag" and what does eac3to do then by default (or how can I see what eac3to does by default and if this is correct?)

[nwg]

Well, FWIW here's what Wikipedia has to say:



So that sounds pretty different to me. On BD it's a core+extension model similar to DTS-HD but using a legacy DD core; on HD DVD it's a whole new codec with different compression and no core.

.

If HD DVD has no AC3 core, then why would the HD DVD to Blu Ray I did come out as AC3 5.1? It also comes out as AC3 5.1 when I play the EVO files from the pc through the same amp. I am using the original DD+ audio from the HD DVD's and muxed with TsMuxer to make a BD.

[rijnton]

Can anyone help me please ? I am getting desparate.
At first I tried remuxing with eac3to, using a trial version of TMT. This went perfect !
So I thought to take the plunge and I bought the retail version of TMT (online download). It turns out to be version 2.1.6.129. But although TMT itself works perfectly, eac3to -test reports that Arcsoft DTS Decoder is not installed.
I already tried a lot of suggested solutions (copying .dll files to eac3to folder, registering commands etc), but nothing works. I also tried the environment path thing, but no success, although I don't know if I did that right. I went to My Computer>Properties>Advanced>Variables and selected Path and clicked Edit and added the value C:\Program Files\Common Files\Arcsoft\Bin. After that under Name there is PATH and under Value of C:\Program Files\Common Files\Arcsoft\Bin. Is that the correct way ?
Btw, before installing the "cracked, trial" version I made a Ghost backup and restored from that before installing the legal, bought version.
As I said, I am getting desperate because I bought TMT only for the decoder and now it turns out to be not working. Please help me.

[Killroy™]

Can anyone help me please ? I am getting desparate.
At first I tried remuxing with eac3to, using a trial version of TMT (or actually a non-working cracked version, resulting in still trial.....). This went perfect !
So I thought to take the plunge and I bought the retail version of TMT (online download). It turns out to be version 2.1.6.129. But although TMT itself works perfectly, eac3to -test reports that Arcsoft DTS Decoder is not installed.
I already tried a lot of suggested solutions (copying .dll files to eac3to folder, registering commands etc), but nothing works. I also tried the environment path thing, but no success, although I don't know if I did that right. I went to My Computer>Properties>Advanced>Variables and selected Path and clicked Edit and added the value C:\Program Files\Common Files\Arcsoft\Bin. After that under Name there is PATH and under Value of C:\Program Files\Common Files\Arcsoft\Bin. Is that the correct way ?
Btw, before installing the "cracked, trial" version I made a Ghost backup and restored from that before installing the legal, bought version.
As I said, I am getting desperate because I bought TMT only for the decoder and now it turns out to be not working. Please help me.

My experience was that the trial version screws up the install of the Retail (bought) version.

Uninstall the trial, Reboot, delete all Arcsoft folders (even hidden ones), Reboot, and then run a reg cleaner (CCleaner or whatever one you want) and delete all the orphaned registries. Reboot.

And then... this is most important.... run regedit and search for "arcsoft" and "TotalMedia" and delete every single thing that has those names. Reboot.

Install the Retail version and see if that helps. I am sure it will.

[rijnton]

My experience was that the trial version screws up the install of the Retail (bought) version.

Uninstall the trial, Reboot, delete all Arcsoft folders (even hidden ones), Reboot, and then run a reg cleaner (CCleaner or whatever one you want) and delete all the orphaned registries. Reboot.

And then... this is most important.... run regedit and search for "arcsoft" and "TotalMedia" and delete every single thing that has those names. Reboot.

Install the Retail version and see if that helps. I am sure it will.

Thanks Killroy. Did exactly what you told me, but unfortunately the same negative result. I really feel screwed-up.

[TruckChase!]

I've tried everything recommended in this thread and the other dedicated to this issue, and still nothing. I've tried 2 different versions on 3 different pcs (32bit XP, 64bit Vista, 32bit Vista) and still no luck. I'm a little disappointed because I dumped a lot of money on a product that I didn't really want (TMT) for anything but it's use with EAC3to.

Is it possible that we could know exactly what eac3to is looking for to help us in our troubleshooting efforts? I know the codecs are working... I can use them in any number of external programs (zoomplayer, etc.) and they show up fine in codec manager apps. If we knew what exactly eac3to was looking for we could compare versions or something and try to figure out the issue....

[laserfan]

Thanks Killroy. Did exactly what you told me, but unfortunately the same negative result. I really feel screwed-up.Are you CERTAIN that eac3to is not working, i.e. not decoding DTS files properly?

I installed the retail TMT and it's always worked, despite that the -test option failed with some versions of eac3to!

[lchiu7]

..

Why would only DD+ be so fundamentally different that it would be the only audio codec that's not at all interchangeable?..

As noted in a Dolby White paper

Unique Aspects of Dolby Digital Plus on Blu-ray Disc vs. HD DVD
There is a fundamental difference in the way programs are carried in the two HD disc formats.

Like DVD-Video, the HD DVD format carries audio and video signals in the MPEG-2 “program stream” format, whereas the Blu-ray Disc format carries them in the MPEG-2 “transport
stream” format, the same manner in which ATSC DTV and DVB signals are broadcast. A result of this otherwise subtle distinction is that there are packetizing differences for the audio
bitstreams between the two disc formats. You may recall that DVD-Video is not able to carry Dolby Digital bitstreams of more than 448 kbps. The same remains generally true for HD DVD,
although due to finer granularity in the choice of data rates, the actual maximum rate increases slightly, to 504 kbps. In order to support the higher bit rates and greater number of channels
offered by Dolby Digital Plus, HD DVD discs will use Dolby Digital Plus bitstreams (which have progressively shorter coding frames as the bit rate increases, thereby always fitting within the
defined audio packets on the disc). For example, standard Dolby Digital has a six-block frame, and in that case the Dolby Digital Plus bit rate can reach 0.5 Mbps; a three-block frame can
reach 1 Mbps; a two-block frame, 1.5 Mbps; and a one-block frame, 3 Mbps.

In contrast, the Blu-ray format has no such packetizing constraint, so one immediate result is the ability to transport Dolby Digital at its maximum 640 kbps rate for the first time. That’s slightly higher than the 576 kbps carried on D-VHS.

I read this as being a difference in the packet sizes which is why the pure DD+ stream extracted from a HD-DVD cannot be played on a BD player.

Not sure how a device like the Popcorn Hour would handle it though

[TruckChase!]

Are you CERTAIN that eac3to is not working, i.e. not decoding DTS files properly?

I installed the retail TMT and it's always worked, despite that the -test option failed with some versions of eac3to!

Sorry, I know this is directed @ rijnton, but I think we're trying to fix the same problem here....

C:\Encoding\Video\eac3to>eac3to.exe M:\encoding\video\work\bluray\FORGETSARAHMAR
SH\BDMV\STREAM 5: M:\encoding\video\work\bluray\FORGETSARAHMARSH\output\test.ac3

M2TS, 3 video tracks, 7 audio tracks, 5 subtitle tracks, 1:57:45, 24p /1.001
1: Chapters, 20 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: h264/AVC, 480p24 /1.001 (20:11)
4: h264/AVC, 480p24 /1.001 (20:11)
5: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
6: DTS, Spanish, 5.1 channels, 24 bits, 768kbps, 48khz
7: DTS, French, 5.1 channels, 24 bits, 768kbps, 48khz
8: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
9: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
10: E-AC3, English, 2.0 channels, 192kbps, 48khz
11: E-AC3, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
12: Subtitle (PGS), English
13: Subtitle (PGS), Spanish
14: Subtitle (PGS), French
15: Subtitle (PGS), Spanish
16: Subtitle (PGS), French
a05 The ArcSoft and Sonic decoders don't seem to work, will use libav instead.
a05 The libav DTS decoder doesn't decode the full DTS-HD information.
a05 Extracting audio track number 5...
a05 Extracting DTS core...
a05 Decoding with libav/ffmpeg...
a05 Remapping channels...
a05 Encoding AC3 <640kbps> with libAften...
a05 Creating file "M:\encoding\video\work\bluray\FORGETSARAHMARSH\output\test.ac
3"...


Thank you for the help! Any more would be greatly appreciated...

[rijnton]

Are you CERTAIN that eac3to is not working, i.e. not decoding DTS files properly?

I installed the retail TMT and it's always worked, despite that the -test option failed with some versions of eac3to!

I get the same result as TruckChase reported above: Arcsoft DTS and Sonic don't seem to work.

[jamos]

I can't reproduce any problems here. I even tried doing the same thing as you (eac3to + Pcm2Tsmu + tsMuxeR) and for me the channel order is always correct. I can only guess that there's something wrong in your playback chain...


Ok some final thoughts on this issue. Tsmuxer is not the issue as it muxes fine using dts 7.1 and plays back fine. The issue must exist in eac3to or Pcm3tsmu as I am getting this with multiple dts-ma 7.1 sources. You have to have a 7.1 reciever to test this as it looks correct in the file but on playback the center channel plays through the right front. I can send you a small sample if you want..else I will just convert to 5.1 no problem.

btw..testing now in xp 32bit to make sure its not a wierd vista 64 bug.

Edit: nope same issue with a different machine and xp...

[laserfan]

Sorry you guys--that eac3to log confirms the problem.I also tried the environment path thing, but no success, although I don't know if I did that right. I went to My Computer>Properties>Advanced>Variables and selected Path and clicked Edit and added the value C:\Program Files\Common Files\Arcsoft\Bin. After that under Name there is PATH and under Value of C:\Program Files\Common Files\Arcsoft\Bin.I don't follow this last sentence. The Path variable Value is a bunch of entries separated by semicolons. You need(ed) to add a semicolon, and then the path e.g. C:\Program Files\Common Files\Arcsoft\Bin after the semicolon.

I looked at mine and I also have an entry as follows:

...;C:\Program Files\Common Files\Arcsoft\Bin;C:\Program Files\Arcsoft\TotalMedia Theatre\Codec

Not sure if it's "the magic" but give it a try (and reboot after making the change).

[TruckChase!]

Sorry you guys--that eac3to log confirms the problem.I don't follow this last sentence. The Path variable Value is a bunch of entries separated by semicolons. You need(ed) to add a semicolon, and then the path e.g. C:\Program Files\Common Files\Arcsoft\Bin after the semicolon.

I looked at mine and I also have an entry as follows:

...;C:\Program Files\Common Files\Arcsoft\Bin;C:\Program Files\Arcsoft\TotalMedia Theatre\Codec

Not sure if it's "the magic" but give it a try (and reboot after making the change).

I can assure you that my path is set correctly, including the codec dir, bin dir, and I also tried including the commonfiles\arcsoft\mpeg engine dir since there are quite a few codecs stored there to no avail. Thanks though!

I can't speak for rijnton, but as for myself I've been doing this stuff for quite awhile and am very comfortable that I've followed all the advice presented here in the forums with no luck. I think there may be something to the newer versions of Arcsoft TMT that eac3to isn't picking up. The directory structure, for example is slightly different on the newest versions. (pointing to a "root" directory of total media extreme and then having a subdir for TMT, even if TMT is all you have installed) This is why I'd really like to know exactly what eac3to is looking for to help troubleshoot this issue. To reiterate, the codecs are registered and do work successfully. I can use them in any directshow application that allows the use of external codecs.... except eac3to. (main example is zoomplayer, but others work as well)

That being said, this is definitely not a registration issue, and based off of all read and attempted it isn't a pathing issue either. I would be very happy to provide madshi with any information required to help troubleshoot this issue on behalf of myself and others experiencing the same problem.

Thanks again!

[rijnton]

@laserfan: Could you tell me, being not so technical, what I exactly should do where, what and in which sequence, after I go to My Computer>Properties>Advanced>Environmental Variables ?
When I am there I see 2 windows User Variables for my username and System Variables (I translate the names from dutch). In the upper window there were 2 lines for Temp and Tmp and now (after maybe doing something wrong) a 3rd line with Path and C:\Program Files\Common Files\Arcsoft\Bin after that.
What should I do /change now and where/what ?

Btw, I discovered something else, which is very strange imo
I tried uninstalling the retail version and installing the trial version (which is Totalmedia Extreme with a patch that is not working) again. eac3to did not complain and I could start Totalmedia Theatre being trial. Then I installed the retail version on top of it, without uninstall. Bingo ! Totalmedia Theatre registered and eac3to still not complaining. BUT...
after that I had TM Extreme, still being trial AND TMT not trial anymore. Then I uninstalled Extreme in Windows software removal and TMT still working, not trial, but eac3to complaining again. So this procedure will not help imo, because when the trial of Extreme expires I will have the same troubles again.

Btw, I have Windows XP and not Vista.

[TruckChase!]

@laserfan: Could you tell me, being not so technical, what I exactly should do where, what and in which sequence, after I go to My Computer>Properties>Advanced>Environmental Variables ?
When I am there I see 2 windows User Variables for my username and System Variables (I translate the names from dutch). In the upper window there were 2 lines for Temp and Tmp and now (after maybe doing something wrong) a 3rd line with Path and C:\Program Files\Common Files\Arcsoft\Bin after that.
What should I do /change now and where/what ?



As long as you don't have a user defined path variable, (which you don't by default) you only need to change system. If you've added it there, remove it, as there is no need for a user specific variable when the system variable will do. This will help us avoid confusion.

Look in the system variables and find the PATH variable. Double click it to open it up... notice that each variable is separated by a ";". (without quotes) You can add to the beginning or end of the line. To avoid confusion, I'm talking about adding at the beginning. Keep the entire contents already there, and paste your full directory structure (c:\program files\common files\blah\blah..) in the beginning. Follow this with a ;. After that entry, add your other entrie(s) for TMT codec locations, following each entry with a ;.

In the end, you should have a path variable that contains all the directories with the arcsoft codecs in question, separated by a ;. There is no need for a ; at the beginning (before the first path) or at the end (after the last path) but having them there won't hurt you either way, as they are merely a delimiter and not actual "data".

[laserfan]

@rijnton I don't know if putting a Path variable up under User Variables works!? What has been suggested is editing the Path variable in the box BELOW, under "System Variables". If you try that (adding a semicolon then C:\Program Files\Common Files\Arcsoft\Bin) take care not to mess-up any of the existing Path values already there!

If that doesn't work then suspect you have hit the same wall as TruckChase!, and hopefully he will find the secret to getting your installs to work. Sounds like he has done everything including the regsvr32 bit with, what is it, ASAudioHD.ax. Not sure madshi wants to help w/Arcsoft but I wish you good luck.

EDIT: I see TC! was advising exact Path instructions whilst I typed here... :)

[rijnton]

Thanks for the replies.
Meanwhile I looked under the System Variables, the bottom window, but I don't see a name PATH there. The only entry with PATH in its name is Pathext. Or do I still misunderstand it.
If not, is something wrong with my system then and what can I do ?

[rijnton]

Although I still don't know what is going on, I FINALLY got it working. (Knock on wood). I could find a trial version of TMT alone (not Extreme and don't ask me where) and installed it after uninstalling the retail. Then I installed the retail again on top of the trial and now everything seems o.k. TMT seems to be o.k. and not trial and version .129 and no problems with eac3to.
It seems that the retail version of TMT messes things up for me. But after installing it on top of a trial version it works !

[jamos]

BTW to all.. TMT will not play in vista 64 (its not supported if you look on thier website) but the dlls should still work to encode dts using eac3to..they do on my vista 64 machine. On my xp32 machine TMT plays fine.

[TruckChase!]

BTW to all.. TMT will not play in vista 64 (its not supported if you look on thier website) but the dlls should still work to encode dts using eac3to..they do on my vista 64 machine. On my xp32 machine TMT plays fine.

Wow; that's weird. It's the exact opposite scenario on my machine. (note that I do believe what you're saying, it's just frustrating) My Vista64 machine will play back fine with TMT, it just won't work with eac3to. That being said I can't get my xp 32 or vista 32 machines to work either. I'm becoming more convinced that this has something to do with the new non-trial releases installing in a different directory structure based off of the total media extreme product. I think the naming of the codecs might be slightly different as well.....

[zeropc]

i never used the demux feature in eac3to, so i wonder if this would be the right cmd (example)...

eac3to "F:\MOVIE\BDMV\STREAM\00001.M2TS" "G:\FILES\" -demux

#edit#

i just tried what i was thinking and it didn't work...

log

eac3to v3.12
command line: "C:\Program Files\encoding-tools\eac3to\eac3to.exe" "I:\broken.arrow.trl.m2ts" "F:\video" -demux
------------------------------------------------------------------------------
M2TS, 1 video track, 1 audio track, 3 subtitle tracks, 0:02:28, 24p /1.001
1: MPEG2, 1080p24 /1.001 (16:9)
2: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
3: Subtitle (PGS)
4: Subtitle (PGS)
5: Subtitle (PGS)
Track 2 is used for destination file "video".
This audio conversion is not supported. <ERROR>

any idea what went wrong?

[yonta]

i never used the demux feature in eac3to, so i wonder if this would be the right cmd (example)...

eac3to "F:\MOVIE\BDMV\STREAM\00001.M2TS" "G:\FILES\" -demux

#edit#

i just tried what i was thinking and it didn't work...

log

any idea what went wrong?

eac3to demuxes to the current folder.
If you want your source file demuxed to a specific folder,
go to the folder and run command there.

F:\video>"C:\Program Files\encoding-tools\eac3to\eac3to.exe" "I:\broken.arrow.trl.m2ts" -demux

[TruckChase!]

Although I still don't know what is going on, I FINALLY got it working. (Knock on wood). I could find a trial version of TMT alone (not Extreme and don't ask me where) and installed it after uninstalling the retail. Then I installed the retail again on top of the trial and now everything seems o.k. TMT seems to be o.k. and not trial and version .129 and no problems with eac3to.
It seems that the retail version of TMT messes things up for me. But after installing it on top of a trial version it works !

Thanks to this info from rijnton I finally got this up and running. It seems that the detection most definitely has something to do with the newer installs from Arcsoft surrounding the Total Media Extreme product family. (which includes TMT) If using an older version that is TMT only it works fine even when the newer codec versions are installed on top of it.

Here's how to get it working:

1. Uninstall all previous versions and do the cleanup steps listed in this thread. (Registry clean, delete orphaned directories, delete instances of "arcsoft" in the registry)

1.Download and install the old version of the TMT trial that was pulled from Arcsoft's site awhile ago:

http://rapidshare.com/files/155298896/arcsoft_totalmediatheatre_engintro.exe.html

2. Do not run the program at this point - reboot
3. download this patch: http://www.arcsoft.com/downloads/digitaltheatre/arcsoft/totalmediatheatre_2.1.6.105_2.1.6.125_Update_ALL.exe
and install it. Again, do not run program. Note that this step may not be necessary. Rijton reported that he did not install this patch and went right to installing the retail w/o problems.
4. Reboot
5. eac3to should work at this point. If you own TMT or TME retail, go ahead and install now.

[idbirch2]

I own several HD-DVDs and would like to (in some instances) use the HD-DVD .eac3 track directly on a BD back-up, but I've never been able to get it to work. I always have to change the audio to something else (DTS, PCM, or standard AC3).dburckh, the same guy who made the TrueHD fix for TSMuxer also makes ToNMT which converts HD-DVD Dolby Digial Plus streams into BluRay compliant DDP streams. Before the BDs came out, I used it to remux the Bourne trilogy HD-DVDs to BluRay and the DDP worked fine.

Edit: Looking back over the ToNMT forums, I think it was QuickFix.cmd that I used. The .dpp track was first demuxed using eac3to then processed with QuickFix then finally remuxed into a BluRay structure using TSMuxer and played back on the PS3. My Denon 1909 picked up the DDP no problem.

[laserfan]

Here's how to get it working:

1.Download the old version of the TMT trial that was pulled from Arcsoft's site awhile ago:

http://rapidshare.com/files/155298896/arcsoft_totalmediatheatre_engintro.exe.html

2. Do not run the program at this point - reboot
3. download this: http://www.arcsoft.com/downloads/digitaltheatre/arcsoft/totalmediatheatre_2.1.6.105_2.1.6.125_Update_ALL.exe
and install it. Again, do not run program
4. Reboot
5. eac3to should work at this point. If you own TMT or TME retail, go ahead and install now.When do you install the trial from step 1?

[odin24]

Can eac3to slow down a DTS HD-MA track from 24 to 23.976fps without re-encoding to DTS 1536?

I only ask because the last time I played a 24fps BD on my NTSC setup the video had a slow but steady flicker which was quite annoying.

[rijnton]

When do you install the trial from step 1?

Of course I cannot speak for TruckChase, but I uninstalled my retail version first, rebooted, then installed the trial version, rebooted and installed the retail again on top of the trial. I didn't use the update in between. In other words, I installed my retail .129 on top of the .120 trial and I now have a full-working (not trial) 2.1.6.129.

Btw, part of the credits must go to HDCompatible from abhdt.net:
http://www.abhdtv.net/forum/showthread.php?t=12314&page=7

And I now understand the PATH thing, although it was not needed anymore. I now see PATH but with regards to Arcsoft only the path to Common Files\Arcsoft\Bin

[tebasuna51]

Can eac3to slow down a DTS HD-MA track from 24 to 23.976fps without re-encoding to DTS 1536?

It is not possible without re-encoding.

[TruckChase!]

When do you install the trial from step 1?

Lol; Sorry... I shouldn't post after going out drinking w/ friends. Post updated with better instructions. :thanks:

[NanoBot]

Hi,

Yesterday I did my first tests in converting a DTS MA audio track to a multichannel flac, which works fine at all. Nevertheless, when the converting process was finished, I ended up with a flac file, which has a bitdepth of 24bit, while eac3to told me, that the bitdepth of the original track was only 16bit. Since using 24bit in the flac file is a waste of space when the original bitdepth is only 16bit, I repeated the process using the -down16 parameter, and now everthing is fine. Both the original DTS MA audio track an the resulting flac file have the same bitdepth of 16bit.

That brings up the question:

Is it possible to use eac3to in a way, where a lossless track, like Dolby TrueHD or DTS MA, is decoded and analysed for it's bitdepth, while the resulting data is not written to the hdd and discarded instead ? This would make it possible to do a faster first pass to scan the track for it's bitdepth, and then do a second pass to convert the track in a flac file of the original bitdepth.

C.U. NanoBot

[odin24]

It is not possible without re-encoding.

Thanks, I sorta figured that. Instead I converted to LPCM, then changed 24fps fo 23.976fps.

[anode]

Hello to madshi and all of you...

i have a short question to subtitle demuxing:

as far i have seen, in the output of eac3to's demuxed sup-files from blu-ray they contain the necessary PTS info but no DTS (decoder time stamp) information. I had some subtitles which do not play on my Samsung BD-P2500 (longer, more complex ones) (remuxed with tsMuxeR to BR-AVCHD). Why are the DTS fields always zero?
When demuxing the same track with tsMuxeR, DTS info is present.

If i use the tsMuxeR-demuxed source, then the subtitles are displayed correctly.
Perhaps you can give me some info about this DTS information?

Thank you very much for all the work on eac3to.

[sidekick2]

dburckh, the same guy who made the TrueHD fix for TSMuxer also makes ToNMT which converts HD-DVD Dolby Digial Plus streams into BluRay compliant DDP streams. Before the BDs came out, I used it to remux the Bourne trilogy HD-DVDs to BluRay and the DDP worked fine.

Edit: Looking back over the ToNMT forums, I think it was QuickFix.cmd that I used. The .dpp track was first demuxed using eac3to then processed with QuickFix then finally remuxed into a BluRay structure using TSMuxer and played back on the PS3. My Denon 1909 picked up the DDP no problem.

According to the tonmt forums, quickfix.cmd changes "TS Type from 0x81 to 0x84 for eac3 tracks" I just tried this 3 different ways on my Bourne Identity hd-dvd, and it still won't play in the ps3 as a blu-ray disk. I used his tonmt gui, I used ts4np, and then I took the output from tsmuxergui and ran quickfix.cmd on the m2ts file and burned. All three causes the ps3 to stutter and display audio as dolby 112k. (no audio can be heard, but the video plays smooth for ~1 sec then stutters)

quickfix does not convert the hd-dvd dolby-digital plus into something that blu-ray players can understand. At least that I can get to work..

Or... the PS3 doesn't like dolby digital plus.

Arcsoft total media theater plays the files and or folder structures no problem. Shows up as dolby digital 1536k.

When burned... no worky... :(

[peterjcat]

If HD DVD has no AC3 core, then why would the HD DVD to Blu Ray I did come out as AC3 5.1? It also comes out as AC3 5.1 when I play the EVO files from the pc through the same amp. I am using the original DD+ audio from the HD DVD's and muxed with TsMuxer to make a BD.

Well, I gather that HD DVD DD+ is designed to be fairly easily downconverted to AC3 on-the-fly so that a compatible player can produce a good AC3 version for output via optical etc. I suppose you could argue that that's similar to having a legacy core, but I would say that it's not, since it still takes a bit of work and you need a new decoder that know what to do.

What hardware and software players are you using? It might be useful to work out which ones either downconvert HD DVD DD+ to AC3 or pass them through a receiver. The hardware players that do may have a chipset in common etc.

[nwg]

Well, I gather that HD DVD DD+ is designed to be fairly easily downconverted to AC3 on-the-fly so that a compatible player can produce a good AC3 version for output via optical etc. I suppose you could argue that that's similar to having a legacy core, but I would say that it's not, since it still takes a bit of work and you need a new decoder that know what to do.

What hardware and software players are you using? It might be useful to work out which ones either downconvert HD DVD DD+ to AC3 or pass them through a receiver. The hardware players that do may have a chipset in common etc.

I am using a Sony S350 and it has DD+ decoding in the specs for it. I am using the coaxial digital output (don't have HD audio capability). The amp is a Sony STR-DB930.

Software is PowerDVD 7.3 and Nero Showtime with the HD DVD/BD plugin.

I recently did a load of movie only HD DVD to Blu Ray's with DD+ straight from the HD DVD's inc the Bourne Trilogy.

[pbjr]

Anyone using Linux and Wine getting this error?

err:virtual:NtMapViewOfSection Sizes larger than 4Gb (66013e000) not supported on this platform
err:virtual:NtMapViewOfSection Sizes larger than 4Gb (66013e000) not supported on this platform
err:virtual:NtMapViewOfSection Sizes larger than 4Gb (66013e000) not supported on this platform
Error reading file ".\BDMV\STREAM\00000.m2ts".

Thanks,
Mike

** Update **

wine bug that was fixed in 1.1.16
http://bugs.winehq.org/show_bug.cgi?id=17373

[HanSolo00]

I have a small but simple feature request :)

Please add support for -24.975fps source framerate (the product of RePAL'd material.) It would be very useful to be able to "-24.975 -ChangeTo24.000" in order to revert audio back to original film speed, to match RePAL+AssumeFPS(24) video.

Thanks for the excellent tool!

[sidekick2]

I am using a Sony S350 and it has DD+ decoding in the specs for it. I am using the coaxial digital output (don't have HD audio capability). The amp is a Sony STR-DB930.

Software is PowerDVD 7.3 and Nero Showtime with the HD DVD/BD plugin.

I recently did a load of movie only HD DVD to Blu Ray's with DD+ straight from the HD DVD's inc the Bourne Trilogy.

I'm using the PS3, thru hdmi to an Onkyo 805. My Dad has a bdp-s350, so I think I'll take one of these bd-re I've burned over there and see if it works on his. If it does, then that means the PS3 doesn't like dd+.

Hmm... another site shows: http://whatsonhdtv.blogspot.com/2008/03/sony-playstation-3-and-blu-ray-audio.html

To date, Sony has not been terribly responsive to these suggestions. Other than confirming that:

* of the four higher-resolution or lossless audio compression codecs, the PS3 can decode Dolby TrueHD and DTS-HD Master Audio, but not Dolby Digital Plus or DTS-HD High Resolution

* the PS3 can pass through the four advanced audio compression codecs as bitstreams, rather than decoding them to linear PCM, but it can pass them through only in their "core" 5.1-channel, low-bitrate forms


This explains why I won't get dolby digital plus on the PS3. It's not supported or decoded. (unless it's a real blu-ray dd+, with a core, and then it will spit out the core only)

Has anyone else got a PS3 and dolby digital plus working? (from an hd-dvd rip?)

[laserfan]

...as far i have seen, in the output of eac3to's demuxed sup-files from blu-ray they contain the necessary PTS info but no DTS (decoder time stamp) information. ...If i use the tsMuxeR-demuxed source, then the subtitles are displayed correctly.Is this only for certain types of subs? When I tried comparing the output of eac3to and tsMuxeR with HxD Hex Editor, its Compare feature says they are bit-for-bit identical...? :confused:

[SomeJoe]

According to the tonmt forums, quickfix.cmd changes "TS Type from 0x81 to 0x84 for eac3 tracks" I just tried this 3 different ways on my Bourne Identity hd-dvd, and it still won't play in the ps3 as a blu-ray disk. I used his tonmt gui, I used ts4np, and then I took the output from tsmuxergui and ran quickfix.cmd on the m2ts file and burned. All three causes the ps3 to stutter and display audio as dolby 112k. (no audio can be heard, but the video plays smooth for ~1 sec then stutters)

quickfix does not convert the hd-dvd dolby-digital plus into something that blu-ray players can understand. At least that I can get to work..

Or... the PS3 doesn't like dolby digital plus.

Arcsoft total media theater plays the files and or folder structures no problem. Shows up as dolby digital 1536k.

When burned... no worky... :(

I tried the QuickFix.cmd from the ToNMT package on an HD-DVD rip. I was able to get the final .m2ts to play Dolby Digital Plus when playing through the PCH A110, although when fast forwarding or rewinding, the audio goes silent until you pause/play several times. This may be a PCH A110 bug rather than a bug in the muxing.

The same .m2ts file would not play through a Sony BDP-S350 when burned to a BD-RE. It was silent (no audio), and video freezes playback after about 15 seconds. So obviously, the muxing isn't 100% compliant. Of course, there was no core AC3 stream in the .m2ts either, which also isn't BD compliant, so there's all sorts of places where the Sony might have a problem.

[sidekick2]

I tried the QuickFix.cmd from the ToNMT package on an HD-DVD rip. I was able to get the final .m2ts to play Dolby Digital Plus when playing through the PCH A110, although when fast forwarding or rewinding, the audio goes silent until you pause/play several times. This may be a PCH A110 bug rather than a bug in the muxing.

The same .m2ts file would not play through a Sony BDP-S350 when burned to a BD-RE. It was silent (no audio), and video freezes playback after about 15 seconds. So obviously, the muxing isn't 100% compliant. Of course, there was no core AC3 stream in the .m2ts either, which also isn't BD compliant, so there's all sorts of places where the Sony might have a problem.

I took my burn of Bourne Identity that would not work and it plays *FINE* in my Dad's bdp-s350! This disk would not play at all in my PS3 (as described in my previous posts). I would have never guessed the PS3 wouldn't be able to decode DD+. The on-screen display on the bdp-s350 showed it as Dolby Digital Plus. I didn't watch the whole disk, just about 5 minutes of the movie. I took the bd-re version that I used ts4np to create blu-ray structure with.

I also attempted to play the disk in the PS3 using bitstream, instead of linear pcm. Still didn't work. On the second attempt, it locked the PS3 up hard, and I had to flip the switch for a reboot.

So, bottom line, some players will play the hd-dvd version of DD+, once properly flagged with quickfix.cmd, even though the DD+ isn't the "true" or "correct" version of DD+ meant for blu-ray.

All we need now, is for Madshi to come up with a DD+ converter! :) Create a core from the hd-dvd DD+, then stick the extra bits on top of the core with some duct tape and spit. Ohh, and write a new firmware for the PS3 so I can play DD+. (minor detail I almost left out)

[NanoBot]

Hi everybody,


my original question


After converting a TrueHD track to flac the first time, I noticed, that eac3to is not showing the bitdepth of a TrueHD track in the track overview, while it does on DTS MA tracks. Nevertheless, eac3to is automatically doing a second pass to reduce the bitdepth of the output file to 16bit, if a 16bit THD was detected during the first pass, which is fine. Therefore it is necessary to rephrase my original question:

Are the bitdepth values of DTS MA tracks, which are shown in the track overview, only nominal values, which are derived from the DTS header ? Or are these values absolutely reliable, so I can be sure that a DTS MA track really has the bitdepth indicated in the track overview ?

If the indicated bitdepth value is only nominal, I would propose that eac3to should operate like with TrueHD tracks by automatically doing a second pass to reduce the bitdepth of the output file, if necessary. On the other hand, if the bitdepth values indicated in the track overview are reliable, I would propose that eac3to should automatically use the "-down16" parameter, if the bitdepth of a DTS MA track is indicated as 16bit.

C.U. NanoBot

[glood1]

Problem remuxing a bluray


eac3to v3.11
command line: "D:\utils\BLURAY\Tools\eac3to\eac3to.exe" "T:\SOURCE_BLURAY\THE_SIGNAL\THE_SIGNAL\" 1) 2: "T:\temp\RipBot264temp\job1\video.mkv" -seekToIFrames 4: "T:\temp\RipBot264temp\job1\audio.1.dts" -core 1: "T:\temp\RipBot264temp\job1\chapters.txt" -progressnumbers
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 3 subtitle tracks, 1:43:16, 24p
1: Chapters, 12 chapters
2: h264/AVC, 1080p24 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
4: DTS Master Audio, French, 5.1 channels, 16 bits, 48khz
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48khz)
5: AC3, English, 2.0 channels, 224kbps, 48khz
6: Subtitle (PGS), French
7: Subtitle (PGS), French
8: Subtitle (PGS), French
[v02] The video framerate is correct, but rather unusual. <WARNING>
Creating file "T:\temp\RipBot264temp\job1\chapters.txt"...
[a04] Extracting audio track number 4...
[v02] Extracting video track number 2...
[v02] Muxing video to Matroska...
[a04] Extracting DTS core...
[a04] Patching bitdepth to 24 bits...
[a04] Creating file "T:\temp\RipBot264temp\job1\audio.1.dts"...
Added fps value to MKV header.
Video track 2 contains 148699 frames.
eac3to processing took 17 minutes, 45 seconds.
Done.


the framerate is not stable (speedup the video track is bad)
a few moment there is shifting frames (lost frames)
it s working with another movie whith 23.XXX

thanks for you help

[zeropc]

madshi,

say, can you add a "set progressive flag" option into eac3to for slowdown clips/movies?

that would make some pal to ntsc conversions proper.

here is why...

source is 1080i50 (film) and doesnt show any interlacing. slowing it down to 23.976fps works fine but the output file is not 1080/24p, it's 1080i48/1001.
this gives playback problems on the ps3 for example. it's all jerky, like frame-dropping.

[FoxRacR17]

I am having a problem and I cannot figure it out. I'm trying to encode a dts master audio file to wma pro 5.1 but the audio keeps coming out all garbled when I play it on my 360. I have arcsoft total media theater 2.1.6.125 installed, and nero 7 installed. It is coming from a .m2ts files, I have tried demuxing using both xport and tsmuxer, and then I even tried doing the ac3 audio that is in the .m2ts file so that the nero codec would be used but the resulting .wma file was still garbled. I also even tried using the sonic dts decoder but I still get messed up sounds. I'm running on Vista x64 sp1, and i'm using eac3to to change the dts-ma file into a single .wav (the resulting file is ALWAYS 4.50 gigs exactly, shouldnt it be bigger?) then i use tranzcode .40 to split it up into individual .wavs for each channel, then I combine them into a .avi using wavavimux, then i use csscript to convert it to a wma pro file. Please someone help me figure this out. I have been trying all day and its getting on my last nerve.

[tebasuna51]

@madshi
To my request:
There are a problem with 'stdout.wav' and automatic second pass for overflow.
When this occurs the output is duplicated but the corrected second part is out of sync (because the second header) with garbage result.

Seems the problem is solved when add -2pass or -normalize parameters. Now seems the first ouput is inhibited and only the second pass is really output.
...

you answer with:
v3.09
...
* fixed: clipping removal 2nd pass was executed even for "stdout"
But now, when a 2nd pass is needed, I get a WARNING:
The 2nd pass can't be started due to "stdout".

And with the 'normalize' parameter (or '2pass', working ok before 2.09) now I get:
eac3to v3.12
command line: "D:\Programa\eac3to\eac3to.exe" "D:\Test\VideoS\VDPrueba\10.ac3" stdout.wav -down2 -normalize
------------------------------------------------------------------------------
AC3, 5.1 channels, 0:01:00, 448kbps, 48khz
The Nero decoder doesn't seem to work, will use libav instead.
Decoding with libav/ffmpeg...
Remapping channels...
Downmixing multi channel audio to stereo...
Writing WAV...
Creating file "stdout.pass1.wav"...
The 2nd pass can't be started due to "stdout". <WARNING>
eac3to processing took 2 seconds.
Done.
The "stdout.pass1.wav" is created but nothing is sended by STDOUT to encoder.

[TinTime]

I'm trying to encode a dts master audio file to wma pro 5.1...It is coming from a .m2ts files, I have tried demuxing using both xport and tsmuxer...and i'm using eac3to to change the dts-ma file into a single .wav...then i use tranzcode .40 to split it up into individual .wavs for each channel, then I combine them into a .avi using wavavimux, then i use csscript to convert it to a wma pro file.

Why not just use eac3to for your first three steps?
eac3to input.m2ts(s) trackno: output.wavs

Anyway, at what stage in your process does the audio become garbled? If the wav file produced by eac3to is ok then it's not an eac3to problem. What happens if you play it?

[monotube]

Sorry, but I have not understood how to use Eac3to to get the sequence in a seamless branching BD.

I write : eac3to folder/wall-e and I get various Mpls which show video info and ALL THE LANGUAGES. In effect I would like to know what is the correct sequence for one language to obtain one m2ts with only one language

Many thanks in advance

[anode]

Is this only for certain types of subs? When I tried comparing the output of eac3to and tsMuxeR with HxD Hex Editor, its Compare feature says they are bit-for-bit identical...? :confused:

I tried it with a few subtitles from various Blu-Ray titles and eac3to's demuxed .sup and tsMuxeR demuxed .sup are different.

The difference is that tsMuxeR .sup Versions contain PTS and DTS timing (Bytes 5-8 after the "PG" identifier), while the eac3to .sup contains only PTS (Bytes 1-4 after "PG"), DTS info is always zero.

I had one subtitle track from Day after Tomorrow (info tables, graphical bitmaps) which doesn't display without DTS info.

[Snowknight26]

Sorry, but I have not understood how to use Eac3to to get the sequence in a seamless branching BD.

Example:
eac3to v3.12
command line: eac3to.exe "G:\Blu-rays\Die Another Day"
------------------------------------------------------------------------------
1) 00001.mpls, 2:12:34
[130+131+133+134+136+137+139+140+142+143+145+146+148+149+151+152+154+155+157+158+160+161+163+164+166+167+169+170+172+173+175+176+178].m2ts
- Chapters, 33 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- DTS Master Audio, English, multi-channel, 48khz
- AC3, Spanish, stereo, 48khz
- AC3, French, multi-channel, 48khz
- AC3, English, stereo, 48khz
- AC3, English, stereo, 48khz
- AC3, English, stereo, 48khz

2) 00099.mpls (angle 2), 2:12:34
[130+132+133+135+136+138+139+141+142+144+145+147+148+150+151+153+154+156+157+159+160+162+163+165+166+168+169+171+172+174+175+177+178].m2ts
- Chapters, 33 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- DTS Master Audio, English, multi-channel, 48khz
- AC3, Spanish, stereo, 48khz
- AC3, French, multi-channel, 48khz
- AC3, English, stereo, 48khz
- AC3, English, stereo, 48khz
- AC3, English, stereo, 48khz

[mrr19121970]

Sorry, but I have not understood how to use Eac3to to get the sequence in a seamless branching BD.

I write : eac3to folder/wall-e and I get various Mpls which show video info and ALL THE LANGUAGES. In effect I would like to know what is the correct sequence for one language to obtain one m2ts with only one language

Many thanks in advance


you could try Clown_BD GUI, ideal for newcomers.

[tvjunky]

Hi Madshi,
while copying my last few Blu-rays i saw something like this:


sXX Subtitle stream XX contains XX normal and XX forced captions


Would it be much work to automatically create a second file containing only the forced captions while extracting this stream?

Thanks for your great work again!
S.

[mrr19121970]

I'll second that request.

[avivahl]

Hi Madshi,
while copying my last few Blu-rays i saw something like this:


sXX Subtitle stream XX contains XX normal and XX forced captions


Would it be much work to automatically create a second file containing only the forced captions while extracting this stream?

Thanks for your great work again!
S.+1 on that request.

[DoomBot]

That would be a great thing added to this already awesome software:)

[jmonier]

Hi Madshi,
while copying my last few Blu-rays i saw something like this:


sXX Subtitle stream XX contains XX normal and XX forced captions


Would it be much work to automatically create a second file containing only the forced captions while extracting this stream?

Thanks for your great work again!
S.

SupRip has an option to only do forced subtitles.

[TinTime]

Madshi

eac3to seems to occasionally use the wrong title number when it's specified in the command.

For example I get the following results getting all titles with Twelve Monkeys on HDDVD:

eac3to v3.12
command line: "D:\Vtemp\programs\eac3to\eac3to.exe" "E:\Vtemp\12_MONKEYS_HDDVD" -log=main.txt
------------------------------------------------------------------------------
1) feature_01.EVO+feature_02.EVO, 2:09:32
"feature"
- VC-1, 1080p (16:9)
- E-AC3, English, 5.1, 48khz
- E-AC3, Italian, 5.1, 48khz
- E-AC3, Spanish, 5.1, 48khz
- E-AC3, English, 2.0, 48khz

2) 12_monkeys_archives.EVO, 0:39:30
"Bonus3"
- VC-1, 1080p (16:9)

3) MAKINGOF12MONKEYSN6LE_HD.EVO, 1:27:28
"Bonus1"
- MPEG2, 480i (4:3)
- E-AC3, English, 2.0, 48khz

4) 12MONKEYSTRAILERN4FE_HD.EVO, 0:02:24
"Bonus2"
- MPEG2, 480i (4:3)
- E-AC3, English, 2.0, 48khz

However if I then specify title 3 I get details back for title 2:

eac3to v3.12
command line: "D:\Vtemp\programs\eac3to\eac3to.exe" "E:\Vtemp\12_MONKEYS_HDDVD" 3) -log=3.txt
------------------------------------------------------------------------------
EVO, 1 video track, 0:39:32
1: Chapters, 237 chapters
2: VC-1, 1080p24 /1.001 (16:9) with pulldown flags

Also specifying title 2 brings back details for title 3. Titles 1 and 4 are ok though.

Is this related to this change back in 2.84?
* title sorting criteria changed: resolution is more important than runtime

It seems that when using eac3to to display all titles then it gives priority to resolution as per the change, but when specifying the title in the command line it's still sorting by runtime.

Thanks.

[Thunderbolt8]

seems like the "a remaining delay of Xms could not be fixed" message is missing when using -slowdown:

eac3to v3.12
command line: G:\eac3to\eac3to D:\movie.ts G:\movie.mkv
------------------------------------------------------------------------------
TS, 1 video track, 1 audio track, 1:29:41, 50i
1: h264/AVC, 1080i50 (16:9)
2: AC3, 2.0 channels, 384kbps, 48khz, dialnorm: -22dB, -2332ms
[v01] The video track contains the (probably incorrect) "full range" flag. <WARNING>
[a02] Extracting audio track number 2...
[a02] Removing AC3 dialog normalization...
[v01] Extracting video track number 1...
[a02] Applying (E-)AC3 delay...
[v01] Removing "full range" flag from video bitstream.
[a02] A remaining delay of +4ms could not be fixed.
[v01] Muxing video to Matroska...
[a02] Creating file "G:\belle - 2 - AC3, 2.0 channels, 384kbps, 48khz.ac3"...
[v01] Detected PTS overflow...
[a02] Detected PTS overflow...
[a02] The last (E-)AC3 frame is incomplete and thus gets skipped. <WARNING>
Added fps value to MKV header.
Video track 1 contains 269074 frames.
eac3to processing took 3 minutes, 6 seconds.
Done.

eac3to v3.12
command line: G:\eac3to\eac3to D:\movie.ts G:\movie.mkv -slowdown
------------------------------------------------------------------------------
TS, 1 video track, 1 audio track, 1:29:41, 50i
1: h264/AVC, 1080i50 (16:9)
2: AC3, 2.0 channels, 384kbps, 48khz, dialnorm: -22dB, -2332ms
[v01] The video track contains the (probably incorrect) "full range" flag. <WARNING>
Disabling DRC for Nero (E-)AC3 decoding...
[a02] Extracting audio track number 2...
[v01] Extracting video track number 1...
[a02] Removing AC3 dialog normalization...
[a02] Decoding with DirectShow (Nero Audio Decoder 2)...
[v01] Removing "full range" flag from video bitstream.
[v01] Writing new framerate "24fps /1.001" to bitstream.
[v01] Muxing video to Matroska...
[a02] DirectShow reports 2.0 channels, 24 bits, 48khz
[a02] Applying RAW/PCM delay...
[a02] Changing FPS from 25.000 to 23.976...
[a02] Encoding AC3 <448kbps> with libAften...
[a02] Creating file "G:\belleslowed - 2 - AC3, 2.0 channels, 384kbps, 48khz.ac3"...
[v01] Detected PTS overflow...
[a02] Detected PTS overflow...
[a02] The last (E-)AC3 frame is incomplete and thus gets skipped. <WARNING>
Added fps value to MKV header.
Video track 1 contains 269074 frames.
eac3to processing took 5 minutes, 26 seconds.
Done.

[Mark_A_W]

Madshi, a little feature request:

I'm trying to convert a mono 32 bit IEEE Float, 96khz .pcm file generated from DRC (Digital Room Correction software), to a mono 32bit IEEE Float 96khz .wav file.

Eac3to works, but only up to 24 bit (if I put "-32", it gives the specify info error).


If I could get eac3to to do this, then I can incorporate it in a giant .bat file which measures all 6 channels in my HT, then processes the measurement files with DRC, then converts to .wav format, ready for use in Convolver directshow plugin (or ffdshow, or foobar). All I do is place the microphone, turn it on, double click the shortcut to the .bat file, and it would all just happen.

I've been searching for ages for a program to do the conversion (other than Audition, which is major overkill), it just occured to me that the program I use to convert my BDs to MKV might do it :) But at the the moment it's close but no cigar..


If it's only a little change...please?

If it's a biggie, no problems, I'll keep hunting for something else.

Thanks

Mark

[TinTime]

I'm trying to convert a mono 32 bit IEEE Float, 96khz .pcm file generated from DRC (Digital Room Correction software), to a mono 32bit IEEE Float 96khz .wav file.


Have you tried sox (http://sox.sourceforge.net)?

[Mark_A_W]

Have you tried sox (http://sox.sourceforge.net)?


Yes.

I tried:

sox leftspeaker.pcm leftspeaker.wav

and got:

"sox formats: no handler for file extension 'pcm' "

(I would expect that sox, like eac3to would need me to specify the bit depth, etc, but it doesn't even seem to recognise pcm at all).

Any other suggestions are welcome (and sorry for the kinda OT..well, it is an audio conversion). I tried Audacity (16 bit it seems). Cooledit pro/audition will do it, but I'd prefer opensource, and command line...not that I'm picky :)

Mark

[TinTime]

Rename the .pcm files to .raw and sox should recognise them. As you said you'll have to specify bitrate, bit depth, etc.

[Mark_A_W]

Rename the .pcm files to .raw and sox should recognise them. As you said you'll have to specify bitrate, bit depth, etc.

Thanks.

Took me all that time to sort out the syntax...

sox -r 96000 -b 32 -c 1 -f lfe.raw lfe.wav (I think that's right..)


Job done :)

[nwg]

How do I work out the delay in the audio is eac3to doesn't give me anything? Is there any software that will? I tried Tsmuxer but that saids 0ms.

[alc0re]

If eac3to doesn't mention any unfixed audio delay, then there isnt any required.

[MuteyM]

I am having a problem and I cannot figure it out. I'm trying to encode a dts master audio file to wma pro 5.1 but the audio keeps coming out all garbled when I play it on my 360. I have arcsoft total media theater 2.1.6.125 installed, and nero 7 installed. It is coming from a .m2ts files...

You could try my HDConcertRipper (http://forum.doom9.org/showthread.php?t=145140) app, it's a GUI that uses eac3to to convert Blu-Ray (or HD DVD) discs to WMA Pro 5.1 (or FLAC). It won't work if all you have is the M2TS file though, it needs a full Blu-Ray BDMV directory structure.

[idbirch2]

I took my burn of Bourne Identity that would not work and it plays *FINE* in my Dad's bdp-s350! This disk would not play at all in my PS3 (as described in my previous posts). I would have never guessed the PS3 wouldn't be able to decode DD+.But it does, it worked fine for me with all 3 Bourne Films, the PS clearly said Dolby Digial Plus. Maybe it's because I authored to AVCHD (4GB splits to a FAT32 HDD) and not burned to disc.

[DrNein]

Is the conversion valid from THD (no core) to THD+AC3? The output audio constantly "drops out" during playback. The original THD is otherwise not recognized by tsMuxeR though and I understand incompatible with BD too.

[lchiu7]

..

All we need now, is for Madshi to come up with a DD+ converter! :) Create a core from the hd-dvd DD+, then stick the extra bits on top of the core with some duct tape and spit. Ohh, and write a new firmware for the PS3 so I can play DD+. (minor detail I almost left out)

And what loss in AQ is there if the DD+ is converted to say DD at 640Kb? I guess only if the DD+ stream is 7.1 but then you could convert to DD-EX?

I am facing a similar situation. Finally got an AVR that has HDMI input ( in fact it decodes DD+ over bitstream) but nonoe of the playback devices I have, the A100, Toshiba A3 or PS3 can output DD+ as bitstream. The A3 outputs it as LPCM, the A100 not at all (perhaps as DD+ converted to PCM) and similarly for the PS3.

So I just convert the DD+ to DD and put up with the AQ reduction, if any

[peterjcat]

And what loss in AQ is there if the DD+ is converted to say DD at 640Kb? I guess only if the DD+ stream is 7.1 but then you could convert to DD-EX?

So I just convert the DD+ to DD and put up with the AQ reduction, if any

DD+ on HD DVD is not the same as DD+ on Blu-ray -- there's no core to extract, so converting DD+ to anything requires a complete lossy recode so you'll technically lose some quality even if you're going from 640k DD+ to 640k DD. Since the majority of DD+ tracks are 1536k you'll lose even more quality going to DD. It doesn't make any difference whether the DD+ is 7.1 or 5.1 (if there are any 7.1 DD+ movies).

However, the libavcodec conversion to 640k DD is pretty good so the quality loss shouldn't be that noticeable.

[peterjcat]

Is the conversion valid from THD (no core) to THD+AC3? The output audio constantly "drops out" during playback. The original THD is otherwise not recognized by tsMuxeR though and I understand incompatible with BD too.

What are you using for playback? And are you using the most recent (.18 or later) version of tsMuxeR?

eac3to's THD to THD+AC3 has almost always worked fine for me (playing back via Popcorn Hour A-110), using old versions of tsMuxeR fixed with ts4Np, or the new version of tsMuxeR.

[lchiu7]

DD+ on HD DVD is not the same as DD+ on Blu-ray -- there's no core to extract, so converting DD+ to anything requires a complete lossy recode so you'll technically lose some quality even if you're going from 640k DD+ to 640k DD. Since the majority of DD+ tracks are 1536k you'll lose even more quality going to DD. It doesn't make any difference whether the DD+ is 7.1 or 5.1 (if there are any 7.1 DD+ movies).

However, the libavcodec conversion to 640k DD is pretty good so the quality loss shouldn't be that noticeable.

What about DD+ to LPCM? Should be no loss at all then but a slight(!) increase in filesize (but you would have saved some on the video presumably?)

[edit]

OK - I can see some challenges with that. Just did the first evo file of a HD-DVD title and while the eac3 track was 978Mb the PCM was 4.6Gb! And I guess that would be almost double for the whole movie! As an aside I can see why folks are clamoring for multichannel FLAC support for the Popcorn Hour

[DrNein]

What are you using for playback? And are you using the most recent (.18 or later) version of tsMuxeR?

eac3to's THD to THD+AC3 has almost always worked fine for me (playing back via Popcorn Hour A-110), using old versions of tsMuxeR fixed with ts4Np, or the new version of tsMuxeR.

Using latest tsMuxeR. Playback with MPC-HC and CyberLink Audio Decoder (though dodgy sound is similar with other decoders). The duration of the converted file is also wrong (too long and the counter runs fast). The THD cannot be played back in MPC-HC (although fine in source with PDVD) and MediaInfo reports no specs. But when converted to DTS is it okay though so eacto must be extracting it correctly and just not converting to THD+AC3 properly.

[Snowknight26]

eac3to seems to report the number of fields for interlaced content, not the number of frames.

eac3to v3.12
command line: eac3to.exe "Z:\temp\threekings.ts" "Z:\Encoding Tools\temp\threekings.source.mkv"
------------------------------------------------------------------------------
TS, 1 video track, 1 audio track, 1:54:49, 50i
1: h264/AVC, 1080i50 (16:9)
2: AC3 Surround, 2.0 channels, 384kbps, 48khz, dialnorm: -27dB, -2081ms
[v01] The video track contains the (probably incorrect) "full range" flag. <WARNING>
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[v01] Removing "full range" flag from video bitstream.
[a02] Removing AC3 dialog normalization...
[a02] Applying (E-)AC3 delay...
[v01] Muxing video to Matroska...
[a02] A remaining delay of -1ms could not be fixed.
[a02] Creating file "Z:\Encoding Tools\temp\threekings.source - 2 - AC3 Surround, 2.0 channels, 384kbps, 48khz.ac3"...
[a02] [1:48:15] The source file seems to be damaged (transport error indicated). <WARNING>
[v01] [1:48:14] The source file seems to be damaged (transport error indicated). <WARNING>
[a02] [1:48:15] The source file seems to be damaged (discontinuity). <WARNING>
[v01] [1:48:14] The source file seems to be damaged (discontinuity). <WARNING>
[a02] [1:48:15] The source file seems to be damaged (discontinuity). <WARNING>
[v01] [1:48:14] The source file seems to be damaged (discontinuity). <WARNING>
[a02] [1:48:15] The source file seems to be damaged (discontinuity). <WARNING>
[v01] [1:48:14] The source file seems to be damaged (discontinuity). <WARNING>
[a02] This track is not clean. <WARNING>
[a02] The last (E-)AC3 frame is incomplete and thus gets skipped. <WARNING>
[a02] Audio has a gap of 96ms at playtime 1:48:13. <WARNING>
[a02] Starting 2nd pass...
[a02] Realizing (E-)AC3 gaps...
[a02] Creating file "Z:\Encoding Tools\temp\threekings.source - 2 - AC3 Surround, 2.0 channels, 384kbps, 48khz.ac3"...
Added fps value to MKV header.
Video track 1 contains 344450 frames.
eac3to processing took 7 minutes, 38 seconds.
Done.

[madshi]

Did you get my file Madshi ?
Yes, but didn't have time to look into it yet...

I think I've found a very, very minor bug. If I run...

eac3to.exe 1.eac3 1.dts -core

...then it causes eac3to to crash.
Thanks, will check that out.

There appears to be a problem converting RAW/PCM 3.0 (Vicky Cristina Barcelona) to WAV. The center channel ends up being tagged as LFE. eac3to does correctly identify it (although PowerDVD identifies it as LPCM 3.0) but then doesn't seem to handle it as such.
Can't reproduce that problem. The WAV file I end up with is reported by eac3to as:

"WAV, 3.0 channels, 0:00:20, 24 bits, 3456kbps, 48khz"

Using -down2 also seems to have a similar problem and apparently throws away what it thinks is the LFE channel.
Downmixing does not properly support anything other than 5.1 or higher. Downmixing funny channel configurations like 3.0 or 2.1 is currently not supported. That's on my to do list, but it doesn't have a high priority...

what does the message mean which eac3to sometimes gives with some captures "this video track contains the (probably incorrect) 'full range' flag" and what does eac3to do then by default (or how can I see what eac3to does by default and if this is correct?)
I've already explained this a number of times, please do a search. Thanks...

Ok some final thoughts on this issue. Tsmuxer is not the issue as it muxes fine using dts 7.1 and plays back fine.
That is not a proof at all. tsMuxeR might work just fine with DTS and still screw up LPCM muxing. Perfectly possible.

The issue must exist in eac3to or Pcm3tsmu as I am getting this with multiple dts-ma 7.1 sources. You have to have a 7.1 reciever to test this as it looks correct in the file but on playback the center channel plays through the right front.
It could also be your receiver which doesn't handle 7.1 LPCM correctly. Or it could be whatever source device you're using. I don't really know where the bug is introduced in your situation. But from what I can see the bug is not caused by eac3to. So there's nothing I can do for you.

i never used the demux feature in eac3to, so i wonder if this would be the right cmd (example)...

eac3to "F:\MOVIE\BDMV\STREAM\00001.M2TS" "G:\FILES\" -demux
eac3to demuxes to the current folder.
If you want your source file demuxed to a specific folder,
go to the folder and run command there.
Yes, yonta is right. Alternatively you can demux to a specific folder by using "eac3to source G:\FILES\movie.*".

Yesterday I did my first tests in converting a DTS MA audio track to a multichannel flac, which works fine at all. Nevertheless, when the converting process was finished, I ended up with a flac file, which has a bitdepth of 24bit, while eac3to told me, that the bitdepth of the original track was only 16bit.
That is not supposed to happen. Are you sure you're using the latest eac3to version? If so, please post the eac3to log.

why is it that Dolby Digital Plus (.eac3) tracks from HD-DVD cannot be muxed to Blu-Ray and played on a Blu-Ray player? Is there that much of a fundamental difference between the .eac3 used on HD-DVD and what should/could be used on BD?
I have never tried and thus cannot answer any questions about what plays on a Blu-Ray standalone player or not.

What I can say is that Dolby Digital Plus is indeed fundamentally different on HD DVD compared to Blu-Ray. Well, at least Blu-Ray main audio. For Blu-Ray secondary audio (e.g. Director's comments) DD+ is identical to HD DVD.

Well, yes, that's what I've read. But it doesn't seem to make a lot of sense. Why would DD+ be so fundamentally different on HD-DVD compared to Blu-Ray?
Because DD+ is optional on Blu-Ray. So they used a core+extension logic for Blu-Ray main audio, where the core is simple Dolby Digital (not Plus). For HD DVD DD+ is mandatory, so they didn't need to provide a compatible core.

ToNMT converts HD-DVD Dolby Digial Plus streams into BluRay compliant DDP streams.
No. It does not.

It doesn't change one bit in the DD+ stream. It just slightly changes the way the DD+ stream is stored in the m2ts container.

All we need now, is for Madshi to come up with a DD+ converter! :)
That's not possible without reencoding. And there is no (affordable) DD+ encoder available. So this is not going to happen.

as far i have seen, in the output of eac3to's demuxed sup-files from blu-ray they contain the necessary PTS info but no DTS (decoder time stamp) information. I had some subtitles which do not play on my Samsung BD-P2500 (longer, more complex ones) (remuxed with tsMuxeR to BR-AVCHD). Why are the DTS fields always zero?
When demuxing the same track with tsMuxeR, DTS info is present.
I didn't think the DTS info would have any value. Well, seems I was wrong. I'll fix that...

Please add support for -24.975fps source framerate (the product of RePAL'd material.) It would be very useful to be able to "-24.975 -ChangeTo24.000" in order to revert audio back to original film speed, to match RePAL+AssumeFPS(24) video.
I'm sorry to say, but the SSRC resampler used by eac3to only supports very specific changes. I don't think it will do "24.975 -> 24.000".

Edit: Scratch that. I just tried, it works. "-24.975" will be supported in the next build.

Problem remuxing a bluray

2: h264/AVC, 1080p24 (16:9)
[v02] The video framerate is correct, but rather unusual. <WARNING>

the framerate is not stable (speedup the video track is bad)
a few moment there is shifting frames (lost frames)
it s working with another movie whith 23.XXX
I don't think this is a bug in eac3to. I think it's your playback chain that somehow doesn't like 24.000 movies. So the best solution would probably be to slow the movie down to 23.976. The eac3to option "-slowdown" will do that for you. Afterwards you'll have a movie which would behave just like your average 23.976 Blu-Ray. You will need to apply "-slowdown" on both audio and video tracks.

say, can you add a "set progressive flag" option into eac3to for slowdown clips/movies?

that would make some pal to ntsc conversions proper.

here is why...

source is 1080i50 (film) and doesnt show any interlacing. slowing it down to 23.976fps works fine but the output file is not 1080/24p, it's 1080i48/1001.
this gives playback problems on the ps3 for example. it's all jerky, like frame-dropping.
A while ago I tried to do that, but I failed. The problem is that especially with h264 progressively encoded movies are very VERY different to interlaced encoded movies. I don't think patching 1080i48 to 1080p24 is possible without doing a full reencoding. At least not with h264. It might be possible with MPEG2 or VC-1. But you're talking about h264, right?

I'm trying to encode a dts master audio file to wma pro 5.1 but the audio keeps coming out all garbled when I play it on my 360.
Try converting to something else, e.g. AC3, and check whether that plays fine. If so, the problem is most likely caused by whatever software you're using after eac3to is done.

I'm running on Vista x64 sp1, and i'm using eac3to to change the dts-ma file into a single .wav (the resulting file is ALWAYS 4.50 gigs exactly, shouldnt it be bigger?)
Depends on the bitdepth and runtime. But 4.5GB is a perfectly reasonable size for a movie audio track.

But now, when a 2nd pass is needed, I get a WARNING:
The 2nd pass can't be started due to "stdout".
That's intended behavior. What would you like eac3to to do instead?

And with the 'normalize' parameter (or '2pass', working ok before 2.09) [...] the "stdout.pass1.wav" is created but nothing is sended by STDOUT to encoder.
That's a bug. Will fix that in the next build.

Would it be much work to automatically create a second file containing only the forced captions while extracting this stream?
I guess it would be possible, but I don't know how difficult it would be. I'll put it on my to do list.

eac3to seems to occasionally use the wrong title number when it's specified in the command.

For example I get the following results getting all titles with Twelve Monkeys on HDDVD
Ouch, thanks.

seems like the "a remaining delay of Xms could not be fixed" message is missing when using -slowdown
When you "slowdown", eac3to has to do a full reencoding of the audio track. In this situation it applies delay on the decoded data. Which means that delay can always be perfectly fixed.

Madshi, a little feature request:

I'm trying to convert a mono 32 bit IEEE Float, 96khz .pcm file generated from DRC (Digital Room Correction software), to a mono 32bit IEEE Float 96khz .wav file.

Eac3to works, but only up to 24 bit (if I put "-32", it gives the specify info error).
Hmmmm... I guess I could add support for that. But "-32" alone will not be enough. Because there is 32bit integer and 32bit float. So you will also have to specify "-float" or something like that...

Is the conversion valid from THD (no core) to THD+AC3? The output audio constantly "drops out" during playback. The original THD is otherwise not recognized by tsMuxeR though and I understand incompatible with BD too.
I'm not aware of any bug in thd+ac3 creation. It should work just fine, and you're the first one who reports a problem. My first guess would be that tsMuxeR does something wrong.

Using latest tsMuxeR. Playback with MPC-HC and CyberLink Audio Decoder (though dodgy sound is similar with other decoders). The duration of the converted file is also wrong (too long and the counter runs fast). The THD cannot be played back in MPC-HC (although fine in source with PDVD) and MediaInfo reports no specs. But when converted to DTS is it okay though so eacto must be extracting it correctly and just not converting to THD+AC3 properly.
Why do people always think that eac3to is at fault if anything goes wrong? :rolleyes:

eac3to seems to report the number of fields for interlaced content, not the number of frames.
That's true. Hmmmmm... I guess I could count frames and fields separately.

[yesgrey]

Why do people always think that eac3to is at fault if anything goes wrong? :rolleyes:

Because eac3to's developer is the only one answering the questions and fixing the bugs when they exist?;)
Maybe if you stop eac3to development people will think that it's already perfect and the problems must be on any other part...:D

[tebasuna51]

Originally Posted by tebasuna51
But now, when a 2nd pass is needed, I get a WARNING:
The 2nd pass can't be started due to "stdout".
That's intended behavior. What would you like eac3to to do instead?
It's OK. Was only to document the next problem:
Originally Posted by tebasuna51
And with the 'normalize' parameter (or '2pass', working ok before 2.09) [...] the "stdout.pass1.wav" is created but nothing is sended by STDOUT to encoder.
That's a bug. Will fix that in the next build.

Thanks.

[jmonier]

Can't reproduce that problem. The WAV file I end up with is reported by eac3to as:

"WAV, 3.0 channels, 0:00:20, 24 bits, 3456kbps, 48khz"


Did you try checking it with something other than eac3to? I would think that it's possible that if eac3to converts it incorrectly it would also report it incorrectly.

In my case, AC3Filter reported it as 2.1 and played it that way as well. And when I fed the WAV directly to my receiver via SPDIF the center channel was not there.

[Kurtnoise]

@madshi: could you add Dirac detection from TS files please ?

> specs (http://diracvideo.org/specifications)
> samples (http://dirac.kw.bbc.co.uk/download/video/maybefinal/)

thanks...

[DrNein]

I'm not aware of any bug in thd+ac3 creation. It should work just fine, and you're the first one who reports a problem. My first guess would be that tsMuxeR does something wrong.

Why do people always think that eac3to is at fault if anything goes wrong? :rolleyes:

I knew that was coming ;)
But the problem appears to occur in the conversion before tsMuxeR is involved as the output plays wrong on its own. Then again, the extracted unconverted track cannot be played or identified outside of PDVD with the tools I have. In PDVD the track is identified as MLP rather than TrueHD as by eac3to. I understand they are related but do not recall seeing that before. It could be the track is an oddball but eac3to does convert it to DTS nicely and the bitrate was not higher anyway.

[Snowknight26]

eac3to can't properly decode this (http://stfcc.org/misc/threekings.ac3) AC3 file to WAV.

[peterjcat]

What about DD+ to LPCM? Should be no loss at all then but a slight(!) increase in filesize (but you would have saved some on the video presumably?)

[edit]

OK - I can see some challenges with that. Just did the first evo file of a HD-DVD title and while the eac3 track was 978Mb the PCM was 4.6Gb! And I guess that would be almost double for the whole movie! As an aside I can see why folks are clamoring for multichannel FLAC support for the Popcorn Hour

You got it. Most DD+ tracks on HD DVD are 1.536mbps, to unpack that to 24-bit 5.1-channel LPCM take 6.9mbps, so it's 4.5x the size to maintain the same quality. Even if you've got plenty of disc space, it's the principle of the thing!

[lchiu7]

You got it. Most DD+ tracks on HD DVD are 1.536mbps, to unpack that to 24-bit 5.1-channel LPCM take 6.9mbps, so it's 4.5x the size to maintain the same quality. Even if you've got plenty of disc space, it's the principle of the thing!

Well the AQ drop to 640Kbs DD shouldn't be that noticeable. Now if we can re-encode as TrueHD that would solve the problems since I think I see Madshi saying there isn't an affordable DD+ encoder around

[tebasuna51]

eac3to can't properly decode this (http://stfcc.org/misc/threekings.ac3) AC3 file to WAV.

That file have the first 136 frames 2.0 and the rest 5.1

Use delaycut to split the file.

[Snowknight26]

Hah, I would have never guessed.

Thanks the the advice, worked perfectly.

[ACrowley]

That file have the first 136 frames 2.0 and the rest 5.1

Use delaycut to split the file.

yep..it works.

I had such AC3 File yesterday. Its a AC3 from a DVB-C HD Capture. The First Frame/032ms is 2.0.
eac3to/tsmuxer etc cant detect the AC3.

Cutting out 32ms works perfect

[idbirch2]

No. It does not.

It doesn't change one bit in the DD+ stream. It just slightly changes the way the DD+ stream is stored in the m2ts container.OK, I'm sorry, I just assumed that seeing as I processed an HD-DVD DD+ track using QuickFix.cmd (which I'm sure is part of ToNMT) and it then played on my PS3 as DD+ that it must be doing some sort of conversion. All I know is, you can play DD+ from a HD-DVD on a PS3 in the AVCHD format.

[madshi]

@madshi: could you add Dirac detection from TS files please ?
Is this actually used in real life anywhere?

I think there is a small bug in EAC3to way of parsing the mpls files and the m2ts files.

I did a test on Der Untergang (Nordic version)
The missing audio tracks and track languages are caused by the very strange way this Blu-Ray is authored. I could probably find a way to fix it, but honestly, I don't care much, because it's the only movie which shows this problem and fixing the problem would require a major rewrite of the playlist/clipinfo reading logic.

There is also a problem with extracting video and audio from this using the mpls file. Because the audio goes out of sync because of the small 00010.m2ts black screen file.
This will be fixed in the next build.

[madshi]

Did you try checking it with something other than eac3to?
No. And I see no reason to do that.

In my case, AC3Filter reported it as 2.1 and played it that way as well.
So maybe AC3Filter has a bug? Or maybe the WAV source filter doesn't forward the channel mask?

And when I fed the WAV directly to my receiver via SPDIF the center channel was not there.
No surprise there. SPDIF only supports 2 channels PCM/WAV transport.

But the problem appears to occur in the conversion before tsMuxeR is involved as the output plays wrong on its own.
The eac3to output plays wrong on its own? How did you test that?

In PDVD the track is identified as MLP rather than TrueHD as by eac3to.
Which demonstrates that PDVD is not a tool you should trust.

------

Guys, I'm very much willing to fix bugs in eac3to. But I'm not going to spend hours and hours on trying to figure out which of the various tools you're using is responsible for a specific problem you have. If you want me to look into a problem then you should better provide some convincing proof that the problem is actually caused by eac3to.

[madshi]

eac3to v3.13 released

http://madshi.net/eac3to.zip

* fields and frames are counted and displayed separately now
* added DIRAC bitstream parser
* added support for "-24.975" and "-changeto24.975"
* Blu-Ray subtitle demuxing: PTS value is now written to both PTS + DTS
* joining MKV files is now declined with a proper error message
* last chapter is now removed, if it's less than 10 seconds from end of movie
* fixed: "-normalize" didn't work with stdout, anymore
* fixed: audio delay was incorrect when 1st m2ts part contained no audio data
* fixed: very small WAV files were not detected correctly
* fixed: "eac3to source.eac3 dest.dts -core" crashed

[madshi]

i have a short question to subtitle demuxing:

as far i have seen, in the output of eac3to's demuxed sup-files from blu-ray they contain the necessary PTS info but no DTS (decoder time stamp) information. I had some subtitles which do not play on my Samsung BD-P2500 (longer, more complex ones) (remuxed with tsMuxeR to BR-AVCHD). Why are the DTS fields always zero?
When demuxing the same track with tsMuxeR, DTS info is present.

If i use the tsMuxeR-demuxed source, then the subtitles are displayed correctly.
Perhaps you can give me some info about this DTS information?
Can you please retry with v2.13? I've done a quick and dirty fix by simply copying the PTS value to the DTS field. I hope that this solves the problem. If not, I'll have to do a proper (and time consuming) fix.

[TinTime]

eac3to v2.13 released

Thanks very much!

Just one thing - it's v3.13 isn't it?

[DrNein]

The eac3to output plays wrong on its own? How did you test that?

The extracted THD is not identified by MediaInfo nor can it be played in MPC-HC with decoders such as CyberLink. The converted THD+AC3 can be be but the duration and playback is wrong (or presumably the playback is "accurate" but the conversion is screwy). The extracted THD can be played with a plugin for Winamp based upon mlpdec module from FFmpeg and as said converts okay to DTS.


Which demonstrates that PDVD is not a tool you should trust.


I thought it might demonstate that the THD track was odd and thus the dodgy conversion to THD+AC3.

But anyway it is not really worth spending anymore time on -especially as the conversion to DTS is fine. Again, I do appreciate your time and excellent tool.

Oops, typo in the new version release above... should be v3.13 -madshi needs a well deserved break ;)

[madshi]

The extracted THD is not identified by MediaInfo nor can it be played in MPC-HC with decoders such as CyberLink.
Isn't that true for every THD+AC3 track?

The converted THD+AC3 can be be but the duration and playback is wrong
Isn't that true for every THD+AC3 track? TrueHD is a variable bitrate codec, so duration is almost always shown wrong. The source filter only does a best guess on the duration.

I thought it might demonstate that the THD track was odd and thus the dodgy conversion to THD+AC3.
No, not at all. Most probably PDVD always reports MLP for every THD+AC3 track.

[laserfan]

eac3to v3.13 released

http://madshi.net/eac3to.zip

* fields and frames are counted and displayed separately now
Thanks for the update. I was kinda hoping it would ID "hybrid" clips which have a mix of I/L and progressive frames, but instead I will continue to rely on neuron2's tools for that. Here's a nasty one, it's actually 29.97:

eac3to v3.13
command line: eac3to E:\MAMMA_MIA!\BDMV\STREAM\00289.m2ts 1: test.vc1
------------------------------------------------------------------------------
M2TS, 1 video track, 1 audio track, 3 subtitle tracks, 0:02:40, 60i /1.001
1: VC-1, 1080i60 /1.001 (16:9)
2: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz, dialnorm: -4dB
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz, dialnorm: -4dB)
3: Subtitle (PGS), English
4: Subtitle (PGS), Spanish
5: Subtitle (PGS), French
[v01] Extracting video track number 1...
[v01] Creating file "test.vc1"...
Video track 1 contains 4800 frames.
eac3to processing took 11 seconds.
Done.

[DrNein]

Isn't that true for every THD+AC3 track?

No, this specific THD (at least) cannot be identified by MediaInfo or played by MPC-HC but after conversion to THD+AC3 it can. As far as I recall, other unaltered THD+AC3 (such as from BD) can likewise be identified and played but I would have to check more to confirm. Would have to check PDVD identification as well.

The duration of the THD+AC3 (converted) is crazy long and the counter plays fast. Plus as said the audio drops out or in other words is a bit warbly.

As for the BD subtitle problem, would that affect their display in MPC-HC? Because I notice that whether they are shown or not is unreliable and sometimes takes toggling off/on in the menu under subtitles and under splitter, but I assumed that was the fault of the internal splitter. Also, did the same problem exist with extraction from HD DVD or is that not applicable? Is there any way to fix already extracted SUP files (copy PTS value to DTS -if that turns out to be the fix)?

[peterjcat]

Well the AQ drop to 640Kbs DD shouldn't be that noticeable. Now if we can re-encode as TrueHD that would solve the problems since I think I see Madshi saying there isn't an affordable DD+ encoder around

No, it shouldn't be THAT noticeable, but again it's the principle of the thing :) And sure, recoding to TrueHD would be great, but no more affordable: you need the same program from Dolby. Surcode has an MLP encoder but I don't know if there's any way to convert the output to useful TrueHD.

[madshi]

Thanks for the update. I was kinda hoping it would ID "hybrid" clips which have a mix of I/L and progressive frames
What is "I/L"?

eac3to doesn't handle interlaced VC-1 streams well in the moment. That's still on my to do list. Frame/field counting should work fine for h264 and MPEG2, though.

No, this specific THD (at least) cannot be identified by MediaInfo or played by MPC-HC but after conversion to THD+AC3 it can.
So probably *NO* THD (without AC3) can be detected by MediaInfo or MPC-HC. Have you checked that? Probably no. But you should before reporting a problem.

The duration of the THD+AC3 (converted) is crazy long and the counter plays fast.
Incorrect duration probably happens for most/all THD+AC3 tracks. Have you checked that? Probably no. But you should before reporting a problem.

These are most probably not eac3to problems. So let's please finally stop discussing this. Unless you find some proof for an eac3to bug.

As for the BD subtitle problem, would that affect their display in MPC-HC?
Why don't you try for yourself?

[laserfan]

What is "I/L"? eac3to doesn't handle interlaced VC-1 streams well in the moment. That's still on my to do list.Sorry. Interlaced, of course! And my post was indeed about a VC-1 clip. If you can detect/identify the messes that are these hybrid VC-1 clips, you will join neuron2 as the only who's figured these out AFAICT.

[TM2-Megatron]

This seems like a very useful tool, however I'm not sure about a few of these additional programs or plug-ins that are required for some of its most useful features.

The Nero HD-DVD/Blu-Ray plug-in isn't even available anymore, so I'm not sure how people who didn't buy it during the relatively short window it was for sale would get it. As for the Arcsoft plug-in, I'm not sure which of their programs this one comes with, however I haven't heard great things about their software so I wouldn't really want to pay for it anyway. And Surcode's DTS plug-in is ridiculously expensive.

Are there any alternatives to some of these impractical plug-ins?

[TinTime]

Madshi

Sorry to do this to you but there seems to be a problem introduced in 3.13. The output audio is corrupted when the input file is a wav.

I ran...
"D:\Vtemp\programs\eac3to\eac3to.exe" mh2.wav mhtest2.flac
...with this (http://www.sendspace.com/file/5qve2p) test file. It's processed correctly in 3.12.

NB if you play back the corrupt output file turn your speakers down. The noise is pretty loud :eek:

Thank you.

[Snowknight26]

C:\unzipped\eac3to>eac3to C:\temp\test.wav test.flac
WAV, 1.0 channels, 0:00:30, 64 bits <float>, 6144kbps, 96khz
Reading WAV...
eac3to crashed...

C:\unzipped\eac3to>eac3to C:\temp\test2.wav test2.flac
WAV, 1.0 channels, 0:00:30, 32 bits <float>, 3072kbps, 96khz
Reading WAV...
eac3to crashed...


C:\unzipped\eac3to>eac3to.exe C:\temp\test3.wav C:\temp\test3.flac
WAV, 1.0 channels, 0:00:30, 32 bits, 3072kbps, 96khz
Reading WAV...
Reducing depth from 32 to 24 bits...
Encoding FLAC with libFlac...
Creating file "test3.flac"...
Original audio track: max 32 bits, average 28 bits, most common 27 bits.
The processed audio track has a constant bit depth of 24 bits.
eac3to processing took 1 second.
Done.

C:\unzipped\eac3to>eac3to.exe C:\temp\test3.wav C:\temp\test3.down24.try2.flac -down24
WAV, 1.0 channels, 0:00:30, 32 bits, 3072kbps, 96khz
Reading WAV...
Reducing depth from 32 to 24 bits...
Encoding FLAC with libFlac...
Creating file "test3.try2.flac"...
Original audio track: max 32 bits, average 28 bits, most common 27 bits.
The processed audio track has a constant bit depth of 24 bits.
eac3to processing took 1 second.
Done.

C:\unzipped\eac3to>eac3to.exe C:\temp\test3.wav C:\temp\test3.down24.flac -down24
WAV, 1.0 channels, 0:00:30, 32 bits, 3072kbps, 96khz
Reading WAV...
Reducing depth from 32 to 24 bits...
Encoding FLAC with libFlac...
Creating file "C:\temp\test3.down24.flac"...
Original audio track: max 32 bits, average 28 bits, most common 27 bits.
The processed audio track has a constant bit depth of 24 bits.
eac3to processing took 1 second.
Done.

C:\unzipped\eac3to>eac3to.exe C:\temp\test3.wav C:\temp\test3.full.flac -full
WAV, 1.0 channels, 0:00:30, 32 bits, 3072kbps, 96khz
Reading WAV...
Reducing depth from 32 to 24 bits...
Encoding FLAC with libFlac...
Creating file "C:\temp\test3.full.flac"...
Original audio track: max 32 bits, average 28 bits, most common 27 bits.
The processed audio track has a constant bit depth of 24 bits.
eac3to processing took 1 second.
Done.

Couple of things to notice:
32-bit and 64-bit floating point WAV files made with Audacity crash eac3to
32-bit doesn't.
All the FLAC files have differen't sizes and aren't identical to test3.wav. Notice the repeated sound in the beginning.
I've included a reference FLAC file named audacity.flac.

http://www.stfcc.org/misc/wavs.zip (51,803,165 bytes) - should be finished uploading within 30 minutes of posting this.

[madshi]

The Nero HD-DVD/Blu-Ray plug-in isn't even available anymore, so I'm not sure how people who didn't buy it during the relatively short window it was for sale would get it.
I don't know. You could try contacting Nero support. But the libav decoder works well enough, too, so it's not really all that necessary to have Nero working.

As for the Arcsoft plug-in, I'm not sure which of their programs this one comes with, however I haven't heard great things about their software so I wouldn't really want to pay for it anyway.
Well, it's your choice. Either you want DTS-HD decoding or you don't want it. If you do want it, Arcsoft is the best choice. The DTS core can be decoded by eac3to without needing any external software. But obviously decoding only the core means that you don't get full quality. The core quality is very good already on its own, though.

And Surcode's DTS plug-in is ridiculously expensive.
Then just use AC3 encoding instead, which doesn't need any external software.

there seems to be a problem introduced in 3.13. The output audio is corrupted when the input file is a wav.
Argh, thanks. One of the bug fixes fired back.

[madshi]

Couple of things to notice:
32-bit and 64-bit floating point WAV files made with Audacity crash eac3to.

[...]

Notice the repeated sound in the beginning.
Should be fixed in next build.

All the FLAC files have differen't sizes and aren't identical to test3.wav.
That is expected behaviour due to dithering. test3.wav is 32bit which (AFAIK) the current libFlac encoder does not support. So eac3to has to dither down to 24bit.

[madshi]

eac3to v3.14 released

http://madshi.net/eac3to.zip

* WAV reading was broken for all but very small files (introduced in v3.13)

[Thunderbolt8]

does the case that gaps/overlaps of seamless branching do not exist any more apply to all kind of audio tracks? Im sure for flac, as flac framelength is only 1ms, but does this also apply to the other formats? would a DTS track created from a seamless branching movie have more gaps or overlaps in between as a flac track created out of the same movie?

[madshi]

Sorry. Interlaced, of course! And my post was indeed about a VC-1 clip. If you can detect/identify the messes that are these hybrid VC-1 clips, you will join neuron2 as the only who's figured these out AFAICT.
I was not aware that there are "hybrid" VC-1 clips. Do you mean that some parts of the clip are encoded interlaced and some progressively? Can I have a sample of such a clip, please? What kind of output would you expect from eac3to for such a clip?

does the case that gaps/overlaps of seamless branching do not exist any more apply to all kind of audio tracks? Im sure for flac, as flac framelength is only 1ms, but does this also apply to the other formats? would a DTS track created from a seamless branching movie have more gaps or overlaps in between as a flac track created out of the same movie?
As soon as you transcode audio, delays and audio gaps/overlaps are removed down to 1ms granularity. So if you e.g. convert a TrueHD track to DTS, you get 1ms granularity, because eac3to has to decode, anyway. If e.g. the movie track is DTS and you want to avoid transcoding by keeping it as DTS, then eac3to has to live with the frame size limitations. E.g. for DTS a frame is usually 10.6ms long, so delay and gap/overlap fixing is done with up to 5.3ms granularity. But don't worry, eac3to is clever enough to not let the granularity "problem" make seamless branching audio go out of sync. Gap/overlap fixing remainders are cumulated and treated correctly with every new gap/overlap, of course...

[Snowknight26]

Should be fixed in next build.

What about the repeated beginning?

[laserfan]

I was not aware that there are "hybrid" VC-1 clips. Do you mean that some parts of the clip are encoded interlaced and some progressively? Can I have a sample of such a clip, please? What kind of output would you expect from eac3to for such a clip?I have uploaded a 20MB sample clip here (http://www.mediafire.com/?zlbnd2sccd5), made with dgsplit. Here is what DGVC1IndexNV has to say about it:
DGVC1IndexFileNV2

D:\mamma mia clip\snip.m2ts
C:\Program Files (x86)\DGVC1DecNV\

STREAM 1
PKTSIZ 192
VPID 4113
FO 2
YUVRGB 1
LUMFILT 0 0
CLIP 0 0 0 0
RANGE 0 0 20971519 30000
AUDIO 1100

SEQ 798 INTERLACE PULLDOWN
ENTRY 820 !BROKEN CLOSED
FRM I FRAME
FRM I FRAME
FRM B FRAME
FRM B FRAME
FRM I FRAME
FRM B FRAME
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SEQ 224478 INTERLACE PULLDOWN
ENTRY 224500 !BROKEN !CLOSED
FRM I FRAME
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SEQ 603294 INTERLACE PULLDOWN
ENTRY 603316 !BROKEN !CLOSED
FRM I FIELD
FRM B FIELD
FRM B FIELD
FRM P FIELD
FRM B FIELD
FRM B FIELD
SEQ 1121694 INTERLACE PULLDOWN
ENTRY 1121716 !BROKEN !CLOSED
FRM I FIELD
FRM B FIELD
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SEQ 1864542 INTERLACE PULLDOWN
ENTRY 1864564 !BROKEN !CLOSED
FRM I FIELD
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SEQ 2636766 INTERLACE PULLDOWN
ENTRY 2636788 !BROKEN !CLOSED
FRM I FIELD
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SEQ 5919774 INTERLACE PULLDOWN
ENTRY 5919796 !BROKEN !CLOSED
FRM I FIELD
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SEQ 9033054 INTERLACE PULLDOWN
ENTRY 9033076 !BROKEN !CLOSED
FRM I FIELD
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SEQ 11802654 INTERLACE PULLDOWN
ENTRY 11802676 !BROKEN !CLOSED
FRM I FIELD
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SEQ 15010974 INTERLACE PULLDOWN
ENTRY 15010996 !BROKEN !CLOSED
FRM I FRAME
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SEQ 17703774 INTERLACE PULLDOWN
ENTRY 17703796 !BROKEN !CLOSED
FRM I FIELD
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SEQ 19860702 INTERLACE PULLDOWN
ENTRY 19860724 !BROKEN !CLOSED
FRM I FIELD
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SEQ 20299038 INTERLACE PULLDOWN
ENTRY 20299060 !BROKEN !CLOSED
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SIZ 1920 x 1080
FPS 30000 / 1001
CODED 274
PLAYBACK 274Honestly I do not *expect* anything of eac3to, beyond what it already does correctly which is to demultiplex, and convert audio. It's just that these clips will crash most decoders, indeed AFAIK neuron2's tool is the only one that deals with them, i.e. after indexing you can re-encode or play with VirtualDub.

Given though that eac3to is "step 1" for any manipulation effort, it would be useful I think if it were to ID these as a mix of interlaced & progressive, if possible (that's where I'd hoped your field/frame count might be way to tell).

[setarip_old]

Anyone know why most recent versions (up to and including v.0.34b) of "Clown_BD" reject input of .MKV files, despite showing .MKV to be an acceptable input format?

[madshi]

What about the repeated beginning?
Should also be fixed. Just give it a try...

I have uploaded a 20MB sample clip
Thanks. Will have a deeper look at this when I finally find some time to fix/improve interlaced VC-1 handling.

[tebasuna51]

What about the repeated beginning?

With 3.13 I can reproduce the problem at the beginning, but 3.14 work fine for me.

[Chumbo]

eac3to v3.14 released

http://madshi.net/eac3to.zip

* WAV reading was broken for all but very small files (introduced in v3.13)
Don't forget to update the first post. ;)

[mrr19121970]

Anyone know why most recent versions (up to and including v.0.34b) of "Clown_BD" reject input of .MKV files, despite showing .MKV to be an acceptable input format?

because I don't have any .MKVs to test. Please post the logs (in the Slysoft forum).

I assume that eac3to CLI can handle your files ?

[setarip_old]

@mrr19121970

Hi!

Thanks for getting back to me.I assume that eac3to CLI can handle your files ?Thusfar, I've only used eac3to via your (very impressive) GUI.

Like you, based on the information in post #1 of this thread, I also presume that it can handle MKV as input.

Any further insight that you can provide will be greatly appreciated.

[madshi]

@mrr19121970, your signature says "eacto" instead of "eac3to"... ;)

[mrr19121970]

:o

Changed it now. 52k views on the Slysoft forum, and nobody noticed it.

:o

@setarip_old

like I said, I've no way to test it.

you need to try the CLI version:

"c:\yourpathto\eac3to.exe" "C:\your.mkv" -demux

[setarip_old]

@mrr19121970

Then, perhaps you should include a note in the download that states that this is presently a non-functioning option of your GUI...

[mrr19121970]

@mrr19121970

Then, perhaps you should include a note in the download that states that this is presently a non-functioning option of your GUI...

The GUI is exactly that, it's only an interface, a presentation layer.

If the underlying programs cannot handle a particular (for whatever reason) then you need to show logs so that the author can address the issue.

[leeperry]

guys, I got a pretty annoying problem....hopefully someone will have an idea :o

I've got the german Revolver BD, w/ the english track in DTS HD MA....I've tried to extract the core w/ eac3to and I get quite a lot of static in different places(not loud ones, so it's not saturation :confused: )
w/ the extracted DTS core, the static is worse in Sonic 4.2 than in AC3filter, I've tried to play the DTSHD file w/ Sonic 4.2 : it's full of glitches but it seems that the static is still there..

a friend of mine tried to play the track in Arcsoft and the static was still there, I guess that's the only way to play DTS HD in its full glory on PC ?

I've read that Sonic 4.3 should also play it, but I've never managed to make it open DTS...does it really play DTS HD ?

either there's some DRC or whatever that's saturating the audio, or the guys at the mastering studio didn't notice that the english track was full of static before encoding it :(

[setarip_old]

@leeperry

Hi!or the guys at the mastering studio didn't notice that the english track was full of static before encoding itAre you saying that the same audio problem is observed when you play your original BluRay disc?

[leeperry]

Are you saying that the same audio problem is observed when you play your original BluRay disc?
well yeah, the DTS core on the original disc has the glitches..from what I know if you play DTSHD in PDVD it will only play the core?

only Arcsoft can decode DTSHD fully, as I understand it? I guess I'm SOL, too bad this happens on my favorite movie :o

I wish I could somehow decode it to FLAC w/o the glitches :D

[dorati]

@madshi:
Today I have demux IceAge2 with eac3to v3.14 (-demux).
After Muxing with TsMuxeR to M2TS, the Blu-Ray subtitles Stream are not Sync.
There is a delay of 2 Min 42 Sec. The subtitles are displayed to early...

I did again with eac3to v3.05. There is everything ok.

[jamos]

That is not a proof at all. tsMuxeR might work just fine with DTS and still screw up LPCM muxing. Perfectly possible.


It could also be your receiver which doesn't handle 7.1 LPCM correctly. Or it could be whatever source device you're using. I don't really know where the bug is introduced in your situation. But from what I can see the bug is not caused by eac3to. So there's nothing I can do for you.



Ok thanks, it probably is tsmuxer. But can you tell me the order that you remap the channels by default when you demux DTS-HD 7.1 to pcm using sonic (ie LF = 0, RF=2, etc)? I can manually remap them using switches like -0,1,2,3,4,5,6,7. if I know the order that you remap them then I could just override it to swap the two channels (FR with Center) so it works with tsmuxer.

It is not the reciever Onkyo 806 handles 7.1 lpcm input fine from my Ps3 and sony bd player. I do not beleive it the source either as this problem occurs with 3 different DTS-HD 7.1 sources. Could be tversity but i doubt that.

thanks for any help...

edit: n/m the new version of tsmuxer 1.8.24 can now import 7.1 wav files and it still mixes the Center with the Right Front. Bug is with tsmuxer thanks for your patience. for those using the pcm demux then using pcm2tsmu workaround you can now use .wav output using eac3to and directly import .wav into the new version of tsmuxer works fine with 5.1 sources.

[jamos]

Ok got DTS-HD 7.1 to LPCM 7.1 to work with Tsmuxer 1.8.24 you need to swap the center and right front channels when outputing to .wav

ex.
eac3to app.dts app.wav -0,2,1,3,4,5,6,7

then import video and wav files in tsmuxer and mux to m2ts.

all channels play fine with lpcm playback in m2ts file streaming to ps3.

This is a bug with tsmuxer.:D

Edit: note that using .wav is very buggy still in tsmuxer..it crashes most of the time. am now trying to use .pcm with remapping.

Ok found a workaround: use pcm output but swap channel 1 and 2 then use the pcm file as input to pcm2tsmu to prepare it for tsmuxer. then use tsmuxer and the channels are now correct. until they fix the issue in tsmuxer this will work if you want correctly mapped channels using streaming 7.1 lpcm to ps3.

ex:
eac3to app.dts app.pcm -0,2,1,3,4,5,6,7
pcm2tsmu app.pcm output.pcm -c 8
then use tsmuxer with your video and use output.pcm as your audio..vola 7.1 lpcm streaming..

[Thunderbolt8]

got a problem with a movie, 24fps AVC, 7.1 DTS-HD MA. there are no problems listed in the log at remuxing, but apparently there seems to be a problem with the flac track, when its muxed together with the video its all running at ~2fps and the sound is only garbage. the remuxed video file alone plays fine, so I guess its either a problem of the track (or what eac3to does with it) or perhaps also with mkvmerge. but I went back to mkvmerge 2.2.0 and its still the same. the playing back the flac track alone, muxed to .mka then it has the same gargabe sound as together with the video.

[madshi]

Today I have demux IceAge2 with eac3to v3.14 (-demux).
After Muxing with TsMuxeR to M2TS, the Blu-Ray subtitles Stream are not Sync.
There is a delay of 2 Min 42 Sec. The subtitles are displayed to early...

I did again with eac3to v3.05. There is everything ok.
Argh, so my "PTS -> DTS" hack probably doesn't work. <sigh>

got a problem with a movie, 24fps AVC, 7.1 DTS-HD MA. there are no problems listed in the log at remuxing, but apparently there seems to be a problem with the flac track, when its muxed together with the video its all running at ~2fps and the sound is only garbage. the remuxed video file alone plays fine, so I guess its either a problem of the track (or what eac3to does with it) or perhaps also with mkvmerge. but I went back to mkvmerge 2.2.0 and its still the same. the playing back the flac track alone, muxed to .mka then it has the same gargabe sound as together with the video.
Does the standalone FLAC track play correctly on its own (not muxed into MKV)? If so, eac3to is most probably not at fault.

[Thunderbolt8]

Does the standalone FLAC track play correctly on its own (not muxed into MKV)? If so, eac3to is most probably not at fault.cant get it to work in non remuxed form, vlc and mpc wont play it by just normally opening it

well mpc loads it, but does not output any sound, as the playback does not progress at all, but only stalls and mpc crashes then when trying to move the progress bar. same when I try to load it with 'mplayer.exe movie.flac'

[madshi]

cant get it to work in non remuxed form, vlc and mpc wont play it by just normally opening it
Well, do you have madFlac installed? If yes, try uninstalling and reinstalling it...

[Thunderbolt8]

aerf, that fixed it -.- thanks!

[HanSolo00]

Thanks for adding support for 24.975fps. However, when I test on a blank 100 sec audio clip, EAC3TO seems to use 25fps in place of 24.975fps... since the following produces the same result:

100 second WAV test clip
-24.975 -ChangeTo24.000 produces 104.149s of audio (should be 104.063s or factor of 25/1.001/24)
-25.000 -ChangeTo24.000 produces 104.149s of audio (correctly a factor of 25/24)

The exact FPS for -24.975 option is 25/1.001 = 24.975025

Thanks again I love this tool!

[Snowknight26]

Any reason why one of my 1080i60 streams makes eac3to output 'xxxxxx fields' while the other one 'xxxxxx frames'?

Also, would be nice if when demuxing audio, if eac3to can't correct a, say, -9ms delay, it would add -9ms to the file name.

[anode]

Can you please retry with v2.13? I've done a quick and dirty fix by simply copying the PTS value to the DTS field. I hope that this solves the problem. If not, I'll have to do a proper (and time consuming) fix.

Thank you very much for looking into this, sorry for the late answer...

I've checked the problematic subtitle stream again with 2.13, but the subtitle doesn't display, as with DTS fields set to zero. Only with the DTS entries from tsMuxeR it is displayed. It is a graphical subtitle (colored info tables) which contains 3 OD fields (object definition) in a row (normally only one OD field is enough for some simple text). The easy subtitles (text) are displayed, even with DTS zero, so this is not such a big problem, I think.
PowerDVD displays all subs without problems, DTS zero or with PTS copy (2.13).
If madshi or anybody has some more infos about the relationsship of PTS to DTS in Blu-ray subtitle specs, I would be happy if you let me know. From my own findings the DTS has to be a little earlier (smaller value) than the PTS. Some definition fields like PD (palette info) and END always have DTS info set to zero.

[0xdeadbeef]

If madshi or anybody has some more infos about the relationsship of PTS to DTS in Blu-ray subtitle specs, I would be happy if you let me know. From my own findings the DTS has to be a little earlier (smaller value) than the PTS. Some definition fields like PD (palette info) and END always have DTS info set to zero.
I'd need some help regarding this very topic as well:
http://forum.doom9.org/showthread.php?t=145654

[DrNein]

FYI, for a minimal install of Burning ROM and audio decoder functionality, I gave Nero Micro another go. After thorough removal of all Ahead data (files and registry), Micro 7.11.10.0 was installed with default audio & video options, then the single file restored from previous install: C:\Program Files\Common Files\Ahead\DSFilter\NeEacDec.dll, and the BD/HD DVD license entered in addition to that for the main package (in this case Ultra).

eac3to (v3.14) is installed
Nero Audio Decoder (Nero 7) works fine.

So, no need for ShowTime and all that Home and Indexing bloat (or futzing disabling and removing) afterall. :cool:

[ps3hacker]

Ive just tried to demux Body Of Lies bluray and get these errors.(log attatched)
The bluaray is not a seemless branching movie and is a single m2ts. Also running latest anydvd.

edit I think maybe the disc has mastering errors do to the fact that at the times posted by eac3to as errors, there is glitches in video and audio from the original disk. This is a new movie and is not scratched in any way. I will just return for another

[piratburner]

What is this ???? v02 The video framerate is correct, but rather unusual. ???????

C:\BT>eac3to "Transporter 2 2005 1080p BluRay MPEG-4 AVC DTS-HD 5.1-Mike" 1)
M2TS, 1 video track, 2 audio tracks, 1 subtitle track, 15.052p
1: Chapters, 12 chapters
2: h264/AVC, 1080p24 (16:9)
3: DTS Hi-Res, French, 5.1 channels, 16 bits, 2046kbps, 48khz
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48khz)
4: DTS Hi-Res, English, 5.1 channels, 16 bits, 2046kbps, 48khz
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48khz)
5: Subtitle (PGS), French
v02 The video framerate is correct, but rather unusual.

C:\BT>

[jamos]

Can someone tell me what is the channel order when you convert DTS-HD MA 7.1 to PCM 7.1 using eac3to?

Thanks!

[Snowknight26]

madshi, how hard would it be to implement reading from the other formats that libavcodec handles? Would be nice to be able to decode Vorbis, or MP3, AAC, and more with eac3to. Wishful thinking on my part?

[Steel]

Can someone tell me what is the channel order when you convert DTS-HD MA 7.1 to PCM 7.1 using eac3to?

Thanks!

I don't know if EAC3TO can do it, but wavavimux can.

[Snowknight26]

The channel order should be FL FR FC LFE BL BR SL SR.

[DrNein]

What is this ???? v02 The video framerate is correct, but rather unusual. ???????

Just because it is 24 FPS rather than 23.976. 24 is included in the BD standard but is relatively rare.

[leeperry]

depending on the sountrack being 1.5mbit DTS core or FLAC(from DTSHD in eac3to), the sound levels are much different ?!
I can hardly hear dialogs in DTSHD FLAC and need to increase the center channel to like 140%

I've got all DRC disabled in AC3Filter and it's the same story w/ ffdshow/sonic 4.2 for DTS core, so what gives ?!

it happens in all the DTSHD>FLAC tracks I've built, and forcing dialog normalization in ea3to doesn't do anything(same CRC whether it's enabled or not)

OTOH transcoding LPCM in FLAC gives proper sound levels, identical to DTS core :o

if anyone's got a clue, I'd love to hear it. thanks!

[evdberg]

I am not sure since what version the following problem is (I had V3.12 and just tried V3.14), but eac3to can not read m2ts files anymore. So when I point to the root of the ripped BD, I get the report as usual. But when I add the title index, eac3to reports there is an error reading xxxxx.m2ts. I am using CrossOver on Mac OS-X 10.5. For the record, this has always worked perfectly fine with older versions. CrossOver has not been updated in the meantime.

[jamos]

Madshi I just got this reply from the tsmuxer folks

the order of PCM 7.1 should be L R C LS Rls Rrs RS lfe

is that the order that you put pcm in for 7.1 with eac3to?

[Snowknight26]

5.0 channel DTS can't seem to be converted to AC3.
The AC3 encoder received a non-supported data format (pcm, 5, 16, -).

[TinTime]

Madshi, not sure if this is a bug report or a feature request but anyway...

If I feed eac3to a duff audio file it reports "The format of the source file could not be detected." but the return code is 0. Could it be changed to non-zero please? It seems to be rc 1 for other source file errors.

Thanks.

[tebasuna51]

5.0 channel DTS can't seem to be converted to AC3.

You can use the external encoder Aften:

eac3to source.dts stdout.wav | aften -b 640 -readtoeof 1 - output.ac3

[lchiu7]

I have Nero 7 installed and just picked up a copy of Arcsoft TMT a friend had and didn't want, primarily to play HD-DVD's since my included PDVD7 with my LG BD/HD-DVD drive only plays BD.

Anyway eac3to can't see either giving the following error messages

Nero Audio Decoder (Nero 7) is not working correctly
http://www.nero.com/eng/store-blu-ray.html
CAUTION: You need Nero 7. Nero 8 won't work with eac3to.
ArcSoft DTS Decoder doesn't seem to be installed
http://www.arcsoft.com/products/totalmediatheatre

According to what I have read the DTS decoder is via the filter ASAudioHD.ax which according to this DS filter chain, is installed

http://www.datamix.info/capture.jpg

Also I know Nero 7 is installed since I use it regularly.

This is on Vista Home Premium 32

Anything else I should be looking out for?

Thanks

[edit]

OK some progress. Looks like a path or environment variable issue. Copied all the DLL's from Arcsoft's directory to the eac3to directory and now the Arcsoft DTS decoder works fine. Still no joy with Nero though

Out of interest what DS filter is eac3to looking for when it checks for Nero?

Thanks

[EDIT again!]

Have gone through this thread again and it seems you need not only Nero 7 but the nero HD-DVD plugin. That's no longer been sold so I guess I am stuck with using libav which seems to be fine

[dchard]

I have 6 mono wavs (synced), and now I don't know how can I make a 5.1 FLAC file from it.

Can somebody help me with this?

Thank you!

Dchard

[tebasuna51]

I have 6 mono wavs (synced), and now I don't know how can I make a 5.1 FLAC file from it.

You can use WaveWizard or Sox (command line) to merge the 6 channels in one multichannel wav file, and Flac or ea3to to convert to flac.

Also you can use an .avs (AviSynth script) and BeHappy to convert directly to flac without the intermediate multichannel wav file.

[mrr19121970]

@mrr19121970

Then, perhaps you should include a note in the download that states that this is presently a non-functioning option of your GUI...

It's working now.

http://forum.slysoft.com/showthread.php?t=25818&page=56

[BLKMGK]

Having an issue with a specific movie - The Game Plan. The video appears to rip just fine but the audio track is a mess :( I have gone after the DTS and the AC3 5.1 tracks to no avail. When I rip it I get normal fine audio and then a few minutes into the track it loops back to the beginning! If I FFWD a bit I get normal dialog again - then a few mins later it loops back to the starting music again. After about the 3rd time I gave up :( I have now ripped it 3 times trying various things to include two copies of the movie to no avail. Has anyone else done this one and what issues if any did you encounter? I am on 3.14 and haven't ever seen this before. Ideas or suggestions?

Here's my log -> http://pastebin.com/m2d93e154

[TinTime]

If I feed eac3to a duff audio file it reports "The format of the source file could not be detected." but the return code is 0. Could it be changed to non-zero please? It seems to be rc 1 for other source file errors.

Thanks.

Following on from this I've been trying to work out why eac3to doesn't like some (but not all) of my input files. I used eac3to to create a wav file from a pcm input:
eac3to v3.14
command line: "D:\Vtemp\programs\eac3to\eac3to.exe" "test.pcm" "E:\Vtemp\test1.wav" -6 -24 -little -96000 -override -log=1.txt
------------------------------------------------------------------------------
RAW/PCM, 5.1 channels, 1:38:26, 24 bits, 13824kbps, 96khz
Reading RAW/PCM...
Writing WAV...
Creating file "E:\Vtemp\test1.wav"...
The original audio track has a constant bit depth of 24 bits.
Caution: The WAV file is bigger than 4GB. <WARNING>
Some WAV readers might not be able to handle this file correctly. <WARNING>
eac3to processing took 9 minutes, 41 seconds.
Done.

If I then run eac3to on the output wav file I get:
eac3to v3.14
command line: "D:\Vtemp\programs\eac3to\eac3to.exe" "E:\Vtemp\test1.wav" -log=t1.txt
------------------------------------------------------------------------------
The format of the source file could not be detected. <ERROR>

I've been trying to find out why this file does not work and other apparently similar ones do and the difference seems to be the amount of silence at the beginning of the file. This file contains 11866608 bytes = 6.86725 seconds of silence at the beginning. I tried editing the first six samples to non-zero values and it's then detected by eac3to.

I've created two test files (http://www.sendspace.com/file/2wi285) (the original is 10GB) containing the first 15MB of the wav file. I've edited the first six samples of test2.wav and it can be processed. I've left test3.wav as is and it can't be detected. The lengths in the header remain as they are from the original file. I found that if I change the lengths in the RIFF and data chunks in test3.wav to zero then eac3to will detect it. However this doesn't work with the full sized original.

Thank you!

[tebasuna51]

Yes, seems a bug.

[Snowknight26]

eac3to can't seem to recognize AAC files it made.

[tebasuna51]

eac3to can't seem to recognize AAC files it made.
Eac3to recognize (and can decode with Nero, without plugins) .aac files. Is AAC format with ADTS headers.

Can't recognize AAC inside mp4 container (.mp4 or .m4a).

The use of the faad decoder (aac/m4a, inside libav) or NeroAacDec is in the todo list of madshi like he say.

Edit: By the moment you can use the Faad decoder included in UsEac3to (http://forum.doom9.org/showthread.php?t=145574)

[LeXXuz]

I have Nero7 on my system to use the e-ac3 decoder. I want to get rid of Nero, because I use other tools for burning stuff and Nero just occupies to much space and resources.

How can I get rid of Nero but keep the e-ac3 decoder and/or the necessary libraries/files? Is that possible? :)

[laserfan]

Easiest just to de- and re-install with only Showtime, taking care to skip the Scout & other unneeded junk, though I think there remains an indexing service you still have to Disable by hand.

[Snowknight26]

Can't recognize AAC inside mp4 container (.mp4 or .m4a).

I hadn't even noticed that the files it was making were .m4a. :o

Thanks for the continued support tebasuna51.

Edit: I just noticed that eac3to displays a 2.0 channel AAC track in mkv as being 24KHz when its really 48KHz.

[DragonM]

Hey, I got a 20gig m2ts file containing several episodes in it. I want to demux the video and audio separately to mkv and flac.

Can I split m2ts files by chapters? If so, how?

[tebasuna51]

Edit: I just noticed that eac3to displays a 2.0 channel AAC track in mkv as being 24KHz when its really 48KHz.

Then is a AAC-HE (with Spectral Band Replication) the high frequencies (24-48KHz) are in the SBR part, not always recognized. With the appropriate decoder you can recover the 48 KHz.

[Snowknight26]

It's AAC LC-SBR, 2 channel, 48KHz according to YAMB.

[tebasuna51]

LC-SBR and HE is the same.

The LC part is 24KHz (and some decoders/players only recognize this) and the high frequency go in SBR part.
Is not easy know when a aac audio file have SBR or not, in the ADTS header from aac the frequency are marqued as 24Khz.

BTW the support for AAC in eac3to is still limited.

[narshorn]

Dear Madshi and all sound addicts @ doom9,

Thanks for this wonderful program eac3to. It's a really handsome tool for audio extraction and conversions !
I really like the command line utility. Plus I like the idea of bit perfect decoding from various formats ... :)

However, I have a small problem when opening .wavs created by eac3to in extraction;
when I try to open some 24/96 or 24/192 wavs in r8brain (wavs created by eac3to), r8brain pops up an error.
It's relative with the wav header I guess. When I open the "defective" wav in Wavelab, I have no problem, so I save the file again
in Wavelab without changing the bitrate or sample frequency, i.e. identical file, but now I can open them in r8brain and downsample
them in very high quality mode ... any thoughts on this ? Should I add a particular switch while doing the .wav extraction from .mlp
to get a r8brain compatible header ? Thanks for your thoughts on this.

Best regards,

narshorn

[kypec]

Try to add -simple switch, it should produce old fashioned WAV header, probably better compatible with r8brain.

More info on undocumented switches of this wonderful program can be found here (http://forum.doom9.org/showthread.php?t=145066)

[narshorn]

Try to add -simple switch, it should produce old fashioned WAV header, probably better compatible with r8brain.

More info on undocumented switches of this wonderful program can be found here (http://forum.doom9.org/showthread.php?t=145066)

Thanks for the info :) will try it in my next conversions :)

Regards,

narshorn

[Pl4yit]

Just wanted to post the following issue that I found using eac3to (v3.14). The sample file that I have used can be found in the link here below.

http://rapidshare.de/files/46162759/extract.h264.html

- The sample is a part of a h264 file extracted directly from blu-ray using eac3to
- When performing the following set of commands on this sample:

eac3to extract.h264 extract.mkv
eac3to extract.mkv 1: extractfrommkv.h264


then you will notice that the result file "extractfrommkv.h264" is different from the original "extract.h264".

Looking a bit more in detail there is a sequence of 162 bytes added at specific places in the file, starting with the following 34 bytes:

00 01 27 64 00 29 AC 7B 01 E0 08 9F 97 FF 00 01 00 01 10 00 00 3E 90 00 0B B8 08 40 00 00 00 01 28 EA

The sequence is the same throughout the file and removing it will give back the original "extract.h264".

The issue has been introduced as of v3.07 (v3.06 works fine), and I think it happens only with AVC/H.264 encoded files with 2 Reframes. Did the test on 4 different films (directly from blu-ray), all with 2 reframes, and they all had the issue, i.e. a lot of artifacts during very fast-moving scenes.
Did some other tests with VC1, MPEG2 and higher-frame H.264 encoded files and did not find a difference between the files before and after the 2 eac3to steps provided here above.

Sorry for the long thread, but I hope this is enough information to easily pin-point the problem, otherwise I'm happy to provide more :D

Pl4yit

[shon3i]

I have problem when i try to demux TsMuxer created Blu-Ray.

M2TS, 1 video track, 1 audio track, 1 subtitle track, 1:24:34, 24p /1.001
1: Chapters, 16 chapters
2: h264/AVC, 1080p23.975 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48khz
4: Subtitle (PGS), English
v02 The video bitstream is encoded in a non-standard framerate.
v02 The video bitstream framerate field doesn't seem to match the timestamps.

but that is not true, i encode to H264 with x264 and input avs have parametar fps=23.976, all programs and players reconize as 23.976, aslo i tryed with checking changefps in TsMuxer and selecting 24000/1001, but still eac3to report message.

Second question is: why eac3to when extracting chapters not include final time anymore?

[BLKMGK]

Having an issue with a specific movie - The Game Plan. The video appears to rip just fine but the audio track is a mess :( I have gone after the DTS and the AC3 5.1 tracks to no avail. When I rip it I get normal fine audio and then a few minutes into the track it loops back to the beginning! If I FFWD a bit I get normal dialog again - then a few mins later it loops back to the starting music again. After about the 3rd time I gave up :( I have now ripped it 3 times trying various things to include two copies of the movie to no avail. Has anyone else done this one and what issues if any did you encounter? I am on 3.14 and haven't ever seen this before. Ideas or suggestions?

Here's my log -> http://pastebin.com/m2d93e154

Any ideas on this guys? I've never had a soundtrack do this and the loops do not appear to be on chapter or segment changes. I'm totally stuck although I have a few (dozen) old versions of eac3to I may try just for grins to see if this is something goofed recently... So far as I can tell this movie hasn't presented AnyDVD-HD any issues.:confused:

[DrNein]

Easiest just to de- and re-install with only Showtime, taking care to skip the Scout & other unneeded junk, though I think there remains an indexing service you still have to Disable by hand.

Alas, all the Nero Home junk (Scout, Indexing, and such) is installed regardless of all the extras being deselected. But yes, it can all be disabled and deleted manually.

However, for the smallest starting point follow my directions here:

http://forum.doom9.org/showpost.php?p=1261179&postcount=8538

After that, Nero Burning ROM could be deleted to save a measly 20MB, essentially just keeping the Common Files including activation and filters (filters could potentially be trimmed down a bit but they are only 8MB as-is).

[setarip_old]

@BLKMGK

Hi!

It appears that you've used the wrong .MPLS- 00139. You should use 00138. Note the playtimes of each:

D:\Video\eac3to>eac3to x:
1) 00139.mpls, 2:07:03
[92+1+93+3+94+5+95+7+96+9+97+11+98+13+99+15+16+100+18+101+20+102+22+103+24+104+26+105+28+106+30+107+32+108+109+35+110
+37+111+39+112+41+113+43+114+45+115+47+116+49+117+51+118+53+119+55+120+57+121+59+122+61+123+63+124+65+125+67+126+69+127+
71+128+73+129+75+130+77+78+131+80+132+82+133+84+134+86+135+88+136+90+137].m2ts
- Chapters, 20 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- RAW/PCM, English, multi-channel, 48khz
- AC3, English, multi-channel, 48khz
- AC3, English, multi-channel, 48khz
- AC3, English, multi-channel, 48khz
- AC3, English, stereo, 48khz

2) 00138.mpls, 1:50:15
[0+1+2+3+4+5+6+7+8+9+10+11+12+13+14+15+16+17+18+19+20+21+22+23+24+25+26+27+28+29+30+31+32+33+34+35+36+37+38+39+40+41+
42+43+44+45+46+47+48+49+50+51+52+53+54+55+56+57+58+59+60+61+62+63+64+65+66+67+68+69+70+71+72+73+74+75+76+77+78+79+80+81+
82+83+84+85+86+87+88+89+90+91].m2ts
- Chapters, 20 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- RAW/PCM, English, multi-channel, 48khz
- AC3, English, multi-channel, 48khz
- AC3, French, multi-channel, 48khz
- AC3, Spanish, multi-channel, 48khz
- AC3, English, stereo, 48khz

[laserfan]

...for the smallest starting point follow my directions here:

http://forum.doom9.org/showpost.php?p=1261179&postcount=8538

After that, Nero Burning ROM could be deleted...But Nero Micro is not a "product" afaict, does not come from Ahead, and thus all downloads of same are of suspicious origin.

Correct me, if I'm wrong, and/or provide a link to a good, legal copy of Micro?

[BLKMGK]

@BLKMGK

Hi!

It appears that you've used the wrong .MPLS- 00139. You should use 00138. Note the playtimes of each:



Play time is listed as 110minutes on the box. The dialog is perfectly synched and correct up until the glitch occurs. When this occurs the audio from the very beginning of the movie - the Disney fireworks music - plays for a bit and then it jumps back to the correct point in the movie and dialog\effects\etc. is all correct and in synch. It's as if incorrect portions of the movie are being dropped into the midst of correct portions. Honestly I figured one of those features was say the Director's cut and the other was the release cut. I ripped both of them initially but erased them both and reripped with another copy of the media hoping for a fix, I can rip the 2hour version to see what it is easily enough I guess. The box doesn't mention that there are different cuts and I've nto tried it in my PS3 <sigh> In any case - sumthin' ain't right with the way it's ripping....

[DrNein]

But Nero Micro is not a "product" afaict, does not come from Ahead, and thus all downloads of same are of suspicious origin.

Correct me, if I'm wrong, and/or provide a link to a good, legal copy of Micro?

Indeed, it is not from Ahead. But like nLite or such, valid licenses are still required.

[NanoBot]

But Nero Micro is not a "product" afaict, does not come from Ahead, and thus all downloads of same are of suspicious origin.

Correct me, if I'm wrong, and/or provide a link to a good, legal copy of Micro?

As DrNein already stated, Nero 7 Micro is not an official product from Ahead. Instead, it is a slimmed down installer provided from a third party. If you do not trust this installer, or if you are in doubt about copyright or license problems using such an installer, you possibly prefer to build your own installer using the NeroLiteSDK available here:

http://updatepack.nl/downloads/nero-lite/

With that SDK you are able to generate your own slimmed down installer from the latest official Nero7 update package.

I am using a similar configuration like DrNein described here:

http://forum.doom9.org/showpost.php?p=1261179&postcount=8538

and I neither have problems to use the Nero EAC3 decoder together with eac3to nor do I have any other problems resulting from the use of Nero7Micro.

C.U. NanoBot

[XhmikosR]

Or you can use this (http://updatepack.nl/) version of Nero Micro English which already has all the necessary files to decode True-HD, E-AC3 etc. Of course, a valid license is needed to use it.
The changes in the above version are not in the "official" Lite/Micro, but hopefully they will be added when the creator of Nero Lite has some free time.

EDIT:
And of course the above setup is clean.

EDIT2:
All of the above have been implemented in the "official" Nero Micro/Lite.

[laserfan]

Or you can use this (http://ikwuva.bay.livefilestore.com/y1pla02NXBT_OjrqNpcCkCEcDyxgSOR8MJP9IdIW-h0cfFbcapxmnusVCoR-v8kSurLZyUgldg4g_tIq25ABndmgw/Nero-7.11.10.0_english_micro.exe) version...The version I bought is a little older but I will give this a try, thanks. What is different about this build vs. "official"?

[XhmikosR]

Just the filters needed for E-AC3, True-HD and DTS Express decoding and some minor improvements to the setup script. I have already contacted the creator of Nero Lite in order to include the changes I've made, but he doesn't seem to have a lot of free time.

[shon3i]

I have problem when i try to demux TsMuxer created Blu-Ray.

M2TS, 1 video track, 1 audio track, 1 subtitle track, 1:24:34, 24p /1.001
1: Chapters, 16 chapters
2: h264/AVC, 1080p23.975 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48khz
4: Subtitle (PGS), English
v02 The video bitstream is encoded in a non-standard framerate.
v02 The video bitstream framerate field doesn't seem to match the timestamps.

but that is not true, i encode to H264 with x264 and input avs have parametar fps=23.976, all programs and players reconize as 23.976, aslo i tryed with checking changefps in TsMuxer and selecting 24000/1001, but still eac3to report message.

Second question is: why eac3to when extracting chapters not include final time anymore?
Did somebody have answers for my questions?

[Chumbo]

Did somebody have answers for my questions?
Did you try forcing the framerate via a CLI switch? What's the exact command line? Always helpful to have the full command line.

[shon3i]

Did you try forcing the framerate via a CLI switch? What's the exact command line? Always helpful to have the full command line.
Yes i tryed but not helped. I use classic cmd eac3to "k:\b13" "k:\video.mkv"

[Chumbo]

Yes i tryed but not helped. I use classic cmd eac3to "k:\b13" "k:\video.mkv"
Put a sample for madshi to look at. Your command line is not workable. It doesn't know what to do if you don't tell it what play list or track and so on. The exact command line is usually at the top of the log file which is helpful to include with any log entries.

[narshorn]

Any reason to why eac3to with -decodeHdcd switch gets 6 dB gain lower than standard decoding ? If so, is it OK to add +6dB as a switch while doing conversions ?

Best regards,

narshorn

[G_M_C]

People, i've got a question, that's (hopefully) just a theoretical one:

Take, for instance, a 5.1 DTS-HD Master Audio stream that has a bitrate of 6900 kbps, with a core DTS stream 5.1 @ 1536 kbps.

The way i understand the DTS-HDMA system works (simplified), is that a 1536 kbps DTS stream is encoded as the core, and that the additional HDMA data contains the differences between the original stream and the DTS stream. When decoded the DTS + the differences end up being completely identical to the source/original stream.

But if I reason this on this / see this correctly (here's the theoretical bit) this would mean that you don't necessarily have to have "the best core track you can make". You could change / master the core-track marginally different, so that the audience that has to use this core-track has the better listening experience. This intentional slight difference can be made up in the (extended) HDMA part of the stream, where you "iron out" this (intentional) slight differences.

So far the theoretical part of this question. The question i'd like to ask is derived from my assumption that a core DTS-stream doesn't necessarily be the best core-track they could make. Wouldn't it be better to fully decode the complete DTS-MA track, and use that (lossless) track to encode a 1536 DTS track yourself ? As opposed to just demuxing the core track off course ?

[easy2Bcheesy]

Hi,

I'm using eac3to to turn .ac3 files into multi-channel WAVs. Decoding is being handled by libav/ffmpeg. The only problem I have is that I'm only getting 2.1 channel output. The command line text refers to remapping channels and reducing depth from 64 to 24-bit.

How can I make libav/ffmpeg give me all six channels instead of just three?

EDIT: A bit of a false alarm. What appears to be happening is that the WAV is absolutely fine. However, when I import it into Adobe Premiere, I am only seeing waveforms on three of the six channels. If I actually play the WAV, I hear it from the centre and rears exactly as you would expect. Even when I edit the audio, export it and re-import it, I still only see three channels, even though sound is coming from all six. Another Adobe cock-up? Is there a freeware tool out there that will show me waveforms on a six channel WAV?

[TinTime]

You can try Audacity (http://audacity.sourceforge.net/). That will show you all six channels.

[StephenB]

Hi,

Thx for the great tool - I've been using it for about a year now, and am quite impressed.

I'm trying to use EAC3TO to process an 8 bit PCM file. The goal is to resample it to 48 kHz and transcode it.

Where I'm having difficulty is that EAC3TO won't expand the bit depth beyond 8 bits. Here's a simple log:

eac3to v3.14
command line: c:\vid2eva\tools\eac3to\eac3to audio_2_l.wav test.flac -16
------------------------------------------------------------------------------
WAV, 1.0 channels, 0:00:10, 8 bits, 88kbps, 11khz
Reading WAV...
Encoding FLAC with libFlac...
The original audio track has a constant bit depth of 8 bits.
Creating file "test.flac"...
eac3to processing took 1 second.
Done.

If I specify the resampling, I end up with a clipped output that is totally unusable.

eac3to v3.14
command line: c:\vid2eva\tools\eac3to\eac3to audio_2_l.wav test.flac -16 -resampleto48000 -r8brain
------------------------------------------------------------------------------
WAV, 1.0 channels, 0:00:10, 8 bits, 88kbps, 11khz
Reading WAV...
Resampling to 48khz...
Reducing depth from 64 to 24 bits...
Encoding FLAC with libFlac...
Clipping detected, a 2nd pass will be necessary. <WARNING>
Creating file "test.flac"...
The original audio track has a constant bit depth of 8 bits.
The processed audio track has a constant bit depth of 24 bits.
Starting 2nd pass...
Reading WAV...
Resampling to 48khz...
Reducing depth from 64 to 24 bits...
Encoding FLAC with libFlac...
The processed audio track has a constant bit depth of 24 bits.
Creating file "test.flac"...
eac3to processing took 4 seconds.
Done.

This fails in the same way, no matter what output format I choose. It seems to me that EAC3TO is not expanding the bit depth properly.

If I first convert to FLAC with ffmpeg

ffmpeg -i Audio_2_L.wav -acodec flac -f flac test.flac

ffmpeg will convert the bit depth to 16 bits.

Resampling with EAC3TO to 48000 kHz then works ok - so the problem is not with the resampling itself.

[tebasuna51]

Try with the correct parameter to change the bitdepth, because:

...
-down16 downconvert decoded audio data to 14..23 bit
...
-16 PCM file is '16' or '24' bit

[SomeJoe]

But if I reason this on this / see this correctly (here's the theoretical bit) this would mean that you don't necessarily have to have "the best core track you can make". You could change / master the core-track marginally different, so that the audience that has to use this core-track has the better listening experience. This intentional slight difference can be made up in the (extended) HDMA part of the stream, where you "iron out" this (intentional) slight differences.

So far the theoretical part of this question. The question i'd like to ask is derived from my assumption that a core DTS-stream doesn't necessarily be the best core-track they could make. Wouldn't it be better to fully decode the complete DTS-MA track, and use that (lossless) track to encode a 1536 DTS track yourself ? As opposed to just demuxing the core track off course ?

Well, there's two parts to your question here.

First is your question about using virtually any core track along with the HDMA portion of the stream to end up with lossless audio. Yes, that is correct. Theoretically, I could use a 768Kbps core track, and then use more HDMA information and still end up with lossless audio. However, this is not in the best interest.

Your goal with lossless audio is still to reduce the bitrate as much as you can (i.e. get the best compression ratio while still maintaining lossless audio). The core portion of the track is a sophisticated encoding algorithm that implements perceptual encoding, temporal similarities, etc. The HDMA portion is simple by comparison, and just consists of compensation information. Thus, the more of the information that you can put in the core track, the less information that has to be in the HDMA portion, and therefore the higher the compression ratio.

So yes, you could use a DTS 768kbps as the core and expand the HDMA information and still end up with lossless audio. But the bitrate of that resulting DTS-HD MA track would end up being higher than a DTS-HD MA track made with a 1536kbps core.

Thus, it's in the best interest of reducing bitrate to make the best core possible.

The second part of your question is in regards to how to encode if you want to end up with a lossy 1536kbps DTS track: 1) Extract the core, or 2) encode your own 1536kbps DTS track using the decoded lossless audio.

Well, the core was originally made off the studio master. Assuming the decoded DTS-HD MA lossless audio is equal to the studio master, if you encode your own core (with a licensed, official DTS encoder), you should end up with an identical core.

Thus, I think it would be easier to just extract the given core rather than encode your own.

[StephenB]

Try with the correct parameter to change the bitdepth, because:

...
-down16 downconvert decoded audio data to 14..23 bit
...
-16 PCM file is '16' or '24' bit

Using -down16 results in the same behavior. On the first command (w/o resampling) I still get a bit depth of 8 -

Audio
Format : FLAC
Format/Info : Free Lossless Audio Codec
Duration : 10s 99ms
Bit rate mode : Variable
Bit rate : 31.4 Kbps
Channel(s) : 1 channel
Sampling rate : 11.024 KHz
Resolution : 8 bits
Stream size : 38.7 KiB (100%)

With resampling, I still get clipping and an unusable output.

[rik1138]

Here's an easy request:

When I convert a 5.1 stream to WAV files, it names the Surround channels .SL.wav and .SR.wav. The encoding software I use for Dolby and DTS expect .LS.wav and .RS.wav (the software will auto-find all the files in their correct channel locations if they are named like this, and it seems to be the industry standard...) Any chance of swapping those letters in a future update? :)

No big deal, obviously... It only takes a few seconds to rename them after all, I just thought I'd suggest it...

I haven't tried to convert a 7.1 to WAVs (assuming eac3to does that), but the encoding software looks for LSS (Left Surround Side), RSS, LSR (Left Surround Rear) and RSR for the surrounds in that case...

Rik

[tebasuna51]

Using -down16 results in the same behavior. On the first command (w/o resampling) I still get a bit depth of 8
Is the expected behaviour, use the minimum bith depth.

With resampling, I still get clipping and an unusable output.
This is a problem. Seems the resample don't work with 8 bit samples.
Use Audacity, WaveWizard or Sox to change the bitdepth, for instance:
sox mono8-11.wav -2 mono16-11.wav

[StephenB]

This is a problem. Seems the resample don't work with 8 bit samples.
Use Audacity, WaveWizard or Sox to change the bitdepth, for instance:
sox mono8-11.wav -2 mono16-11.wav

Or ffmpeg, which is what I am using in the script already.

Though I have a workaround, it seems to me it is a bug, so I am reporting it.

As far as expected behavior goes, it would be useful to be able to specify that you need a 16 or 24 bit depth, without needing to reprocess with another tool.

BTW Madshi, it would be very friendly to batch files if EAC3TO switched to r8brain automatically when ssrc gets tempermental.

[G_M_C]

Well, there's two parts to your question here.

First is your question about using virtually any core track along with the HDMA portion of the stream to end up with lossless audio. Yes, that is correct. Theoretically, I could use a 768Kbps core track, and then use more HDMA information and still end up with lossless audio. However, this is not in the best interest.

Your goal with lossless audio is still to reduce the bitrate as much as you can (i.e. get the best compression ratio while still maintaining lossless audio). The core portion of the track is a sophisticated encoding algorithm that implements perceptual encoding, temporal similarities, etc. The HDMA portion is simple by comparison, and just consists of compensation information. Thus, the more of the information that you can put in the core track, the less information that has to be in the HDMA portion, and therefore the higher the compression ratio.

So yes, you could use a DTS 768kbps as the core and expand the HDMA information and still end up with lossless audio. But the bitrate of that resulting DTS-HD MA track would end up being higher than a DTS-HD MA track made with a 1536kbps core.

Thus, it's in the best interest of reducing bitrate to make the best core possible.

The second part of your question is in regards to how to encode if you want to end up with a lossy 1536kbps DTS track: 1) Extract the core, or 2) encode your own 1536kbps DTS track using the decoded lossless audio.

Well, the core was originally made off the studio master. Assuming the decoded DTS-HD MA lossless audio is equal to the studio master, if you encode your own core (with a licensed, official DTS encoder), you should end up with an identical core.

Thus, I think it would be easier to just extract the given core rather than encode your own.

I've tried the 2 methods on a DTS-MA track (extract core and remuxing myself). The reencoded channel seemd to be better, better balanced between channels, and more pronounced LFE response. But i'm keeping the both to let someone else choose between the two, without letting them know wich is wich.

[evdberg]

Regarding below problem, where to get older versions of eac3to? Just to see when the problem is introduced and see if eac3to is the source of the problem at all.

I am not sure since what version the following problem is (I had V3.12 and just tried V3.14), but eac3to can not read m2ts files anymore. So when I point to the root of the ripped BD, I get the report as usual. But when I add the title index, eac3to reports there is an error reading xxxxx.m2ts. I am using CrossOver on Mac OS-X 10.5. For the record, this has always worked perfectly fine with older versions. CrossOver has not been updated in the meantime.

[Thunderbolt8]

arcsoft totalmedia theatre 3 is out, does anyone know if there could be benefits for eac3to from it?

[honai]

Regarding below problem, where to get older versions of eac3to? Just to see when the problem is introduced and see if eac3to is the source of the problem at all.

http://www.videohelp.com/tools/eac3to/old-versions#download

[deathlord]

How do I disable the 2nd pass? I have a file with discontinuities. Eac3to doesn't care and finishes processing. The result has minor picture and audio dropouts I can easily live with. However, eac3to erroneously sees an audio delay and starts a second pass. Afterwards the audio is no longer in sync from the first discon on. I am pretty sure this would not happen without the second pass. But that is automatically executed and I don't know how to prevent that.
Unfortunately the newest version without automatic 2nd pass (2.80) stops at the discons (even with -ignorediscon switch).

[evdberg]

@honai,
Thanks! After some testing it turns out the problem is introduced in V3.06

@Madshi,
What is changed in V3.06 that can explain this behavior? So in short, eac3to can perfectly read the contents of a BD, but it can not read the m2ts files (using CrossOver on Mac OS-X).

[IanD]

Is there any way to demux a PIP video and its associated soundtrack and keep them in-sync?

Whenever I have attempted to do this, the audio is always wildly out of sync. I'm assuming that either the video or audio is continuous, whilst the corresponding audio or video isn't: often there are long stretches of blank video between the PIP segments.

[saint-francis]

Got an issue with Terminator, the SaraConnor Chronicles. eac3to failed to detect an entire episode. There are three on there and eac3to only found 2. I managed to demux the m2ts file for the episode, but it lacked a chapter file.

PLAYLIST and CLIPINF folder can be found here. (http://www.mediafire.com/?sharekey=910d230ea6085fa036df4e8dca1419699b7d209600d73463b8eada0a1ae8665a)

Thanks madshi for all your work.

[honai]

No post in this thread from madshi for a while. Does this mean he sold the rights for eac3to to SlySoft for their new HD transcoding app? ;)

[rik1138]

@Madshi
I've discovered (well, I've been told about, technically) an issue with eac3to recognizing original, unused DTS-HD files. By unused, I mean the files created by the DTS-HD Master Audio Suite (the software that the studios use to actually create the DTS-HD MA audio). The DTS software creates files with an extra 140 bytes of header information at the beginning of the files. This information is stripped off when the DTS is used on a final disc, so if you rip DTS off a Blu-Ray disc, this data is not there. Thus, eac3to doesn't recognize a DTS file with this information... The header isn't vague, the first 8 characters are DTSHDHDR. It would be great if the software could be updated to recognize DTS files with this header information still attached.

I encoded a 7.1 test stream for someone else on the forum, and a member discovered the problem when trying to use eac3to on the file. The files are here:

http://www.sendspace.com/file/mz71xx

There's a DTS-HD High Resolution stream (I don't see this used very often, basically DTS that can go up to a 3mb bitrate), and a DTS-HD Master Audio stream included.

Thanks, Rik

[3ngel]

Hi,

just one question.

How i tell eac3to to use the maximum compression for a .flac destination encoding?

If it is not possible can a swith be added to do the thing?

Thank you very much

[Snowknight26]

How i tell eac3to to use the maximum compression for a .flac destination encoding?


It already does.

[3ngel]

No, because i've recompressed the flac from eac3to, with SoundOut avisynth plugin and the result is 500meg smaller, so i think the default is to use "medium" compression.

[Inspector.Gadget]

Quick question about eac3to and audio delay in Matroska files: The other day I ripped a DVD and then muxed it directly to Matroska without re-encoding, planning to encode it later. I checked the audio delay with DGIndex before muxing and applied the proper delay at mux. eac3to, when run without any options against the Matroska file, reports the audio delay as -301 ms. This sounds like too much of a delay, but off hand I can't remember what the audio delay reported by DGIndex was. Does anyone know how accurate eac3to's parsing of matroska delay values is?

Eduit: Mkvinfo -v shows that the audio starts 16ms after the video, which jogs my memory and I think was the delay I applied at muxing. What's going on here?

[Thunderbolt8]

No, because i've recompressed the flac from eac3to, with SoundOut avisynth plugin and the result is 500meg smaller, so i think the default is to use "medium" compression.perhaps the source was a 24-bit file and that plugin only compressed it to 16-bit?

[3ngel]

perhaps the source was a 24-bit file and that plugin only compressed it to 16-bit?

No, checked. 24bit in, 24 bit out.
81% from eac3to, 68% from SoundOut.

[StephenB]

How do I disable the 2nd pass? I have a file with discontinuities. Eac3to doesn't care and finishes processing. The result has minor picture and audio dropouts I can easily live with. However, eac3to erroneously sees an audio delay and starts a second pass. Afterwards the audio is no longer in sync from the first discon on. I am pretty sure this would not happen without the second pass. But that is automatically executed and I don't know how to prevent that.
Unfortunately the newest version without automatic 2nd pass (2.80) stops at the discons (even with -ignorediscon switch).

I would also like a switch that suppressed the second pass.

[Snowknight26]

No, because i've recompressed the flac from eac3to, with SoundOut avisynth plugin and the result is 500meg smaller, so i think the default is to use "medium" compression.

http://forum.doom9.org/showpost.php?p=1222771&postcount=7316

[saint-francis]

Got an issue with Terminator, the SaraConnor Chronicles. eac3to failed to detect an entire episode. There are three on there and eac3to only found 2. I managed to demux the m2ts file for the episode, but it lacked a chapter file.

PLAYLIST and CLIPINF folder can be found here. (http://www.mediafire.com/?sharekey=910d230ea6085fa036df4e8dca1419699b7d209600d73463b8eada0a1ae8665a)

Thanks madshi for all your work.

Disk 2 is even worse. It doesn't list any independent episodes.

CLIPINF and PLAYLIST files for disk 2 here (http://www.mediafire.com/?sharekey=910d230ea6085fa036df4e8dca14196954ce36e38a4b5ed3b8eada0a1ae8665a)

[3ngel]

http://forum.doom9.org/showpost.php?p=1222771&postcount=7316

The sense of that post is the opposite. Madshi doesn't want to add "less" compression, 'cause "speed is good enough".

In this sense i think Madshi didn't use a "Ultra" level to do a speed/compression balancing.

But in my case (and i think in other cases too at this point), 500 megs smaller are more than a weight difference.

[dev84]

Hi, simple question:


DTS Master Audio, 5.1 channels, 16 bits, 48khz
(core: DTS, 5.1 channels, 16 bits, 1509kbps, 48khz)
Decoding with ArcSoft DTS Decoder...


does it mean that source file is decoded as lossless or only as lossy core?
thx

[tebasuna51]

...does it mean that source file is decoded as lossless or only as lossy core?

ArcSoft decode DTS-MA as lossless.

With -libav only the core.

[TinTime]

The sense of that post is the opposite. Madshi doesn't want to add "less" compression, 'cause "speed is good enough".

In this sense i think Madshi didn't use a "Ultra" level to do a speed/compression balancing.

But in my case (and i think in other cases too at this point), 500 megs smaller are more than a weight difference.

Madshi's post suggests he's using flac's --best compression level.

If you de-compress the flac files produced by eac3to and SoundOut are the resulting wav files identical?

[Thunderbolt8]

The sense of that post is the opposite. Madshi doesn't want to add "less" compression, 'cause "speed is good enough".but isn't "max compression" max compression already?

[dev84]

ArcSoft decode DTS-MA as lossless.

With -libav only the core.

Thank You

[DoomBot]

Any update on being able to extract forced subtitles only?

I would love to see this feature added it would make things so much easier.

[TinTime]

No, because i've recompressed the flac from eac3to, with SoundOut avisynth plugin and the result is 500meg smaller, so i think the default is to use "medium" compression.

I just ran two test encodes of a small 6 channel wav file. The first encoded to flac using eac3to and the second using the flac CLI encoder with the --best switch.

The resulting flac files were (very nearly) the same size. Perhaps there's a problem with SoundOut? It's not truncating the output file is it? I know that the flac CLI encoder can't cope with large files.

[rik1138]

@Madshi
I've discovered (well, I've been told about, technically) an issue with eac3to recognizing original, unused DTS-HD files. By unused, I mean the files created by the DTS-HD Master Audio Suite (the software that the studios use to actually create the DTS-HD MA audio). The DTS software creates files with an extra 140 bytes of header information at the beginning of the files. This information is stripped off when the DTS is used on a final disc, so if you rip DTS off a Blu-Ray disc, this data is not there. Thus, eac3to doesn't recognize a DTS file with this information... The header isn't vague, the first 8 characters are DTSHDHDR. It would be great if the software could be updated to recognize DTS files with this header information still attached.

I encoded a 7.1 test stream for someone else on the forum, and a member discovered the problem when trying to use eac3to on the file. The files are here:

http://www.sendspace.com/file/mz71xx

There's a DTS-HD High Resolution stream (I don't see this used very often, basically DTS that can go up to a 3mb bitrate), and a DTS-HD Master Audio stream included.

Follow-up to this:

There's more to this than I thought. Apparently the header info is different sized depending on the type of DTS file you've encoded... The header includes the 8 bytes following the string STRMDATA, that seems to be consistent with all the files I've encoded so far.

Rik

[Inspector.Gadget]

Anybody know whether the new Arcsoft 3 will continue to provide the necessary decoders for eac3to DTS decoding or whether the architecture has changed?

[laserfan]

Anybody know whether the new Arcsoft 3 will continue to provide the necessary decoders for eac3to DTS decoding or whether the architecture has changed?Yeah maybe have they made it impossible to use their filters outside of TMT...?

[kypec]

I just ran two test encodes of a small 6 channel wav file. The first encoded to flac using eac3to and the second using the flac CLI encoder with the --best switch.

The resulting flac files were (very nearly) the same size. Perhaps there's a problem with SoundOut? It's not truncating the output file is it? I know that the flac CLI encoder can't cope with large files.
I tried to encode via FLAC CLI (revision 1.2.1b alias the latest official release) with --no-padding together with the --best switch. Despite the fact that madshi has included only version 1.2.0 of libFLAC.dll with his eac3to package the only differences occur in the metadata tags at the beginning of files. Raw encoded audio data are bit-to-bit identical.

How to make FLAC CLI encodes bit-to-bit identical with eac3to
First download the latest FLAC library package (http://downloads.sourceforge.net/flac/flac-1.2.1-devel-win.zip) and replace libFLAC.dll in the eac3to folder with the downloaded one.
Run your CLI encoder with the following options:flac.exe --best --no-seektable --no-padding --tag=WAVEFORMATEXTENSIBLE_CHANNEL_MASK=0X0 --tag=VALID_BITS=16 --tag=HDCD=0 inputfile.wav
You will have to change the tag values accordingly for multichannel and/or different bitdepths, my example demonstrates ordinary 16-bit stereo WAV.

[StephenB]

When extracting from a VOB, EAC3TO 3.14 is frequently outputing duplicate subtitle tracks - for instance

Movie - 6 - Subtitle (DVD), 1857 captions.sup and
Movie - 6 - Subtitle (DVD).sup

The two files have identical contents.

The command line is
EAC3TO vts_01_1.vob+vts_01_2.vob+vts_01_3.vob+vts_01_4.vob+vts_01_5.vob+vts_01_6.vob Movie.mkv -seekToIFrames -lowPriority

[3ngel]

My test was done on a 24bit 5.1 2 hours audio. Pheraps the difference takes place on the long run and high def files?

[Audionut]

Hi madshi.

Could you please take a look at this file.

http://d01.megashares.com/?d01=72e4388

It's recorded from DTV. It has 5.1 channels during main broadcast and switches to 2.0 channel during the ad breaks.
Converting to flac results in the audio during the ad breaks being high pitched.

[TinTime]

My test was done on a 24bit 5.1 2 hours audio. Pheraps the difference takes place on the long run and high def files?

I've just tried it using a 2hr+ 5.1 24 bit source and don't find any significant difference in size between flacs produced by eac3to and soundout.

Are you sure the audio is not being corrupted somewhere in your avisynth / soundout chain? If you convert your target flacs back to wav are they identical?

[tebasuna51]

...
It's recorded from DTV. It has 5.1 channels during main broadcast and switches to 2.0 channel during the ad breaks.
Converting to flac results in the audio during the ad breaks being high pitched.

Use DelayCut to split the ac3 source.

[narshorn]

Sorry to bother, some eac3to -decodeHdcd questions ;

Any reason to why eac3to with -decodeHdcd switch gets 6 dB gain lower than standard decoding ?
If so, is it OK to add +6dB as a switch while doing conversions ?
Madshi, Is there a possibility to add -deEmphasis switch option to eac3to ? :)

Best regards,

narshorn

narshorn

[madshi]

Thanks for adding support for 24.975fps. However, when I test on a blank 100 sec audio clip, EAC3TO seems to use 25fps in place of 24.975fps...
Will be fixed in the next build.

The exact FPS for -24.975 option is 25/1.001 = 24.975025
Argh, why did nobody tell me before!? It makes things much simpler...

Any reason why one of my 1080i60 streams makes eac3to output 'xxxxxx fields' while the other one 'xxxxxx frames'?
Yes, because some 1080i60 streams are encoded using fields while other 1080i60 streams are encoded using frames.

Also, would be nice if when demuxing audio, if eac3to can't correct a, say, -9ms delay, it would add -9ms to the file name.
This might be troublesome. All the GUIs out there don't expect the file names to have such a delay added to them. So if I did that, it would probably break all GUIs. Also I'm not sure if it's really worth it. I mean e.g. 9ms are really small...

I've checked the problematic subtitle stream again with 2.13, but the subtitle doesn't display, as with DTS fields set to zero. Only with the DTS entries from tsMuxeR it is displayed.
Argh, ok, will have to do a full DTS fix then. That might take a while...

Ive just tried to demux Body Of Lies bluray and get these errors.(log attatched)
The bluaray is not a seemless branching movie and is a single m2ts. Also running latest anydvd.

edit I think maybe the disc has mastering errors do to the fact that at the times posted by eac3to as errors, there is glitches in video and audio from the original disk. This is a new movie and is not scratched in any way. I will just return for another
If you run into such a problem, always try using the AnyDVD HD built in ripping tool to copy the disc to harddisk first and then run eac3to on the harddisk folder. If that also fails, the disc might be faulty.

What is this ???? v02 The video framerate is correct, but rather unusual. ???????

C:\BT>eac3to "Transporter 2 2005 1080p BluRay MPEG-4 AVC DTS-HD 5.1-Mike" 1)
M2TS, 1 video track, 2 audio tracks, 1 subtitle track, 15.052p
1: Chapters, 12 chapters
2: h264/AVC, 1080p24 (16:9)
v02 The video framerate is correct, but rather unusual.
Most Blu-Rays are 24/1.001. Yours is 24/1.000. That is unusual. eac3to just posts this warning to alert you that the framerate used by this Blu-Ray is strange. You can ignore the warning. But be aware that you might run into stuttering issues when playing back this Blu-Ray.

madshi, how hard would it be to implement reading from the other formats that libavcodec handles? Would be nice to be able to decode Vorbis, or MP3, AAC, and more with eac3to. Wishful thinking on my part?
libav MP3 decoding is already supported. Vorbis and AAC are not. The AAC decoder is not feature complete yet, so I'm waiting until it is. Vorbis might be added sooner or later...

depending on the sountrack being 1.5mbit DTS core or FLAC(from DTSHD in eac3to), the sound levels are much different ?!
I can hardly hear dialogs in DTSHD FLAC and need to increase the center channel to like 140%

I've got all DRC disabled in AC3Filter and it's the same story w/ ffdshow/sonic 4.2 for DTS core, so what gives ?!

it happens in all the DTSHD>FLAC tracks I've built, and forcing dialog normalization in ea3to doesn't do anything(same CRC whether it's enabled or not)

OTOH transcoding LPCM in FLAC gives proper sound levels, identical to DTS core :o

if anyone's got a clue, I'd love to hear it. thanks!
Maybe your DTS decoder is not working properly? Don't know...

5.0 channel DTS can't seem to be converted to AC3.
AC3 encoding is only supports with 5.1 and 2.0 right now. Other formats will be supported in a future version.

If I feed eac3to a duff audio file it reports "The format of the source file could not be detected." but the return code is 0. Could it be changed to non-zero please?
Will be implemented in the next build.

Having an issue with a specific movie - The Game Plan. The video appears to rip just fine but the audio track is a mess :(
IIRC this movie had several playlists. The main playlist has messed up audio. Try some of the other playlists. At least one of them works fine. This is a mastering problem, not an eac3to bug.

[mrr19121970]

This might be troublesome. All the GUIs out there don't expect the file names to have such a delay added to them. So if I did that, it would probably break all GUIs. Also I'm not sure if it's really worth it. I mean e.g. 9ms are really small...



Clown_BD (http://clownbd.techxt.com/) handles this....

see here (http://forum.slysoft.com/showpost.php?p=181531&postcount=499)

[a06] Creating file "C:\Temp\Batman Begins\Audio_6_English.THD+AC3"...

A_AC3, "C:\Temp\Batman Begins\Audio_6_English DELAY 200ms.THD+AC3", timeshift=200ms, lang=eng

[madshi]

Following on from this I've been trying to work out why eac3to doesn't like some (but not all) of my input files. I used eac3to to create a wav file from a pcm input:
eac3to v3.14
command line: "D:\Vtemp\programs\eac3to\eac3to.exe" "test.pcm" "E:\Vtemp\test1.wav" -6 -24 -little -96000 -override -log=1.txt
------------------------------------------------------------------------------
RAW/PCM, 5.1 channels, 1:38:26, 24 bits, 13824kbps, 96khz
Reading RAW/PCM...
Writing WAV...
Creating file "E:\Vtemp\test1.wav"...
The original audio track has a constant bit depth of 24 bits.
Caution: The WAV file is bigger than 4GB. <WARNING>
Some WAV readers might not be able to handle this file correctly. <WARNING>
eac3to processing took 9 minutes, 41 seconds.
Done.

If I then run eac3to on the output wav file I get:
eac3to v3.14
command line: "D:\Vtemp\programs\eac3to\eac3to.exe" "E:\Vtemp\test1.wav" -log=t1.txt
------------------------------------------------------------------------------
The format of the source file could not be detected. <ERROR>

I've been trying to find out why this file does not work and other apparently similar ones do and the difference seems to be the amount of silence at the beginning of the file. This file contains 11866608 bytes = 6.86725 seconds of silence at the beginning. I tried editing the first six samples to non-zero values and it's then detected by eac3to.
This was caused by the HDCD decoder not being able to lock unto the stream due to the many zeroes. Will be fixed in the next build.

Hey, I got a 20gig m2ts file containing several episodes in it. I want to demux the video and audio separately to mkv and flac.

Can I split m2ts files by chapters? If so, how?
Currently not in any automatic way. You could try splitting the m2ts file manually by using a hexeditor or by using some other external tools.

Just wanted to post the following issue that I found using eac3to (v3.14). The sample file that I have used can be found in the link here below.

http://rapidshare.de/files/46162759/extract.h264.html

- The sample is a part of a h264 file extracted directly from blu-ray using eac3to
- When performing the following set of commands on this sample:

eac3to extract.h264 extract.mkv
eac3to extract.mkv 1: extractfrommkv.h264


then you will notice that the result file "extractfrommkv.h264" is different from the original "extract.h264".

Looking a bit more in detail there is a sequence of 162 bytes added at specific places in the file, starting with the following 34 bytes:

00 01 27 64 00 29 AC 7B 01 E0 08 9F 97 FF 00 01 00 01 10 00 00 3E 90 00 0B B8 08 40 00 00 00 01 28 EA

The sequence is the same throughout the file and removing it will give back the original "extract.h264".

The issue has been introduced as of v3.07 (v3.06 works fine), and I think it happens only with AVC/H.264 encoded files with 2 Reframes. Did the test on 4 different films (directly from blu-ray), all with 2 reframes, and they all had the issue, i.e. a lot of artifacts during very fast-moving scenes.
Did some other tests with VC1, MPEG2 and higher-frame H.264 encoded files and did not find a difference between the files before and after the 2 eac3to steps provided here above.
Thanks for the report. Will be fixed in the next build. FWIW, the MKV file is just fine. It's just the MKV h264 extraction which produced incorrect results.

I have problem when i try to demux TsMuxer created Blu-Ray.

M2TS, 1 video track, 1 audio track, 1 subtitle track, 1:24:34, 24p /1.001
1: Chapters, 16 chapters
2: h264/AVC, 1080p23.975 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48khz
4: Subtitle (PGS), English
v02 The video bitstream is encoded in a non-standard framerate.
v02 The video bitstream framerate field doesn't seem to match the timestamps.

but that is not true, i encode to H264 with x264 and input avs have parametar fps=23.976, all programs and players reconize as 23.976, aslo i tryed with checking changefps in TsMuxer and selecting 24000/1001, but still eac3to report message.
Using "fps=23.976" is wrong. You should encode with 24/1.001 instead. tsMuxeR's conversion to 24/1.001 was buggy, that's why eac3to didn't like that, either. This problem is fixed in the latest tsMuxeR version, though.

Second question is: why eac3to when extracting chapters not include final time anymore?
I've done that by user request! What would you need the final time for?

Any reason to why eac3to with -decodeHdcd switch gets 6 dB gain lower than standard decoding ? If so, is it OK to add +6dB as a switch while doing conversions ?
eac3to is just writing to disk what the HDCD decoder outputs. Of course you can change volume, but please be aware that changing volume involves floating point math, so basically the result will not be 20bit integer, anymore. Instead it will be 64bit floating point. You can then dither that down to whatever bitdepth you want. eac3to by default dithers down to 24bit.

Here's an easy request:

When I convert a 5.1 stream to WAV files, it names the Surround channels .SL.wav and .SR.wav. The encoding software I use for Dolby and DTS expect .LS.wav and .RS.wav (the software will auto-find all the files in their correct channel locations if they are named like this, and it seems to be the industry standard...) Any chance of swapping those letters in a future update? :)
And next week someone else uses a different encoding software which expects "SL" and "SR" or something completely different and I have to change names again? I don't think there's a fixed standard on how to name these files. If there was such a standard, I'd be happy to make use of it. But as long as different encoders expect different names, I'll not adjust to one specific encoder.

I'm trying to use EAC3TO to process an 8 bit PCM file. The goal is to resample it to 48 kHz and transcode it.
8bit handling is buggy right now. Will have to work on that. But it's low on my priority list.

arcsoft totalmedia theatre 3 is out, does anyone know if there could be benefits for eac3to from it?
You could try:

(1) Does the DTS decoder still work for eac3to?
(2) Does it handle those 7.1 DTS-MA tracks with funny speaker placement correctly? That was the only thing that could be improved. Apart from this the old ArcSoft DTS decoder was already pretty perfect.

How do I disable the 2nd pass? I have a file with discontinuities. Eac3to doesn't care and finishes processing. The result has minor picture and audio dropouts I can easily live with. However, eac3to erroneously sees an audio delay and starts a second pass.
Erroneously? How do you know that?

Afterwards the audio is no longer in sync from the first discon on. I am pretty sure this would not happen without the second pass.
How do you know that? You could try pressing "Ctrl+C" when the 2nd pass starts. Depending on how the 2nd pass is executed, this might get you what you want. Eventually eac3to has already deleted first pass results, though. Depends on the circumstances...

What is changed in V3.06 that can explain this behavior? So in short, eac3to can perfectly read the contents of a BD, but it can not read the m2ts files (using CrossOver on Mac OS-X).
Do you have this problem when reading directly from the physical BD? Or when reading a folder structure? Newer eac3to versions use memory mapped file read for physical drive access because that was *a lot* faster. For folder reading the same logic should be used as before, though.

Is there any way to demux a PIP video and its associated soundtrack and keep them in-sync?

Whenever I have attempted to do this, the audio is always wildly out of sync. I'm assuming that either the video or audio is continuous, whilst the corresponding audio or video isn't: often there are long stretches of blank video between the PIP segments.
I've never really tried. But as you say, I think PIP video sometimes has big (very VERY big) areas where there's no video content at all, while secondary audio usually is continuous. That of course makes things very difficult. However, I imagine eac3to should detect the gaps in the video stream and try to repair them when muxing to MKV?

[madshi]

Got an issue with Terminator, the SaraConnor Chronicles. eac3to failed to detect an entire episode. There are three on there and eac3to only found 2. I managed to demux the m2ts file for the episode, but it lacked a chapter file.
As far as I can see, the 3 episodes are 4.m2ts, 12.m2ts and 13.m2ts. Strange enough, there is one playlist for 4.m2ts and one for 13.m2ts, but none for 12.m2ts. So that's the simple reason why eac3to "failed to detect" one episode. eac3to does not detect anything, it just lists the available playlists. If there's no playlist for an episode, naturally eac3to will not list it.

Does this mean he sold the rights for eac3to to SlySoft for their new HD transcoding app? ;)
I don't think they'd like Delphi code... ;)

I've discovered (well, I've been told about, technically) an issue with eac3to recognizing original, unused DTS-HD files. By unused, I mean the files created by the DTS-HD Master Audio Suite (the software that the studios use to actually create the DTS-HD MA audio). The DTS software creates files with an extra 140 bytes of header information at the beginning of the files. This information is stripped off when the DTS is used on a final disc, so if you rip DTS off a Blu-Ray disc, this data is not there. Thus, eac3to doesn't recognize a DTS file with this information... The header isn't vague, the first 8 characters are DTSHDHDR. It would be great if the software could be updated to recognize DTS files with this header information still attached.

I encoded a 7.1 test stream for someone else on the forum, and a member discovered the problem when trying to use eac3to on the file. The files are here:

There's a DTS-HD High Resolution stream (I don't see this used very often, basically DTS that can go up to a 3mb bitrate), and a DTS-HD Master Audio stream included.

[...]

Follow-up to this:

There's more to this than I thought. Apparently the header info is different sized depending on the type of DTS file you've encoded... The header includes the 8 bytes following the string STRMDATA, that seems to be consistent with all the files I've encoded so far.
Interesting. It seems there's a whole file structure. It's very easy to understand, though, so it should be easy adding support for that. Well, "easy" means "not difficult". It does cost some time, though, so it may have to wait a while.

How i tell eac3to to use the maximum compression for a .flac destination encoding?
It already uses max compression by default.

No, because i've recompressed the flac from eac3to, with SoundOut avisynth plugin and the result is 500meg smaller
Then your SoundOut setup is probably broken and does some damage to the audio data. If you want to be sure, you can upload the first 50MB of the eac3to created FLAC file and the first 50MB of the SoundOut created FLAC file. I'll have a look at them then.

Quick question about eac3to and audio delay in Matroska files: The other day I ripped a DVD and then muxed it directly to Matroska without re-encoding, planning to encode it later. I checked the audio delay with DGIndex before muxing and applied the proper delay at mux. eac3to, when run without any options against the Matroska file, reports the audio delay as -301 ms. This sounds like too much of a delay, but off hand I can't remember what the audio delay reported by DGIndex was. Does anyone know how accurate eac3to's parsing of matroska delay values is?

Eduit: Mkvinfo -v shows that the audio starts 16ms after the video, which jogs my memory and I think was the delay I applied at muxing. What's going on here?
Is it possible that the video stream doesn't start with a proper sequence header? eac3to always calculates from the first *displayable* video frame.

I would also like a switch that suppressed the second pass.
If you can find a good reason for that?

Any update on being able to extract forced subtitles only?
It's still on my to do list... :)

When extracting from a VOB, EAC3TO 3.14 is frequently outputing duplicate subtitle tracks - for instance

Movie - 6 - Subtitle (DVD), 1857 captions.sup and
Movie - 6 - Subtitle (DVD).sup

The two files have identical contents.
Can you please post a full eac3to log file (in "[ code ] [ / code ]" brackets, please)?

Is there a possibility to add -deEmphasis switch option to eac3to ? :)
What would that switch do exactly?

Clown_BD handles this....
Nice!

[madshi]

eac3to v3.15 released

http://madshi.net/eac3to.zip

* "24.975" is now interpreted as "25.000/1.001"
* Blu-Ray "sup" are demuxed with DTS set to 0 again, proper fix will come later
* fixed: error code not set for "source file format could not be detected"
* fixed: audio resampling from/to 24.975 didn't work properly
* fixed: WAV files beginning with lots of zeroes were sometimes not accepted

[evdberg]

Do you have this problem when reading directly from the physical BD? Or when reading a folder structure? Newer eac3to versions use memory mapped file read for physical drive access because that was *a lot* faster. For folder reading the same logic should be used as before, though.
I am always reading from a folder on HDD. However, CrossOver maps folders on my Mac HDD to a device in Windows. For example, if I read from '/Volumes/HardDisk1/BD/Movie' it is mapped to 'D:\BD\Movie'. I am pretty sure the file mapping is causing the problem. Maybe you can fallback to normal IO if filemapping fails?

[Thunderbolt8]

thanks for the update :D

[princenigma]

* Blu-Ray "sup" are demuxed with DTS set to 0 again, proper fix will come later

Can someone explain this please ?
Does it means we should wait the proper fix of eac3to to demux sup file ?

[madshi]

I am always reading from a folder on HDD. However, CrossOver maps folders on my Mac HDD to a device in Windows. For example, if I read from '/Volumes/HardDisk1/BD/Movie' it is mapped to 'D:\BD\Movie'. I am pretty sure the file mapping is causing the problem. Maybe you can fallback to normal IO if filemapping fails?
If I knew how to detect that? How does your eac3to log file look like?

* Blu-Ray "sup" are demuxed with DTS set to 0 again, proper fix will come later

Can someone explain this please ?
Does it means we should wait the proper fix of eac3to to demux sup file ?
If you want to convert those sup files to SRT, there's no problem. If you want to remux them into a m2ts structure, then with some movies the current sup files produced by eac3to won't display correctly on some standalone players.

[princenigma]

If you want to remux them into a m2ts structure, then with some movies the current sup files produced by eac3to won't display correctly on some standalone players.

Thanks .

Are PowerDVD and TMT display correctly?

[DoomBot]

So if i have already demuxed the subtitles and when the problem gets fixed can i run the demuxed subs back through eac3to and be fixed then or do i have to go back to the blu-ray it self and do it again?

[laserfan]

If you want to remux them into a m2ts structure, then with some movies the current sup files produced by eac3to won't display correctly on some standalone players.I wondered why a couple of my re-muxed .sups didn't display.

Many thanks madshi for the update--your attention to detail is VERY MUCH APPRECIATED! :)

[evdberg]

If I knew how to detect that? How does your eac3to log file look like?
I just get an error message like this:
Error reading file "D:\BD\Movie\BDMV\STREAM\00001.m2ts".

[narshorn]

[/QUOTE]


What would that switch do exactly?


[/QUOTE]

Some rare CDs have pre-emphasis. Extracting them in secure mode with EAC leaves the file emphasised. eac3to could have a switch to
get proper gain figure at high frequencies when processing the wavs.

WaveEmph110 does the job, but a switch in eac3to could be a nice feature, although rare.

Regards,

narshorn

[napalm-187]

hi

i have buyed arcsoft TMT3 eac3 dosnet work with this decoder
is there any chance to runs this audiodecoder?

zhx

[Snowknight26]

Glad to see you back in action madshi.

Small feature request while you're still around. Would it be possible to add a -down7 switch that makes 6.1 channel output from 8 (7.1) channel soures?

[StephenB]

Originally Posted by StephenB
When extracting from a VOB, EAC3TO 3.14 is frequently outputing duplicate subtitle tracks - for instance

Movie - 6 - Subtitle (DVD), 1857 captions.sup and
Movie - 6 - Subtitle (DVD).sup

The two files have identical contents.
Can you please post a full eac3to log file (in "[ code ] [ / code ]" brackets, please)?
This is from a script that automatically called EAC3TO again if it detected a gaps file

eac3to v3.14
command line: "c:\VID2EVA\tools\eac3to\eac3to.exe" "c:\VID2EVA\Videos\Vid2eva_0.vob"+"c:\VID2EVA\Videos\Vid2eva_1.vob"+"c:\VID2EVA\Videos\Vid2eva_2.vob"+"c:\VID2EVA\Videos\Vid2eva_3.vob"+"c:\VID2EVA\Videos\Vid2eva_4.vob"+"c:\VID2EVA\Videos\Vid2eva_5.vob"+"c:\VID2EVA\Videos\Vid2eva_6.vob"+"c:\VID2EVA\Videos\Vid2eva_7.vob"+"c:\VID2EVA\Videos\Vid2eva_8.vob"+"c:\VID2EVA\Videos\Vid2eva_9.vob"+"c:\VID2EVA\Videos\Vid2eva_10.vob"+"c:\VID2EVA\Videos\Vid2eva_11.vob"+"c:\VID2EVA\Videos\Vid2eva_12.vob"+"c:\VID2EVA\Videos\Vid2eva_13.vob"+"c:\VID2EVA\Videos\Vid2eva_14.vob"+"c:\VID2EVA\Videos\Vid2eva_15.vob"+"c:\VID2EVA\Videos\Vid2eva_16.vob"+"c:\VID2EVA\Videos\Vid2eva_17.vob" "c:\VID2EVA\Videos\Vid2eva.mkv" -seekToIFrames -lowPriority
------------------------------------------------------------------------------
VOB, 1 video track, 3 audio tracks, 3 subtitle tracks, 2:00:55
1: Joined VOB file
2: MPEG2, 480p24 /1.001 (4:3) with pulldown flags
3: AC3 Surround, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, 24ms
4: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB, 24ms
5: AC3 Surround, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, 24ms
6: Subtitle (DVD)
7: Subtitle (SRT)
8: Subtitle (DVD)
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a04] Extracting audio track number 4...
[a05] Extracting audio track number 5...
[s07] Extracting subtitle track number 7...
[a03] Removing AC3 dialog normalization...
[a04] Removing AC3 dialog normalization...
[a05] Removing AC3 dialog normalization...
[v02] Muxing video to Matroska...
[a03] Applying (E-)AC3 delay...
[a04] Applying (E-)AC3 delay...
[a05] Applying (E-)AC3 delay...
[a03] A remaining delay of -8ms could not be fixed.
[a04] A remaining delay of -8ms could not be fixed.
[a05] A remaining delay of -8ms could not be fixed.
[a04] Creating file "c:\VID2EVA\Videos\Vid2eva - 4 - AC3, 5.1 channels, 448kbps, 48khz.ac3"...
[a03] Creating file "c:\VID2EVA\Videos\Vid2eva - 3 - AC3 Surround, 2.0 channels, 192kbps, 48khz.ac3"...
[a05] Creating file "c:\VID2EVA\Videos\Vid2eva - 5 - AC3 Surround, 2.0 channels, 192kbps, 48khz.ac3"...
[s08] Extracting subtitle track number 8...
[s06] Extracting subtitle track number 6...
[s06] Creating file "c:\VID2EVA\Videos\Vid2eva - 6 - Subtitle (DVD).sup"...
[s08] Creating file "c:\VID2EVA\Videos\Vid2eva - 8 - Subtitle (DVD).sup"...
[v02] Video has a gap of 7 frames at playtime 1:57:07. <WARNING>
[v02] The MKV file was created without making use of the gap/overlap information. <WARNING>
[v02] Please check whether audio is in sync. If it is in sync everything is fine. <WARNING>
[v02] Otherwise ask eac3to to repeat the muxing. It will then automatically make <WARNING>
[v02] use of the detailed gap/overlap information. <WARNING>
[s07] Creating file "c:\VID2EVA\Videos\Vid2eva - 7 - Subtitle (SRT).srt"...
Added fps value to MKV header.
Video track 2 contains 174101 frames.
Subtitle track 6 contains 1857 captions.
Subtitle track 8 contains 1421 captions.
eac3to processing took 10 minutes, 11 seconds.
Done.

eac3to v3.14
command line: "c:\VID2EVA\tools\eac3to\eac3to.exe" "c:\VID2EVA\Videos\Vid2eva_0.vob"+"c:\VID2EVA\Videos\Vid2eva_1.vob"+"c:\VID2EVA\Videos\Vid2eva_2.vob"+"c:\VID2EVA\Videos\Vid2eva_3.vob"+"c:\VID2EVA\Videos\Vid2eva_4.vob"+"c:\VID2EVA\Videos\Vid2eva_5.vob"+"c:\VID2EVA\Videos\Vid2eva_6.vob"+"c:\VID2EVA\Videos\Vid2eva_7.vob"+"c:\VID2EVA\Videos\Vid2eva_8.vob"+"c:\VID2EVA\Videos\Vid2eva_9.vob"+"c:\VID2EVA\Videos\Vid2eva_10.vob"+"c:\VID2EVA\Videos\Vid2eva_11.vob"+"c:\VID2EVA\Videos\Vid2eva_12.vob"+"c:\VID2EVA\Videos\Vid2eva_13.vob"+"c:\VID2EVA\Videos\Vid2eva_14.vob"+"c:\VID2EVA\Videos\Vid2eva_15.vob"+"c:\VID2EVA\Videos\Vid2eva_16.vob"+"c:\VID2EVA\Videos\Vid2eva_17.vob" "c:\VID2EVA\Videos\Vid2eva.mkv" -seekToIFrames -lowPriority
------------------------------------------------------------------------------
VOB, 1 video track, 3 audio tracks, 3 subtitle tracks, 2:00:55
1: Joined VOB file
2: MPEG2, 480p24 /1.001 (4:3) with pulldown flags
3: AC3 Surround, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, 24ms
4: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB, 24ms
5: AC3 Surround, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, 24ms
6: Subtitle (DVD)
7: Subtitle (SRT)
8: Subtitle (DVD)
[v02] Video gap description file detected, will be used for muxing...
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a04] Extracting audio track number 4...
[a05] Extracting audio track number 5...
[s07] Extracting subtitle track number 7...
[a03] Removing AC3 dialog normalization...
[v02] Muxing video to Matroska...
[a04] Removing AC3 dialog normalization...
[a05] Removing AC3 dialog normalization...
[a03] Applying (E-)AC3 delay...
[a04] Applying (E-)AC3 delay...
[a05] Applying (E-)AC3 delay...
[a03] A remaining delay of -8ms could not be fixed.
[a04] A remaining delay of -8ms could not be fixed.
[a05] A remaining delay of -8ms could not be fixed.
[a04] Creating file "c:\VID2EVA\Videos\Vid2eva - 4 - AC3, 5.1 channels, 448kbps, 48khz.ac3"...
[a03] Creating file "c:\VID2EVA\Videos\Vid2eva - 3 - AC3 Surround, 2.0 channels, 192kbps, 48khz.ac3"...
[a05] Creating file "c:\VID2EVA\Videos\Vid2eva - 5 - AC3 Surround, 2.0 channels, 192kbps, 48khz.ac3"...
[s08] Extracting subtitle track number 8...
[s06] Extracting subtitle track number 6...
[s06] Creating file "c:\VID2EVA\Videos\Vid2eva - 6 - Subtitle (DVD).sup"...
[s08] Creating file "c:\VID2EVA\Videos\Vid2eva - 8 - Subtitle (DVD).sup"...
[s07] Creating file "c:\VID2EVA\Videos\Vid2eva - 7 - Subtitle (SRT).srt"...
Added fps value to MKV header.
Video track 2 contains 174101 frames.
Subtitle track 6 contains 1857 captions.
Subtitle track 8 contains 1421 captions.
eac3to processing took 11 minutes, 24 seconds.
Done.

After this completed, the directory contained:

03/26/2009 04:42 PM 174,273,024 Vid2eva - 3 - AC3 Surround, 2.0 channels, 192kbps, 48khz.ac3
03/26/2009 04:42 PM 406,637,056 Vid2eva - 4 - AC3, 5.1 channels, 448kbps, 48khz.ac3
03/26/2009 04:42 PM 174,273,024 Vid2eva - 5 - AC3 Surround, 2.0 channels, 192kbps, 48khz.ac3
03/26/2009 04:30 PM 3,126,648 Vid2eva - 6 - Subtitle (DVD), 1857 captions.sup
03/26/2009 04:42 PM 3,126,648 Vid2eva - 6 - Subtitle (DVD).sup
03/26/2009 04:42 PM 204,990 Vid2eva - 7 - Subtitle (SRT).srt
03/26/2009 04:30 PM 2,338,892 Vid2eva - 8 - Subtitle (DVD), 1421 captions.sup
03/26/2009 04:42 PM 2,338,892 Vid2eva - 8 - Subtitle (DVD).sup
03/26/2009 04:42 PM 2,768 Vid2eva - Log.txt
03/26/2009 04:42 PM 5,461,622,129 Vid2eva.mkv
03/26/2009 04:30 PM 16 Vid2eva.mkv.gaps

[SamNZDat]

Welcome home madshi!

I'm trying to serve a live 16-bit 44.1 FLAC stream via Tribler, but so far, so stumped. Tribler uses Python to convert live input (FLAC in my case) to .mpegts, but thus far, none of my .mpegts output is readable. There is one thing I should mention about the FLAC I use as input: VLC can play it as a live stream (from another instance of VLC) but foobar2000 can only play it once it has been saved as a finished file.

Any tips on how to proceed? madshi, especially would appreciate your weighing in here.

[deathlord]

Erroneously? How do you know that?


How do you know that? You could try pressing "Ctrl+C" when the 2nd pass starts. Depending on how the 2nd pass is executed, this might get you what you want. Eventually eac3to has already deleted first pass results, though. Depends on the circumstances...

Eac3to says there is an audio delay of 600ms after the discontinuity. After the second pass, eac3to should have corrected that, so audio should be in sync. However, it is not, it is out of sync by something that looks like approx. 600 ms.
So I conlude that if eac3to had not "corrected" the "delay", it would still be in sync.

I was going to try Ctrl+C, but I thought I would ask first because I don't like it. Maybe I'm too late and the files are already changed and might not work any longer.

I assume it would be easy for you to add an undocumented switch "-no2pass" or similar.
If it means a lot of trouble for you, please forget about it.

[laserfan]

I'm confused about the handling of Dolby Digital, Dolby Digital Plus (aka E-AC3), and TrueHD. If I encounter a track like this:

3: TrueHD/AC3, English, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -30dB)

and I don't want the TrueHD track, do I convert to "ordinary AC3" by using the -core option, or -640 option, or neither e.g. "eac3to 3: audio.ac3" ? :confused:

[TinTime]

Use the last option you mentioned, i.e...

"eac3to 3: audio.ac3"

[rik1138]

And next week someone else uses a different encoding software which expects "SL" and "SR" or something completely different and I have to change names again? I don't think there's a fixed standard on how to name these files. If there was such a standard, I'd be happy to make use of it. But as long as different encoders expect different names, I'll not adjust to one specific encoder.

It's not just one specific encoder, but the encoders written by DTS and Dolby (both of them). There is a fixed standard on how to name the files in the DVD/Blu-Ray industry:
For 5.1:
L, R, C, LFE, Ls, Rs
For 6.1:
L, R, C, LFE, Ls, Rs, Cs
For 7.1:
L, R, C, LFE, Lss, Rss, Lsr, Rsr

All capitals are acceptable. These are the extensions that are printed on the labels of the tapes delivered from the studios to be encoded for Blu-Ray. These extensions are also used by the two professional audio encoding suites: DTS-HD Master Audio Suite Encoder and the Dolby Media Encoder. You can see both of these encoders expecting these extensions here:
Dolby:
http://www.dolby.com/professional/pro_audio_engineering/DMP_02.html
(Click on the image to make it bigger)

DTS:
http://www.dts.com/Pro-Audio_Software/DTS-HD_Master_Audio_Suite/DTS-HD_Encoder.aspx
(Again, click on the image...)

These two pieces of software will auto-fill the WAV files when you select one if the file names are correct.

That's the only reason I mentioned it... People keep reversing the letters, or in some cases, use completely different letters, and it can get confusing sometimes...

Again, it's not really a big deal (there's probably more imortant things you have to worry about)... It's not like it's hard to rename the files. I was just making an observation... :cool:

[kypec]

Hello madshi,
any chance you could put the most recent version of libFLAC.dll (http://forum.doom9.org/showpost.php?p=1266310&postcount=8628) in your package at next release? Just for the sake of having up-to-date libraries included with eac3to.

[tebasuna51]

It's not just one specific encoder, but the encoders written by DTS and Dolby (both of them). There is a fixed standard on how to name the files in the DVD/Blu-Ray industry:
For 5.1:
L, R, C, LFE, Ls, Rs
For 6.1:
L, R, C, LFE, Ls, Rs, Cs
For 7.1:
L, R, C, LFE, Lss, Rss, Lsr, Rsr
...

You are free to name the channels inside DD/DTS, but still the channels are in wav format you must use the wav names and order:

Front Left - FL
Front Right - FR
Front Center - FC
Low Frequency – LF
Back Left - BL
Back Right - BR
Front Left of Center - FLC
Front Right of Center - FRC
Back Center - BC
Side Left - SL
Side Right - SR
...

[kartman_canada]

I've been pulling some of my older DVDs to disk for storage on my server. Being a user of EAC3TO for HD-DVD and BD, I was trying to stick with a similar process. Does EAC3TO do "joined VOB" correctly?

command line: eac3to.exe VTS_01_1.VOB+VTS_01_2.VOB+VTS_01_3.VOB+VTS_01_4.VOB+VTS_01_5.VOB+VTS_01_6.VOB 2: movie.m2v 3: movie.ac3

The info returned identifies the source as a "joined VOB" but the the total length is way off and I often get errors such as large gaps in the audio or video overlaps during the demuxing of the streams.

Am I doing something wrong or is this just not the correct way to use the tool?

[SamNZDat]

Welcome home madshi!

I'm trying to serve a live 16-bit 44.1 FLAC stream via Tribler, but so far, so stumped. Tribler uses Python to convert live input (FLAC in my case) to .mpegts, but thus far, none of my .mpegts output is readable. There is one thing I should mention about the FLAC I use as input: VLC can play it as a live stream (from another instance of VLC) but foobar2000 can only play it once it has been saved as a finished file.

Any tips on how to proceed? madshi, especially would appreciate your weighing in here.

Hmm (hear effete Brit accent in your mind's ear) ... I have serious doubts as to the efficacy of message boards for achieving results ... supposing one were to pay for them?

[Jeff Flowerday]

I've been pulling some of my older DVDs to disk for storage on my server. Being a user of EAC3TO for HD-DVD and BD, I was trying to stick with a similar process. Does EAC3TO do "joined VOB" correctly?

command line: eac3to.exe VTS_01_1.VOB+VTS_01_2.VOB+VTS_01_3.VOB+VTS_01_4.VOB+VTS_01_5.VOB+VTS_01_6.VOB 2: movie.m2v 3: movie.ac3

The info returned identifies the source as a "joined VOB" but the the total length is way off and I often get errors such as large gaps in the audio or video overlaps during the demuxing of the streams.

Am I doing something wrong or is this just not the correct way to use the tool?

No it doesn't. Currently eac3to isn't capable of reading IFO and handling SD DVDs. When or if Madshi will add support can only be answered by him.

Use DVD Decrypter.

[The_Keymaker]

I just ripped my copy of Iron Man Blu-Ray using eac3to and have two questions.

1. It appears that MKmerge does not support muxing of TrueHD audio tracks (at least the version I'm using (v 2.53) doesn't. How can I mux a TreuHD audio track into mkv? Is this possible? My audio file has the extension *.thd.

2. For playback i use Zoom Player which as you know is highly configurable, allowing me to select directshow filters for play back. Howver, I can't seem to get the Iron Man TrueHD audio track to play. In graphedit I tried using Nero File source (Async) > Nero Splitter > Nero Audio Decoder 2 > Audio Renderer. But nothing connects (yes, I renamed graphedit to "Recode"). Any suggestions on a working graph for TrueHD playback??

Thanks,
The_Keymaker

[Mark_A_W]

I just ripped my copy of Iron Man Blu-Ray using eac3to and have two questions.

1. It appears that MKmerge does not support muxing of TrueHD audio tracks (at least the version I'm using (v 2.53) doesn't. How can I mux a TreuHD audio track into mkv? Is this possible? My audio file has the extension *.thd.

2. For playback i use Zoom Player which as you know is highly configurable, allowing me to select directshow filters for play back. Howver, I can't seem to get the Iron Man TrueHD audio track to play. In graphedit I tried using Nero File source (Async) > Nero Splitter > Nero Audio Decoder 2 > Audio Renderer. But nothing connects (yes, I renamed graphedit to "Recode"). Any suggestions on a working graph for TrueHD playback??

Thanks,
The_Keymaker


You could convert the TrueHD track losslessly to FLAC - that's what most of use do.

Flac is easy to decode, thanks to Madshi writing the Madflac decoder ;)

[kypec]

No it doesn't. Currently eac3to isn't capable of reading IFO and handling SD DVDs. When or if Madshi will add support can only be answered by him.

Use DVD Decrypter.
Yes, definitely use DVD Decrypter and pay attention you choose only one PGC to decrypt in IFO mode. Select None file splitting as to finish with only one (rather huge) VOB file instead of many smaller ones. eac3to has (probably as many other tools) problems with multiangle VOB stuff and such. I recently used eac3to with 2 joined VOBs ripped via IFO mode (one PGC only) successfully - The Lion King 2 Special Edition.:)

[kypec]

Hmm (hear effete Brit accent in your mind's ear) ... I have serious doubts as to the efficacy of message boards for achieving results ... supposing one were to pay for them?
No offense man but what has your problem to do with eac3to? Tribler (http://en.wikipedia.org/wiki/Tribler) seems to me as completely unrelated to HD-DVD/BD/DVD audio/video processing for which eac3to is targeted. You shouldn't put your hopes in getting fast and effective answers from the forum members when you post at the wrong forum in the first place. If you anticipate to receive help directly from madshi on your problems you better try to contact him personally via PM/IM.
If I was in a bad mood I'd say your post is nothing but violating our forum rules (http://forum.doom9.org/forum-rules.htm), especially point 12. :readrule:
My advice to you is find Tribler discussion forum and try to ask for solution there.

[The_Keymaker]

You could convert the TrueHD track losslessly to FLAC - that's what most of use do.

Flac is easy to decode, thanks to Madshi writing the Madflac decoder ;)

Thanks Mark_A_W,

Converting to FLAC is what I usually do, but I've noticed, at least on my system (Krell PrePro, Pass Labs Amplification and B&W Nautilus 802s) that the bit streamed audio track sounds a little better than the same audio track when it is FLAC encoded/decoded. Technically there should be no difference as FLAC is a lossless compression format, but it sounds like something is missing...

In any event, if there is not a way to mux and playback a TrueHD audio track, i'll have to revert back to FLAC encoding/decoding.

Thanks!

[xkodi]

Technically there should be no difference as FLAC is a lossless compression format, but it sounds like something is missing...

In any event, if there is not a way to mux and playback a TrueHD audio track, i'll have to revert back to FLAC encoding/decoding

and why are you sure that TrueHD decoder you are using is bit-perfect decoder like bit-perfect eac3to to FLAC conversion and doesn't do some post-processing which makes the difference?

[rik1138]

You are free to name the channels inside DD/DTS, but still the channels are in wav format you must use the wav names and order:

Front Left - FL
Front Right - FR
Front Center - FC
Low Frequency – LF
Back Left - BL
Back Right - BR
Front Left of Center - FLC
Front Right of Center - FRC
Back Center - BC
Side Left - SL
Side Right - SR
...

I'm not sure I follow you... The letter codes I posted are the file names for the WAV files themselves. It's got nothing to do with being inside DD/DTS. That's how the individual WAV files are supposed to be named. It's a professional industry standard in Blu-Ray and DVD production. I've never seen WAV files or audio channels labeled like you've mentioned above... I'm not an audio engineer though, that might be where those designations originally came from...

Are you talking about when making a 7.1 channel single WAV file? Maybe that's where the confusion is coming from... I'm talking about when each channel is a separate mono WAV file. That's all the professional encoders will accept, they don't recognize multi-channel WAVs (well, anything above a stereo WAV that is). Uncompressed audio is never delivered that way to a BD/DVD production facility. It's always delivered as each speaker in it's own, mono WAV file... That's helpful, too, if there's a problem with one channel, they don't have to re-deliver or re-capture the entire audio track, just that one channel.

For example, if we were to get .WAV files for a 5.1 project, here's what they'd be named:
Dolby_C.wav
Dolby_L.wav
Dolby_LFE.wav
Dolby_LS.wav
Dolby_R.wav
Dolby_RS.wav

Each one is a 24-bit, mono WAV file. Those are the extensions that would be required by the encoding software...

Hope that helps explain what I was referring to... Individual WAV files, not multi-channel WAVs.

Rik

[tebasuna51]

That's all the professional encoders will accept, they don't recognize multi-channel WAVs
For what? professional?
Hope that helps explain what I was referring to... Individual WAV files, not multi-channel WAVs.
Yes I say individual mono wav files:

Channel_FL.wav
Channel_FR.wav
Channel_FC.wav
Channel_LF.wav
Channel_BL.wav
Channel_BR.wav
...

[The_Keymaker]

and why are you sure that TrueHD decoder you are using is bit-perfect decoder like bit-perfect eac3to to FLAC conversion and doesn't do some post-processing which makes the difference?

I'm not sure. I can only trust my ears.

By the way legions of posters on AVSforum have observed the same thing: Bitstreamed audio sounds different (better) than LPCM (decoded FLAC) from the SAME audio track. Some suspect it has something to do with the metadata.

In any event Until there are HDMI audio/video cards that support non-Protected Audio Path bit streaming, I'll have to make do with FLAC.

The_Keymaker

[honai]

By the way legions of posters on AVSforum have observed the same thing: Bitstreamed audio sounds different (better) than LPCM (decoded FLAC) from the SAME audio track.

Where's the double-blind listening test to support that statement? ;)

[StephenB]

Yes, definitely use DVD Decrypter and pay attention you choose only one PGC to decrypt in IFO mode. Select None file splitting as to finish with only one (rather huge) VOB file instead of many smaller ones. eac3to has (probably as many other tools) problems with multiangle VOB stuff and such. I recently used eac3to with 2 joined VOBs ripped via IFO mode (one PGC only) successfully - The Lion King 2 Special Edition.:)

This often works, but not always. The Studio splash stuff at the beginning often creates a gap that results in lipsync errors. Processing again with the video gap info sometimes fixes it, but often does not.

I've had better luck with the old PGCDemux tool, using its "by VOB id" mode. This creates one file per VID. You can then delete the very short ones at the beginning, and use EAC3TO to join/process the remainder.

[TinTime]

By the way legions of posters on AVSforum have observed the same thing: Bitstreamed audio sounds different (better) than LPCM (decoded FLAC) from the SAME audio track. Some suspect it has something to do with the metadata.

I think there are four options (none to do with metadata, unless you prefer DRC and channel down-mixing - I'm guessing not :D):

1. Transcoding TrueHD to flac with eac3to (i.e. libavcodec or libflac) is not perfect.
2. Windows and/or audio drivers are interfering with LPCM en route.
3. The decoding device processes TrueHD differently to LPCM in a way that sounds more pleasing, but is not perfect.
4. Placebo.

Point 1 seems unlikely as it can be tested (and I assume has been?) by anyone with access to a TrueHD encoder.
Point 2 is a distinct possibility and kind of hard to check. A DTS wav file is one way, although it's only 2 channel.
3 and 4 can be sorted out with the double-blind listening test mentioned by honai. It might also allow points 1 and 2 to be ignored :)

[The_Keymaker]

I think there are four options (none to do with metadata, unless you prefer DRC and channel down-mixing - I'm guessing not :D):

1. Transcoding TrueHD to flac with eac3to (i.e. libavcodec or libflac) is not perfect.
2. Windows and/or audio drivers are interfering with LPCM en route.
3. The decoding device processes TrueHD differently to LPCM in a way that sounds more pleasing, but is not perfect.
4. Placebo.

Point 1 seems unlikely as it can be tested (and I assume has been?) by anyone with access to a TrueHD encoder.
Point 2 is a distinct possibility and kind of hard to check. A DTS wav file is one way, although it's only 2 channel.
3 and 4 can be sorted out with the double-blind listening test mentioned by honai. It might also allow points 1 and 2 to be ignored :)

I trust eac3to and I can rule out placebo effect (listening to a movie I THOUGHT was bit streamed but sounded odd, is what set me on my quest). So I guess 2 and 3 are possibilities (based on the choices given).

By the way, the differences seem to be greater with the TrueHD/DTS-MA vs LPCM than the DD/DTS vs LPCM.

I have a high resolution audio playback system and suspect many people's systems do not possess enough resolution to distinguish the small differences I am hearing (Indeed, I recall reading once that studies show some can't distinguish the differences between 640kbps vs High bit rate formats).

Anyway, thanks for all the help. I do not want to go off-topic anymore in this thread!

The_Keymaker

[rik1138]

For what? professional?

Yes I say individual mono wav files:

Channel_FL.wav
Channel_FR.wav
Channel_FC.wav
Channel_LF.wav
Channel_BL.wav
Channel_BR.wav
...

Yeah, that's incorrect (for DVD/Blu-Ray delivery). Files are never delivered from the studios with names like that.

When I say professional encoders, I'm referring to Dolby's Media Encoder and DTS's DTS-HD Master Audio Suite Encoder as I mentioned in a previous post. These are the encoders the studios use for their titles. Neither one of these encoders would be able to automatically assign any of the WAV files you named above to a channel.

From the Manual for the DTS-HD Master Audio Suite Encoder:

Abbreviation Description
L Left
R Right
C Center
LFE Low Frequency Effects
Ls Left Surround
Rs Right Surround
Cs Center Surround
ss Surround Side
sr Surround Rear
S Surround
Lt/Rt Left Total/Right Total

The manual spells it out in even more detail when discussing every possible abbreviation that it recognizes (any other abbreviation is ignored and would have to be manually entered into the encoder):

Channel ID Definitions
L Left Speaker
R Right Speaker
Lt Left Total
Rt Right Total
C Center Speaker
LFE, LF, or SW Low Frequency Effects / Subwoofer
Ls Left Surround Speaker
Rs Right Surround Speaker
Lsr Left Surround Rear Speaker
Rsr Right Surround Rear Speaker
Lss Left Surround Side Speaker
Rss Right Surround Side Speaker
Lw Left Wide Speaker
Rw Right Wide Speaker
Lh Left Height Speaker
Rh Right Height Speaker
Lhs Left Height Side Speaker
Rhs Right Height Side Speaker
Cs Center Surround Speaker
S Surround Speaker
Ch Center Height Speaker
Oh Overhead Speaker

The Dolby Media Encoder expects the same abbreviations... Nothing is ever referred to as 'Front' or 'Back'. Front is assumed if there's no designation, and 'Back' is always referred to as 'Rear'.

Where did you get your abbreviations from? Is that from ProTools or something? I've never seen anything used in DVD/Blu-Ray production that uses those abbreviations...

[yesgrey]

By the way legions of posters on AVSforum have observed the same thing: Bitstreamed audio sounds different (better) than LPCM (decoded FLAC) from the SAME audio track. Some suspect it has something to do with the metadata.
I suspect they are not bypassing microsoft internal resamplers (kmixer in XP and WaveRT in Vista). Are they using reclock with Kernel Streaming in XP or with WASAPI in VISTA?

[tebasuna51]

Where did you get your abbreviations from? Is that from ProTools or something? I've never seen anything used in DVD/Blu-Ray production that uses those abbreviations...

Nope, is for wav specs (no dts or dd or ...)

This is a old doc http://www.cs.bath.ac.uk/~rwd/mulchaud.rtf

but you can see the same in M$ docs:
http://msdn.microsoft.com/en-us/library/aa474697.aspx

[73ChargerFan]

Where did you get your abbreviations from?
Microsoft programming documentation.

On another note, mkvmerge now muxes TrueHD.

[rik1138]

Nope, is for wav specs (no dts or dd or ...)

This is a old doc http://www.cs.bath.ac.uk/~rwd/mulchaud.rtf

but you can see the same in M$ docs:
http://msdn.microsoft.com/en-us/library/aa474697.aspx

I suppose that makes sense... Like I said, I only work in the DVD/Blu-Ray industry, not audio or recording. Dolby and DTS do not use the WAV file naming conventions created by Microsoft (and you don't use WAV files on DVD and Blu-Ray discs either, only as a source file before encoding/converting to DD/DTS/PCM). Microsoft created the WAV format, so they created the channel naming abbreviations. So, for a WAV file, that's probably accurate, if it's going to stay a WAV file. If the intent is to convert the WAV file to another format, the naming convention of that format should be used. Especially if the tools used to convert the file expect a certain naming convention.

I only brought this up because Madshi asked if there is a set standard for channel naming. The channel naming standard for DVD/Blu-Ray is like I mentioned above, not the one created by Microsoft. :D

His program is widely used in the professional industry, so I thought he might want to conform the file naming to match the other industry tools... It's up to him I suppose.

[The_Keymaker]

Microsoft programming documentation.

On another note, mkvmerge now muxes TrueHD.

Thanks ChargerFan, I'll download the latest revision from Mosu.

The_Keymaker

[TinTime]

I suppose that makes sense... Like I said, I only work in the DVD/Blu-Ray industry, not audio or recording. Dolby and DTS do not use the WAV file naming conventions created by Microsoft (and you don't use WAV files on DVD and Blu-Ray discs either, only as a source file before encoding/converting to DD/DTS/PCM). Microsoft created the WAV format, so they created the channel naming abbreviations. So, for a WAV file, that's probably accurate, if it's going to stay a WAV file. If the intent is to convert the WAV file to another format, the naming convention of that format should be used. Especially if the tools used to convert the file expect a certain naming convention.

I only brought this up because Madshi asked if there is a set standard for channel naming. The channel naming standard for DVD/Blu-Ray is like I mentioned above, not the one created by Microsoft. :D

His program is widely used in the professional industry, so I thought he might want to conform the file naming to match the other industry tools... It's up to him I suppose.

I guess the point is that eac3to doesn't know what your intent is. Its task in this case is to split an audio file into its individual channels and store them as wavs. That's it. Given that this is the end of its role it makes sense to follow MS naming conventions for what is an MS format. After all, if you wanted to process the wavs in some way and join them back together to make a multichannel wav then you'd want to stick with MS names anyway.

The only way around this is a switch for eac3to to tell it what your future intention is for its output, i.e. whether you want MS names, or DTS names, or Dolby names, or whatever. I think madness lies down that route :eek:

As you say it's up to madshi. My vote would be to stick with names as they are though, not least because suddenly changing them would probably cause grief to anyone with batch scripts containing the current names. Like me :)

[rik1138]

I guess the point is that eac3to doesn't know what your intent is. Its task in this case is to split an audio file into its individual channels and store them as wavs. That's it.

I would tend to disagree with that. The entire purpose of almost every application discussed here is ways to convert video/audio from one format to another for playback on computers, DVD players, Blu-Ray players, etc... I doubt anyone here leaves their audio in WAV format for that. You use eac3to to extract from one format and encode into another format... Thus the WAVs created from eac3to are specifically used to be re-encoded into another format, again, for the most part. Most people are trying to convert DTS to AC3, or the other way around, or to convert lossless codecs into PCM or FLAC. Me, I use it mostly to convert HD DVD Dolby to Blu-Ray Dolby/DTS. I also use it to extract DTS-HD MA audio and re-encode it into Dolby TrueHD (or the other way) so I can have both formats on my disc. That's good for A/B testing (or showing off) of sound systems and such... Not too many commercial discs put both lossless formats on the disc (technically, it's probably kind of pointless for the average viewer...)

And, as far as I know, any audio editing tool of any kind will recognize the DTS/Dolby standard naming convention, as well as the Microsoft one. You see, it's not just DTS/Dolby, it's _everyone else_, and then Microsoft (which is a common thing with Microsoft anyway... :rolleyes: ). Pro Tools, master tape layback, etc all use the same naming convention that DTS/Dolby use... Which Dolby/DTS probably just adopted from the standard convention used in the video industry. So if you put a switch in the tool for the output files names, the switch would be 'Microsoft names' or 'Everyone else's names'. That's all you'd need... Dolby and DTS do use the same naming.

As someone who works in the industry, I was just trying to help bring the tool up to industry standards, that's all... I didn't think it would cause such a huge debate! :D

It's funny you mention that your batch application wouldn't work if he changed the naming convention... Well, none of _my_ tools will work with current naming convention, and I don't have the ability to change that (since I didn't write the tools...) But I guess if everyone is used to using it the way it is, maybe it should just be left alone... I can rename files easily enough... :cool:

[kastrom]

On another note, mkvmerge now muxes TrueHD.

How about eac3? I can see it in the changelog but I havn't been able to mux in eac3 in an mkv. Is it possible? Should be very nice to have the original sound in the mkv.

/Kent

[xkodi]

I also use it to extract DTS-HD MA audio and re-encode it into Dolby TrueHD

and what tool are you using to create Dolby TrueHD?

[ACrowley]

and what tool are you using to create Dolby TrueHD?

theres NO Software Encoder or "Tool" available on Windwos Plattform. Not one single one!

So he can only use Dolby Media Encoder (SE) for MAC
http://www.dolby.com/professional/pro_audio_engineering/DMP_02.html

other Questions :
1. Its not possible to extract the German KingKong Bluray with eac3to.
Nothing happens after i select the Playlist.....

But it was possible to join the M2TS Files for the extended Version with TSMuxer

2. eac3to is patching the Bitdepth from DTS-HR Audio Tracks from 16 to 24bit ? I think it cant cause Problems/trouble (HDMi Passtrough for example)?

[rik1138]

theres NO Software Encoder or "Tool" available on Windows Plattform. Not one single one!

Officially, you are correct. Dolby has no plans to release a PC based encoder, which _REALLY_ sucks as the entire DVD/Blu-Ray productions workflow is PC based (film capture, encoding, authoring...) Dolby claims that there is no market or interest in a PC based encoder. And the Mac encoder is slower than sin... It can literally take half a day to encode a TrueHD stream. (We could probably put it on a faster Mac, but still... everyone complains about how slow it is...)

But, there's is a PC based MLP encoder that Dolby has written, but it's a beta only, and they will only loan it to you (as in, loan you the entire computer...) if you convince them you have a need for it (and they probably have no intention of releasing it). Currently it's required for doing segmented multi-angle features, so that's about the only place it's used. It only encodes the MLP stream (lossless part) of the TrueHD though, you still have to encode the AC3 separately and then use Scenarist or something to mux it together. But it encodes MLP 100s of times faster than the Media Encoder which makes it worth using just for that. They are incorporating the multi-segment processing into the Mac tool, so that will eliminate the 'need' for the PC encoder I suppose...

Fortunately, DTS-HD MA works perfectly with segmented features, so that's a far easier-to-use option... And it encodes faster than Dolby on either platform...

[nonymous]

I Love eac3to! It's great and versatile. Used in on a couple of occasions and always find it being able to do the one thing I need to do.

Would you be open to a Linux / Mac OSX port? I'd be willing to help with the autoconfig, make and conditional compilation tricks required! Especially since I would myself love to have a command line version for my mac. I understand that you have hook-ups with Nero and other things but eac3to is still VERY USEFUL even without ANY of those external programs. Let me know, I'd love to help. I understand you may not want to share the code but it could still be distributed in the binary form (we could have a rpm for linux and dmg for mac or something like that). PM me please.

[buzzqw]

Hi Madshi

could you add support for avs (audio) files ? as wavi ?

would be handy for using a unique tool for audio encoding (i use eac3to for decoding AND piping to nero/ogg/lame...)

thanks!

BHH

[tebasuna51]

could you add support for avs (audio) files ? as wavi ?

I can't understand your request, please explain it.

[ACrowley]

Officially, you are correct. Dolby has no plans to release a PC based encoder, which _REALLY_ sucks as the entire DVD/Blu-Ray productions workflow is PC based (film capture, encoding, authoring...) Dolby claims that there is no market or interest in a PC based encoder. And the Mac encoder is slower than sin... It can literally take half a day to encode a TrueHD stream. (We could probably put it on a faster Mac, but still... everyone complains about how slow it is...)

But, there's is a PC based MLP encoder that Dolby has written, but it's a beta only, and they will only loan it to you (as in, loan you the entire computer...) if you convince them you have a need for it (and they probably have no intention of releasing it). Currently it's required for doing segmented multi-angle features, so that's about the only place it's used. It only encodes the MLP stream (lossless part) of the TrueHD though, you still have to encode the AC3 separately and then use Scenarist or something to mux it together. But it encodes MLP 100s of times faster than the Media Encoder which makes it worth using just for that. They are incorporating the multi-segment processing into the Mac tool, so that will eliminate the 'need' for the PC encoder I suppose...

Fortunately, DTS-HD MA works perfectly with segmented features, so that's a far easier-to-use option... And it encodes faster than Dolby on either platform...

yeah youre right.
When i would have the choce : Ofcourse i would by the DTS HD MAS Suite. But its to expensive (~1500$)

[Jeff Flowerday]

I Love eac3to! It's great and versatile. Used in on a couple of occasions and always find it being able to do the one thing I need to do.

Would you be open to a Linux / Mac OSX port? I'd be willing to help with the autoconfig, make and conditional compilation tricks required! Especially since I would myself love to have a command line version for my mac. I understand that you have hook-ups with Nero and other things but eac3to is still VERY USEFUL even without ANY of those external programs. Let me know, I'd love to help. I understand you may not want to share the code but it could still be distributed in the binary form (we could have a rpm for linux and dmg for mac or something like that). PM me please.

You do know that madshi develops in delphi?

[buzzqw]

I can't understand your request, please explain it.

eac3to.exe audio.avs 1: stdout.wav | lame - -cbr 128 audio.mp3

where audio.avs is
nicac3source("c:\audio.ac3",2)


as example

BHH

[Kurtnoise]

why not using wavi for that ?

or this ?
eac3to.exe audio.ac3 1: stdout.wav | lame - -cbr 128 audio.mp3

[tebasuna51]

If eac3to is not used to decode, or to demux from a container, for what you need eac3to?

Like Kurtnoise say use wavi if you need avs for a complex change or use directly the eac3to decoders (the same than nicaudio at least).

[buzzqw]

within avs i can use directshowsource for input file

and i prefer to not use another program (like wavi) but, if possibile (as request) to add avs support to eac3to

for example FLV video file isn't supported by eac3to, BUT with AVS/DSS support would be easy to handle it

BHH

[rik1138]

yeah youre right.
When i would have the choce : Ofcourse i would by the DTS HD MAS Suite. But its to expensive (~1500$)

It's cheaper than Dolby's though. :D Dolby Media Encoder SE is about $2500-3000.

Rik

[StephenB]

Quote:
Originally Posted by StephenB
I would also like a switch that suppressed the second pass.

If you can find a good reason for that?

When I use EAC3TO to extract from vob files, on about 1 DVD out of 20 the audio gap processing causes a lip sync error. (The vob files seem clean BTW, and only contain the PGC I am interested in). Re-processing (using the video gap) does not help. If I do the same extraction with version 2.80 using -ignoreDiscon there is no lipsync problem. I'd rather not have to keep version 2.80 around; I'd prefer it if the flag was put back.

[Snowknight26]

madshi, mkv files edited with the header editor in mkvmerge aren't recognized by eac3to.

[kalia]

Hi Madshi

could you add support for avs (audio) files ? as wavi ?

would be handy for using a unique tool for audio encoding (i use eac3to for decoding AND piping to nero/ogg/lame...)

thanks!

BHH

yeah madshi, it would definitely be handy! http://www.photopile.info/img/c/8.gif

[sucker]

i tried importing a converted TrueHD2WAV track into Scenarist, but scenarist MUI Generator tells me "The Scpecified ChannelAssign is not corresponding to data"
i also tried the same TrueHD track converted to PCM and PCM2tsmu both won´t be accepted by Scenarist, so i´m hoping it´s a mixup between the eac3to channel layout and the layout that Scenarist expects, anybody able to help here?

[rik1138]

Well, we know eac3to doesn't follow the standards used by the DVD/Blu-Ray industry for channel naming, it's using Microsoft's naming convention instead.. It's possible that the layout is affected the same way. I'm not sure exactly how Scenarist knows what channel is what, unless that's indicated in the WAV header/meta somewhere (and it might be expecting a specific layout). It would be expecting the channels to be in this order: L, C, R, LS, RS, LFE for 5.1

Just extract your TrueHD into individual mono WAVs instead, and use the MUI Generator to create your multi-channel PCM that way. (Open the MUI Generator, select your _LEFT_ channel WAV in the ES File box. Then select 'Use Multi LPCM function' and click Input LPCM Files. You can then select the rest of your files and it will create a .VES file you can drop into Scenarist.)

[AnryV]

eac3to v3.15
command line: eac3to eng.thd eng.wavs
------------------------------------------------------------------------------
TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
Removing TrueHD dialog normalization...
Decoding with libav/ffmpeg...
Writing WAVs...
Creating file "eng.R.wav"...
Creating file "eng.SL.wav"...
Creating file "eng.LFE.wav"...
Creating file "eng.SR.wav"...
Creating file "eng.C.wav"...
Creating file "eng.L.wav"...
The original audio track has a constant bit depth of 16 bits.
Superfluous zero bytes detected, will be stripped in 2nd pass.
Starting 2nd pass...
Removing TrueHD dialog normalization...
Decoding with libav/ffmpeg...
Reducing depth from 24 to 16 bits...
Writing WAVs...
Creating file "eng.R.wav"...
Creating file "eng.C.wav"...
Creating file "eng.SL.wav"...
Creating file "eng.L.wav"...
Creating file "eng.SR.wav"...
Creating file "eng.LFE.wav"...
The processed audio track has a constant bit depth of 16 bits.
eac3to processing took 20 minutes, 35 seconds.
Done.

eac3to v3.15
command line: eac3to eng.thd eng.wavs -nero
------------------------------------------------------------------------------
TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
Disabling DRC for Nero (E-)AC3 decoding...
Removing TrueHD dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
DirectShow reports 5.1 channels, 24 bits, 48khz
Writing WAVs...
Creating file "eng.SR.wav"...
Creating file "eng.C.wav"...
Creating file "eng.L.wav"...
Creating file "eng.SL.wav"...
Creating file "eng.R.wav"...
Creating file "eng.LFE.wav"...
The original audio track has a constant bit depth of 24 bits.
eac3to processing took 12 minutes, 9 seconds.
Done.

What bit depth the original track has? 24 or 16?

[rik1138]

If it's TrueHD, it should be 24bit original audio. This line is interesting:

"Superfluous zero bytes detected, will be stripped in 2nd pass."

Maybe libav can't do true 24 bit, so it's making all the bits above 16 zeros, which eac3to then strips off...

I'd stick with the nero decode.

[TinTime]

Although TrueHD is always 24 bit not all of those bits are necessarily used. There are plenty of TrueHD tracks with only 16 or 20 bits per sample used. When eac3to decodes TrueHD it does so as 24 bit but if it reaches the end of file and discovers that 8 of the 24 bits are just zero padding it will strip them and produce a 16 bit output file, as has happened here.

What I'm not sure about is why Nero output isn't identical. I thought it was for 5.1 channels or under. It is outputting a message "Disabling DRC for Nero (E-)AC3 decoding..." but not mentioning TrueHD. If DRC is still being applied by the Nero TrueHD decoder then that would account for it. I suspect it's just a generic message for the Nero filter though.

Because of Nero's limitation to 5.1 channels, libav (i.e. the default) is the recommended decoder to use for TrueHD.

[The_Keymaker]

Microsoft programming documentation.

On another note, mkvmerge now muxes TrueHD.

OK, I've installed the latest release of MKVmerge and it still gives me an error message and says it is an unknown type when I try to mux the TrueHD audio of Iron man.

Am I missing something?

The_Keymaker

[poisondeathray]

OK, I've installed the latest release of MKVmerge and it still gives me an error message and says it is an unknown type when I try to mux the TrueHD audio of Iron man.

Am I missing something?

The_Keymaker

The beta version is supposed to add support. Scroll down for more info, and download link
http://forum.doom9.org/showthread.php?t=120648&page=42

[sucker]

Just extract your TrueHD into individual mono WAVs instead, and use the MUI Generator to create your multi-channel PCM that way. (Open the MUI Generator, select your _LEFT_ channel WAV in the ES File box. Then select 'Use Multi LPCM function' and click Input LPCM Files. You can then select the rest of your files and it will create a .VES file you can drop into Scenarist.)

i´m pretty new to Scenarist and didn´t notice this possibility, thx alot for the info
btw is eac3to able to apply delays to TrueHD streams when demuxing to TrueHD or only when the TrueHD is beeing converted at the same time to PCM/WAV...?

[The_Keymaker]

The beta version is supposed to add support. Scroll down for more info, and download link
http://forum.doom9.org/showthread.php?t=120648&page=42


Found it!

Thanks Poisondeathray :thanks:

[AnryV]

Because of Nero's limitation to 5.1 channels...

How it is connected with the asked question about bitdepth?
I wish to be assured that libav/ffmpeg decoder does not lose the information.

[nurbs]

It doesn't lose quality. Nero will output 24 bit no matter what the actual bitdepht of the TrueHD file is.

[rik1138]

He was asking about libav/ffmpeg. Can libav output 24bit wavs if the source really is full 24bit?

Is there an easy way to determine if a 24-bit WAV really is 24-bit (and not just padded)?

[TinTime]

He was asking about libav/ffmpeg. Can libav output 24bit wavs if the source really is full 24bit?

Yes :) It is decoding to 24 bits here. It's just when it reaches the end that eac3to confirms that only 16 bits are used so it runs a second pass.

Is there an easy way to determine if a 24-bit WAV really is 24-bit (and not just padded)?

Just run it through eac3to:
eac3to.exe input.wav output.wav

If it's got superfluous bits eac3to will strip them. It's probably worth trying this with AnryV's output from the Nero decoder.

@AnryV - Try converting to a single wav with Nero and then running that wav through eac3to again.

I would have expected this to happen automatically with the output from Nero in this case but maybe it doesn't. I thought libav and Nero behave identically, except for the 5.1 channel limitation for Nero. I may be wrong about this though. I'm confident that madshi will have picked the best decoder for the job though so I'd stick with libav. It doesn't say in the eac3to help that for best TrueHD decoding you need Nero. It used to IIRC, before libav TrueHD decoding was refined.

[AnryV]

@AnryV - Try converting to a single wav with Nero and then running that wav through eac3to again.


eac3to v3.15
command line: eac3to eng.C.wav eng.C1.wav
------------------------------------------------------------------------------
WAV, 1.0 channels, 1:55:25, 24 bits, 1152kbps, 48khz
Reading WAV...
Writing WAV...
Creating file "eng.C1.wav"...
The original audio track has a constant bit depth of 24 bits.
eac3to processing took 1 minute, 27 seconds.
Done.

eng.C.wav was created by Nero decoder as mentioned above.

???:confused:

[kastrom]

I'm trying to rip Total Recall from HD-DVD and I received a gaps file and have huge sync problems. I have run it again and the log files says it uses the gap file but there is still over three seconds differences between picture and the sound. Is is possible to fix the sync?

/Kent

[The_Keymaker]

OK, I loaded the beta version of MKVtoolnix (MKVmerge) and muxed the Iron man TrueHD+AC3 audio and Video.

However, when I attempted to play back the file using Zoom Player, only the VIDEO showed up in Zoom Player. An audio stream was not found.

I'm assuming this is an MKVmerge issue and not an eac3to issue since I was able to play the TrueHD+AC3 file fine as a standalone file (i.e., not in a MKV container).

I'll take this issue up on the MKVtoolnix forum. Just wanted to share my experience in case any is trying to do the same thing.

[Mark_A_W]

OK, I loaded the beta version of MKVtoolnix (MKVmerge) and muxed the Iron man TrueHD+AC3 audio and Video.

However, when I attempted to play back the file using Zoom Player, only the VIDEO showed up in Zoom Player. An audio stream was not found.

I'm assuming this is an MKVmerge issue and not an eac3to issue since I was able to play the TrueHD+AC3 file fine as a standalone file (i.e., not in a MKV container).

I'll take this issue up on the MKVtoolnix forum. Just wanted to share my experience in case any is trying to do the same thing.


At a wild guess you are using the Haali Splitter? The Haali Splitter does not "see" TrueHD streams. You need to use the MPC-HC Mpeg2 splitter (or MPC HC itself, using the internal splitter).

Or convert to FLAC. FLAC just works.

[leeperry]

Is there an easy way to determine if a 24-bit WAV really is 24-bit (and not just padded)?
both Wavelab & Ozone will show you whether all the bits are actually in use...at least it tells you right away, no need to wait 10 minutes ;)

[The_Keymaker]

At a wild guess you are using the Haali Splitter? The Haali Splitter does not "see" TrueHD streams. You need to use the MPC-HC Mpeg2 splitter (or MPC HC itself, using the internal splitter).

Or convert to FLAC. FLAC just works.

Yes you are correct I am using the haali splitter. I will acquire and and try the MPC HC Splitter.

Also, I've tried FLAC but it just does not sound the same (in my system) as bitstream (TrueHd, DD, DTS etc.). I've done some research since I discovered this fact and it appears some think it has to do with excessive jitter wrought by the HDMI transmission interface ( I use HDMI from my HTPC rather than SPDIF). This jitter apparently is less of an issue when bit streaming since the way bitstreams are decoded is less dependent on the master clock.

See this link (post #6016): http://www.avsforum.com/avs-vb
/showthread.php?p=16205544#post16205544

And here: http://www.avsforum.com/avs-vb/showthread.php?t=1134289

UPDATE: i installed the MPC HC splitter (from here: http://sourceforge.net/project/showfiles.php?group_id=170561&package_id=264678) but still no joy. Can someone that has successfully muxed (with video) and played back a TrueHD track post their graph? You can PM me if you don't want to tard up the eac3to thread, although there may be others interested as well.

[Mark_A_W]

Yes you are correct I am using the haali splitter. I will acquire and and try the MPC HC Splitter.

Also, I've tried FLAC but it just does not sound the same (in my system) as bitstream (TrueHd, DD, DTS etc.). I've done some research since I discovered this fact and it appears some think it has to do with excessive jitter wrought by the HDMI transmission interface ( I use HDMI from my HTPC rather than SPDIF). This jitter apparently is less of an issue when bit streaming since the way bitstreams are decoded is less dependent on the master clock.

See this link: http://www.avsforum.com/avs-vb
/showthread.php?p=16205544#post16205544

And here: http://www.avsforum.com/avs-vb/showthread.php?t=1134289

UPDATE: i installed the MPC HC splitter (from here: http://sourceforge.net/project/showfiles.php?group_id=170561&package_id=264678) but still no joy. Can someone that has successfully muxed (with video) and played back a TrueHD track post their graph? You can PM me if you don't want to tard up the eac3to thread, although there may be others interested as well.


Actually, I think the latest Haali splitter may "see" TrueHD, but it doesn't label it (which is ok if it's the only audio stream present).

[TinTime]

both Wavelab & Ozone will show you whether all the bits are actually in use...at least it tells you right away, no need to wait 10 minutes ;)

But if you have a 24 bit wav file with 24 valid bits per sample set in the header how can Wavelab & Ozone tell for sure without spending 10 minutes reading the whole file?

I suppose they can read the first few seconds and see what that looks like, but the only sure-fire way to check all the bits is to check all the bits!

[TinTime]

What bit depth the original track has? 24 or 16?

I've tried two TrueHD tracks (16 bit from "I Am Legend" and 24 bit from "Cloverfield") with both decoders and couldn't replicate your problem. How did you create the .thd file? What movie did it come from and how did you demux it, or have you encoded it yourself?

[AnryV]

I've tried two TrueHD tracks (16 bit from "I Am Legend" and 24 bit from "Cloverfield") with both decoders and couldn't replicate your problem. How did you create the .thd file? What movie did it come from and how did you demux it, or have you encoded it yourself?
"Appaloosa."
.thd demuxed by eac3to.

[dstoe]

rather old, but still relevant:

Thank you. Right now there's no way to donate. Maybe I'll make that available later...

any update on this?

[leeperry]

But if you have a 24 bit wav file with 24 valid bits per sample set in the header how can Wavelab & Ozone tell for sure without spending 10 minutes reading the whole file?

I suppose they can read the first few seconds and see what that looks like, but the only sure-fire way to check all the bits is to check all the bits!
both Ozone & Wavelab have bit meters, but sure this won't be as accurate as checking the whole file...but if the bottom bits lighten up, it's a good guess that you do have a true 24bit file :o

from the Ozone help file :

http://www.image-load.eu/out.php/t156901_ozone.png (http://www.image-load.eu/out.php/i156901_ozone.png)

[leeperry]

if you see this, this is 24bit 100% sure and it took 10 secs to find out :)

http://www.image-load.eu/out.php/i156904_24.png

[TinTime]

"Appaloosa."
.thd demuxed by eac3to.

According to this list (http://www.avsforum.com/avs-vb/showthread.php?t=760714) Appaloosa's TrueHD track is 16 bit, so it looks like something funny is happening with Nero decoding here. This is really a question for madshi. Or stick with the default decoding which seems to be correct in this case.

if you see this, this is 24bit 100% sure and it took 10 secs to find out :)

http://www.image-load.eu/out.php/i156904_24.png

You're quite right - I didn't think of it that way around. If the file is genuinely 24 bit you can find out immediately. If it's 16 bit you need to read the whole lot to confirm.

[leeperry]

If it's 16 bit you need to read the whole lot to confirm.
not really, if it's 16bit it will look like this :

http://www.image-load.eu/out.php/i156906_16.png

you can use Ozone in ffdshow :)

[TinTime]

not really, if it's 16bit it will look like this :

http://www.image-load.eu/out.php/i156906_16.png

you can use Ozone in ffdshow :)

If you've got a film with something like an introduction from the director at the beginning, and the audio introduction is 16 bit and the main movie audio is 24 bit, then what does Ozone show you when you're watching the intro? If it's the above picture then it would lead to an incorrect conclusion about the full audio track.

I'm being kind of pedantic here (but what the hell :devil:) because I don't know if such a soundtrack exists (although there's no reason why it couldn't). If you think that a 24 bit audio file is only 16 bit then the only way to confirm it is to read the file until you find a > 16 bit sample. If it's entirely 16 bit then you'll end up reading the whole file.

[leeperry]

If you've got a film with something like an introduction from the director at the beginning, and the audio introduction is 16 bit and the main movie audio is 24 bit
haha sure, but Ozone works in realtime in ffdshow....so you can seek a number of times and if the bottom 8bit remain turned off you know it's 16bit, anyway I was simply answering to rik1138 who was seeking an easy way to find out ;)

Is there an easy way to determine if a 24-bit WAV really is 24-bit (and not just padded)?

[leeperry]

just a last word on bit meters...you'll have to use Wavelab because the winamp plugins work in 16int only in ffdshow audio(even if you only check 32float processing) :o

here's a test on a 24/96 stereo FLAC, captured via graphedit.
left side is ffdshow audio w/ 3 filters : volume/mixer/EQ, right side is w/ a winamp2 plugin afterwards.

http://www.image-load.eu/out.php/i156949_eeeeq.png

[Blue_MiSfit]

Is cool!

@Madshi:

I have found an unusual MPEG Program Stream that contains MPEG-2 video an AC3 audio, but eac3to doesn't recognize the audio stream (thankfully I was able to use DGIndex to extract the AC3). It complains like this:

Z:\>eac3to Alaska-SpiritoftheWild.mpg
MPG, 1 video track
1: MPEG2, 1080p24 (16:9)
v01 The video framerate is correct, but rather unusual.


The MPEG-2 is indeed 24.0fps exactly, but it seems after this point eac3to gives up.

Here's a 25mb sample:

http://www.mediafire.com/?sharekey=47de74c3f78774007f7ec40ada4772a66f2980d865a0ad6c5621d66e282a0ee8

Please let me know if you need more!

~MiSfit

[odin24]

Two new features added: Sony wave64 support and mkv chapters.

While demuxing BDs with MeGUI (which uses eac3to), should I use W64 for PCM tracks. I'm really only familiar with the three main HD audio extensions, MeGUI doesn't offer a PCM extension.

Thanks.

[tebasuna51]

While demuxing BDs with MeGUI (which uses eac3to), should I use W64 for PCM tracks. I'm really only familiar with the three main HD audio extensions, MeGUI doesn't offer a PCM extension.

Yes, w64 is like wav file (PCM data with a header) but without the wav limits (2/4 GB).

With multichannel audio and high bitdepth/samplerate is easy reach this wav limits then the w64 header is recommended.

NicAudio AviSynth plugin can read w64 files then can be used with MeGUI also.

[tebasuna51]

(Seems TsMuxeR 1.9.1 still don't work with w64 files)

@madshi. Seems there are a bug in eac3to with w64 files like the recent with wav files.

This file, created by eac3to, is not recognized by eac3to. (work fine with other softs):

6x321.w64 (http://www.sendspace.com/file/h196ug)

[madshi]

Sorry for the lack of replies lately. I'll reply to all questions and comments and get back to eac3to bug fixing (at least) this or next weekend.

If you're interested in why eac3to development has been slow recently, you may want to look here:

http://forum.doom9.org/showthread.php?t=146228

:D

[TinTime]

If you're interested in why eac3to development has been slow recently, you may want to look here:

http://forum.doom9.org/showthread.php?t=146228

:D

Now that looks very interesting...

If madVR is half as good as eac3to you're on to a winner.


Keep up the good work! :thanks:

[jolson]

Tried to convert a 5.1 TrueHD file to 1.5Mbit DTS, but got an error:

eac3to v3.12
command line: "C:\Program Files\eac3to\eac3to.exe" th.ac3 dm.dts -resampleto48000 -1536
------------------------------------------------------------------------------
TrueHD/AC3, 5.1 channels, 96khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB)
Extracting TrueHD stream...
Removing TrueHD dialog normalization...
Decoding with libav/ffmpeg...
Resampling to 48khz...
Reducing depth from 64 to 32 bits...
Writing WAVs...
Creating file "dm.R.wav"...
Creating file "dm.C.wav"...
Creating file "dm.LFE.wav"...
Creating file "dm.SL.wav"...
Creating file "dm.SR.wav"...
Creating file "dm.L.wav"...
Clipping detected, a 2nd pass will be necessary. <WARNING>
The original audio track has a constant bit depth of 24 bits.
The processed audio track has a constant bit depth of 32 bits.
Starting 2nd pass...
Extracting TrueHD stream...
Removing TrueHD dialog normalization...
Decoding with libav/ffmpeg...
Resampling to 48khz...
Reducing depth from 64 to 32 bits...
Writing WAVs...
Creating file "dm.LFE.wav"...
Creating file "dm.SR.wav"...
Creating file "dm.L.wav"...
Creating file "dm.R.wav"...
Creating file "dm.C.wav"...
Creating file "dm.SL.wav"...
The processed audio track has a constant bit depth of 32 bits.
Encoding DTS <1536kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.21.0.
Starting Surcode DTS Encoder failed. <ERROR>

Any ideas what can be wrong, and is there an option to have eac3to not remove the temporary .wav files, at least unless the DTS file creation is successful?

[tebasuna51]

For what you need a dts file if you have an ac3 640 kb/s?

Maybe you can make a flac, but with a -down24, the 64 or 32 bits product of resampling is enough.
Or a flac with 24/96.

BTW, maybe Surcode don't work with 32 bit float.

[TinTime]

I think it's using 32 bit int which should be ok for Surcode.

[jolson]

For what you need a dts file if you have an ac3 640 kb/s?
I don't yet have a "Media Extender" that can play TrueHD/DTS Master Audio (only HDI's Dune HD Center & BD Prime can do that) but I want to get the best quality I can play at the moment - which is 1,5Mbps DTS. So I convert to that from the Dolby TrueHD track.

I think my problem mentioned above was a firewall issue, eac3to didn't have permission to call Surcode but the dialogue wasn't modal so when I didn't see it and answer it disappeared after a while. When I reran the same command I got to answer yes to allow that call and all went through.

The created DTS track plays well in Foobar, but when I try to use tsMuxerGUI to mux together the vc1 track with the dts track the program complains "Can't detect stream type". Seems like Surcode doesn't leave a well-formed result...

But after running that DTS track through eac3to it becomes usable for tsNuxerGUI :)

[jamos]

(Seems TsMuxeR 1.9.1 still don't work with w64 files)

@madshi. Seems there are a bug in eac3to with w64 files like the recent with wav files.

This file, created by eac3to, is not recognized by eac3to. (work fine with other softs):

6x321.w64 (http://www.sendspace.com/file/h196ug)

tsmuxer 1.9.4 works with your file though..cheers no more need for pcm2tsmu

n/m got a error while muxing a big w64 file about 1/4 of the way through...not sure if its eac3to extract to w64 format or tsmuxer that is the problem.

[tebasuna51]

...but I want to get the best quality I can play at the moment - which is 1,5Mbps DTS. So I convert to that from the Dolby TrueHD track.

Yes, but I doubt than a dts 1536 Kb/s generated by Surcode is better than the core ac3 640 Kb/s.

[Blender]

Madshi,
I cannot get EAC3TO to write to 6tb raid arraywith 4k block size(to defeat>2tb limit in windows XP) .

Areca 1130 controller.

Is this a known issue? Is there a work around?

[73ChargerFan]

Yes, but I doubt than a dts 1536 Kb/s generated by Surcode is better than the core ac3 640 Kb/s.

It has better bass & rear separation. DTS has always been better than dolby digital.

[deathlord]

Madshi,
I cannot get EAC3TO to write to 6tb raid arraywith 4k block size(to defeat>2tb limit in windows XP) .

Areca 1130 controller.

Is this a known issue? Is there a work around?

Hm, I have no problems writing to such an array (highpoint controller).

[kurt]

It's probably not the best place to ask, but what am I supposed to do with 23,975 mkvs? I think they don't play stutterfree on Popcorn Hour and what I'm doing righ now is to demux all streams and mux it new with mkvmerge while changing fps in the container to 24000/1001 --> no judder on PCH. I guess changeto23.976 would lead to the same effect.

But what about the audio? is the difference neglectable? or should I use changeto23,976 to the audiostreams? is there a quality loss when changing fps to the DTS stream?

Edit:

I did some math:

2h movie = 7200s

0,001 frames difference for 1s --> 7,2 frames in total.

1s ---- 23,976 frames
xs ---- 7,2 frames

x = 0,300s = 300 ms

hm, 300 ms is quite a bit. Should be noticed at the end, I think 25-50 ms is noticable... I looked into some remuxed videos and didn't see such difference at the end of the movies....

Am I missing something? maybe mkvmerge did changing the fps for both, video and audio? (which would be great btw)

[tebasuna51]

It has better bass & rear separation. DTS has always been better than dolby digital.

Is your respectable opinion, but not the results of blind test.

Remember we are talking about the same source encoded with ac3 (supposed well encoded) and dts (Surcode).

Better bass and rear separation?
The bass never is a problem for a encoder.
The rear separation must be in the source.

[tbean]

When I test eac3to, the message regarding Nero Audio Decoder says:
"Nero Audio Decoder (Nero 7) is not working correctly
http://www.nero.com/eng/store-blu-ray.html"

When I try to access the link http://www.nero.com/eng/store-blu-ray.html, the page that comes up says nothing about the Nero Audio Decoder.

Can someone tell me the correct path to find the Nero Audio Decoder eac3to wants?

Thanks,
Tom

[TinTime]

Unfortunately the blu-ray / hd-dvd plugin is not available any more.

[ps3hacker]

Is your respectable opinion, but not the results of blind test.

A dts file created from surcode will most definitely sound better than a core ac3 file if the source is the same. Such as a lossless audio track, like true hd. Remember, there is no magic way the studios have to encode an ac3 file. The same could be said of dts. Surcode outputs the same dts file as a studio would(if not very, very, close)

[buzzqw]

@madshi

since eac3to (ffmpeg) is unable to decode correctly aac SBR and so want use nero decoder, would be possible to add support for Monogram AAC decoder (from radlight) ?
it's free/gpl

so, when aac sbr is to decode, check nero, else monogram, else ffmpeg

thanks

BHH

[jamos]

@madshi

I am having issues with w64 madshi. same as tabasuna51 is. also when I use a w64 in tsmuxer it will give a error after a minute or so. Roman tried a large w64 that he created then used tsmuxer and it works fine for him. He suggested it may be a header error. This only happens using DTS-ma converted to W64. true-hd to W64 does not give a error but the w64 file seems small.

link
http://forum.doom9.org/showthread.php?p=1272478#post1272478

[jamos]

If you're interested in why eac3to development has been slow recently, you may want to look here:

http://forum.doom9.org/showthread.php?t=146228

:D

very nice! love to see a GPU based renderer..:D

[jamos]

Is it correct to have a w64 file be less size than the original true-hd file?

thats what I am getting when I convert a true-hd file to w64 using eac3to. When I convert the true-hd file to pcm it is 7 gig. compared to the original 2 gig. true-hd file (which seems correct).

[ACrowley]

Mh, Strange.
eac3to gives Warnings on a lot of 23.976Fps 1080p x264 MKV/M2TS

M2TS, 1 video track, 2 audio tracks, 1:27:08, 24p /1.001
1: h264/AVC, 1920x800 23.976p (12:5)
2: DTS Hi-Res, German, 5.1 channels, 24 bits, 3018kbps, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
3: DTS, English, 5.1 channels, 24 bits, 755kbps, 48khz
[v01] The video bitstream is encoded in a non-standard framerate. <WARNING>

But the Video is surely encoded in 23.976FPs (trough AVS with DGIndex/Decode). All other Tools/Palyser shows me correct 23.976Fps.

Whats wrong there?

[honai]

Mh, Strange.
eac3to gives Warnings on a lot of 23.976Fps 1080p x264 MKV/M2TS

M2TS, 1 video track, 2 audio tracks, 1:27:08, 24p /1.001
1: h264/AVC, 1920x800 23.976p (12:5)
2: DTS Hi-Res, German, 5.1 channels, 24 bits, 3018kbps, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
3: DTS, English, 5.1 channels, 24 bits, 755kbps, 48khz
[v01] The video bitstream is encoded in a non-standard framerate. <WARNING>

But the Video is surely encoded in 23.976FPs (trough AVS with DGIndex/Decode). All other Tools/Palyser shows me correct 23.976Fps.

Whats wrong there?

The container says 24/1001 (which is a fraction, not a fixed number), but the video inside the container says 23.976 (which is a fixed number, not a fraction), so because of that mismatch you get a warning.

[ACrowley]

ah ok thank you

But some x264 Files with 23.976fps are working without Warnings but other Files not ?

Found this Post by Madshi
http://forum.doom9.org/showpost.php?p=1244513&postcount=8111

[shon3i]

A dts file created from surcode will most definitely sound better than a core ac3 file if the source is the same. Such as a lossless audio track, like true hd. Remember, there is no magic way the studios have to encode an ac3 file. The same could be said of dts. Surcode outputs the same dts file as a studio would(if not very, very, close)
I don't think so, DTS 1.5 have near quality as DD 448 and DTS is somewhere better, somewhere DD, but DD 640 produce alot better than any DTS codec including DTS Pro Suite Encoder, and Surcode. I doing some deep ABX, chanel vs chanel vs original with HD 650 earphones. Anyway EBU aslo find same in their Multichanel ABX and here is publicy doc http://www.ebu.ch/CMSimages/en/tec_doc_t3324-2007_tcm6-53801.pdf. DTS is very bad option for archiving except DTS HD-MA offcourse. It produce very large file and quality are not superb, not for my ears :) IMHO

[magic144]

hi madshi, just to report...

"Bitstream parsing for track 3 failed" msg reported when running eac3to (3.15) on an mkvtoolnix/mkvmerge(2.6.0)-created .mkv file for a muxed VobSub track (3) created with BDSup2Sub 3.1.0 (i.e. idx/sub file pair source)
(other/first 2 tracks are h.264 720p video and DTS audio)

.mkv file plays just fine with ZoomPlayer using VSFilter

[djesteban]

Guys, is there an advantage using eac3to to demux audio/video track instead of working with tsMuxer?... or do they do the exact same thing on this level?
Thanks

[laserfan]

Guys, is there an advantage using eac3to to demux audio/video track instead of working with tsMuxer?...AFAICT wrt demuxing, eac3to properly handles seamless branching discs (with attendant audio gaps) while tsMuxeR does not.

[SomeJoe]

I don't think so, DTS 1.5 have near quality as DD 448 and DTS is somewhere better, somewhere DD, but DD 640 produce alot better than any DTS codec including DTS Pro Suite Encoder, and Surcode. I doing some deep ABX, chanel vs chanel vs original with HD 650 earphones. Anyway EBU aslo find same in their Multichanel ABX and here is publicy doc http://www.ebu.ch/CMSimages/en/tec_doc_t3324-2007_tcm6-53801.pdf. DTS is very bad option for archiving except DTS HD-MA offcourse. It produce very large file and quality are not superb, not for my ears :) IMHO

Be careful when looking through that EBU document. Many of the test codecs are Dolby Digital Plus, not Dolby Digital.

In all the phase 1 codecs, DTS 1.5 Mbps outperformed every other codec. It outperformed DD 448kbps by a pretty substantial margin, and also outperformed (barely) DD+ 448 kbps.

Unfortunately, DD 640kbps was not tested. However, from some trends that you can identify throughout that data, DD640 should be very close to DTS 1.5M, maybe exceeds it, maybe not.

All performance measurements were done by ABX testing, by the way.

[sucker]

i have a 6.1 DTS track from a PAL DVD (25 fps) and want to sync it to a Blu-ray (23.976 fps), what are my options for lossless conversion?
suppose i have no special encoder installed, i bet my only option would be the eac3to slowdown and conversion to PCM/WAV?
if i would have the Surcode DTS encoder i would be limited to 5.1 iirc?
just for a later possible case for a 5.1 track, would the reencoding with the Surcode DTS Encoder result in a lossless conversion or not? (in relation to the already lossy compressed DTS source)

[ACrowley]

i have a 6.1 DTS track from a PAL DVD (25 fps) and want to sync it to a Blu-ray (23.976 fps), what are my options for lossless conversion?
suppose i have no special encoder installed, i bet my only option would be the eac3to slowdown and conversion to PCM/WAV?
if i would have the Surcode DTS encoder i would be limited to 5.1 iirc?
just for a later possible case for a 5.1 track, would the reencoding with the Surcode DTS Encoder result in a lossless conversion or not? (in relation to the already lossy compressed DTS source)

lossless conversion is impossible. Reencoding (timestretch/Resampling+reencoding in your case) is always lossy (either you use lossless Formats like FLAC )

eac3to.exe yoursourcedts.dts output.wavs -slowdown -r8brain

You will get 7 mono waves (when its dts es 6.1 discrete, but only 5.1 output when its dts es matrix)

That will provide a "very" conversion from DTS ES 6.1 25fps to 23.976Fps

Also i wouldnt use Surcodode DTS Encoder. Its a low quality and outdated encoder. And you cant encode dts es 6.1 with surcode

DTS Pro Series is a certified Reference Encoder from DTS labs and outputs perfect quality.

But you can remux your Video to 25FPS lossless with MKVmerge. Then you can use your untouched 25fps DTS Track without any Conversion.
But ist not possible to timestrech Audio without reencoding.

[sucker]

btw how do i get the 6.1 discrete (according to the packaging) track off the DVD? DVD Audio Ripper only shows 5.1 (direct demux option)
what would the -r8brain option in eac3to do, i believe it´s a better quality conversion or something like that, if yes what´s the difference to the standard eac3to conversion method? and what´s a "very" conversion? i´m still learning sorry for going on your nerves ;)
can i use the Wav64 or PCM option in combination with r8brain to remux later with tsmuxer? tsmuxer can´t handle mono wavs afaik?

[jj666]

Also i wouldnt use Surcodode DTS Encoder. Its a low quality and outdated encoder.

Given the price tag for Surcode, is there any proof to that claim?

Cheers,

-jj-

[Blue_MiSfit]

I tend to think that pricetag is invesely proportional with quality :)

Look at Aften versus Apple's AC3 encoder, or a $30k Dolby rackmount encoding server...

~MiSfit

[Blue_MiSfit]

Here's a feature request that I'd absofreakinglutely _LOVE_ to see implemented:

QuickTime support :)

I deal with ProRes + PCM sources a lot, and currently use an AviSynth + SoundOut workflow to handle them.

It would be very nice to just do this:

eac3to movie.mov 2:audio.ac3 -384 -0,1,2,3,4,5

(to grab the first 6 channels and toss the rest)

~MiSfit

[shon3i]

It outperformed DD 448kbps by a pretty substantial marginWell both codecs are in EXCELENT range which means they have tied scores, DTS have bettere averge score but not by substantial margin as you said, they are very close, on some samples DTS show better scores, on somes DD show's. Since DD 640 not tested i tested myself and prove my expectations. Like i said earlier i run deep ABX, and comapre every channel

Many of the test codecs are Dolby Digital Plus, not Dolby Digital.I know what i am looking for ;)

[SomeJoe]

Well both codecs are in EXCELENT range which means they have tied scores, DTS have bettere averge score but not by substantial margin as you said, they are very close, on some samples DTS show better scores, on somes DD show's. Since DD 640 not tested i tested myself and prove my expectations. Like i said earlier i run deep ABX, and comapre every channel


Do you feel from your own personal experience and ABX testing that you would choose DD 640kbps encoded by eac3to.exe (uses libAften.dll for this conversion, I believe) over DTS 1.5 Mbps encoded by SurCode, given that the source for each encoder would be the same?

[zn]

Nero AAC Encoder could not be located
http://www.nero.com/eng/nero-aac-codec.html


URL changed, unfortunately they dont make 301 redirect:

http://www.nero.com/eng/technologies-aac-codec.html

[turbojet]

Any idea why eac3to doesn't detect any of these AVCHHD/BD images from TSMuxer (http://www.sendspace.com/file/81wpv2) as bluray?

I also experience this when splitting with TSMuxer but full movies seem ok, does eac3to have a threshold on running time?

[tebasuna51]

...choose DD 640kbps encoded by eac3to.exe (uses libavcodec.dll for this conversion, I believe)...

Eac3to uses libAften.dll to encode to ac3.

[SomeJoe]

Eac3to uses libAften.dll to encode to ac3.

Noted, post edited, thanks.

[BlackJack1]

About year ago I dmuxed TRAINING DAY HDDVD using eac3to to mkv.
Here is the log:
eac3to v2.46
command line: "C:\Program Files\eac3to\eac3to.exe" "N:\HD\New z Shell'a\Training.Day.HD.DVD.1080p.VC-1.DDPlus.5.1\HVDVD_TS\FEATURE_1.EVO"+"N:\HD\New z Shell'a\Training.Day.HD.DVD.1080p.VC-1.DDPlus.5.1\HVDVD_TS\FEATURE_2.EVO" 2: m:\training\chapters.txt 3: m:\training\training2.mkv 5: m:\training\training2.dts -1536
------------------------------------------------------------------------------
EVO, 1 video track, 5 audio tracks, 4 subtitle tracks, 2:02:02
"Feature Presentation"
1: Joined EVO file
2: Chapters, 28 chapters with names
3: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
4: E-AC3, English, 5.1 channels, 640kbit/s, 48khz, dialnorm: -27dB, -13ms
5: TrueHD, English, 5.1 channels, 48khz, dialnorm: -27dB, -1ms
6: E-AC3, French, 5.1 channels, 640kbit/s, 48khz, dialnorm: -27dB, -13ms
7: E-AC3, Spanish, 2.0 channels, 192kbit/s, 48khz, dialnorm: -27dB, -29ms
8: E-AC3, English, 2.0 channels, 192kbit/s, 48khz, dialnorm: -27dB, -29ms
"Commentary 1"
9: Subtitle, English
10: Subtitle, English, "SDH"
11: Subtitle, French
12: Subtitle, Polish
Creating file "m:\training\chapters.txt"...
Audio gap description file detected, will be used for processing...
[v03] Extracting video track number 3...
[v03] Muxing video to Matroska...
[a05] Extracting audio track number 5...
[a05] Removing dialog normalization...
[a05] Applying RAW/PCM delay...
[a05] Writing WAVs...
[a05] Realizing RAW/PCM gaps...
[a05] Creating file "m:\training\training2.L.wav"...
[a05] Creating file "m:\training\training2.SR.wav"...
[a05] Creating file "m:\training\training2.C.wav"...
[a05] Creating file "m:\training\training2.LFE.wav"...
[a05] Creating file "m:\training\training2.R.wav"...
[a05] Creating file "m:\training\training2.SL.wav"...
Found Surcode DTS Encoder version 1.0.23.0.
Surcode says/asks: "Destination file m:\training\training2.dts exists. Overwrite?".
Surcode encoding successfully started. Please wait...
Closing Surcode...
Added fps value to MKV header.
Video track 3 contains 175570 frames.
eac3to processing took 11 minutes, 30 seconds.
Surcode encoding took 16 minutes, 57 seconds.
Done.
Now tried to make Blu-ray disc using newest TsMuxer accepting mkv.
I'm suprised that Mediainfo recognised VC1 video in mkv container as 29.97fps (!)
The same TsMuxer - 1080i 29.97 (pulldown)...

After TsMuxer finished the job BDinfo showing:
QUICK SUMMARY:

Disc Title: Training.Day.BD.from.HD.DVD
Disc Size: 17*336*030*328 bytes
Protection: AACS
BD-Java: No
Playlist: 00000.MPLS
Size: 0 bytes
Length: 2:02:02
Total Bitrate: 0,00 Mbps
Video: VC-1 Video / 0 kbps / 1080i / 29,970 fps / 16:9 / Advanced Profile 3
Audio: English / DTS Audio / 5.1 / 48 kHz / 1509 kbps / 24-bit
Subtitle: Polish / 0,000 kbps

Why eac3to changed oryginal video VC1 from 1080p 23.976fps to 1080i 29.97fps?

[jj666]

You didn't strip the pulldown flags from the stream, hence it shows 29.97fps, this is the format on the HDDVD disk. You can take them out using either EAC3TO (it does it by default these days) or TSMUXER (tick the remove pulldown box). The stream will then show as 23.976fps.

Cheers,

-jj-

[BlackJack1]

Thx for help. :)

[kurt]

feature request: support for chapters in mkv while demuxing :)

[shadowolf64]

Is it possible to convert a 6 channel aac file to a 2 channel aac file with eac3to? And if so how? I couldn't find anything that looked like it could do that but I'm not very proficient with eac3to or audio files for that matter.

[TinTime]

Yes, sort of:

eac3to input.aac output.m4a -down2

The output is aac in m4a, not raw aac. I think you need the Nero directshow audio decoder to decode the input though. If not you'll need to decode your aac first to wav (not sure how) and do the same:

eac3to input.wav output.m4a -down2

[shadowolf64]

Ah thank you for the quick response. This is exactly what I was looking for, and m4a is fine I just need it to be in the aac format.

[umaximus]

Hi madshi, been awhile since I last posted here, but still using eac3to everyday :)

Anyway, have a question;

What is the difference if I just demux video stream with eac3to and mux it in mkv with mkvmerge instead of mux it directly into mkv with eac3to? Do stream/mkv header loose some important data?

Im asking this, as I found out that some AVC titles muxed in mkv with eac3to are causing small stutter on PCH but if demux this mkv with eac3to and mux it back with mkvmerge (inputing elementary raw video stream) then the stutter is gone. But since its double the work with this, am I safe to just demux the streams at first place?

btw, this is the link from nmt forum regarding this PCH problem;
http://www.networkedmediatank.com/showthread.php?tid=10602

[jamos]

Madshi I know your busy with the other renderer project. if you can release source code for eac3to that would be most excellent if you no longer have the time..

[sidekick2]

My older version of arcsoft is bugging me constantly about updating to version 3. I assume that the new version 3 stuff won't work with eac3to, which is why I'm holding off on upgrading. Are there some files to hold onto that I can copy elsewhere for eac3to to use? Which file does eac3to need in order to work properly?

Or, has anyone tried upgrading arcsoft to version 3 and had eac3to still work?

I really don't want to lose my eac3to tools!

Thanks!

[Greif]

Madshi I know your busy with the other renderer project. if you can release source code for eac3to that would be most excellent if you no longer have the time..

Is there a bug that you are experiencing or a feature you would like to see implemented? If so, post it here and I'm sure madshi will work on it in due time.

[xkodi]

Or, has anyone tried upgrading arcsoft to version 3 and had eac3to still work?

I really don't want to lose my eac3to tools!

Thanks!

i haven't tried, but DTS-HD decoded DLL in version 3 is exactly the same version as in version 2, so i guess eac3to will work .

[ACrowley]

i haven't tried, but DTS-HD decoded DLL in version 3 is exactly the same version as in version 2, so i guess eac3to will work .

eac3to( or Dshow Apps/Players) wont work with TMT v3

1. Arcsoft TMT 3 doesnt install Filters/Decoders in C:\Programme(x86)\CommonFiles and theyre not registered from installer in Dshow.
2. Its not possible to reg them manually .You get a Error Message from reg32. It looks like that we cant use the Filters from TMT v3 outside TMT.
However, the Decoder Versions are not updated so i dont care

Workaround :
-Install Arcsoft TMT 3 as usual and use it.
-Install Arsoft TMT 2.25 Standalone Decoder Package to use the Decoders in eac3to and other Apps.
It works in eac3to and TMT 3 use its on Filters from installation Folder

@umaximus
I cannot reproduce your Problem. All Files are working fine on my NMT A110.
Remuxed with eac3to and/or MKVmerge with all types of Audio (including DTSHD,TrueHD etc)

[sidekick2]

ACrowley,

I'm not quite understanding that 2nd part... Are you saying to install the old version of Arcsoft TMT 2.25 after installing 3.0? Or are you saying to download a specific decoder package from somewhere? If so, where? I've looked on Arcsoft's site, and googled, and I'm not finding a decoder only package?!?

Hmm.. looking in my C:\Program Files\ArcSoft\TotalMedia Extreme\Digital Theatre\Codec I see the files in there. Should I just save a copy of that folder somewhere, or what?


Thanks!

eac3to( or Dshow Apps/Players) wont work with TMT v3

Workaround :
-Install Arcsoft TMT 3 as usual and use it.
-Install Arsoft TMT 2.25 Standalone Decoder Package to use the Decoders in eac3to and other Apps.
It works in eac3to and TMT 3 use its on Filters from installation Folder

[rica]

Hi guys,

What program do you suggest me to clean the audio files ?

[Thunderbolt8]

for ac3 files? delaycut 1.3

[jj666]

Hi Madshi,

I'm remuxing the main movie from Blu-ray disks and keeping all DTS-HD audio files intact. Would there be any way you would consider to make a note of the audio bitrate when demuxing as there is no easy way to ascertain the bitrate on VBR DTS-HD files (which 99% seem to be) as Mediainfo is not geared up to handle that type of file. I did already ask Zenitram about adding, I think on his list of "to-do's" for the far future as it would mean Mediainfo needs to scan the entire file which is not possible at the moment. I'm forced to use BDInfo at the moment to scan the entire disk, which can be time consuming.

Cheers,

-jj-

[tebasuna51]

...Would there be any way you would consider to make a note of the audio bitrate when demuxing as there is no easy way to ascertain the bitrate on VBR DTS-HD files ...
Is easy to know the average bitrate if you have the DTS-HD demuxed:

bitrate = filesize_in_bits / duration_in_seconds

[jj666]

Thank you for your insight tebasuna51.

I'm just testing a file I have with the following information:

DTS-HD MA:
1.36 gigabytes = (11408506.88 kilobits) calculator (http://www.matisse.net/bitcalc/?input_amount=1.36&input_units=gigabytes&notation=legacy)
1 hour 37 minutes 26 seconds (5846 seconds)
= bitrate 1951.50 kbps

compares adequately to the mentioned av. bitrate of 1991 kbps in BDINFO after full scan :-).

Thanks again, this saves me a lot of time!

Cheers,

-jj-

[stinman]

Malwarebytes' Anti-Malware 1.36
Database version: 2051
Windows 6.0.6001 Service Pack 1

4/27/2009 8:28:17 PM
mbam-log-2009-04-27 (20-28-13).txt

Scan type: Quick Scan
Objects scanned: 69890
Time elapsed: 2 minute(s), 19 second(s)

Memory Processes Infected: 0
Memory Modules Infected: 0
Registry Keys Infected: 0
Registry Values Infected: 0
Registry Data Items Infected: 0
Folders Infected: 0
Files Infected: 4

Memory Processes Infected:
(No malicious items detected)

Memory Modules Infected:
(No malicious items detected)

Registry Keys Infected:
(No malicious items detected)

Registry Values Infected:
(No malicious items detected)

Registry Data Items Infected:
(No malicious items detected)

Folders Infected:
(No malicious items detected)

Files Infected:
C:\Program Files\avcodec.dll (Spyware.OnlineGames) -> No action taken.
C:\Program Files\avutil-49.dll (Spyware.OnlineGames) -> No action taken.
C:\Program Files\libaften.dll (Spyware.OnlineGames) -> No action taken.
C:\Program Files\libFLAC.dll (Spyware.OnlineGames) -> No action taken.

This is the log file,should I send it to malwarebytes or could it of got contaminated.I sure do not believe these filters are viruses.I do not how how they got spyware online gaming.I think it is from eac3to or megui.

Mark

[rack04]

Can anyone tell me if the following extensions are correct for extracting audio from Blu-ray/HD DVD?

if "%INPUT_AUDIO_TYPE%" EQU "ac3" goto ac3
if "%INPUT_AUDIO_TYPE%" EQU "eac3" goto eac3
if "%INPUT_AUDIO_TYPE%" EQU "thd" goto thd
if "%INPUT_AUDIO_TYPE%" EQU "dts" goto dts
if "%INPUT_AUDIO_TYPE%" EQU "dtshd" goto dtshd
if "%INPUT_AUDIO_TYPE%" EQU "pcm" goto pcm

:ac3
Echo.
Echo.
Echo.
@Echo on
"%eac3to_PATH%" "%INPUT_DIRECTORY%" "%MAIN_MOVIE%") %AUDIO_TRACK%: "%WORKING_DIRECTORY%\%OUTPUT_FILENAME%.ac3"
@Echo off
goto subtitle

:eac3
Echo.
Echo.
Echo.
@Echo on
"%eac3to_PATH%" "%INPUT_DIRECTORY%" "%MAIN_MOVIE%") %AUDIO_TRACK%: "%WORKING_DIRECTORY%\%OUTPUT_FILENAME%.eac3"
@Echo off
goto subtitle

:thd
Echo.
Echo.
Echo.
@Echo on
"%eac3to_PATH%" "%INPUT_DIRECTORY%" "%MAIN_MOVIE%") %AUDIO_TRACK%: "%WORKING_DIRECTORY%\%OUTPUT_FILENAME%.thd+ac3"
@Echo off
goto subtitle

:dts
Echo.
Echo.
Echo.
@Echo on
"%eac3to_PATH%" "%INPUT_DIRECTORY%" "%MAIN_MOVIE%") %AUDIO_TRACK%: "%WORKING_DIRECTORY%\%OUTPUT_FILENAME%.dts"
@Echo off
goto subtitle

:dtshd
Echo.
Echo.
Echo.
@Echo on
"%eac3to_PATH%" "%INPUT_DIRECTORY%" "%MAIN_MOVIE%") %AUDIO_TRACK%: "%WORKING_DIRECTORY%\%OUTPUT_FILENAME%.dtshd"
@Echo off
goto subtitle

:pcm
Echo.
Echo.
Echo.
@Echo on
"%eac3to_PATH%" "%INPUT_DIRECTORY%" "%MAIN_MOVIE%") %AUDIO_TRACK%: "%WORKING_DIRECTORY%\%OUTPUT_FILENAME%.w64"
@Echo off
goto subtitle

[mrr19121970]

AC3
E-AC3
DTS
DTS Master Audio
DTS Hi-Res
DTS-HD (no core)
DTS Express
FLAC
RAW/PCM
MLP
TrueHD
TrueHD/AC3
WAV
W64RF64
DTSWAV
AC3WAV
AAC
MP1
MP2
MP3
Unknown audio track

these are all the possible ones.

[rfic]

i use eac3to to convert dts fps from 25 to 23.976

i have SurCode DVD DTS 1.0.21; but eac3to reports SurCode can't read .wave file whatever, then exit; the worst thing is it delete all generated .wave files and i can't even bebug what's wrong there

can i somehow at least keep those .wave files and use Surcode directly?

thanks

and here's log file

===================

eac3to v3.15
command line: eac3to.exe JACKY.track_4132.dts Jacky.23.976.dts -25.000 -changeTo23.976 -resampleTo48000
------------------------------------------------------------------------------
DTS Master Audio, 5.1 channels, 24 bits, 96khz
(core: DTS-96/24, 5.1 channels, 24 bits, 1509kbps, 96khz)
Decoding with ArcSoft DTS Decoder...
Changing FPS from 25.000 to 23.976...
Resampling to 48khz...
Reducing depth from 64 to 32 bits...
Writing WAVs...
Creating file "Jacky.23.976.SL.wav"...
Creating file "Jacky.23.976.LFE.wav"...
Creating file "Jacky.23.976.SR.wav"...
Creating file "Jacky.23.976.R.wav"...
Creating file "Jacky.23.976.C.wav"...
Creating file "Jacky.23.976.L.wav"...
Clipping detected, a 2nd pass will be necessary. <WARNING>
The original audio track has a constant bit depth of 24 bits.
The processed audio track has a constant bit depth of 32 bits.
Caution: The WAV file is bigger than 2GB. <WARNING>
Some WAV readers might not be able to handle this file correctly. <WARNING>
Starting 2nd pass...
Decoding with ArcSoft DTS Decoder...
Changing FPS from 25.000 to 23.976...
Resampling to 48khz...
Reducing depth from 64 to 32 bits...
Writing WAVs...
Creating file "Jacky.23.976.L.wav"...
Creating file "Jacky.23.976.R.wav"...
Creating file "Jacky.23.976.C.wav"...
Creating file "Jacky.23.976.SR.wav"...
Creating file "Jacky.23.976.LFE.wav"...
Creating file "Jacky.23.976.SL.wav"...
The processed audio track has a constant bit depth of 32 bits.
Caution: The WAV file is bigger than 2GB. <WARNING>
Some WAV readers might not be able to handle this file correctly. <WARNING>
Encoding DTS <1536kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.21.0.
Surcode says/asks: "At least one valid source file must be specified to encode.".
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...

[hubblec4]

hello

you can try this one:
command line: eac3to.exe JACKY.track_4132.dts Jacky.23.976.wavs -25.000 -changeTo23.976 -resampleTo48000

and then start SurCode load the files manually and convert them.

hubble

[Joniii]

I'm demuxing E-AC3 track from HD DVD, eac3to says remapping channels. Is there something wrong, I thought E-AC3 tracks from HD DVD shouldn't be remapped?

[rfic]

hello

you can try this one:
command line: eac3to.exe JACKY.track_4132.dts Jacky.23.976.wavs -25.000 -changeTo23.976 -resampleTo48000

and then start SurCode load the files manually and convert them.

hubble

thanks, hubble

it works -- i can get all 6 channel of .wav

and after i use Surcode to synth 6 channel of .wav to .dts, it finish successfully; but somehow, when i try to add this .dts to Tsmuxer, it doesn't recognize

anything i need to do?

thanks

[hoju3508]

I have the same issue with Tsmuxer and Ripbot264 not recognizing my .dts file. I just saw this in another post:

http://forum.doom9.org/showthread.php?p=1271882#post1271882

He ran the .dts file through eac3to again and it's working for him. I'm going to try this myself.

thanks, hubble

it works -- i can get all 6 channel of .wav

and after i use Surcode to synth 6 channel of .wav to .dts, it finish successfully; but somehow, when i try to add this .dts to Tsmuxer, it doesn't recognize

anything i need to do?

thanks

[ggking7]

The latest eac3to breaks Linux support via wine for me. I now get:

$ wine ~/eac3to/eac3to video "1)" 2: video.vc1 3: audio.ac3
Error reading file "video\BDMV\STREAM\00004.m2ts".

This same command worked perfectly on a previous version of eac3to (can't remember which one). I think it's because eac3to uses \ instead of / characters for the path. Does anyone know how to work around this?

[tebasuna51]

and after i use Surcode to synth 6 channel of .wav to .dts, it finish successfully; but somehow, when i try to add this .dts to Tsmuxer, it doesn't recognize


When you encode with Surcode use the .cpt (compact) format instead the .dts (padded) format. After rename to dts.

[hoju3508]

I see what you mean. I ran it through eac3to again and it removed the padding. Now it works with TsMuxer.

Thanks,

-Hoju


eac3to v3.15
command line: ..\..\eac3to\eac3to.exe wdky.dts wdky_new.dts
------------------------------------------------------------------------------
DTS (zero padded), 5.1 channels, 1:40:04, 24 bits, 1510kbps, 48khz
Removing DTS zero padding...
Creating file "wdky_new.dts"...
eac3to processing took 2 minutes, 8 seconds.
Done.




When you encode with Surcode use the .cpt (compact) format instead the .dts (padded) format. After rename to dts.

[Someone_123]

I have a problem. I load the track in eac3to and select to decode it, but nothing happens... and I wait, and wait. I tried everyting: changed the version of the program with newer, reinstalled the codecs, but the problem still is the same. Usually when i load the track in the program and select to decode it, it opens CMD for a second or two and makes an analyse, then i can do my job.
What's the problem? And how can I fix it?

[tebasuna51]

@Someone_123
Put here your command line and log (if any).

[Someone_123]

I load the track in eac3to and select to decode it, but nothing happens... the program just minimizes and there is no CMD or sh*t.... eac3to just stays minimized... :(
Where's the problem?!?

[tebasuna51]

I load the track in eac3to and select to decode it...
How make you this?

[rfic]

I see what you mean. I ran it through eac3to again and it removed the padding. Now it works with TsMuxer.

Thanks,

-Hoju

me too here, thanks guys for help

i use remuxer and it works

[Someone_123]

How make you this?

First I extract my audio track. For exmaple TrueHD (*.thd) track. I open the eac3to GUI, which has all of its plugins like SurCode DTS. Then I click on "Add Source File(s)" and input the *.thd audio track. After that I go to the next tab "Eac3to Audio" and in the upper right I choose the "Input File(s) or Folder(s)". (I just select the only file that I added earlier)
The program usually runs CMD to analyse the source file and after the analyse I can select my options and docode the audio without any problem. But this time the program minimizes and CMD doesn't pop up like usual.
I runned CMD to see if it works - it works. I changed the codecs, changed the eac3to with newer but the problem still is the same and i can't decode my TrueHD tracks to DTS like I did before. I'm doing everything like before.
Can you tell me what could be the problem? How can i fix it?

P.S: Sorry for my English. I'm from Bulgaria.

[tebasuna51]

First I extract my audio track. For exmaple TrueHD (*.thd) track.
How?

I open the eac3to GUI, which has all of its plugins like SurCode DTS. Then I click on "Add Source File(s)" and input the *.thd audio track. After that I go to the next tab "Eac3to Audio" and in the upper right I choose the "Input File(s) or Folder(s)". (I just select the only file that I added earlier)
Then you use yr_eac3to_more_gui.
I think is a GUI problem, if your source is from BD, the thd identifier TrueHD/AC3 (generated by eac3to) maybe is not recognized by yr_eac3to_more_gui.

Try using another GUI

[Someone_123]

Thank you for the help! I downloaded this GUI http://www.digital-digest.com/software/download.php?sid=1519&ssid=1&did=3 . Which isn't better then the last one and I have to write code to decode audio but the most important thing is that I can do my job right. :)

[wolfbane5]

So I've used eac3to several times now to rip my bluray movies to the comp and have never encountered a problem. I was ripping the firefly bluray to my comp the other day and every episode i ripped, eac3to said:

eac3to v3.15
command line: eac3to h:\ 2) 2: d:\e3\e3.mkv 3: d:\e3\e3dts.dts -core
------------------------------------------------------------------------------
M2TS, 1 video track, 5 audio tracks, 6 subtitle tracks, 0:43:59, 24p /1.001
1: Chapters, 12 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz, -8ms
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a03] Extracting DTS core...
[a03] Applying DTS delay...
[a03] A remaining delay of +2ms could not be fixed.
[v02] Muxing video to Matroska...
[a03] Creating file "d:\e3\e3dts.dts"...
Added fps value to MKV header.
Video track 2 contains 63262 frames.
eac3to processing took 21 minutes, 53 seconds.
Done.

The delay that could not be fixed varied with each episode from -2ms to +3ms. I didn't think much of it until I went and encoded it with RipBot264 and noticed that the audio went further and further out of sync as the episode progressed. I didn't think it was a RipBot264 problem since I've ripped blurays and encoded them to 720p (mkv) with RipBot264 previously and have never had a problem.

If someone could help me out it would be appreciated. If it is indeed a RipBot264 problem, please let me know and I'll post this in the appropriate forum.

[madshi]

feature request: support for chapters in mkv while demuxing :)
Is already on the to do list.

What is the difference if I just demux video stream with eac3to and mux it in mkv with mkvmerge instead of mux it directly into mkv with eac3to? Do stream/mkv header loose some important data?
When using eac3to for MKV video muxing, you can later demux again (by using eac3to) and get the same data back without any missing information. If you use mkvtoolnix for muxing, depending on the codec you can lose information. Also eac3to sometimes produces better timecodes than mkvtoolnix.

Im asking this, as I found out that some AVC titles muxed in mkv with eac3to are causing small stutter on PCH but if demux this mkv with eac3to and mux it back with mkvmerge (inputing elementary raw video stream) then the stutter is gone.
Strange. Never heard of such a problem yet. Is it always the same place in the movie where the stuttering occurs? Does it also occur when playing back on a HTPC?

Malwarebytes' Anti-Malware 1.36
That's clearly a false alarm. Could you please report that to the AV writers? Thanks!

I'm demuxing E-AC3 track from HD DVD, eac3to says remapping channels. Is there something wrong, I thought E-AC3 tracks from HD DVD shouldn't be remapped?
When demuxing there should be no channel remapping going on. When demuxing + decoding, channels must eventually be remapped.

So I've used eac3to several times now to rip my bluray movies to the comp and have never encountered a problem. I was ripping the firefly bluray to my comp the other day and every episode i ripped, eac3to said:

"a03 Extracting DTS core...
a03 Applying DTS delay...
a03 A remaining delay of +2ms could not be fixed."

The delay that could not be fixed varied with each episode from -2ms to +3ms. I didn't think much of it until I went and encoded it with RipBot264 and noticed that the audio went further and further out of sync as the episode progressed. I didn't think it was a RipBot264 problem since I've ripped blurays and encoded them to 720p (mkv) with RipBot264 previously and have never had a problem.

If someone could help me out it would be appreciated. If it is indeed a RipBot264 problem, please let me know and I'll post this in the appropriate forum.
Those "a remaining delay of" messages have nothing to do with the problem you're reporting. FWIW, I've remuxed my Firefly Blu-Rays to MKV by using eac3to and they play perfectly fine for me. So I can only suspect that it's a RipBot264 problem.

I cannot get EAC3TO to write to 6tb raid arraywith 4k block size(to defeat>2tb limit in windows XP) .

Areca 1130 controller.

Is this a known issue?
No, never heard of such a problem yet.

It's probably not the best place to ask, but what am I supposed to do with 23,975 mkvs?
It depends on which framerate is 23.975 and which is maybe not. There is a framerate written in the video bitstream header structure. The container also has a framerate field. And then there are the container timestamps which can again have a different framerate. So basically there are 3 things which can be 23.975 or 23.976 or something else, and all 3 can be different. So there isn't really a simple answer to your question.

since eac3to (ffmpeg) is unable to decode correctly aac SBR and so want use nero decoder, would be possible to add support for Monogram AAC decoder (from radlight) ?
it's free/gpl
Every decoder I support costs quite a lot of time for me to implement. Because of that I'd prefer to wait until libav has added support for SBR etc. Also, I can't ship a GPL decoder with eac3to due to license restrictions.

I am having issues with w64 madshi. same as tabasuna51 is. also when I use a w64 in tsmuxer it will give a error after a minute or so. Roman tried a large w64 that he created then used tsmuxer and it works fine for him. He suggested it may be a header error. This only happens using DTS-ma converted to W64. true-hd to W64 does not give a error but the w64 file seems small.
Is it correct to have a w64 file be less size than the original true-hd file?

thats what I am getting when I convert a true-hd file to w64 using eac3to. When I convert the true-hd file to pcm it is 7 gig. compared to the original 2 gig. true-hd file (which seems correct).
The w64 file should have virtually the same size as the wav file. Try writing to wav. Does that produce a small or big file? Make sure your harddisk has enough free space.

There is a bug in eac3to, but I don't really think that it results in incorrectly written w64 files. I rather guess that maybe your harddisk was full?

"Bitstream parsing for track 3 failed" msg reported when running eac3to (3.15) on an mkvtoolnix/mkvmerge(2.6.0)-created .mkv file for a muxed VobSub track (3)
IIRC VobSub is not fully supported by eac3to yet.

Here's a feature request that I'd absofreakinglutely _LOVE_ to see implemented:

QuickTime support :)

I deal with ProRes + PCM sources a lot, and currently use an AviSynth + SoundOut workflow to handle them.

It would be very nice to just do this:

eac3to movie.mov 2:audio.ac3 -384 -0,1,2,3,4,5

(to grab the first 6 channels and toss the rest)
It's a possibility, but obviously it would cost quite a bit of time to implement. So if it ever comes, it will take quite a while...

Some rare CDs have pre-emphasis. Extracting them in secure mode with EAC leaves the file emphasised. eac3to could have a switch to
get proper gain figure at high frequencies when processing the wavs.

WaveEmph110 does the job, but a switch in eac3to could be a nice feature, although rare.
Analyzing high frequencies etc is too scientific for me. I don't have enough knowledge to implement such features...

any chance you could put the most recent version of libFLAC.dll (http://forum.doom9.org/showpost.php?p=1266310&postcount=8628) in your package at next release? Just for the sake of having up-to-date libraries included with eac3to.
Is there any practical reason for updating? IIRC someone recently tried the newest libFLAC.dll and the FLAC files produced with the new dll were bit by bit identical to the ones produced with the dll shipping with eac3to. Updating to a newer dll could break something. So I don't really want to do that, unless there is a real benefit.

could you add support for avs (audio) files ? as wavi ?

would be handy for using a unique tool for audio encoding (i use eac3to for decoding AND piping to nero/ogg/lame...)
I've never done any AVS programming, so that's a bit over my head. Would have to read into the whole topic first. And the main problem is that eac3to is used to have full control over the input file (seeking, reading etc). Not sure how easy that would be to do with AVS input. That's also a reason why stdin is currently not supported by eac3to.

What bit depth the original track has? 24 or 16?
Whenever Nero decodes to 24bit and libav decodes to 16bit, you can usually trust libav and discard Nero. Nero likes to apply dialnorm, which mutates a 16bit track into 24bit. Usually eac3to is able to convince Nero to not apply dialnorm. Maybe in your case that failed. I'd like to get a small sample of the TrueHD track, so that I can try to find out why Nero outputs 24bit.

I'm trying to rip Total Recall from HD-DVD and I received a gaps file and have huge sync problems. I have run it again and the log files says it uses the gap file but there is still over three seconds differences between picture and the sound. Is is possible to fix the sync?
Strange. I once ripped that HD DVD and had no problems. I've since sold the HD DVD, unfortunately, so I can't double check. Can anybody else confirm the problem?

madshi, mkv files edited with the header editor in mkvmerge aren't recognized by eac3to.
Works for me (latest mkvtoolnix version).

When I use EAC3TO to extract from vob files, on about 1 DVD out of 20 the audio gap processing causes a lip sync error. (The vob files seem clean BTW, and only contain the PGC I am interested in). Re-processing (using the video gap) does not help. If I do the same extraction with version 2.80 using -ignoreDiscon there is no lipsync problem. I'd rather not have to keep version 2.80 around; I'd prefer it if the flag was put back.
Ok, will add a switch to disable 2pass processing.

URL changed, unfortunately they dont make 301 redirect
The URL still seems to work fine for me with the current eac3to version.

It's not just one specific encoder, but the encoders written by DTS and Dolby (both of them). There is a fixed standard on how to name the files in the DVD/Blu-Ray industry:
For 5.1:
L, R, C, LFE, Ls, Rs
For 6.1:
L, R, C, LFE, Ls, Rs, Cs
For 7.1:
L, R, C, LFE, Lss, Rss, Lsr, Rsr

All capitals are acceptable. These are the extensions that are printed on the labels of the tapes delivered from the studios to be encoded for Blu-Ray. These extensions are also used by the two professional audio encoding suites: DTS-HD Master Audio Suite Encoder and the Dolby Media Encoder.
Ok, these are good arguments.

But what about DTS speaker placements? There are many many different versions for 7.1 DTS tracks. In my 7.1 DTS-HD sample collection I have at least 4 different combinations of speakers. Surprsingly "L, R, C, LFE, Lss, Rss, Lsr, Rsr" seems to be the exception and not the norm. So that is a problem for me. If I switched to these names, which names should I use for 7.1?

His program is widely used in the professional industry
It is!? That is news to me! :eek:

[madshi]

I have found an unusual MPEG Program Stream that contains MPEG-2 video an AC3 audio, but eac3to doesn't recognize the audio stream
Thanks, will be fixed in the next build.

[madshi]

This file, created by eac3to, is not recognized by eac3to. (work fine with other softs)
Will be fixed in the next build.

[madshi]

eac3to v3.16 released

http://madshi.net/eac3to.zip

* added undocumented "-no2ndpass" switch to turn off 2nd pass processing
* fixed: two pass processing sometimes produced superfluous sup files
* fixed: MPG/EVO/VOB audio tracks with "PES extension 2" were not detected
* fixed: very small W64/RF64 files were not detected correctly
* fixed: when processing was aborted, log file was sometimes not created
* fixed: sometimes specifying a title number addressed the wrong HD DVD title

[shon3i]

Thanks madshi, where we can find list of that undocumented switches :)

[Thunderbolt8]

thanks!

[madshi]

where we can find list of that undocumented switches :)
In the changelog. Simply search for "undocumented".

[tebasuna51]

Thanks madshi.

[Chumbo]

Thanks for the update madshi.

[Snowknight26]

Works for me (latest mkvtoolnix version).

At the time of the posting, the latest version of mkvtoolnix made files unreadable to eac3to, but I'm glad it's not the case anymore.

[sidekick2]

Thanks again Madshi!

[mrr19121970]

eac3to v3.16 released

* fixed: when processing was aborted, log file was sometimes not created
* fixed: sometimes specifying a title number addressed the wrong HD DVD title

Thanks madshi !

[rik1138]

Ok, these are good arguments.

But what about DTS speaker placements? There are many many different versions for 7.1 DTS tracks. In my 7.1 DTS-HD sample collection I have at least 4 different combinations of speakers. Surprsingly "L, R, C, LFE, Lss, Rss, Lsr, Rsr" seems to be the exception and not the norm. So that is a problem for me. If I switched to these names, which names should I use for 7.1?

Like what other kind of speaker combinations? When audio is mixed for 7.1 for use on Blu-Ray movies, it's mixed with that 7.1 configuration. The speaker arrangement in the mixing bay looks like the speaker diagram seen here:

http://www.dts.com/Pro-Audio_Software/DTS-HD_Master_Audio_Suite/DTS-HD_Master_Audio_Suite_Encoder_Demo.aspx

And those are the labels used for the different channels...

Originally Posted by rik1138
His program is widely used in the professional industry
It is!? That is news to me! :eek:

Yep, it's one of the standard tools most people in the industry (compressionists and authors specifically) have heard of, and usually have handy in case it's needed. The industry doesn't seem to want to make tools to reverse the process of making DVDs/Blu-Rays, so if we have to go backwards for any reason, we rely on you guys. :cool: Doom9 is also widely recognized as the place to go if you have to do anything that the pro tools don't support... There always seems to someone here that's written a tool to do damn near anything someone can think of.

I've been using eac3to for well over a year now, originally to get audio from the HD DVD format converted to a DVD compatible format for a client, and more recently to convert lossless audio back to WAVs to it can be encoded in other formats (hence my desire for the naming to match what the encoding software expects...), and to strip out core streams which comes in handy a lot too. It's the best tool out there for converting all these formats.

TSMuxer is another popular tool... It's _SO_ much faster to run a video/audio stream through TSMuxer to get a playable Blu-Ray image than to try to do it in Scenarist. If you just need audio/video for QC playback or something, it can save a lot of time...

[ggking7]

Could I download an older version of eac3to? I'd like to go back to one that works via wine on Linux.

[Brazil2]

Could I download an older version of eac3to? I'd like to go back to one that works via wine on Linux.
Which version do you want ?

Check this page:
http://www.videohelp.com/tools/eac3to/old-versions#download

[Greif]

In the changelog. Simply search for "undocumented".

Then it wouldn't be undocumented :p

Thx for the update.

[StephenB]

* added undocumented "-no2ndpass" switch to turn off 2nd pass processing
* fixed: two pass processing sometimes produced superfluous sup files

:thanks:

[juGGaKNot]

Hi

Great tool, fun to use.

The AAC LC 2 ch is ps3/xbox compatible in mp4 no ? neroAacEnc.exe in the folder :

http://i286.photobucket.com/albums/ll105/juGGaKNot4cs/eac3to.jpg

"%mypath%\bin\eac3to\eac3to.exe" %mypath%\%mymovie%.wav %mypath%\bin\@temp\%mymovie%.m4a -quality=0.5

"%mypath%\bin\mkvtoolnix\mkvmerge.exe" --default-duration %mykfps% --default-language eng --aac-is-sbr 0 -o %mypath%\The_Final_Movie.mp4 "%mypath%\bin\@temp\%mymovie%.264" "%mypath%\bin\@temp\%mymovie%.m4a"

Or is this wrong ?

[dbknightx]

Nevermind - I was being stupid.... :)

DELETED

[mrr19121970]

@Madshi

Concerning your post here:

http://forum.doom9.org/showthread.php?p=1258562#post1258562

Quote:
Originally Posted by tvjunky
Would it be much work to automatically create a second file containing only the forced captions while extracting this stream?
I guess it would be possible, but I don't know how difficult it would be. I'll put it on my to do list.

do you have any ETA on when this might be available ?

Thanks.

[madshi]

do you have any ETA on when this might be available ?
I generally do not give out ETAs for my freeware. ;)

Currently my focus is on madVR, though, so new features for eac3to will have to wait...

[Octo-puss]

what is MadVR please? I wasn't reading the forums for quite some time.


All: is there any up to date GUI for eac3to?

[madshi]

what is MadVR please?
There is no MadVR, only madVR. Do a search and you'll find it easily.

All: is there any up to date GUI for eac3to?
All GUIs are listed on the first page. Two of them are still constantly updated.

[hubblec4]

eac3to v3.16 released

http://madshi.net/eac3to.zip


* fixed: two pass processing sometimes produced superfluous sup files


i demuxed a bluray and saw that the *.sups are generated without the number of captions (like this: 00129 - 14 - Subtitle (PGS), German, 1374 captions.sup). have you removed this feature?

Very Great Work!!


hubble

[ggking7]

I rolled back to 3.02 and it works on Linux via wine again. I'd be happy to test any changes to the current version for wine compatibility. I think it's just a problem with the \ character versus the / character.

Here is the error I get:

$ wine ~/eac3to/eac3to video "1)" 2: video.vc1 3: audio.ac3
Error reading file "video\BDMV\STREAM\00004.m2ts".

[Snowknight26]

I was demuxing my The Hitcher HD DVD when I saw that eac3to output:
a07 Audio has a gap of 157664ms at playtime 0:48:20.
I really hope that's it syncs perfectly. :x

[ggking7]

I'm trying to demux a series of unencrypted AOB files but I get:

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

or:

"HD DVD / Blu-Ray disc structure not found."

depending on how I execute eac3to. Can eac3to handle AOB? This link claims it can:

http://en.wikibooks.org/wiki/Eac3to/Features

[aleg]

Can eac3to handle AOB?

I use DVDA-Explorer to extract the mlps which I then convert using eac3to.
See this section of eac3to-thread http://forum.doom9.org/showthread.php?p=1102171#post1102171

[Mark_A_W]

Is there any way to use Eac3to to convert a DTS-WAV track to a 6 channel FLAC track?

I tried it, but it sees the wav file as a stereo wav, not a DTS wav.

[tebasuna51]

Is there any way to use Eac3to to convert a DTS-WAV track to a 6 channel FLAC track?

I tried it, but it sees the wav file as a stereo wav, not a DTS wav.

Please :logfile:

No problem here:
eac3to v3.16
command line: eac3to "D:\Corazon espinado.wav" "D:\Corazon espinado.flac"
------------------------------------------------------------------------------
DTSWAV, 5.1 channels, 0:04:36, 24 bits, 1235kbps, 44.1khz
Reading DTSWAV...
Decoding with ArcSoft DTS Decoder...
Encoding FLAC with libFlac...
Creating file "D:\Corazon espinado.flac"...
eac3to processing took 48 seconds.
Done.

Maybe eac3to don't recognize your wav like DTSWAV.
You can try also BeSplit:

D:\BeSpV09b7\BeSplit.exe -core( -input "your_dts.wav" -prefix "your_wav" -type dtswav -fix )
and encode after the wav to flac.

[Mark_A_W]

Please :logfile:

No problem here:
eac3to v3.16
command line: eac3to "D:\Corazon espinado.wav" "D:\Corazon espinado.flac"
------------------------------------------------------------------------------
DTSWAV, 5.1 channels, 0:04:36, 24 bits, 1235kbps, 44.1khz
Reading DTSWAV...
Decoding with ArcSoft DTS Decoder...
Encoding FLAC with libFlac...
Creating file "D:\Corazon espinado.flac"...
eac3to processing took 48 seconds.
Done.

Maybe eac3to don't recognize your wav like DTSWAV.
You can try also BeSplit:

D:\BeSpV09b7\BeSplit.exe -core( -input "your_dts.wav" -prefix "your_wav" -type dtswav -fix )
and encode after the wav to flac.

Interesting.

It no workee for me:

eac3to v3.16
command line: eac3to The_Run.wav The_Run.flac
------------------------------------------------------------------------------
WAV, 2.0 channels, 0:03:47, 16 bits, 1411kbps, 44.1khz
Reading WAV...
Encoding FLAC with libFlac...
Creating file "The_Run.flac"...
The original audio track has a constant bit depth of 16 bits.
eac3to processing took 7 seconds.
Done.


It's definitely a 5.1 channel DTS encoded wav. If I play it back normally I get noise, if I send it via SPDIF to my surround processor it goes into DTS mode, and it's 6 channels.

I tried a few, with the same result. I will try BeSplit.


Edit: Tried Besplit, it sees 2 channels too. Tebasuna can you try it too? You can download "The Run" here: http://www.sr.se/cgi-bin/mall/artikel.asp?ProgramID=2446&Artikel=740607



Thanks

Mark

[mrr19121970]

one guy is having trouble here:

http://forum.slysoft.com/showpost.php?p=196092&postcount=806


D:\eac3to>eac3to.exe h:\ 1) -demux
M2TS, 1 video track, 2 audio tracks, 1 subtitle track, 1:36:12, 24p /1.001
1: Chapters, 16 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: DTS Master Audio, German, 3.0 channels, 16 bits, 48khz
(core: DTS, 3.0 channels, 16 bits, 1509kbps, 48khz)
4: DTS Master Audio, English, 3.0 channels, 16 bits, 48khz
(core: DTS, 3.0 channels, 16 bits, 1509kbps, 48khz)
5: Subtitle (PGS), German
Creating file "00000 - Chapters.txt"...
s05 Extracting subtitle track number 5...
v02 Extracting video track number 2...
a03 Extracting audio track number 3...
a04 Extracting audio track number 4...
a04 Creating file "00000 - 4 - DTS Master Audio, English, 3.0 channels, 16 bits, 48khz.dtsma"...
a03 Creating file "00000 - 3 - DTS Master Audio, German, 3.0 channels, 16 bits, 48khz.dtsma"...
v02 Creating file "00000 - 2 - VC-1, 1080p24.vc1"...
s05 Creating file "00000 - 5 - Subtitle (PGS), German.sup"...



Then I tried the conversion from dtsma to ac3:

D:\eac3to>eac3to 3.dtsma 3.ac3
DTS Master Audio, 3.0 channels, 16 bits, 48khz
(core: DTS, 3.0 channels, 16 bits, 1509kbps, 48khz)
The ArcSoft decoder doesn't seem to work, will use Sonic instead.
Decoding with DirectShow (Sonic Audio Decoder)...
DirectShow reports 3.0 channels, 16 bits, 48khz
The AC3 encoder received a non-supported data format (pcm, 3, 16, -).
Aborted at file position 1190930124.


I've tried it with Arcsoft, Sonic and libav and alwasy get the same error.

is there any way to convert the 3 channel DTS-MA to AC3 ?

[Xorp]

Would be .mpls file reading be a reasonable feature to add? i.e. just load a mpls playlist file in eac3to and get the order the m2ts files will be played in, regardless if you have the rest of the disc structure or not. Would probably also be useful for GUIs versions.

[tebasuna51]

is there any way to convert the 3 channel DTS-MA to AC3 ?

Use the external encoder Aften with eac3to STDOUT:

eac3to 3.dtsma stdout.wav | Aften -b 320 - 3.ac3

[tebasuna51]

It no workee for me:...
"The Run" don´t work here with eac3to. There are many trailing '0' before a valid dts header and the automatic detection mode of eac3to don't work here.

Many other dtswav of this page work fine.
"The Firebird" work with eac3to.

Edit: Tried Besplit, it sees 2 channels too. Tebasuna can you try it too?
BTW, BeSplit work always, you need the last beta version:
BeSplit (v0.9b7) (http://schudy.de/dts/softarchiv-e.htm)

Tested and work with "The Run"

[Mark_A_W]

"The Run" don´t work here with eac3to. There are many trailing '0' before a valid dts header and the automatic detection mode of eac3to don't work here.

Many other dtswav of this page work fine.
"The Firebird" work with eac3to.


BTW, BeSplit work always, you need the last beta version:
BeSplit (v0.9b7) (http://schudy.de/dts/softarchiv-e.htm)

Tested and work with "The Run"


Umm...ahh little help with your working BeSplit command line please??

Thanks for testing The Run Tebasuna.

[Mark_A_W]

"The Run" don´t work here with eac3to. There are many trailing '0' before a valid dts header and the automatic detection mode of eac3to don't work here.

Many other dtswav of this page work fine.
"The Firebird" work with eac3to.


BTW, BeSplit work always, you need the last beta version:
BeSplit (v0.9b7) (http://schudy.de/dts/softarchiv-e.htm)

Tested and work with "The Run"


Yes, Firebird works with Eac3to.

It would be nice if we could force DTSWAV decoding when auto detection fails (Madshi? -dtswav switch?).

When I use dtswav as type in BeSplit, it crashes..

[tebasuna51]

Umm...ahh little help with your working BeSplit command line please??

With BeSplit.exe and Track01.wav (The Run) at same folder:

BeSplit -core( -input "Track01.wav" -prefix "prefix" -type dtswav -fix )

to obtain a prefix01.dtswav (rename to "The Run.dts")

BTW, I use a dts.bat file with:
for %%a in (*.wav) do D:\Programa\Audio\BeSp09b7\BeSplit.exe -core( -input "%%a" -prefix "%%a" -type dtswav -fix )
pause

to convert all .wav (dtswav) in a folder

[Mark_A_W]

Thanks Tebasuna

I now having a working bat file with:

for %%a in (*.wav) do BeSplit.exe -core( -input "%%a" -prefix "%%a" -type dtswav -fix )
for %%a in (*.dtswav) do eac3to.exe "%%a" "%%a".flac
pause

And it all just happens :)

[rebkell]

I have a question about audio gaps, I have the following:


[a02] Audio has a gap of 39ms at playtime 0:03:39. -> +32(-7)
[a02] Audio has a gap of 41ms at playtime 0:03:55. -> +32(-16)
[a02] Audio has a gap of 41ms at playtime 0:16:51. -> +32(-25)
[a02] Audio has a gap of 44ms at playtime 0:25:56. -> +32(-37)
[a02] Audio has a gap of 22ms at playtime 0:37:43. -> +32(-25)
[a02] Audio has a gap of 54ms at playtime 0:47:29. -> +32(-47)
[a02] Audio has a gap of 29ms at playtime 0:55:48. -> +32(-44)
[a02] Audio has a gap of 28ms at playtime 1:03:05. -> +32(-40)
[a02] Audio has a gap of 21ms at playtime 1:11:36. -> +32(-29)
[a02] Audio has a gap of 58ms at playtime 1:27:22. -> +32(-55)



I put in the red, just assuming you would pad 32ms, the number in parenthesis would be the running total left over in ms at each spot, just wondering how you would actually fill the gaps. The video is 23.976FPS and the audio is 48KHz AC3.

[Wakka]

I'm using eac3to alot, awesome program :cool:

But having a problem converting a TrueHD 7.1 into DTS-ES 6.1

If I want to convert the TrueHD 7.1 into .agm, it doesn't work...
"agm writer doesn't support this channel configuration"

Used agm output before, also for 7.1 into 6.1, but with this source it won't work :(

Tried the -7 switch, to make a wav file first, but it doesn't get remuxed into 6.1 channel format...

I want to remux the channels into 6.1 automatically,
and pass the .agm into DTS Pro encoder.

any ideas?

[TinTime]

I don't think eac3to does downmixing itself to 6.1 - it's either 5.1 or 7.1. The -7 switch is for the ArcSoft decoder, which isn't used for TrueHD.

One solution... If the DTS Pro encoder can take individual wavs as input you could create 6.1 wavs. To do this you'll need to decode the TrueHD track to 8 wavs using eac3to and then merge the two back channels with something like sox to end up with 7 wavs. Don't know if this is any use to you.

[Wakka]

I don't think eac3to does downmixing itself to 6.1 - it's either 5.1 or 7.1. The -7 switch is for the ArcSoft decoder, which isn't used for TrueHD.

One solution... If the DTS Pro encoder can take individual wavs as input you could create 6.1 wavs. To do this you'll need to decode the TrueHD track to 8 wavs using eac3to and then merge the two back channels with something like sox to end up with 7 wavs. Don't know if this is any use to you.

Ah, thats why it worked before with DTS-HD 7.1 (which only had a 5.1 core, hence the conversion)

Yeah, I'm already looking into the alternate method, but its time-consuming...and easy to make mistakes.
DTS Pro has a tool to merge seperate mono wavs into one .agm for DTS encodig.


Thanks for the help :)

[florinandrei]

Demuxing M2TS files produced by an HD camcorder: Canon HF100. The camcorder produces almost a BD structure on the flash (or is it a complete BD structure?).

Anyway, it seems like files shorter than 10 seconds or so can be demuxed just fine:

http://dl.getdropbox.com/u/29966/demux/00001.MTS

eac3to v3.16
command line: eac3to 00001.MTS 1: 00001.mkv 2: 00001.ac3 -seekToIFrames
------------------------------------------------------------------------------
M2TS, 1 video track, 1 audio track, 0:00:11, 60i /1.001
1: h264/AVC, 1080i60 /1.001 (16:9)
2: AC3, 2.0 channels, 256kbps, 48khz
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[v01] Muxing video to Matroska...
[a02] Creating file "00001.ac3"...
Added fps value to MKV header.
Video track 1 contains 658 fields.
eac3to processing took 1 second.
Done.

However, files longer than 20 sec or so cannot be demuxed - eac3to fails at some point.

http://dl.getdropbox.com/u/29966/demux/00002.MTS

eac3to v3.16
command line: eac3to 00002.MTS 1: 00002.mkv 2: 00002.ac3 -seekToIFrames
------------------------------------------------------------------------------
M2TS, 1 video track, 1 audio track, 0:00:21, 60i /1.001
1: h264/AVC, 1080i60 /1.001 (16:9)
2: AC3, 2.0 channels, 256kbps, 48khz
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[v01] Muxing video to Matroska...
Unfortunately the Haali Muxer cannot handle this source file. <ERROR>
It doesn't contain enough seek/recovery points. <ERROR>
The "-seekToIFrames" option may work around that.
However, using this option may result in artifacts when seeking.
Aborted at file position 35651584. <ERROR>

The position where it's aborted is not always exactly the same, but the values are pretty similar - always 30 ... 40 million.

What's going on? Does it fail after a GOP or something?

The files are valid, I can open them in various programs and they work very well.

[TinTime]

Ah, thats why it worked before with DTS-HD 7.1 (which only had a 5.1 core, hence the conversion)

Yeah, I'm already looking into the alternate method, but its time-consuming...and easy to make mistakes.
DTS Pro has a tool to merge seperate mono wavs into one .agm for DTS encodig.


Thanks for the help :)


This will merge two wavs with sox so you can automate the 7.1 TrueHD to 6.1 wavs stage at least, using eac3to and sox:

sox.exe --show-progress -V3 -m -v 0.5 "back left.wav" -v 0.5 "back right.wav" "back centre.wav"

[rica]

Hi madshi,
Is it possible to add Radscorpion's Monogram AAC Decoder support with the option of "-mmaac"?
Using this, i got the best results even with problematic aac files.
If you need any problematic aac _supposed to be- which eac3to can not figure out- i mean nero audio decoder2 or libav inside eac3to-, i can pm a sample to you.
Best.

[Wakka]

This will merge two wavs with sox so you can automate the 7.1 TrueHD to 6.1 wavs stage at least, using eac3to and sox:

sox.exe --show-progress -V3 -m -v 0.5 "back left.wav" -v 0.5 "back right.wav" "back centre.wav"

Thanks, already mixed it, but didn't half the volume.

Why should I half the volume, the mix should have the same vol as the source, when they get mixed?

[TinTime]

Argh, sorry, I copied the command line from a script where I use it for something else. As you say both the -v 0.5s should be -v 1.0 (and check for clipping). Sorry about that.

But you've done it now anyway :)

[tebasuna51]

Thanks, already mixed it, but didn't half the volume.

Why should I half the volume, the mix should have the same vol as the source, when they get mixed?
To avoid clip. The mix must be 0.5 + 0.5 = 1

[Blackwalker]

hi, can i convert a video with VC-1 in to h.264 using eac3to?
if yes, which is the command to do this?
thx! :)

[buzzqw]

hi, can i convert a video with VC-1 in to h.264 using eac3to?
if yes, which is the command to do this?
thx! :)

no, you cannot use use eac3to for this pourpose

eac3to is an advanced audio processor AND can MUX/DEMUX (no encoding is done) into MKV

you cannot use eac3to to encode a vc1 (or whatever codec) to another codec

BHH

[zn]

If I have DTS and AC3 tracks from PAL, and need to make AC3 for NTSC, DTS source will be better for trancoding?

[shadowolf64]

Is there no way to convert a 6 channel AAC file to a 2 channel AAC without the Nero 7 decoder?

[tebasuna51]

Is there no way to convert a 6 channel AAC file to a 2 channel AAC without the Nero 7 decoder?
No with eac3to. There are others methods (all with NeroAacEnc like encoder) and with support for .aac/.m4a/.mp4 input:

- Command line with Faad v2.7 (http://kurtnoise.free.fr/index.php?dir=mp4tools/&file=faad2-2.7.7z):
faad -w -d multchan.aac | NeroAacEnc -ignorelength -if - -of stereo.m4a
- Foobar2000 (foo_dsp_downmix.dll)
- AviSynth methods BeHappy/MeGUI/Wavi/SoundOut (Bass or DirectShow decoders)

[shadowolf64]

No with eac3to. There are others methods (all with NeroAacEnc like encoder) and with support for .aac/.m4a/.mp4 input:

- Command line with Faad v2.7 (http://kurtnoise.free.fr/index.php?dir=mp4tools/&file=faad2-2.7.7z):
faad -w -d multchan.aac | NeroAacEnc -ignorelength -if - -of stereo.m4a
- Foobar2000 (foo_dsp_downmix.dll)
- AviSynth methods BeHappy/MeGUI/Wavi/SoundOut (Bass or DirectShow decoders)
Ah thank you that's very helpful! :D

[StephenB]

EAC3TO has problems with mpg clips from all HP digicams.

For instance, with this one: http://www.steves-digicams.com/2006_reviews/hp_r927/samples/hpim0054.mpg

it does the following:

eac3to v3.16
command line: c:\vid2eva\tools\eac3to\eac3to hpim0054.mpg test.mkv
------------------------------------------------------------------------------
The format of the source file could not be detected. <ERROR>

Here's the mediainfo:

General
Complete name : C:\HD\Cameras\temp\hpim0054.mpg
Format : MPEG-PS
File size : 10.4 MiB
Duration : 13s 916ms
Overall bit rate : 6 249 Kbps

Video
ID : 224 (0xE0)
Format : MPEG Video
Format version : Version 1
Format settings, Matrix : Default
Duration : 13s 916ms
Bit rate mode : Variable
Width : 640 pixels
Height : 480 pixels
Display aspect ratio : 4/3
Frame rate : 24.000 fps
Scan type : Progressive

Audio #1
ID : 192 (0xC0)
Format : MPEG Audio
Format version : Version 1
Format profile : Layer 2
Duration : 13s 896ms
Bit rate mode : Constant
Bit rate : 96.0 Kbps
Channel(s) : 1 channel
Sampling rate : 44.1 KHz
Resolution : 16 bits
Video delay : 23ms
Stream size : 163 KiB (2%)

Audio #2
ID : 194 (0xC2)

Text
ID : 0 (0x0)
Format : RLE
Format/Info : Run-length encoding

Menu
Format : DVD-Video


All the other HP digicam mpgs I've tried have the same error.

[CapnBob]

Eac3to has a big problem with seemless branching movies with THD soundtracks eg. Spiderman 2, Cars (reg.B), Ratatouille (reg.b).

It's a strange one but let me explain further. WHen converting a lossless THD track to either FFLAC or WAV it appears that eac3to does not do the 2nd pass to correct overlaps that it does when it converts the same THD track to regular DD for instance.

If I take any of the examples above I can convert THD to DD and FLAC and whereas the DD track will be in sync with the movie, the FLAC (or WAV) conversion will be hugely out of sync by the end of the film.

I found it confusing why one track would have different syncs when converted to different formats and I think I've found the solution. The FLAC tracks do not appear to be using audio GAP information to correct gaps in a 2nd pass. By using an older version of eac3to before the 2nd pass for shortening gaps was created automatically I managed to copy the DD.gaps file and rename it FLAC.gaps and use that information in the gaps file to get a FLAC track that was nearly in sync as opposed to the 5 seconds or so that was the sync gap with the current eac3to.

Madshi, is it possible for you to have a look at this because for those of us with seamless branching THD BD's who want lossless audio are screwed at the moment and features such as seamless branching are only going to become more prevalent as time goes on.

If you need any logs or tests done with such files, let me know and I'll post the results for you.

tia

[setarip_old]

@StephenB

Hi!Format : MPEG Video
Format version : Version 1If you read the initial post to this thread, I believe you'll find no reference to processing MPEG-1 video (Only MP1 audio)...

[StephenB]

@StephenB

Hi!If you read the initial post to this thread, I believe you'll find no reference to processing MPEG-1 video (Only MP1 audio)...

IF that were it, wouldn't it be a different error? Unknown video format, or something like that?

[kypec]

IF that were it, wouldn't it be a different error? Unknown video format, or something like that?
Slightly different wording, yet exactly same meaning:The format of the source file could not be detected.
What precisely you can't understand in that error message?

[alpha_125]

Hi, i'm using Hdconverttox that use eac3to to muxing/demuxing/decoding/encoding...
Recently i'm having this problem constantly when I start from a mkv with aac audio tracks.

eac3to v3.16
command line: "F:\HDC\applications\eac3to.exe" "f:\hdc\6\sg1\TEST.mkv" 2: "F:\HDC\Temp\hdencodertox-pid2.ac3" -448
------------------------------------------------------------------------------
MKV, 1 video track, 2 audio tracks, 2 subtitle tracks, 1:56:31, 24p /1.001
1: h264/AVC, English, 720p23.976
"TEST"
2: AAC, Italian, 5.1 channels, 24khz
"Italian 5.1"
3: AAC, English, 5.1 channels, 24khz
"English 5.1"
4: Subtitle (SRT), Italian, "Italian"
5: Subtitle (SRT), English, "English"
[v01] The video bitstream is encoded in a non-standard framerate. <WARNING>
Bitstream parsing for tracks 2 and 3 failed. <WARNING>
Demuxing these tracks may still produce correct results - or not. <WARNING>
This audio conversion is not supported. <ERROR>

I knew that there was a bitstream parsing problem with vobsub...but aac also?

Strangely using eac3to to convert the aac into the mkv to ac3...<ERROR>

if I use eac3to to extract the track.aac from mkv it works
and then if I use eac3to to convert track.aac to track.ac3...it works again.

I don't know! all dec/enc/splitter used by eac3to are good:

F:\HDC\cianfrusaglie>F:\0\applications\eac3to.exe -test
eac3to (v3.16) is up to date
Nero Audio Decoder (Nero 7) works fine
ArcSoft DTS Decoder (1.1.0.0) works fine
Sonic Audio Decoder (4.3.0.169) works fine
Haali Matroska Muxer (2009-01-11) is up to date
Nero AAC Encoder (1.3.3.0) is up to date
Surcode DTS Encoder (1.0.21.0) is installed
MkvToolnix (2.8.0.0, release version) is up to date

there is some error that I can not see?


ps. sorry for my miserable english :D

[alpha_125]

errata corrige: using eac3to to extract aac generates a corrupt aac... :(

[poisondeathray]

@alpha_125: What happens when you use mkvextract to extract aac from .mkv?

It might be that 24Khz is not standard for the conversion to AC3, maybe try adding -resampleTo48000

[StephenB]

Slightly different wording, yet exactly same meaning:The format of the source file could not be detected.
What precisely you can't understand in that error message?

I understand the error message, it means that EAC3TO doesn't think the file itself is mpg, and doesn't know what it is. You get precisely the same error if you run EAC3TO on a text file, or any other non-media file.

That is a very different meaning from not being able to deal with a particular codec within a media file, which is what the "unknown video format" error is about.

What I don't understand is why EAC3TO can't identify the file type, since it seems to be an ordinary mpeg program stream.

[setarip_old]

@StephenBWhat I don't understand is why EAC3TO can't identify the file type, since it seems to be an ordinary mpeg program stream.You do understand, don't you, that MPEG-1 and MPEG-2 are two different formats?

"eac3to" does process MPEG-2 video but doesn't process MPEG-1 video (video files or video streams)...

[alpha_125]

@alpha_125: What happens when you use mkvextract to extract aac from .mkv?

It might be that 24Khz is not standard for the conversion to AC3, maybe try adding -resampleTo48000

mkvextract make his job fine, it ask only the extention AAC...and extract.

My problem is now the encoding to ac3...maybe tomorrow will be a more lucky day :)

[Momber]

The transcoding to FLAC from PCM 3.0 of the English audio stream on the "Vicky Cristina Barcelona" BluRay failed miserably with respect to channel mapping.
The dialogue is in the left surround speaker of the finished FLAC stream.
AC3 encoding of same PCM 3.0 stream failed (using the options -down2 and -448) with the message "non supported data format".

[StephenB]

@StephenBYou do understand, don't you, that MPEG-1 and MPEG-2 are two different formats?

"eac3to" does process MPEG-2 video but doesn't process MPEG-1 video (video files or video streams)...

You do understand the difference between an mpeg program stream and an elementary stream? Since a valid MPEG-1 program stream is a valid MPEG-2 program stream as well, I expected EAC3TO to recognize the container.

I now know that EAC3TO doesn't support mp1v video anyway, so I have modified my program to use another tool for those streams.

[shadowolf64]

Does eac3to have any way to convert from OGG Vorbis audio to AAC?

[CapnBob]

Eac3to has a big problem with seemless branching movies with THD soundtracks eg. Spiderman 2, Cars (reg.B), Ratatouille (reg.b).

It's a strange one but let me explain further. WHen converting a lossless THD track to either FFLAC or WAV it appears that eac3to does not do the 2nd pass to correct overlaps that it does when it converts the same THD track to regular DD for instance.

If I take any of the examples above I can convert THD to DD and FLAC and whereas the DD track will be in sync with the movie, the FLAC (or WAV) conversion will be hugely out of sync by the end of the film.

I found it confusing why one track would have different syncs when converted to different formats and I think I've found the solution. The FLAC tracks do not appear to be using audio GAP information to correct gaps in a 2nd pass. By using an older version of eac3to before the 2nd pass for shortening gaps was created automatically I managed to copy the DD.gaps file and rename it FLAC.gaps and use that information in the gaps file to get a FLAC track that was nearly in sync as opposed to the 5 seconds or so that was the sync gap with the current eac3to.

Madshi, is it possible for you to have a look at this because for those of us with seamless branching THD BD's who want lossless audio are screwed at the moment and features such as seamless branching are only going to become more prevalent as time goes on.

If you need any logs or tests done with such files, let me know and I'll post the results for you.

tia

Anyone got any clues on this? Awkward seamless branching movies that add a delay to audio when ripping the lossless THD track to FLAC are PEREFCTLY IN SYNC when converted to DD. Something is happening during conversion to dolby digital that is not happening when converting to flac. It looks like the same gaps file information that is used for DD is not being used for FLAC. If this bug could be fixed then eac3to would be batting 100% for all blurays disks as far as I'm aware.

Anyone else curious about this?

[Snowknight26]

Are you using the latest version of eac3to? I just did Ratatouille and the FLAC track was fine.

[CapnBob]

Are you using the latest version of eac3to? I just did Ratatouille and the FLAC track was fine.

I'll bet a million bucks that your copy of Ratatouille was Region A and not the seamless branching Region B version. Pixar seem to be rendering any text in their movies in native European languages via small .m2ts files that are seamlessly branched in. If you want an example in your region then Spiderman 2/2.1 will fit the bill nicely. People have been unable to keep the audio in sync when converting to FLAC with this either.

What I did notice after I tried ripping for the umpteenth time is that eac3to WILL fix the gaps if converting from THD to DD but not when converted to FLAC. Using an old version of eac3to (2.77) which did not automatically fix the gaps but required a 2nd pass I was able to extract the DD.gaps file, rename it to FLAC.gaps and it very nearly fixed the FLAC file (still out by a small amount but drastically better than letting eac3to 3.xx rip the movie).

If eac3to will correctly convert to one file type but not another then that's a bug, right?

I'm not crying about the loss of these 3 BD's but there will be a lot of miffed people in Europe who can't rip Toystory 1 & 2 and The incredibles etc.

[Snowknight26]

I'll bet a million bucks that your copy of Ratatouille was Region A and not the seamless branching Region B version.
True, it was Region A, but the Region A one still uses seamless branching.

[CapnBob]

True, it was Region A, but the Region A one still uses seamless branching.

Yup, but Ratatouille Region A doesn't have Dolby Digital True HD. I believe it has LPCM (24/48) a luxury that was probably forgone for Europe due to the multiple extra languages required for that release.

So we've concluded that ripping the lossless audio from seamless branching movies with LPCM (Pixar Region A) and DTS-MA (Rambo) are OK but seamless branching True HD movies (Spiderman 2, Cars -Region B, Ratatouille -Region B) do not work as they get progressively further and further out of sync. What I have found out is that eac3to will perfectly sync said movies when converting to a lossy format such as Dolby Digital (and probably DTS as well as I have seen such rips on the interweb but lack the surcode DTS encoder to check myself).

This being the case there is a large bug in the way that eac3to converts to lossless formats such as flac and wav (yup, tried that too) for seamless branching THD movies. I could try converting to LPCM but is there anyway to mux that back together as I don't think mkvmerge will touch that format?

Any way to flag this up to Madshi?

As stated before, I am happy to post logs if this will help solve this problem.

Cheers!

[honai]

So we've concluded that ripping the lossless audio from seamless branching movies with LPCM (Pixar Region A) and DTS-MA (Rambo) are OK but seamless branching True HD movies (Spiderman 2, Cars -Region B, Ratatouille -Region B) do not work

No, we didn't conclude anything. It's just the speculation of a single user. FWIW, my Ratatouille Region B Blu-ray does not have TrueHD but PCM audio. Also, I have successfully converted Spiderman 2 with eac3to in the past, no issues.

[Thunderbolt8]

strange thing with the "A bug's life" BD, according to the log its 39.395p. but is this only a display bug, because anything else tells me its 23.976fps, when I play the remuxed file (length is 1:34:53; playlist info tells 24/1.001 as well)?

eac3to v3.16
command line: eac3to X:\movie 1) 2: X:\life.mkv 3: X:\life.flac 3: X:\life.dtsma 6: X:\lifecomment.ac3 7: X:\life.sup
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 6 subtitle tracks, 1:34:53, 39.395p
1: Chapters, 36 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
(core: DTS-ES, 5.1 channels, 24 bits, 1509kbps, 48khz)
4: AC3, French, 5.1 channels, 640kbps, 48khz
5: AC3, Spanish, 5.1 channels, 640kbps, 48khz
6: AC3, English, 2.0 channels, 192kbps, 48khz
7: Subtitle (PGS), English
8: Subtitle (PGS), French
9: Subtitle (PGS), Spanish
10: Subtitle (PGS), English
11: Subtitle (PGS), French
12: Subtitle (PGS), Spanish
[a03] Extracting audio track number 3...
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[s07] Extracting subtitle track number 7...
[v02] Muxing video to Matroska...
[s10] Extracting subtitle track number 10...
[a06] Extracting audio track number 6...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "X:\life.dtsma"...
[a03] Creating file "X:\life.flac"...
[a06] Creating file "X:\lifecomment.ac3"...
[s07] Creating file "X:\life.sup"...
[a03] The original audio track has a constant bit depth of 24 bits.
[a03] Audio overlaps for 9ms at playtime 0:27:11. <WARNING>
[a03] Audio overlaps for 9ms at playtime 1:28:00. <WARNING>
[a03] Audio overlaps for 9ms at playtime 0:27:11. <WARNING>
[a03] Audio overlaps for 9ms at playtime 1:28:00. <WARNING>
[a06] Audio overlaps for 13ms at playtime 0:00:27. <WARNING>
[a06] Audio overlaps for 17ms at playtime 0:27:11. <WARNING>
[a06] Audio overlaps for 30ms at playtime 1:28:00. <WARNING>
[a06] Audio overlaps for 12ms at playtime 1:28:28. <WARNING>
[a03] Starting 2nd pass...
[a03] Decoding FLAC...
[a03] Realizing RAW/PCM gaps...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "X:\life.flac"...
[a03] The processed audio track has a constant bit depth of 24 bits.
[a03] Starting 2nd pass...
[a03] Realizing DTS gaps...
[a03] Creating file "X:\life.dtsma"...
[a06] Starting 2nd pass...
[a06] Realizing (E-)AC3 gaps...
[a06] Creating file "X:\lifecomment.ac3"...
Added fps value to MKV header.
Video track 2 contains 136511 frames.
Subtitle track 7 contains 1497 captions.
Subtitle track 10 contains 1878 captions.
eac3to processing took 42 minutes, 13 seconds.
Done.

[mrr19121970]

@Madshi

I noticed a couple of small features:

using

eac3to D:\ -progressnumbers
eac3to D:\ 1) -progressnumbers
doesn't work as I imagined, eac3to goes straight into processing.

also when doing this on a HD-DVD:

eac3to D:\ 1)

multiple times, the results are different each time - like a counter is not being reset so the (results are 1), the 2), then 3) etc)

[mrr19121970]

http://forum.doom9.org/showthread.php?p=1286012#post1286012

The transcoding to FLAC from PCM 3.0 of the English audio stream on the "Vicky Cristina Barcelona" BluRay failed miserably with respect to channel mapping.
The dialogue is in the left surround speaker of the finished FLAC stream.
AC3 encoding of same PCM 3.0 stream failed (using the options -down2 and -448) with the message "non supported data format".

I asked the same question a few pages back:

http://forum.doom9.org/showthread.php?p=1283264#post1283264

and tebasuna51 (http://forum.doom9.org/member.php?u=71585) answered already:

http://forum.doom9.org/showthread.php?p=1283462#post1283462

I have built this into Clown_BD so you don't need to mess with the CLI if you prefer. Checkout the link in my SIG

[honai]

I get the same problem as Thunderbolt8 with Cars BD. Not sure if the fps in the stream is set correctly, haven't remuxed yet, but it would be helpful if in the log the true fps were indicated:

Added fps value [xx.yyy] to MKV header.

[odin24]

Does the -down6 switch mix the entire rear channels with the sides, or does it work in a different way?

Thanks.

[zn]

Does r8brain (pro) produce better quality than SSRC for pal/ntsc conversions?

What FPS values are supported? Is it planned to make it possible to use any custom FPS?

-23.976/... define source fps to be "23.976", "24.000", "25.000", ...
-changeTo24.000 change source fps to "23.976", "24.000", "25.000", ...

What arguments I can use with ssrc_hp.exe to get same settings?

Did you compare SSRC with SoX Resampler (http://sox.sourceforge.net/SoX/Resampling) (GPL)? Code (http://sox.cvs.sourceforge.net/viewvc/sox/sox/src/rate.c?view=markup) is under active developement (http://sox.cvs.sourceforge.net/viewvc/sox/sox/src/rate.c?view=log) right now.
This code also used for foobar2000 resampler plugin (http://www.hydrogenaudio.org/forums/index.php?showtopic=67376), users feedback looks positive.

[hoju3508]

Using Clown_BD to create iso of main movie, eng track and sub. Is this something to be concerned about?

[a03] Audio overlaps for 9ms at playtime 0:27:11. <WARNING>
[a03] Audio overlaps for 9ms at playtime 1:28:00. <WARNING>


:thanks:


eac3to v3.16
command line: "C:\eac3to\eac3to.exe" "D:\" 1) 1: "C:\Clown_BD_WORK\Chapters_1.txt" 2: "C:\Clown_BD_WORK\Video_2.*" 3: "C:\Clown_BD_WORK\Audio_3_English.DTS" 7: "C:\Clown_BD_WORK\Subtitles_7_English.sup" -LOG="C:\Clown_BD_WORK\eac3to_PASS3_LOG.LOG"
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 6 subtitle tracks, 1:34:53, 39.395p
1: Chapters, 36 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
(core: DTS-ES, 5.1 channels, 24 bits, 1509kbps, 48khz)
4: AC3, French, 5.1 channels, 640kbps, 48khz
5: AC3, Spanish, 5.1 channels, 640kbps, 48khz
6: AC3, English, 2.0 channels, 192kbps, 48khz
7: Subtitle (PGS), English
8: Subtitle (PGS), French
9: Subtitle (PGS), Spanish
10: Subtitle (PGS), English
11: Subtitle (PGS), French
12: Subtitle (PGS), Spanish
Creating file "C:\Clown_BD_WORK\Chapters_1.txt"...
[a03] Extracting audio track number 3...
[v02] Extracting video track number 2...
[s07] Extracting subtitle track number 7...
[v02] Creating file "C:\Clown_BD_WORK\Video_2.h264"...
[a03] Creating file "C:\Clown_BD_WORK\Audio_3_English.DTS"...
[s07] Creating file "C:\Clown_BD_WORK\Subtitles_7_English.sup"...
[a03] Audio overlaps for 9ms at playtime 0:27:11. <WARNING>
[a03] Audio overlaps for 9ms at playtime 1:28:00. <WARNING>
[a03] Starting 2nd pass...
[a03] Realizing DTS gaps...
[a03] Creating file "C:\Clown_BD_WORK\Audio_3_English.DTS"...
Video track 2 contains 136511 frames.
Subtitle track 7 contains 1497 captions.
eac3to processing took 21 minutes, 27 seconds.
Done.

[Chiamos]

Using Clown_BD to create iso of main movie, eng track and sub. Is this something to be concerned about?

[a03] Audio overlaps for 9ms at playtime 0:27:11. <WARNING>
[a03] Audio overlaps for 9ms at playtime 1:28:00. <WARNING>


:thanks:

I've seen that similar warning maybe once, but do not recall the Title, as never noticed anything that effected playback, but maybe some here can provide the cause and possible effect.:)

[Killroy™]

Seamless branching does that. They don't show up if ac3to does a second pass.

[tebasuna51]

...
[a03] Audio overlaps for 9ms at playtime 0:27:11. <WARNING>
[a03] Audio overlaps for 9ms at playtime 1:28:00. <WARNING>
[a03] Starting 2nd pass...
[a03] Realizing DTS gaps...
[a03] Creating file "C:\Clown_BD_WORK\Audio_3_English.DTS"...

All is OK, the gaps/overlaps was corrected.

[mrr19121970]

@ hoju3508 (http://forum.doom9.org/member.php?u=139770)


you should probably use a different naming convention than:

C:\Clown_BD_WORK\

this would be better

C:\Clown_BD_WORK\NAME_OF_MOVIE

this way you'll get an M2TS, TS or ISO called NAME_OF_MOVIE rather than Clown_BD_WORK

[hoju3508]

I've seen that similar warning maybe once, but do not recall the Title, as never noticed anything that effected playback, but maybe some here can provide the cause and possible effect.:)

Seamless branching does that. They don't show up if ac3to does a second pass.

All is OK, the gaps/overlaps was corrected.

:thanks:

@ hoju3508 (http://forum.doom9.org/member.php?u=139770)


you should probably use a different naming convention than:

C:\Clown_BD_WORK\

this would be better

C:\Clown_BD_WORK\NAME_OF_MOVIE

this way you'll get an M2TS, TS or ISO called NAME_OF_MOVIE rather than Clown_BD_WORK

Yeah, I noticed that. It was my first attempt at using Clown_BD.

:thanks:

[leeperry]

Does r8brain (pro) produce better quality than SSRC for pal/ntsc conversions?
using SineGen & WaveSpectra you can easily compare them yourself.

I used them to compare various samplerates in Reclock : http://forum.slysoft.com/showpost.php?p=195873&postcount=103

some comparison w/ r8brain Pro would be most instructive.

for what it's worth, the THD+N values were unacceptable w/ Wavelab's resampler :o

[gillie]

Has anyone seen a problem with eac3to where it incorrectly identifies the chapter list on The Da Vinci Code and returns a list with over a hundred chapters as opposed to the 16 it actually has.

I found an application called Chapter Grabber that allowed me to get the correct chapter list.

Using eac3to v3.16 and its the UK PAL Blu-Ray of Da Vinci Code Extended Code.

[mrr19121970]

Has anyone seen a problem with eac3to where it incorrectly identifies the chapter list on The Da Vinci Code and returns a list with over a hundred chapters as opposed to the 16 it actually has.

I found an application called Chapter Grabber that allowed me to get the correct chapter list.

Using eac3to v3.16 and its the UK PAL Blu-Ray of Da Vinci Code Extended Code.


if you zip up your .mpls and .clpi files (eh anydvd logfile) and PM them to madshi he can have a look, if not then all he can do is say "that's nice"....

[rica]

Madshi, i know you are busy enough with madvr but please take a look from time to time to your own thread.

[nyonya]

Has anyone been able to rip the 'Waiting...' Blu-Ray? When I type in
c:\eac3to.exe e: 2)
with 2) being the main movie, it just hangs there for 20 seconds or so, and then brings me back to the prompt. Every other disc I have tried has worked just fine. Thoughts?

[loekf]

Maybe a stupid question, but is there a way to force eac3to into silent mode ? I haven't been able to find any commandline option to disable to "buzzer" sound after an error has occurred or the other sound after completion.

I'm using eac3to in a video transcoding tool I'm writing and these sounds are, I won't say annoying.... , but not preferred... ;-)

[TinTime]

I think you can just delete the two wav files that eac3to uses.

[Xorp]

If a Blu-ray has multiple 7.1 tracks, lets say all three lossless/uncompressed varieties, and I want to downmix to 5.1, does it matter which one I use? I'm guessing the PCM track might better since the layout is more straight-forward.

[Snowknight26]

How can there be clipping when encoding an E-AC3 track to AC3? Isn't the E-AC3 track decoded to floating point before being fed to libAften?

[tebasuna51]

How can there be clipping when encoding an E-AC3 track to AC3? Isn't the E-AC3 track decoded to floating point before being fed to libAften?
Yes, when you decode a lossy format to float you can have artifacts over the Max. value (can't be present at original source).

If you only decode and recode (without downmix, resample or other audio change) you can safely use the new -no2ndpass parameter because any clip is a encoder artifact.

[Snowknight26]

If you only decode and recode (without downmix, resample or other audio change) you can safely use the new -no2ndpass parameter because any clip is a encoder artifact.

So this would be the libAften's fault?:

eac3to v3.16
command line: eac3to.exe ..\temp\asd.eac3 ..\temp\asd.ac3
------------------------------------------------------------------------------
E-AC3, 5.1 channels, 1:40:23, 640kbps, 48khz
The Nero decoder doesn't seem to work, will use libav instead.
Decoding with libav/ffmpeg...
Remapping channels...
Encoding AC3 <640kbps> with libAften...
Creating file "..\temp\asd.ac3"...
Clipping detected, a 2nd pass will be necessary. <WARNING>
Starting 2nd pass...
Decoding with libav/ffmpeg...
Remapping channels...
Encoding AC3 <640kbps> with libAften...
Creating file "..\temp\asd.ac3"...
eac3to processing took 7 minutes, 42 seconds.
Done.

[Mtz]

madshi, can you implement extracting the subtitles from HDTV recordings? The subtitles can be viewed if playing the files with VLC.

Here (http://www.sendspace.com/file/a2ep4l) is one example with DVB Subtitles.
TS, 1 video track, 4 audio tracks, 8 subtitle tracks, 0:02:40, 50i
1: h264/AVC, 1080i50 (16:9)
2: MP2, English, 2.0 channels, 256kbps, 48khz, -903ms
3: MP2, Polish, 2.0 channels, 256kbps, 48khz, 18ms
4: MP2, Hungarian, 2.0 channels, 256kbps, 48khz, -149ms
5: MP2, Czech, 2.0 channels, 256kbps, 48khz, -68ms
6: Subtitle (DVB), Dutch
7: Subtitle (DVB), Serbian
8: Subtitle (DVB), Slovenian
9: Subtitle (DVB), Croatian
10: Subtitle (DVB), Swedish
11: Subtitle (DVB), Danish
12: Subtitle (DVB), Romanian
13: Subtitle (DVB), Norwegian
[v01] The video track contains the (probably incorrect) "full range" flag. <WARNING>
This subtitle conversion is not supported. <ERROR>

Here (http://www.sendspace.com/file/ktzc52) is one example with teletext subtitle and DVB Subtitle.
Here (http://www.sendspace.com/file/yovv7y) is one sample with teletext subtitle.
Here (http://www.sendspace.com/file/oegamw) is one TRP file with no subtitles.

Also, the audio delay is not correct on all files.

enjoy,
Mtz

[tebasuna51]

So this would be the libAften's fault?:

Fault?
For what fault?
For what libAften?

All is ok.

[Snowknight26]

You said that if there is any clipping when decoding then reencoding (in one step), its the encoders fault. So in this case its libAften.. which isn't plausible.

[tebasuna51]

The clip is at decoder pass (not encoder) and is normal with lossy formats.

[StephenB]

The clip is at decoder pass (not encoder) and is normal with lossy formats.


Do you know why it happens though?

Is this because the decoder is not quite compliant?

[tebasuna51]

Do you know why it happens though?

Is this because the decoder is not quite compliant?
Because is lossy compression.
If the source file have some peaks at 0 dB, some of them are encoded/decoded to obtain peaks between -0.1dB (no problem) and +0.1dB (clip to -> 0 dB)

The clipped peak recover the original volume, maybe the problem is in the peak at -0.1 dB.

If you want exact translation you need lossless encoders.

[StephenB]

Because is lossy compression.
If the source file have some peaks at 0 dB, some of them are encoded/decoded to obtain peaks between -0.1dB (no problem) and +0.1dB (clip to -> 0 dB)

The clipped peak recover the original volume, maybe the problem is in the peak at -0.1 dB.

If you want exact translation you need lossless encoders.

Well, I do understand that lossy codecs are not exact.

But it seems to me that a well designed encoder would not exceed the clipping limit on any signal it encodes.

After all, all compliant decoders should obtain the same (non-exact) result (at least with standard floating point hardware), so it seems to me that the encoder algorithm can/should prevent the clipping.

[StephenB]

Madshi,

It looks like SOPHOS antivirus software is generating a false positive on EAC3TO V3.16 You can see this here:

http://virscan.org/report/aa2d3e03ec288e0f26c03f7f9438f232.html

[Jeff Flowerday]

Madshi,

It looks like SOPHOS antivirus software is generating a false positive on EAC3TO V3.16 You can see this here:

http://virscan.org/report/aa2d3e03ec288e0f26c03f7f9438f232.html

Well then report it to them as a false positive not madshi, he has no control over their virus scanner.

[jruggle]

Well, I do understand that lossy codecs are not exact.

But it seems to me that a well designed encoder would not exceed the clipping limit on any signal it encodes.

After all, all compliant decoders should obtain the same (non-exact) result (at least with standard floating point hardware), so it seems to me that the encoder algorithm can/should prevent the clipping.
An AC-3 encoder actually has very little control over this. I guess it could, in theory, imploy an internal decoder and clipping detection, try to determine the frequency of clipped areas, then add bits to those frequencies using delta bit allocation, hoping removing bits from other places would not cause clipping elsewhere. I very highly doubt that Dolby's encoder even does this, as it would be very impractical, not completely reliable, and would decrease the quality.

It is explictly mentioned in the AC-3 specification that the output signal from the decoder may exceed 100%. Here is the exact text from A/52B section 7.9.4:
"Since the output signal consists of the original signal plus coding error, it is possible for the output signal to exceed 100 percent level even though the original input signal was less than or equal to 100 percent level."

It is up to the decoder what to do about this. Most will probably clip the results though. The specification mentions it so that those implementing a decoder will not assume a certain output range in their calculations.

[StephenB]

Madshi,

It looks like SOPHOS antivirus software is generating a false positive on EAC3TO V3.16 You can see this here:

http://virscan.org/report/aa2d3e03ec288e0f26c03f7f9438f232.html

SOPHOS says they have fixed this.

[StephenB]

An AC-3 encoder actually has very little control over this. I guess it could, in theory, imploy an internal decoder and clipping detection, try to determine the frequency of clipped areas, then add bits to those frequencies using delta bit allocation, hoping removing bits from other places would not cause clipping elsewhere. I very highly doubt that Dolby's encoder even does this, as it would be very impractical, not completely reliable, and would decrease the quality.

It is explictly mentioned in the AC-3 specification that the output signal from the decoder may exceed 100%. Here is the exact text from A/52B section 7.9.4:
"Since the output signal consists of the original signal plus coding error, it is possible for the output signal to exceed 100 percent level even though the original input signal was less than or equal to 100 percent level."

It is up to the decoder what to do about this. Most will probably clip the results though. The specification mentions it so that those implementing a decoder will not assume a certain output range in their calculations.

It still seems odd to me that professionally mastered AC3 tracks on commercial disks would be encoded so that every AC3 decoder on the planet has to clip.

[ron spencer]

quick question...if I am changing 640 AC3 to 448 with EAC3to and the 640 is stated to have bit depth of 24 bits, will the encode via libAften to 448 reduce bit depth to 16 automatically or must it be specified first?

thanks

-rs

[Snowknight26]

Lossy formats don't have a bit-depth.

[TinTime]

The AC3 decoder (the Nero one anyway) decodes to 24 bits. It's not the AC3 itself that's 24 bit. If encoding to AC3 just pass it the best quality audio possible - 24 bit in this case. There's no need to reduce to 16 bit.

[tebasuna51]

If you use eac3to to recode ac3 640Kbs to 448Kbs the decoder (at least libav) suply audio samples with 32 bits float, and the Aften encoder (libaften) read these 32 bit float to obtain the ac3 stream.
The ac3 stream don't have a fix bitdepth, the quality is measured in bitrate instead.

[Ryu77]

I had some recent difficulty re-authoring a disc with a TrueHD track and the only thing I can pin it on is EAC3to. I will explain more so you can let me know if this is normal behavior for this application.

I demuxed a Dolby TrueHD (TrueHD+AC3) track with EAC3to and then re-encoded the video, after that remuxed back together with tsMuxeR.

My Pioneer SC-LX81 AVR made some spluttering noises and the audio was quite garbled (bitstreamed from LG BD370). My PS3 seemed to decode the audio and send the AVR the LPCM track with no problem.

So I decided to take the disc to work and on 4 other AVR's in the store, the info display showed "DIAL NORM -27" (I think that was the number, I could be off a little but I am sure it was in it's twenties). The AVR's at work seemed to be able to play the audio ok. However, I have never seen such a large number show up on my AVR. It is usually somewhere between +8 and -8 (from memory???). After doing some reading it appears that -27 could be the default dialog normalization figure before any metadata changes are made. This suggested to me that maybe EAC3to removed the dialog normalization when demuxing. To further solidify my thoughts, I remuxed the same tracks, only this time avoiding using the demuxed (thd+ac3) track from EAC3to, instead I loaded the original m2ts file into tsMuxeR and selected the TrueHD track that way.

This time it played without any problems at all, except now no DIAL NORM notification comes up on my AVR. To sum this post up...

Does EAC3to remove or alter dialog normalization when demuxing? I am aware how this could affect encoding but surely it shouldn't be removed or altered when demuxing, should it?

[tebasuna51]

Does EAC3to remove or alter dialog normalization when demuxing?
Yes, change any value of Dialog Normalization to -31 dB
I am aware how this could affect encoding but surely it shouldn't be removed or altered when demuxing, should it?
The change only affect this field value in headers (and CRC), is a lossless change without reencode.
This change can't be the source of your problems.

[TinTime]

If you want to keep the existing dialnorm value then you can use the -keepDialnorm option with eac3to.

A value of -27 is very usual.

As to whether the original value should be kept or not when demuxing, opinion is divided. It's really personal preference. I keep it myself.

[Ryu77]

Yes, change any value of Dialog Normalization to -31 dB

The change only affect this field value in headers (and CRC), is a lossless change without reencode.
This change can't be the source of your problems.

Well it obviously was the source of my problem as my receiver didn't like the audio track with the dialog normalization removed. Once remuxed with the dial norm untouched my AVR played it without any issue.


If you want to keep the existing dialnorm value then you can use the -keepDialnorm option with eac3to.

Thank you, I forgot about that command. I always thought this wasn't a problem with demuxing though. I always thought this was only removed for re-encoding purposes. Now I know different. :D

[tebasuna51]

Well it obviously was the source of my problem as my receiver didn't like the audio track with the dialog normalization removed. Once remuxed with the dial norm untouched my AVR played it without any issue.
Nope, can't be the problem.
The DialNorm isn't removed, is changed to a valid value (-31).

When the receiver read a -27 dB value in DialNorm make a global attenuation of 4 dB.

When the receiver read a -31 dB value in DialNorm make a global attenuation of 0 dB.

Maybe is other problem related to the demux-remux.
:logfile:

[Ryu77]

Nope, can't be the problem.
The DialNorm isn't removed, is changed to a valid value (-31).

When the receiver read a -27 dB value in DialNorm make a global attenuation of 4 dB.

When the receiver read a -31 dB value in DialNorm make a global attenuation of 0 dB.

Maybe is other problem related to the demux-remux.
:logfile:

Thank you for your help but as mentioned earlier when I muxed with tsMuxeR only (avoiding EAC3to), the disc created played with no problem at all. So that leaves the only different viarable as EAC3to.

Another oddity I noticed is that when I extracted the AC3 stream from the TrueHD+AC3 track with tsMuxeR, which was originally demuxed with EAC3to, the dial norm registered as +4 on my AVR. Why this reverted back with the core stream, I have no idea. Maybe tsMuxeR altered the metadata back to it's original state somehow.

If you still want the log file, I can post that when I get home from work later today. I thought that the problem was pretty clear cut (and solved) but I can certainly post it later. I posted this as I had originally forgotten about the "-KeepDialnorm command" that TinTime reminded me of. Now that I know that it exists, I will use it to keep the audio as the studio intended.

[peterjcat]

Does EAC3to remove or alter dialog normalization when demuxing? I am aware how this could affect encoding but surely it shouldn't be removed or altered when demuxing, should it?

eac3to removes dialnorm by setting it to -31dB, which is the right way of doing it but the Pioneer receivers are known not to deal with -31dB properly for some reason. Use the -keepdialnorm tag in relation to every Dolby (DD/TrueHD) track you want to demux and you should be fine.

[tebasuna51]

eac3to removes dialnorm by setting it to -31dB, which is the right way of doing it but the Pioneer receivers are known not to deal with -31dB properly for some reason.

Nope, the problem in Pionner receiver was when the value is set to 0 dB, but this was corrected at:
v2.85
...
* AC3 and E-AC3 dialnorm removal now uses "-31db" instead of "-0db"

The -31dB must work fine on all players/recievers.

[TinTime]

The -31dB must work fine on all players/recievers.

Yep. My Yamaha amp works with -31dB but not 0. Or rather, it works with 0dB Dialnorm but is really quiet, as you would expect.

eac3to removes dialnorm by setting it to -31dB, which is the right way of doing it but the Pioneer receivers are known not to deal with -31dB properly for some reason. Use the -keepdialnorm tag in relation to every Dolby (DD/TrueHD) track you want to demux and you should be fine.

Pioneer amps must be able to handle a value of -31dB surely? This is a valid value. What if the source already has a value set of -31dB? Then the -keepdialnorm switch won't make any difference.

[rapscallion]

I have both the Sonic 4.2 and 4.3 versions and have been using 4.3 since it came out.

I have read somewhere (don't remember where, but on a forum) yesterday that a number of people have had problems with ver 4.3 decoding DTS to AC-3 and that 4.2 was more stable and the one to use.

Can anyone comment on this and confirm if it's true or not?

[Ryu77]

Nope, the problem in Pionner receiver was when the value is set to 0 dB, but this was corrected at:
v2.85
...
* AC3 and E-AC3 dialnorm removal now uses "-31db" instead of "-0db"

The -31dB must work fine on all players/recievers.

What about TrueHD? As mentioned earlier it was only the TrueHD track that caused the problem. When I extracted the AC3 core with tsMuxeR from the same problematic track the core somehow reverted back to dial norm +4. Also, my Pioneer AVR didn't display a dial norm fugure, it was one of the AVR's at work that showed something like -27 but the wierd part was they seemed to manage to play the audio.

I don't know what caused the problem but all I know is when I used tsMuxeR exclusively to handle the demuxing that everything was ok.

[rack04]

Why does eac3to remove the fullrange flag from h264 streams?

[honai]

Because there are almost always wrong, mostly due to a bug in the old Tandberg h.264 encoder that most European tv stations use.

[Ryu77]

Just to follow up on my previous posts...

I tested the disc created with the audio track (Dolby TrueHD) demuxed with EAC3to and then remuxed with tsMuxeR (after re-encoding video) on an Onkyo TXNR906 at work (again) and the display showed DIAL NORM -27dB. This same disc is the one that was spluttered and garbled on my Pioneer SC-LX81 (SC-07 in the USA). Strangely enough the disc displaying -27 seemed a little louder at the same volume setting on the Onkyo AVR.

However, the disc created with the audio track demuxed/remuxed exclusively with tsMuxeR didn't display anything at all on the Onkyo AVR but displays DIAL NORM +4 on my Pioneer AVR (and plays perfectly on both).

Another point is, it was said earlier that EAC3to doesn't remove dialog normalization but instead changes it to -31. If that is the case, why does the Onkyo AVR display -27? Also, why does the Dolby Digital core extracted from the same TrueHD track revert back to +4?

Also, another interesting note. I tested Transformers with a regular Dolby Digital track and it displayed DIAL NORM +4 on both the Onkyo TXNR906 at work and my SC-LX81 at home... ??? So that would indicate that they both indeed use the same measurement system. Maybe my Pioneer AVR didn't like the DIAL NORM -27 metadata setting. It also seems that EAC3to may have handled this TrueHD track different to usual.

[SomeJoe]

Ryu77,

I've had issues with Dolby TrueHD and my Pioneer receiver for a long time. (See this (http://forum.doom9.org/showthread.php?t=143986) thread). I'm virtually certain the issue is some kind of metadata in the TrueHD stream that's being altered by eac3to, tsMuxer, or one of the other common tools that happens to cause the audio on the Pioneers to not play back properly.

I'm going to try again with the latest tsMuxer, and use the -keepdialnorm option on eac3to and see if it fixes the problem.

Gotto go dig out the Iron Man BD for the 57th time ... :rolleyes:

[Ryu77]

Ryu77,

I've had issues with Dolby TrueHD and my Pioneer receiver for a long time. (See this (http://forum.doom9.org/showthread.php?t=143986) thread). I'm virtually certain the issue is some kind of metadata in the TrueHD stream that's being altered by eac3to, tsMuxer, or one of the other common tools that happens to cause the audio on the Pioneers to not play back properly.

I'm going to try again with the latest tsMuxer, and use the -keepdialnorm option on eac3to and see if it fixes the problem.

Gotto go dig out the Iron Man BD for the 57th time ... :rolleyes:

Hello Joe,

I did state above that when using tsMuxeR exclusively (avoiding EAC3to) to demux/remux that the TrueHD track plays without any problem on my Pioneer SC-LX81 (SC-07). I have authored a few discs with TrueHD and DTS-HD MA and they all play fine. Previously I used tsMuxeR to do the demuxing. It was only recently that I started using EAC3to to handle the demuxing and that is when the TrueHD problem occured.

I do like the ease of use that EAC3to offers when locating the main movie playlist and also I believe it handles seamless branching better. However, I am not sure what it is doing to the dialog normalization to make my Pioneer AVR to reject the track. So for the time being I may need to stick with tsMuxeR for all the work. :)

[simps]

Hi,

I have a DTS-ES 6.1 24bits track, and I would like to convert it, to a 2CH AC3 192.
Is this the correct command line for it, or am I missing something?

eac3to c:\a\a.dts c:\a\a.ac3 -down2 -normalize -192

Do I have to worry about that 24bit, and use -down16 too?

Thanks.

[DoomBot]

Ryu77,

I've had issues with Dolby TrueHD and my Pioneer receiver for a long time. (See this (http://forum.doom9.org/showthread.php?t=143986) thread). I'm virtually certain the issue is some kind of metadata in the TrueHD stream that's being altered by eac3to, tsMuxer, or one of the other common tools that happens to cause the audio on the Pioneers to not play back properly.

I'm going to try again with the latest tsMuxer, and use the -keepdialnorm option on eac3to and see if it fixes the problem.

Gotto go dig out the Iron Man BD for the 57th time ... :rolleyes:


Does this happen on every movie with TrueHD or is from movie to movie with TrueHD i ask because im having what seems to be the same problem, theres movies that are messed up and others that are not and im using a Pioneer receiver aswell. If the audio has problems do you hear it right away meaning through out the movie or is it parts of the audio during the movie?

I take it you have no problems with DTS-HD Master or PCM audio when demuxed and then remuxed?

[73ChargerFan]

In the log file, if the command was for something like stream 1) please list the .m2ts files that would be reported when listing 1), e.g. 00000.m2ts+00183.m2ts+00184.m2ts

[mrr19121970]

isn't your request already catered for ?

eac3to v3.03
command line: "E:\TVIX\eac3to\eac3to.exe" "R:\" -LOG="D:\DEMUX\Test1\PASS1_LOG.LOG"
------------------------------------------------------------------------------
1) 00211.mpls, 1:56:54
[105+149+107+109+110+113+114+116+117+119+120+122+123+126+127+129+130+132+134+136+137+139+140+142+143+145+146+148+133].m2ts
- h264/AVC, 1080p24 /1.001 (16:9)
- TrueHD, English, multi-channel, 48khz
- TrueHD, German, multi-channel, 48khz
- AC3, Spanish, multi-channel, 48khz
- AC3, English, stereo, 48khz
- AC3, English, stereo, 48khz
- AC3, English, stereo, 48khz
- AC3, English, stereo, 48khz
- AC3, Portuguese, multi-channel, 48khz

[73ChargerFan]

That is an BD inquiry. This is the log file I'm talking about:
eac3to v3.16
command line: eac3to 1) 1: chapters.txt 2: taken.mkv 4: taken.dtsma
------------------------------------------------------------------------------
M2TS, 2 video tracks, 6 audio tracks, 6 subtitle tracks, 1:33:25, 24p /1.001
1: Chapters, 24 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: h264/AVC, 480p24 /1.001 (20:11)
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
5: AC3, Spanish, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
6: AC3, French, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
7: AC3, French, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB
8: AC3, English, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB
9: DTS Express, English, 1.0 channels, 24 bits, 96kbps, 48khz
10: Subtitle (PGS), English
11: Subtitle (PGS), English
12: Subtitle (PGS), Spanish
13: Subtitle (PGS), French
14: Subtitle (PGS), English
15: Subtitle (PGS), English
Creating file "chapters.txt"...
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[v02] Muxing video to Matroska...
[a04] Creating file "taken.dtsma"...
[a04] Audio overlaps for 8ms at playtime 0:14:32. <WARNING>
[a04] Audio overlaps for 6ms at playtime 0:24:29. <WARNING>
[a04] Audio overlaps for 6ms at playtime 0:46:23. <WARNING>
[a04] Audio overlaps for 6ms at playtime 0:48:28. <WARNING>
[a04] Audio overlaps for 12ms at playtime 1:06:57. <WARNING>
[a04] Audio overlaps for 5ms at playtime 1:12:37. <WARNING>
[a04] Audio overlaps for 8ms at playtime 1:25:07. <WARNING>
[a04] Starting 2nd pass...
[a04] Realizing DTS gaps...
[a04] Creating file "taken.dtsma"...
Added fps value to MKV header.
Video track 2 contains 134392 frames.
Video track 3 contains 134392 frames.
eac3to processing took 20 minutes, 46 seconds.
Done.

[marklar]

Hi madshi,

do you plan to add VobSub format support of subtitles to eac3to? It was already disscussed here that it is not supported. But is it planned?

Thanks!

[wildchild22]

I have a bug report for eac3to when using seamless branching blurays and .w64 audio extension. eac3to creates a w64 file then it detects the gaps and re writes the w64 file. The first file is 4.5 gigs but after it re-writes the w64 file it is only 465 megs. It seems to only write the gap information up to the first gap then it stops. Using the same file with the dts or pcm file extension creates a file the correct size.

[DoomBot]

Where is madshi, we need him back on eac3to fixing W64 and what ever is wrong with truehd after demuxing and using Pioneer receivers the truehd thing kills me more.

[DoomBot]

Hello Joe,

I did state above that when using tsMuxeR exclusively (avoiding EAC3to) to demux/remux that the TrueHD track plays without any problem on my Pioneer SC-LX81 (SC-07). I have authored a few discs with TrueHD and DTS-HD MA and they all play fine. Previously I used tsMuxeR to do the demuxing. It was only recently that I started using EAC3to to handle the demuxing and that is when the TrueHD problem occured.

I do like the ease of use that EAC3to offers when locating the main movie playlist and also I believe it handles seamless branching better. However, I am not sure what it is doing to the dialog normalization to make my Pioneer AVR to reject the track. So for the time being I may need to stick with tsMuxeR for all the work. :)

Your right about EAC3to doing something to the demuxed file with truehd, i have tried many movies with truehd demuxing with EAC3to and then muxing them with tsMuxeR and most dont work correctly but when doing the same movies only through tsMuxeR and avoiding EAC3to they all work perfectly. So i sure would like to know what EAC3to is doing to mess the truehd files up.

[DoomBot]

I have a seamless branching movie with truehd will EAC3to correct the audio gaps or do i have to convert it to pcm and whats the best way of going at that?

[tebasuna51]

I have a seamless branching movie with truehd will EAC3to correct the audio gaps or do i have to convert it to pcm and whats the best way of going at that?
¿The best way? There are many.
For me the best is convert to flac.

[DoomBot]

Well i dont want to use flac so how do you convert truehd to pcm step by step? Or is it as simple as telling EAC3to to out put .pcm file?

[wildchild22]

for pcm yes tell eac3to to eac3to audiotruehd audio.pcm and make sure you use the pcm2tsmu if you plan on muxing it or you will get static noise.

for pcm2tsmu check tebasuna51 signature.

[DoomBot]

for pcm yes tell eac3to to eac3to audiotruehd audio.pcm and make sure you use the pcm2tsmu if you plan on muxing it or you will get static noise.

for pcm2tsmu check tebasuna51 signature.

I have some seamless branching movies to do that have truehd, will the audio gaps get fixed and will it say in eac3to? I know that seamless branching movies with truehd with gaps can't be fixed so this is the only reason im doing pcm.

Can't i just run the pcm file back through eac3to as a .w64 and not get any static noise?

[tebasuna51]

I have a bug report for eac3to when using seamless branching blurays and .w64 audio extension. eac3to creates a w64 file then it detects the gaps and re writes the w64 file. The first file is 4.5 gigs but after it re-writes the w64 file it is only 465 megs. It seems to only write the gap information up to the first gap then it stops. Using the same file with the dts or pcm file extension creates a file the correct size.

And using .wav output?

If work we can use:

eac3to input.thd output.wav
eac3to output.wav output.w64

With pcm:

eac3to input.thd output.pcm
eac3to output.pcm output.w64

but maybe the .pcm (raw audio data without header) need more parameters:

eac3to output.pcm output.w64 -override -big -6 -16 -48000

if is a 5.1, 16 bitdepth and 48KHz.

[wildchild22]

Yes it works using a 2 step method but it would be nicer if the w64 file could be created in 1 step.

[tebasuna51]

Yes it works using a 2 step method but it would be nicer if the w64 file could be created in 1 step.
Work with .wav?

Sorry, I haven't a seamless branching bluray to test

[wildchild22]

the problem is with .w64 files only now with any others as far as I can tell. But a temp fix is to to convert to pcm then to w64

[tebasuna51]

the problem is with .w64 files only now with any others as far as I can tell. But a temp fix is to to convert to pcm then to w64

If wav work is more easy convert wav to w64 beacuse only the header must be changed, the audio data is exact.
Convert pcm to w64 need change the channel mapping and the audio data from big endian to little endian.

Also wav support more channels config and can be traslated to w64 without lose info.

[DoomBot]

If wav work is more easy convert wav to w64 beacuse only the header must be changed, the audio data is exact.
Convert pcm to w64 need change the channel mapping and the audio data from big endian to little endian.

Also wav support more channels config and can be traslated to w64 without lose info.


So are you saying movies with out the seamless branching will work just fine converting truehd to wav then wav to w64?

Converting truehd to either pcm, wav, or w64 there isn't any quality loss its bit for bit correct? there all doing the same thing?

So what your saying is converting truehd to wav then wav to w64 just takes less time to convert the file?

[tebasuna51]

Is better use wav than pcm like intermediate file to obtain a w64 accepted by TsMuxer.

[Adub]

Hi, I am seem to be encountering the same troubles as some previous users with the 300 HD-DVD.

Here is the log from Eac3to 3.16
C:\Documents and Settings\Merlin>"C:\Documents and Settings\Merlin\Desktop\HDCon
vertToX_1.8.332.2935\applications\eac3to.exe" G:\300_HD_DVD\HVDVD_TS\feature_300
NDOM6LF1VC1_HD1.EVO+G:\300_HD_DVD\HVDVD_TS\feature_300NDOM6LF1VC1_HD1_Divide.EVO
2: "D:\300\chapters.txt" 3: "D:\300\video.mkv" 6: "D:\300\audio1.m4a" 9: "D:\30
0\audio2.m4a" 10: "D:\300\sub1.sup" 11: "D:\300\sub2.sup" 12: "D:\300\sub3.sup"
EVO, 2 video tracks, 5 audio tracks, 4 subtitle tracks, 1:56:32
"feature"
1: Joined EVO file
2: Chapters, 30 chapters
3: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
4: VC-1, 480p30 /1.001 (3:2)
5: E-AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
6: TrueHD, English, 5.1 channels, 48khz, dialnorm: -27dB
7: E-AC3, French, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
8: E-AC3, Spanish, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
9: E-AC3, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
10: Subtitle (DVD), English
11: Subtitle (DVD), English
12: Subtitle (DVD), French
13: Subtitle (DVD), Spanish
Creating file "D:\300\chapters.txt"...
a09 The Nero decoder doesn't seem to work, will use libav instead.
v03 Extracting video track number 3...
a06 Extracting audio track number 6...
a09 Extracting audio track number 9...
v03 Writing new framerate "24fps /1.001" to bitstream.
a06 Removing TrueHD dialog normalization...
a09 Removing E-AC3 dialog normalization...
v03 Removing VC-1 pulldown...
a06 Decoding with libav/ffmpeg...
a09 Decoding with libav/ffmpeg...
v03 Muxing video to Matroska...
a06 Encoding AAC <0.50> with NeroAacEnc...
a09 Reducing depth from 64 to 32 bits...
a09 Encoding AAC <0.50> with NeroAacEnc...
s11 Extracting subtitle track number 11...
s10 Extracting subtitle track number 10...
s12 Extracting subtitle track number 12...
s11 Creating file "D:\300\sub2.sup"...
s10 Creating file "D:\300\sub1.sup"...
s12 Creating file "D:\300\sub3.sup"...
a06 libav End of stream indicated
a06 The original audio track has a constant bit depth of 16 bits.
-------------------------------------------------------------------------------
After displaying the above error, eac3to sits idle, without actually closing any streams or completing.

Is there any fix for this problem as of yet?

[DoomBot]

Is better use wav than pcm like intermediate file to obtain a w64 accepted by TsMuxer.

So i shouldn't have any problems with a seamless branching movie with no over gaps using wav output?

[TinTime]

So i shouldn't have any problems with a seamless branching movie with no over gaps using wav output?

I always convert TrueHD to wav with eac3to and haven't encountered any problems, seamless branching movie or otherwise. The only possible issue is the one that eac3to warns you about - oversized (>2GB) wav files. As you're converting the wav to w64 with eac3to that doesn't matter though as it handles large wavs.

Converting truehd to either pcm, wav, or w64 there isn't any quality loss its bit for bit correct? there all doing the same thing?

Yep. As tebasuna51 said wav is the best bet for an intermediate file in this case (it's just pcm data with a header describing it stuck at the front).

[DoomBot]

I always convert TrueHD to wav with eac3to and haven't encountered any problems, seamless branching movie or otherwise. The only possible issue is the one that eac3to warns you about - oversized (>2GB) wav files. As you're converting the wav to w64 with eac3to that doesn't matter though as it handles large wavs.



Yep. As tebasuna51 said wav is the best bet for an intermediate file in this case (it's just pcm data with a header describing it stuck at the front).

Thanks for your response that's exactly what i wanted to hear. :thanks:

[Atak_Snajpera]

Is better use wav than pcm like intermediate file to obtain a w64 accepted by TsMuxer.
TSmuxer also accepts large .wavs. I tried 7GB wav and it was correctly muxed to .m2ts :) So in fact we don't have to use .w64 for now.

[wildchild22]

When eac3to outputs wavs all I used to get was sound for the first 15-30 minutes then white noise.


TSmuxer also accepts large .wavs. I tried 7GB wav and it was correctly muxed to .m2ts :) So in fact we don't have to use .w64 for now.

[tebasuna51]

TSmuxer also accepts large .wavs. I tried 7GB wav and it was correctly muxed to .m2ts :) So in fact we don't have to use .w64 for now.

If is correct is a new and undocumented feature. But this is the eac3to thread.

When eac3to outputs wavs all I used to get was sound for the first 15-30 minutes then white noise.
The audio data in wav is ok, maybe your player can't play it.
There are many soft to manage wav files >4GB (encoders, AviSynth, ...)

[Tarnatos]

Hello,

i have a problem with dts-es matrixed files.

If i try to create the waves from the source file with:
X:\temp>eac3to demo.dts c:\temp\demo.wavs
DTS-ES, 5.1 channels, 0:01:55, 16 bits, 755kbps, 48khz
Decoding with ArcSoft DTS Decoder...
Patching bitdepth to 24 bits...
Writing WAVs...
Creating file "c:\temp\demo.R.wav"...
Creating file "c:\temp\demo.L.wav"...
Creating file "c:\temp\demo.C.wav"...
Creating file "c:\temp\demo.SL.wav"...
Creating file "c:\temp\demo.LFE.wav"...
Creating file "c:\temp\demo.SR.wav"...

Everthing works fine.

But i want that 7th channel so i use the -7 cmd to force the extraction.

But i doesn't work:
X:\temp>eac3to demo.dts c:\temp\demo.wavs -7
DTS-ES, 5.1 channels, 0:01:55, 16 bits, 755kbps, 48khz
Decoding with ArcSoft DTS Decoder...
Patching bitdepth to 24 bits...
Writing WAVs...

Eac3to only creates a log file and interrups.

Here the content of the log file:
eac3to v3.16
command line: eac3to demo.dts c:\temp\demo.wavs -7
------------------------------------------------------------------------------
DTS-ES, 5.1 channels, 0:01:55, 16 bits, 755kbps, 48khz
Decoding with ArcSoft DTS Decoder...
Patching bitdepth to 24 bits...
Writing WAVs...


Every other file i had tested worked fine.

Here is a little part of that file so you can test by yourselves.

DL: http://rapidshare.com/files/244067335/demo.dts


##EDIT

With Sonic it doen't work, too!
eac3to demo.dts demo.wavs -sonic
DTS-ES, 5.1 channels, 0:01:55, 16 bits, 755kbps, 48khz
Patching bitdepth to 24 bits...
Decoding with DirectShow (Sonic Audio Decoder)...
DirectShow reports 5.1 channels, 24 bits, 48khz
Writing WAVs...
Creating file "demo.L.wav"...
Creating file "demo.SL.wav"...
Creating file "demo.LFE.wav"...
Creating file "demo.C.wav"...
Creating file "demo.SR.wav"...
Creating file "demo.R.wav"...

eac3to demo.dts demo.wavs -sonic -7
DTS-ES, 5.1 channels, 0:01:55, 16 bits, 755kbps, 48khz
Patching bitdepth to 24 bits...
Decoding with DirectShow (Sonic Audio Decoder)...
DirectShow reports 5.1 channels, 24 bits, 48khz
Writing WAVs...
Creating file "demo.L.wav"...
Creating file "demo.SL.wav"...
Creating file "demo.LFE.wav"...
Creating file "demo.C.wav"...
Creating file "demo.SR.wav"...
Creating file "demo.R.wav"...

With libAv it's the same. Only 6 channels.

[aleg]

Is it possible to downmix a 2/2 recording (an old quadrophonics with L-R-SL-SR) to stereo?

The -down2 option leaves it as a 4-channel recording.

[tebasuna51]

...
DTS-ES, 5.1 channels, 0:01:55, 16 bits, 755kbps, 48khz
...
But i want that 7th channel so i use the -7 cmd to force the extraction.
...

-7 don't force to extract 7 channel (-7 means the pcm input have 7 channels)

Like you see is a 5.1 and the BackCenter channel is an analog matrix mix in BackLeft and BackRight channels.

There are other kind of DTS-ES with a discrete BackCenter than can be extracted, eac3to inform with this message:
DTS-ES, 6.1 channels, 0:00:22, 16 bits, 1510kbps, 48khz

[Tarnatos]

Than it is not possible to extract the matrixed backcenter channel?

[tebasuna51]

Than it is not possible to extract the matrixed backcenter channel?
For what?
You can play the 5.1 and the phantom BackCenter is there.

BTW, you can try to make a stereo with BL,BR and use a dpl decoder to 5.0, the new FL' is the original BL, the new FC' the original BC, and the new FR' the original BR (the news BL', BR' must be empty).

To do this conversion you can use Foobar2000 and the foo_dsp_fsurround.dll (http://www.hydrogenaudio.org/forums/index.php?showtopic=52235) plugin.

[cybpsy]

Hello!

I'm using eac3to 3.16. I have 7.1 TrueHD audio track. I'm trying to recode it to flac or to wavs, but I get such error:
eac3to.exe 00003.track_4352.ac3 audio.wavs
TrueHD/AC3, 7.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB)
Extracting TrueHD stream...
Removing TrueHD dialog normalization...
Decoding with libav/ffmpeg...
Remapping channels...
Writing WAVs...
libav Invalid channel 13 specified as output from matrix
The libav decoder reported an error while decoding.
Aborted at file position 262144.

I understand that it's problem of audio track or decoder, but may be there is some work around in eac3to? :)

I've made sample of the audio: http://rapidshare.com/files/244408548/00003.track_4352.ac3 (105 MB)

[tebasuna51]

I understand that it's problem of audio track or decoder, but may be there is some work around in eac3to? :)

Or maybe the tool used to extract the track (.ac3 with thd content?).
Let eac3to extract the track to the flac or wavs from the container.

[cybpsy]

Or maybe the tool used to extract the track (.ac3 with thd content?).
Let eac3to extract the track to the flac or wavs from the container.

The track is .ac3 only because it was demuxed by tsmuxer. I have also tried eac3to in two ways: decoding from BD directly and demuxing from BD (thd+ac3) and decoding to flac or wavs. Just the same error.

[tebasuna51]

The track is .ac3 only because it was demuxed by tsmuxer.
But eac3to can't imagine a TrueHD when say:
eac3to.exe 00003.track_4352.ac3 audio.wavs
TrueHD/AC3, 7.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB)
Extracting TrueHD stream...
I have also tried eac3to in two ways: decoding from BD directly and demuxing from BD (thd+ac3) and decoding to flac or wavs. Just the same error.

Then the stream is corrupt

[odin24]

Has anybody encountered this before? If so, is there anyway around it? I've examined all of the m2ts files in the two playlists with tsMuxeR, each branch has identical streams within the whole playlist... both 1) and 2).
Thanks.

http://img4.imageshack.us/img4/9439/eac3toerror.jpg

[setarip_old]

There are two observable differences:

1) A total 13 second time difference

2) Different sequence of the audio tracks

What is the title of your BluRay disc?

[mrr19121970]

@odin24

why have you blanked out your name in 3 out of 4 C:\Users\XXXXX> ?

and yes, I saw it yesterday. With the 'March of the Pengiuns' German version, Kinowelt have added a 13 second trailer to the start of the film. There were 3 playlists all with 00026.mts (from memory). The chapter points are different.

1) 1:22:26 mins long one with the trailer.
2) can't remember
3) 1:22:15 mins long - was clearly the original film without the trailer

However on demuxing either 1) or 3) I always got the longest version.

Here are the logs...

eac3to v3.16
command line: "E:\TVIX\Clown_BD\eac3to\eac3to.exe" "s:\" -LOG="D:\DEMUX\March of the Penguins\eac3to_PASS1_LOG.LOG"
------------------------------------------------------------------------------
1) 00027.mpls, 00026.m2ts, 1:22:29
- Chapters, 12 chapters
- VC-1, 1080i50 (16:9)
- DTS Hi-Res, German, multi-channel, 48khz
- DTS Hi-Res, German, multi-channel, 48khz
- DTS, French, multi-channel, 48khz
- DTS, German, multi-channel, 48khz
- AC3, German, stereo, 48khz
- AC3, French, stereo, 48khz

2) 00037.mpls, 00026.m2ts, 1:22:16
- Chapters, 12 chapters
- VC-1, 1080i50 (16:9)
- DTS Hi-Res, German, multi-channel, 48khz
- DTS Hi-Res, German, multi-channel, 48khz
- DTS, French, multi-channel, 48khz
- DTS, German, multi-channel, 48khz
- AC3, German, stereo, 48khz
- AC3, French, stereo, 48khz

3) 00028.mpls, 00026.m2ts, 1:22:15
- Chapters, 12 chapters
- VC-1, 1080i50 (16:9)
- DTS Hi-Res, German, multi-channel, 48khz
- DTS Hi-Res, German, multi-channel, 48khz
- DTS, French, multi-channel, 48khz
- DTS, German, multi-channel, 48khz
- AC3, German, stereo, 48khz
- AC3, French, stereo, 48khz

4) 00002.mpls, 00002.m2ts, 0:52:50
- MPEG2, 576i50 (16:9)
- AC3, French, stereo, 48khz
- AC3, French, stereo, 48khz

5) 00005.mpls, 00005.m2ts, 0:52:49
- MPEG2, 576i50 (16:9)
- AC3, French, stereo, 48khz

6) 00004.mpls, 00004.m2ts, 0:46:03
- MPEG2, 576i50 (16:9)
- AC3, French, stereo, 48khz

7) 00038.mpls, 0:32:49
[14+10+17+12+13+21+24+18+11+23+25+20+16+15+9+19+22].m2ts
- Chapters, 17 chapters
- MPEG2, 576i50 (16:9)
- AC3, French, stereo, 48khz

8) 00003.mpls, 00003.m2ts, 0:19:21
- MPEG2, 576i50 (4:3)
- AC3, French, stereo, 48khz

[BZeeme]

I get the same results as you using eac3to on CJ7. I'm trying to back it up using BD Rebuilder (movie only). The movie only result would play with PowerDvD but not my Sony standalone (audio, subtitles, but no video).

[pascalwil]

From Page 1:

I do have Nero 7.0.8.2 and the Blu-ray/HD-DVD plug-in registered.

I can use Nero WaveEditor or Nero Burning ROM to convert a 5.1 AAC file to a 5.1 AC3 file but I can't seem to be able to get eac3to to use the Nero AAC decoder to do the same job. I wonder what I've done wrong. eac3to reports the following:
eac3to v3.16
command line: "G:\Bin\eac3to316\eac3to.exe" -test
------------------------------------------------------------------------------
eac3to (v3.16) is installed
Nero Audio Decoder (Nero 6 or older) doesn't seem to be installed
http://www.nero.com/eng/store-blu-ray.html
CAUTION: You need Nero 7. Nero 8 won't work with eac3to.
ArcSoft DTS Decoder (1.1.0.0) works fine
Sonic Audio Decoder (4.3.0.169) works fine
Haali Matroska Muxer doesn't seem to be installed
http://haali.net/mkv
Nero AAC Encoder (1.3.3.0) is up to date
Surcode DTS Encoder (1.0.21.0) is installed
MkvToolnix (2.2.0.0, 2008-03-04) is installed

Of course I can use Sonic but I would love to use Nero inside eac3to. Can someone help? Many thanks.

And the log gives:
eac3to v3.16
command line: "G:\Bin\eac3to316\eac3to.exe" "Y:\Music (Video)\Movie\Original - x264\Movie - en-English.aac" "Y:\Music (Video)\Movie\Original - x264\Movie - 2005 - en-English - eac3to 3.16 with GUI - nero.ac3" -384 -nero
------------------------------------------------------------------------------
AAC, 5.1 channels, 48khz
I'd like to use the Nero decoder, but it doesn't seem to work. <ERROR>

Madshi could you tell us a little bit on how eac3to checks for Nero Audio Decoder installation?
Maybe with some inputs from you I could check whether I meet your requirements. Thanks.

[odin24]

There are two observable differences:

1) A total 13 second time difference

2) Different sequence of the audio tracks

What is the title of your BluRay disc?

The time difference is playlist 1 has "Columbia Pictures" intro, and 2) has "Sony Pictures Classics".

Also, one other thing I didn't notice last night in tsMuxeR... 00011.m2ts has a slightly different order to the tracks as the rest of the m2ts files in 1). Same goes for 2).

The movie in question is CJ7... I just wasn't sure I was allowed to post titles.

@odin24

why have you blanked out your name in 3 out of 4 C:\Users\XXXXX> ?

and yes, I saw it yesterday. With the 'March of the Pengiuns' German version, Kinowelt have added a 13 second trailer to the start of the film. There were 3 playlists all with 00026.mts (from memory). The chapter points are different.

1) 1:22:26 mins long one with the trailer.
2) can't remember
3) 1:22:15 mins long - was clearly the original film without the trailer

However on demuxing either 1) or 3) I always got the longest version.

I can update the post later with the logs.

LOL... i posted my initial concern just before bed, I was pretty tired... now that you all know my true identity... I must kill you!!!... Just kidding.

I'm pretty sure I know why, the first m2ts in each playlist is different from the rest of the m2ts' in rest of the movie. I'm pretty sure manually demuxing the rest, excluding the first m2ts will work... I just haven't tried yet.

I get the same results as you using eac3to on CJ7. I'm trying to back it up using BD Rebuilder (movie only). The movie only result would play with PowerDvD but not my Sony standalone (audio, subtitles, but no video).

I am working with CJ7 too. I think the only way to get this going properly is to manually demux, excluding the first m2ts file. Then remux with tsMuxeR, then feed that structure into BD-rebuilder.

Thanks for the responses people.

[rapscallion]

I've searched through this thread and found conflicting opinions re Surcode as a DTS Encoder.
For example, the 2 below....

One poster states : "Also I wouldnt use Surcodode DTS Encoder. Its a low quality and outdated encoder. And you cant encode dts es 6.1 with Surcode"

While another states : A dts file created from surcode will most definitely sound better than a core ac3 file if the source is the same. Such as a lossless audio track, like true hd. Remember, there is no magic way the studios have to encode an ac3 file. The same could be said of dts. Surcode outputs the same dts file as a studio would(if not very, very, close)

Which of these opinions have most of you found to be the most accurate ?

My goal is to convert/recode a 1536 dts file to a 768 bitrate. I've been using eacto_more gui , Sonic to decode and Surcode to encode, so am curious if this is the best way or is their another simpler one I should be using ? Thanks

[tebasuna51]

My goal is to convert/recode a 1536 dts file to a 768 bitrate.

For what?
http://www.ebu.ch/CMSimages/en/tec_doc_t3324-2007_tcm6-53801.pdf

[73ChargerFan]

My goal is to convert/recode a 1536 dts file to a 768 bitrate.
I'd listen to a 640 AC3 soundtrack, and compare it to a 768dts. I'd think they'd be about the same, and the 640 would probably be better (done by the dvd mixing company).

[qyot27]

I was looking to decode the MP3 Surround (http://en.wikipedia.org/wiki/MP3_Surround) files from the diatonis web site (http://www.diatonis.com/downloads.html), but I don't know if the process I'm using is correct concerning the channel ordering.

Only the Fraunhofer decoder package can decode MP3 Surround files to actual 5.1-channel PCM Waves, but while the output files have WAVE_FORMAT_EXTENSIBLE headers, the channels aren't defined. Running these Waves through eac3to produces files with the channels defined, but I have no idea if the Fraunhofer decoder reorders the channels or not (I would assume it does, though, if that's needed; however, I'd rather ask and be certain than just assume things).

[Kuningas]

OS - Windows 7.
eac3to doesn't see Nero Audio Decoder (Nero 6 or older). But I have Nero 7.11.10.0 and the Blu-ray/HD-DVD plug-in registered.
It works fine on Vista.

Is there any decision for Windows 7?

[darthcloud]

OS - Windows 7.
eac3to doesn't see Nero Audio Decoder (Nero 6 or older). But I have Nero 7.11.10.0 and the Blu-ray/HD-DVD plug-in registered.
It works fine on Vista.

Is there any decision for Windows 7?

I'm running Windows 7 x64 RC1 and everything work fine, nero BD/HDDVD plugin, arcsoft, nero aac, surcode...

[Kuningas]

darthcloud

I'm running Windows 7 x64 RC1 and everything work fine, nero BD/HDDVD plugin, arcsoft, nero aac, surcode...

What's you version of Nero?

[ggking7]

Sure was cool when the latest version of eac3to worked via wine.

[10L23r]

what's the equivalent of besweet's pregain in eac3to? i couldn't find such an option...

[tebasuna51]

You can use:
-normalize

[Steel]

Are there any known problems when demuxing a 7.1 source track such as a DTS-MA 7.1 into wav files with EAC3TO?

[tebasuna51]

I was looking to decode the MP3 Surround (http://en.wikipedia.org/wiki/MP3_Surround) files from the diatonis web site (http://www.diatonis.com/downloads.html), but I don't know if the process I'm using is correct concerning the channel ordering.

Only the Fraunhofer decoder package can decode MP3 Surround files to actual 5.1-channel PCM Waves, but while the output files have WAVE_FORMAT_EXTENSIBLE headers, the channels aren't defined. Running these Waves through eac3to produces files with the channels defined, but I have no idea if the Fraunhofer decoder reorders the channels or not (I would assume it does, though, if that's needed; however, I'd rather ask and be certain than just assume things).

The wav output from the command line mp3sDecoder.exe have the WAVE_FORMAT_EXTENSIBLE header, but the MaskChannel field have a value of '0' and can be rejected by some audio soft.

I make a test with mp3sEncoder -> mp3sDecoder and the channel order is correct then you can use eac3to to obtain a wav with a correct MaskChannel value:
MaskChannels : 1551 (FL FR FC LF SL SR)

Other correct value for MaskChannel is:
MaskChannels : 63 (FL FR FC LF BL BR)

[tebasuna51]

Are there any known problems when demuxing a 7.1 source track such as a DTS-MA 7.1 into wav files with EAC3TO?
You need use ArcSoft decoder.

This:
"Using eac3to with ArcSoft decoder there are a problem, the Side and Back channels are interchanged" (http://forum.doom9.org/showthread.php?p=1265484#post1265484)

Was solved with a more recent version of dtsdecoderdll.dll (http://forum.doom9.org/showthread.php?p=1266679#post1266679)

[qyot27]

The wav output from the command line mp3sDecoder.exe have the WAVE_FORMAT_EXTENSIBLE header, but the MaskChannel field have a value of '0' and can be rejected by some audio soft.

I make a test with mp3sEncoder -> mp3sDecoder and the channel order is correct then you can use eac3to to obtain a wav with a correct MaskChannel value:
MaskChannels : 1551 (FL FR FC LF SL SR)

Other correct value for MaskChannel is:
MaskChannels : 63 (FL FR FC LF BL BR)
Thanks. So the difference between the two is basically the difference between Dolby 5.1 and DTS 5.1 configurations?

[xkodi]

You need use ArcSoft decoder.

This:
"Using eac3to with ArcSoft decoder there are a problem, the Side and Back channels are interchanged" (http://forum.doom9.org/showthread.php?p=1265484#post1265484)

Was solved with a more recent version of dtsdecoderdll.dll (http://forum.doom9.org/showthread.php?p=1266679#post1266679)

TMT 3 has version 1.1.0.1 of "dtsdecoderdll.dll"
TMT 3.5 has version 1.1.0.10 of "dtsdecoderdll.dll"

have you tried this new versions?

the latest 1.1.0.10 doesn't work with eac3to though. anyway, it's interesting if they fixed the problem with DTS-HD MA tracks reported by eac3to as "7.1 (strange setup)". as you know such tracks are not bit-perfect decoded by old 1.1.0.0 versions of "dtsdecoderdll.dll"

[tebasuna51]

Thanks. So the difference between the two is basically the difference between Dolby 5.1 and DTS 5.1 configurations?
What?
Don't exist difference between Dolby 5.1 and DTS 5.1 or any 5.1 config, is always 3 front channels, 2 surround channels and LFE.
The Surround channels must be at +-120º from Central channel (more or less)

With 7.1 config the order in wav files is: FL FR FC LF BL BR SL SR
and mean:
BL BR = Back Channels at 150º (more or less)
SL SR = Side Channels at 90º (more or less)

Like you can see don't exist (in wav specs) a exact correspondence with Surround channels in 5.1 (120º)
Before XP SP2 the FL FR FC LF BL BR maskchannels was used for 5.1, after the FL FR FC LF SL SR was recommended but the previous one is also accepted.

[tebasuna51]

TMT 3 has version 1.1.0.1 of "dtsdecoderdll.dll"
TMT 3.5 has version 1.1.0.10 of "dtsdecoderdll.dll"

have you tried this new versions?

Nope, I'm not very interested in 7.1 DTS-MA, I only have 5.1 systems and the 'core' DTS (or AC3) is enough for my old ears.

Sorry.

[rickardk]

TMT 3 has version 1.1.0.1 of "dtsdecoderdll.dll"
TMT 3.5 has version 1.1.0.10 of "dtsdecoderdll.dll"

have you tried this new versions?

the latest 1.1.0.10 doesn't work with eac3to though. anyway, it's interesting if they fixed the problem with DTS-HD MA tracks reported by eac3to as "7.1 (strange setup)". as you know such tracks are not bit-perfect decoded by old 1.1.0.0 versions of "dtsdecoderdll.dll"

Maybe the new version can decode titles like Black Hawk Down DTS-HD MA 7.1 without the static noise that is introduced when decoded now.

[Thunderbolt8]

for me the 5.1 LPCM track of the US disc sounds better anyway, maybe its just a crappy made 7.1 track