eac3to - audio conversion tool

[n0mag!c]

Originally Posted by madshi

-quality=high slowdown/speedup/resampling quality (low/high/ultra)

eac3to.exe pal.ac3 z:ntsc.ac3 -slowdown -quality=ultra -448
Command line parameter "quality=ultra" is unknown.

I guess 1st message description is outdated?
(And option "-r8brain" is correct now?)

[nautilus7]

Yes: http://forum.doom9.org/showthread.ph...16#post1211516

[rickardk]

Can the final filesize of a FLAC track created by eac3to tell something about the quality?

I just compared two tracks from two diffrent releases of a movie. The FLAC created from the LPCM track is about 0.9 GB in size. And the FLAC created from the DTS-HD High Resolution track is about 2.2 GB.

Number of channels, bit depth and sample rate is the same. (The length of the LPCM track is about one minute shorter. Shorter credits)

[nautilus7]

Most probably you're mistaken about bitdepth. Which movie is this? And which releases? Can you post the logfiles?

Generally, bigger size, better quality.

[rickardk]

Originally Posted by nautilus7

Most probably you're mistaken about bitdepth. Which movie is this? And which releases? Can you post the logfiles?

Generally, bigger size, better quality.

The Sixth Sense. US vs. German.

I'm pretty sure eac3to said that the DTS-HD High Resolution track was 16 bit before decoding. But after looking at the spec of the created FLAC I see that it's actually 24 bit.

[Thunderbolt8]

ive also converted these 2 and can say that the sizes are correct. afaik DTS-HD gets patched to 24-bit for some reason, thus the size increase. the LPCM track is only 16-bit, thats why it is so small.

[nautilus7]

DTS(-HD) dynamic range is equivalent of 24bits, thus patching bitdepth. If you don't want that you can use the undocumented switch -dontPatchDts.

[gregt]

Anyone having problems with this title? I get the following:
N:\>eac3to BATTLE_PLANET_APES
1) 00013.mpls, 1:36:26
[65+54+68+56+70+57+72+58+74+59+76+60+78+61+80+62+82+63+84+64+86+87].m2ts
- h264/AVC, 1080p24 /1.001 (16:9)
- DTS Master Audio, English, multi-channel, 48khz
- AC3, English, stereo, 48khz
- AC3, Spanish, stereo, 48khz
- AC3, French, multi-channel, 48khz
- DTS Master Audio, English, multi-channel, 48khz

2) 00016.mpls, 1:26:32
[65+66+68+69+70+71+72+73+74+75+76+77+78+79+80+81+82+83+84+85+86+87].m2ts
- h264/AVC, 1080p24 /1.001 (16:9)
- DTS Master Audio, English, multi-channel, 48khz
- AC3, English, stereo, 48khz
- AC3, Spanish, stereo, 48khz
- AC3, French, multi-channel, 48khz
- DTS Master Audio, English, multi-channel, 48khz

3) 00018.mpls, 00046.m2ts, 0:16:34
- h264/AVC, 1080p24 /1.001 (16:9)
- AC3, English, stereo, 48khz

N:\>eac3to BATTLE_PLANET_APES 1)
This TS/M2TS file seems to be damaged (sync byte missing).
This TS/M2TS file seems to be damaged (sync byte missing).
The format of the source file could not be detected.

N:\>eac3to BATTLE_PLANET_APES 1) 2: BATTLE_PLANET_APES.mkv 3: BATTLE_PLANET_APES
.flac
This TS/M2TS file seems to be damaged (sync byte missing).
This TS/M2TS file seems to be damaged (sync byte missing).
The format of the source file could not be detected.

N:\>eac3to BATTLE_PLANET_APES 2)
This TS/M2TS file seems to be damaged (sync byte missing).
This TS/M2TS file seems to be damaged (sync byte missing).
The format of the source file could not be detected.

I did not have any problems with the other Planet of the Apes discs. I am using v2.74.

[woah!]

as all the ape films have new BD+ protection which hasnt be solved yet, i suppose it is an issue with how you extracted the m2ts files. have you seen the results of the other ape films you say worked ok?

[itsancho]

hi all, and madshi, 10x again for your great work!
well, this i think is strange, at least for me. German Blu-ray, the movie is Tais-toi!

eac3to v2.75
command line: z\eac3to "K:\Tais-toi! (2003) (Ruby & Quentin) (DE)Blu-ray VC-1 dts-HD Hi-Res" 1) 1: l:\Chapters.txt 2: l:\Tais.mkv 4: f:\Tais.dts -dontPatchDts 5: f:\DE.sup
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 1 subtitle track, 1:27:07
1: Chapters, 12 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: DTS Hi-Res, German, 5.1 channels, 16 bits, 3072kbps, 48khz
(core: DTS, 5.1 channels, 16 bits, 1536kbps, 48khz)
4: DTS Hi-Res, French, 5.1 channels, 16 bits, 1774kbps, 48khz
(core: DTS, 5.1 channels, 16 bits, 1536kbps, 48khz)
5: Subtitle (PGS), German
Creating file "l:\Chapters.txt"...
[a04] Extracting audio track number 4...
[s05] Extracting subtitle track number 5...
[v02] Extracting video track number 2...
[v02] Muxing video to Matroska...
[a04] Creating file "f:\Tais.dts"...
[s05] Creating file "f:\DE.sup"...
[a04] The last DTS frame is incomplete and thus gets skipped.
Added fps value to MKV header.
Video track 2 contains 125327 frames.
eac3to processing took 7 minutes, 46 seconds.
Done.

well, everything is OK, but the *.dts file is 2.34 GB and MediaInfo 0.7.7.8 (+ DTS: detection of DTS-HRA, DTS-MA, DTS-Express, thanks to http://madshi.net) reporting 3867 Kbps and also Haali Media Splitter...
sample: http://rapidshare.de/files/40920267/...mple_.dts.html

[madshi]

Originally Posted by n0mag!c

eac3to.exe pal.ac3 z:ntsc.ac3 -slowdown -quality=ultra -448
Command line parameter "quality=ultra" is unknown.

I guess 1st message description is outdated?
(And option "-r8brain" is correct now?)

Will update the first post. But I'm wondering: Is there a specific reason why you want to use r8brain instead of SSRC?

Originally Posted by rickardk

Can the final filesize of a FLAC track created by eac3to tell something about the quality?

It gives you a hint about the bitdepth. A higher bitdepth does not always have to mean that the quality is better. So you can not look at the file size alone. E.g. I'd consider a 16bit lossless track to be ever so slightly better than a DTS-HD High Resolution track (which is decoded by eac3to to 24bit).

Originally Posted by rickardk

I'm pretty sure eac3to said that the DTS-HD High Resolution track was 16 bit before decoding. But after looking at the spec of the created FLAC I see that it's actually 24 bit.

The DTS bitdepth reported by eac3to is a header field in the DTS track which tells us which bitdepth was fed into the DTS encoder. For lossy tracks this is just an information and has no use whatsoever for the decoder. So eac3to decodes to 24bit instead. For lossless tracks the situation is different, though...

Originally Posted by gregt

Anyone having problems with this title?

This could be either a ripping bug in AnyDVD HD or it could be an authoring fault. I'd suggest reporting this problem to the AnyDVD HD guys and see what they say. I'd say there is a good chance that this is a bug in AnyDVD HD, but I can't say for certain...

Originally Posted by itsancho

well, this i think is strange, at least for me. German Blu-ray, the movie is Tais-toi! well, everything is OK, but the *.dts file is 2.34 GB and MediaInfo 0.7.7.8 (+ DTS: detection of DTS-HRA, DTS-MA, DTS-Express, thanks to http://madshi.net) reporting 3867 Kbps...

This looks like a bug in eac3to. Will be fixed in the next build...

[odyssey]

I tried installing Nero 7 Showtime only from the Ultra package, but eac3to says "The Nero decoder doesn't seem to work, will use libav instead". How can I install the filter correctly? Better yet, can I install the filter without installing any Nero components? (such as extracting the nessesary files and registering them)

[odin24]

Originally Posted by odyssey

I tried installing Nero 7 Showtime only from the Ultra package, but eac3to says "The Nero decoder doesn't seem to work, will use libav instead". How can I install the filter correctly? Better yet, can I install the filter without installing any Nero components? (such as extracting the nessesary files and registering them)

You need the BD/HDDVD plugin installed too. I think you can have just Showtime installed and have eac3to work with it.

[odyssey]

Also if I'm just using it to transcode regular DVD AC3?

[odin24]

Originally Posted by odyssey

Also if I'm just using it to transcode regular DVD AC3?

Nero would be the preferable decoder, but libav does a good job too, which comes packaged with eac3to.

[nautilus7]

Originally Posted by odyssey

Also if I'm just using it to transcode regular DVD AC3?

You need the plugin so you can use the decoder. It's irrelevant of the source you're using.

[mochevolete]

Originally Posted by gregt

Anyone having problems with this title? I get the following:
N:\>eac3to BATTLE_PLANET_APES
1) 00013.mpls, 1:36:26
[65+54+68+56+70+57+72+58+74+59+76+60+78+61+80+62+82+63+84+64+86+87].m2ts
- h264/AVC, 1080p24 /1.001 (16:9)
- DTS Master Audio, English, multi-channel, 48khz
- AC3, English, stereo, 48khz
- AC3, Spanish, stereo, 48khz
- AC3, French, multi-channel, 48khz
- DTS Master Audio, English, multi-channel, 48khz

2) 00016.mpls, 1:26:32
[65+66+68+69+70+71+72+73+74+75+76+77+78+79+80+81+82+83+84+85+86+87].m2ts
- h264/AVC, 1080p24 /1.001 (16:9)
- DTS Master Audio, English, multi-channel, 48khz
- AC3, English, stereo, 48khz
- AC3, Spanish, stereo, 48khz
- AC3, French, multi-channel, 48khz
- DTS Master Audio, English, multi-channel, 48khz

3) 00018.mpls, 00046.m2ts, 0:16:34
- h264/AVC, 1080p24 /1.001 (16:9)
- AC3, English, stereo, 48khz

N:\>eac3to BATTLE_PLANET_APES 1)
This TS/M2TS file seems to be damaged (sync byte missing).
This TS/M2TS file seems to be damaged (sync byte missing).
The format of the source file could not be detected.

N:\>eac3to BATTLE_PLANET_APES 1) 2: BATTLE_PLANET_APES.mkv 3: BATTLE_PLANET_APES
.flac
This TS/M2TS file seems to be damaged (sync byte missing).
This TS/M2TS file seems to be damaged (sync byte missing).
The format of the source file could not be detected.

N:\>eac3to BATTLE_PLANET_APES 2)
This TS/M2TS file seems to be damaged (sync byte missing).
This TS/M2TS file seems to be damaged (sync byte missing).
The format of the source file could not be detected.

I did not have any problems with the other Planet of the Apes discs. I am using v2.74.

had the same problem with another BR,
the BR itself played flawless (on the HD), but eac3to gave me the same error.

Solved remuxing the BR into another BR using this simple guide:
http://forum.slysoft.com/showpost.ph...81&postcount=1

then eac3to managed to do the job

Nik

[madshi]

eac3to v2.76 released

http://madshi.net/eac3to.zip

* "-slowdown" now works to convert 24.000 movies to 23.976
* "-speedup" now works to convert 24.000 movies to 25.000
* option "-xx.xxx" (e.g. "-24.000") sets the FPS of the source track
* option "-changeToXx.xxx" (e.g. "-changeTo23.976") changes video/audio FPS
* modified FPS information is written to video bitstream (VC-1, MPEG2, h264)
* demuxing with FPS change option now activates audio track transcoding
* SSRC resampling parameters modified slightly to reduce steepness and ringing
* fixed incorrect h264 movie slowdown gap/overlap complaints
* fixed DTS-HD High Resolution bitrate calculation
* dithering is now done differently per channel

Let me explain the new FPS changing options:

(1) If you have a source which contains FPS information, you don't need to tell eac3to which FPS the source has. E.g. if you feed eac3to a m2ts, TS, EVO or VOB file with a video track in it, eac3to will know which FPS the video and audio tracks have. So in this case you can just use "-slowdown" to convert both 24.000 and 25.000 movies to 23.976 fps. Or you can use "-speedup" to convert both 23.976 and 24.000 movies to 25.000 fps.

(2) If eac3to does not know the source FPS (which is usually the case if you feed eac3to with a demuxed audio track), you can still use the "-slowdown" and "-speedup" options. If you do that, eac3to will apply 25.000 -> 23.976 (slowdown) or 23.976 -> 25.000 (speedup) conversion.

(3) If you want to do any funny conversions. E.g. if you want to convert a 25.000 source to 24.000, you can use the option "-changeTo24.000". If eac3to knows the source FPS, that's all you need to do. If eac3to doesn't know the source FPS, you should do "-25.000 -changeTo24.000" to do a 25.000 -> 24.000 conversion.

(4) Generally, if the source is a container which contains both video and audio tracks, doing either "eac3to source -demux -anyFpsChangeOptions" or "eac3to source some.mkv -anyFpsChangeOptions" will result in eac3to doing FPS conversion for all audio and video tracks. Audio tracks will be transcoded in this situation. Lossless tracks will be transcoded to 24bit FLAC. Lossy tracks will be transcoded to 640kbps AC3.

(5) Video FPS changes will as usual not only result in adjusted MKV timestamps. eac3to will also automatically adjust the video bitstream itself to reflect the FPS value change.

[AnryV]

Originally Posted by tebasuna51

You can use WaveWizard or Sox command line:

Sox -M FL.wav FR.wav FC.wav LF.wav SL.wav SR.wav multichannel.wav

How to create the 7.1 multichannel wav? What is the correct order of the channells/files ?

[Thunderbolt8]

Originally Posted by madshi

* fixed incorrect h264 movie slowdown gap/overlap complaints

thanks, will test and report back in case something should still be wrong!

btw. if I slow down an audio track with delay, whats done first: the delay applied and then the track is slowed or the other way round (the message order on screen is first the fps conversion and after that applying the delay)?

[tebasuna51]

Originally Posted by AnryV

How to create the 7.1 multichannel wav? What is the correct order of the channells/files ?

Sox -M FL.wav FR.wav FC.wav LF.wav BL.wav BR.wav SL.wav SR.wav multichannel.wav

[Thunderbolt8]

madshi, the robocop cap is still more than 3 mins too long (its 1h46min59sec, should be ~1h43min11sec; there's no gap file this time)

eac3to v2.76
command line: eac3to G:\robo.ts G:\robo.mkv -slowdown
------------------------------------------------------------------------------
TS, 1 video track, 1 audio track, 1:38:57
1: h264/AVC, 1080i50 (16:9)
2: AC3, English, 5.1 channels, 384kbps, 48khz, dialnorm: -27dB, 57ms
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[v01] Muxing video to Matroska...
[a02] Removing AC3 dialog normalization...
[a02] Decoding with DirectShow (Nero Audio Decoder 2)...
[a02] DirectShow reports 5.1 channels, 24 bits, 48khz
[a02] Changing FPS from 25.000 to 23.976...
[a02] Applying RAW/PCM delay...
[a02] Encoding AC3 <640kbps> with libAften...
[a02] Creating file "G:\robo - 2 - AC3, English, 5.1 channels, 384kbps, 48khz.ac3"...
[a02] The last (E-)AC3 frame is incomplete and thus gets skipped.
24000/1001
Added fps value to MKV header.
Video track 1 contains 296870 frames.
eac3to processing took 23 minutes, 48 seconds.
Done.

the other cap:

eac3to v2.76
command line: eac3to G:\cube.ts 1: G:\cube.mkv -slowdown          [audio needs to be fixed with delaycut]
------------------------------------------------------------------------------
TS, 1 video track, 2 audio tracks, 1:26:58
1: h264/AVC, 1080i50 (16:9)
2: AC3, English, 5.1 channels, 384kbps, 48khz, dialnorm: -27dB, 86ms
3: MP2, English, 2.0 channels, 256kbps, 48khz, 83ms
[v01] Extracting video track number 1...
[v01] Muxing video to Matroska...
24000/1001
Adding fps value to MKV header failed.
Video track 1 contains 260926 frames.
eac3to processing took 10 minutes, 16 seconds.
Done.

should be 1h30min41s (is 1h30min36s), but have to test with the audio first. maybe its allright.

[rica]

Originally Posted by madshi

eac3to v2.76 released

http://madshi.net/eac3to.zip

Another revolutionary update, thanks.

_ _ _

[madshi]

Originally Posted by Thunderbolt8

btw. if I slow down an audio track with delay, whats done first: the delay applied and then the track is slowed or the other way round?

Good question. Just checked: Resampling/slowdown is done *before* delay, which I think is wrong. Will change that in the next build.

Originally Posted by Thunderbolt8

the robocop cap is still more than 3 mins too long (its 1h46min59sec, should be ~1h43min11sec; there's no gap file this time)

Hmmmm... The sample you uploaded is probably too short for me to reproduce the problem, right? Can you please PM me for more details on that movie source?

Originally Posted by rica

Another revolutionary update, thanks.

Pleasure!

[n0mag!c]

Originally Posted by madshi

I'm wondering: Is there a specific reason why you want to use r8brain instead of SSRC?

This is first time I'm using eac3to to slowdown/speedup audio (usually I use SForge for this) and I just wanted to compare speed of these two methods (SSRC and r8brain).

[Thunderbolt8]

Originally Posted by Thunderbolt8

should be 1h30min41s (is 1h30min36s), but have to test with the audio first. maybe its allright.

nope, its unfortunately not. seems to vary a little, at the beginning its at least closer to being correct as towards the end.

for those 2 clips it always seems to be the video which is wrong, when you count the number of seconds for original 25fps audio for both movies and then calculate it to 23.976 fps, then the slowed down audio track seem to be fine.

tested the robocop without -slowdown, and its already wrong then, the video duration:

1:38:57 original .ts length
1:42:37 remuxed .mkv without -slowdown
1:43:11 supposed video length for -slowdown
1:46:XX video length with -slowdown


same for cube without -slowdown:

1:26:58 original .ts length
1:26:53 remuxed .mkv without -slowdown
1:30:41 supposed video length for -slowdown
1:30:36 video length with -slowdown

at least here it seems the difference of 5 seconds seems to be stable after the slowdown process.

[shambles]

Originally Posted by madshi

eac3to v2.76 released

thank thank thank you!

it's amazing how one little app can make so many things so much easier.. you are really spoiling us, madshi

[gillie]

Originally Posted by madshi

eac3to v2.76 released

http://madshi.net/eac3to.zip

* "-slowdown" now works to convert 24.000 movies to 23.976
* "-speedup" now works to convert 24.000 movies to 25.000
* option "-xx.xxx" (e.g. "-24.000") sets the FPS of the source track
* option "-changeToXx.xxx" (e.g. "-changeTo23.976") changes video/audio FPS
* modified FPS information is written to video bitstream (VC-1, MPEG2, h264)
* demuxing with FPS change option now activates audio track transcoding
* SSRC resampling parameters modified slightly to reduce steepness and ringing
* fixed incorrect h264 movie slowdown gap/overlap complaints
* fixed DTS-HD High Resolution bitrate calculation
* dithering is now done differently per channel

Let me explain the new FPS changing options:

(1) If you have a source which contains FPS information, you don't need to tell eac3to which FPS the source has. E.g. if you feed eac3to a m2ts, TS, EVO or VOB file with a video track in it, eac3to will know which FPS the video and audio tracks have. So in this case you can just use "-slowdown" to convert both 24.000 and 25.000 movies to 23.976 fps. Or you can use "-speedup" to convert both 23.976 and 24.000 movies to 25.000 fps.

(2) If eac3to does not know the source FPS (which is usually the case if you feed eac3to with a demuxed audio track), you can still use the "-slowdown" and "-speedup" options. If you do that, eac3to will apply 25.000 -> 23.976 (slowdown) or 23.976 -> 25.000 (speedup) conversion.

(3) If you want to do any funny conversions. E.g. if you want to convert a 25.000 source to 24.000, you can use the option "-changeTo24.000". If eac3to knows the source FPS, that's all you need to do. If eac3to doesn't know the source FPS, you should do "-25.000 -changeTo24.000" to do a 25.000 -> 24.000 conversion.

(4) Generally, if the source is a container which contains both video and audio tracks, doing either "eac3to source -demux -anyFpsChangeOptions" or "eac3to source some.mkv -anyFpsChangeOptions" will result in eac3to doing FPS conversion for all audio and video tracks. Audio tracks will be transcoded in this situation. Lossless tracks will be transcoded to 24bit FLAC. Lossy tracks will be transcoded to 640kbps AC3.

(5) Video FPS changes will as usual not only result in adjusted MKV timestamps. eac3to will also automatically adjust the video bitstream itself to reflect the FPS value change.

Appreciate this might be a little of the beaten track but just wondering whether the -slowdown or -speedup settings can be used to help produce .ts files which will auto-trigger 24p settings on a TViX player.

I use eac3to to demux my Blu-ray disks and have to say it works brilliantly. However, if I mux video and audio streams into an .mkv file my TViX player autoswitches into 24p mode. If I mux exactly the same video and audio streams into a .ts file it doesn't, unless the video source is MPEG2 which most recent Blu-rays aren't.

The reason I want to use the .ts format is that the TViX plays this format well as opposed to .mkv's which have video stuttering and audio dropouts.

Is there a "header" in the .ts container that can force 24p mode? Can eac3to create that header?

[AnryV]

Originally Posted by tebasuna51

Sox -M FL.wav FR.wav FC.wav LF.wav BL.wav BR.wav SL.wav SR.wav multichannel.wav

Thanks!

[madshi]

Originally Posted by Thunderbolt8

nope, its unfortunately not. seems to vary a little, at the beginning its at least closer to being correct as towards the end.

for those 2 clips it always seems to be the video which is wrong, when you count the number of seconds for original 25fps audio for both movies and then calculate it to 23.976 fps, then the slowed down audio track seem to be fine.

tested the robocop without -slowdown, and its already wrong then, the video duration:

1:38:57 original .ts length
1:42:37 remuxed .mkv without -slowdown
1:43:11 supposed video length for -slowdown
1:46:XX video length with -slowdown


same for cube without -slowdown:

1:26:58 original .ts length
1:26:53 remuxed .mkv without -slowdown
1:30:41 supposed video length for -slowdown
1:30:36 video length with -slowdown

at least here it seems the difference of 5 seconds seems to be stable after the slowdown process.

Ok, thanks, will see what I can do - next week.

Originally Posted by gillie

Appreciate this might be a little of the beaten track but just wondering whether the -slowdown or -speedup settings can be used to help produce .ts files which will auto-trigger 24p settings on a TViX player.

I use eac3to to demux my Blu-ray disks and have to say it works brilliantly. However, if I mux video and audio streams into an .mkv file my TViX player autoswitches into 24p mode. If I mux exactly the same video and audio streams into a .ts file it doesn't, unless the video source is MPEG2 which most recent Blu-rays aren't.

The reason I want to use the .ts format is that the TViX plays this format well as opposed to .mkv's which have video stuttering and audio dropouts.

Is there a "header" in the .ts container that can force 24p mode? Can eac3to create that header?

There's a "FPS" like field in the MKV header. But for TS files there's no such field (at least as far as I'm aware). Still the TViX could detect the right framerate by looking at the video bitstream. It seems that it does that correctly for MPEG2, but not for VC-1 and h264. There's nothing I can do about that. You'll have to ask the TViX guys to add support for that. It's not too difficult to realize. If they need help about how to parse the VC-1 and h264 bitstream, you can tell them to contact me...

[madshi]

eac3to v2.77 released

http://madshi.net/eac3to.zip

* pcm/raw audio delay is now applied before resampling and fps change
* parsing of command line with multiple sources files sometimes failed

[odyssey]

Originally Posted by nautilus7

You need the plugin so you can use the decoder. It's irrelevant of the source you're using.

No it's not. odin24 claims that I need BD/HDDVD plugin installed in order to decode AC3 audio, but It's logical to think that it's not needed when it's JUST for old AC3, not EAC3!

Also, noone answered why the filter didn't work - I already installed Nero Showtime 7, why would it not work?

[nautilus7]

I suggest reading the answers you 're given more carefully.

The 1st post says that you need nero 7 and the plugin. That's all it says. And that should be enough to understand that you need both. Period. You need both so you can use the nero decoder. What you will do it doesn't say much to eac3to.

As additional info, you can find (by searching this trhead or the wiki) that you need only showtime from the whole nero package and you also need nero 7.8.5.0 or later nero 7. The plugin is necessary in any case.

[odyssey]

Originally Posted by nautilus7

I suggest reading the answers you 're given more carefully.

I suggest someone updates the post so new users like me would even KNOW that a wiki existed!!!

[techouse]

Whoa! Thanx for he super update

[madshi]

Originally Posted by odyssey

I suggest someone updates the post so new users like me would even KNOW that a wiki existed!!!

I suggest that you stop flooding this thread with comments about a question which has already been answered in 3 different prominent places:

(1) The very first post in this thread.
(2) The eac3to help text.
(3) Several direct user replies to your question.

All of which say the same (correct) thing.

[Chumbo]

@madshi,
I was wondering if you could provide logging at the play list level please? When running eac3to on a folder which lists the play lists, the -log does not work. Once you use -log, it processes similar to "eac3to bdmv" rather than "eac3to folder" which is a bummer. I can pipe the output of the play list to a file, but it's back to having to clean up those files from all the extra CRLFs.

Also, one other request if you have time. When I'm running an instance of eac3to that's extracting or whatever and then run another instance just to log and get info about another title, the -log doesn't work. It seems to just copy what's already there from the first instance. It would really be helpful if the logging is done independently in each eac3to command. Thank you so much.

[himan2001]

@madshi:

New Bug-Report for the "-core"-command:

Situation: DTS-MA 7.1 File with embedded 5.1 core, and ArcSoft-Decoder:

Parameters: eac3to <src.m2ts> audio.ac3 -448 -core

Brings this error:

eac3to v2.77
command line: eac3to 00007.m2ts 3: audio.ac3 -448 -core
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 5 subtitle tracks, 1:56:07
1: Chapters, 20 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 7.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
4: DTS, English, 2.0 channels, 24 bits, 255kbps, 48khz
5: DTS, English, 2.0 channels, 24 bits, 255kbps, 48khz
6: DTS Master Audio, German, 5.1 channels, 24 bits, 48khz, dialnorm: -1dB
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz, dialnorm: -1dB)
7: Subtitle (PGS), English
8: Subtitle (PGS), German
9: Subtitle (PGS), German
10: Subtitle (PGS), English
11: Subtitle (PGS), German
[a03] Extracting audio track number 3...
[a03] Extracting DTS core...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] The AC3 encoder received a non-supported data format.
Aborted at file position 32768.


after adding -sonic, all was fine!

After your fixx using the 5.1 from the "-core", the arcsoft-decoder is "internaly" still "feeded" with the 7.1 information, i asume.

So the "-core" Option doesn´t work with arcsoft-decoder, only with Sonic at the moment.

[nautilus7]

Sonic can't output 7.1 (is limited to 5.1 in this case) that's why it works.

[himan2001]

i know it, nautilus

but is still internaly false reported to the arcsoft

[73ChargerFan]

Originally Posted by nautilus7

As additional info, you can find (by searching this trhead or the wiki) ...

Madshi,
I didn't even remember there was a wiki under development.
Would you please add a link to it in the first post?

[rica]

Originally Posted by 73ChargerFan

Madshi,
I didn't even remember there was a wiki under development.
Would you please add a link to it in the first post?

At least for now:

http://en.wikibooks.org/wiki/Eac3to/How_to_Use

(by nautilus7)

[nautilus7]

Wikibooks is a place where everyone can contribute. The fact that i started it, doesn't make it my property. Plus i haven't much time lately so it's up to other eac3to users to continue the development, so it doesn't stay "under construction" for a life.

[odyssey]

Originally Posted by madshi

I suggest that you stop flooding this thread with comments about a question which has already been answered in 3 different prominent places:

(1) The very first post in this thread.
(2) The eac3to help text.
(3) Several direct user replies to your question.

All of which say the same (correct) thing.

Well apprearently it's not clear enough.

Also you need to update the wiki links (they are broken). I have searched all over the net, but yet to find the plugin package you all talk about.

[sehgal.v7]

@Madshi

It took some time but finally here it is.. upload first 4mins of movie Iron Man..

_http://rapidshare.com/files/164894642/1_-_DTS_Master_Audio__English__5.1_channels__16_bits__48khz.dtsma
_http://rapidshare.com/files/164889007/2_-_TrueHD_AC3__English__5.1_channels__48khz.thd

Regards.
& lemme knoe if u found noticeable diff.

[nautilus7]

Originally Posted by odyssey

Well apprearently it's not clear enough.

Also you need to update the wiki links (they are broken). I have searched all over the net, but yet to find the plug-in package you all talk about.

Can you find anywhere in Nero site some info regarding how to buy nero 7? No, because both nero 7 and the plug in are discontinued now (the plug-in is included in nero 9 actually).

[nwg]

Is there any difference in a AC3 file created from a DTA-HD MA 7.1 and a 5.1?

I have so far created a AC3 5.1 640kbps file straight from the DT-HD MA 5.1. Now with a DTHD MA 7.1 am I better to extract the DTS core and create a AC3 5.1 from that? I only ask as I have just "AC3 encoding does not suport back channels, will mix them into the surround". Will that make the surrounds different?

[nautilus7]

One thing is the extra channels.
But another more important is that in one case you use the master audio tracks as source (lossless) and in the other you use just the lossy dts core.

[bigotti5]

Seems there is a bug in eac3to.
Tried to concatenate some mts files from my Sony AVCHD camera.
Video presentation order:
BBIBPBPBPBPBPBBIBPBP...
Stream order:
IBBPBPBPBPBPBIBBPBPB...
Two leading B-frames encoded using only backward prediction misleads eac3to to assume -80 ms audio delay.
First audio PTS: 93600 - 00:00:01:040
First I-Frame PTS (first frame stored but third frame shown): 100800 - 00:00:01:120
First B-Frame PTS (the first frame shown): 93600 - 00:00:01:040

You can dl an example here

[nwg]

Originally Posted by nautilus7

One thing is the extra channels.
But another more important is that in one case you use the master audio tracks as source (lossless) and in the other you use just the lossy dts core.

I suppose my point is are these extra channels going to cause audio problems for the rears as the channels don't exist.

[nautilus7]

What kind of problems? I don't think so.

[Thunderbolt8]

does normal DTS or DTS-ES have the same frame length like DTS-HD (MA) = 10.666ms ?

[nautilus7]

Yes, afaik.

[odin24]

Is there a similar command to view the properties of TrueHD track as there is for DTS (-logdts)?

[Nullity]

Could someone please describe the process of automatically applying an audio delay? I have an HDDVD with an EAC3 track with a delay of -84ms. From what I understand, the delay cannot be corrected exactly. I am not familiar with all the terminology, but is it correct that the audio can only be corrected in 32ms "chunks"? If so, how then would my example track be fixed? Would it apply a +64ms delay, leaving -20ms, or would it apply a +96ms delay, changing it to +12ms?

Also, is there a switch to skip the automatic delay removal, so that I can handle it on my own later (I did not see one listed on the wiki)? If not, could this option please be added?

[Thunderbolt8]

Originally Posted by Nullity

Could someone please describe the process of automatically applying an audio delay? I have an HDDVD with an EAC3 track with a delay of -84ms. From what I understand, the delay cannot be corrected exactly. I am not familiar with all the terminology, but is it correct that the audio can only be corrected in 32ms "chunks"? If so, how then would my example track be fixed? Would it apply a +64ms delay, leaving -20ms, or would it apply a +96ms delay, changing it to +12ms?

Also, is there a switch to skip the automatic delay removal, so that I can handle it on my own later (I did not see one listed on the wiki)? If not, could this option please be added?

it applies the delay which leaves the closest gap to the needed value, so +96ms in your case, leaving only +12ms rest delay. dont ask me what happens with values like 15m-17ms though, in which direction its applied then. (maybe test it with mediainfo and compare before after)

[Joniii]

I just demuxed Interview With The Vampire Blu-ray with eac3to v2.77. It has VC1 and AC3 5.1 640kbps + some subs. I demuxed all tracks with "eac3to X:\iwv.m2ts -demux". Then remuxed audio and video back to m2ts with TSMuxer. Now there is really low sound volume with remuxed m2ts (I need to tweak the volume in my Yamaha receiver nearly to max so that I hear any sound, with original m2ts sound level is fine). Anyone know whats causing this?

[BlackJack1]

Sorry for n00b question but if I've got ac3 DVD audio track DL from internet but with dialog normalization present can I remove it using this tool?
What command (option) must I use...?
Thx in advance.

[Momber]

Originally Posted by odyssey

I suggest someone updates the post so new users like me would even KNOW that a wiki existed!!!

I agree. This is information that would be well placed in post no. 1 of this thread, actually.

Well, now that I know a wiki exists I have looked at it and found it very helpful. Thanks to everyone involved in creating it!

[Chumbo]

Originally Posted by BlackJack1

Sorry for n00b question but if I've got ac3 DVD audio track DL from internet but with dialog normalization present can I remove it using this tool?
What command (option) must I use...?
Thx in advance.

Did you even try to read the available switches that are clearly listed on page 1 or by just running eac3to w/out any parameters?

-keepDialnorm     disables dialog normalization removal (not recommended)

Hmmm, I wonder what not using this command means?

I would guess "eac3to source.ac3 target.ac3" would remove it.

[odin24]

Guys, I need some advice. I have a demuxed BD that would be slightly larger than a SL BD-RE, my playback device is my PS3. I could recode the video to fit with the THD track on a BD-RE, which would consume a considerable amount of time, or I could convert the THD track to PCM and play as m2ts from the HDD.

My concern is if I convert the audio, will the PCM file be as close to a replica of the TrueHD track as possible... aside for the CBR and obvioulsy different file type? This would be the first time I convert the audio as I usually process the video with good results. Can anybody vouch for the quality of the audio conversion.

Thanks.

[rica]

odin,
think you unzip a file;
here, zipped file is lossless (THD) and unzipped file is uncompressed (pcm)

[odin24]

Originally Posted by rica

odin,
think you unzip a file;
here, zipped file is lossless (THD) and unzipped file is uncompressed (pcm)

I get that a TrueHD track is a lossless compressed version of the studio track, but I'm not sure how eac3to goes about creating the new PCM file. Does it just "unzip" as you say creating a replicate of the studio track, or does it write a completely new version of the decoded TrueHD track.

Thanks again.

[Snowknight26]

When you go from lossless to lossless, say in this case, the first stream is decoded to RAW audio, then written to a file.

[odin24]

I'm talking about the TrueHD track as the source file, not the BD structure or m2ts.

I just want to know, how true the newly created PCM file by eac3to will be to the BD TrueHD file... or is it worth it to just keep the TrueHD track and recode my video instead.

[rica]

Originally Posted by odin24

I get that a TrueHD track is a lossless compressed version of the studio track, but I'm not sure how eac3to goes about creating the new PCM file. Does it just "unzip" as you say creating a replicate of the studio track, or does it write a completely new version of the decoded TrueHD track.

Thanks again.

OK, i may explain in this way:
think you have a thd decoder as default on your PC; say it is arcsoft audio decoder hd.
If you play an original THD with this decoder it is an unzip processs.
If you decode and re-encode to pcm with your default decoder, the final file can be considered as unzipped .
But in our case libav will be used for decoding in re-encoding process.
So it should be considered a rewrite of the original file.

[odin24]

Originally Posted by rica

OK, i can explain this in this way:
think you have a thd decoder as default on your PC; say it is arcsoft audio decoder hd.
If you play an original THD with this decoder it is an unzip processs.
If you decode and re-encode to pcm with your default decoder to pcm, the final file can be considered as unzipped .
But in our case libav will be used for decoding in re-encoding process.
So it should be considered a rewrite of the original file.

Great, thanks. That's exactly what I needed to know.

Thanks again.

[Steel]

Originally Posted by tebasuna51

Sox -M FL.wav FR.wav FC.wav LF.wav BL.wav BR.wav SL.wav SR.wav multichannel.wav

what app is SOX?

[tebasuna51]

Originally Posted by Steel

what app is SOX?

This is the Sox Homepage

[Thunderbolt8]

got a problem with converting a 2.0 ac3 track to .wavs: both outcoming wav files are slowed massively down, meaning all sounds and voices are played like ultra slowmotion.
can't say though if this is a specific problem of this single file or a general problem of 2.0 ac3 tracks. only had a 5.1 ac3 track to compare and that one played normaly after wave conversion.

edit: just noticed the track is from a mpeg2 cap and according to mpegrepair 5.1, while eac3to only recognizes it as 2.0

20mb sample of the ac3 track: http://www.sendspace.com/file/nbry8n

50mb sample of the .ts: http://www.sendspace.com/file/03nqcj

[dant3s]

Hello Madshi,
There is a little bug when using 2 separated threads of EAC3to, generated logs are the same, even if the source file is different and is located in a different folder.
Log is nicely generated in the prompt window but not in the text log, think my text is clear...
dant3s

[tebasuna51]

Originally Posted by Thunderbolt8

got a problem with converting a 2.0 ac3 track to .wavs: both outcoming wav files are slowed massively down, meaning all sounds and voices are played like ultra slowmotion....

Your answer in DelayCut thread ...

[itsancho]

hi all!
madshi, thank you for the great work!
well, is this a bug or it's nothing serious?

eac3to v2.77
command line: z\eac3to "E:\Music and Lyrics (2007) Blu-ray VC-1 DD" 1) 1: f:\Chapters.txt 2: f:\Music.mkv 3: a:\Music.ac3 9: a:\Music.sup 10: a:\Music1.sup
------------------------------------------------------------------------------
M2TS, 1 video track, 6 audio tracks, 14 subtitle tracks, 1:44:07
1: Chapters, 25 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
...
9: Subtitle (PGS), English
10: Subtitle (PGS), English
...
22: Subtitle (PGS), Swedish
Creating file "f:\Chapters.txt"...
[v02] Extracting video track number 2...
[s09] Extracting subtitle track number 9...
[a03] Extracting audio track number 3...
[s10] Extracting subtitle track number 10...
[a03] Removing AC3 dialog normalization...
[v02] Muxing video to Matroska...
[a03] Creating file "a:\Music.ac3"...
[s10] Creating file "a:\Music1.sup"...
[s09] Creating file "a:\Music.sup"...
24000/1001
Adding fps value to MKV header failed.
Video track 2 contains 149775 frames.
eac3to processing took 11 minutes, 25 seconds.
Done.

with 2.76 is the same, but with 2.75...

eac3to v2.75
command line: zzz\eac3to "E:\Music and Lyrics (2007) Blu-ray VC-1 DD" 1) 2: f:\Music.mkv
------------------------------------------------------------------------------
M2TS, 1 video track, 6 audio tracks, 14 subtitle tracks, 1:44:07
1: Chapters, 25 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
...
9: Subtitle (PGS), English
10: Subtitle (PGS), English
...
22: Subtitle (PGS), Swedish
[v02] Extracting video track number 2...
[v02] Muxing video to Matroska...
Added fps value to MKV header.
Video track 2 contains 149775 frames.
eac3to processing took 9 minutes, 23 seconds.
Done.

[Daodan]

I noticed latest version makes monowavs from 5.1 to BL and BR instead of old style SL and SR.
Now my question is, in case of 7.1 track, which are the extra 2 (compared to 5.1), SB,SL or BR,BL?
Thank you.

[tebasuna51]

Originally Posted by Daodan

Now my question is, in case of 7.1 track, which are the extra 2 (compared to 5.1), SB,SL or BR,BL?

Your question have not sense, the surround channels in 5.1 aren't the Side or Back channels in 7.1, maybe a mix.

[Thunderbolt8]

Originally Posted by tebasuna51

Your answer in DelayCut thread ...

thanks

[Daodan]

Originally Posted by tebasuna51

Your question have not sense, the surround channels in 5.1 aren't the Side or Back channels in 7.1, maybe a mix.

I don't see what doesn't make sense. It's the way they are labeled by eac3to and I want to now what is what. Following channels were created: L,R,C,LFE,SL,SR,BL,BR

[Snowknight26]

Any chances of speed increases for demuxing? Just demuxed the PCM track and an AC3->WAV track from A Few Good Men Blu-ray on a RAID array that can sustain 700MB/s, yet eac3to purportedly took 20 minutes.

[Chumbo]

Originally Posted by Daodan

I don't see what doesn't make sense. It's the way they are labeled by eac3to and I want to now what is what. Following channels were created: L,R,C,LFE,SL,SR,BL,BR

Your question doesn't make sense. It's like asking which one of the stereo channels is used for the mono track, Left or Right? It's how it's mixed. A mono recording would still contain all the info, but in one channel.

It doesn't matter what they're labeled. You only have 2 rear channels in 5.1 which will include the entire surround mix.

[Daodan]

But those channels are from somewhere, they didn't drop from the sky. They are from a 7.1 track and I want to know what label correspons to what channel. Just like you know the LFE or C mean, I want to know what's the difference between SR and BR.

[rica]

Originally Posted by Daodan

But those channels are from somewhere, they didn't drop from the sky. They are from a 7.1 track and I want to know what label correspons to what channel. Just like you know the LFE or C mean, I want to know what's the difference between SR and BR.

SR= Surround Right
BR= Back Right

[Daodan]

Lovely. But which of them corresponds to the extra channels from 7.1 vs 5.1?

[Thunderbolt8]

got a DTS 5.1 track here, eac3to reports it as DTS 5.1, while PowerDVD says it's DTS-ES 5.1

so which is right in this case?

http://www.sendspace.com/file/79229m

[odin24]

Originally Posted by Thunderbolt8

got a DTS 5.1 track here, eac3to reports it as DTS 5.1, while PowerDVD says it's DTS-ES 5.1

so which is right in this case?

http://www.sendspace.com/file/79229m

Just DTS, possibly with the ES extension stripped?

eac3to v2.72
command line: e3\eac3to  c:\videos\test\dts sample.dts c:\videos\test\log.txt -logdts
------------------------------------------------------------------------------
+ DTS-Core
  - frameSize            1006
  - DTS-ES               -
  - channelNo            5
  - lfe                  1
  - channelDescr         5.1
  - samplingRate         48000
  - bitDepth             24
  - bitrate              768000
  - samplesPerFrame      512
  - copyHistory          1
DTS, 5.1 channels, 0:09:16, 24 bits, 768kbps, 48khz

[rica]

Originally Posted by Daodan

Lovely. But which of them corresponds to the extra channels from 7.1 vs 5.1?

Rear (or back) channels in 5.1 are shared to back and surround in 7.1.

[Daodan]

So you say both have the same content? (SR and BR)
I analized the peaks and they are very slightly different.

[rica]

Originally Posted by Daodan

So you say both have the same content? (SR and BR)
I analized the peaks and they are very slightly different.

I did not say this.

[lithiumus]

I just got the Wall E Blu-ray and it's supposed to have DTS-HD Master Audio 6.1 but eac3to shows only 5.1.

I was wondering what the "($f)" is under Active speakers.

Edit: let me clarify, the box says 5.1 DTS MA but blu-ray.com says 6.1 DTS-MA and several reviews have indicated 6.1 so maybe I'm just reaching for something that just isn't there...

D:\Program Files\eac3to>eac3to.exe "c:\Audio Testing\20000.m2ts" -logdts
+ DTS-Core
  - frameSize            2012
  - DTS-ES               +
  - channelNo            5
  - lfe                  1
  - channelDescr         5.1
  - samplingRate         48000
  - bitDepth             24
  - bitrate              1509000
  - samplesPerFrame      512
  - copyHistory          1
+ DTS-HD
  - fullSize             2776
  - headerSize           32
  - refClockCode         1/48000
  - frameDurationCode    1
  - activeMasks          [1], [[1]]
  + Asset [0]
    - fullSize             2744
    - headerSize           14
    - corePackets          Core
    - extSubStrPackets     XLL
    - bitResolution        24
    - maxSampleRate        48000
    - totalNumChannels     6
    - activeSpeakers       C L R Ls Rs LFE ($f)
M2TS, 3 video tracks, 3 audio tracks, 3 subtitle tracks, 1:37:26
1: Chapters, 32 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-ES, 5.1 channels, 24 bits, 1509kbps, 48khz)
6: AC3 Surround, English, 2.0 channels, 192kbps, 48khz
7: AC3 Surround, English, 2.0 channels, 192kbps, 48khz
8: Subtitle (PGS), English
9: Subtitle (PGS), English
10: Subtitle (PGS), English

D:\Program Files\eac3to>

[shanghai2004]

Originally Posted by madshi

eac3to v2.76 released

http://madshi.net/eac3to.zip

.......
* SSRC resampling parameters modified slightly to reduce steepness and ringing
.........
* dithering is now done differently per channel

Madshi, those changes are large enough to make it useful to process my audio files again (96/24 -> 48/16)?

[hubblec4]

hi madshi

when i encode an audio stream (e.g. audio.thd+ac3) to dts, in first the audio stream will demux to 6 wavs and after dts encoding the 6 wavs were deleted.

its possible to keep the 6wavs?

hubble

[sehgal.v7]

yeap..
eac3to audio.thd+ac3 audio.wavs

[Chumbo]

Originally Posted by hubblec4

hi madshi

when i encode an audio stream (e.g. audio.thd+ac3) to dts, in first the audio stream will demux to 6 wavs and after dts encoding the 6 wavs were deleted.

its possible to keep the 6wavs?

hubble

Another "trick" is to set the file flag to Read-Only AFTER the dts conversion starts as the wave files will be complete.

If you just need the wave files, then you can just do as already suggested and convert directly to waves.

[Thunderbolt8]

madshi, could you please implement a kind of detection which reports 2.0 tracks being mono or stereo tracks? would be useful for older movies, sometimes those original tracks are either 1.0 or 2.0 mono. then it would be possible to distinguish if a studio included an original 2.0 mono track or if they did use a modified stero track.

[rickardk]

F:\>eac3to sin.m2ts 1: e:\sin.mkv 3: e:\sin.flac
M2TS, 1 video track, 2 audio tracks, 4 subtitle tracks, 2:03:59
1: h264/AVC, 1080p24 /1.001 (16:9)
2: AC3, 5.1 channels, 640kbps, 48khz
3: DTS Master Audio, 7.1 (strange setup) channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1536kbps, 48khz)
4: Subtitle (PGS)
5: Subtitle (PGS)
6: Subtitle (PGS)
7: Subtitle (PGS)
CAUTION: Decoding this track with ArcSoft results in low volume.
[v01] Extracting video track number 1...
[a03] Extracting audio track number 3...
[a03] Decoding with ArcSoft DTS Decoder...
[v01] Muxing video to Matroska...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "e:\sin.flac"...



"Strange setup" and "CAUTION: Decoding this track with ArcSoft results in low volume"

What does this mean?

[Thunderbolt8]

add a sample

[rickardk]

www.earselect.se/sample.m2ts

[rica]

Originally Posted by rickardk

www.earselect.se/sample.m2ts

When you extract video and audio at the same time, you get this:

 C:\>eac3to\eac3to C:\sample.m2ts C:\s_out_video.h264 C:\s_out_audio.flac
M2TS, 1 video track, 2 audio tracks, 4 subtitle tracks, 0:00:26
1: h264/AVC, 1080p24 /1.001 (16:9)
2: AC3, 5.1 channels, 640kbps, 48khz
3: DTS Master Audio, 7.1 (strange setup) channels, 24 bits, 48khz
   (core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
4: Subtitle (PGS)
5: Subtitle (PGS)
6: Subtitle (PGS)
7: Subtitle (PGS)
Track 3 is used for destination file "s_out_audio.flac".
CAUTION: Decoding this track with ArcSoft results in low volume.
CAUTION: Decoding this track with ArcSoft results in low volume.
[a03] The Arcsoft DTS Decoder only allows one operation at a time.

What is strange is eac3to extracts/re-encode the tracs individually:

 C:\>eac3to\eac3to C:\sample.m2ts  C:\s_out_audio.flac
M2TS, 1 video track, 2 audio tracks, 4 subtitle tracks, 0:00:26
1: h264/AVC, 1080p24 /1.001 (16:9)
2: AC3, 5.1 channels, 640kbps, 48khz
3: DTS Master Audio, 7.1 (strange setup) channels, 24 bits, 48khz
   (core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
4: Subtitle (PGS)
5: Subtitle (PGS)
6: Subtitle (PGS)
7: Subtitle (PGS)
Track 3 is used for destination file "s_out_audio.flac".
CAUTION: Decoding this track with ArcSoft results in low volume.
[a03] Extracting audio track number 3...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "C:\s_out_audio.flac"...
[a03] The last DTS frame is incomplete and thus gets skipped.
[a03] The original audio track has a constant bit depth of 24 bits.
Video track 1 contains 622 frames.
eac3to processing took 7 seconds.
Done.

_ _ _ _

[rica]

When you give the adrress of video and audio; this is the result:

eac3to v2.77
command line: eac3to\eac3to  C:\sample.m2ts 1: C:\s_video.h264 3: C:\s_audio.flac
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 4 subtitle tracks, 0:00:26
1: h264/AVC, 1080p24 /1.001 (16:9)
2: AC3, 5.1 channels, 640kbps, 48khz
3: DTS Master Audio, 7.1 (strange setup) channels, 24 bits, 48khz
   (core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
4: Subtitle (PGS)
5: Subtitle (PGS)
6: Subtitle (PGS)
7: Subtitle (PGS)
CAUTION: Decoding this track with ArcSoft results in low volume.
[a03] Extracting audio track number 3...
[v01] Extracting video track number 1...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Encoding FLAC with libFlac...
[v01] Creating file "C:\s_video.h264"...
[a03] Creating file "C:\s_audio.flac"...
[a03] The last DTS frame is incomplete and thus gets skipped.
[a03] The original audio track has a constant bit depth of 24 bits.
Video track 1 contains 622 frames.
eac3to processing took 8 seconds.
Done.

[DoomBot]

Originally Posted by lithiumus

I just got the Wall E Blu-ray and it's supposed to have DTS-HD Master Audio 6.1 but eac3to shows only 5.1.

I was wondering what the "($f)" is under Active speakers.

Edit: let me clarify, the box says 5.1 DTS MA but blu-ray.com says 6.1 DTS-MA and several reviews have indicated 6.1 so maybe I'm just reaching for something that just isn't there...

D:\Program Files\eac3to>eac3to.exe "c:\Audio Testing\20000.m2ts" -logdts
+ DTS-Core
  - frameSize            2012
  - DTS-ES               +
  - channelNo            5
  - lfe                  1
  - channelDescr         5.1
  - samplingRate         48000
  - bitDepth             24
  - bitrate              1509000
  - samplesPerFrame      512
  - copyHistory          1
+ DTS-HD
  - fullSize             2776
  - headerSize           32
  - refClockCode         1/48000
  - frameDurationCode    1
  - activeMasks          [1], [[1]]
  + Asset [0]
    - fullSize             2744
    - headerSize           14
    - corePackets          Core
    - extSubStrPackets     XLL
    - bitResolution        24
    - maxSampleRate        48000
    - totalNumChannels     6
    - activeSpeakers       C L R Ls Rs LFE ($f)
M2TS, 3 video tracks, 3 audio tracks, 3 subtitle tracks, 1:37:26
1: Chapters, 32 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-ES, 5.1 channels, 24 bits, 1509kbps, 48khz)
6: AC3 Surround, English, 2.0 channels, 192kbps, 48khz
7: AC3 Surround, English, 2.0 channels, 192kbps, 48khz
8: Subtitle (PGS), English
9: Subtitle (PGS), English
10: Subtitle (PGS), English

D:\Program Files\eac3to>

Yeah, i thought it was supposed to be 6.1 as well and it shows the same thing for me, 5.1 i guess all the reviews are wrong or something oh well.

[madshi]

Originally Posted by Chumbo

I was wondering if you could provide logging at the play list level please? When running eac3to on a folder which lists the play lists, the -log does not work. Once you use -log, it processes similar to "eac3to bdmv" rather than "eac3to folder" which is a bummer. I can pipe the output of the play list to a file, but it's back to having to clean up those files from all the extra CRLFs.

Also, one other request if you have time. When I'm running an instance of eac3to that's extracting or whatever and then run another instance just to log and get info about another title, the -log doesn't work. It seems to just copy what's already there from the first instance. It would really be helpful if the logging is done independently in each eac3to command.

Will see what I can do...

Originally Posted by himan2001

New Bug-Report for the "-core"-command:

Situation: DTS-MA 7.1 File with embedded 5.1 core, and ArcSoft-Decoder:

Parameters: eac3to <src.m2ts> audio.ac3 -448 -core

Brings this error:

eac3to v2.77
command line: eac3to 00007.m2ts 3: audio.ac3 -448 -core 
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 5 subtitle tracks, 1:56:07
1: Chapters, 20 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 7.1 channels, 24 bits, 48khz
   (core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
4: DTS, English, 2.0 channels, 24 bits, 255kbps, 48khz
5: DTS, English, 2.0 channels, 24 bits, 255kbps, 48khz
6: DTS Master Audio, German, 5.1 channels, 24 bits, 48khz, dialnorm: -1dB
   (core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz, dialnorm: -1dB)
7: Subtitle (PGS), English
8: Subtitle (PGS), German
9: Subtitle (PGS), German
10: Subtitle (PGS), English
11: Subtitle (PGS), German
[a03] Extracting audio track number 3...
[a03] Extracting DTS core...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] The AC3 encoder received a non-supported data format.
Aborted at file position 32768.

Will be fixed in next build.

Originally Posted by sehgal.v7

It took some time but finally here it is.. upload first 4mins of movie Iron Man..

_http://rapidshare.com/files/164894642/1_-_DTS_Master_Audio__English__5.1_channels__16_bits__48khz.dtsma
_http://rapidshare.com/files/164889007/2_-_TrueHD_AC3__English__5.1_channels__48khz.thd

Regards.
& lemme knoe if u found noticeable diff.

The DTS-HD track is only 16bit, while the TrueHD track is 24bit. Also there are more differences: The TrueHD track is slightly louder and the LFE contains much higher peaks compared to the DTS-HD track. Don't know why that is...

Originally Posted by bigotti5

Seems there is a bug in eac3to.
Tried to concatenate some mts files from my Sony AVCHD camera.
Video presentation order:
BBIBPBPBPBPBPBBIBPBP...
Stream order:
IBBPBPBPBPBPBIBBPBPB...
Two leading B-frames encoded using only backward prediction misleads eac3to to assume -80 ms audio delay.
First audio PTS: 93600 - 00:00:01:040
First I-Frame PTS (first frame stored but third frame shown): 100800 - 00:00:01:120
First B-Frame PTS (the first frame shown): 93600 - 00:00:01:040

Thanks for the report & sample. Will be fixed in the next build.

Originally Posted by odin24

Is there a similar command to view the properties of TrueHD track as there is for DTS (-logdts)?

No.

Originally Posted by Joniii

I just demuxed Interview With The Vampire Blu-ray with eac3to v2.77. It has VC1 and AC3 5.1 640kbps + some subs. I demuxed all tracks with "eac3to X:\iwv.m2ts -demux". Then remuxed audio and video back to m2ts with TSMuxer. Now there is really low sound volume with remuxed m2ts (I need to tweak the volume in my Yamaha receiver nearly to max so that I hear any sound, with original m2ts sound level is fine). Anyone know whats causing this?

Try playing the demuxed AC3 file as it is. Does that work alright? My best guess is that the AC3 decoder you're using doesn't like 640kbit/s. Some (older) decoders have problems with that bitrate, because DVD was limited to 448kbit/s.

Originally Posted by Thunderbolt8

got a problem with converting a 2.0 ac3 track to .wavs: both outcoming wav files are slowed massively down, meaning all sounds and voices are played like ultra slowmotion.
can't say though if this is a specific problem of this single file or a general problem of 2.0 ac3 tracks. only had a 5.1 ac3 track to compare and that one played normaly after wave conversion.

edit: just noticed the track is from a mpeg2 cap and according to mpegrepair 5.1, while eac3to only recognizes it as 2.0

eac3to currently doesn't like mixed 2.0/5.1 bitstreams. I have such bitstreams on my todo list. But it doesn't have a very high priority, because usually if you have such a broadcast, home cinema freaks like us replace the audio track with a DVD audio track, anyway. And DVD audio tracks are usually straight and not such weird 2.0/5.1 mixtures.

Originally Posted by itsancho

hi all!
madshi, thank you for the great work!
well, is this a bug or it's nothing serious?
with 2.76 is the same, but with 2.75...

Seems to work for me with my latest (work in progress) sources. Can you please retry with v2.78, once it's out? If the problem still occurs, a sample (with which I can reproduce the problem) would be helpful.

Originally Posted by Daodan

I noticed latest version makes monowavs from 5.1 to BL and BR instead of old style SL and SR.
Now my question is, in case of 7.1 track, which are the extra 2 (compared to 5.1), SB,SL or BR,BL?

I'm confused. The latest version should use SL/SR for 5.1! Which format does your audio source have?

In case of a 7.1 track, SL/SR are the channels intended for the side speakers while BL/BR are intended for the back speakers. In case of a 5.1 track, the surround channel pair can be played through SL/SR with the back channels not playing any sound. Or you can use some funny algorithms to upconvert 5.1 to 7.1.

If you play a 7.1 track on a 5.1 speaker setup (or if the studio downmixes 7.1 to 5.1) the 5.1 surround channel pair usually contains a mix of the 7.1 SR/SR and BL/BR channel pairs. It doesn't matter much whether you name the 5.1 surround channel pair SL/SR or BL/BR. But IMO SL/SR is the correct name.

Originally Posted by Snowknight26

Any chances of speed increases for demuxing? Just demuxed the PCM track and an AC3->WAV track from A Few Good Men Blu-ray on a RAID array that can sustain 700MB/s, yet eac3to purportedly took 20 minutes.

AC3->WAV is not simple demuxing. It's demuxing+decoding, which is much slower than simple demuxing.

Originally Posted by Thunderbolt8

got a DTS 5.1 track here, eac3to reports it as DTS 5.1, while PowerDVD says it's DTS-ES 5.1

so which is right in this case?

The track begins with DTS 5.1, but after a while it mutates into DTS-ES. eac3to only checks the beginning of the track, while PowerDVD probably updates its status display all the time while playing the track. So is this track DTS 5.1 or DTS-ES 5.1? Neither nor. It's a mixture of both.

Originally Posted by lithiumus

I just got the Wall E Blu-ray and it's supposed to have DTS-HD Master Audio 6.1 but eac3to shows only 5.1.

I was wondering what the "($f)" is under Active speakers.

Edit: let me clarify, the box says 5.1 DTS MA but blu-ray.com says 6.1 DTS-MA and several reviews have indicated 6.1 so maybe I'm just reaching for something that just isn't there...

- activeSpeakers       C L R Ls Rs LFE ($f)

The reviewers often simply report what is printed on the box without double checking it. The Wall E track definitely begins as only DTS-HD Master Audio 5.1 ES. I only have a few MBs of the audio track, though. Maybe after the intro it changes to 6.1? I don't know.

The "($f)" is the hex value of the "activeSpeakers" bitstream element. What it means is shown in the log, namely "C L R Ls Rs LFE". These are the speakers contained in the DTS-HD track. You can see, there is no Back channel in the track, just plain 5.1.

Originally Posted by shanghai2004

Madshi, those changes are large enough to make it useful to process my audio files again (96/24 -> 48/16)?

I do not know. I'm still waiting for a reply from a guy who compares a lot of resampling algorithms. I hope that the new parameters are slightly better, but I'm not 100% sure. They could also be slightly worse...

Originally Posted by Thunderbolt8

madshi, could you please implement a kind of detection which reports 2.0 tracks being mono or stereo tracks? would be useful for older movies, sometimes those original tracks are either 1.0 or 2.0 mono. then it would be possible to distinguish if a studio included an original 2.0 mono track or if they did use a modified stero track.

Don't know how to do that. Do you have a few 2.0 mono samples?

Originally Posted by rickardk

F:\>eac3to sin.m2ts 1: e:\sin.mkv 3: e:\sin.flac
M2TS, 1 video track, 2 audio tracks, 4 subtitle tracks, 2:03:59
1: h264/AVC, 1080p24 /1.001 (16:9)
2: AC3, 5.1 channels, 640kbps, 48khz
3: DTS Master Audio, 7.1 (strange setup) channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1536kbps, 48khz)
CAUTION: Decoding this track with ArcSoft results in low volume.

Which movie is that? The warning means what it says: The ArcSoft decoder decodes this track correctly, but it lowers the volume a bit. You can more or less undo the volume change by adding "+3db" to the eac3to command line. However, perfect losslessness is lost in any case.

Why does ArcSoft lower the volume? Don't ask me. It's caused by the speaker mapping the studio has chosen. DTS-HD supports a big number of different speaker mappings for 7.1 streams. There are at least 3 different mappings the ArcSoft decoder decodes perfectly. But this specific speaker mapping used for this track seems to confuse the ArcSoft decoder, which makes it decode the track with lower volume. It's not a terribly bad thing, it's more or less similar to the effect DialNorm has. If you want, you can report this problem to the ArcSoft guys (together with a small sample). Should be easy for them to fix. Just ask them to decode the sample as 5.1 and then as 7.1. The 7.1 decoding volume will be lower, which doesn't really make any sense. With almost every other 7.1 track on the planet the 7.1 decoding volume is not lower. You can use "-logdts" to see which speaker mappings a specific 7.1 DTS-HD track uses...

[madshi]

Originally Posted by himan2001

There are a lot more "old" BD Releases which contains seemless branching and "splitted" Releases that produce Audio-Spikes
and loud noise at the cutting/fixxing Points after timecode-rerun

Something is borked when using DTS, no matter if it´s Arcsoft or Sonic. With AC3-Track splitting/joining is ok.

As i can remember, the same BD´s using an older eac3to version < 2.58 doesn´t produce this errors.

A good try will be Conair/German BD. Try to assemble
the GERMAN DTS Track. On the cut point there is a loud
spike on the decoded file (destination format can be WAV or AC3 - result is the same)

This should finally be fixed in the next build. The problem was caused by the RAW/PCM gap/overlap fixing code. The code was working just fine, but due to how LPCM sampling curves work, just removing a number of audio samples from an audio track can result in spikes. This problem doesn't seem to occur if the gap fixing is done on the AC3/DTS bitstream. Now the next version will contain a new post processing filter which will adjust the audio signal 0.5ms before and after the m2ts join points to make sure that there are no spikes in the final audio stream...

[bigotti5]

Can you correct delay calculation in the next build?
Closed GOPs at the beginning of a stream are misinterpreted in eac3to.
eac3to calculates delay from first I-frame, but this I-frame is third in presentation order, so -66 ms for NTSC and -80 ms for PAL is calculated.

TS, 1 video track, 1 audio track, 0:00:03
1: MPEG2, 480p30 /1.001 (4:3)
2: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB, -66ms

----
File Name: closed_gop.ts
File Size: 3 533 272
Stream Type: Transport
Packets Count: 19332

.....
.....
0x00002400 PES Packet { stream_id = 0xE0 (video stream)}
packet_length = 0
PES_scrambling_control = 0
PES_priority = 0
data_alignment_indicator = 1
copyright = 0
original_or_copy = 0
PTS_DTS_flags = 3
ESCR_flag = 0
ES_rate_flag = 0
DSM_trick_mode_flag = 0
additional_copy_info_flag = 0
PES_CRC_flag = 0
PES_extension_flag = 0
PES_header_data_length = 10
PTS = 0: 10: 0: 066 (54 006 006)
DTS = 0: 9: 59: 966 (53 996 997)

0x00002413 Sequence Header
horizontal_size_value = 720
vertical_size_value = 480
aspect_ratio_information = 2 (0.673500)
frame_rate_code = 4 (29.970000)
bit_rate_value = 22500 (9000000)
marker_bit = 1
vbv_buffer_size = 112
constrained_parameters_flag = 0
load_intra_quantiser_matrix = 0
load_non_intra_quantiser_matrix = 0

0x0000241F Sequence Extention
profile_and_level_indication = 72 (Main@Main)
progressive_sequence = 1
chroma_format = 1 (4:2:0)
horizontal_size_extension = 0
vertical_size_extension = 0
bit_rate_extension = 0
marker_bit = 1
vbv_buffer_size_extension = 0
low_delay = 0
frame_rate_extension_n = 0
frame_rate_extension_d = 0

0x00002429 Sequence Display Extention
video_format = 2
colour_description = 1
colour_primaries = 6
transfer_characteristics = 6
matrix_coefficients = 6
display_horizontal_size = 720
marker_bit = 1
display_vertical_size = 480

0x00002435 User Data {}

0x00002487 Group of Picture Header #0
time = 0:0:0:0 closed_gop = 1
broken_link = 0

0x0000248F Picture Header - I Frame #0
temporal_reference = 2
picture_coding_type = 1
vbv_delay = 65535

0x00002497 Picture Coding Extention
f_code[0][0] = 15
f_code[0][1] = 15
f_code[1][0] = 15
f_code[1][1] = 15
intra_dc_precision = 2
picture_structure = 3 (Frame picture)
top_field_first = 1
frame_pred_frame_dct = 1
concealment_motion_vectors = 0
q_scale_type = 0
intra_vlc_format = 1
alternate_scan = 0
repeat_first_field = 0
chroma_420_type = 1
progressive_frame = 1
composite_display_flag = 0

.......
.......

0x00013BC8 Transport Packet { PID = 0x1011, Payload = Yes (184), Counter = 14, Start indicator }

0x00013BCC PES Packet { stream_id = 0xE0 (video stream)}
packet_length = 26939
PES_scrambling_control = 0
PES_priority = 0
data_alignment_indicator = 1
copyright = 0
original_or_copy = 0
PTS_DTS_flags = 2
ESCR_flag = 0
ES_rate_flag = 0
DSM_trick_mode_flag = 0
additional_copy_info_flag = 0
PES_CRC_flag = 0
PES_extension_flag = 0
PES_header_data_length = 5
PTS = 0: 10: 0: 000 (54 000 000)
0x00013BDA Picture Header - B Frame #1
temporal_reference = 0picture_coding_type = 3
vbv_delay = 65535
full_pel_forward_vector = 0
forward_f_code = 7
full_pel_backward_vector = 0
backward_f_code = 7

0x00013BE3 Picture Coding Extention
f_code[0][0] = 1
f_code[0][1] = 1
f_code[1][0] = 1
f_code[1][1] = 1
intra_dc_precision = 2
picture_structure = 3 (Frame picture)
top_field_first = 1
frame_pred_frame_dct = 1
concealment_motion_vectors = 0
q_scale_type = 0
intra_vlc_format = 0
alternate_scan = 0
repeat_first_field = 0
chroma_420_type = 1
progressive_frame = 1
composite_display_flag = 0
.....
.....
0x0001A7C0 PES Packet { stream_id = 0xFD (extended_stream_id)}
packet_length = 1803
PES_scrambling_control = 0
PES_priority = 0
data_alignment_indicator = 1
copyright = 0
original_or_copy = 0
PTS_DTS_flags = 2
ESCR_flag = 0
ES_rate_flag = 0
DSM_trick_mode_flag = 0
additional_copy_info_flag = 0
PES_CRC_flag = 0
PES_extension_flag = 1
PES_header_data_length = 8
PTS = 0: 10: 0: 000 (54 000 000)

0x0001A7D1 AC3 Frame
SyncInfo():
CRC1 = 22870
fscod = 0
frmsizecod = 30
SamplingRate = 48000
FrameSize = 1792
BitRate = 448000
Duration = 0.032000
BSI():
bsid = 8
bsmod = 0
acmod = 7
cmixlev = 0
surmixlev = 0
lfeon = 1
dialnorm = 27
compre = 1
compr = 4
langcode = 0
Channels = 6

Here is the above examble

eac3to should calculate either from GOP header or PTS from b-picture with temporal_reference = 0

Same in h264 files. My Sony AVCHD Cam e.g. creates h264 streams with 2 b-frames at the beginnung of the stream.

[rickardk]

Originally Posted by madshi

Which movie is that? The warning means what it says: The ArcSoft decoder decodes this track correctly, but it lowers the volume a bit. You can more or less undo the volume change by adding "+3db" to the eac3to command line. However, perfect losslessness is lost in any case.

Why does ArcSoft lower the volume? Don't ask me. It's caused by the speaker mapping the studio has chosen. DTS-HD supports a big number of different speaker mappings for 7.1 streams. There are at least 3 different mappings the ArcSoft decoder decodes perfectly. But this specific speaker mapping used for this track seems to confuse the ArcSoft decoder, which makes it decode the track with lower volume. It's not a terribly bad thing, it's more or less similar to the effect DialNorm has. If you want, you can report this problem to the ArcSoft guys (together with a small sample). Should be easy for them to fix. Just ask them to decode the sample as 5.1 and then as 7.1. The 7.1 decoding volume will be lower, which doesn't really make any sense. With almost every other 7.1 track on the planet the 7.1 decoding volume is not lower. You can use "-logdts" to see which speaker mappings a specific 7.1 DTS-HD track uses...

Sin City (Swedish).
So ArcSoft will lower all channels with 3dB during decoding?

I will try to find where I can send the ArcSoft guys a sample.


with logdts switch:
F:\>eac3to sample.m2ts 3: e:\sample.flac -logdts
+ DTS-Core
- frameSize 2012
- DTS-ES -
- channelNo 5
- lfe 1
- channelDescr 5.1
- samplingRate 48000
- bitDepth 24
- bitrate 1536000
- samplesPerFrame 512
- copyHistory 1
+ DTS-HD
- fullSize 84
- headerSize 32
- refClockCode 1/48000
- frameDurationCode 1
- activeMasks [1], [[1]]
+ Asset [0]
- fullSize 52
- headerSize 14
- corePackets Core
- extSubStrPackets XLL
- bitResolution 24
- maxSampleRate 48000
- totalNumChannels 8
- activeSpeakers C L R Ls Rs LFE Lsr Rsr ($4f)
M2TS, 1 video track, 2 audio tracks, 4 subtitle tracks, 0:00:26
1: h264/AVC, 1080p24 /1.001 (16:9)
2: AC3, 5.1 channels, 640kbps, 48khz
3: DTS Master Audio, 7.1 (strange setup) channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1536kbps, 48khz)
4: Subtitle (PGS)
5: Subtitle (PGS)
6: Subtitle (PGS)
7: Subtitle (PGS)
CAUTION: Decoding this track with ArcSoft results in low volume.

[madshi]

Originally Posted by bigotti5

Can you correct delay calculation in the next build?

Look two comments above yours. I already replied to you there. Delay should be correct with the next build. At least the current work in progress sources don't report any audio delay for both of your samples...

Originally Posted by rickardk

So ArcSoft will lower all channels with 3dB during decoding?

With this specific track: Yes. With most other 7.1 tracks: No.

[bigotti5]

Look two comments above yours. I already replied to you there. Delay should be correct with the next build. At least the current work in progress sources don't report any audio delay for both of your samples...

Sorry..overlooked and big thanks for your appreciated work.

[madshi]

eac3to v2.78 released

http://madshi.net/eac3to.zip

* fixed: h264 interlaced muxing to MKV could result in too long runtime
* fixed: transcoding DTS-HD/E-AC3 core sometimes failed to work correctly
* improved TS/m2ts audio delay detection
* added filter to remove spikes when fixing gaps/overlaps in RAW/PCM audio
* each eac3to instance has its own log file now
* playlist output now also works with "-log" option
* default bitrate for mono & stereo AC3 encodes lowered to 448kbps
* default bitrate for mono & stereo DTS encodes lowered to 768kbps
* it should be possible to handle TsSplitter splitted TS files via "+" now

[nwg]

Thanks for the new version.

[Chumbo]

Originally Posted by madshi

eac3to v2.78 released

http://madshi.net/eac3to.zip

...
* each eac3to instance has its own log file now
...

Thanks so much for all the fixes and improvements and especially for adding this one. Much appreciated.

[Thunderbolt8]

Originally Posted by madshi

eac3to v2.78 released

thanks! will test those interlaced h264 movies during the week!

regarding that improved m2ts audio delay correction, have there been any problems (maybe also such which werent indicated by the log?) and it was a fix, or just more like getting the already fine working detection (for most movies) more towards perfection?

[madshi]

Originally Posted by Thunderbolt8

regarding that improved m2ts audio delay correction, have there been any problems (maybe also such which werent indicated by the log?) and it was a fix, or just more like getting the already fine working detection (for most movies) more towards perfection?

Check out bigotti5's last two posts in this thread. Delay correction itself worked just fine, but delay detection was off by 2 video frames in two samples he provided. I think this problem only occurred with some movies, though, not with all. I think most Blu-Ray movies shouldn't have this problem.

[bigotti5]

Thx - works in ts-streams.
------------
Delay correction in VOB files regarding closed GOP:
- Closed GOP Video - 2 leading B-frames
- Audio AC3 - no delay
eac3to reports:

VOB, 1 video track, 1 audio track, 0:00:08
1: MPEG2, 704x576 50i (4:3)
2: AC3, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, 3ms

in VOB files delay should be calculated using 'PTS first audio' and 'Vobu Start Presentation Time'

[nurbs]

Small feature request:
Could you make the bitrate switch "-xxx" also work with the nero aac encoder? That would save me some typing.

[madshi]

Originally Posted by bigotti5

Delay correction in VOB files regarding closed GOP:
- Closed GOP Video - 2 leading B-frames
- Audio AC3 - no delay
eac3to reports:

in VOB files delay should be calculated using 'PTS first audio' and 'Vobu Start Presentation Time'

Are you sure about that? What happens if:

- vobu start presentation time: X
- PTS first audio: X + 10ms
- PTS first video: X + 5ms

Now if you demux audio and video, audio should be delayed by 5ms and not by 10ms, or am I wrong? I can't "delay" video, so I have to delay audio by the difference between first audio and video PTS, no?

Originally Posted by nurbs

Could you make the bitrate switch "-xxx" also work with the nero aac encoder? That would save me some typing.

Using CBR for AAC encoding is not really a good thing for quality. VBR encoding gives better quality per average bitrate. So why would you want to use CBR?

[yesgrey]

Originally Posted by madshi

I'm still waiting for a reply from a guy who compares a lot of resampling algorithms.

madshi,
This "guy" is not me, am I?
I'm still working on my resampler's test, and it could take a while to finish it, I have to do a few things first.
But my first test gave some interesting results, I only don't know if the results are valid enough...

[madshi]

Originally Posted by yesgrey3

This "guy" is not me, am I?
I'm still working on my resampler's test, and it could take a while to finish it, I have to do a few things first.
But my first test gave some interesting results, I only don't know if the results are valid enough...

I meant the maintainer of this comparison website:

http://src.infinitewave.ca/

He gave me some feedback on my early SSRC implementation, based on which I tweaked the SSRC parameters a bit.

But I'm still interesting in your comparison, too. Would be nice if you could use the latest eac3to version, because of the tweaked SSRC parameters...

As far as I understand the technical comparison website above, SSRC is a rather steep resampling filter with good results, but with "normal" ringing. r8brain filters out quite a lot of the high frequencies, but on the positive side r8brain has very reduced ringing (see pulse graph). So both filters have their advantages and disadvantages, technically.

[bigotti5]

Are you sure about that? What happens if:

- vobu start presentation time: X
- PTS first audio: X + 10ms
- PTS first video: X + 5ms

not possible - vobu start presentation time == start time of first video frame (presentation order)

[madshi]

Originally Posted by bigotti5

not possible - vobu start presentation time == start time of first video frame (presentation order)

Is that written in some documentation/specification? You seem to be very sure about it. Thanks...

[nurbs]

Originally Posted by madshi

Using CBR for AAC encoding is not really a good thing for quality. VBR encoding gives better quality per average bitrate. So why would you want to use CBR?

Because sometimes I need to hit a certain filesize. I have no problem with quality based encoding as the default, but adding the option isn't much work and it would save me some typing compared to manually piping the stdout from eac3to to the nero encoder.

[bigotti5]

Is that written in some documentation/specification?

5000$ + Non Disclosure Agreement....

But look in "Philips DVD Verifier" documentation page 127

[A3] A VOBU‘s video presentation start time is given by the presentation start time of its first picture in DISPLAY ORDER ! Notice that in coding order this first picture (which is always an I-picture) may be preceded by some B-pictures.

[madshi]

Originally Posted by nurbs

Because sometimes I need to hit a certain filesize. I have no problem with quality based encoding as the default, but adding the option isn't much work and it would save me some typing compared to manually piping the stdout from eac3to to the nero encoder.

Ok, will add that to my to do list.

Originally Posted by bigotti5

5000$ + Non Disclosure Agreement....

But look in "Philips DVD Verifier" documentation page 127

Thanks!! Would you mind uploading the first part of that "3ms" VOB sample?

[yesgrey]

Originally Posted by madshi

I meant the maintainer of this comparison website:

http://src.infinitewave.ca/

From that website and my first tests the winner is SOX. Have you tried it?
http://sox.sourceforge.net/

I will post my results when I have a better understanding of their correctness... I also want to compare it by earing, but I'm currently in the process of upgrading my audiogear... maybe I should post first the technical objective tests and let the subjective tests to later?...

[bigotti5]

Here the sample

[madshi]

Originally Posted by yesgrey3

From that website and my first tests the winner is SOX.

What leads you to that conclusion? Have you compared the original SSRC results or the latest eac3to implementation?

The preliminary graphs of the latest eac3to SSRC resampling implementation (v2.78) look almost identical to the SOX graphs. Have a look for yourself:





And these may not the final results yet. I might be able to further improve noise floor (2nd image).

[madshi]

Originally Posted by bigotti5

Here the sample

Thanks! Will look at that later...

[rebkell]

I know this a stupid question, but I sometimes get confused about negative delays. If I have a video w/AC3 audio and eac3to reports -100ms delay, then basically eac3to will just drop the first three audio frames of the AC3 stream(around 96ms). Is that a correct assumption, and if it was 100ms delay, then it would add three 32ms frames at the start.

Would that be a correct assumption?

[yesgrey]

Originally Posted by madshi

What leads you to that conclusion?

My first tests. Since that website results are almost identical it made me think that my tests could show something interesting...

Originally Posted by madshi

Have you compared the original SSRC results or the latest eac3to implementation?

I have used the original SSRC HQ version. I have done this a few weeks back. I will try with your implementation, I hope it surpasses SOX!...

[madshi]

Originally Posted by rebkell

Would that be a correct assumption?

Yes.

Originally Posted by yesgrey3

My first tests. Since that website results are almost identical it made me think that my tests could show something interesting...

Do you have a software (free to share) which can produce similar graphs to those on that website? I know how to create such a sweep graph, but I don't know how to get the other ones, especially the 2nd one.

[rack04]

I'm now experiencing errors when converting the follwing DTS files to AC3.

http://www.sendspace.com/file/meghs0

eac3to v2.78
command line: eac3to "C:\Personal\Videos\dts.hires.71.24.96.2604.dtshd" "C:\Personal\Videos\dts.hires.71.24.96.2604.ac3"
------------------------------------------------------------------------------
DTS Hi-Res, 7.1 channels, 0:00:16, 24 bits, 2559kbps, 96khz
(core: DTS-ES, 5.1 channels, 0:00:16, 24 bits, 1509kbps, 48khz)
AC3 encoding doesn't support back channels. Will mix them into the surround.
Decoding with ArcSoft DTS Decoder...
Mixing surround channels...
Loading white noise (needed for dithering)...
Encoding AC3 <640kbps> with libAften...
Initialization of the AC3 encoder failed.
Aborted at file position 16384.

eac3to v2.78
command line: eac3to "C:\Personal\Videos\nature01.50ch.96kHz.24bit.ma.dtshd" "C:\Personal\Videos\nature01.50ch.96kHz.24bit.ma.ac3"
------------------------------------------------------------------------------
DTS Master Audio, 5.0 channels, 24 bits, 96khz
(core: DTS, 5.0 channels, 24 bits, 1509kbps, 48khz)
Decoding with ArcSoft DTS Decoder...
The AC3 encoder received a non-supported data format (pcm, 5, 24, -).
Aborted at file position 16384.

eac3to v2.78
command line: eac3to "C:\Personal\Videos\nature02.50ch.96kHz.24bit.ma.dtshd" "C:\Personal\Videos\nature02.50ch.96kHz.24bit.ma.ac3"
------------------------------------------------------------------------------
DTS Master Audio, 5.0 channels, 24 bits, 96khz
(core: DTS, 5.0 channels, 24 bits, 1509kbps, 48khz)
Decoding with ArcSoft DTS Decoder...
The AC3 encoder received a non-supported data format (pcm, 5, 24, -).
Aborted at file position 16384.

Also when using the -test command I get the following errors:

[yesgrey]

Originally Posted by madshi

Do you have a software (free to share) which can produce similar graphs to those on that website? I know how to create such a sweep graph, but I don't know how to get the other ones, especially the 2nd one.

No. I also don't know how to get the other ones... but I think I have a file with instructions for it, let me look in my PC to find it.

Do you want to keep this discussion in this thread? I think it would be a better idea starting a new thread just about resamplers... If you want I can start it and post my first test results...

[tebasuna51]

Originally Posted by rack04

I'm now experiencing errors when converting the follwing DTS files to AC3:
...
DTS Hi-Res, 7.1 channels, 0:00:16, 24 bits, 2559kbps, 96khz

Ac3 don't support 96 KHz. This work for me:

eac3to "dts.hires.71.24.96.2604.dtshd" xx.ac3 -resampleTo48000

...
DTS Master Audio, 5.0 channels, 24 bits, 96khz

Also need the 48 KHz conversion, but I don't remember if 5.0 is supported by eac3to (only 2.0 or 5.1?). This workaround can be used:

eac3to "nature02.50ch.96kHz.24bit.ma.dtshd" stdout.wav -resampleTo48000 | Aften -b 640 - xx.ac3

[bigotti5]

As stated earlier in this thread I tried to concatenate AVCHD clips from my Sony Cam using eac3to and its gap/overlapping feature.
But concatenating clips results in increasing negative audio delay.

First demux log from eac3to:

M2TS, 1 video track, 1 audio track, 1 subtitle track, 0:17:06
1: h264/AVC, 1440x1080 50i (16:9)
2: AC3, 5.1 channels, 448kbps, 48khz
3: Subtitle (PGS)
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[v01] Creating file "D:\MKV\video.h264"...
[a02] Creating file "D:\MKV\audio.ac3"...
[a02] Audio overlaps for 36ms at playtime 0:00:45.
[a02] Audio overlaps for 28ms at playtime 0:01:44.
[a02] Audio overlaps for 36ms at playtime 0:01:56.
[a02] Audio overlaps for 36ms at playtime 0:02:59.
[a02] Audio overlaps for 28ms at playtime 0:03:11.
[a02] Audio overlaps for 44ms at playtime 0:03:30.
[a02] Audio overlaps for 20ms at playtime 0:03:47.
[a02] Audio overlaps for 44ms at playtime 0:04:08.
[a02] Audio overlaps for 44ms at playtime 0:05:13.
[a02] Audio overlaps for 44ms at playtime 0:07:37.
[a02] Audio overlaps for 44ms at playtime 0:08:38.
[a02] Audio overlaps for 44ms at playtime 0:09:29.
[a02] Audio overlaps for 20ms at playtime 0:10:35.
[a02] Audio overlaps for 44ms at playtime 0:11:51.
[a02] Audio overlaps for 28ms at playtime 0:13:07.
[a02] Audio overlaps for 28ms at playtime 0:13:42.
[a02] Audio overlaps for 28ms at playtime 0:14:19.
[a02] Audio overlaps for 36ms at playtime 0:15:30.
[a02] Audio overlaps for 44ms at playtime 0:15:52.
[a02] The audio file was demuxed without making use of the gap/overlap information.
[a02] Please rerun the same eac3to command line. That will correct the gaps/overlaps.
Video track 1 contains 51348 frames.
eac3to processing took 58 seconds.
Done.

I just analyzed my clips and found a 20 ms difference to eac3to
Start Delay in all clips is 1:040

Clip1
last Video PTS: 45:720
correct Start Delay: 44:680
Duration (adding duration of last video frame): 44:720
last Audio PTS: 45:744
correct Start Delay: 44:704
Duration (adding duration of last audio frame): 44:736
16 ms

Clip2
last Video 1.00:800 = 59:760 = 59:800
last Audio 1.00:816 = 59:776 = 59:808
8 ms

Clip3
last Video 12:440 = 11:400 = 11:440
last Audio 12:464 = 11:424 = 11:456
16 ms

Clip4
last Video 1.04:440 = 1.03:400 = 1.03:440
last Audio 1.04:464 = 1.03:424 = 1.03:456
16 ms

....
....

So I assume these differences results in increasing delay.

[madshi]

Originally Posted by rack04

Also when using the -test command I get the following errors

Argh, will fix that in the next build. For now you can simply delete the whole "plugin" folder. It's not needed yet, anyway.

Originally Posted by yesgrey3

Do you want to keep this discussion in this thread? I think it would be a better idea starting a new thread just about resamplers... If you want I can start it and post my first test results...

Starting a new thread would make sense...

Originally Posted by tebasuna51

Ac3 don't support 96 KHz. This work for me:

eac3to "dts.hires.71.24.96.2604.dtshd" xx.ac3 -resampleTo48000

Ah yes. eac3to automatically downmixes 7.1 to 5.1, but it doesn't automatically activate 96khz -> 48khz resampling yet. I'll add that to my to do list...

Originally Posted by tebasuna51

Also need the 48 KHz conversion, but I don't remember if 5.0 is supported by eac3to (only 2.0 or 5.1?).

5.0 AC3 encoding is not supported yet by eac3to.

Originally Posted by bigotti5

As stated earlier in this thread I tried to concatenate AVCHD clips from my Sony Cam using eac3to and its gap/overlapping feature.
But concatenating clips results in increasing negative audio delay.

The big question is whether the video in your clips is encoded as single interlaced fields or as interlaced frames? In the first case a video field is 20ms long, in the 2nd case the frame is 40ms long. But now that I think about it, I think eac3to's gap/overlap correction doesn't properly detect these cases. I think for an interlaced stream (regardless of whether the stream is encoded as fields or frames) eac3to always calculates with only 20ms. Which is not correct. But still the results you get could be correct if your stream consists of single encoded fields, only. Is that the case?

[evdberg]

Is it correct that eac3to does not detect DD+ tracks in M2TS files (streamtype == 0x84)? Also it seems that a DD+ track on BD has a DD core inside, just like TrueHD. The PS3 shows a 640kbps 5.1 DD track when playing the file with a 7.1 DD+ track.

[bigotti5]

I think for an interlaced stream (regardless of whether the stream is encoded as fields or frames) eac3to always calculates with only 20ms. Which is not correct. But still the results you get could be correct if your stream consists of single encoded fields, only. Is that the case?

Stream consists of single encoded fields but PES packet header containing PTS spans always two fields.
Duration of clips is always a multiple of 40 ms (PAL).
An example of such a stream is in post #7054

[madshi]

Originally Posted by evdberg

Is it correct that eac3to does not detect DD+ tracks in M2TS files (streamtype == 0x84)? Also it seems that a DD+ track on BD has a DD core inside, just like TrueHD. The PS3 shows a 640kbps 5.1 DD track when playing the file with a 7.1 DD+ track.

Currently eac3to doesn't support Blu-Ray style main audio DD+ track which have a DD core (Blu-Ray commentary tracks don't have a DD core, these are supported by eac3to). The reason for that is that the only such sample world wide seems to be from a Dolby demo disc. Do you have a real movie disc with such a track?

Originally Posted by bigotti5

Stream consists of single encoded fields but PES packet header containing PTS spans always two fields.
Duration of clips is always a multiple of 40 ms (PAL).

Ok, good to know. But how is it done with 60i video and movie content? I guess with 60i video there are also always 2 fields for one PTS timestamp? So I'd have to use 33.366ms, right? How about movies with pulldown flags? 41.70833ms or 33.366ms?

[yesgrey]

When resampling a 16 bit audio file the result should not be also a 16 bit audio file? eac3to is giving me a 24 bit audio file...
Here is my log:

eac3to v2.78
command line: eac3to white44.1_16.wav white44.16_eac3to.wav -resampleTo48000
------------------------------------------------------------------------------
WAV, 2.0 channels, 0:00:10, 16 bits, 1411kbps, 44.1khz
Reading WAV...
Resampling to 48khz...
Reducing depth from 64 to 24 bits...
Writing WAV...
Loading white noise (needed for dithering)...
Creating file "white44.16_eac3to.wav"...
The original audio track has a constant bit depth of 16 bits.
The processed audio track has a constant bit depth of 24 bits.
eac3to processing took 1 second.
Done.

[madshi]

Originally Posted by yesgrey3

When resampling a 16 bit audio file the result should not be also a 16 bit audio file? eac3to is giving me a 24 bit audio file...

Resampling is done in 64bit floating point. If you want to end up with a 16bit audio track, use the "-down16" parameter. But even downconverting to 24bit already reduces quality. If you want "full" quality (I mean best resampling comparison graphs) you should use "-down32" (32bit PCM) or "-full" (64bit floating point). The "sweep" SSRC High Precision graph on the resampling comparison website looks only that bad because the SSRC standalone tool doesn't support 32bit PCM output. The dithering down to 24bit is responsible for the dark blue background in that graph.

[evdberg]

Originally Posted by madshi

Currently eac3to doesn't support Blu-Ray style main audio DD+ track which have a DD core (Blu-Ray commentary tracks don't have a DD core, these are supported by eac3to). The reason for that is that the only such sample world wide seems to be from a Dolby demo disc. Do you have a real movie disc with such a track?

No ... I have the Dolby demo disc ... I can not get DD+ without DD core to work on BD, so I assumed that a DD core is mandatory. Which discs do have DD+ commentary tracks without DD core?

[nautilus7]

Originally Posted by evdberg

I can not get DD+ without DD core to work on BD, so I assumed that a DD core is mandatory.

You mean you can't mux DD+ w/o a core inside m2ts (using tsmuxer maybe?)

Originally Posted by evdberg

Which discs do have DD+ commentary tracks without DD core?

Transformers blu-ray.

[evdberg]

Originally Posted by nautilus7

You mean you can't mux DD+ w/o a core inside m2ts (using tsmuxer maybe?)

I can mux it with tsmuxer, but the result won't give any sound ... at least not on the PS3. The PS3 detects a DD track with variable bitrate. This is not strange, considering that txmuxer tags the DD+ track with streamtype 0x81 (DD) instead of 0x84.

Originally Posted by nautilus7

Transformers blu-ray.

I have that one only on HD-DVD ...

[bigotti5]

Ok, good to know. But how is it done with 60i video and movie content? I guess with 60i video there are also always 2 fields for one PTS timestamp? So I'd have to use 33.366ms, right?

Imho yes

How about movies with pulldown flags? 41.70833ms or 33.366ms?

Never 41.70833
Frame duration in pulldowned videos is always 33.366 ms.

[madshi]

Originally Posted by bigotti5

Imho yes

Ok, thanks. Will fix that problem in the next build.

[bigotti5]

I did a test counting video frames and audio frames in one of my cam files.

Video frames: 1495 (2990 fields) = 59.800 sec | eac3to: 2988 = 59.760 sec
Audio frames: 1869 = 59.808 | eac3to: 1868 = 59.776

eac3to reports 28 ms overlapping

Concatenating this clip e.g five times results in 14948 fields reported by eac3to (2988*5 = 14940)

[madshi]

Originally Posted by bigotti5

I did a test counting video frames and audio frames in one of my cam files.

Video frames: 1495 (2990 fields) = 59.800 sec | eac3to: 2988 = 59.760 sec
Audio frames: 1869 = 59.808 | eac3to: 1868 = 59.776

eac3to reports 28 ms overlapping

Concatenating this clip e.g five times results in 14948 fields reported by eac3to (2988*5 = 14940)

Hmmmm... eac3to can not be sure whether the last PES packets are complete (some streams have wrong length information in the headers, so eac3to is ignoring the length field in the PES headers). Because of that reason the last video and audio frame is currently ignored. However, if you concatenate multiple m2ts files via "+" in the command line, the last video and audio frame in each file should NOT be ignored. In other words: I don't understand why eac3to behaves that way. Can you (once again) upload that sample you tested with? Thanks!

[bigotti5]

Because of that reason the last video and audio frame is currently ignored.

If so it should report 16 ms (59.760 <-> 59.776)

edit:

via "+" in the command line, the last video and audio frame in each file should NOT be ignored.

then 14948 is correct

[madshi]

Originally Posted by bigotti5

If so it should report 16 ms (59.760 <-> 59.776)

Well, the current eac3to build adds 20ms to the last video PTS for PAL interlaced content when concatenating 2 clips (as I said, will be fixed in the next build). So I'd expect eac3to to report 36ms. Don't know why it ends up with 28ms. How long is that clip you're talking about? Can you upload it for me to check out?

[bigotti5]

Misunderstanding.....16 ms is for the single file.
Concatenating does not ignore each last frame, 8 ms will be correct - adding 20 ms results in 28 ms.
So next build will fix this, thx.

Here is the sample (75 mb)

[madshi]

Originally Posted by bigotti5

Misunderstanding.....16 ms is for the single file.
Concatenating does not ignore each last frame, 8 ms will be correct - adding 20 ms results in 28 ms.
So next build will fix this, thx.

Here is the sample (75 mb)

Yep, eac3to reports 2988 frames for the single file and 5978 (= 2990 + 2988) frames, if I use "sample.mts+sample.mts". So it works as intended. And my latest (work in progress) sources report 8ms for each overlap...

[cavediver]

I have figured out how to mux seemlessly branched blu-ray's to mkv using eac3to and how to create truehd, pcm and ac3 audio tracks with eac3to. But what I haven't figured out is how to put them all back together into an mkv. I've tried using both Haali and Mkvtoolnix to put both the video file and audio files into an mkv, but both indicate that the pcm and thd+ac3 audio files are not supported media files. I've been successful using tsmuxer to put the files back together, but the truehd tracks won't play in my PCH A-110 even after remuxing with txremux. So, how do I put all of the files I've created using eac3to into an mkv container?

[mikeathome]

Hi,
might have been reported already, did not read thru all 358 posts ;-)

Downsampling 6ch AAC to 2 ch AAC did not work for me. Created a 6ch ACC instead (= did nothing)

CMDLine: eac3to 6ch.aac 2ch.aac -down2

Am I missing something?

mike

[jmonier]

Originally Posted by cavediver

I have figured out how to mux seemlessly branched blu-ray's to mkv using eac3to and how to create truehd, pcm and ac3 audio tracks with eac3to. But what I haven't figured out is how to put them all back together into an mkv. I've tried using both Haali and Mkvtoolnix to put both the video file and audio files into an mkv, but both indicate that the pcm and thd+ac3 audio files are not supported media files. I've been successful using tsmuxer to put the files back together, but the truehd tracks won't play in my PCH A-110 even after remuxing with txremux. So, how do I put all of the files I've created using eac3to into an mkv container?

As I understand it, mkv does not support pcm or thd. For a lossless audio stream you need to convert them to flac. Or you can convert them to straight ac3.

[Skinleech]

Hhhm. I've not seen this message before, using Arcsoft to decode:

M2TS, 1 video track, 2 audio tracks, 4 subtitle tracks, 2:03:59
1: Chapters, 10 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48khz
4: DTS Master Audio, English, 7.1 (strange setup) channels, 24 bits, 48khz
   (core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
5: Subtitle (PGS), Swedish
6: Subtitle (PGS), Danish
7: Subtitle (PGS), Norwegian
8: Subtitle (PGS), Finnish
Creating file "f:\sin.txt"...
CAUTION: Decoding this track with ArcSoft results in low volume.
[a04] Extracting audio track number 4...
[v02] Extracting video track number 2...
[v02] Muxing video to Matroska...
[a04] Decoding with ArcSoft DTS Decoder...
[a04] Encoding FLAC with libFlac...
[a04] Creating file "f:\sin.flac"...
[a04] The last DTS frame is incomplete and thus gets skipped.
[a04] The original audio track has a constant bit depth of 24 bits.
Added fps value to MKV header.
Video track 2 contains 178367 frames.
eac3to processing took 57 minutes, 10 seconds.
Done.

Sonic doesn't apply that error message, but I can only get 5.1 channels via that. Any idea? Thanks in advance.

[madshi]

Originally Posted by cavediver

both indicate that the pcm and thd+ac3 audio files are not supported media files.

MKV doesn't support TrueHD (yet?). You can use either FLAC or PCM. MKV does support PCM, but muxing it with mkvtoolnix only works through WAV. So you need to ask eac3to to create a WAV file and then mux that with mkvtoolnix. I don't know if mkvtoolnix can handle WAV files bigger than 2GB (4GB), though.

Originally Posted by mikeathome

Downsampling 6ch AAC to 2 ch AAC did not work for me. Created a 6ch ACC instead (= did nothing)

That's a bug. Will be fixed in the next build. For now you can work around it by first doing "eac3to source.ac3 temp.wav -down2 -full" and then in an extra step "eac3to temp.wav dest.aac".

Originally Posted by Skinleech

I've not seen this message before

Just do a search on this thread for the very warning message and you'll find the explanation you're looking for. (And maybe next time you can do that before posting... )

[Skinleech]

Originally Posted by madshi

Just do a search on this thread for the very warning message and you'll find the explanation you're looking for. (And maybe next time you can do that before posting... )

Thanks. I did try searching, and it only brought up the 7000+ reply thread, not the posts related to the error.

I'll try again.

EDIT: Got it now; for some reason I didn't have the 'search this thread' option before, but it's there now.

[itsancho]

Originally Posted by madshi

MKV doesn't support TrueHD (yet?). You can use either FLAC or PCM. MKV does support PCM, but muxing it with mkvtoolnix only works through WAV. So you need to ask eac3to to create a WAV file and then mux that with mkvtoolnix. I don't know if mkvtoolnix can handle WAV files bigger than 2GB (4GB), though.

yep!

2008-05-15 Moritz Bunkus <moritz@bunkus.org>

* mkvmerge: new feature: Improved support for WAV files bigger
than 4 GB which only contain a single DATA chunk and a wrong
length field for this DATA chunk (e.g. eac3to creates such files).

and i'm waiting for TrueHD support, too...

[Jeff Flowerday]

Question for madshi or anyone smart like him:

SD DVDs with a mpeg2 video stream that is a combination of progressive and interlaced. I've seen it numerous times when I try to strip the pulldown flag. Just wondering what's going on, is there an acutal frame rate change? I assumed DVDs were all progressive on the source by nature.

[gillie]

Hoping someone can help.
I've been using the following process to produce .ts files of Blu-Ray rips which I can then play via my TVix M6500 "media streamer".

1. Rip Blu-ray disc to hard disk with AnyDVD HD
2. Use eac3to to extract video to .mkv and the audio to .dts or .ac3
3. Use tsMuxeR GUI to create .ts file container the video and audio streams

This process has worked fine for weeks but has now started creating a .ts file which has either no video playback or corrupted video playback. The audio is fine.

If I use tsMuxeR to extract the video and audio directly from the Blu-ray hard disk copy then the .ts file works okay. But using this method I'm unable to modify the audio output. My preference is to extract the lossless audio stream to DTS using eac3to and Surcode DTS encoder.

If I use MKVmerge to create a .mkv file of the same streams the output is fine, but my TviX box is rubbish with large .mkv files with high bit rates so this isn't an option for me.

Also tried using eac3to to extract the video to an elementary stream (.h264) then tsmuxed the streams into a .ts file and all works fine.

Looking at the mediainfo details for the .ts files which don't work it looks like something odd is going on. The "duration" of the .ts file is twice what it should be. But the individual .mkv and .dts files both have the correct duration. Think it may be the latest versions of eac3to which are producing odd .mkv files.

Would welcome anyones thoughts on this one.

[xkodi]

so, the studios are now using the DTS-HD MA 7.1 tracks with the "strange setup" on real movies too and not only on some BD demos.

[madshi]

Originally Posted by Jeff Flowerday

SD DVDs with a mpeg2 video stream that is a combination of progressive and interlaced. I've seen it numerous times when I try to strip the pulldown flag. Just wondering what's going on, is there an acutal frame rate change? I assumed DVDs were all progressive on the source by nature.

This is a complicated topic. MPEG2 streams can be encoded as separate interlaced fields or as frames. A frame doesn't have to be progressive. It's very much possible to encode two interlaced fields in one frame. So even if an MPEG2 stream consists of frames, it can still be interlaced in nature. Now on top of that you can build the MPEG2 bitstream in such a way that only 24 frames per second are actually encoded, but the stream still plays at 60i, because there are flags in the stream which tell the decoder which fields are supposed to be repeated to do the 48i -> 60i pulldown.

In real life you can not trust the DVD bitstream encoding. You can not trust MPEG2 broadcasts, either. Both often switch between all the various bitstream encoding formats (60 interlaced fields, 30 frames, 24 frames with pulldown flags). Even if you have a stream with 24 frames and pulldown flags throughout the whole DVD, you still cannot always trust that everything is progressive. It's still possible that the wrong fields were encoded together into frames or that the pulldown flags are plain wrong. Because of this a good DVD player usually ignores all this stuff and simply decodes the stream as 60i and then uses a real video processing chip (e.g. HQV Reon) to figure out which of the fields belong together by analyzing the actual image content of all the fields.

So IMHO DVDs should never be treated as being progressive - unless you know for a fact that they are encoded perfectly.

Originally Posted by gillie

Looking at the mediainfo details for the .ts files which don't work it looks like something odd is going on. The "duration" of the .ts file is twice what it should be. But the individual .mkv and .dts files both have the correct duration.

If the individual mkv and dts files seem to work correct on their own then why do you think the problem has anything to do with eac3to? It seems that the files produced by eac3to work perfectly fine on their own, but the TS file produced by tsMuxeR does not. So this looks like a tsMuxeR bug to me. And that's going to be my stance - unless you can find some evidence that something is wrong with the MKV and/or audio files produced by eac3to.

Originally Posted by xkodi

so, the studios are now using the DTS-HD MA 7.1 tracks with the "strange setup" on real movies too and not only on some BD demos.

It seems so. I'm not sure how many different movies are out there with such tracks, but at least one movie is. Unfortunately I'm not in a position to fix this. It's technically too complicated. The only proper solution would be for people who have licensed ArcSoft and who own such a problematic 7.1 Blu-Ray disc to report the problem to ArcSoft and ask them for a DTS-HD decoder fix.

[rica]

gillie ,
Haali and TSMuxer do not like VC1mkvs.
If your final container will be a TS, i'd advise this way:

Demux vc1 and audio with eac3to and remux to ts with TSMuxer.


If your final container will be an mkv, i'd advise this way:

Demux vc1 and audio with eac3to.
Open your vc1 with TSMuxer and remux into a ts container.
Convert ts to mkv with Graphstudio:

Haali splitter > Haali matroska muxer

Open this mkv file with MkvMergeGui and remux it with audio into an mkv.

(MkvMerge gui doesn't recognize raw vc1; so you have to convert it to mkv.
Haali splitter doesn't recognize vc1; so you have to convert it to ts)

[madshi]

Originally Posted by rica

If your final container will be an mkv, i'd advise this way

I disagree. What you suggest is complicated, time consuming and results in an MKV file which has suboptimal timestamps.

If the target container is MKV, there's no reason not to use eac3to for VC-1 muxing. Actually eac3to will give you optimal results. You can't get any better VC-1 MKV than those created by eac3to, as far as I'm aware.

Muxing to TS is another topic. It might be true that letting eac3to demux to a raw video stream is preferable if you want to create a TS with tsMuxeR. But if that's the case, it's caused by a bug in tsMuxeR and not by eac3to.

[xkodi]

Originally Posted by madshi

It seems so. I'm not sure how many different movies are out there with such tracks, but at least one movie is. Unfortunately I'm not in a position to fix this. It's technically too complicated. The only proper solution would be for people who have licensed ArcSoft and who own such a problematic 7.1 Blu-Ray disc to report the problem to ArcSoft and ask them for a DTS-HD decoder fix.

yes, you mentioned before that patching DTS-HA MA headers is too complicated, because of the variable frame size if i remember correctly.

i doubt that Arcsoft will fix it soon, if fix it at all, e.g. Arcsoft do not decode correctly also all DTS-HD MA 7.1 16bit samples that i have in GraphEdit, but at least Arcsoft+eac3to decode them correctly.

[Thunderbolt8]

Originally Posted by itsancho

yep! and i'm waiting for TrueHD support, too...

theres some ironie, mkvmerge can mux DTS-HD MA, but theres no directshow decoder for it yet. but ffdshow supports thruehd support (soon), yet we cannot mux it yet

[madshi]

Originally Posted by xkodi

i doubt that Arcsoft will fix it soon, if fix it at all

Not sure about that. Has anybody even brought this problem to ArcSoft's attention yet? Of course they won't fix it, if nobody reports it...

[madshi]

Originally Posted by Thunderbolt8

theres some ironie, mkvmerge can mux DTS-HD MA, but theres no directshow decoder for it yet.

Doesn't the Sonic decoder work just fine? (Of course limited to 5.1, but still?)

The problem with TrueHD -> MKV muxing is simply that it's a totally separated codec. DTS-HD streams are muxed just like DTS streams are, so that was easy. Muxing TrueHD requires a whole new definition of how to do it exactly in MKV.

[rica]

Originally Posted by madshi

But if that's the case, it's caused by a bug in tsMuxeR and not by eac3to.

Sorry but did i tell something like this?

I just wanted to share my experience; that's all.

 which has suboptimal timestamps.

i disagree either.

[xkodi]

Originally Posted by madshi

Not sure about that. Has anybody even brought this problem to ArcSoft's attention yet? Of course they won't fix it, if nobody reports it...

actually, i believe that ArcSoft can't fix anything about DTS-HD MA decoding, they just buy the DTS-HD decoding library from DTS Labs and use it.

if you look with hex editor in dtsdecoderdll.dll (Arcsoft), CinemasterAudio.dll (Sonic) and dtshddec.dll (Corel WinDVD9) you can find the version of the DTS Labs DTS-HD decoding library with which the corresponding DLL is linked.

Arcsoft uses the most recent version of all of these 3 DLLs, but it's still old compared to the version that DTS Labs DTS-HD MAS suite version 1.6 is using.

so, i'm under the impression that when DTS Labs make newer version of the DTS-HD decoding library they don't supply it to their customers of the old versions of the library for free, otherwise i can't understand why Arcsoft, Sonic and Corel don't update the version they are using to the latest one, but instead they use old versions.

[Thunderbolt8]

Originally Posted by madshi

Doesn't the Sonic decoder work just fine? (Of course limited to 5.1, but still?)

frankly said never tried it. i thought it was only used for eac3 decoding, didnt know I can use it to play dts-hd ma tracks as well. it plays them completely without drc, dialnorm; this is only applied for (e)ac3?

[gillie]

Originally Posted by madshi

I disagree. What you suggest is complicated, time consuming and results in an MKV file which has suboptimal timestamps.

If the target container is MKV, there's no reason not to use eac3to for VC-1 muxing. Actually eac3to will give you optimal results. You can't get any better VC-1 MKV than those created by eac3to, as far as I'm aware.

Muxing to TS is another topic. It might be true that letting eac3to demux to a raw video stream is preferable if you want to create a TS with tsMuxeR. But if that's the case, it's caused by a bug in tsMuxeR and not by eac3to.

Thanks Madshi. I have to agree that using eac3to to create .mkv video stream has always worked flawlessly for me. Likewise using it to create DTS audio stream from lossless audio on original Blu-ray source works great as well. My preference would always be this route and then use MKVmergeGUI to create single .mkv file, however the DViX M6500 is absolutley hopeless at streaming .mkv files with high bit rates, i.e. 20Mbps+ hence my reason for using .ts which the DViX plays without a problem even though the video and audio steams are identical to those in the .mkv container.

Have any changes been made to eac3to recently which change the header info in an .mkv video stream which may now be confusing tsMuxeR?

[rica]

Think it may be the latest versions of eac3to which are producing odd .mkv files.

Who told this? Me?

[Nullity]

Originally Posted by madshi

MKV doesn't support TrueHD (yet?).

Originally Posted by itsancho

yep! and i'm waiting for TrueHD support, too...

I sent Mosu (the mkvtoolnix maintainer) an email about this a couple months ago, asking if he was planning on adding TrueHD support. His response was not quite what I was hoping for...

Probably not. I don't have specs for Dolby TrueHD, nor do I have such files.

Regards,
Mosu

I have no idea if he has changed his stance, but madshi, you have much greater influence in this area, perhaps you could convince him or give him a hand?

[madshi]

Originally Posted by xkodi

actually, i believe that ArcSoft can't fix anything about DTS-HD MA decoding, they just buy the DTS-HD decoding library from DTS Labs and use it.

May be true. But I'm not fully sure. E.g. do they get source code from DTS or just a static lib to link with? Also I'm not sure if it's really the decoder which is lowering the volume or if it's ArcSoft's own post processing...

Originally Posted by Thunderbolt8

it plays them completely without drc, dialnorm; this is only applied for (e)ac3?

Yes. BUT the decoder is slow and limited to 5.1. And I'm not sure if it accepts the Haali Media Splitter as input.

Originally Posted by gillie

Have any changes been made to eac3to recently which change the header info in an .mkv video stream which may now be confusing tsMuxeR?

No, not at all. But I've heard multiple times that tsMuxeR generally doesn't like VC-1 MKVs. Don't know if that's true. And I really don't care much.

Originally Posted by Nullity

I have no idea if he has changed his stance, but madshi, you have much greater influence in this area, perhaps you could convince him or give him a hand?

The bigger problem is that there's no specification as of yet on how TrueHD should be stored in MKV. And it's not up to Mosu to make up such a specification himself.

[Momber]

Hi!
I just finished converting a True-HD track from HD DVD (Patch Adams) to lpcm and got this strange result:

[a04] Original audio track, L+R+C+LFE: constant bit depth of 16 bits.
[a04] Original audio track, SL+SR: constant bit depth of 24 bits.

How can a track have a bit depth of 16 and 24 bits at the same time?
How should I treat the resulting lpcm track for further processing - as 16 or 24 bit?

Greets
S.

[banker_rishad]

Can anybody help with eac3to guides like to how to%. Step by step guide. madshi plz advise

[xkodi]

Originally Posted by madshi

May be true. But I'm not fully sure. E.g. do they get source code from DTS or just a static lib to link with?

static lib for sure, DTS Labs will never give them source code. also, i think that when you buy license for the DTS decoding lib, seems you get license only for particular version and not for the future updated versions, because for example let's take Sonic - in all version of their decoder DLL (the latest version of CinemasterAudio.dll 4.3.0.230 is only 1-2 months old) they always link to the same version of the DTS decoding lib and it is very old version.

same applies to Corel and Arcsoft with only the difference that Corel uses more recent version of the DTS-HD decoding lib than Sonic and Arcsoft uses more recent version than Corel, but still Arcsoft version is far from the latest that DTS-HD MAS uses. i look at this few month ago i and don't remember the exact version of the DTS lib that each DLL is using, but it was DTS-HD MAS > Arcsoft > Corel > Sonic. maybe, i will look at this again and post a table here in format "dll -> dts lib version".

[madshi]

Originally Posted by Momber

I just finished converting a True-HD track from HD DVD (Patch Adams) to lpcm and got this strange result:

[a04] Original audio track, L+R+C+LFE: constant bit depth of 16 bits.
[a04] Original audio track, SL+SR: constant bit depth of 24 bits.

How can a track have a bit depth of 16 and 24 bits at the same time?

TrueHD always decodes to 24bit. A "16bit" TrueHD track also decodes to 24bit, but has the lower 8 bits zeroed out. So with this specific movie, obviously the surround channels have a true bitdepth of 24bit (all 24bits being made use of) while the other channels have the 8 lower bits zeroed out.

Originally Posted by Momber

How should I treat the resulting lpcm track for further processing - as 16 or 24 bit?

If you want to conserve the full quality you should treat it as 24bit track. E.g. FLAC compression will still work very well. The final FLAC filesize will be somewhere between a typical 16bit and a typical 24bit FLAC size.

Of course the surround channels are slightly less important than the left/center/right channels. So you could also say since the more important channels are only 16bit, anyway, you can use "-down16" to convert the whole track to 16bit. However, doing this will result in eac3to applying dither to the main channels, too, unfortunately. So I'd recommend to keep the track as 24bit.

[Momber]

Thanks for your reply madshi!
I will treat it with pcm2tsmu as 24 bit... let's see how that works...

[siella]

I ask convert dts audio from here

[n0mag!c]

Originally Posted by madshi

TrueHD always decodes to 24bit. A "16bit" TrueHD track also decodes to 24bit, but has the lower 8 bits zeroed out.
So you could also say since the more important channels are only 16bit, anyway, you can use "-down16" to convert the whole track to 16bit. However, doing this will result in eac3to applying dither to the main channels, too, unfortunately.

Maybe you can implement "smart" "-down16"-switch behaviour then? Which will truncate lower 8 bits instead of making "standard TPDF dithering" for these channels? Or there are hidden disadvantages?

[madshi]

Originally Posted by n0mag!c

Maybe you can implement "smart" "-down16"-switch behaviour then? Which will truncate lower 8 bits instead of making "standard TPDF dithering" for these channels? Or there are hidden disadvantages?

That'd be hard to realize. If you do "-down16" eac3to doesn't know yet which channel has which true bitdepth. So eac3to would have to apply dithering. Only at the end of the whole processing eac3to would notice "ooops, some of those channels were only 16bit to begin with" and thus would have to redo the whole processing from the get go. That's quite complicated and honestly I don't really think it's worth the effort...

[Momber]

Originally Posted by Momber

I will treat it with pcm2tsmu as 24 bit... let's see how that works...

It worked! Which is surprising, because usually if you feed 16 bit PCM into pcm2tsmu without the -i 16 option the result gets garbled. It didn't in this case however, which would indicate the lpcm track out of eac3to was true 24 bit after all. At least that's how I make sense of it...

Greets
S.

[Thunderbolt8]

madshi, could you please add a switch that just demuxes audio stream(s) in their state as they originally are (e.g. without applying any kind of delay and dialnorm removal etc.)? im hinting at those tv caps with corrupted ac3 (or dts) audio frames. in that case I always have to switch to xport to demux the track and then process it with delaycut. since this step cannot be avoided anyway, why not let eac3to demux them, 1 less program to use in between. there could be a message then like "just demuxing track without applying any delay or processing dialnorm etc." so that the user knows this stuff is still applied to the track and he then can get rid of it/fix it at a later time (e.g. after delaycut) if he likes to.

it just doenst make sense to go eac3to -> xport -> delaycut, when it could be eac3to -> eac3to -> delaycut. perhaps it would make sense to add a kind of -autoaudiodemux switch which then, in case that audio problems are detected and eac3to would have to abort, eac3to would directly start with the same command line again (or the audio track only then if it saves time and theres no thing as gaps), but this time then just as described as pure demuxing without any processing of that track.

[quantum]

Originally Posted by madshi

..If the target container is MKV, there's no reason not to use eac3to for VC-1 muxing. Actually eac3to will give you optimal results. You can't get any better VC-1 MKV than those created by eac3to, as far as I'm aware.

Muxing to TS is another topic. It might be true that letting eac3to demux to a raw video stream is preferable if you want to create a TS with tsMuxeR. But if that's the case, it's caused by a bug in tsMuxeR and not by eac3to.

Is there any reason not to demux to raw video? I've found .TS works better on my Networked Media Tank so I've been going that way. I've been demuxing my Blu-rays and HD-DVDs to raw video and audio using eac3to, then remuxing with tsMuxer. So far I haven't visually noticed any issues. Am I losing timecodes this way?

[Momber]

Originally Posted by madshi

I've checked The Descent and it's stored as 7.1, but BL and BR are identical and the case also sais 6.1. But still it's stored as 7.1.

I'm working on The Descent right now and eac3to gives me only 5.1 when demuxing to lpcm. I've also tried the -8 option but the output was the same: 5.1.
What am I doing wrong here?

TIA
S.

[madshi]

Originally Posted by Momber

It worked! Which is surprising, because usually if you feed 16 bit PCM into pcm2tsmu without the -i 16 option the result gets garbled. It didn't in this case however, which would indicate the lpcm track out of eac3to was true 24 bit after all. At least that's how I make sense of it...

Well, if eac3to reports 16bit in its bitdepth statistics, that's just saying how many bits are filled with actual information. The bitdepth analyzation statistic does not say in which bitdepth the data is packaged. You can have 16 actual bits in a 24bit transport, or 20 actual bits in a 24bit transport, or 16 actual bits in a 32bit transport. That's all possible. A TrueHD track is usually something between 16-24 actual bits, but it's always a 24bit transport. eac3to does change the transport from 24bit to 16bit (by stripping the zero bytes) if the whole TrueHD track is only 16bit. So in such cases you have to use "-i 16" for pcm2tsmu. But for all other TrueHD tracks the transport is left at 24bit.

Or in other words: The pcm2tsmu switches must be set to the transport bitdepth and not to the number of bits which are non-zero inside of the transport. pcm2tsmu doesn't care how many bits are zero or non-zero.

Originally Posted by Thunderbolt8

madshi, could you please add a switch that just demuxes audio stream(s) in their state as they originally are

Igoring or working around errors has been requested a thousand times already and it's on my to do list - just like a dozen of other important things...

Originally Posted by quantum

Is there any reason not to demux to raw video?

Depends on your final target. If you want to end up with an MKV then it's best to let eac3to create that MKV. If you want to end up with a TS, then you should do whatever works best with tsMuxeR. If tsMuxeR handles raw video streams better than MKVs then I don't see any problems using raw video streams. You do lose the timestamps created by eac3to in that case, but that's not really a problem - unless there are gaps/overlaps in the video track, which is extremely rare...

Originally Posted by Momber

I'm working on The Descent right now and eac3to gives me only 5.1 when demuxing to lpcm. I've also tried the -8 option but the output was the same: 5.1.
What am I doing wrong here?

My English The Descent PCM track is 7.1 (reported as 6.1 on the back cover). Maybe you have a different version of The Descent? Does eac3to report 5.1 or 7.1 in the track listing?

[Momber]

Originally Posted by madshi

My English The Descent PCM track is 7.1 (reported as 6.1 on the back cover). Maybe you have a different version of The Descent? Does eac3to report 5.1 or 7.1 in the track listing?

eac3to reports 7.1 and so does every other tool known to man
The demuxed track is however only 5.1 (which also correlates with its size).

[madshi]

Originally Posted by Momber

eac3to reports 7.1 and so does every other tool known to man
The demuxed track is however only 5.1 (which also correlates with its size).

Can I have a small sample, please? 20MB should do - but please double check whether the problem also occurs with the sample. Thanks!

[Jeff Flowerday]

I've got some 6.1 DTS-ES and DTS Hi Res that I want to convert into flac. Is there anyway to tell it to create 7.1 flac by doubling that back channel?

Not sure if madflac isn't liking the resultant flac from eac3to or the resultant PCM from madflac isn't being liked by the audio renderer. Either way I'm getting no sound and graphedit won't render the mkv.

[madshi]

Originally Posted by Jeff Flowerday

I've got some 6.1 DTS-ES and DTS Hi Res that I want to convert into flac. Is there anyway to tell it to create 7.1 flac by doubling that back channel?

Not sure if madflac isn't liking the resultant flac from eac3to or the resultant PCM from madflac isn't being liked by the audio renderer. Either way I'm getting no sound and graphedit won't render the mkv.

You can use the "-double7" option. Or maybe you can also use the ffdshow raw audio processor to do a similar thing at runtime? Not sure...

[Snowknight26]

When running a DTS track through eac3to just to get its info (eac3to file.dts), would it be possible to state whether it was padded or not? The only way to find out at the moment is to output a DTS file (eacto input.dts output.dts).

[Jeff Flowerday]

Originally Posted by madshi

You can use the "-double7" option. Or maybe you can also use the ffdshow raw audio processor to do a similar thing at runtime? Not sure...

Not sure about ffdshow either, it's easier to just do convert to 7.1

Doing it right now, Thanks!!!

[kurt]

Originally Posted by banker_rishad

Can anybody help with eac3to guides like to how to%. Step by step guide. madshi plz advise

I don't know a specific guide for eac3to but there is at least this wiki: http://en.wikibooks.org/wiki/Eac3to/How_to_Use

Maybe it'll help you...

Originally Posted by quantum

I've found .TS works better on my Networked Media Tank ...

that's because the NMT can decode TS/M2TS in hardware. mkv not. I'm also going over raw vc1 (from eac3to) to TS with tsmuxer and didn't experience any problems so far....

[madshi]

Originally Posted by bigotti5

Stream consists of single encoded fields but PES packet header containing PTS spans always two fields.
Duration of clips is always a multiple of 40 ms (PAL).

I've found that while what you say is correct for your sample, it's not correct for German PAL HDTV broadcasts. Here every single interlaced fields seems to be stored in its own PES packet and has its own PTS value. So the duration is 20ms for these broadcasts.

I'm now using the average video PES duration. That seems to work pretty well.

Originally Posted by Snowknight26

When running a DTS track through eac3to just to get its info (eac3to file.dts), would it be possible to state whether it was padded or not? The only way to find out at the moment is to output a DTS file (eacto input.dts output.dts).

Done.

[madshi]

eac3to v2.79 released

http://madshi.net/eac3to.zip

* improved m2ts file joining overlap detection (mainly for interlaced video)
* vob/evo audio delay detection now uses "vobu start presentation time"
* program streams which are neither VOB nor EVO are now reported as "MPG"
* resampling is now automatically activated for AC3/DTS encoding, if necessary
* "Mersenne Twister" random number generator is used for dithering now
* zero padded DTS tracks are now displayed as such
* fixed: 32bit PCM conversion to floating point was broken
* fixed: with some (rare) movies first subtitle began after 50 minutes runtime
* only plugins with the extension *.dll are loaded now

[Snowknight26]

When Haali Media Splitter isn't installed and I specify .mkv as the output container for a video stream, this happens:

[v02] Extracting video track number 2...
[v02] Muxing video to Matroska...
[v02] Getting "Haali Matroska Muxer" instance failed.
----

The progress bar keeps going so I just cancelled it.

When I do have it installed, however, as I reported before, it still stops when the output file size is 2,752,512 bytes. Any way to track that issue down? Maybe a debug build of some sort?

[madshi]

Originally Posted by Snowknight26

it still stops when the output file size is 2,752,512 bytes. Any way to track that issue down?

Try muxing with gdsmux. Does that also freeze? Which video format is the movie in? Eventually there are too many frames without a new key frame. That usually makes the Haali Muxer freeze, unfortunately...

[Snowknight26]

It happens with any video, be it VC-1, H.264, etc. I've tried nearly a dozen sources.

With gdsmux, when I use the 1st file from Die Another Day (00130.m2ts [H.264] - Blu-ray is seamlessly branched) and I have all the tracks checked, gdsmux goes from 0-100% but the output mkv file size is 4,325,376 bytes. When I select only the video stream, it stopped at ~11% and the file size was the same. I checked any audio stream, and it did the same 0-100% thing, same output file size. Unchecked the audio stream once again to make sure, and it again stopped at 11.2% with an output file size of 4,325,376 bytes.

Tried it again with the Doomsday Blu-ray (00000.m2ts [VC-1]) and with all tracks selected, goes form 0-100% but the output is still 4,325,376 bytes. Unchecked the audio tracks, % stopped at 0.8 this time.

Tried it with the 2 Fast 2 Furious HD DVD (FEATURE_1.EVO [VC-1]), stopped at 8,462,336 bytes. The Apollo 13 HD DVD (PEVOB_1.EVO [VC-1]) stopped at 4,325,376.

Tried an mkv as a source file but it didn't work either.

Won't work with ES video tracks either but thats because there is 'no combination of intermediate filters [...] to make the connection.'

[madshi]

Originally Posted by Snowknight26

It happens with any video, be it VC-1, H.264, etc. I've tried nearly a dozen sources. Will test gdsmux though.

If it's source independent then it probably indicates a general problem with your PC/installation, because video muxing seems to be working for everyone else. My suggestion would be to uninstall Haali's Media Splitter, then cleanup the harddisk and the registry to remove all what might be left, then reinstall Haali. Maybe that helps, maybe not.

[Snowknight26]

I've done that several times now. Removed files, registry entries, etc, still nothing. (Edited my post above.)

[bigotti5]

Concatenating my Cam files work like a charm now.....
Thx

I've found that while what you say is correct for your sample, it's not correct for German PAL HDTV broadcasts. Here every single interlaced fields seems to be stored in its own PES packet and has its own PTS value. So the duration is 20ms for these broadcasts.

I'm now using the average video PES duration. That seems to work pretty well.

Yes - but video output length has to be a multiple of 40 ms

[madshi]

Originally Posted by Snowknight26

With gdsmux, when I use the 1st file from Die Another Day (00130.m2ts [H.264] - Blu-ray is seamlessly branched) and I have all the tracks checked, gdsmux goes from 0-100% but the output mkv file size is 4,325,376 bytes.

Sorry, mate, but this is a problem which is obviously not related in any way to eac3to. It seems that something with your Haali installation is broken. Can't help you with that, unfortunately...

[survivant001]

what that means ?

eac3to v2.79
command line: "D:\DVD-tools\RipBot264v1.11.5\Tools\eac3to\eac3to.exe" "F:\Batman-7.ts" 2:"C:\temp\RipBot264temp\job3\audio.1.mp2" -progressnumbers
------------------------------------------------------------------------------
TRP, 1 video track, 2 audio tracks
1: MPEG2, 704x480 60i /1.001 (4:3)
2: MP2, 2.0 channels, 160kbps, 48khz, -226ms
3: MP2, 2.0 channels, 160kbps, 48khz, 15038ms
[a02] Extracting audio track number 2...
[a02] Applying MPx delay...
[a02] Creating file "C:\temp\RipBot264temp\job3\audio.1.mp2"...
[a02] This track is not clean. Processing aborted.
[a02] Please clean the track with delaycut and then retry eac3to.
Aborted at file position 303874048.

how can I fix that ?

[Snowknight26]

Originally Posted by madshi

Sorry, mate, but this is a problem which is obviously not related in any way to eac3to. It seems that something with your Haali installation is broken. Can't help you with that, unfortunately...

How about internal matroska read/write support?

[madshi]

Originally Posted by survivant001

what that means ?

[a02] This track is not clean. Processing aborted.

how can I fix that ?

That means that your source file is probably damaged/corrupt. eac3to currently doesn't handle such files well. You'll have to use a different tool for extracting the video/audio tracks from this TS file.

Originally Posted by Snowknight26

How about internal matroska read/write support?

So many other important things to do first...

[madshi]

For those interested, the eac3to (SSRC) resampling graphs are now online here:

http://src.infinitewave.ca

As far as I can see, eac3to belongs into the top group of steep resamplers. It's intentionally not as steep as the original SSRC algorithm, but still belongs to the steepest algorithms in the comparison. However, if you are willing to sacrifice high frequency response and if you don't mind some aliasing artifacts, there are other resamplers which have noticeably less ringing. I've learned that there's not one "best" resampler. Going steeper gives you get better high frequency response, but you buy it with stronger ringing. So the "best" resampling algorithm/parameters depend on the material and also on your taste...

[survivant001]

@madshi

2.79 works with my file that I tried to convert, but I'm not able to extract the audio track from this file : (10 megs)

http://www.mediafire.com/?sharekey=f...db6fb9a8902bda

still a cartoon reordered from my FTA.

here the info from mediainfo


General
Complete name : D:\DVD-convertion\done\Batman-5.TSSplit.1-57.ts
Format : MPEG-TS
Format profile : No PAT/PMT
File size : 10.0 MiB
Duration : 1mn 18s
Overall bit rate : 1 067 Kbps

Video
ID : 6690 (0x1A22)
Format : MPEG Video
Format version : Version 2
Format profile : Main@Main
Format settings, Matrix : Default
Duration : 1mn 18s
Bit rate mode : Constant
Bit rate : 705 Kbps
Nominal bit rate : 809 Kbps
Width : 704 pixels
Height : 480 pixels
Display aspect ratio : 4/3
Frame rate : 29.970 fps
Colorimetry : 4:2:0
Bits/(Pixel*Frame) : 0.080

Audio #1
ID : 6691 (0x1A23)
Format : MPEG Audio
Format version : Version 1
Format profile : Layer 2
Bit rate mode : Constant
Bit rate : 160 Kbps
Channel(s) : 2 channels
Sampling rate : 48.0 KHz
Resolution : 16 bits
Video delay : -742ms

Audio #2
ID : 6692 (0x1A24)
Format : MPEG Audio
Format version : Version 1
Format profile : Layer 2
Bit rate mode : Constant
Bit rate : 160 Kbps
Channel(s) : 2 channels
Sampling rate : 48.0 KHz
Resolution : 16 bits
Video delay : -742ms

[survivant001]

Originally Posted by madshi

That means that your source file is probably damaged/corrupt. eac3to currently doesn't handle such files well. You'll have to use a different tool for extracting the video/audio tracks from this TS file.


So many other important things to do first...

can you suggest me a tool that can do that ?

do you think you can add where the audio track failed ? like 341megs : 23:34.112

like that I could just split my movie.. and convert before and after that time.

[madshi]

Originally Posted by survivant001

can you suggest me a tool that can do that ?

You can use TsRemux or tsMuxeR to do the demuxing/extracting. Then you can run the audio track(s) through delaycut to fix them. Finally, if there's any other audio processing, you can then use eac3to.

Originally Posted by survivant001

do you think you can add where the audio track failed ? like 341megs : 23:34.112

It's already there: "Aborted at file position 303874048."

So the problem is somewhere around 303874048 bytes (probably a few bytes before that).

[asarian]

Madshi (or anyone else knowledgeable in these matters), i have a Japanese LCPM 2.0 sound track (Macross Frontier, Blu-Ray), which I'd like to convert to DTS 1.5 Mb/s, if possible. I have SurCode 1.0.29. But when I run eac3to, like this:

eac3to 00002.m2ts 3: c:\video\mf4.dts

Then eac3to starts to create two wav files (for left and right, it seems). Not quite what I was looking for. Am I missing something? I've done a lot of DTS decoding, just never encoding.

Thanks

[nautilus7]

It's simple, your source track is 2.0 only. Thus the DTS will be stereo.

[asarian]

Originally Posted by nautilus7

It's simple, your source track is 2.0 only. Thus the DTS will be stereo.

Thanks. I see the two wavs were just an intermediary state, prior to Surcode starting.

[nautilus7]

Oh, your question had to do with the existence of the wav files, not the number of them...

Yes, surcode needs to be fed with mono wav files.

[asarian]

Hmm, on my Vmware box I now get the following:

command line: eac3to 00002.m2ts 3: c:\video\mf2.dts -1536
------------------------------------------------------------------------------
M2TS, 1 video track, 1 audio track, 1:12:28
1: Chapters, 7 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: RAW/PCM, Japanese, 2.0 channels, 16 bits, 48khz
[a03] Extracting audio track number 3...
[a03] Reading RAW/PCM...
[a03] Swapping endian...
[a03] Writing WAVs...
[a03] Creating file "c:\video\mf2.R.wav"...
[a03] Creating file "c:\video\mf2.L.wav"...
[a03] The original audio track has a constant bit depth of 16 bits.
Encoding DTS <1536kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.29.0.
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...

The eac3to FAQ says: "Surcode doesn't like long filenames/paths. Change them and you 'll be ok." I don't see any long filenames, though. Anyone else has any idea?

Thanks

[nautilus7]

It should be the vmware. Surcode is generally a strange application.

[tebasuna51]

Originally Posted by asarian

i have a Japanese LCPM 2.0 sound track (Macross Frontier, Blu-Ray), which I'd like to convert to DTS 1.5 Mb/s, if possible. I have SurCode 1.0.29. But when I run eac3to, like this:

eac3to 00002.m2ts 3: c:\video\mf4.dts

Then eac3to starts to create two wav files (for left and right, it seems). Not quite what I was looking for. Am I missing something? I've done a lot of DTS decoding, just never encoding.

Surcode only can make 5.0 or 5.1 encodes.

BTW, I can't understand why you need convert a lossless format:
LPCM 2.0 16 bit 48 KHz -> 48000 x 16 x 2 = 1536 Kb/s
to a lossy format:
DTS 2.0 1536 Kb/s

Use flac to less space or preserve LPCM

[asarian]

Originally Posted by tebasuna51

Surcode only can make 5.0 or 5.1 encodes.

BTW, I can't understand why you need convert a lossless format:
LPCM 2.0 16 bit 48 KHz -> 48000 x 16 x 2 = 1536 Kb/s
to a lossy format:
DTS 2.0 1536 Kb/s

I don't have an HDMI 1.3 receiver (yet). So, since I'll be streaming this BD series to my PS3, using the optical out, I figured I wouldn't be able to do LPCM over the 'bitstream' channel. But now I'm not so sure anymore. it seems "LPCM 2.0 48Khz" is supported over optical. So maybe it's just 7.1 LPCM that needs to be done over HDMI per se? This warrants some tests.

[tebasuna51]

Originally Posted by asarian

I don't have an HDMI 1.3 receiver (yet). So, since I'll be streaming this BD series to my PS3, using the optical out, I figured I wouldn't be able to do LPCM over the 'bitstream' channel. But now I'm not so sure anymore. it seems "LPCM 2.0 48Khz" is supported over optical. So maybe it's just 7.1 LPCM that needs to be done over HDMI per se?

Exact.
"LPCM 2.0 48Khz" is supported over optical.
7.1 LPCM needs to be done over HDMI.

[Snowknight26]

eac3to can't detect this FLAC track but madFLAC Source accepts it and is decoded fine with ffdshow.. well, apart from time issue.
http://www.stfcc.org/misc/departed.flac

Only thing I can do is make a graph of madFLAC Source -> ffdshow audio decoder -> Dump, then use eac3to to convert that PCM track to FLAC. :P

[madshi]

Originally Posted by Snowknight26

eac3to can't detect this FLAC track but madFLAC Source accepts it and is decoded fine with ffdshow.. well, apart from time issue.

Will be fixed in the next build.

[shanghai2004]

Originally Posted by madshi

eac3to v2.79 released

http://madshi.net/eac3to.zip

* improved m2ts file joining overlap detection (mainly for interlaced video)
* vob/evo audio delay detection now uses "vobu start presentation time"
* program streams which are neither VOB nor EVO are now reported as "MPG"
* resampling is now automatically activated for AC3/DTS encoding, if necessary
* "Mersenne Twister" random number generator is used for dithering now
* zero padded DTS tracks are now displayed as such
* fixed: 32bit PCM conversion to floating point was broken
* fixed: with some (rare) movies first subtitle began after 50 minutes runtime
* only plugins with the extension *.dll are loaded now

Madshi, seems somthing wrong now....

F:\Chicago\HVDVD_TS>c:\eac3to\eac3to hv001t01.evo+hv001t02.evo+hv001t03.evo+hv00
1t04.evo+hv001t05.evo+hv001t06.evo+hv001t07.evo+hv001t08.evo+hv001t09.evo+hv001t
10.evo+hv001t11.evo+hv001t12.evo+hv001t13.evo+hv001t14.evo+hv001t15.evo+hv001t16
.evo+hv001t17.evo+hv001t18.evo+hv001t19.evo+hv001t20.evo+hv001t21.evo+hv001t22.e
vo+hv001t23.evo+hv001t24.evo+hv001t25.evo+hv001t26.evo
EVO, 1 video track, 2 audio tracks, 1 subtitle track, 14:47:35
1: Joined EVO file
2: h264/AVC, 1080i60 /1.001 (16:9)
3: E-AC3, 5.1 channels, 894kbps, 48khz
(core: E-AC3, 5.1 channels, 3024kbps, 48khz)
4: E-AC3, 2.0 channels, 448kbps, 48khz
5: Subtitle

Wrong duration is listed and something seems to be wrong with the 5.1 track listing. When the files are processed, an endless list of 13ms audio overlaps are reported in the 5.1 track.

Tried eeac3to v2.79 and v2.78 with this result, same EVO files where processed before with older version of eac3to without problems (cannot remember what version though... if important I can check)

[madshi]

Originally Posted by shanghai2004

Wrong duration is listed and something seems to be wrong with the 5.1 track listing. When the files are processed, an endless list of 13ms audio overlaps are reported in the 5.1 track.

Tried eeac3to v2.79 and v2.78 with this result, same EVO files where processed before with older version of eac3to without problems (cannot remember what version though... if important I can check)

Can you upload a little sample for me? Maybe 2 of those EVO parts which are rather small (if there are any such)?

[Snowknight26]

Originally Posted by madshi

Will be fixed in the next build.

Just out of curiosity, what was the issue? Bad FLAC track possibly?

[Thunderbolt8]

Originally Posted by madshi

Don't know how to do that. Do you have a few 2.0 mono samples?

this ac3 track here is 2.0 and at least supposed to be mono:

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

[madshi]

Originally Posted by Snowknight26

Just out of curiosity, what was the issue? Bad FLAC track possibly?

No runtime information in the FLAC track. That threw eac3to off.

Originally Posted by Thunderbolt8

this ac3 track here is 2.0 and at least supposed to be mono

Unfortunately this behaves just like any true stereo track does. The header says stereo, there are 2 full channels in there and they are *not* bit perfect identical. So I don't see any reasonable way to find out that this track is mono instead of stereo. Ok, technically I could probably decode the whole track and check whether there are any "big" differences in the waveform anywhere. But I don't think it's worth it...

[alc0re]

Why is it that whenever I check any dts file extracted by eac3to with MediaInfo, the length is always slightly shorter than the video or any ac3 file extracted?

Examples:

1) Skinwalkers Bluray

eac3to v2.79
command line: eac3to c:\BDRip\Skinwalkers 2) 1:"C:\BDRip\Skinwalkers\Demuxed\Chapters.txt" 2:"C:\BDRip\Skinwalkers\Demuxed\Skinwalkers.mkv" 4:"C:\BDRip\Skinwalkers\Demuxed\Skinwalkers.dts" -core
------------------------------------------------------------------------------
M2TS, 1 video track, 3 audio tracks, 2 subtitle tracks, 1:31:50
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 (16:9)
3: DTS Master Audio, French, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)

Media Info on Video MKV : 1h 31mn
Media Info on dts : 1h 30mn

2) The Untouchables Bluray

eac3to v2.78
command line: eac3to c:\BDRip\TheUntouchables 1) 1:"C:\BDRip\TheUntouchables\Demuxed\Chapters.txt" 2:"C:\BDRip\TheUntouchables\Demuxed\TheUntouchables.mkv" 3:"C:\BDRip\TheUntouchables\Demuxed\TheUntouchables.ac3" 4:"C:\BDRip\TheUntouchables\Demuxed\TheUntouchables.dts"
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 4 subtitle tracks, 1:59:27
1: Chapters, 24 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: AC3 EX, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
4: DTS-ES, English, 6.1 channels, 24 bits, 1509kbps, 48khz, dialnorm: -4dB

Media Info on Video MKV : 1h 59mn
Media Info on Audio DTS : 1h 57mn
Media Info on Audio AC3 : 1h 59mn


Am I doing something wrong? Am I mistaken to think this is going to cause an audio sync issue as the movie gets further along? Could this be a bug with MediaInfo?

Its also listing the skinwalkers core dts file as 1536Kbps when eac3to said its 1509. I open the mkv and the ac3 files I get in Zoom player and they both are the correct length for the movie, but I can't get .dts files to open in Zoom player to check the lenght of the DTS file...and I do have ac3filter installed...

[Snowknight26]

Probably a calculation bug. 1:59:27 * (1509/1536) = 1:57:21.

[alc0re]

Yah, I think you're right about a calculation error, although it doesnt do it with .ac3 files. Anyways, I kinda figured out its fine.

I just muxed the .dts files in question by themselves to .m2ts files with TSMuxer, and checked the resulting .m2ts file. They are the correct length.

[madshi]

Originally Posted by alc0re

Why is it that whenever I check any dts file extracted by eac3to with MediaInfo, the length is always slightly shorter than the video or any ac3 file extracted?

What length is displayed if you do "eac3to Skinwalkers.dts"?

[asarian]

Hello,

I'm not sure whether this is a tsMuxeR issue or an eac3to one (but I assume the latter, sorry), but when I use eac3to to extract a RAW/PCM stream from a BD of Macross Frontier, tsMuxeR doesn't recognize the resultant .pcm file any more ("Can't detect stream type"), whereas it is recognized when I select it directly from the m2ts. I uploaded a sample:

3: RAW/PCM, Japanese, 2.0 channels, 16 bits, 48khz
http://rapidshare.com/files/169776965/mf2-test.pcm.html

I extracted it with the .pcm extension.

Thanks

[G_M_C]

Originally Posted by asarian

Hello,

I'm not sure whether this is a tsMuxeR issue or an eac3to one (but I assume the latter, sorry), but when I use eac3to to extract a RAW/PCM stream from a BD of Macross Frontier, tsMuxeR doesn't recognize the resultant .pcm file any more ("Can't detect stream type"), whereas it is recognized when I select it directly from the m2ts. I uploaded a sample:

3: RAW/PCM, Japanese, 2.0 channels, 16 bits, 48khz
http://rapidshare.com/files/169776965/mf2-test.pcm.html

I extracted it with the .pcm extension.

Thanks

It's a known tsMuxeR problem, go to that thread and



You'll get to a tool that converts the BD PCM track to a format that tsMuxeR can use.

[tebasuna51]

Originally Posted by asarian

I'm not sure whether this is a tsMuxeR issue or an eac3to one (but I assume the latter, sorry), but when I use eac3to to extract a RAW/PCM stream from a BD of Macross Frontier, tsMuxeR doesn't recognize the resultant .pcm file any more ("Can't detect stream type"), whereas it is recognized when I select it directly from the m2ts.

Is your issue. TsMuxer don't accept lpcm files. These files are raw audio data without header and can't be recognized out of a container than inform about bitdepth, channels, samplerate and endian.

Select wav like output file and can be recognized by TsMuxer if is <4GB (probably because 2 C and 16 bit). For wav files > 4GB (5.1 and > 130 min.) you need pcm output and Pcm2Tsmu.

[madshi]

Originally Posted by Zwitterion

Another idea:
DIRAC time stretching. The LE version is free to implement.
I don't know any free tool that does high-quality timestretching, so that would definitely be another killer feature and very useful for 23.976 --> 25 audio conversions.

I've looked into this. The DIRAC documentation contains this text:

2.3 Phase locked multi-channel processing vs. multiple channel processing

The STUDIO version of DIRAC supports stereo while DIRAC PRO supports an infinite number of
channels (memory permitting) that it can process in a phase-locked (synced) manner at the same time. All of these
simultaneous channels are being processed using a phase-locked processing algorithm that ensures that the stereo
(or surround/multi-channel) phase relationship is preserved.

It is important to understand how this works and what this means exactly.

In a stereo recording, important localization cues are provided to the listener through the relative timing of a sound
source between the left and the right ear (channel). If a time stretching process changes the relative timing of the
two channels by even a minimal amount, the stereo image will be perceived as “distorted”. Also, mono
compatibility will no longer be guaranteed, which means that if you mix down the two stereo channels to a mono
channel (as is the case in some TV and radio equipment) you will end up with very audible artifacts perceived as
phasing or even cancellations.

If you have the situation that the relative phase between channels matters, it is imperative to use the multi-channel
processing mode of DIRAC STUDIO and PRO (all channels are being processed at the same time). As a rule of
thumb, phase is always important with stereo recordings, or recordings of the same sound source that were made
simultaneously through different microphones. It is almost always the case with the channels in a surround mix. In
these cases, you should use DIRAC in multi-channel mode, by setting up a single DIRAC object for multiple
channels.

So I contacted the DIRAC company and asked about whether it would make any sense at all to use the free DIRAC version for movie tracks. Here's the reply I received:

"If you are planning on time stretching and pitch shifting 5.1 and 7.1 recordings relative phase is essential. You would need to use the PRO version of DIRAC in order to do this."

In other words: The free DIRAC version is useless for our needs, sadly.

[asarian]

Originally Posted by tebasuna51

Is your issue. TsMuxer don't accept lpcm files. These files are raw audio data without header and can't be recognized out of a container than inform about bitdepth, channels, samplerate and endian.

Select wav like output file and can be recognized by TsMuxer if is <4GB (probably because 2 C and 16 bit). For wav files > 4GB (5.1 and > 130 min.) you need pcm output and Pcm2Tsmu.

Okay, thanks. Hadn't dealt with LPCM before, and didn't realize they were that raw.

[rack04]

Any help with the following error? The source is Hellboy 2 Blu-ray.

eac3to v2.79
command line: eac3to "F:\Blu-ray\HELLBOY2_D1" 1) 1: "F:\Blu-ray\Hellboy 2.txt" 2: "F:\Blu-ray\Hellboy 2.h264" 4: "F:\Blu-ray\Hellboy 2.ac3" 10: "F:\Blu-ray\Hellboy 2.sup"
------------------------------------------------------------------------------
M2TS, 2 video tracks, 6 audio tracks, 5 subtitle tracks, 1:59:49
1: Chapters, 21 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: DTS, Spanish, 5.1 channels, 24 bits, 768kbps, 48khz
6: DTS, French, 5.1 channels, 24 bits, 768kbps, 48khz
7: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
8: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
9: DTS Express, English, 2.0 channels, 24 bits, 192kbps, 48khz
10: Subtitle (PGS), English
11: Subtitle (PGS), Spanish
12: Subtitle (PGS), French
13: Subtitle (PGS), Spanish
14: Subtitle (PGS), French
Creating file "F:\Blu-ray\Hellboy 2.txt"...
[a04] AC3 encoding doesn't support back channels. Will mix them into the surround.
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[s10] Extracting subtitle track number 10...
[a04] Decoding with ArcSoft DTS Decoder...
[a04] Mixing surround channels...
[a04] Encoding AC3 <640kbps> with libAften...
[v02] Creating file "F:\Blu-ray\Hellboy 2.h264"...
[a04] Creating file "F:\Blu-ray\Hellboy 2.ac3"...
[s10] Creating file "F:\Blu-ray\Hellboy 2.sup"...
[a04] This TS/M2TS file seems to be damaged (sync byte missing).
[v02] This TS/M2TS file seems to be damaged (sync byte missing).
[s10] This TS/M2TS file seems to be damaged (sync byte missing).
Aborted at file position 16013541376.

[asarian]

Originally Posted by rack04

Any help with the following error? The source is Hellboy 2 Blu-ray.

eac3to v2.79

[a04] Creating file "F:\Blu-ray\Hellboy 2.ac3"...
[s10] Creating file "F:\Blu-ray\Hellboy 2.sup"...
[a04] This TS/M2TS file seems to be damaged (sync byte missing).
[v02] This TS/M2TS file seems to be damaged (sync byte missing).
[s10] This TS/M2TS file seems to be damaged (sync byte missing).
Aborted at file position 16013541376.

I believe I had the exact same error with this disc. If I recall correctly, what solved it for me was to demux the audio track first with tsMuxeR. Then eac3to would convert the demuxed stream properly.

[Chumbo]

@madshi,
A request please as you have time. When going from the same format to a target of the same format, i.e., using -slowdown for example, can you use the same bitrate as the source by default please? Below is an example of one track I was slowing down and noticed it was being reencoded at 640Kbps rather than just defaulting to what the source is which, in this case, is 448Kbps. All I had to do was use the -448 to make sure the target is at least the same, but it would be nice to have the feature. Thank you.

eac3to v2.79
command line: eac3to movie.1.ts 3: audio.slow.ac3 -slowdown -log=aud-slow.txt
------------------------------------------------------------------------------
TS, 1 video track, 3 audio tracks, 0:10:40
1: h264/AVC, 1080i50 (16:9)
2: AC3, German, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB, -280ms
3: AC3, English, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB, -280ms
4: AC3 Surround, Achinese, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, -280ms
[a03] Extracting audio track number 3...
[a03] Removing AC3 dialog normalization...
[a03] Decoding with DirectShow (Nero Audio Decoder 2)...
[a03] DirectShow reports 5.1 channels, 24 bits, 48khz
[a03] Applying RAW/PCM delay...
[a03] Changing FPS from 25.000 to 23.976...
[a03] Encoding AC3 <640kbps> with libAften...
[a03] Creating file "audio.slow.ac3"...
Video track 1 contains 1075 frames.
eac3to processing took 15 seconds.
Done.

[madshi]

Originally Posted by rack04

Any help with the following error? The source is Hellboy 2 Blu-ray.

[a04] This TS/M2TS file seems to be damaged (sync byte missing).
Aborted at file position 16013541376.

This looks like a damaged source file. Either it's a bad rip, or a bug in AnyDVD HD, or an authoring fault. eac3to only accepts clean sources at this time...

Originally Posted by Chumbo

When going from the same format to a target of the same format, i.e., using -slowdown for example, can you use the same bitrate as the source by default please?

No, I won't do that. When reencoding audio, the source bitrate is totally independent of the target bitrate. Doing 448kbps -> slowdown -> 448kbps results in worse audio quality compared to 448kbps -> slowdown -> 640kbps.

[Beastie Boy]

I have become a bit confused about encoding AC3 from a THD file containing a AC3 core. If I do

eac3to input.thd output.ac3

will this extract the core AC3 track or will it encode a new track from the lossless data. What I am trying to do is encode from the lossless portion since I have read that most of the audible difference between the two tracks can be due to a different mix being used, or different masters.

It is quite clear regarding DTS with the -core option, but I need some advice regarding Dolbly. If necessary, I can convert to FLAC first.

Cheers, Beastie.

[sehgal.v7]

@Beastie
eac3to input.thd output.ac3
It will convert TrueHD to ac3

eac3to input.thd+ac3 output.ac3
It will extract embedded ac3 track.

[madshi]

The file extension of the source file doesn't really matter at all. But basically sehgal.v7 is right: If the source file contains a TrueHD/AC3 interweaved stream, asking eac3to for the AC3 file will result in a simple extract of the studio provided AC3 track. If the source file contains a straight TrueHD track, only (as is the case with HD DVDs), eac3to will encode a new AC3 track. Currently there's no way to directly force the encoding of a new AC3 track, if there's already an existing one, unless you choose one of the modification options (e.g. a different bitrate, or a volume change or something similar). But you can work around this by first converting to a TrueHD only track (name the target file "*.thd") and then in a separate step transcoding that to AC3. And yes, doing an intermediate FLAC step would have the same effect.

[Beastie Boy]

Thanks for the replies. I'll use an intermediate THD only track since this doesn't require any encoding and I can save it for future use (FLAC support on the NMTs maybe )

Cheers, Beastie.

[Chumbo]

Originally Posted by madshi

No, I won't do that. When reencoding audio, the source bitrate is totally independent of the target bitrate. Doing 448kbps -> slowdown -> 448kbps results in worse audio quality compared to 448kbps -> slowdown -> 640kbps.

Wow, I didn't know that. Thanks for the explanation.

[Atak_Snajpera]

eac3to detects incorrect audio delay

eac3to v2.79
command line: "C:\Users\Dawidos\Documents\Delphi_Projects\RipBot264\Tools\eac3to\eac3to.exe"  "D:\_Video_Samples\ts\premiere-paff.ts"
------------------------------------------------------------------------------
TS, 1 video track, 1 audio track, 0:00:29
1: h264/AVC, 1080i50 (16:9)
2: AC3, English, 2.0 channels, 448kbps, 48khz, dialnorm: -27dB, -3404ms

DGAVCDec reports correct value -719 ms.

sample:http://www.mediafire.com/?onyymjatwjq

eac3to v2.79
command line: "C:\Users\Dawidos\Documents\Delphi_Projects\RipBot264\Tools\eac3to\eac3to.exe"  "D:\_Video_Samples\ts\Digiturk.HD.Sirius.2.4.8E.10.jul.2007.ts"
------------------------------------------------------------------------------
TS, 1 video track, 1 audio track, 0:00:24
1: h264/AVC, 1080i50 (16:9)
2: MP2, English, 2.0 channels, 320kbps, 48khz, -2532ms

DGAVCDec reports correct value -980 ms.

http://x264.nl/h.264.samples/force.p...10.jul.2007.ts

[madshi]

Originally Posted by Atak_Snajpera

eac3to detects incorrect audio delay

DGAVCDec reports correct value -719 ms.

You can't directly compare the audio delay values reported by eac3to and DGAVCDec because eac3to removes all video frames before the first sequence header, while DGAVCDec doesn't do that (I think). Because of that eac3to's audio delay values can be higher.

However, the audio delay with your two test streams was not really correct with the current eac3to version. The next build will use a different delay calculation for such streams with video frames before the first sequence header. The delay numbers will be higher compared to DGAVCDec, but audio should be in sync.

[Ryu77]

Madshi,

Previously I was using a trial version of ArcSoft TotalMedia Theatre and eac3to recognised the DTS decoder without a problem.

I found a 40% discount coupon for this software so I decided to purchase it. I uninstalled the trial version, then installed the retail version, now eac3to informs me that the DTS decoder isn't installed. Any idea why?

[Jeff Flowerday]

Originally Posted by Ryu77

Madshi,

Previously I was using a trial version of ArcSoft TotalMedia Theatre and eac3to recognised the DTS decoder without a problem.

I found a 40% discount coupon for this software so I decided to purchase it. I uninstalled the trial version, then installed the retail version, now eac3to informs me that the DTS decoder isn't installed. Any idea why?

Is the TMT bin directory in your path? If not give that a try.

[Ryu77]

Originally Posted by Jeff Flowerday

Is the TMT bin directory in your path? If not give that a try.

I am not sure exactly what you are asking... Would you be able to clarify exactly what I should check?

Thank you. :-)

Also, I was wondering if it must be the TotalMedia Extreme suite or is it ok to only install TotalMedia Theatre?

I ask this because I previously had TotalMedia Extreme trial installed and now I only purchased TotalMedia Theatre retail.

[odin24]

Originally Posted by Ryu77

I am not sure exactly what you are asking... Would you be able to clarify exactly what I should check?

Thank you. :-)

Also, I was wondering if it must be the TotalMedia Extreme suite or is it ok to only install TotalMedia Theatre?

I ask this because I previously had TotalMedia Extreme trial installed and now I only purchased TotalMedia Theatre retail.

TM Theatre should be fine, it's all I have installed and the decoder works fine.

Could the old install directory still have the trial information. Maybe uninstall everything Arcsoft, delete the old install directories then re-install your full version.

Who know, it might work.

[madshi]

Search for "environment" in this thread. Adding the ArcSoft DLL path to that variable has helped some people in the past...

[Ryu77]

Originally Posted by madshi

Search for "environment" in this thread. Adding the ArcSoft DLL path to that variable has helped some people in the past...

Still not working...

I added the path as suggested. My full path environment variable is as follows... %SystemRoot%\system32;%SystemRoot%;%SystemRoot%\System32\Wbem;C:\Program Files\Common Files\ArcSoft\Bin\

The strange thing is I have tried every way imaginable. I have installed it over a trial version where the DTS decoder was working, the full version then removes the ability for eac3to to access it. I have also tried installing the new TotalMedia Theatre on a fresh install of Windows... Nothing seems to be working.

I installed TotalMedia Theatre v2.1.6.126 if that helps.

[komisar]

Ryu77, try:

md c:\as
copy "%CommonProgramFiles%\ArcSoft\Bin\*.*" c:\as
copy "%CommonProgramFiles%\ArcSoft\MPEG Engine\*.*" c:\as

and add to you PATH environment variable c:\as

[Ryu77]

Originally Posted by komisar

Ryu77, try:

md c:\as
copy "%CommonProgramFiles%\ArcSoft\Bin\*.*" c:\as
copy "%CommonProgramFiles%\ArcSoft\MPEG Engine\*.*" c:\as

and add to you PATH environment variable c:\as

Ok, I've added those two lines "%CommonProgramFiles%\ArcSoft\Bin\*.*" & "%CommonProgramFiles%\ArcSoft\MPEG Engine\*.*" without the quotation marks... I am not sure what you mean by adding c:\as?

[komisar]

Ryu77, open cmd.exe and enter this 3 command. Go to "MyComputer->Properties->Advanced->Environment Variables" and add in "User variables" variable Path with value c:\as. If you already have Path variable, add to end of value ;c:\as

[Ryu77]

Thank you for your help komisar... But still no luck.

I have added the C:\Program Files\Common Files\ArcSoft\Bin and C:\as to the path in environment variables and I have run those command prompts you suggested but it's still not recognising the DTS decoder.

I'm starting to feel like I was better off with a "trial" version.

[komisar]

Ryu77, one more explanation. You need all files from ArcSoft\Bin and ArcSoft\MPEG Engine in ONE folder (in my example c:\as). Also you need point Path to this folder (in my example c:\as).
P.S. I also spent a lot of time searching for the decision. And now this work for me...

[Ryu77]

Originally Posted by komisar

Ryu77, one more explanation. You need all files from ArcSoft\Bin and ArcSoft\MPEG Engine in ONE folder (in my example c:\as). Also you need point Path to this folder (in my example c:\as).
P.S. I also spent a lot of time searching for the decision. And now this work for me...

I do appreciate your persistence with helping me but nothing seems to be working.

I really can't understand this. I checked my other PC that has the DTS decoder working and I can see that C:\Program Files\Common Files\ArcSoft\Bin has been already added to the environment variables. This must have been done automatically with the older trial install. I guess newer retail versions don't do that anymore. However, one would think manually adding it in as we have done would rectify this... In my case this doesn't seem to work. :-(

[asarian]

Originally Posted by Ryu77

Thank you for your help komisar... But still no luck.

I have added the C:\Program Files\Common Files\ArcSoft\Bin and C:\as to the path in environment variables and I have run those command prompts you suggested but it's still not recognising the DTS decoder.

I'm starting to feel like I was better off with a "trial" version.

TotalMedia Theatre, in the commercial variant of TotalMedia Extreme (which I have purchased too), is at:

C:\Program Files\ArcSoft\TotalMedia Extreme\Digital Theatre

You don't need to add anything to your path; though I'd add eac3to to your path, like: "C:\Program Files\eac3to" (or wherever it resides on your system)

[Ryu77]

Thank you asarian but I purchased TotalMedia Theatre not the Extreme suite.

I fixed my problem now...

This is what I did...

1) Uninstalled full retail version, then did a restart.

2) Installed trial version, tested eac3to and it worked, then restarted.

3) Tested eac3to again, still good. Ran full retail install again but chose "repair".

It's all good now. The trial (or older versions) must do something that the newer full version doesn't. Oh well, it's all working now... Yay. :-D

For anyone that's interested in purchasing ArcSoft TotalMedia Theatre or TotalMedia Extreme, the code that I used to get 40% off is "earlyxmas09". Just type this into the field marked "Please enter your coupon code here:" and you will receive 40% off.

[Thunderbolt8]

got an Arcsoft error when decoding a movie's DTS-HD MA track to flac:

eac3to v2.79
command line: eac3to x:\path 1) 4: x:\test.flac
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 6 subtitle tracks, 3:08:47
1: Chapters, 10 chapters
2: h264/AVC, 1080p24 (16:9)
3: RAW/PCM, Swedish, 1.0 channels, 24 bits, 48khz
4: DTS Master Audio, Swedish, 1.0 channels, 24 bits, 48khz
   (core: DTS, 1.0 channels, 24 bits, 768kbps, 48khz)
5: Subtitle (PGS), Dutch
6: Subtitle (PGS), Swedish
7: Subtitle (PGS), Norwegian
8: Subtitle (PGS), Finnish
9: Subtitle (PGS), English
10: Subtitle (PGS), Swedish
[a04] Extracting audio track number 4...
[a04] Decoding with ArcSoft DTS Decoder...
[a04] The ArcSoft DTS Decoder reported an error while decoding.
Aborted at file position 16384.

the movie consists of 2 main .m2ts files, its similar with the 2nd one (position 32768). I've added 2x50mb samples of both here:

http://www.sendspace.com/file/d5yek8

and this happened when I used the -sonic switch for that track instead:

eac3to v2.79
command line: eac3to x:\path 1) 2: G:\test.mkv 3: G:\testPCM.flac 4: G:\testDTSMA.flac -sonic 4: G:\test.dtsma 9: G:\test.sup
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 6 subtitle tracks, 3:08:47
1: Chapters, 10 chapters
2: h264/AVC, 1080p24 (16:9)
3: RAW/PCM, Swedish, 1.0 channels, 24 bits, 48khz
4: DTS Master Audio, Swedish, 1.0 channels, 24 bits, 48khz
   (core: DTS, 1.0 channels, 24 bits, 768kbps, 48khz)
5: Subtitle (PGS), Dutch
6: Subtitle (PGS), Swedish
7: Subtitle (PGS), Norwegian
8: Subtitle (PGS), Finnish
9: Subtitle (PGS), English
10: Subtitle (PGS), Swedish
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a04] Extracting audio track number 4...
[s09] Extracting subtitle track number 9...
[a04] Extracting audio track number 4...
[a03] Reading RAW/PCM...
[v02] Muxing video to Matroska...
[a04] Decoding with DirectShow (Sonic Audio Decoder)...
[a03] Swapping endian...
[a03] Encoding FLAC with libFlac...
[a04] Creating file "G:\test.dtsma"...
[a03] Creating file "G:\testPCM.flac"...
[s09] Creating file "G:\test.sup"...
[a03] The original audio track has a constant bit depth of 24 bits.
[a03] The processed audio track has a constant bit depth of 24 bits.
[a04] The last DTS frame is incomplete and thus gets skipped.
[a04] The FLAC encoder didn't receive the format information.
Aborted at file position 45477599232.

-sonic doesn't give me any output file, nor the "Creating file" message and aborted in the end :S

btw. when I decode the 1.0 RAW/PCM track to .flac I get the "Swapping endian" message, it this intended to appear for 1.0 or 2.0 tracks?

[WildTexasChef]

Am I mistaken or will eac3to NOT convert DTS audio to AC3?

I tried running eac3to to convert a DTS audio track to AC3 at a bit rate of 448 and it would not convert. It would work fine at a 640k bit rate but not at 448.

Any ideas?

WTC

[nautilus7]

Post you exact command line and log file.

[n0mag!c]

Originally Posted by madshi

You would need to use the PRO version of DIRAC in order to do this.

I dare to disagree with this statement. If you figure this out further this would mean that each time when we do stretching we'll get different result, not the same. Our fear and ignorance are warmed up by smart marketing stuff. But this is easy to check up! We must process stereo file and two mono channels independently and just compare results.

Originally Posted by madshi

If you have the situation that the relative phase between channels matters, it is imperative to use the multi-channel
processing mode of DIRAC STUDIO and PRO (all channels are being processed at the same time).

I don't really think that this algorithm PURPOSELY produces different result each time it runs.

[vucloutr]

Hi, I got a little question regarding AAC encoding.

eac3to can encode AAC in quality ranging from (-quality=) 0.01 to 0.99
but the Nero AAC encoder which eac3to uses can do a float from 0..1.
So how is the eac3to range mapped to the Nero AAC encoder range?
Does eac3to 0.99 mean Nero 1.00 or does eac3to 0.99 mean Nero 0.99 (which is what I assume) ?
If it is the latter and only a minor change to make:
Could you be so kind to allow also -quality=0.00 and 1.00, so that one can use the full range the Neor AAC encoder offers ?

[madshi]

Originally Posted by Thunderbolt8

got an Arcsoft error when decoding a movie's DTS-HD MA track to flac

Interesting sample, thanks.

It seems that neither ArcSoft nor Sonic can decode this one. They don't seem to like mono DTS tracks. You'll have to use either Nero or libav to decode it. Of course both only decode the core...

Originally Posted by n0mag!c

If you figure this out further this would mean that each time when we do stretching we'll get different result, not the same.

No, that's not true.

Originally Posted by n0mag!c

Our fear and ignorance are warmed up by smart marketing stuff.

I talked to the developer of the algorithm, not to the marketing department...

Originally Posted by n0mag!c

I don't really think that this algorithm PURPOSELY produces different result each time it runs.

It produces the same result when running on the same data. But the part you didn't understand is that the algorithm adopts itself to the waveform of the audio track. So if you split a stereo file into 2 separate mono files and process each separately from each other, the DIRAC algorithm may do different things to each channel, depending on the waveform. And this can distort stereo phase information. Same thing with surround tracks.

If you use phase locked processing (only available in the non-free DIRAC versions), the DIRAC algorithm makes sure that it does exactly the same thing to all channels. This makes sure that stereo and surround phase information is not distorted.

Originally Posted by vucloutr

Could you be so kind to allow also -quality=0.00 and 1.00, so that one can use the full range the Neor AAC encoder offers ?

Will be added in the next build.

[Thunderbolt8]

Originally Posted by madshi

Interesting sample, thanks.

It seems that neither ArcSoft nor Sonic can decode this one. They don't seem to like mono DTS tracks. You'll have to use either Nero or libav to decode it. Of course both only decode the core...

does any of these 2 still gives out a slightly better quality or are both equally bad here?

[madshi]

Originally Posted by Thunderbolt8

does any of these 2 still gives out a slightly better quality or are both equally bad here?

I don't know. Nero has a reference decoder, so it might be better. Or not...

[xkodi]

@madshi

actually, Thunderbolt8's DTS-HD MA mono samples work with Arcsoft in GraphEdit.

i set the "ArcSoft Audio Decoder HD" to decode to "2 channel stereo" and the result is file with empty Right channel (all zeroes) and all the information is in the Left channel. so, it is save to assume that the decode is bit-perfect.

however, if i use eac3to to force Arcsoft to use 2 channels:

C:\eac3to279>eac3to.exe c:\FAAsplit\FAAsplit1.m2ts 3:c:\1.wav -2

i get error:

[a03] Decoding with ArcSoft DTS Decoder...
[a03] The ArcSoft DTS Decoder reported an error while decoding.
Aborted at file position 16384.

so, actually eac3to is wrong - somehow it should decode with ArcSoft to 2 channels like GraphEdit do and then remove the empty Right channel.

please, add to your TODO list - fix for decoding mono tracks with ArcSoft.

[edit] Sonic also decode the mono samples in GraphEdit.

[edit2] Sonic and Arcsoft decodes in GraphEdit are bit identical, but duration is 28-29sec as the core. so, Sonic and Arcsoft decode the DTS core not the DTS-HD MA in GraphEdit with these mono samples.

[edit3] Nero doesn't work at all here with these mono samples. libav is OK.

[madshi]

@xkodi, which exact graph are you using? And are you using the m2ts file or the demuxed DTS file? Are you sure that the splitter you used didn't output the PCM file instead of the DTS file? On a quick check I couldn't get the DTS track to work in GraphEdit with ArcSoft or Sonic, but of course I might have missed something...

[yonta]

evo audio delay detection has a problem with v2.79

v2.78
D:\samples>eac3to.exe "The.Interpreter.evo"
EVO, 1 video track, 5 audio tracks, 5 subtitle tracks, 0:01:02
1: h264/AVC, 1080p24 /1.001 (16:9) with pulldown flags
2: E-AC3, 5.1 channels, 1536kbps, 48khz, dialnorm: -27dB, -125ms
3: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz, dialnorm: -4dB, -125ms
4: E-AC3, 5.1 channels, 768kbps, 48khz, dialnorm: -27dB, -125ms
5: E-AC3, 5.1 channels, 768kbps, 48khz, dialnorm: -27dB, -125ms
6: E-AC3 Surround, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, -29ms

v2.79
D:\samples>eac3to "The.Interpreter.evo"
EVO, 1 video track, 5 audio tracks, 5 subtitle tracks, 0:01:02
1: h264/AVC, 1080p24 /1.001 (16:9) with pulldown flags
2: E-AC3, 5.1 channels, 1536kbps, 48khz, dialnorm: -27dB, -543ms
3: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz, dialnorm: -4dB, -543ms
4: E-AC3, 5.1 channels, 768kbps, 48khz, dialnorm: -27dB, -543ms
5: E-AC3, 5.1 channels, 768kbps, 48khz, dialnorm: -27dB, -543ms
6: E-AC3 Surround, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, -447ms

[madshi]

Originally Posted by yonta

evo audio delay detection has a problem with v2.79

Have you tried which delay is more correct? I've changed to a different VOB/EVO delay calculation method in v2.79, but I found myself that it doesn't work very well, so I've already changed in back for the next build.

[xkodi]

@madshi

i missed the fact that there is also LPCM track and yes you are right i actually decoded the LPCM track instead of the DTS-HD MA track. so, nothing you can do, Arcsoft and Sonic simply can't handle such tracks in GraphEdit too.

[yonta]

Originally Posted by madshi

Have you tried which delay is more correct? I've changed to a different VOB/EVO delay calculation method in v2.79, but I found myself that it doesn't work very well, so I've already changed in back for the next build.

v2.78 is way more correct.
I would say, in this specific case, v2.78 is correct while v2.79 is not.

Tried some more files.
Some are correct while others are about 300~400ms off.

[laserfan]

Not a bug, but maybe this would bother someone else:

eac3to v2.79
command line: eac3to coredaudio.1.dts audio.ac3 -448
------------------------------------------------------------------------------
DTS, 5.1 channels, 2:29:59, 24 bits, 1509kbps, 48khz
The ArcSoft decoder doesn't seem to work, will use Sonic instead.
Decoding with DirectShow (Sonic Audio Decoder)...
DirectShow reports 5.1 channels, 24 bits, 48khz
Encoding AC3 <448kbps> with libAften...
Creating file "audio.ac3"...
eac3to processing took 28 minutes, 11 seconds.
Done.

I don't have Arcsoft on my PC so this took me aback when I saw it. Or maybe I should ask too: Is Arcsoft preferred over Sonic for DTS decoding? I was thinking of getting their TMT anyway...

[poisondeathray]

Originally Posted by laserfan

I don't have Arcsoft on my PC so this took me aback when I saw it. Or maybe I should ask too: Is Arcsoft preferred over Sonic for DTS decoding? I was thinking of getting their TMT anyway...

That is answered on page 1 under "evaluation of available decoders"

Cheers

[asarian]

Originally Posted by laserfan

eac3to v2.79
command line: eac3to coredaudio.1.dts audio.ac3 -448
------------------------------------------------------------------------------
DTS, 5.1 channels, 2:29:59, 24 bits, 1509kbps, 48khz
The ArcSoft decoder doesn't seem to work, will use Sonic instead.
Decoding with DirectShow (Sonic Audio Decoder)...
DirectShow reports 5.1 channels, 24 bits, 48khz
Encoding AC3 <448kbps> with libAften...
Creating file "audio.ac3"...
eac3to processing took 28 minutes, 11 seconds.
Done.

Just out of curiosity, why would you downgrade the AC3 stream to 448kbps (from 1536kbps DTS), instead of just using the default 640kbps?

[Thunderbolt8]

Originally Posted by madshi

Have you tried which delay is more correct? I've changed to a different VOB/EVO delay calculation method in v2.79, but I found myself that it doesn't work very well, so I've already changed in back for the next build.

is this problem only restricted to delay or does it affect gaps/overlapping as well?

[n0mag!c]

Originally Posted by madshi

It produces the same result when running on the same data. But the part you didn't understand is that the algorithm adopts itself to the waveform of the audio track. So if you split a stereo file into 2 separate mono files and process each separately from each other, the DIRAC algorithm may do different things to each channel, depending on the waveform. And this can distort stereo phase information. Same thing with surround tracks.

Aha, I've figured that out lately, DIRAC algorithm (due to its purpose) can shift phase in different degree depending on the input data.

Originally Posted by madshi

If you use phase locked processing (only available in the non-free DIRAC versions), the DIRAC algorithm makes sure that it does exactly the same thing to all channels. This makes sure that stereo and surround phase information is not distorted.

Phase locked processing isn't available in free stereo version? Damn. At least we could feed it with pairs (C+R, C+L, C+SL, C+SR, C+LFE) to get phase synced channels but lose in speed.

[madshi]

Originally Posted by Thunderbolt8

is this problem only restricted to delay or does it affect gaps/overlapping as well?

Only delay. So no need to worry...

Originally Posted by n0mag!c

Phase locked processing isn't available in free stereo version? Damn. At least we could feed it with pairs (C+R, C+L, C+SL, C+SR, C+LFE) to get phase synced channels but lose in speed.

Yeah, unfortunately the free version can't even do stereo phase locked processing. Only mono. That's why I think it's useless for our purposes. The only situation where the free DIRAC version can be used is for producers/studios/musicians who record and edit one track after the other in mono.

[madshi]

eac3to v2.80 released

http://madshi.net/eac3to.zip

* fixed: FLAC files with missing runtime information were not accepted
* gone back to old VOB/EVO auto delay calculation method, more reliable for me
* improved TS broadcast audio delay detection
* added support for constant bitrate AAC encoding
* added support for AAC encoding 0.00 and 1.00 quality

[nautilus7]

Thanks man. I really love to know that i can expect a new version every Sunday night.

[Thunderbolt8]

Originally Posted by nautilus7

Thanks man. I really love to know that i can expect a new version every Sunday night.

+1

can you please comment on the improved .ts audio delay correction, in far was it unprecise before? only a few ms? or XXms? do I need to check again now those I've already remuxed?

[vucloutr]

Originally Posted by madshi

eac3to v2.80 released

http://madshi.net/eac3to.zip

* ...
* added support for AAC encoding 0.00 and 1.00 quality

Thank you !

PS: just the description isn't updated yet

[madshi]

Originally Posted by nautilus7

Thanks man. I really love to know that i can expect a new version every Sunday night.

At one time in the past I realized that I was spending too much time on a non-profit software like eac3to. So now I'm limiting myself to only work on it on Sundays... Of course there's no guarantee that I'll have time every Sunday. But so far there are usually at least a few bugfixes every week...

Originally Posted by Thunderbolt8

can you please comment on the improved .ts audio delay correction, in far was it unprecise before? only a few ms? or XXms? do I need to check again now those I've already remuxed?

It could be quite incorrect (up to several hundreds of ms) for such movies where there were video frames before the first sequence header in the source file. That is not the case for any HD DVD or Blu-Ray movies I know. Also cleanly cut broadcast movies shouldn't have this, either. But randomly cut broadcasts could show this problem. You would have noticed that, though, cause the delay was usually way off in such cases. So if you did a quick audio sync check on every movie and you found it to be ok, there's nothing to worry about...

Originally Posted by vucloutr

PS: just the description isn't updated yet

Done.

[FredThompson]

Originally Posted by xkodi

available Intro and Trial (Try and Buy) versions of Arcsoft TotalMediaTheatre (TMT) and TotalMediaExtreme (TME) and which of them work with eac3to:

1. TMT Intro version 2.1.6.120 - "TotalMediaTheatre_Retail_INTRO_E.exe" (md5 77ea5ecdc52af9f20d646cc352389cc2):

http://base.arcsoft.com/downloads/in...il_INTRO_E.exe

The link is broken for me. That might be redirection blocking, I don't know. Your referenced post is a few months old so I'm querying to see if there is updated sourcing information. Goal is 7.1 with as little overhead as possible, of course.

[jimz06]

A search on the Forums reveals that at least several people on this thread have managed to remux the Babel HD DVD so I was hoping that one of those fortunate people could share some wisdom.

I've lost track of the various ways and tools I've used produce a file. The last attempt was to use eac3to to extract the AC3 track, EVOdemux to extract the H264 file, and then mkvmerge to put them together. The end result always seems to be the same, a break up of the picture every 10 seconds or so and loss of audio sync. When I play the H264 stream alone it plays fine without the breakups and the framerate counter in MPC says it is playing at 24 fps. I can listen to the AC3 stream separately also. Am I missing some critical setting on any of these tools that is causing me fits? All of the Blu Rays I've remuxed have been a dream compared to this HD DVD.

[Beastie Boy]

@jimz06

Use eac3to to produce a mkv file containing the video (just use .mkv as the output extension). Then use mkvmerge to join your video and audio.

Cheers, Beastie.

[hubblec4]

Originally Posted by madshi

eac3to v2.80 released

http://madshi.net/eac3to.zip

* added support for constant bitrate AAC encoding
quality

how i can realize that?


hubble

[nautilus7]

I suppose like you specify the bitrate for other format encoding. Try it.

[Thunderbolt8]

madshi, regarding that DTS-HD MA 1.0 movie I sent you 2 samples from, there is one gap/overlap detected for the DTS-HD MA track between the sync point of both main m2ts files. but when I process the LPCM -> FLAC track from it, I dont get this message. is this ok ? (I know that LPCM frames are rather short and the delay difference would be indiscernible, but could this be a general bug, or did eac3to only decide that the correction of this minor delay wouldnt be worth it and therefore did not put out a gap/overlap message?)

[nurbs]

Originally Posted by hubblec4

how i can realize that?

Haven't tried yet, but I guess its the -xxx switch, same as ac3.

BTW thanks madshi

[Jeff Flowerday]

Originally Posted by jimz06

A search on the Forums reveals that at least several people on this thread have managed to remux the Babel HD DVD so I was hoping that one of those fortunate people could share some wisdom.

I've lost track of the various ways and tools I've used produce a file. The last attempt was to use eac3to to extract the AC3 track, EVOdemux to extract the H264 file, and then mkvmerge to put them together. The end result always seems to be the same, a break up of the picture every 10 seconds or so and loss of audio sync. When I play the H264 stream alone it plays fine without the breakups and the framerate counter in MPC says it is playing at 24 fps. I can listen to the AC3 stream separately also. Am I missing some critical setting on any of these tools that is causing me fits? All of the Blu Rays I've remuxed have been a dream compared to this HD DVD.

Sounds more like a play back problem, try reinstalling haali media splitter.

[Devrethman]

Okay, stupid n00b question here: I just bought Band of Brothers on Blu-ray, it's the first disc I've seen that has DTS "Master Audio" which I have gathered to be the same thing as DTS-HD. When I run eac3to to check what the track order is, it says that the audio conversion is not supported. However, the first post also says that the ArcSoft decoder (which I have) works well for DTS-HD, and when I start decoding it to FLAC, it decodes with the arcsoft decoder just fine. is it actually decoding the HD track or just the DTS core? If the latter, is it possible to get it to decode the whole track or is that not supported yet?

[Thunderbolt8]

got this message with the german dances with wolves BD:

eac3to v2.80
command line: X:\eac3to\eac3to X:\movie 1) 2: X:\dances.mkv 4: X:\dances.flac 4: X:\dances.dtsma 11: X:\dances.sup
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 5 subtitle tracks, 3:56:38
1: Chapters, 31 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, German, 7.1 channels, 16 bits, 48khz
   (core: DTS, 5.1 channels, 16 bits, 1509kbps, 48khz)
4: DTS Master Audio, English, 7.1 channels, 16 bits, 48khz
   (core: DTS, 5.1 channels, 16 bits, 1509kbps, 48khz)
5: AC3, English, 2.0 channels, 192kbps, 48khz
6: AC3, English, 2.0 channels, 192kbps, 48khz
7: Subtitle (PGS), German
8: Subtitle (PGS), German
9: Subtitle (PGS), German
10: Subtitle (PGS), German
11: Subtitle (PGS), English
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[s11] Extracting subtitle track number 11...
[a04] Extracting audio track number 4...
[v02] Muxing video to Matroska...
[a04] Decoding with ArcSoft DTS Decoder...
[a04] Encoding FLAC with libFlac...
[a04] Creating file "G:\dances.dtsma"...
[a04] Creating file "G:\dances.flac"...
[s11] Creating file "G:\dances.sup"...
[a04] The last DTS frame is incomplete and thus gets skipped.
[a04] Original audio track, L+R+C+BL+BR+SL+SR: constant bit depth of 16 bits.
[a04] Processed audio track, LFE: bitdepth analyzation failed.
[a04] Processed audio track, L+R+C+BL+BR+SL+SR: constant bit depth of 16 bits.
[a04] Processed audio track, LFE: bitdepth analyzation failed.
[a04] The last DTS frame is incomplete and thus gets skipped.
Added fps value to MKV header.
Video track 2 contains 340422 frames.
eac3to processing took 1 hour, 2 minutes.
Done.

50mb sample:

http://www.sendspace.com/file/xkpcl3

[madshi]

Originally Posted by jimz06

The last attempt was to use eac3to to extract the AC3 track, EVOdemux to extract the H264 file, and then mkvmerge to put them together. The end result always seems to be the same, a break up of the picture every 10 seconds or so and loss of audio sync.

As Beastie Boy suggested, try "eac3to HdDvdSourceFolder movie.mkv". Then play the MKV file just as it is (without the audio). Are there still break ups? If so, probably your playback system is borked (as suggested by Jeff Flowerday). If there are no breaks up when playing the video only MKV, mux the audio track to the video MKV by using mkvtoolnix. Does the final result play fine? If not, again probably something is wrong with your DirectShow filter setup. Try different video/audio decoders and maybe also different video/audio renderers...

Originally Posted by hubblec4

how i can realize that?

nautilus7 and nurbs have guessed right. It's the same way you can specify bitrates for AC3 and DTS. E.g. "eac3to source.whatever dest.aac -192".

If you don't specify anything for AAC encoding, eac3to will use quality 0.5. If you specify a different quality level, eac3to will use that instead. And if you specify a bitrate, eac3to will use constant bitrate encoding instead of quality based encoding.

Originally Posted by Thunderbolt8

madshi, regarding that DTS-HD MA 1.0 movie I sent you 2 samples from, there is one gap/overlap detected for the DTS-HD MA track between the sync point of both main m2ts files. but when I process the LPCM -> FLAC track from it, I dont get this message. is this ok ?

Yes. LPCM gaps usually only get big enough to be reported if there are a whole lot of m2ts parts. The threshould for reporting a gap is 7ms, IIRC.

Originally Posted by Devrethman

When I run eac3to to check what the track order is, it says that the audio conversion is not supported.

You most probably have used an incorrect command line then.

Originally Posted by Devrethman

However, the first post also says that the ArcSoft decoder (which I have) works well for DTS-HD, and when I start decoding it to FLAC, it decodes with the arcsoft decoder just fine. is it actually decoding the HD track or just the DTS core?

That is explained in the first post of this thread. The full DTS-HD track is decoded bit perfectly, when the ArcSoft decoder is used by eac3to.

[madshi]

Originally Posted by Thunderbolt8

got this message with the german dances with wolves BD:

4: DTS Master Audio, English, 7.1 channels, 16 bits, 48khz
[a04] Original audio track, L+R+C+BL+BR+SL+SR: constant bit depth of 16 bits.
[a04] Processed audio track, LFE: bitdepth analyzation failed.

That is *VERY* bad. It means that the English audio track on the Dances with Wolves Blu-Ray is broken. The LFE is channel is empty. That's why eac3to can't find out which bitdepth it has.

Now it's not the end of the world. We could extract the LFE channel of the German audio track and use that for the English audio track. But for all intends and purposes the Blu-Ray should be considered broken and a repressing should be requested from the studio.

[Thunderbolt8]

hm how can I do that to have it in flac lossless as well? extract the german dts-hd track to .wavs and then just convert the LFE channel to flac? how to combine it then with the other 4 channels from the english track all together to one single flac track again?

[madshi]

Originally Posted by Thunderbolt8

hm how can I do that to have it in flac lossless as well? extract the german dts-hd track to .wavs and then just convert the LFE channel to flac? how to combine it then with the other 4 channels from the english track all together to one single flac track again?

Decode both the English and German audio tracks to "WAVs". Then swap the LFE WAV channel WAV files of both tracks. Next combine the separate WAV files for the English track to one big WAV file. Unfortunately eac3to can't do that, but I think wavewizard can. Finally do "eac3to combined.wav english.flac" to encode the final FLAC file.

[Thunderbolt8]

madshi (or someone else), im not familiar with wavewizard, but it looks to me like this programm cant do it. I can change channel mapping with it or convert wav files to another format, but I cant find any option where I can join multiple files to a big single one.

is there perhaps another program who can do this?

[b66pak]

try using m$ wavavimux (google it - its free) to mux 2, 6 or 8 wav mono channels in one giant .avi which can be used as an input be many audio tools
_

[tebasuna51]

Originally Posted by Thunderbolt8

madshi (or someone else), im not familiar with wavewizard, but it looks to me like this programm cant do it. I can change channel mapping with it or convert wav files to another format, but I cant find any option where I can join multiple files to a big single one.

is there perhaps another program who can do this?

WaveWizard instructions:
- Add monowavs in standard order: FL,FR,FC,LF,SL,SR
- Edit -> Preferences:
All options unchecked but:
Stream manipulation -> Merge files
Output format -> Wave PCM
(or WaveFormatEx but the ChannelMask value is always 0)
- Convert

You can use also the sox command line:

sox -M FL.wav FR.wav FC.wav LF.wav BL.wav BR.wav multichannel.wav

[Thunderbolt8]

thanks. but its a 7.1 track (BL, BR, C, L, R, SL, SR, LFE), which is the standard order for that then?

[b66pak]

left.wav right.wav center.wav lfe.wav ls.wav rs.wav aux1.wav aux2.wav....it meens l, r, c, lfe, sl, sr, bl, br
_

[bmnot]

Is there/will there be a way to remove pulldown from already made remuxes? I just checked a bunch old HD DVD remuxes in tsmuxer and it detects the video as 29.97fps with pulldown. I tried to output as a Blu-ray disc, but it failed to work. It couldn't remove the pulldown or fix the framerate. Using tsmuxer on a recently remuxed HD DVD, it detects the video as 23.976 and outputs perfect Blu-ray disc structure, perfect for burning. So anyway to fix those old HD DVD remuxes? Unfortunately I sold most of them...

[crazydane]

I have been ripping with eac3to for some time and I have never really had any major issues, but now all of a sudden when I rip, I get the following error regardless of what title I try to rip:

"One of the FLAC encoder's callbacks returned a fatal error"

I was on version 2.78 and just upgraded to 2.80, but the issue remains. I rebooted the machine as well to no avail.

Other than installing the latest Intel graphics/HDMI driver (I have a G45 based motherboard), I have not changed anything recently in my Vista 32-bit environment.

Here are some examples of the error:

The Untouchables:




The Hunt for Red October:



Eac3to errors out after a few minutes of processing, something like 4 "-"s or so, so it's not instant.

I can rip these titles just fine to hd using AnyDVD HD, and then run eac3to on the files on the hd, but why would I be getting this error all of a sudden when pointing eac3to directly to the BD disc in the drive?

Any ideas?

Thanks!

[tebasuna51]

Originally Posted by b66pak

left.wav right.wav center.wav lfe.wav ls.wav rs.wav aux1.wav aux2.wav....it meens l, r, c, lfe, sl, sr, bl, br
_

No! Mus be:
l, r, c, lfe, BackL, BackR, SideL, SideR

sox -M FL.wav FR.wav FC.wav LF.wav BL.wav BR.wav SL.wav SR.wav multichannel.wav

[jimz06]

Originally Posted by madshi

As Beastie Boy suggested, try "eac3to HdDvdSourceFolder movie.mkv". Then play the MKV file just as it is (without the audio). Are there still break ups? If so, probably your playback system is borked (as suggested by Jeff Flowerday). If there are no breaks up when playing the video only MKV, mux the audio track to the video MKV by using mkvtoolnix. Does the final result play fine? If not, again probably something is wrong with your DirectShow filter setup. Try different video/audio decoders and maybe also different video/audio renderers...

Thanks to everyone for the helpful advice. I was trying to create this file to play on a PopcornHour. The one thing I hadn't tried was playing it on the PCH. As it turns out it plays just fine on the PCH, so as was indicated it must be the setup on my MPC-HC player.

However, some enlightenment would be appreciated about how this all works. When I used EVOdemux to create the H264 stream I got this:

Complete name              :E:\FEATURE_1_MERGED.H.264.stream.0.mpv
Format                           : AVC
Format/Info                      : Advanced Video Codec
File size                        : 22.3 GiB

Video
Format                           : AVC
Format/Info                      : Advanced Video Codec
Format profile                   : High@L4.1
Format settings, CABAC           : Yes
Format settings, ReFrames        : 2 frames
Width                            : 1 920 pixels
Height                           : 1 080 pixels
Display aspect ratio             : 16/9
Frame rate                       : 29.970 fps
Standard                         : Component
Resolution                       : 24 bits
Colorimetry                      : 4:2:0
Scan type                        : Progressive

This played fine on MPC but the fps counter showed 23.97.

When I muxed this into an mkv (no audio) I got this:

Complete name               : E:\ToNMT\Babel.mkv
Format                          : Matroska
File size                        : 22.3 GiB
Duration                         : 2h 23mn
Overall bit rate                 : 22.2 Mbps
Encoded date                     : UTC 2008-12-09 00:33:12
Writing application              : eac3to
Writing library                  : Haali DirectShow Matroska Muxer 1.8.122.18

Video
Format                           : AVC
Format/Info                      : Advanced Video Codec
Format profile                   : High@L4.1
Format settings, CABAC           : Yes
Format settings, ReFrames        : 2 frames
Muxing mode                      : Container profile=Unknown@0.0
Codec ID                         : V_MPEG4/ISO/AVC
Duration                         : 2h 23mn
Bit rate                         : 21.4 Mbps
Width                            : 1 920 pixels
Height                           : 1 080 pixels
Display aspect ratio             : 16/9
Frame rate                       : 23.976 fps
Standard                         : Component
Resolution                       : 24 bits
Colorimetry                      : 4:2:0
Scan type                        : Progressive

Here is the eac3to log:

eac3to v2.80
command line: "E:\eac3to275\eac3to.exe" "D:\Babel\BABEL\HVDVD_TS\FEATURE_1.EVO"+"D:\Babel\BABEL\HVDVD_TS\FEATURE_2.EVO" 3: "E:\ToNMT\Babel.mkv"
------------------------------------------------------------------------------
EVO, 1 video track, 2 audio tracks, 6 subtitle tracks, 2:23:29
"Main Movie"
1: Joined EVO file
2: Chapters, 24 chapters with names
3: h264/AVC, 1080p24 /1.001 (16:9) with pulldown flags
4: E-AC3, English, 5.1 channels, 1536kbps, 48khz, dialnorm: -27dB, 1918ms
5: E-AC3, French, 5.1 channels, 1536kbps, 48khz, dialnorm: -27dB, 1918ms
6: Subtitle, English
7: Subtitle, English, "SDH"
8: Subtitle, French
9: Subtitle, Spanish
10: Subtitle, English, "Forced"
11: Subtitle, French, "Forced"
[v03] Extracting video track number 3...
[v03] Removing h264 pulldown...
[v03] Muxing video to Matroska...
Added fps value to MKV header.
Video track 3 contains 206438 frames.
eac3to processing took 19 minutes, 56 seconds.
Done.

Played back in MPC, I get the regular video break ups using either Haali or the internal splitter. No problems with other H264 Blu Ray rips. I assume something is going on with that 30 fps to 24 fps that is wreaking havoc on my MPC setup?

[73ChargerFan]

Originally Posted by bmnot

Is there/will there be a way to remove pulldown from already made remuxes? I just checked a bunch old HD DVD remuxes in tsmuxer and it detects the video as 29.97fps with pulldown. I tried to output as a Blu-ray disc, but it failed to work. It couldn't remove the pulldown or fix the framerate. Using tsmuxer on a recently remuxed HD DVD, it detects the video as 23.976 and outputs perfect Blu-ray disc structure, perfect for burning. So anyway to fix those old HD DVD remuxes? Unfortunately I sold most of them...

Try muxing as an m2ts (if not already in that format) and then demux with eac3to. It should remove the pulldown automatically.

[Thunderbolt8]

Originally Posted by tebasuna51

No! Mus be:
l, r, c, lfe, BackL, BackR, SideL, SideR

sox -M FL.wav FR.wav FC.wav LF.wav BL.wav BR.wav SL.wav SR.wav multichannel.wav

aerf, which now, backs first or sides first? the whole process takes quite long :S

[madshi]

Originally Posted by bmnot

Is there/will there be a way to remove pulldown from already made remuxes? I just checked a bunch old HD DVD remuxes in tsmuxer and it detects the video as 29.97fps with pulldown.

You can either try to remux the MKV to m2ts or ts with tsMuxeR (as 73ChargerFan suggested) and then ask eac3to to demux the video or to remux it to MKV. Or if tsMuxeR can't handle the MKV properly, you can demux the video track from the original MKV file by using mkvextract and then you can do "eac3to source.vc1 dest.vc1" to remove the pulldown completely. Please note that mkvextract may need different parameters depending on the video codec. Do a search for mkvextract in this thread to find out which parameters must be used for which video codec. Also for h264 you may need to run the stream through h264info to add AUDs (access unit delimiters) into the h264 stream, or else eac3to won't be able to handle the stream.

Originally Posted by crazydane

I have been ripping with eac3to for some time and I have never really had any major issues, but now all of a sudden when I rip, I get the following error regardless of what title I try to rip:

"One of the FLAC encoder's callbacks returned a fatal error"

I was on version 2.78 and just upgraded to 2.80, but the issue remains. I rebooted the machine as well to no avail.

Other than installing the latest Intel graphics/HDMI driver (I have a G45 based motherboard), I have not changed anything recently in my Vista 32-bit environment.

Here are some examples of the error:

Eac3to errors out after a few minutes of processing, something like 4 "-"s or so, so it's not instant.

I can rip these titles just fine to hd using AnyDVD HD, and then run eac3to on the files on the hd, but why would I be getting this error all of a sudden when pointing eac3to directly to the BD disc in the drive?

I've no idea why it's going wrong. But the fact that it works well when ripping to harddisk first means that it can't be eac3to's fault. Don't know what else to say...

Originally Posted by jimz06

However, some enlightenment would be appreciated about how this all works. When I used EVOdemux to create the H264 stream I got this:

Frame rate                       : 29.970 fps

When I muxed this into an mkv (no audio)

I got this:

Frame rate                       : 23.976 fps

Played back in MPC, I get the regular video break ups using either Haali or the internal splitter. No problems with other H264 Blu Ray rips. I assume something is going on with that 30 fps to 24 fps that is wreaking havoc on my MPC setup?

EvoDemux does not remove the h264 pulldown, so the video is 1080i60. eac3to removes the h264 pulldown, so the video is 1080p24. Don't know why your playback system has problems with that. It seems to work well for everyone else. If all else fails you can tell eac3to to not remove the pulldown by using the "-keepPulldown" switch.

Originally Posted by Thunderbolt8

aerf, which now, backs first or sides first? the whole process takes quite long :S

tebasuna51 is usually right. Just as in this case.

[yesgrey]

Which compressing level is used in the conversion to FLAC, the default? Could this value be specified by the user?

Thanks.

[madshi]

Originally Posted by yesgrey3

Which compressing level is used in the conversion to FLAC, the default? Could this value be specified by the user?

FLAC encoding is hardcoded to max compression level and I don't plan on changing that because I think the compression speed is good enough. So IMHO there's no need to make things more complicated by adding further options just to save a tiny bit of time.

[odin24]

As to why when a DTS track is encoded using Surcode it is 1536kb/s, then when the Zero Padding is removed it is reported as 1510kb/s? MediaInfo and tsMuxeR both report as 1536kb/s.

Another question... whenever I convert a THD track to PCM it always starts as ....24bit.pcm until the actual bit depth is revealed later in the conversion where it reports a constant bit depth of 16 bits. I'm assuming the PCM output will actually be 16 bits... correct?

Thanks for the time.

eac3to v2.78
command line: e3\eac3to  f: 1) 2: c:\videos\tdk\tdk.vc1 4: c:\videos\tdk\tdk.thd+ac3 4: c:\videos\tdk\tdk.pcm 4: c:\videos\tdk\tdk.dts 8: c:\videos\tdk\subs1.sup
------------------------------------------------------------------------------
M2TS, 1 video track, 5 audio tracks, 5 subtitle tracks, 2:32:13
1: Chapters, 40 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: AC3, English, 5.1 channels, 640kbps, 48khz
4: TrueHD/AC3, English, 5.1 channels, 48khz
   (embedded: AC3, 5.1 channels, 640kbps, 48khz)
5: AC3 Surround, English, 2.0 channels, 192kbps, 48khz
6: AC3, French, 5.1 channels, 640kbps, 48khz
7: AC3, Spanish, 5.1 channels, 640kbps, 48khz
8: Subtitle (PGS), English
9: Subtitle (PGS), French
10: Subtitle (PGS), Spanish
11: Subtitle (PGS), French
12: Subtitle (PGS), Spanish
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[a04] Extracting audio track number 4...
[a04] Extracting audio track number 4...
[s08] Extracting subtitle track number 8...
[a04] Extracting TrueHD stream...
[a04] Extracting TrueHD stream...
[a04] Decoding with libav/ffmpeg...
[a04] Decoding with libav/ffmpeg...
[a04] Swapping endian...
[a04] Writing WAVs...
[a04] Remapping channels...
[a04] Creating file "c:\videos\tdk\tdk.24bit.pcm"...
[v02] Creating file "c:\videos\tdk\tdk.vc1"...
[a04] Creating file "c:\videos\tdk\tdk.thd+ac3"...
[a04] Creating file "c:\videos\tdk\tdk.L.wav"...
[a04] Creating file "c:\videos\tdk\tdk.R.wav"...
[a04] Creating file "c:\videos\tdk\tdk.C.wav"...
[a04] Creating file "c:\videos\tdk\tdk.SR.wav"...
[a04] Creating file "c:\videos\tdk\tdk.LFE.wav"...
[a04] Creating file "c:\videos\tdk\tdk.SL.wav"...
[s08] Creating file "c:\videos\tdk\subs1.sup"...
[a04] The original audio track has a constant bit depth of 16 bits.
[a04] The zero bytes were successfully removed.
[a04] The original audio track has a constant bit depth of 16 bits.
Encoding DTS <1536kbps> with Surcode...Found Surcode DTS Encoder version 1.0.21.0.
Surcode encoding successfully started. Please wait...
Closing Surcode...
Video track 2 contains 218979 frames.
eac3to processing took 49 minutes, 32 seconds.
Surcode encoding took 27 minutes, 18 seconds.
Done.

eac3to v2.78
command line: e3\eac3to  c:\videos\tdk\tdk.dts c:\videos\tdk\tdk_fixed.dts
------------------------------------------------------------------------------
DTS, 5.1 channels, 2:32:13, 24 bits, 1510kbps, 48khz
Removing DTS zero padding...
Creating file "c:\videos\tdk\tdk_fixed.dts"...
eac3to processing took 2 minutes, 52 seconds.
Done.

[yesgrey]

Originally Posted by madshi

FLAC encoding is hardcoded to max compression level...

If it's set to the max level no need to change... For me the important is not the speed, but the size.

[asarian]

Got a question.

For streaming, the PS3 sadly supports only DD. 2.0 for VC-1 in a WMV container. So, I go to thinking the following: what if you were to 'matrix-encode' five disccrete channels of audio, ala the Dolby ProLogic II scheme, onto a 640kbps DD 2.0 stream in such a way that a Dolby ProLogic II decoder could split those matrixed channels back to the original five channels, would that work? I mean, to take an existing DD 5.1 stream, and turn it into a 2.0 stream that Dolby ProLogic II could then extract more-or-less the same original channels from. Would be a nice trick to get near DD 5.1 sound (whilst it really is just DD 2.0 surround).

Question, of course, is: can eac3to do it? Can it convert a DD 5.1 track to a Dolby ProLogic II 'splittable' 2.0 track? I mean, the -down convert would have to follow the exact Dolby ProLogic II scheme, as if it were a Dolby ProLogic II encoder, as it were.

It's kinda early here (near 6 AM, and I've yet to turn in, LOL), so I may be missing somehing; but with a 640kbps 2.0 stream, I don't see how you couldn't at least get a half-way decent Dolby ProLogic II 5.1 result from it (and thus be able to bypass the need to reencode VC-1 content when streaming).

Feel free to enlighten me (or desillusion me, whichever the case may be).

[madshi]

Originally Posted by odin24

As to why when a DTS track is encoded using Surcode it is 1536kb/s, then when the Zero Padding is removed it is reported as 1510kb/s? MediaInfo and tsMuxeR both report as 1536kb/s.

The nominal bitrate is 1536kbps. However, the actual bitrate is lower. eac3to calculates & reports the actual bitrate, while MediaInfo and tsMuxeR report the nominal bitrate...

Originally Posted by odin24

Another question... whenever I convert a THD track to PCM it always starts as ....24bit.pcm until the actual bit depth is revealed later in the conversion where it reports a constant bit depth of 16 bits. I'm assuming the PCM output will actually be 16 bits... correct?

A TrueHD track can have 16bit data or 24bit data or anything in between. eac3to only knows for sure when the whole track is fully decoded. The PCM output consequently is either 16bit (for TrueHD tracks which only contain 16bit worth of audio data) or 24bit (for TrueHD tracks which have 17-24bit of information). eac3to will tell you the bitdepth of the final PCM file.

Originally Posted by yesgrey3

For me the important is not the speed, but the size.

Agreed.

Originally Posted by asarian

what if you were to 'matrix-encode' five disccrete channels of audio, ala the Dolby ProLogic II scheme, onto a 640kbps DD 2.0 stream in such a way that a Dolby ProLogic II decoder could split those matrixed channels back to the original five channels, would that work?

Yes. But the end result will not be perfect. It should be quite ok, but keeping the channels separated is of course noticeably better. One problem with DPLII is that the LFE channel is more or less lost.

Originally Posted by asarian

Question, of course, is: can eac3to do it? Can it convert a DD 5.1 track to a Dolby ProLogic II 'splittable' 2.0 track?

Yes. "-down2".

Originally Posted by asarian

but with a 640kbps 2.0 stream, I don't see how you couldn't at least get a half-way decent Dolby ProLogic II 5.1 result from it (and thus be able to bypass the need to reencode VC-1 content when streaming).

Half-way decent sounds about right.

[komisar]

madshi
Can you add to eac3to ability for "custom-comman-line-parameters"?
E.g. for neroaacenc:

eac3to.exe input.ac3 output.aac -quality=0.23 -lc

[tebasuna51]

Originally Posted by komisar

madshi
Can you add to eac3to ability for "custom-comman-line-parameters"?
E.g. for neroaacenc:

eac3to.exe input.ac3 output.aac -quality=0.23 -lc

Encoders can have many parameters.
You can always acces to all command line parameters for a encoder using the pipe option:

eac3to.exe <input> stdout.wav <eac3to_parameters> | <encoder> <full_encoder_parameters> <-> <output>

Example:
eac3to.exe input.ac3 stdout.wav | neroaacenc -ignorelength -q 0.23 -lc -if - -of output.m4a

[komisar]

tebasuna51, Ok. Thnx.

[Thunderbolt8]

got a problem with a TrueHD track here:

50mb sample: http://www.sendspace.com/file/y2diub

eac3to v2.80
command line: eac3to X:\movie 1) 2: G:\movie.mkv 3: G:\movie.flac 3: G:\movie.thd 5: G:\moviecomment1.ac3 6: G:\moviecomment2.ac3 9: G:\movie.sup
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 5 subtitle tracks, 2:04:01
1: Chapters, 29 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), -13ms
4: TrueHD/AC3, Japanese, 5.1 channels, 48khz, dialnorm: -27dB
   (embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB), -13ms
5: AC3, English, 2.0 channels, 256kbps, 48khz, dialnorm: -27dB, -13ms
6: AC3, English, 2.0 channels, 256kbps, 48khz, dialnorm: -27dB, -13ms
7: Subtitle (PGS), Japanese
8: Subtitle (PGS), Japanese
9: Subtitle (PGS), English
10: Subtitle (PGS), Japanese
11: Subtitle (PGS), Japanese
[a03] Extracting audio track number 3...
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a06] Extracting audio track number 6...
[a05] Extracting audio track number 5...
[s09] Extracting subtitle track number 9...
[a03] This track is not clean.
[a03] This track is not clean.
[a03] Extracting TrueHD stream...
[v02] Muxing video to Matroska...
[a03] Extracting TrueHD stream...
[a03] Removing TrueHD dialog normalization...
[a03] Decoding with libav/ffmpeg...
[libav] Stream parameters not seen; skipping frame
[a03] The libav decoder output an unexpected bitdepth (1).
Aborted at file position 49152.

and when I do it with -nero (nero also decodes TrueHD 5.1 completely lossless, right?) then I get:

G:\eac3to>eac3to X:\movie 1) 2: G:\movie.mkv 3: G:\movie.flac -nero 3: G:\movie.thd 5: G:\moviecomment1.ac3 6: G:\moviecomment2.ac3 9: G:\movie.sup
M2TS, 1 video track, 4 audio tracks, 5 subtitle tracks, 2:04:01
1: Chapters, 29 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), -13ms
4: TrueHD/AC3, Japanese, 5.1 channels, 48khz, dialnorm: -27dB
   (embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB), -13ms
5: AC3, English, 2.0 channels, 256kbps, 48khz, dialnorm: -27dB, -13ms
6: AC3, English, 2.0 channels, 256kbps, 48khz, dialnorm: -27dB, -13ms
7: Subtitle (PGS), Japanese
8: Subtitle (PGS), Japanese
9: Subtitle (PGS), English
10: Subtitle (PGS), Japanese
11: Subtitle (PGS), Japanese
[a03] Extracting audio track number 3...
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a06] Extracting audio track number 6...
[a05] Extracting audio track number 5...
[s09] Extracting subtitle track number 9...
[a03] This track is not clean.
[a03] This track is not clean.
[v02] Muxing video to Matroska...
[a03] Extracting TrueHD stream...
[a03] Extracting TrueHD stream...
[a03] Removing TrueHD dialog normalization...
[a03] Decoding with DirectShow (Nero Audio Decoder 2)...
[a03] Removing TrueHD dialog normalization...
[a05] Removing AC3 dialog normalization...
[a06] Removing AC3 dialog normalization...
[a03] DirectShow reports 5.1 channels, 24 bits, 48khz
[a03] Applying RAW/PCM delay...
[a03] This audio track contains more than 16 bits of information.
[a03] Encoding FLAC with libFlac...
[a03] Creating file "G:\movie.thd"...
[a03] Creating file "G:\movie.24bit.flac"...
[a06] Creating file "G:\moviecomment2.ac3"...
[a05] Creating file "G:\moviecomment1.ac3"...
[s09] Creating file "G:\movie.sup"...
[a03] Original audio track, L+R+C+LFE: constant bit depth of 16 bits.
[a03] Original audio track, BL+BR: constant bit depth of 24 bits.
[a03] Processed audio track, L+R+C+LFE: constant bit depth of 16 bits.
[a03] Processed audio track, SL+SR: constant bit depth of 24 bits.
[a03] Audio has a gap of 8ms at playtime 0:00:00.
[a03] The audio file was demuxed without making use of the gap/overlap information.
[a03] Please rerun the same eac3to command line. That will correct the gaps/overlaps.
[a03] Audio has a gap of 8ms at playtime 0:00:00.
[a03] The audio gaps/overlaps technically can't be removed from the TrueHD bitstream.
[a03] In order to remove them you'll have to transcode the audio to another format.
Added fps value to MKV header.
Video track 2 contains 178415 frames.
eac3to processing took 59 minutes, 53 seconds.
Done.

and after re-running the cmd line:

eac3to v2.80
command line: eac3to X:\movie 1) 2: G:\movie.mkv 3: G:\movie.flac -nero 3: G:\movie.thd 5: G:\moviecomment1.ac3 6: G:\moviecomment2.ac3 9: G:\movie.sup
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 5 subtitle tracks, 2:04:01
1: Chapters, 29 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), -13ms
4: TrueHD/AC3, Japanese, 5.1 channels, 48khz, dialnorm: -27dB
   (embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB), -13ms
5: AC3, English, 2.0 channels, 256kbps, 48khz, dialnorm: -27dB, -13ms
6: AC3, English, 2.0 channels, 256kbps, 48khz, dialnorm: -27dB, -13ms
7: Subtitle (PGS), Japanese
8: Subtitle (PGS), Japanese
9: Subtitle (PGS), English
10: Subtitle (PGS), Japanese
11: Subtitle (PGS), Japanese
Audio gap description file detected, will be used for processing...
Audio gap description file detected, can't be used for TrueHD/MLP, though.
[a06] Extracting audio track number 6...
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a03] Extracting audio track number 3...
[s09] Extracting subtitle track number 9...
[a05] Extracting audio track number 5...
[a03] This track is not clean.
[a03] This track is not clean.
[v02] Muxing to Matroska...
[a03] Extracting TrueHD stream...
[a03] Extracting TrueHD stream...
[a03] Decoding with DirectShow (Nero Audio Decoder 2)...
[a03] Removing TrueHD dialog normalization...
[a03] Removing TrueHD dialog normalization...
[a05] Removing AC3 dialog normalization...
[a06] Removing AC3 dialog normalization...
[a03] DirectShow reports 5.1 channels, 24 bits, 48khz
[a03] Applying RAW/PCM delay...
[a03] This audio track contains more than 16 bits of information.
[a03] Encoding FLAC with libFlac...
[a03] Creating file "G:\movie.thd"...
[a03] Creating file "G:\movie.24bit.flac"...
[a05] Creating file "G:\moviecomment1.ac3"...
[a06] Creating file "G:\moviecomment2.ac3"...
[s09] Creating file "G:\movie.sup"...
[a03] Original audio track, L+R+C+LFE: constant bit depth of 16 bits.
[a03] Original audio track, BL+BR: constant bit depth of 24 bits.
[a03] Processed audio track, L+R+C+SL+SR: max 24 bits, average 19 bits.
[a03] Processed audio track, LFE: max 23 bits, average 16 bits.
[a03] Audio has a gap of 8ms at playtime 0:00:00.
[a03] The audio gaps/overlaps technically can't be removed from the TrueHD bitstream.
[a03] In order to remove them you'll have to transcode the audio to another format.
Added fps value to MKV header.
Video track 2 contains 178415 frames.
eac3to processing took 1 hour, 1 minute.
Done.

so regarding the difference between the original and the processed track with nero at the re-run, is the resulting flac not lossess or whats going on there?
and when gaps are detected, is there a way to find out if the delay needed is positive or negative? then it would be possible to add that delay to the demuxed truehd track at a later point.

[73ChargerFan]

Report on Mask BD, Extra stream 00024.m2ts using eac3to v2.78
This is the first time I've encountered this, so I thought I'd post it here.

eac3to v2.78
command line: eac3to 2) 2: extra.mkv
------------------------------------------------------------------------------
M2TS, 1 video track, 5 audio tracks, 2 subtitle tracks, 0:27:17
1: Chapters, 2 chapters
2: VC-1, 480p24 /1.001 (3:2) with pulldown flags
3: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
4: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
5: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
6: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
7: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
8: Subtitle (PGS), English
9: Subtitle (PGS), German
[v02] Extracting video track number 2...
[v02] Removing VC-1 pulldown...
[v02] Muxing video to Matroska...
[v02] Video overlaps for 1 frames at playtime 0:00:00.
[v02] Video overlaps for 1 frames at playtime 0:00:00.
.
.
[v02] Video overlaps for 1 frames at playtime 0:34:01.
[v02] The MKV file was created without making use of the gap/overlap information.
[v02] Please check whether audio is in sync. If it is in sync everything is fine.
[v02] Otherwise you can ask eac3to to repeat the muxing. It will then automatically
[v02] make use of the detailed gap/overlap information.
Added fps value to MKV header.
Video track 2 contains 49058 frames.

gave 9684 "video overlaps" messages, or 1 every 10 seconds,
and a playtime of 0:34:06, which is precisely 30/24 times the correct playtime 0:27:17.

Running the command second time used the gaps file and the new mkv file has the correct video length.

The playtime of the first mkv file was checked using mpc-hc 918 and MediaInfo 0.7.7.4.

---------------------------------------------

This number of video overlaps in a file necessitates a second run.

fyi

---------------------------------------------
edit:

This didn't work as I thought. The video jitters and jutters every few seconds.

[digitlman]

Thanks for the great tools. they work just wonderfully! however i have a dilemma. I have used "eac3to and more gui" to easily convert bluray to mkv with flac audio in it. now i have a popcorn hour and it can't play the flac audio. i need to know the fastest/best/easiest method to batch convert some movies back into a m2ts file with the raw video and the audio converted back into LPCM which i can stream to my receiver. i know once i can get the h264 and lpcm files i can run the patch on the pcm to fix the header and use tsmuxer to make the m2ts file. but is there and easier way? or what is the best method to demux and convert the mkv?

thanks

[deathlord]

madshi,
Got a problem with The Simpsons Movie BD (eac3to 2.80.0.0):

eac3to 1) 2: g:\Simpsons\Video.mkv 3: g:\Simpsons\Engli
sh_5.1_dts-hdma.flac -down16
M2TS, 1 video track, 9 audio tracks, 24 subtitle tracks, 1:41:28
1: Chapters, 25 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
   (core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
4: DTS, French, 5.1 channels, 24 bits, 768kbps, 48khz
5: DTS, German, 5.1 channels, 24 bits, 768kbps, 48khz
6: AC3, Dutch, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
7: AC3, Dutch, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
8: AC3, Finnish, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
9: AC3, Swedish, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
10: AC3, English, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB
11: AC3, English, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB
12: Subtitle (PGS), English
13: Subtitle (PGS), English
14: Subtitle (PGS), French
15: Subtitle (PGS), French
16: Subtitle (PGS), German
17: Subtitle (PGS), German
18: Subtitle (PGS), Dutch
19: Subtitle (PGS), Dutch
20: Subtitle (PGS), Finnish
21: Subtitle (PGS), Finnish
22: Subtitle (PGS), Swedish
23: Subtitle (PGS), Swedish
24: Subtitle (PGS), English
25: Subtitle (PGS), French
26: Subtitle (PGS), German
27: Subtitle (PGS), Dutch
28: Subtitle (PGS), Finnish
29: Subtitle (PGS), Swedish
30: Subtitle (PGS), English
31: Subtitle (PGS), French
32: Subtitle (PGS), German
33: Subtitle (PGS), Dutch
34: Subtitle (PGS), Finnish
35: Subtitle (PGS), Swedish
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[v02] Muxing video to Matroska...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Reducing depth from 24 to 16 bits...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "g:\Simpsons\English_5.1_dts-hdma.flac"...
[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] The h264 muxer doesn't support this stream type yet.
[v02] Please send a 20MB sample to dear@madshi.net
Aborted at file position 22046294016.

Do you need a sample, and if yes, what exactly should I make the sample from?

Also, this movie is supposed to have force subs. But how do I see which it is?

[madshi]

Originally Posted by Thunderbolt8

got a problem with a TrueHD track here:

50mb sample: http://www.sendspace.com/file/y2diub

[a03] This track is not clean.
[a03] Decoding with libav/ffmpeg...
[libav] Stream parameters not seen; skipping frame
[a03] The libav decoder output an unexpected bitdepth (1).
Aborted at file position 49152.

Obviously the source file is not clean. The libav decoder doesn't seem to be able to handle this situation. Are you sure that the rip is not corrupt? I'd strongly suggest reripping the disk.

Originally Posted by Thunderbolt8

and when I do it with -nero (nero also decodes TrueHD 5.1 completely lossless, right?) then I get:

and after re-running the cmd line:

Audio gap description file detected, will be used for processing...
[a03] Original audio track, L+R+C+LFE: constant bit depth of 16 bits.
[a03] Original audio track, BL+BR: constant bit depth of 24 bits.
[a03] Processed audio track, L+R+C+SL+SR: max 24 bits, average 19 bits.
[a03] Processed audio track, LFE: max 23 bits, average 16 bits.

so regarding the difference between the original and the processed track with nero at the re-run, is the resulting flac not lossess or whats going on there?

I've recently added a post processing filter which smooths the audio cut points where gaps/overlaps are fixed to remove audio spikes. This post processing filter creates 24bit samples. That's why the bitdepth analyzation reports different results after the gaps have been fixed. In your specific case that's no problem cause the stream was overall (up to) 24bit, anyway. So it's still practically lossless. But for 16bit the behavior of the smoothing filter is not very good. So I'll change it for the next build so that it only creates 16bit samples.

Originally Posted by Thunderbolt8

and when gaps are detected, is there a way to find out if the delay needed is positive or negative? then it would be possible to add that delay to the demuxed truehd track at a later point.

When there's a gap you need to apply a positive delay. For overlaps you need to apply a negative delay. The gap/overlap amount (in milliseconds) is listed in the eac3to log of the first run.

Originally Posted by 73ChargerFan

Report on Mask BD, Extra stream 00024.m2ts using eac3to v2.78
This is the first time I've encountered this, so I thought I'd post it here.

2: VC-1, 480p24 /1.001 (3:2) with pulldown flags

gave 9684 "video overlaps" messages, or 1 every 10 seconds,
and a playtime of 0:34:06, which is precisely 30/24 times the correct playtime 0:27:17.

I'm aware of this problem. It occurs with VC-1 extras which are encoded as 480p30 with pulldown flags to 480i60. eac3to incorrectly identifies these as "480p24 with pulldown flags". So basically a wrong framerate is detected which results in the problems you noticed. I have on my to do list to fix this. It only occurs with some VC-1 extras, though, so it's not too bad...

Originally Posted by digitlman

I have used "eac3to and more gui" to easily convert bluray to mkv with flac audio in it. now i have a popcorn hour and it can't play the flac audio.

So please ask Syabas and Popcorn Hour to add support for MKV files with multichannel FLAC in it. It's about time they finally support that!

Originally Posted by digitlman

i need to know the fastest/best/easiest method to batch convert some movies back into a m2ts file with the raw video and the audio converted back into LPCM which i can stream to my receiver. i know once i can get the h264 and lpcm files i can run the patch on the pcm to fix the header and use tsmuxer to make the m2ts file. but is there and easier way? or what is the best method to demux and convert the mkv?

You can use mkvextract the demux the video and audio tracks. Then you can use eac3to to convert the FLAC to whatever format you need. Can't help you with m2ts muxing. That's outside the scope of this thread.

Originally Posted by deathlord

Got a problem with The Simpsons Movie BD (eac3to 2.80.0.0):

[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] Detected PTS break, increasing PTS by 41.7ms...
[v02] The h264 muxer doesn't support this stream type yet.
[v02] Please send a 20MB sample to dear@madshi.net
Aborted at file position 22046294016.

That's weird. Especially the PTS break before the muxing failure report. My best guess is that your source is corrupt and that the corruption is responsible for the problems. Can you please rerip the disk? Please rip to harddisk first and then run eac3to on the folder on harddisk. This is known to be the most stable way if you have problems with corruption.

Has anyone else remuxed the Simpsons movie with eac3to yet? Did it work ok for you?

[Jeff Flowerday]

Originally Posted by digitlman

Thanks for the great tools. they work just wonderfully! however i have a dilemma. I have used "eac3to and more gui" to easily convert bluray to mkv with flac audio in it. now i have a popcorn hour and it can't play the flac audio. i need to know the fastest/best/easiest method to batch convert some movies back into a m2ts file with the raw video and the audio converted back into LPCM which i can stream to my receiver. i know once i can get the h264 and lpcm files i can run the patch on the pcm to fix the header and use tsmuxer to make the m2ts file. but is there and easier way? or what is the best method to demux and convert the mkv?

thanks

If your flac was created from the lossless soundtrack at this point your only choice is to convert the audio to LPCM before stuffing it in a m2ts container. Unfortunaty your file size will increase dramatically, I've seen 4-7GB per file.

I ended up dumping the popcorn hour and putting a HTPC in it's place. With madshi's madflac directshow filter, my MKV's work flawlessly and the Media Portal interface is sooooooo much richer than the popcorn hour and/or the YAMJ plugin for iHome.

[rack04]

Madshi,

Could you lend any advice to neuron2 regarding how to detect dts-hd ma, dts-hd hr, and truehd?

http://forum.doom9.org/showthread.ph...45#post1223545

Thanks.

[rica]

Can somebody advise me how to cut the last part of any media.
-50MB option cuts the first 50MB part.

[nautilus7]

I don't think it can be done with eac3to. Use a hex editor instead.

[rica]

Oooo, i hate hex editors.
Thanks anyways.

[banker_rishad]

Madshi sir i tried converting vob to dts but the output is dts that is running away. I mean the audio is running or the tempo is high. what to do. plz advise.

[yonta]

Originally Posted by rica

Can somebody advise me how to cut the last part of any media.
-50MB option cuts the first 50MB part.

tail.exe can do that.

[Snowknight26]

Or clip.exe.

[asarian]

Originally Posted by tebasuna51

Is your issue. TsMuxer don't accept lpcm files. These files are raw audio data without header and can't be recognized out of a container than inform about bitdepth, channels, samplerate and endian.

Select wav like output file and can be recognized by TsMuxer if is <4GB (probably because 2 C and 16 bit). For wav files > 4GB (5.1 and > 130 min.) you need pcm output and Pcm2Tsmu.

This LPCM conversion is very exciting. I'm still not getting it entirely right. Here's an example:

eac3to c:\video\20000.m2ts 4: c:\video\wall-e.pcm
M2TS, 3 video tracks, 3 audio tracks, 3 subtitle tracks, 1:37:26
1: h264/AVC, 1080p24 /1.001 (16:9)
2: h264/AVC, 480p24 /1.001 (20:11)
3: h264/AVC, 480p24 /1.001 (20:11)
4: DTS Master Audio, 5.1 channels, 24 bits, 48khz
   (core: DTS-ES, 5.1 channels, 24 bits, 1509kbps, 48khz)
5: AC3 Surround, 2.0 channels, 192kbps, 48khz
6: AC3 Surround, 2.0 channels, 192kbps, 48khz
7: Subtitle (PGS)
8: Subtitle (PGS)
9: Subtitle (PGS)
[a04] Extracting audio track number 4...
[a04] Decoding with ArcSoft DTS Decoder...
[a04] Swapping endian...
[a04] Remapping channels...
[a04] Creating file "c:\video\wall-e.pcm"...
[a04] The last DTS frame is incomplete and thus gets skipped.
[a04] The original audio track has a constant bit depth of 24 bits.
Video track 1 contains 140154 frames.
Video track 2 contains 140154 frames.
Video track 3 contains 140154 frames.
eac3to processing took 13 minutes, 20 seconds.
Done.

So far so good (?). Now:

Pcm2Tsmu c:\video\wall-e.pcm c:\video\wall-e-lpcm.pcm -c 6

This kinda works. I get an audible 5.1 track (of around 5G), but a lot seems missing: like if the front channels were gone. Mind you, there's sound in the front channels, but it's almost as if things are miswired, as you can hear voices only very faintly, and other major stuff. Weird.

Maybe this has to do with the -16 switches (both for eac3to and Pcm2Tsmu). But if I use those, like:

eac3to c:\video\20000.m2ts 4: c:\video\wall-e.pcm -16
Pcm2Tsmu c:\video\wall-e.pcm c:\video\wall-e-lpcm.pcm -i 16 -c 6

Then my PS3 just hangs at the start of the movie (doesn't even start playing the video). So, my question is, how is this down-converting to 16bit supposed to go then? Or, in case down-converting isn't necessary, why am I missing so much audio on the 24bit mix? Other than clearly some major channels/sound missing, what is being output actually sound superb, btw. So I guess I'm close.

Thanks

[asarian]

^^ Seems I'm in bad luck. I send the LPCM over a Toslink, and apparently that only does stereo (I thought it would do 5.1 LPCM as well). Alas.

Interestingly, though, the PS3 actually lists the audio as "Linear PCM, 5.1ch, 6.9Mbps", while streaming (via Twonky). The latter is remarkable, as I always thought, until now, that 640kbps AC3 was the limit for streaming on the PS3. And now it turns out it does full 5.1 LPCM, and at near 7Mbps even! (converted from the DTS-HD track).

@Madshi: still, I'm not a man without hope, as I can now still try my Dolby ProLogic II trick; this time, however, not with a measly 640kbps AC3 stream, but with a 7Mbps LPCM stereo stream: 11x as much bandwidth.

Still, seems the PS3 streams a whole lot more than 640kbps AC3, after all. And that is good news. That I just need to buy a better amp is irrelevant here.

[n0mag!c]

Originally Posted by rica

Can somebody advise me how to cut the last part of any media.
-50MB option cuts the first 50MB part.

To cut last part of any file you can use CutTools.
It works in primitive binary mode and doesn't know anything about frames or packets of specific file types. You can calculate cut point for yourself if you know frame or packet length.

[telmoMRC]

some days ago i have convert one dts audio to ac3 384 but now doesn´t work the dts audio [input.dts] is in the same directory the eac3to and give this error source file "input.dts" not found wy do not work ???

screen
http://e.imagehost.org/0702/nao_d.jpg

[madshi]

Originally Posted by banker_rishad

i tried converting vob to dts but the output is dts that is running away. I mean the audio is running or the tempo is high.

I don't really know what you mean. And posting attachments to this forum doesn't make sense, unfortunately, because it usually takes years until they are approved.

Originally Posted by asarian

This LPCM conversion is very exciting. I'm still not getting it entirely right.

Both Pcm2Tsmu and tsMuxeR have their own threads. Please ask there for help. I cannot help you and this topic doesn't really belong here.

Originally Posted by telmoMRC

some days ago i have convert one dts audio to ac3 384 but now doesn´t work the dts audio [input.dts] is in the same directory the eac3to and give this error source file "input.dts" not found wy do not work ???

It seems that your OS is configured in such a way that the file extensions of files which are known to the OS are not displayed. So my best guess is that the file's real name is "input.dts.dts".

[madshi]

Originally Posted by komisar

Can you add to eac3to ability for "custom-comman-line-parameters"?
E.g. for neroaacenc:

eac3to.exe input.ac3 output.aac -quality=0.23 -lc

Not sure how many funny parameters neroaacenc has. Right now I'd have to implement every single one by hand to make them all work. That's no fun, so I'm probably not gonna do it. I might sooner or later find a way to automatically pass all unknown parameters to the encoder module. Then what you're looking for would automatically work.

[telmoMRC]

Originally Posted by madshi

It seems that your OS is configured in such a way that the file extensions of files which are known to the OS are not displayed. So my best guess is that the file's real name is "input.dts.dts".

i try that and doesn´t work what can i do ?

[rica]

Originally Posted by yonta

tail.exe can do that.

Originally Posted by Snowknight26

Or clip.exe.

Thanks guys.

Can you please give me a command line sample for tail.exe and clip.exe?

usage: tail <infile> <outfile> <length>

For tail i gave it a go with:

C:\>clip\tail  E:\BDMV\stream\00181.m2ts C:\HD\out.m2ts 500MB

But nothing has happened.

For clip.exe...???

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

***madshi, thanks for your understanding.

[asarian]

Originally Posted by madshi

Both Pcm2Tsmu and tsMuxeR have their own threads. Please ask there for help. I cannot help you and this topic doesn't really belong here.

Fortunately, for me, I have neither a tsMuxeR, nor a Pcm2Tsmu issue. I just like to know one eac3to thing; when I do this:

eac3to 20000.m2ts 4: wall-e.pcm -down2

eac3to lowers the bitrate to about 2Mbps. Any way I can keep the original bitrate for the two channels?

[yonta]

If you want to cut out the last 50MBytes

tail.exe infile outfile 50000000

[digitlman]

i dunno which place to post this as i am not sure which program is the problem but i will start here...

i have tried this with 2 different mkv+flac files and neither flac file is recognized by eac3to. is there an option with eac3to i need to change or is it something wrong with my mkvextract options?

mediainfo recognizes the flac files just fine, says they are 6channel 24bit 48000hz. i also tried using the --no-ogg option with same results

i am just trying to convert the flac file to lpcm.


C:\Program Files\MKVtoolnix>mkvextract tracks "f:\Pirates Of The Caribbean 1 AVC
LPCM24.mkv" 1:d:\p1.h264 2:d:\p1.flac
Extracting track 1 with the CodecID 'V_MPEG4/ISO/AVC' to the file 'd:\p1.h264'.
Container format: AVC/h.264 elementary stream
Extracting track 2 with the CodecID 'A_FLAC' to the file 'd:\p2.flac'. Container
format: Ogg (FLAC in Ogg)
progress: 100%

C:\Program Files\MKVtoolnix>

C:\eac3to>eac3to d:\p1.flac
The format of the source file could not be detected.

C:\eac3to>

C:\eac3to>eac3to d:\cars.flac h:\cars3.pcm
The format of the source file could not be detected.

[rica]

Originally Posted by yonta

If you want to cut out the last 50MBytes

tail.exe infile outfile 50000000

Thanks, i'd like to have a sample CMD line of clip.exe as well?

"start offset" ?

[yonta]

You can cut out 1920000 bytes from a file starting from the 193rd byte.

clip infile outfile 193 1920000

[madshi]

Originally Posted by telmoMRC

i try that and doesn´t work what can i do ?

First of all configure your OS correctly. Letting the OS hide known extensions is the best way to confuse yourself.

Originally Posted by asarian

I just like to know one eac3to thing; when I do this:

eac3to 20000.m2ts 4: wall-e.pcm -down2

eac3to lowers the bitrate to about 2Mbps. Any way I can keep the original bitrate for the two channels?

The bitrate for PCM is a simple calculation:

bitrate = bitDepth * channels * sampleRate

So if you go from 6 to 2 channels, obviously the bitrate decreases. There's nothing you can do about that, and there's no reason to do anything about it. Bitrate doesn't matter with PCM, it only matters for lossy compression.

Originally Posted by digitlman

i dunno which place to post this as i am not sure which program is the problem but i will start here...

i have tried this with 2 different mkv+flac files and neither flac file is recognized by eac3to. is there an option with eac3to i need to change or is it something wrong with my mkvextract options?

mediainfo recognizes the flac files just fine, says they are 6channel 24bit 48000hz. i also tried using the --no-ogg option with same results

You do have to use the "no ogg" option, because eac3to doesn't support the OGG container. My best guess is that you put the option at the wrong place in the mkvextract command line. That has happened before...

Originally Posted by rica

Can you please give me a command line sample for tail.exe and clip.exe?

I hope that you are asking these questions because you want to make a sample for me? Because if not you're once again very much out of topic here...

[asarian]

Originally Posted by madshi

The bitrate for PCM is a simple calculation:

bitrate = bitDepth * channels * sampleRate

So if you go from 6 to 2 channels, obviously the bitrate decreases. There's nothing you can do about that, and there's no reason to do anything about it.

The reason to keep the high bitrate was to use the Dolby Prologic II trick, so as to use the full 6.9Mbps for the 2 channels.

Anyway, looks like I'll have to buy the Marantz SR-8002 soon, and do things properly.

Bitrate doesn't matter with PCM, it only matters for lossy compression.

? I thought bitrate always matters. I mean, isn't a 8Mbps LPCM stream better than a 640kpbs one?

[madshi]

Originally Posted by asarian

? I thought bitrate always matters. I mean, isn't a 8Mbps LPCM stream better than a 640kpbs one?

Have you tried to understand the LPCM bitrate formula I gave you? If so, you would have understood by now that the bitrate you got is lower because you decreased the number of channels. The only way to increase the bitrate again is to either increase the samplerate or the bitdepth. Both of which would not improve audio quality in your case. That's the last thing I'll say on this topic.

[kurt]

how to demux mpeg2 video properly?

eac3to xxx.m2ts video.m2v does not work (audio conversion not supported). neither than mpeg2, mpeg, mpg.....

(I know there is the -demux switch)

[nautilus7]

Well, use the track number format. Something like:

eac3to xxx.m2ts 2: video.m2v

[madshi]

Originally Posted by kurt

how to demux mpeg2 video properly?

eac3to xxx.m2ts video.m2v does not work (audio conversion not supported). neither than mpeg2, mpeg, mpg.....

It should work the way you did it. Please post the full eac3to log (between [ code ][ / code ]).

[kurt]

Originally Posted by madshi

It should work the way you did it. Please post the full eac3to log (between [ code ][ / code ]).

sorry, my fault. it works with *.m2v as destination file. I think my problem was a space character in destination folder. thx, madshi!

@ nautilus: jap, Í know. just forgot to write the number above

[LessThanJake]

Hello,

eac3to can handle whole BD and HD DVD structures.
are there plans to add full DVD-structure with chapter support too?

greets
LTJ

[rica]

Originally Posted by madshi

I hope that you are asking these questions because you want to make a sample for me? Because if not you're once again very much out of topic here...

I'm out of the topic at neuron2's thread i suppose

http://forum.doom9.org/showthread.ph...02#post1224102

_ _ _ _

[crazydane]

Madshi,

A while back the user itsancho had an issue with Basic Instinct, where you told him to use the -seekToIFrames option to overcome a known bug in Haali's muxer. Using that flag worked like a charm for him.

However, when I try doing the same thing, I still get the same error.

Here's my log:

eac3to v2.80
command line: eac3to BASIC_INSTINCT 1) 1: chapters.txt 2: video.mkv 3: audio.flac -seekToIFrames
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 2 subtitle tracks, 2:08:15
1: Chapters, 17 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Hi-Res, English, 6.1 channels, 24 bits, 3039kbps, 48khz, dialnorm: -4dB
(core: DTS-ES, 6.1 channels, 24 bits, 1509kbps, 48khz, dialnorm: -4dB)
4: AC3 EX, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
5: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
6: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
7: Subtitle (PGS), English
8: Subtitle (PGS), Spanish
Creating file "chapters.txt"...
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[v02] Muxing video to Matroska...
[a03] Remapping channels...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Removing DTS dialog normalization...
[a03] Encoding FLAC with libFlac...
[a03] Creating file "audio.flac"...
Unfortunately the Haali Muxer cannot handle this source file.
It doesn't contain enough seek/recovery points.
The "-seekToIFrames" option may work around that.
However, using this option may result in artifacts when seeking.
Aborted at file position 147783680.

Any ideas?

Btw, my issue with running eac3to directly from the DVD drive dissappeared for no reason. Strange...

[Thunderbolt8]

its a known probelem, use the switch -seektoIframes here.

edit: misread the beginning :S
actually it should work, at least it did some versions ago for me (probably around 2.70)

[deathlord]

Originally Posted by madshi

That's weird. Especially the PTS break before the muxing failure report. My best guess is that your source is corrupt and that the corruption is responsible for the problems. Can you please rerip the disk? Please rip to harddisk first and then run eac3to on the folder on harddisk. This is known to be the most stable way if you have problems with corruption.

You were right, madshi. Reripping helped.
Thanks for your continued support!

[madshi]

Originally Posted by LessThanJake

are there plans to add full DVD-structure with chapter support too?

Plans, yes.

Originally Posted by crazydane

command line: eac3to BASIC_INSTINCT 1) 1: chapters.txt 2: video.mkv 3: audio.flac -seekToIFrames

You're asking eac3to to apply the "-seekToIFrames" option on the audio track 3.

[madshi]

eac3to v2.81 released

http://madshi.net/eac3to.zip

* audio gap/overlap fixing is now automatically done in a 2nd pass
* option "-normalize" maximizes the volume of the audio data, needs 2 passes
* audio clipping is detected and automatically removed in a 2nd pass
* "-2pass" activates 2 pass mode (can speedup seamless branching processing)
* superfluous zero bytes are now automatically removed in 2nd pass
* "-phaseShift" shifts surround channel phase by 90° for DPL II downmixing
* spike removal post processing filter now always produces 16bit samples
* empty channels are now reported by the bitdepth analyzer as "no audio data"
* option "-shutdown" shuts the PC down automatically after processing is done
* the HD DVD XPL with the longest title is now loaded instead of VPLST000.XPL
* eac3to can now open specific XPL files (e.g. "eac3to ADV_OBJ\VPLS002.XPL")
* eac3to can now open specific mpls files (e.g. "eac3to PLAYLIST\00002.mpls")
* fixed: TrueHD streams starting with a non-major header failed to decode
* fixed: WAV files created by eac3to with empty channels had incorrect header
* fixed: RAW/PCM gap/overlap remover sometimes didn't work correctly

This is quite a big release. Especially the automated 2 pass processing is a change which goes quite deep. Because of that there might be new bugs. As a backup plan I've added the previous v2.80 version into the v2.81 zip file, so if all else fails you can use v2.80 instead.

"Gaps" files don't exist for audio tracks, anymore, now. However, for video tracks there may still be "gaps" files. And if gaps/overlaps are detected for video tracks, you'll still need to manually restart processing to fix them. I've decided to not automate gap/overlap fixing for video tracks because (1) video gaps/overlaps are extremely rare and (2) even if there are gaps/overlaps detected for video tracks, sometimes it's better not to fix them. But audio gaps/overlaps are now automatically fixed. If possible, this is done without reading the big movie source file again. Also the video track is only ever written once, regardless of whether there are audio gaps/overlaps or not.

[Thunderbolt8]

thanks!

which might be the situations itll be better not to correct video gaps? (apart from those that I can spot a quite a delay on a first glance of course)

[nautilus7]

Thanks! Now I need a seamless branching blu-ray to test the changes.

[rica]

Man, i like your revolutions.

[rica]

eac3to v2.81
command line: eac3to\eac3to  E: 1) 4: C:\HD\audio.dtshd -8000MB
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 20 subtitle tracks, 2:17:13
1: Chapters, 20 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz
   (embedded: AC3, 5.1 channels, 448kbps, 48khz)
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
   (core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
5: Subtitle (PGS), English
*********
[a04] Extracting audio track number 4...

[a04] Creating file "C:\HD\audio.dtshd"...
[a04] Audio overlaps for 7ms at playtime 0:23:06.
[a04] Audio overlaps for 6ms at playtime 0:26:06.
[a04] Audio overlaps for 11ms at playtime 0:30:37.

[a04] Starting 2nd pass...
[a04] Realizing DTS gaps...
[a04] Creating file "C:\HD\audio.dtshd"...
eac3to processing took 15 minutes, 37 seconds.
Done.

[mrr19121970]

@madshi

Thanks for all the hard work man. It's appreciated.

I've a feature request concering the -demux switch and the order the titles are presented... Currently the longest title is presented 1st, and generally correct. However in Clone-Wars and a couple of others - this is not the case. The SD 'bonus feature' has a longer duration than the HD 'main feature'. My request is can HD content take priority over SD content before length of feature ?



Thanks. Mike.

[madshi]

Originally Posted by Thunderbolt8

which might be the situations itll be better not to correct video gaps?

Generally if there are video gaps/overlaps reported I'd recommend to first not fix them and mux everything to the final target container (e.g. MKV or m2ts). Then check the runtime where eac3to reported video gaps/overlaps and check whether audio sync is lost at that runtime or not. If you don't notice anything weird, just leave things as they are and don't fix the reported video gaps/overlaps. Let eac3to fix that only if you notice that audio goes out of sync at the reported runtime.

Sometimes eac3to reports video gaps or overlaps for every second of runtime throughout the whole movie. In that case there are most probably no gaps/overlaps, but instead eac3to detected the movie's FPS incorrectly.

Originally Posted by mrr19121970

I've a feature request concering the -demux switch and the order the titles are presented... Currently the longest title is presented 1st, and generally correct. However in Clone-Wars and a couple of others - this is not the case. The SD 'bonus feature' has a longer duration than the HD 'main feature'. My request is can HD content take priority over SD content before length of feature ?

Yes, that makes sense. Will add that to my to do list.

[Momber]

Originally Posted by madshi

option "-shutdown" shuts the PC down automatically after processing is done

You beauty
Thanks so much!

[Greif]

Originally Posted by rica

eac3to v2.81
command line: eac3to\eac3to  E: 1) 4: C:\HD\audio.dtshd -8000MB

Is that setting a target size of 800MB for the DTS-HD track? I didn't know you could do that.

[nautilus7]

Actually it's 8000MB, not 800MB. And it refers to the source file. It should read the source file till it reaches 8000MB. The output file size will be unknown.

Am i the only one that have troubles with v2.81? I can't do anything with it. It won't even display the file properties if i do eac3to input.file. I have tried with ac3 and flac. I tried also a conversion, but nothing. It simply does nothing, no progress bar, nothing.

[madshi]

Originally Posted by nautilus7

Am i the only one that have troubles with v2.81? I can't do anything with it. It won't even display the file properties if i do eac3to input.file. I have tried with ac3 and flac. I tried also a conversion, but nothing. It simply does nothing, no progress bar, nothing.

Please try downloading again. Maybe something went corrupt? Does the v2.80 exe in the v2.81 zip work for you?

[nautilus7]

I have already downloaded again. Both times same result. And v2.80 from the same zip works fine.

[Chumbo]

Originally Posted by madshi

Please try downloading again. Maybe something went corrupt? Does the v2.80 exe in the v2.81 zip work for you?

I'm having the same issue. I just tested it on a vob and a ts file and nothing. You can't even ctrl-c to exit. I had to kill the eac3to process with task manager.

The included 2.80 works fine. If you run 2.81 without parameters, it does show the help stuff, but doesn't seem to work with any switches. I don't think the zip file is corrupt or we couldn't unzip it successfully.

[nautilus7]

Exact same issues. I couldn't stop it with ctrl+c when used a flac file for input (just it), but i can stop it with ac3 input. But other symptoms are the same.

[madshi]

Weird. Does "eac3to -test" work? How about "eac3to nonExistingFile"? Please download this:

http://madshi.net/madTraceProcess.zip

And then run it and try to create a report for the stuck eac3to process. If you succeed getting a report, please post it here. Thanks!

[nautilus7]

-test works. And using a non existing file works as it should. It also works with (m2)ts container and normal commands related to it (track extraction etc).

This is from an existing file:

removed as it contains personal data

[madshi]

eac3to v2.82 released

http://madshi.net/eac3to.zip

* fixed: sometimes eac3to stalled before processing (introduced in v2.81)

[nautilus7]

Thanks! You're the best!

(PS: others who will get it make sure that you get v2.82.)

[Thunderbolt8]

is there actually a possibility to get rid of short fps spikes for video stream when rewriting the fps? when I slow down a movie which has these short spikes, the fps rate still increases a little at those points and therefore any external audio I'd try to sync to it goes oos from that point then. the internal audio from the .ts seems to spike as well then, as it stays in sync.

[madshi]

Originally Posted by rica

I've just tried the new version; where are the overlaps?

The first time you had used -8000MB and not -800MB.

Originally Posted by Thunderbolt8

is there actually a possibility to get rid of short fps spikes for video stream when rewriting the fps? when I slow down a movie which has these short spikes, the fps rate still increases a little at those points and therefore any external audio I'd try to sync to it goes oos from that point then. the internal audio from the .ts seems to spike as well then, as it stays in sync.

What kind of fps spikes to you mean? The only times I've seen video played sped up is when the audio track is a mixture of 5.1 and 2.0. In that case the audio decoder gets confused and reports funny timecodes and that results in video fps getting confused, as well. If that's the problem you're seeing then there's nothing that can (and should!) be done with video muxing, cause it's already correct. Just fetch a proper AC3 track from a DVD and mux that with the video.

[rica]

Originally Posted by madshi

The first time you had used -8000MB and not -800MB.

Please read again the previous post; it was 800MB.

[madshi]

Originally Posted by rica

Please read again the previous post; it was 800MB.

Please read again your own v2.81 log.

[rica]

Originally Posted by madshi

Please read again your own v2.81 log.

Yes, i read the original log as well; i must admit you're right.
And 8000MB includes four m2ts files so this is why i selected 8000.
By mistake i remembered it as 800.
Sorry for the inconvenience.
@all, forget it.

Edit: Previous post has been deleted by myself.... False info...

Btw, this has been deleted in the same post as well: (restored back)

Originally Posted by Greif

Is that setting a target size of 800MB for the DTS-HD track? I didn't know you could do that.

No, this is the first part of the whole m2ts' cut lenght. (8000MB)
Nobody knows what lenght audio will be, as nautilus said.
I used this cut option to see the result immediately. (this lenght has been selected since it included four m2ts files so as to see the gaps/overlaps)

[Blackwalker]

hi all!
this is the first time i'm trying to use eac3to.

i had problem with audio demuxing blu-ray films with tsmuxer,
so someone suggest me to use this program.

there is a "Guide" for a noob , to extract Audio & Video
from a ".m2ts" file?

thx!!!

[deathlord]

Originally Posted by Blackwalker

hi all!
this is the first time i'm trying to use eac3to.

i had problem with audio demuxing blu-ray films with tsmuxer,
so someone suggest me to use this program.

there is a "Guide" for a noob , to extract Audio & Video
from a ".m2ts" file?

thx!!!

Hi

Read the first post of this thread! It contains a link to the eac3to wikibook. There you can find many examples on how to use eac3to.

deathlord

[Blackwalker]

Originally Posted by deathlord

Hi

Read the first post of this thread! It contains a link to the eac3to wikibook. There you can find many examples on how to use eac3to.

deathlord

i just check it, but my english is not so good...

if there is a guide with screenshots to how extract
audio and video form a ".m2ts" file it'd be cool!!!
if no i have to take my dictionary and see if i can understand
how i can use it

ho well, tyvm anyway!

[jmonier]

Originally Posted by Blackwalker

i just check it, but my english is not so good...

if there is a guide with screenshots to how extract
audio and video form a ".m2ts" file it'd be cool!!!
if no i have to take my dictionary and see if i can understand
how i can use it

ho well, tyvm anyway!

Try this (as listed in the first post):

http://forum.doom9.org/showthread.php?t=135095

Extract from an iso NOT from m2ts

[Blackwalker]

Originally Posted by jmonier

Try this (as listed in the first post):

http://forum.doom9.org/showthread.php?t=135095

Extract from an iso NOT from m2ts

you mean i have to create a "ISO" and not a ".m2ts" file from
blu-ray??

anyway, the thread talk about mkv and not about extracting Audio & Video from a ".m2ts" or "Blu-ray" disc...or my english is
so bad???

[jmonier]

Originally Posted by Blackwalker

you mean i have to create a "ISO" and not a ".m2ts" file from
blu-ray??

Yes

anyway, the thread talk about mkv and not about extracting Audio & Video from a ".m2ts" or "Blu-ray" disc...or my english is so bad???

To actually play it back you will have to combine Audio & Video into an mkv using mkvmerge. It will just extract them if that's what you want to do.

This is really OT for this thread. Post further questions in that thread and I'll try to answer them.

[Blackwalker]

Originally Posted by jmonier

Yes



To actually play it back you will have to combine Audio & Video into an mkv using mkvmerge. It will just extract them if that's what you want to do.

This is really OT for this thread. Post further questions in that thread and I'll try to answer them.

This is OT?

someone on this forum point me here and told me that eac3to is the program i need to extract from a ".m2ts" the audio & video!
i'm not talking about mkv....

Maybe i'm wrong, but i DONT think its OT.

someone else can confirm this, this program, eac3to, can do what i need? extractint the audio & video form a ".m2ts" file or ".ts" file?

i'm a still a noob with blu-ray,but i think i need someone more expert!
thx!

[nautilus7]

All these posts asking how to use eac3to...
You have been directed to where you can find examples. You can also search this thread. Saying that your English is bad is no excuse.

Anyway, is it so hard to look a few pages back and find an example with m2ts file?

[jmonier]

Originally Posted by Blackwalker

This is OT?

someone on this forum point me here and told me that eac3to is the program i need to extract from a ".m2ts" the audio & video!
i'm not talking about mkv....

Maybe i'm wrong, but i DONT think its OT.

someone else can confirm this, this program, eac3to, can do what i need? extractint the audio & video form a ".m2ts" file or ".ts" file?

i'm a still a noob with blu-ray,but i think i need someone more expert!
thx!

What was OT was discussing the other (GUI) program and how it makes it easy to use eac3to and how it provides screenshots (which is what you asked for). I did try to answer your question but felt it would be better to not clutter up this thread with questions about that program.

I do use eac3to via that GUI constantly so I am at least considerably more expert than you. Since your English is so bad it's easy for you to misunderstand. You should realize this and at least be more courteous with someone who is trying to help you.

[Blackwalker]

wasnt my intenction to be rude in any way, really.

...and my "bad english" is not a excuse...bha!

Anyway, is it so hard to look a few pages back and find an example with m2ts file?

about 400 pages...

i'm "just" asking how to use eac3to...i dont think its a crime!
at last you can reply like you just do it! or maybe you need a "+1" on your post count?

like many other i'm asking this, because i didnt see any real guide, like many other programs
with screenshoots on how to use the programs,
at last i didnt find them, thats why i'm asking.

(do you see for example the guide of ripbot? that is a real guide)


ok, i'll check by myself...i see that asking is not accepted, thx anyway and have a nice day!

i'll check some pages back or i'll read all the 369 PAGES!!

[nautilus7]

I wrote "a few pages back". All people here are dealing with m2ts files. Someone has already posted a log file/example recently.

[Blackwalker]

i'll check around, thx anyway nautilus7

[rica]

Have a look at this:

http://en.wikibooks.org/wiki/Eac3to/How_to_Use

More details may be found at:

http://en.wikibooks.org/wiki/Eac3to

[Beastie Boy]

Is eac3to or any of the default filters known to lower lfe channel volume?
I have been converting lossless audio tracks (THD & DTSHD MA) to 640kbps ac3. I have done a listening test, and the only detectable difference is a noticably weaker bass.

I always convert from the lossless data, not core. This morning, I tested the DTS MA track on Mamma Mia and bass is certainly stronger on the DTS track compared to my DD track. I noticed the same effect with Iron Man which was True-HD.

Would I need to rip to wavs first and apply a boost to the lfe channel only?

Cheers, Beastie.

EDIT: If I do: eac3to DTSHD_MA_Track.ts New_ac3_track.ac3

will this use the HD data or the core data as the source?

[tebasuna51]

Originally Posted by Beastie Boy

Is eac3to or any of the default filters known to lower lfe channel volume?

No, Aften have a LFE low-pass filter but the default is don't use.

I have been converting lossless audio tracks (THD & DTSHD MA) to 640kbps ac3. I have done a listening test, and the only detectable difference is a noticably weaker bass.

These problems are, most the times, by different settings in players. The sources are identical (more os less).

Would I need to rip to wavs first and apply a boost to the lfe channel only?

This never solve the problem, only add more problems.

will this use the HD data or the core data as the source?

If you have ArcSoft/Sonic the HD, if you only decode with libav the core.

[madshi]

Look here for more information about LFE volume:

http://www.avsforum.com/avs-vb/showthread.php?t=748147

It's a complicated topic, but as tebasuna51 says, probably caused by settings in either your player or your receiver/amp. Strange enough if there are problems it's usually DTS which has weaker bass compared to AC3. So something is funny in your setup...

[Beastie Boy]

Thanks for your replies. I'm bitstreaming both codecs to my amp (Marantz SR8002) so maybe it treats the two codecs differently. Just as a comparison, I examined a portion of the lfe wave file from both the core and the HD data, and both seem to be at the same level.
I'll test playback of the core DTS track and see if there's an improvement.

Cheers, Beastie.

[Beastie Boy]

[offtopic, but follow-up to earlier post]
After comparing samples, the levels for all the codecs are the same. However, after some listening tests, it seems as though my amp way overblows the LFE channel of HD codecs. Both AC3 and standard DTS play correctly but, THD and DTS-HD bass totally dominates the soundtrack. I guess I'll be sticking with AC3.

Cheers, Beastie.

[73ChargerFan]

Weird, I was going to ask about low LFE also. I don't have a clue yet, though, as to what to do.

[/ offtopic]

[Killroy™]

Tried 2.82 for the first time today with a seamless branch BD and kept getting this....

[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[a04] Decoding with ArcSoft DTS Decoder...
[a04] Encoding FLAC with libFlac...
[v02] Muxing video to Matroska...
[a04] Creating file "c:\bd\movie.flac"...
[a04] The last DTS frame is incomplete and thus gets skipped.
[a04] The original audio track has a constant bit depth of 24 bits.
[a04] Audio overlaps for 10ms at playtime 0:21:43.
[a04] Audio overlaps for 10ms at playtime 0:45:09.
[a04] Audio overlaps for 9ms at playtime 0:46:59.
[a04] Audio overlaps for 12ms at playtime 0:55:06.
[a04] Audio overlaps for 9ms at playtime 1:00:39.
[a04] Audio overlaps for 9ms at playtime 1:02:56.
[a04] Audio overlaps for 6ms at playtime 1:03:40.
[a04] Audio overlaps for 6ms at playtime 1:08:03.
[a04] Audio overlaps for 8ms at playtime 1:09:37.
[a04] Audio overlaps for 9ms at playtime 1:12:00.
[a04] Audio overlaps for 10ms at playtime 1:14:02.
[a04] Audio overlaps for 8ms at playtime 1:16:25.
[a04] Audio overlaps for 13ms at playtime 1:41:02.
[a04] Audio overlaps for 7ms at playtime 1:45:20.
[a04] Audio overlaps for 6ms at playtime 1:47:45.
[a04] Audio overlaps for 9ms at playtime 1:52:03.
[a04] Audio overlaps for 7ms at playtime 1:59:16.
[a04] Audio overlaps for 6ms at playtime 2:01:06.
[a04] Audio overlaps for 9ms at playtime 2:44:49.
[a04] Audio overlaps for 45ms at playtime 2:48:33.
[a04] Source file joining currently doesn't work for FLAC files.

Also tried -2pass and it also failed.

[nautilus7]

Well, i think the last line is self explained.

[Killroy™]

It worked in previous versions.

[Atak_Snajpera]

Can somebody send me sample (.m2ts or .evo) with audio overlaps? Unfortunately all my blu-rays are clean.

[madshi]

Originally Posted by Killroy™

Tried 2.82 for the first time today with a seamless branch BD and kept getting this....

Can you please post the full log, so that I can see the track listing and your exact command line? Thanks!

[Killroy™]

Originally Posted by madshi

Can you please post the full log, so that I can see the track listing and your exact command line? Thanks!

Sure thing:

eac3to v2.82
command line: eac3to  k:\unrar\movie 1) 2: c:\bd\movie.mkv 4: c:\bd\movie.flac
------------------------------------------------------------------------------
M2TS, 2 video tracks, 4 audio tracks, 3 subtitle tracks, 2:55:45
1: Chapters, 21 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: VC-1, 480p24 /1.001 (3:2)
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
   (core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
5: DTS, French, 5.1 channels, 24 bits, 768kbps, 48khz
6: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
7: DTS Express, English, 2.0 channels, 24 bits, 192kbps, 48khz
8: Subtitle (PGS), English
9: Subtitle (PGS), Spanish
10: Subtitle (PGS), French
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[a04] Decoding with ArcSoft DTS Decoder...
[a04] Encoding FLAC with libFlac...
[v02] Muxing video to Matroska...
[a04] Creating file "c:\bd\movie.flac"...
[a04] The last DTS frame is incomplete and thus gets skipped.
[a04] The original audio track has a constant bit depth of 24 bits.
[a04] Audio overlaps for 10ms at playtime 0:21:43.
[a04] Audio overlaps for 10ms at playtime 0:45:09.
[a04] Audio overlaps for 9ms at playtime 0:46:59.
[a04] Audio overlaps for 12ms at playtime 0:55:06.
[a04] Audio overlaps for 9ms at playtime 1:00:39.
[a04] Audio overlaps for 9ms at playtime 1:02:56.
[a04] Audio overlaps for 6ms at playtime 1:03:40.
[a04] Audio overlaps for 6ms at playtime 1:08:03.
[a04] Audio overlaps for 8ms at playtime 1:09:37.
[a04] Audio overlaps for 9ms at playtime 1:12:00.
[a04] Audio overlaps for 10ms at playtime 1:14:02.
[a04] Audio overlaps for 8ms at playtime 1:16:25.
[a04] Audio overlaps for 13ms at playtime 1:41:02.
[a04] Audio overlaps for 7ms at playtime 1:45:20.
[a04] Audio overlaps for 6ms at playtime 1:47:45.
[a04] Audio overlaps for 9ms at playtime 1:52:03.
[a04] Audio overlaps for 7ms at playtime 1:59:16.
[a04] Audio overlaps for 6ms at playtime 2:01:06.
[a04] Audio overlaps for 9ms at playtime 2:44:49.
[a04] Audio overlaps for 45ms at playtime 2:48:33.
[a04] Source file joining currently doesn't work for FLAC files.

[madshi]

Originally Posted by Killroy™

Sure thing: [...]

Ouch. Pretty serious bug. So:

[madshi]

eac3to v2.83 released

http://madshi.net/eac3to.zip

* fixed: gap/overlap correction didn't work for FLAC and WAV files
* fixed: when clipping was detected, 2nd pass was not always executed correctly

[Killroy™]

Originally Posted by madshi

eac3to v2.83 released

http://madshi.net/eac3to.zip

* fixed: gap/overlap correction didn't work for FLAC and WAV files
* fixed: when clipping was detected, 2nd pass was not always executed correctly

Wow, that was fast!!! Thanks.

I'll give it a shot on the next seamless branch.

I fixed the other one with v2.69.

[rica]

2.81 and 2.83.
What are the the differences?
Can you make a change log between them?

[nautilus7]

What do you mean?
madshi always posts change logs with new versions. In addition a complete eac3to change log is included within eac3to's zip file.

[rica]

Originally Posted by nautilus7

What do you mean?
madshi always posts change logs with new versions. In addition a complete eac3to change log is included within eac3to's zip file.

Nothing.
I just wanted to say there are no appearent changes between them as .81 was already making gaps/overlaps correction on transcoding flac.
The question; what is beyond this?

[zeropc]

madshi, is there any chance for mkv import into eac3to or are there any complications of the lacking format support?

[Thunderbolt8]

Originally Posted by rica

Nothing.
I just wanted to say there are no appearent changes between them as .81 was already making gaps/overlaps correction on transcoding flac.
The question; what is beyond this?

bug fixes

[madshi]

Originally Posted by zeropc

is there any chance for mkv import into eac3to

It may come some day.

[NimaBeamer]

When creating a PCM or LPCM from a seamless branched disc with TrueHD for example should the audio gaps i.e. the second pass occur or not? I am asking because it is not with the newest version.

[madshi]

Originally Posted by NimaBeamer

When creating a PCM or LPCM from a seamless branched disc with TrueHD for example should the audio gaps i.e. the second pass occur or not? I am asking because it is not with the newest version.

Depends on how many m2ts parts the movie consists of. If there are only 2 or 3 parts, then maybe the gap amount is too small to be of concern for eac3to. If there are more than 2-3 parts and there are still no gaps reported then that might be worth worrying about...

[NimaBeamer]

Madshi the movie consits of over 10 M2TS files

The Log:
eac3to v2.82
command line: "F:\xxxxxxxxxxxxx\EAC3to\eac3to\eac3to.exe" "G:\xxxx\" 1) 2: "G:\xxxx\video.mkv" 4: "G:\xxxx\audio.pcm"
------------------------------------------------------------------------------
M2TS, 2 video tracks, 6 audio tracks, 19 subtitle tracks, 1:42:14
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
(embedded: AC3, 5.1 channels, 448kbps, 48khz)
5: AC3, English, 5.1 channels, 448kbps, 48khz, dialnorm: -30dB
6: TrueHD/AC3, German, 5.1 channels, 48khz
(embedded: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB)
7: AC3, Russian, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
8: AC3, Ukrainian, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
9: DTS Express, English, 2.0 channels, 24 bits, 192kbps, 48khz
10: Subtitle (PGS), English
11: Subtitle (PGS), English
12: Subtitle (PGS), German
13: Subtitle (PGS), Dutch
14: Subtitle (PGS), Danish
15: Subtitle (PGS), Estonian
16: Subtitle (PGS), Finnish
17: Subtitle (PGS), Modern Greek
18: Subtitle (PGS), Hindi
19: Subtitle (PGS), Lithuanian
20: Subtitle (PGS), Norwegian
21: Subtitle (PGS), Swedish
22: Subtitle (PGS), Turkish
23: Subtitle (PGS), English
24: Subtitle (PGS), German
25: Subtitle (PGS), Dutch
26: Subtitle (PGS), Russian
27: Subtitle (PGS), Ukrainian
28: Subtitle (PGS), English
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[a04] Extracting TrueHD stream...
[v02] Muxing video to Matroska...
[a04] Decoding with libav/ffmpeg...
[a04] Swapping endian...
[a04] Remapping channels...
[a04] Creating file "G:\xxxx\audio.pcm"...
[a04] The original audio track has a constant bit depth of 16 bits.
Added fps value to MKV header.
Video track 2 contains 147076 frames.
Video track 3 contains 147076 frames.
eac3to processing took 35 minutes, 53 seconds.
Done.

[madshi]

Which Blu-Ray is that exactly? I'd suggest doing another run with eac3to v2.80, just to be sure that it's not a new bug introduced by the automatic 2pass logic. Also please check audio sync at the beginning and end of the movie. Is sync slightly off at the end of the movie?

[NimaBeamer]

Madshi,

it's Hancock. I remuxed the PCM (after Pcm2Ts) and the MKV with TsMuxer but the thing will not even play in my Popcorn Hour. The MKV alone plays fine though. Will try 2.80. Shouldn't make a difference whether I use the GUI or not should it?

[madshi]

Originally Posted by NimaBeamer

it's Hancock.

The German one, I suppose? I have that Blu-Ray sitting on my shelf, still sealed, waiting to be remuxed. So I guess I'll move it up a bit in priority...

[NimaBeamer]

Jup that one.

No change with 2.80 :-(

a04] Extracting audio track number 4...
[a04] Extracting TrueHD stream...
[a04] Decoding with libav/ffmpeg...
[a04] Swapping endian...
[a04] Remapping channels...
[a04] Creating file "G:\Hancock\audio2.pcm"...
[a04] The original audio track has a constant bit depth of 16 bits.
Video track 2 contains 147076 frames.
Video track 3 contains 147076 frames.
eac3to processing took 28 minutes, 12 seconds.
Done.

[madshi]

Originally Posted by NimaBeamer

No change with 2.80 :-(

Then it's probably alright. How big the gaps are is a bit random. Probably they are just too small to register. But I'll double check with this Blu-Ray, just to be sure...

[NimaBeamer]

But why does it register gaps when extracting the AC3?

[a05] Extracting audio track number 5...
[a05] Removing AC3 dialog normalization...
[a05] Creating file "G:\Hancock\audioac3.ac3"...
[a05] Audio overlaps for 27ms at playtime 0:05:51.
[a05] Audio overlaps for 23ms at playtime 0:27:55.
[a05] Audio overlaps for 16ms at playtime 0:32:31.
[a05] Audio overlaps for 9ms at playtime 0:34:13.
[a05] Audio overlaps for 20ms at playtime 0:48:54.
[a05] Audio overlaps for 18ms at playtime 0:49:56.
[a05] Audio overlaps for 17ms at playtime 0:52:28.
[a05] Audio overlaps for 29ms at playtime 0:53:10.
[a05] Audio overlaps for 31ms at playtime 0:57:34.
[a05] Audio overlaps for 8ms at playtime 1:04:46.
[a05] Audio overlaps for 25ms at playtime 1:07:09.
[a05] Audio overlaps for 7ms at playtime 1:10:47.
[a05] Audio overlaps for 23ms at playtime 1:13:47.
[a05] Audio overlaps for 24ms at playtime 1:15:20.
[a05] Audio overlaps for 18ms at playtime 1:25:56.
[a05] Starting 2nd pass...
[a05] Realizing (E-)AC3 gaps...
[a05] Creating file "G:\Hancock\audioac3.ac3"...

[madshi]

Because AC3 has the biggest frames (in terms of audio samples per frame) of all codecs. Biggest overlaps are with AC3, then comes DTS, then TrueHD and least affected is PCM.

[nautilus7]

I have the same disc. v2.83 doesn't find any gaps for the embedded ac3 in the truehd track (#4). Isn't it weird?

[NimaBeamer]

For the AC3 I picked audio stream number 5 which said 6ch AC3 English. Still no luck with this.

[Jeff Flowerday]

I converted the TrueHD in Hancock to FLAC via eac3to and watched the movie and it was perfect. I don't remember the version I was running but it was before 2.8 for sure.

[madshi]

Originally Posted by nautilus7

I have the same disc. v2.83 doesn't find any gaps for the embedded ac3 in the the truehd track (#4). Isn't it weird?

It's a bit weird since AC3 normally does result in gaps when there are multiple parts. I'll look into this.

Originally Posted by NimaBeamer

Still no luck with this.

You seem to think that there MUST be a message about gaps/overlaps, or else there's something wrong. But that's not the case.

Originally Posted by Jeff Flowerday

I converted the TrueHD in Hancock to FLAC via eac3to and watched the movie and it was perfect.

Thanks for the confirmation.

[williewonton]

madshi

I've noted two titles that break eac3to both 2_81 and 2_82 as well as 2_83. Seems to relate to handling over audio gaps/overlaps. Eac3 eventually fails with "The temp file could not be interpreted correctly."

The original ISO's play fine in PDVD8 and TMT

Logs are attached (I hope)

[madshi]

@willie, attachments don't work well in this forum. Please post the logs here inside of a "[ code ] [ / code ]" block (without the spaces).

Which movies are these 2 and from which country?

[Boulder]

When converting mono AC3 tracks to MP4 using Nero, it seems that they are converted to stereo. Is there a way around this?

[DrNein]

When demuxing an AVC TS, the resolution changes from broadcast standard 1088i to 1084i. Why is that? Is there a way to leave it alone or else change to 1080?

[madshi]

Originally Posted by Boulder

When converting mono AC3 tracks to MP4 using Nero, it seems that they are converted to stereo. Is there a way around this?

It's most probably something happening inside of the Nero encoder. That means you need to ask the Nero guys for a fix, not me. Or if you want to make sure that the Nero encoder is at fault and not eac3to, ask eac3to to create a mono WAV file and then manually encode that to AAC with the Nero command line encoder. Is the result still stereo? Then Nero is at fault...

Originally Posted by DrNein

When demuxing an AVC TS, the resolution changes from broadcast standard 1088i to 1084i. Why is that? Is there a way to leave it alone or else change to 1080?

That's very weird. eac3to is trying to crop the resolution to 1080i. Don't know why you get 1084i. Must be a bug in eac3to. It works fine for German 1080i50 broadcasts, though. Can I please have a small sample of the TS file? 20MB should be enough. Thanks!

[tebasuna51]

Originally Posted by Boulder

When converting mono AC3 tracks to MP4 using Nero, it seems that they are converted to stereo. Is there a way around this?

Maybe the problem is mp4 decoders because in my test:

eac3to mono.ac3 mono.mp4

NeroAacDec mono.mp4 -> mono.wav
Foobar mono.mp4 -> mono.wav

but
ffdshow mono.mp4 -> stereo.wav
Faad mono.mp4 -> stereo.wav
Bass mono.mp4 -> stereo.wav

[Boulder]

Originally Posted by madshi

It's most probably something happening inside of the Nero encoder. That means you need to ask the Nero guys for a fix, not me. Or if you want to make sure that the Nero encoder is at fault and not eac3to, ask eac3to to create a mono WAV file and then manually encode that to AAC with the Nero command line encoder. Is the result still stereo? Then Nero is at fault...

I used to decode mono AC3s to mono WAV and then encode them manually in Nero encoder so it does work The Nero decoder outputs a mono file in eac3to properly, I just tried it.

By the way, this command line writes the output in the eac3to folder with the name "enarnia.wav":

eac3to "y:narnia.ac3" "e:narnia.wav"

If I use e:\temp\narnia.wav as the output, it works fine.

[Boulder]

Originally Posted by tebasuna51

Maybe the problem is mp4 decoders because in my test:

eac3to mono.ac3 mono.mp4

NeroAacDec mono.mp4 -> mono.wav
Foobar mono.mp4 -> mono.wav

but
ffdshow mono.mp4 -> stereo.wav
Faad mono.mp4 -> stereo.wav
Bass mono.mp4 -> stereo.wav

I opened the file in MediaInfo, it showed that it's a stereo file. Also the avg bitrate is pretty much the same as a regular stereo file has at q 0.4.

[tebasuna51]

Originally Posted by Boulder

I opened the file in MediaInfo, it showed that it's a stereo file. Also the avg bitrate is pretty much the same as a regular stereo file has at q 0.4.

Please put your complete log files to see if is a eac3to bad detection or a decoder problem, like here:

eac3to v2.83
command line: "D:\Test\AudioN\eac3to\eac3to.exe" "D:\Test\AudioD\Samples\ac3\1a100.ac3" mono.mp4
------------------------------------------------------------------------------
AC3, 1.0 channels, 0:00:20, 96kbps, 48khz
The Nero decoder doesn't seem to work, will use libav instead.
Decoding with libav/ffmpeg...
Reducing depth from 64 to 32 bits...
Encoding AAC <0.50> with NeroAacEnc...
eac3to processing took 1 second.
Done.

This mono.mp4 is detected by MediaInfo like:

...
Codec                            : AAC LC
Bit rate mode                    : VBR
Bit rate                         : 16.0 Kbps
Maximum bit rate                 : 103 Kbps
Channel(s)                       : 1 channel
Channel positions                : C
Sampling rate                    : 48.0 KHz
...

[Boulder]

Strange..now I'm unable to reproduce the problem - the final mp4 file is in mono I must have chosen the incorrect file in MediaInfo because of the bug I posted (the path in the eac3to command line is parsed a bit wrong). I usually cd to working directories in MS-DOS prompt and then use the drive letters only for paths.

Sorry

[mrr19121970]

I have a disk that -demux es OK with V2.79 and not V2.83

[williewonton]

madshi

@willie, attachments don't work well in this forum. Please post the logs here inside of a "[ code ] [ / code ]" block (without the spaces).

Which movies are these 2 and from which country?

OK thanks for the tip

Game Plan - Region B
Open Season - Not Locked

[mrr19121970]

I have a disk that -demux es OK with V2.79 and not V2.83

eac3to v2.79
command line: E:\TVIX\eac3toV279\eac3to.exe  "R:\" 1) -demux
------------------------------------------------------------------------------
M2TS, 1 video track, 8 audio tracks, 25 subtitle tracks, 1:56:52
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz
   (embedded: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB)
4: TrueHD/AC3, German, 5.1 channels, 48khz
   (embedded: AC3, 5.1 channels, 448kbps, 48khz)
5: AC3, Spanish, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
6: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -29dB
7: AC3 Surround, English, 2.0 channels, 192kbps, 48khz
8: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -29dB
9: AC3 Surround, English, 2.0 channels, 192kbps, 48khz
10: AC3, Portuguese, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
11: Subtitle (PGS), English
12: Subtitle (PGS), English
13: Subtitle (PGS), German
14: Subtitle (PGS), Spanish
15: Subtitle (PGS), Portuguese
16: Subtitle (PGS), German
17: Subtitle (PGS), Spanish
18: Subtitle (PGS), Portuguese
19: Subtitle (PGS), German
20: Subtitle (PGS), Spanish
21: Subtitle (PGS), Portuguese
22: Subtitle (PGS), German
23: Subtitle (PGS), Spanish
24: Subtitle (PGS), Portuguese
25: Subtitle (PGS), German
26: Subtitle (PGS), Spanish
27: Subtitle (PGS), Portuguese
28: Subtitle (PGS), Turkish
29: Subtitle (PGS), French
30: Subtitle (PGS), Dutch
31: Subtitle (PGS), Korean
32: Subtitle (PGS), English
33: Subtitle (PGS), French
34: Subtitle (PGS), Dutch
35: Subtitle (PGS), Korean
Creating file "00105 - Chapters.txt"...
Audio gap description file detected, can't be used for TrueHD/MLP, though.
Audio gap description file detected, will be used for processing...
Audio gap description file detected, can't be used for TrueHD/MLP, though.
Audio gap description file detected, will be used for processing...
Audio gap description file detected, will be used for processing...
Audio gap description file detected, will be used for processing...
Audio gap description file detected, will be used for processing...
Audio gap description file detected, will be used for processing...
Audio gap description file detected, will be used for processing...
Audio gap description file detected, will be used for processing...
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a03] Extracting audio track number 3...
[a04] Extracting audio track number 4...
[a10] Extracting audio track number 10...
[a04] Extracting audio track number 4...
[a06] Extracting audio track number 6...
[a05] Extracting audio track number 5...
[a07] Extracting audio track number 7...
[a08] Extracting audio track number 8...
[a09] Extracting audio track number 9...
[s11] Extracting subtitle track number 11...
[s12] Extracting subtitle track number 12...
[s13] Extracting subtitle track number 13...
[s14] Extracting subtitle track number 14...
[s19] Extracting subtitle track number 19...
[s15] Extracting subtitle track number 15...
[s18] Extracting subtitle track number 18...
[s17] Extracting subtitle track number 17...
[s32] Extracting subtitle track number 32...
[s21] Extracting subtitle track number 21...
[s20] Extracting subtitle track number 20...
[s31] Extracting subtitle track number 31...
[s27] Extracting subtitle track number 27...
[s24] Extracting subtitle track number 24...
[s22] Extracting subtitle track number 22...
[s16] Extracting subtitle track number 16...
[s23] Extracting subtitle track number 23...
[s25] Extracting subtitle track number 25...
[s26] Extracting subtitle track number 26...
[s33] Extracting subtitle track number 33...
[s30] Extracting subtitle track number 30...
[s28] Extracting subtitle track number 28...
[s29] Extracting subtitle track number 29...
[s34] Extracting subtitle track number 34...
[s35] Extracting subtitle track number 35...
[a03] Extracting TrueHD stream...
[a04] Extracting TrueHD stream...
[a03] Extracting TrueHD stream...
[a04] Extracting TrueHD stream...
[a03] Decoding with libav/ffmpeg...
[a04] Decoding with libav/ffmpeg...
[a03] Encoding AC3 <640kbps> with libAften...
[a04] Encoding AC3 <640kbps> with libAften...
[a05] Removing AC3 dialog normalization...
[a10] Removing AC3 dialog normalization...
[a05] Realizing (E-)AC3 gaps...
[a10] Realizing (E-)AC3 gaps...
[a08] Removing AC3 dialog normalization...
[a07] Realizing (E-)AC3 gaps...
[a09] Realizing (E-)AC3 gaps...
[a06] Removing AC3 dialog normalization...
[a08] Realizing (E-)AC3 gaps...
[a06] Realizing (E-)AC3 gaps...
[v02] Creating file "00105 - 2 - h264, 1080p24.h264"...
[a03] Creating file "00105 - 3 - TrueHD+AC3, English, 5.1 channels, 48khz.thd+ac3"...
[a04] Creating file "00105 - 4 - TrueHD+AC3, German, 5.1 channels, 48khz.thd+ac3"...
[a05] Creating file "00105 - 5 - AC3, Spanish, 5.1 channels, 448kbps, 48khz.ac3"...
[a10] Creating file "00105 - 10 - AC3, Portuguese, 5.1 channels, 448kbps, 48khz.ac3"...
[a07] Creating file "00105 - 7 - AC3 Surround, English, 2.0 channels, 192kbps, 48khz.ac3"...
[a06] Creating file "00105 - 6 - AC3 Surround, English, 2.0 channels, 192kbps, 48khz.ac3"...
[a08] Creating file "00105 - 8 - AC3 Surround, English, 2.0 channels, 192kbps, 48khz.ac3"...
[a09] Creating file "00105 - 9 - AC3 Surround, English, 2.0 channels, 192kbps, 48khz.ac3"...
[s25] Creating file "00105 - 25 - Subtitle (PGS), German.sup"...
[s19] Creating file "00105 - 19 - Subtitle (PGS), German.sup"...
[libav] Lossless check failed - expected d1, calculated 34
[s21] Creating file "00105 - 21 - Subtitle (PGS), Portuguese.sup"...
[s27] Creating file "00105 - 27 - Subtitle (PGS), Portuguese.sup"...
[s26] Creating file "00105 - 26 - Subtitle (PGS), Spanish.sup"...
[s20] Creating file "00105 - 20 - Subtitle (PGS), Spanish.sup"...
[s16] Creating file "00105 - 16 - Subtitle (PGS), German.sup"...
[s22] Creating file "00105 - 22 - Subtitle (PGS), German.sup"...
[s24] Creating file "00105 - 24 - Subtitle (PGS), Portuguese.sup"...
[s18] Creating file "00105 - 18 - Subtitle (PGS), Portuguese.sup"...
[s17] Creating file "00105 - 17 - Subtitle (PGS), Spanish.sup"...
[s23] Creating file "00105 - 23 - Subtitle (PGS), Spanish.sup"...
[s14] Creating file "00105 - 14 - Subtitle (PGS), Spanish.sup"...
[s13] Creating file "00105 - 13 - Subtitle (PGS), German.sup"...
[s12] Creating file "00105 - 12 - Subtitle (PGS), English.sup"...
[s11] Creating file "00105 - 11 - Subtitle (PGS), English.sup"...
[s28] Creating file "00105 - 28 - Subtitle (PGS), Turkish.sup"...
[s15] Creating file "00105 - 15 - Subtitle (PGS), Portuguese.sup"...
[libav] Lossless check failed - expected ab, calculated 50
[libav] Lossless check failed - expected 9f, calculated 1b
[libav] Lossless check failed - expected 97, calculated 52
[libav] Lossless check failed - expected 88, calculated 1d
[libav] Lossless check failed - expected b1, calculated 95
[libav] Lossless check failed - expected ed, calculated a0
[libav] Lossless check failed - expected 4, calculated b0
[libav] Lossless check failed - expected 4a, calculated 71
[libav] Lossless check failed - expected 24, calculated 8a
[libav] Lossless check failed - expected 27, calculated 94
[libav] Lossless check failed - expected 15, calculated a2
[libav] Lossless check failed - expected 86, calculated 3b
[libav] Lossless check failed - expected 93, calculated 80
[libav] Lossless check failed - expected 3e, calculated 49
[libav] Lossless check failed - expected fe, calculated 80
[libav] Lossless check failed - expected 67, calculated c0
[libav] Lossless check failed - expected f6, calculated 6c
[libav] Lossless check failed - expected fc, calculated df
[libav] Lossless check failed - expected d4, calculated 8c
[libav] Lossless check failed - expected 38, calculated 18
[a03] Audio overlaps for 5ms at playtime 0:31:59.
[a03] Audio overlaps for 5ms at playtime 1:23:12.
[a03] The audio gaps/overlaps technically can't be removed from the TrueHD bitstream.
[a03] In order to remove them you'll have to transcode the audio to another format.
[a03] The original audio track has a constant bit depth of 16 bits.
[a04] Audio overlaps for 5ms at playtime 0:31:59.
[a04] Audio overlaps for 5ms at playtime 1:23:12.
[a04] The audio gaps/overlaps technically can't be removed from the TrueHD bitstream.
[a04] In order to remove them you'll have to transcode the audio to another format.
[a04] The original audio track has a constant bit depth of 16 bits.
[s29] Creating file "00105 - 29 - Subtitle (PGS), French.sup"...
[s30] Creating file "00105 - 30 - Subtitle (PGS), Dutch.sup"...
[s31] Creating file "00105 - 31 - Subtitle (PGS), Korean.sup"...
[s32] Creating file "00105 - 32 - Subtitle (PGS), English.sup"...
[s33] Creating file "00105 - 33 - Subtitle (PGS), French.sup"...
[s34] Creating file "00105 - 34 - Subtitle (PGS), Dutch.sup"...
[s35] Creating file "00105 - 35 - Subtitle (PGS), Korean.sup"...
Video track 2 contains 168155 frames.
eac3to processing took 29 minutes, 5 seconds.
Done.

[mrr19121970]

eac3to v2.83
command line: E:\TVIX\eac3to\eac3to.exe  "R:\" 1) -demux
------------------------------------------------------------------------------
M2TS, 1 video track, 8 audio tracks, 25 subtitle tracks, 1:56:52
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz
   (embedded: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB)
4: TrueHD/AC3, German, 5.1 channels, 48khz
   (embedded: AC3, 5.1 channels, 448kbps, 48khz)
5: AC3, Spanish, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
6: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -29dB
7: AC3 Surround, English, 2.0 channels, 192kbps, 48khz
8: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -29dB
9: AC3 Surround, English, 2.0 channels, 192kbps, 48khz
10: AC3, Portuguese, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
11: Subtitle (PGS), English
12: Subtitle (PGS), English
13: Subtitle (PGS), German
14: Subtitle (PGS), Spanish
15: Subtitle (PGS), Portuguese
16: Subtitle (PGS), German
17: Subtitle (PGS), Spanish
18: Subtitle (PGS), Portuguese
19: Subtitle (PGS), German
20: Subtitle (PGS), Spanish
21: Subtitle (PGS), Portuguese
22: Subtitle (PGS), German
23: Subtitle (PGS), Spanish
24: Subtitle (PGS), Portuguese
25: Subtitle (PGS), German
26: Subtitle (PGS), Spanish
27: Subtitle (PGS), Portuguese
28: Subtitle (PGS), Turkish
29: Subtitle (PGS), French
30: Subtitle (PGS), Dutch
31: Subtitle (PGS), Korean
32: Subtitle (PGS), English
33: Subtitle (PGS), French
34: Subtitle (PGS), Dutch
35: Subtitle (PGS), Korean
Creating file "00105 - Chapters.txt"...
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a05] Extracting audio track number 5...
[a07] Extracting audio track number 7...
[a03] Extracting audio track number 3...
[a09] Extracting audio track number 9...
[s19] Extracting subtitle track number 19...
[a06] Extracting audio track number 6...
[a04] Extracting audio track number 4...
[a04] Extracting audio track number 4...
[s11] Extracting subtitle track number 11...
[s23] Extracting subtitle track number 23...
[s20] Extracting subtitle track number 20...
[s13] Extracting subtitle track number 13...
[a08] Extracting audio track number 8...
[s31] Extracting subtitle track number 31...
[s27] Extracting subtitle track number 27...
[a10] Extracting audio track number 10...
[s26] Extracting subtitle track number 26...
[s25] Extracting subtitle track number 25...
[s21] Extracting subtitle track number 21...
[s32] Extracting subtitle track number 32...
[s17] Extracting subtitle track number 17...
[s16] Extracting subtitle track number 16...
[s12] Extracting subtitle track number 12...
[s28] Extracting subtitle track number 28...
[s22] Extracting subtitle track number 22...
[s33] Extracting subtitle track number 33...
[s29] Extracting subtitle track number 29...
[s14] Extracting subtitle track number 14...
[s24] Extracting subtitle track number 24...
[s15] Extracting subtitle track number 15...
[s35] Extracting subtitle track number 35...
[s18] Extracting subtitle track number 18...
[s30] Extracting subtitle track number 30...
[s34] Extracting subtitle track number 34...
[a03] Extracting TrueHD stream...
[a03] Extracting TrueHD stream...
[a04] Extracting TrueHD stream...
[a04] Extracting TrueHD stream...
[a03] Decoding with libav/ffmpeg...
[a04] Decoding with libav/ffmpeg...
[a03] Encoding AC3 <640kbps> with libAften...
[a04] Encoding AC3 <640kbps> with libAften...
[a05] Removing AC3 dialog normalization...
[a10] Removing AC3 dialog normalization...
[a06] Removing AC3 dialog normalization...
[a08] Removing AC3 dialog normalization...
[v02] Creating file "00105 - 2 - h264, 1080p24.h264"...
[a03] Creating file "00105 - 3 - TrueHD+AC3, English, 5.1 channels, 48khz.thd+ac3"...
[a04] Creating file "00105 - 4 - TrueHD+AC3, German, 5.1 channels, 48khz.thd+ac3"...
[a05] Creating file "00105 - 5 - AC3, Spanish, 5.1 channels, 448kbps, 48khz.ac3"...
[a10] Creating file "00105 - 10 - AC3, Portuguese, 5.1 channels, 448kbps, 48khz.ac3"...
[a08] Creating file "00105 - 8 - AC3 Surround, English, 2.0 channels, 192kbps, 48khz.ac3"...
[a06] Creating file "00105 - 6 - AC3 Surround, English, 2.0 channels, 192kbps, 48khz.ac3"...
[a09] Creating file "00105 - 9 - AC3 Surround, English, 2.0 channels, 192kbps, 48khz.ac3"...
[a07] Creating file "00105 - 7 - AC3 Surround, English, 2.0 channels, 192kbps, 48khz.ac3"...
[s25] Creating file "00105 - 25 - Subtitle (PGS), German.sup"...
[s19] Creating file "00105 - 19 - Subtitle (PGS), German.sup"...
[libav] Lossless check failed - expected 8f, calculated 73
[s21] Creating file "00105 - 21 - Subtitle (PGS), Portuguese.sup"...
[s27] Creating file "00105 - 27 - Subtitle (PGS), Portuguese.sup"...
[s26] Creating file "00105 - 26 - Subtitle (PGS), Spanish.sup"...
[s20] Creating file "00105 - 20 - Subtitle (PGS), Spanish.sup"...
[s16] Creating file "00105 - 16 - Subtitle (PGS), German.sup"...
[s22] Creating file "00105 - 22 - Subtitle (PGS), German.sup"...
[s24] Creating file "00105 - 24 - Subtitle (PGS), Portuguese.sup"...
[s18] Creating file "00105 - 18 - Subtitle (PGS), Portuguese.sup"...
[s17] Creating file "00105 - 17 - Subtitle (PGS), Spanish.sup"...
[s23] Creating file "00105 - 23 - Subtitle (PGS), Spanish.sup"...
[s14] Creating file "00105 - 14 - Subtitle (PGS), Spanish.sup"...
[s13] Creating file "00105 - 13 - Subtitle (PGS), German.sup"...
[s12] Creating file "00105 - 12 - Subtitle (PGS), English.sup"...
[s11] Creating file "00105 - 11 - Subtitle (PGS), English.sup"...
[s28] Creating file "00105 - 28 - Subtitle (PGS), Turkish.sup"...
[s15] Creating file "00105 - 15 - Subtitle (PGS), Portuguese.sup"...

[mrr19121970]

V2.83 log continued

[libav] Lossless check failed - expected f7, calculated a1
[libav] Lossless check failed - expected 9f, calculated 1b
[libav] Lossless check failed - expected ce, calculated 7a
[libav] Lossless check failed - expected a, calculated cb
[libav] Lossless check failed - expected b1, calculated 95
[libav] Lossless check failed - expected ed, calculated a0
[libav] Lossless check failed - expected 4, calculated b0
[libav] Lossless check failed - expected 4a, calculated 71
[libav] Lossless check failed - expected ab, calculated 99
[libav] Lossless check failed - expected 27, calculated 94
[libav] Lossless check failed - expected dc, calculated 26
[libav] Lossless check failed - expected 86, calculated 3b
[libav] Lossless check failed - expected 93, calculated 80
[libav] Lossless check failed - expected ba, calculated 76
[libav] Lossless check failed - expected fe, calculated 80
[libav] Lossless check failed - expected 67, calculated c0
[libav] Lossless check failed - expected 10, calculated 72
[libav] Lossless check failed - expected aa, calculated b4
[libav] Lossless check failed - expected d4, calculated 8c
[libav] Lossless check failed - expected 45, calculated a2
[libav] Lossless check failed - expected 38, calculated 18
[a03] Audio overlaps for 5ms at playtime 0:31:59.
[a03] Audio overlaps for 5ms at playtime 1:23:12.
[a03] The audio gaps/overlaps technically can't be removed from the TrueHD bitstream.
[a03] In order to remove them you'll have to transcode the audio to another format.
[a03] The original audio track has a constant bit depth of 16 bits.
[a03] Audio overlaps for 5ms at playtime 0:31:59.
[a03] Audio overlaps for 5ms at playtime 1:23:12.
[a04] Audio overlaps for 5ms at playtime 0:31:59.
[a04] Audio overlaps for 5ms at playtime 1:23:12.
[a04] The audio gaps/overlaps technically can't be removed from the TrueHD bitstream.
[a04] In order to remove them you'll have to transcode the audio to another format.
[a04] The original audio track has a constant bit depth of 16 bits.
[a04] Audio overlaps for 5ms at playtime 0:31:59.
[a04] Audio overlaps for 5ms at playtime 1:23:12.
[a05] Audio overlaps for 26ms at playtime 0:02:23.
[a05] Audio overlaps for 24ms at playtime 0:19:05.
[a05] Audio overlaps for 14ms at playtime 0:20:38.
[a05] Audio overlaps for 19ms at playtime 0:21:42.
[a05] Audio overlaps for 25ms at playtime 0:22:50.
[a05] Audio overlaps for 13ms at playtime 0:26:29.
[a05] Audio overlaps for 8ms at playtime 0:29:09.
[a05] Audio overlaps for 8ms at playtime 0:30:16.
[a05] Audio overlaps for 6ms at playtime 0:31:58.
[a05] Audio overlaps for 5ms at playtime 0:55:26.
[a05] Audio overlaps for 29ms at playtime 0:56:58.
[a05] Audio overlaps for 11ms at playtime 1:04:33.
[a05] Audio overlaps for 13ms at playtime 1:05:10.
[a05] Audio overlaps for 16ms at playtime 1:07:23.
[a05] Audio overlaps for 15ms at playtime 1:08:12.
[a05] Audio overlaps for 13ms at playtime 1:18:20.
[a05] Audio overlaps for 18ms at playtime 1:21:53.
[a05] Audio overlaps for 16ms at playtime 1:23:12.
[a05] Audio overlaps for 30ms at playtime 1:31:13.
[a05] Audio overlaps for 11ms at playtime 1:32:08.
[a05] Audio overlaps for 17ms at playtime 1:33:01.
[a05] Audio overlaps for 7ms at playtime 1:33:42.
[a05] Audio overlaps for 25ms at playtime 1:38:23.
[a05] Audio overlaps for 10ms at playtime 1:43:40.
[a05] Audio overlaps for 17ms at playtime 1:44:25.
[a06] Audio overlaps for 26ms at playtime 0:02:23.
[a06] Audio overlaps for 24ms at playtime 0:19:05.
[a06] Audio overlaps for 14ms at playtime 0:20:38.
[a06] Audio overlaps for 19ms at playtime 0:21:42.
[a06] Audio overlaps for 25ms at playtime 0:22:50.
[a06] Audio overlaps for 13ms at playtime 0:26:29.
[a06] Audio overlaps for 8ms at playtime 0:29:09.
[a06] Audio overlaps for 8ms at playtime 0:30:16.
[a06] Audio overlaps for 6ms at playtime 0:31:58.
[a06] Audio overlaps for 5ms at playtime 0:55:26.
[a06] Audio overlaps for 29ms at playtime 0:56:58.
[a06] Audio overlaps for 11ms at playtime 1:04:33.
[a06] Audio overlaps for 13ms at playtime 1:05:10.
[a06] Audio overlaps for 16ms at playtime 1:07:23.
[a06] Audio overlaps for 15ms at playtime 1:08:12.
[a06] Audio overlaps for 13ms at playtime 1:18:20.
[a06] Audio overlaps for 18ms at playtime 1:21:53.
[a06] Audio overlaps for 16ms at playtime 1:23:12.
[a06] Audio overlaps for 30ms at playtime 1:31:13.
[a06] Audio overlaps for 11ms at playtime 1:32:08.
[a06] Audio overlaps for 17ms at playtime 1:33:01.
[a06] Audio overlaps for 7ms at playtime 1:33:42.
[a06] Audio overlaps for 25ms at playtime 1:38:23.
[a06] Audio overlaps for 10ms at playtime 1:43:40.
[a06] Audio overlaps for 17ms at playtime 1:44:25.
[a07] Audio overlaps for 26ms at playtime 0:02:23.
[a07] Audio overlaps for 24ms at playtime 0:19:05.
[a07] Audio overlaps for 14ms at playtime 0:20:38.
[a07] Audio overlaps for 19ms at playtime 0:21:42.
[a07] Audio overlaps for 25ms at playtime 0:22:50.
[a07] Audio overlaps for 13ms at playtime 0:26:29.
[a07] Audio overlaps for 8ms at playtime 0:29:09.
[a07] Audio overlaps for 8ms at playtime 0:30:16.
[a07] Audio overlaps for 6ms at playtime 0:31:58.
[a07] Audio overlaps for 5ms at playtime 0:55:26.
[a07] Audio overlaps for 29ms at playtime 0:56:58.
[a07] Audio overlaps for 11ms at playtime 1:04:33.
[a07] Audio overlaps for 13ms at playtime 1:05:10.
[a07] Audio overlaps for 16ms at playtime 1:07:23.
[a07] Audio overlaps for 15ms at playtime 1:08:12.
[a07] Audio overlaps for 13ms at playtime 1:18:20.
[a07] Audio overlaps for 18ms at playtime 1:21:53.
[a07] Audio overlaps for 16ms at playtime 1:23:12.
[a07] Audio overlaps for 30ms at playtime 1:31:13.
[a07] Audio overlaps for 11ms at playtime 1:32:08.
[a07] Audio overlaps for 17ms at playtime 1:33:01.
[a07] Audio overlaps for 7ms at playtime 1:33:42.
[a07] Audio overlaps for 25ms at playtime 1:38:23.
[a07] Audio overlaps for 10ms at playtime 1:43:40.
[a07] Audio overlaps for 17ms at playtime 1:44:25.
[a08] Audio overlaps for 26ms at playtime 0:02:23.
[a08] Audio overlaps for 24ms at playtime 0:19:05.
[a08] Audio overlaps for 14ms at playtime 0:20:38.
[a08] Audio overlaps for 19ms at playtime 0:21:42.
[a08] Audio overlaps for 25ms at playtime 0:22:50.
[a08] Audio overlaps for 13ms at playtime 0:26:29.
[a08] Audio overlaps for 8ms at playtime 0:29:09.
[a08] Audio overlaps for 8ms at playtime 0:30:16.
[a08] Audio overlaps for 6ms at playtime 0:31:58.
[a08] Audio overlaps for 5ms at playtime 0:55:26.
[a08] Audio overlaps for 29ms at playtime 0:56:58.
[a08] Audio overlaps for 11ms at playtime 1:04:33.
[a08] Audio overlaps for 13ms at playtime 1:05:10.
[a08] Audio overlaps for 16ms at playtime 1:07:23.
[a08] Audio overlaps for 15ms at playtime 1:08:12.
[a08] Audio overlaps for 13ms at playtime 1:18:20.
[a08] Audio overlaps for 18ms at playtime 1:21:53.
[a08] Audio overlaps for 16ms at playtime 1:23:12.
[a08] Audio overlaps for 30ms at playtime 1:31:13.
[a08] Audio overlaps for 11ms at playtime 1:32:08.
[a08] Audio overlaps for 17ms at playtime 1:33:01.
[a08] Audio overlaps for 7ms at playtime 1:33:42.
[a08] Audio overlaps for 25ms at playtime 1:38:23.
[a08] Audio overlaps for 10ms at playtime 1:43:40.
[a08] Audio overlaps for 17ms at playtime 1:44:25.
[a09] Audio overlaps for 26ms at playtime 0:02:23.
[a09] Audio overlaps for 24ms at playtime 0:19:05.
[a09] Audio overlaps for 14ms at playtime 0:20:38.
[a09] Audio overlaps for 19ms at playtime 0:21:42.
[a09] Audio overlaps for 25ms at playtime 0:22:50.
[a09] Audio overlaps for 13ms at playtime 0:26:29.
[a09] Audio overlaps for 8ms at playtime 0:29:09.
[a09] Audio overlaps for 8ms at playtime 0:30:16.
[a09] Audio overlaps for 6ms at playtime 0:31:58.
[a09] Audio overlaps for 5ms at playtime 0:55:26.
[a09] Audio overlaps for 29ms at playtime 0:56:58.
[a09] Audio overlaps for 11ms at playtime 1:04:33.
[a09] Audio overlaps for 13ms at playtime 1:05:10.
[a09] Audio overlaps for 16ms at playtime 1:07:23.
[a09] Audio overlaps for 15ms at playtime 1:08:12.
[a09] Audio overlaps for 13ms at playtime 1:18:20.
[a09] Audio overlaps for 18ms at playtime 1:21:53.
[a09] Audio overlaps for 16ms at playtime 1:23:12.
[a09] Audio overlaps for 30ms at playtime 1:31:13.
[a09] Audio overlaps for 11ms at playtime 1:32:08.
[a09] Audio overlaps for 17ms at playtime 1:33:01.
[a09] Audio overlaps for 7ms at playtime 1:33:42.
[a09] Audio overlaps for 25ms at playtime 1:38:23.
[a09] Audio overlaps for 10ms at playtime 1:43:40.
[a09] Audio overlaps for 17ms at playtime 1:44:25.
[a10] Audio overlaps for 26ms at playtime 0:02:23.
[a10] Audio overlaps for 24ms at playtime 0:19:05.
[a10] Audio overlaps for 14ms at playtime 0:20:38.
[a10] Audio overlaps for 19ms at playtime 0:21:42.
[a10] Audio overlaps for 25ms at playtime 0:22:50.
[a10] Audio overlaps for 13ms at playtime 0:26:29.
[a10] Audio overlaps for 8ms at playtime 0:29:09.
[a10] Audio overlaps for 8ms at playtime 0:30:16.
[a10] Audio overlaps for 6ms at playtime 0:31:58.
[a10] Audio overlaps for 5ms at playtime 0:55:26.
[a10] Audio overlaps for 29ms at playtime 0:56:58.
[a10] Audio overlaps for 11ms at playtime 1:04:33.
[a10] Audio overlaps for 13ms at playtime 1:05:10.
[a10] Audio overlaps for 16ms at playtime 1:07:23.
[a10] Audio overlaps for 15ms at playtime 1:08:12.
[a10] Audio overlaps for 13ms at playtime 1:18:20.
[a10] Audio overlaps for 18ms at playtime 1:21:53.
[a10] Audio overlaps for 16ms at playtime 1:23:12.
[a10] Audio overlaps for 30ms at playtime 1:31:13.
[a10] Audio overlaps for 11ms at playtime 1:32:08.
[a10] Audio overlaps for 17ms at playtime 1:33:01.
[a10] Audio overlaps for 7ms at playtime 1:33:42.
[a10] Audio overlaps for 25ms at playtime 1:38:23.
[a10] Audio overlaps for 10ms at playtime 1:43:40.
[a10] Audio overlaps for 17ms at playtime 1:44:25.
[s29] Creating file "00105 - 29 - Subtitle (PGS), French.sup"...
[s30] Creating file "00105 - 30 - Subtitle (PGS), Dutch.sup"...
[s31] Creating file "00105 - 31 - Subtitle (PGS), Korean.sup"...
[s32] Creating file "00105 - 32 - Subtitle (PGS), English.sup"...
[s33] Creating file "00105 - 33 - Subtitle (PGS), French.sup"...
[s34] Creating file "00105 - 34 - Subtitle (PGS), Dutch.sup"...
[s35] Creating file "00105 - 35 - Subtitle (PGS), Korean.sup"...
[a03] Starting 2nd pass...
[a03] Extracting TrueHD stream...
[a03] The thd ac3 joiner doesn't have the correct connections.
Aborted at file position 16384.

[menlvd]

1. processing

eac3to v2.83
command line: "D:\minep\me_gui\tools\eac3_to\eac3to.exe" "G:\bd-dr\BDMV\STREAM\00286.m2ts"+"G:\bd-dr\BDMV\STREAM\00309.m2ts"+"G:\bd-dr\BDMV\STREAM\00287.m2ts"+"G:\bd-dr\BDMV\STREAM\00310.m2ts"+"G:\bd-dr\BDMV\STREAM\00288.m2ts"+"G:\bd-dr\BDMV\STREAM\00311.m2ts"+"G:\bd-dr\BDMV\STREAM\00289.m2ts"+"G:\bd-dr\BDMV\STREAM\00312.m2ts"+"G:\bd-dr\BDMV\STREAM\00290.m2ts"+"G:\bd-dr\BDMV\STREAM\00313.m2ts"+"G:\bd-dr\BDMV\STREAM\00291.m2ts"+"G:\bd-dr\BDMV\STREAM\00314.m2ts"+"G:\bd-dr\BDMV\STREAM\00292.m2ts"+"G:\bd-dr\BDMV\STREAM\00315.m2ts"+"G:\bd-dr\BDMV\STREAM\00293.m2ts"+"G:\bd-dr\BDMV\STREAM\00316.m2ts"+"G:\bd-dr\BDMV\STREAM\00294.m2ts"+"G:\bd-dr\BDMV\STREAM\00317.m2ts"+"G:\bd-dr\BDMV\STREAM\00295.m2ts"+"G:\bd-dr\BDMV\STREAM\00318.m2ts"+"G:\bd-dr\BDMV\STREAM\00296.m2ts"+"G:\bd-dr\BDMV\STREAM\00319.m2ts"+"G:\bd-dr\BDMV\STREAM\00297.m2ts"+"G:\bd-dr\BDMV\STREAM\00320.m2ts"+"G:\bd-dr\BDMV\STREAM\00298.m2ts" 4: "I:\Death_eng.wavs"
------------------------------------------------------------------------------
M2TS, 2 video tracks, 5 audio tracks, 5 subtitle tracks, 1:50:35
1: Chapters, 21 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: DTS, Spanish, 5.1 channels, 24 bits, 768kbps, 48khz
6: DTS, French, 5.1 channels, 24 bits, 768kbps, 48khz
7: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
8: DTS Express, English, 2.0 channels, 24 bits, 192kbps, 48khz
9: Subtitle (PGS), English
10: Subtitle (PGS), Spanish
11: Subtitle (PGS), French
12: Subtitle (PGS), Spanish
13: Subtitle (PGS), French
[a04] Extracting audio track number 4...
[a04] Decoding with ArcSoft DTS Decoder...
[a04] Writing WAVs...
[a04] Creating file "I:\Death_eng.L.wav"...
[a04] Creating file "I:\Death_eng.LFE.wav"...
[a04] Creating file "I:\Death_eng.SR.wav"...
[a04] Creating file "I:\Death_eng.C.wav"...
[a04] Creating file "I:\Death_eng.R.wav"...
[a04] Creating file "I:\Death_eng.SL.wav"...
[a04] The last DTS frame is incomplete and thus gets skipped.
[a04] The original audio track has a constant bit depth of 24 bits.
[a04] Audio overlaps for 8ms at playtime 0:08:45.
[a04] Audio overlaps for 10ms at playtime 0:13:47.
[a04] Audio overlaps for 10ms at playtime 0:16:07.
[a04] Audio overlaps for 10ms at playtime 0:19:55.
[a04] Audio overlaps for 12ms at playtime 0:24:00.
[a04] Audio overlaps for 8ms at playtime 0:28:21.
[a04] Audio overlaps for 9ms at playtime 0:29:08.
[a04] Audio overlaps for 6ms at playtime 0:30:49.
[a04] Audio overlaps for 6ms at playtime 0:44:22.
[a04] Audio overlaps for 12ms at playtime 0:49:05.
[a04] Audio overlaps for 8ms at playtime 0:53:40.
[a04] Audio overlaps for 11ms at playtime 1:02:06.
[a04] Audio overlaps for 9ms at playtime 1:03:24.
[a04] Audio overlaps for 6ms at playtime 1:35:09.
[a04] Audio overlaps for 8ms at playtime 1:35:58.
[a04] Audio overlaps for 11ms at playtime 1:44:33.
[a04] Starting 2nd pass...
[a04] Extracting audio track number 4...
[a04] Decoding with ArcSoft DTS Decoder...
[a04] Writing WAVs...
[a04] Realizing RAW/PCM gaps...
[a04] Creating file "I:\Death_eng.L.wav"...
[a04] Creating file "I:\Death_eng.SL.wav"...
[a04] Creating file "I:\Death_eng.R.wav"...
[a04] Creating file "I:\Death_eng.LFE.wav"...
[a04] Creating file "I:\Death_eng.C.wav"...
[a04] Creating file "I:\Death_eng.SR.wav"...
[a04] The last DTS frame is incomplete and thus gets skipped.
[a04] The original audio track has a constant bit depth of 24 bits.
[a04] The processed audio track has a constant bit depth of 24 bits.
eac3to processing took 35 minutes, 49 seconds.
Done.

got fail sync

2. extracting first dtshd and then decoding to wavs

eac3to v2.83
command line: "D:\minep\me_gui\tools\eac3_to\eac3to.exe" "i:\Death_eng.dtshd" "i:\Death_eng1.wavs"
------------------------------------------------------------------------------
DTS Master Audio, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
Decoding with ArcSoft DTS Decoder...
Writing WAVs...
Creating file "i:\Death_eng1.LFE.wav"...
Creating file "i:\Death_eng1.SR.wav"...
Creating file "i:\Death_eng1.C.wav"...
Creating file "i:\Death_eng1.SL.wav"...
Creating file "i:\Death_eng1.R.wav"...
Creating file "i:\Death_eng1.L.wav"...
The original audio track has a constant bit depth of 24 bits.
eac3to processing took 13 minutes, 58 seconds.
Done.

sync OK!!!

[madshi]

@menlvd, which Blu-Ray is that? Movie name and release country, please. Guys, please always state which Blu-Ray you have problems with. Makes my life much easier...

[menlvd]

@madshi, Death Race 2008 Unrated BD

[cavediver]

I have "Cream Royal Albert Hall" HD DVD that has a DTS-HD MA audio track. I want to extract out the DTS-HD MA track intact. Is this possible with eac3to? If so, do I need any additional software besides eac3to? If not, what additional software if any do I need to convert it to pcm?

[rica]

Originally Posted by cavediver

I have "Cream Royal Albert Hall" HD DVD that has a DTS-HD MA audio track. I want to extract out the DTS-HD MA track intact. Is this possible with eac3to? If so, do I need any additional software besides eac3to? If not, what additional software if any do I need to convert it to pcm?

1) Yes: eac3to inputfile outputfile (*.dtshd/dts) -both will give the same result except you use -core option. "-core" extracts the core not DTS-HD.
2) -If not: Arcsoft TMT. (or Sonic Cinemaster)

[mrr19121970]

Originally Posted by madshi

Movie name and release country, please. Guys, please always state which Blu-Ray you have problems with. Makes my life much easier...

My disk was 'Leg dich nicht mit Zohan an - Unrated [Blu-ray]'

[madshi]

Originally Posted by NimaBeamer

When creating a PCM or LPCM from a seamless branched disc with TrueHD for example should the audio gaps i.e. the second pass occur or not? I am asking because it is not with the newest version.

Originally Posted by madshi

Then it's probably alright. How big the gaps are is a bit random. Probably they are just too small to register. But I'll double check with this Blu-Ray, just to be sure...

Originally Posted by NimaBeamer

But why does it register gaps when extracting the AC3?

Originally Posted by madshi

Because AC3 has the biggest frames (in terms of audio samples per frame) of all codecs. Biggest overlaps are with AC3, then comes DTS, then TrueHD and least affected is PCM.

Actually I was wrong. Here's how much overlap there can be per m2ts part of a seamless branching movie:

(E-)AC3: somewhere between 0ms and 32ms
DTS(-HD): somewhere between 0ms and 10.6ms
LPCM: somewhere between 0ms and 5ms
TrueHD: somewhere between 0ms and 0.8ms

So you can see that with TrueHD tracks there must be a whole lot of m2ts parts to make the overlaps big enough for eac3to to care. Accumulated overlaps must be bigger than 7ms, or else eac3to won't even bother reporting them. Practically, if every m2ts part contains the max amount of audio overlap, there must be at least 10 m2ts parts, or else a TrueHD track cannot have any overlaps (worth reporting). And since usually the overlaps per m2ts part is not the max value of 0.8ms, I'd guess that TrueHD overlaps begin with maybe 15-20 m2ts parts.

[madshi]

Originally Posted by williewonton

I've noted two titles that break eac3to both 2_81 and 2_82 as well as 2_83. Seems to relate to handling over audio gaps/overlaps. Eac3 eventually fails with "The temp file could not be interpreted correctly."

This will be fixed in the next build.

Originally Posted by mrr19121970

I have a disk that -demux es OK with V2.79 and not V2.83

This will also be fixed in the next build.

[madshi]

Originally Posted by nautilus7

I have the same disc. v2.83 doesn't find any gaps for the embedded ac3 in the truehd track (#4). Isn't it weird?

I've just tried with the first 4 m2ts parts (not full movie to save processing time) of the unrated cut and I'm already getting an overlap in the embedded AC3 track:

eac3to 00123.m2ts+00141.m2ts+00125.m2ts+00142.m2ts
 3: test.ac3
M2TS, 2 video tracks, 6 audio tracks, 19 subtitle tracks, 0:27:55
1: h264/AVC, 1080p24 /1.001 (16:9)
2: h264/AVC, 480p24 /1.001 (20:11)
3: TrueHD/AC3, English, 5.1 channels, 48khz
   (embedded: AC3, 5.1 channels, 448kbps, 48khz)
4: AC3, English, 5.1 channels, 448kbps, 48khz, dialnorm: -30dB
5: TrueHD/AC3, German, 5.1 channels, 48khz
   (embedded: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB)
6: AC3, Russian, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
7: AC3, Ukrainian, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
8: DTS Express, English, 2.0 channels, 24 bits, 192kbps, 48khz
[a03] Extracting audio track number 3...
[a03] Extracting AC3 stream...
[a03] Creating file "test.ac3"...
[a03] Audio overlaps for 27ms at playtime 0:05:51.
[a03] Starting 2nd pass...
[a03] Realizing (E-)AC3 gaps...
[a03] Creating file "test.ac3"...
eac3to processing took 6 minutes, 37 seconds.
Done.

Are you sure that you used the correct command line for testing?

[madshi]

Originally Posted by DrNein

When demuxing an AVC TS, the resolution changes from broadcast standard 1088i to 1084i.

I've double checked my code and I don't see anything wrong. 1088i should be cropped to 1080i by eac3to (and in my tests it works as expected). Which tool shows you 1084i? Are you sure that that tool is showing the correct information? What does eac3to say about the final demuxed video track? Does it also report 1084i? A sample would be very helpful. Thanks...

[madshi]

Originally Posted by menlvd

got fail sync

This will also be fixed in the next build.

[madshi]

eac3to v2.84 released

http://madshi.net/eac3to.zip

* fixed: 2nd pass gap removal was tried (and failed) for TrueHD+AC3 targets
* fixed: processing aborted when trying to fix gaps in PCM destination files
* fixed: more than one RAW/PCM overlaps resulted in lost sync (since v2.81)
* fixed: demuxing TrueHD+AC3 stream by title number didn't renew the AC3 part
* new option for removing or looping audio data, e.g. "-edit=0:20:47,-100ms"
* title sorting criteria changed: resolution is more important than runtime
* new option "-lowPriority" sets eac3to to background/idle priority
* libav warnings are now assigned to the affected audio track
* fixed: "lossless check failed" false alarms for seamless branching movies
* fixed: spike removal filter was not active for the very last overlap/gap
* improved muxing h264 streams which begin with double sequence headers
* source files are now opened with "share read + write access"
* destination files are now opened with "share read access"

Hopefully the new 2 pass processing logic introduced in v2.81 is working reliable now...

The new "-edit" option can be useful to manually sync audio tracks to movies. E.g. if you need to remove 150ms worth of audio data at the runtime of exactly 30 minutes, you can do this: "-edit=0:30:00,-150ms". Or if you need to add/loop 100ms audio data at runtime 15 minutes, 10 seconds and 500 milliseconds, you can do: "-edit=0:15:10.500,+100ms". You can only do one edit at a time, though. The editing feature should work for all supported audio formats.

[jj666]

Editing feature is great news for anyone syncing DVD audio to satellite caps. Thanks very much Madshi! When will you open up a facility for donations ;-)

Cheers,

-jj-

[Thunderbolt8]

sound great!

[mrr19121970]

Originally Posted by madshi

This will also be fixed in the next build.

It works, thanks....

[mrr19121970]

Is there a simple way to get the track details (like this):

M2TS, 1 video track, 8 audio tracks, 25 subtitle tracks, 1:56:52
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz
   (embedded: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB)
4: TrueHD/AC3, German, 5.1 channels, 48khz
   (embedded: AC3, 5.1 channels, 448kbps, 48khz)
5: AC3, Spanish, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
6: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -29dB
7: AC3 Surround, English, 2.0 channels, 192kbps, 48khz
8: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -29dB
9: AC3 Surround, English, 2.0 channels, 192kbps, 48khz
10: AC3, Portuguese, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
11: Subtitle (PGS), English
12: Subtitle (PGS), English
13: Subtitle (PGS), German
14: Subtitle (PGS), Spanish
15: Subtitle (PGS), Portuguese
16: Subtitle (PGS), German
17: Subtitle (PGS), Spanish
18: Subtitle (PGS), Portuguese
19: Subtitle (PGS), German
20: Subtitle (PGS), Spanish
21: Subtitle (PGS), Portuguese
22: Subtitle (PGS), German
23: Subtitle (PGS), Spanish
24: Subtitle (PGS), Portuguese
25: Subtitle (PGS), German
26: Subtitle (PGS), Spanish
27: Subtitle (PGS), Portuguese
28: Subtitle (PGS), Turkish
29: Subtitle (PGS), French
30: Subtitle (PGS), Dutch
31: Subtitle (PGS), Korean
32: Subtitle (PGS), English
33: Subtitle (PGS), French
34: Subtitle (PGS), Dutch
35: Subtitle (PGS), Korean

without -demux ing, and the pausing ? as I ultimately only want to do this ?

eac3t0 R:\ 1) 2: Movie.h264 3: English.ac3 4: German.ac3

Thanks in advance. Mike.

[rebkell]

Originally Posted by jj666

Editing feature is great news for anyone syncing DVD audio to satellite caps. Thanks very much Madshi! When will you open up a facility for donations ;-)

Cheers,

-jj-

yes, that is a great feature, I've got a few captures right now, I'll probably try to get it working on, They are pretty close, but they need a tweak here and there.

[cavediver]

Originally Posted by rica

1) Yes: eac3to inputfile outputfile (*.dtshd/dts) -both will give the same result except you use -core option. "-core" extracts the core not DTS-HD.
2) -If not: Arcsoft TMT. (or Sonic Cinemaster)

Thank you.

[madshi]

Originally Posted by mrr19121970

Is there a simple way to get the track details [...] without -demux ing [...]?

Yes:

eac3to R:\ 1)

[frenchglen]

Has anyone tried decoding the 6.1ch DTS-HD MA track on the Wall-e Blu-ray?
Sonic thinks it's only 5.1ch.

[asarian]

Originally Posted by frenchglen

Has anyone tried decoding the 6.1ch DTS-HD MA track on the Wall-e Blu-ray?
Sonic thinks it's only 5.1ch.

Probably because it is 5.1.

eac3to 20000.m2ts
M2TS, 3 video tracks, 3 audio tracks, 3 subtitle tracks, 1:37:26
1: h264/AVC, 1080p24 /1.001 (16:9)
2: h264/AVC, 480p24 /1.001 (20:11)
3: h264/AVC, 480p24 /1.001 (20:11)
4: DTS Master Audio, 5.1 channels, 24 bits, 48khz
(core: DTS-ES, 5.1 channels, 24 bits, 1509kbps, 48khz)
5: AC3 Surround, 2.0 channels, 192kbps, 48khz
6: AC3 Surround, 2.0 channels, 192kbps, 48khz
7: Subtitle (PGS)
8: Subtitle (PGS)
9: Subtitle (PGS)

[Thunderbolt8]

perhaps it only switches to 6.1 at a later stage of the movie.

[asarian]

Originally Posted by Thunderbolt8

perhaps it only switches to 6.1 at a later stage of the movie.

I doubt it. That's the m2ts for the main title. I don't see how it could change to 6.1 midstream.

EDIT: Could be they released a disc with different tracks in France, of course.

[rebkell]

Originally Posted by madshi

eac3to v2.84 released

http://madshi.net/eac3to.zip

The new "-edit" option can be useful to manually sync audio tracks to movies. E.g. if you need to remove 150ms worth of audio data at the runtime of exactly 30 minutes, you can do this: "-edit=0:30:00,-150ms". Or if you need to add/loop 100ms audio data at runtime 15 minutes, 10 seconds and 500 milliseconds, you can do: "-edit=0:15:10.500,+100ms". You can only do one edit at a time, though. The editing feature should work for all supported audio formats.

I assume this will work like the delay, if I have an AC3@48KHz, it would drop 3 32msec frames if I put in -100ms, and add 96ms if I put in +100ms. I really love this feature, I've needed this for a long time. Now I've got something to fix my encodes that end with random sync problems because of corrupted ts streams, now I just have to figure out how much I need to tweak to get things back in sync.

[madshi]

Originally Posted by rebkell

I assume this will work like the delay, if I have an AC3@48KHz, it would drop 3 32msec frames if I put in -100ms, and add 96ms if I put in +100ms.

Yes, that's correct. The new "edit" feature is basically just reusing the same code which is fixing gaps/overlaps in seamless branching movies.

[odin24]

This might be a matter of preference, what would produce better quality audio... A HDDVD with EAC3 @ 1536kb/s with dialnorm or that same track transcoded to DTS @1536kb/s for Blu-ray. Would there would be some quality loss in the conversion or would all of the data be retained?

Thanks.

[Atak_Snajpera]

It's like asking which track sounds better mp3 @ 320 kbps or aac @ 320 kbps (or which is better AVC @ 50 Mbps or VC-1 @ 50Mbps?) Conclusion is that you won't hear any difference. (assuming that your source is uncompressed)
BTW. Transcoding from one lossy format to another always means loss in quality.

[williewonton]

@madshi

Version 2.84 works a treat. Thanks.

[nautilus7]

Originally Posted by madshi

I've just tried with the first 4 m2ts parts (not full movie to save processing time) of the unrated cut and I'm already getting an overlap in the embedded AC3 track:

eac3to 00123.m2ts+00141.m2ts+00125.m2ts+00142.m2ts
 3: test.ac3
M2TS, 2 video tracks, 6 audio tracks, 19 subtitle tracks, 0:27:55
1: h264/AVC, 1080p24 /1.001 (16:9)
2: h264/AVC, 480p24 /1.001 (20:11)
3: TrueHD/AC3, English, 5.1 channels, 48khz
   (embedded: AC3, 5.1 channels, 448kbps, 48khz)
4: AC3, English, 5.1 channels, 448kbps, 48khz, dialnorm: -30dB
5: TrueHD/AC3, German, 5.1 channels, 48khz
   (embedded: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB)
6: AC3, Russian, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
7: AC3, Ukrainian, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
8: DTS Express, English, 2.0 channels, 24 bits, 192kbps, 48khz
[a03] Extracting audio track number 3...
[a03] Extracting AC3 stream...
[a03] Creating file "test.ac3"...
[a03] Audio overlaps for 27ms at playtime 0:05:51.
[a03] Starting 2nd pass...
[a03] Realizing (E-)AC3 gaps...
[a03] Creating file "test.ac3"...
eac3to processing took 6 minutes, 37 seconds.
Done.

Are you sure that you used the correct command line for testing?

Tested again, but with 2.84. And it finds many gaps, so i guess there's nothing to worry about now.

[odin24]

Originally Posted by Atak_Snajpera

It's like asking which track sounds better mp3 @ 320 kbps or aac @ 320 kbps (or which is better AVC @ 50 Mbps or VC-1 @ 50Mbps?) Conclusion is that you won't hear any difference. (assuming that your source is uncompressed)
BTW. Transcoding from one lossy format to another always means loss in quality.

OK then, thanks for the reply.

[krosswindz]

I have a DTS HD MA 6.1ch track @ 16 bit. When I try to extract the core or convert it to 5.1ch 1.5 MBps DTS eac3to seems to patch to 24bit DTS. Is there any way I can prevent it from patching it to 24bit and keep it at 16 bit. -16 switch works only with RAW/PCM audio I suppose.

[Chrishel]

First off, I want to thank madshi for creating such a robust conversion program! I'm trying to archive my high-resolution audio collection, and eac3to works great for converting mlps from DVD-Audios, but I can't seem to get it to work easily with my DTS CDs.

I start off by ripping the DTS CDs to dtswavs using Exact Audio Copy. The subsequent dtswavs are detected as having DTS streams and play as they should through directshow filters, but when I try to use eac3to to convert them, it treats them as 2 channel wav files. Here is the contents of the log file eac3to produces:

eac3to v2.84
command line: "\Program Files (x86)\eac3to\eac3to.exe"  "01 - Tequila Sunrise.dtswav" "01 - Tequila Sunrise.flac"
------------------------------------------------------------------------------
WAV, 2.0 channels, 0:03:26, 16 bits, 1411kbps, 44.1khz
Reading WAV...
Encoding FLAC with libFlac...
Creating file "01 - Tequila Sunrise.flac"...
The original audio track has a constant bit depth of 16 bits.
eac3to processing took 6 seconds.
Done.

If I open the dtswav in DTSParser, it detects the file as having an open bitrate, with a format of DTS 20/44.1 in 6.1. If I use DTSParser to rebuild the stream and save a .dts file, the subsequent dts file also plays fine, but eac3to gives the error "The format of the source file could not be detected.". In case it is relevant, the original dtswav is 34.6 MB and the rebuilt dts file is 30.2 MB.

The only way I've found to come close to completing this conversion is to use VLC to save a multichannel wav file, and then use eac3to to convert that. However, the channels from that appear to be 16-bit, and according to DTSParser, the original is 20-bit, so I'd rather not lose the quality if I can avoid it.

Do you have any suggestions as to anything I could try to get eac3to to detect the dtswavs properly. Is it possible Exact Audio Copy is adding or removing some header information which is interfering with eac3to's ability to detect them?

[Rectal Prolapse]

madshi, can you enter a more finely-grained timestamp for the audio editing? For example, to the nearest 100th of a millisecond? Instead of "01:30:02" can it be "01:30:02.250"?

Thanks!

[Snowknight26]

Should probably read his post again.
http://forum.doom9.org/showpost.php?...postcount=7460

[bmnot]

madshi, eac3to is hands down one of the best designed and supported pieces of software I've ever used. Is there anyway I can donate a little something for all the hard work you put into it?

[williewonton]

madshi

While 2.84 isn't broken by the title "Game Plan" I've found that the audio isn't correctly sequenced. I'll do some more looking and see if I can make sense of it.

[Rectal Prolapse]

Originally Posted by Snowknight26

Should probably read his post again.
http://forum.doom9.org/showpost.php?...postcount=7460

Hmm somehow I missed that. Everything afterwards never used that example. (Signal-to-noise is kind of low in this thread lol)

Thank you, snow kiniggit! You're my *hero*.

[madshi]

Originally Posted by krosswindz

I have a DTS HD MA 6.1ch track @ 16 bit. When I try to extract the core or convert it to 5.1ch 1.5 MBps DTS eac3to seems to patch to 24bit DTS. Is there any way I can prevent it from patching it to 24bit and keep it at 16 bit. -16 switch works only with RAW/PCM audio I suppose.

Why does the patching bother you?

Originally Posted by Chrishel

Do you have any suggestions as to anything I could try to get eac3to to detect the dtswavs properly.

Yes. Make a sample available to me...

Originally Posted by bmnot

madshi, eac3to is hands down one of the best designed and supported pieces of software I've ever used. Is there anyway I can donate a little something for all the hard work you put into it?

Thanks! And no, right now there's no way to donate, cause such a donation would be difficult for me to handle tax wise. But I might change my mind next year...

Originally Posted by williewonton

While 2.84 isn't broken by the title "Game Plan" I've found that the audio isn't correctly sequenced. I'll do some more looking and see if I can make sense of it.

What do you mean exactly with "correctly sequenced"?

[Chrishel]

Originally Posted by madshi

Yes. Make a sample available to me...

Check your PMs. :-)

[madshi]

Originally Posted by Chrishel

Check your PMs. :-)

I'm checking my PMs, anyway, so there's no need to flood this thread with such posts.

-------

Your DTSWAV sample is strange. It begin with a lot of zero data and then half a DTS frame and then finally with the real DTS data. eac3to cannot automatically detect such a file as a valid DTSWAV file. Don't know, maybe the EAC extra features (which are very useful for ripping normal WAV data) are harmful for DTSWAV tracks? You could try playing around with the EAC options to see whether that makes any difference.

Furthermore the DTS data in the DTSWAV file is kind of broken. That's probably not the fault of EAC. The framesize is supposed to be 3585 bytes, but the data only contains 3584 bytes per frame. That's why eac3to didn't accept the DTS file extracted by DTSParser, either. I'll implement a workaround for such broken DTS files into the next eac3to build. So the next eac3to build will be able to detect the DTSParser extracted DTS file as:

DTS, 5.1 channels, 0:03:25, 20 bits, 1235kbps, 44.1khz

Also decoding will be possible with ArcSoft, libav and Nero, but not with Sonic.

[Chrishel]

Thanks so much!

[tebasuna51]

Originally Posted by krosswindz

I have a DTS HD MA 6.1ch track @ 16 bit. When I try to extract the core or convert it to 5.1ch 1.5 MBps DTS eac3to seems to patch to 24bit DTS. Is there any way I can prevent it from patching it to 24bit and keep it at 16 bit. -16 switch works only with RAW/PCM audio I suppose.

Maybe you are confused, only DTS MA (lossless) can have a exact bitdepth.

The core, or a downmix to 5.1 reencoded to a standard DTS (lossy), don't have any bitdepth, the samples are stored in frequency domain with a equivalent precission to 20-24 bits in uncompressed format, not matter if the source is 16 or 24 bits.

A -16 switch with a dts output (or other lossy format) have not sense.

[krosswindz]

Originally Posted by madshi

Why does the patching bother you?

If I can say bit depth is something equivalent to resolution of the sample stored. If the source is 16bit then while re-encoding if it is re-encoded as 24bit would this mean the encoder is filling up the missing data.

Originally Posted by tebasuna51

Maybe you are confused, only DTS MA (lossless) can have a exact bitdepth.

The core, or a downmix to 5.1 reencoded to a standard DTS (lossy), don't have any bitdepth, the samples are stored in frequency domain with a equivalent precission to 20-24 bits in uncompressed format, not matter if the source is 16 or 24 bits.

A -16 switch with a dts output (or other lossy format) have not sense.

Very true I am confused on this.

What I am trying to do is to keep everything at the same except downmix to 5.1ch keeping every other same as possible.

edit: @madshi it would be great if there was an option to prevent patching and let the default be that the audio be patched.

[tebasuna51]

Originally Posted by krosswindz

If I can say bit depth is something equivalent to resolution of the sample stored. If the source is 16bit then while re-encoding if it is re-encoded as 24bit would this mean the encoder is filling up the missing data.

Still confused. After a lossy encode forget the source resolution, if we can recover the source resolution (even if is poor) we have a lossless encoder.

For a lossy output the best procedure is, like madshi know, decode to high resolution (64 or 32 bit float) make functions (mix, speed, resample, ..) with max resolution and, at end, down to the max resolution supported by the encoder (24 bit int for Surcode DTS, 32 bit float for NeroAacEnc, ...).

In lossless formats the precision is know by the bitdepth, in lossy formats by the bitrate (forget the bitdepth).

[Thunderbolt8]

its basically that the sound quality in it remains at 16-bit, but it seems like the way it is stored can only be 24-bit in that case. so theres no downmixing or artificial upgrading of the original quality, it remains untouched, its only the 'package' which transports that sound content which gets changed. but apparently it cannot be made another way.

(correct me if im wrong)

[williewonton]

madshi

The video runs apparently OK, but the audio track repeats the opening theme over and over and doesn't progress to the dialog etc. So it is an odd one. In a seamless branching title, what is the maximum you allow for/handle?

[madshi]

Originally Posted by krosswindz

If I can say bit depth is something equivalent to resolution of the sample stored.

If you're talking about a LPCM or WAV track then you're right. If you're talking about a lossy DTS track then you're flat out wrong. As tebasuna51 already explained, DTS tracks do not store samples in any specific bitdepth. The DTS "bitdepth" header field only has pure informational character and tells us which bitdepth the original PCM audio master had which was fed into the DTS encoder. But the DTS track itself is not bound to any specific bitdepth.

Originally Posted by krosswindz

@madshi it would be great if there was an option to prevent patching

No, because IMO such an option would be totally useless. If you find a technically correct argument for adding such an option, then please let me know. But right now I don't see any such argument...

Originally Posted by williewonton

The video runs apparently OK, but the audio track repeats the opening theme over and over and doesn't progress to the dialog etc. So it is an odd one. In a seamless branching title, what is the maximum you allow for/handle?

There's no max. You can have thousands of m2ts parts, no problem, just takes longer.

I remember that there was a Blu-Ray disc which really had a broken audio track just like you describe it. Are you sure that eac3to borked up this audio track? I rather guess that the audio data in the m2ts files is broken. However, IIRC there's another audio track which is working fine. So please check all audio tracks. You may find one which is correctly working...

[krosswindz]

@madshi/tebasuna51 thanks for the explanation, I didnt know that the DTS "bitdepth" header field was only for informational purpose. I know have a better understanding of what you mean.

@madshi: Really love your tool. Is there any future plan for supporting DTS pro series encoders?

[rica]

Originally Posted by Atak_Snajpera

Transcoding from one lossy format to another always means loss in quality.

Not always, this is a myth which everybody believes in.

It is up to the capabilities of mostly decoder, plus encoder.

I have lots of dts samples sound better than the original ac3s.

Edit: Before getting the objections, i should say:
Why do you use an Hi-End CD Player while you can listen the same CD on an ordinary player?
Because of its HW decoder (plus of its DAC); correct?
So if you can match the Hi-Fi decoder filter with an appropriate encoder filter; why not?
This is why i said "decoders are more important than the encoders in transcoding" before...
And madshi knows what they are and selected those Hi-Fi decoders as default in his tool...
So extracting wavs using an Hi-Fi decoder and re-encoding them with an external Pro encoder would be the best choice.
_ _ _ _ _ _

[tebasuna51]

Originally Posted by rica

Not always, this is a myth which everybody believes in.

Well you can filter noise or modify anything wrong in the original. But, with a correct source, isn't a myth.

It is up to the capabilities of mostly decoder, plus encoder.

No.

I have lots of dts samples sound better than the original ac3s.

Please check your players settings.

"decoders are more important than the encoders in transcoding"

No.

[rica]

"No" would not explain anything.
Except telling the same story...

[alc0re]

Sorry if this has been asked before or if it is a stupid question :

I have been using eac3to to extract video/audio/subs from bluray disks.

I noticed something though and I have a question.

What exactly does the removal of dialog normalization do?

I ask because when I extract an ac3 (dolby digital) track with dialog normalization, and let eac3to remove the dialog normalization, when I play my final encoded avchd structured dvd9 in either my bluray player or my PS3 the audio is really really low and I have to crank up my receiver's volume almost all the way up to hear anything. If I fast forward at like 1.5x realtime where you can still hear the audio its normal level, but as soon as you hit play again it drops the audio down again.

I blamed a PS3 firmware update for awhile but I exchanged my PS3 for a panasonic DMP-BD35K bluray player and it has the same behavior.

I finally got around to trying the -keepDialnorm and now I can hear the audio with my receiver's volume at a normal range. What exactly is removing the dialog normalization buying me and why is it not recommended to use the -keepDialnorm switch? Is there a way to extract the ac3 audio that already has had its dialog normalization removed from my dvd9s and do the reverse process that removing the dialog normalization does? Anyone else have/seen this issue? Perhaps its my receiver? (Yamaha HTR-5940)

[Mercury_22]

Sorry for n00b question but How can I demux to specific location using "-demux " command ?

[madshi]

Originally Posted by krosswindz

Is there any future plan for supporting DTS pro series encoders?

There is AGM output support (for older DTS pro encoders). Don't know if anything more than that will come.

Originally Posted by rica

Not always, this is a myth which everybody believes in.

No, it is not a myth. It's a technical fact that every encoding process to a lossy format throws away information. There may be situations where transcoding from lossy to lossy can result in an improvement. E.g. (as tebasuna51 said) if you post process the audio data in a good way. Or e.g. if one of the decoders in your receiver is broken. But these are really exceptions. It's a fact that every straight transcoding from lossy to lossy format loses audio information.

Originally Posted by alc0re

What exactly does the removal of dialog normalization do?

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

Originally Posted by alc0re

I ask because when I extract an ac3 (dolby digital) track with dialog normalization, and let eac3to remove the dialog normalization, when I play my final encoded avchd structured dvd9 in either my bluray player or my PS3 the audio is really really low and I have to crank up my receiver's volume almost all the way up to hear anything. If I fast forward at like 1.5x realtime where you can still hear the audio its normal level, but as soon as you hit play again it drops the audio down again.

I blamed a PS3 firmware update for awhile but I exchanged my PS3 for a panasonic DMP-BD35K bluray player and it has the same behavior.

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

[madshi]

Originally Posted by Mercury_22

Sorry for n00b question but How can I demux to specific location using "-demux " command ?

By changing the command prompt "current directory" to the wanted location. (e.g. "d:", followed by "cd movies").

[madshi]

Originally Posted by DrNein

When demuxing an AVC TS, the resolution changes from broadcast standard 1088i to 1084i. Why is that? Is there a way to leave it alone or else change to 1080?

Originally Posted by madshi

I've done several checks:

(1) MediaInfo reports 1920x1088 for the original video track. And it reports 1920x1080 for the eac3to processed track.

(2) When playing back the original TS file with the Cyberlink h264 decoder, the video renderer gets 1920x1088 pixels. When playing back the eac3to processed video track with the same decoder, the video renderer gets 1920x1080 pixels.

(3) Same as (2), but with Sonic h264 decoder.

(4) Same as (2), but with ffmpeg/libav h264 decoder.

I think that are enough proofs that eac3to's processing is 100% correct. The cropping down to 1080 from 1088 is a (surprisingly) complicated calculation and tsMuxeR evidently does it in the wrong way. It's so complicated because the cropping size must be multiplied several times, depending on some conditions. E.g. with your sample, eac3to writes a cropping value of "2" into the video bitstream. This must be multipled by 2 twice to get to the correct cropping size of "8". tsMuxeR seems to forget one of the multiplications, that's why it incorrectly reports 1084i instead of 1080i.

Short summary: This is a(nother) bug in tsMuxeR.

Originally Posted by DrNein

Well, I suppose that is sort of good news then

Hopefully, tsMuxeR will be corrected for this and a few other things soon (such as 25.01 FPS output and THD handling). In the meantime, do you think it is safe to use tsMuxeR to mux such files? That is, is it only reporting wrong but not altering them?

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

Originally Posted by DrNein

Do you plan to add TS and/or M2TS output to eac3to?

No.

Originally Posted by DrNein

Also, I have not really noticed any problem with playback of most most 1088 AVC files however 1088 MPEG-2 with DXVA often display a grey bar in MPC-HC but not in PowerDVD. Can eac3to correct those MPEG-2?

No. In MPEG2 it's not as easy as in h264.

Originally Posted by DrNein

Is there any potential drawback to eac3to automatically processing to 1080 (with either codec)?

I see none. Actually I think it has several advantages. E.g. with the original file the video renderer actually gets 1088 lines. What will the renderer do with that? Display all of them? Then it must downscale the image, which means we don't have 1:1 pixel mapping, anymore. With the eac3to cropped video bitstream the video renderer is only getting 1080 lines, so we get perfect 1:1 pixel mapping. You can see that for yourself: If you play the original TS file, you'll see some garbage lines at the bottom of the screen. With the eac3to processed stream these garbage lines are gone.

[Mercury_22]

Originally Posted by madshi

By changing the command prompt "current directory" to the wanted location. (e.g. "d:", followed by "cd movies").

Thanks !

[mrr19121970]

Originally Posted by madshi

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

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

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

:beg: :beg:

[piratburner]

Originally Posted by mrr19121970

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

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

:beg: :beg:

That's should be something

[Thunderbolt8]

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

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

[alc0re]

Originally Posted by madshi

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


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

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

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

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

[n0mag!c]

Originally Posted by madshi

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


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

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

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

[Atak_Snajpera]

Wake up man! Download 2.84 !!!

[jonathonsunshine]

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

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

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

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

[Easy123]

I think I got a bug with v2.84.

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

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

[73ChargerFan]

Originally Posted by alc0re

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

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

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

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

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

[wolfbane5]

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

[alc0re]

Originally Posted by n0mag!c

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

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

n0magic :

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

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

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

[tebasuna51]

Originally Posted by alc0re

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

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

[Thunderbolt8]

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

[williewonton]

madshi

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

[yfed]

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

Here's my cmd:

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

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

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

[madshi]

Originally Posted by n0mag!c

Please don't be hurry to change it while only one man states the issue!

Don't worry. The change I'm planning to do should have no effect at all on proper decoders. And it might help fix bad decoders.

Originally Posted by jonathonsunshine

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

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

eac3to was never able to fix overlaps/gaps in TrueHD streams, if you just demux them to "*.thd" or "*.thd+ac3". However, eac3to can fix the overlaps/gaps if you transcode to another format (e.g. WAV or FLAC).

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

Originally Posted by Easy123

I think I got a bug with v2.84.

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

Yes, your suspicion seems to be true. I think this only occurs when using the "-2pass" option, though. Just try without it and everything should probably be fine. You never have to use the "-2pass" option, eac3to will do 2 passes in any case where it's necessary, regardless of whether you used the "-2pass" option or not. The only sense of that option is to tell eac3to that probably 2 passes are necessary. In that situation eac3to does some things differently to speed up the processing a bit. It's only a performance optimization for situations where 2 passes are necessary. But it's never necessary to use that option...

Originally Posted by Thunderbolt8

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

There's nothing you have to do. This "detected PTS break" simply means that the timestamps in the source file are not continuous, but there's a jump in the timestamps somewhere. Now whenever there's such a jump in the timestamps, eac3to of course has to handle that somehow. It does so by assuming that the first video frame after the timestamp jump is supposed to be played exactly one frame after the last video frame before the timestamp jump. That's all this log message says. So again, you don't have to do anything. However, this message indicates that either the source is corrupt or that it was cut (e.g. to remove advertising). It's quite possible that you won't notice the problem and everything is just fine.

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

Originally Posted by wolfbane5

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

eac3to gets the mkvtoolnix path from one of these registry values:

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

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

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

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

Originally Posted by williewonton

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

I've checked back. Please read these posts:

http://forum.doom9.org/showpost.php?...postcount=5854
http://forum.doom9.org/showpost.php?...postcount=5954
http://forum.doom9.org/showpost.php?...postcount=5956

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

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

Originally Posted by yfed

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

Here's my cmd:

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

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

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

[n0mag!c]

Originally Posted by alc0re

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

Ok, let's wait for changes, wish it will help you!

Originally Posted by alc0re

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

By the way, I'm bitstreaming via HDMI. And sometimes I switch to LPCM output from PS3. My receiver is Denon 2308.

[mrr19121970]

@madshi

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

nice to have #1 would be:

eac3to d:\ 1) -demux *h264* *english* *german*

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




nice to have #2 would be:

eac3to d:\ 1) 2: Movie.* 3: English.ac3 4: German.ac3

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



Thanks for looking & Happy Christmas.

[mrr19121970]

I just tried (Atonement HD-DVD):

eac3to q:\

and it only gave me:

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



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

[madshi]

Originally Posted by mrr19121970

I just tried (Atonement HD-DVD):

eac3to q:\

and it only gave me:

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



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

Can you please upload the XPL files? Should be just a few KBs...

[mrr19121970]

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

[Thunderbolt8]

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

[wolfbane5]

Originally Posted by madshi

Surcode is located by checking these registry values:

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

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

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

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

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

[zeropc]

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

thanks for help

[Chumbo]

Originally Posted by zeropc

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

thanks for help

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

[EDIT] Here's the post: tebasuna51 knows all

[asarian]

Originally Posted by n0mag!c

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

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

[alc0re]

Originally Posted by asarian

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

Read the post.

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

[tebasuna51]

Originally Posted by asarian

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

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

[rebkell]

Originally Posted by tebasuna51

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

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

[zeropc]

Originally Posted by Chumbo

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

[EDIT] Here's the post: tebasuna51 knows all

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

[Thunderbolt8]

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

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

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

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

[nautilus7]

Originally Posted by zeropc

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

Don't you have Nero 7 installed?

Merry Christmas to all!!

[rickardk]

Originally Posted by Thunderbolt8

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

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

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

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

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

[Chumbo]

Originally Posted by zeropc

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

Ah, sorry, you're right. I miss-read your post. Ugh...

[Chumbo]

Originally Posted by nautilus7

...Merry Christmas to all!!

Merry Christmas to you as well!

    *
   / \
  /o  \
 /   o \
/_ o   _\
   | |

[Thunderbolt8]

Originally Posted by rickardk

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

same here :S

[Thunderbolt8]

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

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

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

[alc0re]

Originally Posted by rebkell

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

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

[rebkell]

Originally Posted by alc0re

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

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

[odin24]

Deleted.

[itsancho]

strange, but minor bug in 2.84...

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

2) 00066.mpls, 2:23:08
[30+31+38+39+40+64].m2ts
- MPEG2, 480i60 /1.001 (16:9)
- AC3, English, stereo, 48khz

eac3to "H:\Hellboy (2004) Blu-ray AVC PCM" 1)
M2TS, 1 video track, 1 audio track, 5 subtitle tracks, 2:23:08
1: Chapters, 7 chapters
2: MPEG2, 480i60 /1.001 (16:9)
3: AC3 Surround, English, 2.0 channels, 192kbps, 48khz
4: Subtitle (PGS), English
5: Subtitle (PGS), French
6: Subtitle (PGS), German
7: Subtitle (PGS), Dutch
8: Subtitle (PGS), Korean

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

Merry Christmas everyone! And Thank You madshi!

[tebasuna51]

Originally Posted by Thunderbolt8

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

The ac3 extracted with eac3to from your sample with:

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

have 6 frames 2.0 and after change to 5.1. Delaycut log:

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

[Snowknight26]

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

[tebasuna51]

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

[dorati]

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

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

Merry X-Mas

[Thunderbolt8]

Originally Posted by tebasuna51

The ac3 extracted with eac3to from your sample with:

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

have 6 frames 2.0 and after change to 5.1. Delaycut log:

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

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

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

[ggking7]

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

[odin24]

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

Thanks,

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

[Atak_Snajpera]

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

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

[Snowknight26]

Originally Posted by tebasuna51

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

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

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

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

[vucloutr]

hi there, i found a minor bug with .aac encoding.
eac3to handles "-quality=0.01" like "-quality=0.10", "-quality=0.02" like "-quality=0.20", .. , "-quality=0.09" like "-quality=0.90".

i noticed it when i saw these assumingly wrong filesizes:

[tebasuna51]

Originally Posted by Snowknight26

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

Of course.

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

Yes. Now:
6 = Side Left
7 = Side Right

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

Nope, the eac3to remapping works fine.

[Snowknight26]

G:\Encoding Tools\eac3to>eac3to.exe ..\bhd.broken.cut.flac
FLAC, 5.1 channels, 16 bits, 48khz

G:\Encoding Tools\eac3to>eac3to.exe ..\bhd.broken.cut.flac bhd.broken.cut.flac
FLAC, 5.1 channels, 16 bits, 48khz
Decoding FLAC...
Encoding FLAC with libFlac...
Creating file "bhd.broken.cut.flac"...
The original audio track has a constant bit depth of 16 bits.
eac3to processing took 5 seconds.
Done.

G:\Encoding Tools\eac3to>eac3to.exe bhd.broken.cut.flac
FLAC, 5.1 channels, 0:01:05, 16 bits, 1233kbps, 48khz

Any reason why the 1st track didn't show the length?..

G:\Encoding Tools\eac3to>eac3to.exe bhd.cut.flac
FLAC, 5.1 channels, 2:24:20, 16 bits, 9kbps, 48khz

G:\Encoding Tools\eac3to>eac3to.exe ..\bhd.cut.flac bhd.cut.flac
FLAC, 5.1 channels, 2:24:20, 16 bits, 9kbps, 48khz
Decoding FLAC...
Encoding FLAC with libFlac...
Creating file "bhd.cut.flac"...
The original audio track has a constant bit depth of 16 bits.
eac3to processing took 5 seconds.
Done.

G:\Encoding Tools\eac3to>eac3to.exe bhd.cut.flac
FLAC, 5.1 channels, 0:01:05, 16 bits, 1235kbps, 48khz

..and this one shows different ones?

http://www.stfcc.org/misc/bhd.cut.flac (before reencoding to FLAC)
http://www.stfcc.org/misc/bhd.broken.cut.flac (before reencoding to FLAC)

(Don't mind the broken in the name, I was messing with channel orders - broken refers to the one with the wrong channel order.)

[madshi]

Originally Posted by wolfbane5

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

If you own a proper Surcode DTS encoder license and if you have installed it properly, these folders are supposed to be there. So my suggestion would be to uninstall, get the latest version from Surcode and install it. Please note that AFAIK there are two different products: One for CD (stereo) encoding and one for movie encoding. eac3to only supports the latter one...

Originally Posted by Thunderbolt8

eac3to lists the 7.1 DTS-HD MA track of the nordic black hawk down edition with "(strange setup)" and "CAUTION: Decoding this track with ArcSoft results in low volume" [...].

so I guess arcsoft does not process this track very well, is the channel order wrong?

No. The log already tells you what is wrong: The volume is lower than it should be. There's no problem with the channel order...

Originally Posted by rickardk

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

Hmmmm... Could you guys please report this problem to ArcSoft? I mean it should also occur if you play the full physical Blu-Ray disc with the ArcSoft player (as long as your speakers are setup to be 7.1), right? So this is something ArcSoft should really be willing to fix...

I've already seen this problem once. I'm not sure if the DTS 5.1 -> 7.1 channel extension is broken in this specific movie or whether there's a bug in the ArcSoft decoder. Probably the latter. You can work around it by either using Sonic for decoding or by using the "-6" parameter to force ArcSoft to output 6 channels, only. I'm not fully sure whether the "-6" removes the problem completely. Maybe it does, maybe not. You'll have to check that yourself. Using Sonic for this track might be safer. Of you can decode only the core with ArcSoft by using the "-core" switch. That's safe, too, but of course loses the Master Audio information...

Originally Posted by itsancho

strange, but minor bug in 2.84...

Thanks for the report, will be fixed in the next build (I hope).

Originally Posted by dorati

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

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

I'm currently only demuxing the subtitle stream. I'm not even trying to "understand" it. So I currently don't have the capability to extract only a part of the subtitle stream. Of course it would be possible to add. But I'm not sure how difficult it is, and to be honest, I still have lots of more important things on my to do list, which are probably easier to implement...

Originally Posted by ggking7

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

I'm not a fan of storing video/audio in a suboptimal format just to work around some special requirements imposed on you by your hardware. What happens if in 2 years from now you have a different display which works better with pulldown stripped? IMHO you should store video/audio in the best possible format and that is with pulldown removed. But that's just my opinion.

Now specifically about 120Hz: With pulldown you have 2*60Hz = 120Hz. Without pulldown you have 5*24Hz = 120Hz. So in theory a good 120Hz display should be able to handle the video track fine, regardless of whether you remove the pulldown or not. With pulldown removed it should even be better. But in pratical life things are often different than theory suggests...

Originally Posted by odin24

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

    - corePackets          Core+XCh
    - extSubStrPackets     XBR

It means that the core already contains the back channel ("XCh") extension. The DTS-HD data only contains an XBR extension, which is additional bitrate. I don't know if the additional bitrate is only for 5.1 or whether it's for 6.1, but that doesn't really matter. The DTS-HD data does improve audio quality of the 5.1 channels in your specific case.

Originally Posted by vucloutr

hi there, i found a minor bug with .aac encoding.
eac3to handles "-quality=0.01" like "-quality=0.10", "-quality=0.02" like "-quality=0.20", .. , "-quality=0.09" like "-quality=0.90".

i noticed it when i saw these assumingly wrong filesizes:

Thanks for the report, will be fixed in the next build.

Originally Posted by Snowknight26

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

Should work just fine. If you can reproduce that it doesn't work, then please post your eac3to log. You should check channel order in a WAV editor like e.g. Audacity.

Originally Posted by Snowknight26

Any reason why the 1st track didn't show the length?

Yes, the length information is not stored in that file. It's as easy as that.

[madshi]

Originally Posted by zeropc

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

The problem is that the standalone Nero AAC decoder only supports *.mp4 files as input, while eac3to only supports demuxed raw AAC streams. So those two tools don't match well together, unfortunately.

[n0mag!c]

Originally Posted by madshi

Now specifically about 120Hz: With pulldown you have 2*60Hz = 120Hz. Without pulldown you have 5*24Hz = 120Hz.

Unless 24p mode is activated both on player and TV, player transmits content at 30/25fps, and TV is refreshing with multiplier of 4 at 120/100 Hz. And in 24p mode there is no matter with what multiplier TV is refreshing - 4 or 5 (I ain't trust TV in this mode, that its refresh rate stays 120Hz), because surely no additional frames are interpolated, the same frame is shown continually.
But 24p mode can only be activated when pulldown is removed, so this is another reason why pulldown can be removed.

Originally Posted by madshi

So in theory a good 120Hz display should be able to handle the video track fine, regardless of whether you remove the pulldown or not. With pulldown removed it should even be better. But in pratical life things are often different than theory suggests...

[wolfbane5]

Originally Posted by madshi

If you own a proper Surcode DTS encoder license and if you have installed it properly, these folders are supposed to be there. So my suggestion would be to uninstall, get the latest version from Surcode and install it. Please note that AFAIK there are two different products: One for CD (stereo) encoding and one for movie encoding. eac3to only supports the latter one...

It's possible that I don't have those folders since I'm using v1.0.23 which I believe is not the latest version. However, when I fire up the program it does say 'Surcode DVD-Pro DTS Encoder' so I'm assuming I'm using the movie encoding one. Truth be told, I borrowed SurCode and ArcSoft from a buddy of mine who's a sound engineer since he hardly uses them. I wanted to see if they'd actually work for my purposes, but apparently they're throwing some curve balls. By the way, what about ArcSoft? Where do I find that in the registry? I'd like to be able to use '-core' in eac3to.

[madshi]

Originally Posted by wolfbane5

It's possible that I don't have those folders since I'm using v1.0.23 which I believe is not the latest version.

v1.0.23 is supposed to work. Try reinstalling with admin rights...

Originally Posted by wolfbane5

By the way, what about ArcSoft? Where do I find that in the registry? I'd like to be able to use '-core' in eac3to.

The "-core" switch doesn't need ArcSoft. You only need ArcSoft for best quality decoding...

[madshi]

eac3to v2.85 released

http://madshi.net/eac3to.zip

* using "eac3to source video.h264" doesn't demux audio/subtitle tracks, anymore
* using "eac3to source movie.*" demuxes video, audio and subtitle tracks
* using "eac3to source 1: video.* 2: audio.*" demuxes the specified tracks
* AC3 and E-AC3 dialnorm removal now uses "-31db" instead of "-0db"
* workaround for DTS files where last byte is missing in each audio frame
* fixed: v2.84 sometimes crashed when parsing HD DVD XML files
* fixed: v2.84 sometimes chose incorrect XML file
* fixed: v2.84 sometimes chose wrong m2ts playlist file
* fixed: some actions were eventually applied twice when "-2pass" was used
* fixed: AAC encoding quality "quality=0.0x" was passed to Nero as "0.x"

On a side note, I looked into making Nero 9 DirectShow decoders work with eac3to, but I gave up on it because of these reasons:

(1) AC3 decoding works, but gives slightly different results compared to Nero 7 and Sonic.
(2) E-AC3 decoding is not supported at all, as far as I can say (which kinda makes sense, since Nero 9 doesn't support HD DVD, anymore).
(3) The decoder didn't accept my TrueHD data. Not sure, maybe they changed something and maybe it would be possible to make it work again with more effort.

All in all I just don't see the value of making Nero 9 work, since E-AC3 support seems to be missing, anyway. Instead I'll try to make the ArcSoft (E-)AC3 decoder work, sooner or later...

[mrr19121970]

Great, Atonment HD-DVD works again now.

Thanks.

[nautilus7]

Thanks man. Happy holidays!

Adding support for Arcsoft EAC3 decoder will be great. I will get rid of Nero at last.

[AnryV]

There is DTS audio track extracted from BD by using eac3to.

Info from eac3to:
DTS, 5.1 channels, 1:59:41, 24 bits, 768kbps, 48khz

Info from Delaycut:
====== INPUT FILE INFO ========================
File is dts
Bitrate (kbit/s) 768
Act rate (kbit/s) 754.500
File size (bytes) 689363968
Channels mode C+L+R+SL+SR
Sampling Frec 48000
Low Frec Effects LFE: Present
Duration 02:01:49.359
Frame length (ms) 10.857522
Frames/second 92.102051
Num of frames 673207
Bytes per Frame 1024.0000
Size % Framesize 0
CRC present: NO
=============================================

As consequence at attempt to connect this sound as external unsync appears. If to include it in mkv that everything is all right.
What's wrong with this file and how this can be corrected?

[Thunderbolt8]

Originally Posted by madshi

eac3to v2.85 released

thanks!

[hubblec4]

Originally Posted by madshi

eac3to v2.85 released


On a side note, I looked into making Nero 9 DirectShow decoders work with eac3to, but I gave up on it because of these reasons:

(1) AC3 decoding works, but gives slightly different results compared to Nero 7 and Sonic.
(2) E-AC3 decoding is not supported at all, as far as I can say (which kinda makes sense, since Nero 9 doesn't support HD DVD, anymore).
(3) The decoder didn't accept my TrueHD data. Not sure, maybe they changed something and maybe it would be possible to make it work again with more effort.

All in all I just don't see the value of making Nero 9 work, since E-AC3 support seems to be missing, anyway. Instead I'll try to make the ArcSoft (E-)AC3 decoder work, sooner or later...

big thanks for your good work on it.

please. dont change to nero9 (i dont like it and i wont install it anymore) nero7, the decoder works!!


hubble

[Snowknight26]

madshi, are you going to update libFLAC to 1.2.1?

[Thunderbolt8]

Originally Posted by Snowknight26

madshi, are you going to update libFLAC to 1.2.1?

which advantages does it have, better compression?

[wolfbane5]

Originally Posted by madshi

v1.0.23 is supposed to work. Try reinstalling with admin rights...

The "-core" switch doesn't need ArcSoft. You only need ArcSoft for best quality decoding...

Okay, I'll reinstall Surcode with admin rights and see what happens. As for DTS-HD decoding, to extract the DTS core, (correct me if I'm wrong) don't you need either Sonic or ArcSoft?

[Snowknight26]

No, eac3to can do that on its own.

[n0mag!c]

Originally Posted by AnryV

There is DTS audio track extracted from BD by using eac3to.

Info from eac3to:
DTS, 5.1 channels, 1:59:41, 24 bits, 768kbps, 48khz

Info from Delaycut:
====== INPUT FILE INFO ========================
File is dts
Bitrate (kbit/s) 768
Act rate (kbit/s) 754.500
File size (bytes) 689363968
Channels mode C+L+R+SL+SR
Sampling Frec 48000
Low Frec Effects LFE: Present
Duration 02:01:49.359
Frame length (ms) 10.857522
Frames/second 92.102051
Num of frames 673207
Bytes per Frame 1024.0000
Size % Framesize 0
CRC present: NO
=============================================

As consequence at attempt to connect this sound as external unsync appears. If to include it in mkv that everything is all right.
What's wrong with this file and how this can be corrected?

Actually, DTS must be 93.75 frames/second. I guess headers was broken using MKV stretch feature, so maybe you'll try to fix this with the same MKV feature?

[AnryV]

Originally Posted by n0mag!c

Actually, DTS must be 93.75 frames/second. I guess headers was broken using MKV stretch feature, so maybe you'll try to fix this with the same MKV feature?

DTS track was ONLY demuxed from .m2ts by eac3to. When I merge it into mkv it play correctly.

How to fix header?

[madshi]

Originally Posted by AnryV

There is DTS audio track extracted from BD by using eac3to.

As consequence at attempt to connect this sound as external unsync appears. If to include it in mkv that everything is all right.
What's wrong with this file and how this can be corrected?

Conventional DTS tracks from HD media (or the core of DTS-HD tracks) are slightly different from normal DTS tracks. The number of bytes per frame varies slightly. This can confuse the DTS source filter. My best guess is that the DTS source filter you're using is at fault.

Originally Posted by Snowknight26

are you going to update libFLAC to 1.2.1?

Can't see anything in the changelog that would make upgrading worthwhile to me.

[AnryV]

Originally Posted by madshi

My best guess is that the DTS source filter you're using is at fault.

What I have to do? I use AC3Filter.

[madshi]

Originally Posted by AnryV

What I have to do? I use AC3Filter.

Don't know. Probably AC3Filter is doing something wrong. I'm not responsible for every HTPC software out there...

[AnryV]

Originally Posted by madshi

Conventional DTS tracks from HD media (or the core of DTS-HD tracks) are slightly different from normal DTS tracks. The number of bytes per frame varies slightly.

Number of bytes per frame may varies but Frames per second connected with duration.

[n0mag!c]

Originally Posted by AnryV

What I have to do? I use AC3Filter.

Try DTSSource.ax filter maybe it helps you:
http://www.doom9.org/Soft21/Filters/test.rar

[Atak_Snajpera]

All in all I just don't see the value of making Nero 9 work, since E-AC3 support seems to be missing, anyway. Instead I'll try to make the ArcSoft (E-)AC3 decoder work, sooner or later...

Is something wrong with libav that we need support for other decoders?

[nautilus7]

Nero and ArcSoft use Dolby's reference code for their decoders, so they might offer better sound quality, but the difference should be negligible, if any.

[magic144]

hi Madshi,
just wondering if possible in a future update, can the audio edit function be used to insert silence for a +delay, rather than looping the audio at that point - would be superb if that could be done for at least AC3 and DTS

specifically, this fn from 2.84 rel notes...

* new option for removing or looping audio data, e.g. "-edit=0:20:47,-100ms"

cheers and keep up the great work,
m

[wolfbane5]

Originally Posted by Snowknight26

No, eac3to can do that on its own.

Had no idea eac3to could extract the dts core by itself; well that solves my problem there.

madshi, I was curious as to why there isn't support for .m2ts and .ts outputs? Demuxing video or simply creating an .mkv is great (from a blu-ray disc or whatnot), but sometimes creating a .m2ts or .ts is handy. Just a suggestion. Thanks for the great work thus far

[yesgrey]

madshi,
You are still including eac3to 2.80 in the pack. Is 2.85 not yet stable enough?

[The_Keymaker]

Hello madshi and forum members,

I've been away for awhile but am now back to archiving my movie collection. However, I have a problem.

I tried to decode the 7.1 DTS-HD audio of the Blu-Ray disc "The Golden Compass" and received an error message about the Arcsoft decoder. I ran eac3to -test and received the dreaded "Arcsoft DTS decoder does not seem to be installed".

I have the LATEST purchased (non-trial) version of Total Media Theater (v 2.1.6.128) and my version of the dtsdecoderdll.dll is 1.1.0.0. I've read about some issues with the Arcsoft decoder and eac3to (on page 267 and 268 of this thread) but found no solutions.

I am running on Windows Vista 32 and i am using eac3to version 2.84.

Can you (or anyone else) offer any help?

Thanks
The_Keymaker

[Chumbo]

Originally Posted by The_Keymaker

Hello madshi and forum members,

I've been away for awhile but am now back to archiving my movie collection. However, I have a problem.

I tried to decode the 7.1 DTS-HD audio of the Blu-Ray disc "The Golden Compass" and received an error message about the Arcsoft decoder. I ran eac3to -test and received the dreaded "Arcsoft DTS decoder does not seem to be installed".

I have the LATEST purchased (non-trial) version of Total Media Theater (v 2.1.6.128) and my version of the dtsdecoderdll.dll is 1.1.0.0. I've read about some issues with the Arcsoft decoder and eac3to (on page 267 and 268 of this thread) but found no solutions.

I am running on Windows Vista 32 and i am using eac3to version 2.84.

Can you (or anyone else) offer any help?

Thanks
The_Keymaker

Not sure if it was here, but awhile back someone had an issue with the purchased version and wound up installing the trial first and then updating or something like that which then worked. It may be in this thread or elsewhere. I'm sorry I can't remember where I saw this.

[The_Keymaker]

Hello Chumbo! Good to see you here.

I re-read the older posts and found the solution. I had to add the the following to my path statement:

C:\Program Files\Common Files\ArcSoft\Bin

However, since this issue is prevalent and the solution arcane, I would humbly suggest madshi make note of it in his fist post.

Regards,
The_Keymaker

[madshi]

Originally Posted by Atak_Snajpera

Is something wrong with libav that we need support for other decoders?

Originally Posted by nautilus7

Nero and ArcSoft use Dolby's reference code for their decoders, so they might offer better sound quality, but the difference should be negligible, if any.

Yes, my main reason for supporting other decoders is that they're reference decoders and they do output slightly different results compared to libav, although I won't dare to judge whether there's any audible difference. The one thing which makes me prefer the reference decoders is that I can be (relatively) safe that the decoding will be bug free, while with libav there may be a higher danger of decoding bugs...

Originally Posted by magic144

just wondering if possible in a future update, can the audio edit function be used to insert silence for a +delay, rather than looping the audio at that point - would be superb if that could be done for at least AC3 and DTS

That would be possible to do, but regardless of whether we loop audio or add silence, we still should search for an edit point where audio volume is low and nobody is talking etc, right? So if we have such an edit point, wouldn't looping audio actually be preferred? I've found that in the edits I've done so far, looping audio most often resulted in a "perfect" result. I usually couldn't even find the edit points after the sync was complete. Of course this only works that well if you choose a good edit point (low volume, no talk, see above). So based on my experience I really don't see a need for an option to include silence - unless we're talking about edits that are bigger than just a few hundred milliseconds.

Your opinion?

Originally Posted by wolfbane5

Had no idea eac3to could extract the dts core by itself; well that solves my problem there.

Generally eac3to does most of the processing on its own. There are only a few exceptions where externally installed components are needed. The most important being MKV muxing and full audio decoding of some audio formats (not all).

Originally Posted by wolfbane5

I was curious as to why there isn't support for .m2ts and .ts outputs?

Because it's really difficult to implement in a proper way.

Originally Posted by yesgrey3

You are still including eac3to 2.80 in the pack. Is 2.85 not yet stable enough?

Up until v2.84 there still have been new bugs found which didn't exist in v2.80 yet. So v2.80 did have a purpose at least until now. Hopefully v2.85 is now better than v2.80 in every way. But we'll have to wait and see about that. If no new bugs show up this week, I might delete v2.80 from the zip in the next release.

Originally Posted by The_Keymaker

I re-read the older posts and found the solution. I had to add the the following to my path statement:

C:\Program Files\Common Files\ArcSoft\Bin

However, since this issue is prevalent and the solution arcane, I would humbly suggest madshi make note of it in his fist post.

Will do that...

[mrr19121970]

Originally Posted by madshi

Because it's really difficult to implement in a proper way.

We use tsmuxer to build the streams back together again at the end. I've tried to automate this as best I can here:

http://forum.slysoft.com/showthread.php?t=24388

but it would be super if (in the absence of direct .m2ts & .ts support) that you could find a way to dynamically build the meta file based on:

Chapter List
Video Type
Audio Type & Language

Thanks, Mike.

[madshi]

Originally Posted by mrr19121970

We use tsmuxer to build the streams back together again at the end. I've tried to automate this as best I can here:

http://forum.slysoft.com/showthread.php?t=24388

but it would be super if (in the absence of direct .m2ts & .ts support) that you could find a way to dynamically build the meta file based on:

Chapter List
Video Type
Audio Type & Language

The problem with this approach is that you'd have to tell eac3to exactly which tracks you want to have stored into the final m2ts file. If there are commentary tracks and maybe multiple audio dubs, the eac3to command line would be monstrous. Currently I'm usually simply doing "eac3to source movie.mkv" and then I'm deleting all audio tracks and subtitle tracks I don't need. Doing this in one step is difficult. If there are multiple commentary tracks, I even test all of them and choose only those which seem to be worthwhile to me. Such a thing could only be done in one step if there was a way to make all decisions before doing the all-in-one-step-processing. Maybe for you it works that way, if you are not interested in testing audio tracks before deciding. But for me it doesn't work that way, because I need to test audio tracks before deciding...

[mrr19121970]

yes, for me it's quite simple. my command line is generally:

eac3to q:\ 1) 2: video.* 3: english.ac3 4: german.ac3

it's only when that deviates (say english only, german & french) that things get tricky. in this case eac3to already knows what format for the video & audio, what names have been chosen and the language.

i was thinking of some sort of scraping script, but at the moment i can live as is.

[nautilus7]

Originally Posted by madshi

That would be possible to do, but regardless of whether we loop audio or add silence, we still should search for an edit point where audio volume is low and nobody is talking etc, right? So if we have such an edit point, wouldn't looping audio actually be preferred? I've found that in the edits I've done so far, looping audio most often resulted in a "perfect" result. I usually couldn't even find the edit points after the sync was complete. Of course this only works that well if you choose a good edit point (low volume, no talk, see above). So based on my experience I really don't see a need for an option to include silence - unless we're talking about edits that are bigger than just a few hundred milliseconds.

Your opinion?

I totally agree. An option for adding silence would be nice for such (longer) edits.

[The_Keymaker]

Hello madshi,

My main backup server is nearly Identical (file and program locations) to my workstation but when i run eac3to -test it tells me Surcode, MKVtoolNix and Arcsoft don't seem to be installed. It tells me Arcsoft isn't installed even after I modified my Path environment variable.

Can you tell me how eac3to checks for the presence of these programs so I can modify my Path environment and directories so eac3to can find them?

Thanks!
The_Keymaker

[htpc@rischer.at]

There ist a bug in 2.85 obviously: look at subtitle 19, the bolded lines

E:\..\EAC3TO>eac3to j: 3: oc3.vc1 4: oc3_eng.ac3 6: oc3_deu.ac3 19: oc3.sup
EVO, 1 video track, 4 audio tracks, 12 subtitle tracks, 2:02:09
"Feature Presentation"
1: Joined EVO file
2: Chapters, 34 chapters without names
3: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
4: E-AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
5: E-AC3, French, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
6: E-AC3, German, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
7: E-AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
"Commentary"
8: Subtitle, English
9: Subtitle, French
10: Subtitle, German
11: Subtitle, German, "SDH"
12: Subtitle, Dutch, "Dutch"
13: Subtitle, Danish, "Danish"
14: Subtitle, Finnish, "Finnish"
15: Subtitle, Norwegian, "Norwegian"
16: Subtitle, Swedish, "Swedish"
17: Subtitle, English, "Forced"
18: Subtitle, French, "Forced"
19: Subtitle, German, "Forced"
Track 8 is used for destination file "oc3.sup".
[a04] The Nero decoder doesn't seem to work, will use libav instead.
[a06] The Nero decoder doesn't seem to work, will use libav instead.
[v03] Extracting video track number 3...
[v03] Removing VC-1 pulldown...
[a06] Extracting audio track number 6...
[a04] Extracting audio track number 4...
[a04] Removing E-AC3 dialog normalization...
[a04] Decoding with libav/ffmpeg...
[a04] Remapping channels...
[a04] Encoding AC3 <640kbps> with libAften...
[a06] Removing E-AC3 dialog normalization...
[a06] Decoding with libav/ffmpeg...
[a06] Remapping channels...
[a06] Encoding AC3 <640kbps> with libAften...
[v03] Creating file "oc3.vc1"...
[a04] Creating file "oc3_eng.ac3"...
[a06] Creating file "oc3_deu.ac3"...
[s08] Extracting subtitle track number 8...
[s08] Creating file "19oc3.sup"...

[The_Keymaker]

A little progress...

For Arcsoft, I had to add the following to the environment path: C:\Program Files\Common Files\ArcSoft\Bin

AND

Manually register filters AsAudio.ax, AsAudioHD.ax and AsAudioEffectHD.

The_Keymaker

[magic144]

@madshi
regarding silence edit/insertion - I agree, these insert points should be short and at 'quiet' moments - I've only ever had to use such a feature when the audio has been b0rked, e.g. by a bad TV commercial edit or some other error/glitch

I happened to hear from Tebasuna51 about this and it was mentioned that to insert silence requires lots of individual samples at all different rates and encoding types to have 'at hand' for use by the tool, so it wasn't straightforward, whereas looping merely requires duplicating what is already there, so I think it would be a bit of an effort to do for now - I guess delaycut can do the job, it was just a "nice to have included in eac3to" option in my way of thinking - if it could all be automated in the one tool (and eac3to is rapidly becoming the swiss-army-knife of all tools!)

thanks for giving it some attention anyway - bottom line is I use it quite infrequently and can use delaycut instead!

[wolfbane5]

The_Keymaker: I've been having the same issue as you - surcode, mkvtoolnix and arcsoft aren't being recognized by eac3to. madshi offered this advice in an earlier post if you happened to miss it:

Originally Posted by madshi

eac3to gets the mkvtoolnix path from one of these registry values:

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

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

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

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

madshi also mentioned that re-installing the programs with admin rights may solve the issue.

Originally Posted by madshi

making the ArcSoft DTS decoder work in eac3to:

Sometimes the ArcSoft DTS decoder doesn't work even after you've installed the full retail software. This can usually be fixed by manually adding the ArcSoft "Bin" folder (e.g. "C:\Program Files\Common Files\ArcSoft\Bin") to your environment search "path".

How do I do this?

[madshi]

Originally Posted by magic144

regarding silence edit/insertion - I agree, these insert points should be short and at 'quiet' moments

That's not what I said. If you need long edits then you need long edits. "Should be" or not doesn't matter...

Originally Posted by magic144

I happened to hear from Tebasuna51 about this and it was mentioned that to insert silence requires lots of individual samples at all different rates and encoding types to have 'at hand' for use by the tool, so it wasn't straightforward, whereas looping merely requires duplicating what is already there, so I think it would be a bit of an effort to do for now

Actually adding in silence instead of looping is not difficult at all (technically). My only question is whether it actually makes sense. Because I think (for short edits at least) looping gives better results. So why adding an option for an inferior solution? I don't like adding options to eac3to, unless there's a really good reason for that...

Originally Posted by magic144

bottom line is I use it quite infrequently and can use delaycut instead!

Again you're misunderstanding me. I have not declined your request (yet). I've just posted my opinion that looping is better than silence. If you find a good argument for why adding silence instead of looping would be better then I'm all ears...

Originally Posted by wolfbane5

How do I do this?

Search for "environment path" on google.

[magic144]

@madshi,
sorry for any confusion - I only meant to convey that I use it *very* infrequently and I can't think of a compelling reason to add it into eac3to vs. using delaycut/other - except my own laziness (which I freely admit was my main reason for proposing it) in having to use 2+ tools instead of 1 :-)
...others who may be interested in this facility may want to add their 2 cents if they want it for specific reasons

looping is perhaps not as good as silence insertion in long edits where you have to otherwise loop so many packets as to end up looping preceding/following non-silent packets in the insert, but like I said, this is only useful if the audio is quite badly b0rked and is therefore a very infrequent occurrence...
(specifically, I was playing around with a broken TV episode yesterday which needed a +200ms adjustment at one point, and the eac3to edit looping thing just didn't avoid repeating audible stuff around the gap - in the end I used Audition to insert silence at the exact (short but already silent) point in the stereo signal and took the 'lossy' hit and reencoded the ABR mp3)

so there it is - I am but one user of your fabulous free product, so I leave it to others to add their voices as to if they think it would be something really worthwhile and why

thanks!
m

[madshi]

Can we find a specific edit size where looping stops being useful? E.g. everything smaller than 1 second: Looping. Everything larger than 1 second: Silence? I'm asking because I'm trying to avoid adding another option. Every additional option makes using eac3to more complicated. But if there's no specific edit size we can agree on, then maybe adding an option for silence is necessary...

[mrr19121970]

One of the the guys here:

http://forum.slysoft.com/showthread....122#post159122

is wanting to convert DTS-Express. there is nothing about it on page #1 here, so I just tried it. It doesn't work with libav/ffmpeg. i was wondering if this was by design ?

regards. mike.

[wolfbane5]

Originally Posted by madshi

making the ArcSoft DTS decoder work in eac3to:

Sometimes the ArcSoft DTS decoder doesn't work even after you've installed the full retail software. This can usually be fixed by manually adding the ArcSoft "Bin" folder (e.g. "C:\Program Files\Common Files\ArcSoft\Bin") to your environment search "path".

So I googled it but I'm still not quite clear on what I'm doing. I just haven't used the command prompt that much yet, hence why I'm not that familiar with it. My best guess would be:

PATH=C:\Program Files\Common Files\ArcSoft\Bin ?

[rebkell]

Originally Posted by wolfbane5

So I googled it but I'm still not quite clear on what I'm doing. I just haven't used the command prompt that much yet, hence why I'm not that familiar with it. My best guess would be:

PATH=C:\Program Files\Common Files\ArcSoft\Bin ?

On an XP machine, go to My Computer -> View System Information -> [Advanced Tab] -> Environment Variables
Highlight the Path Variable and click the edit button, at the end of the others items in the path, add
;C:\Program Files\Common Files\ArcSoft\Bin

the semicolon(; ) separates the other directories in the path

[The_Keymaker]

ThanKs to Wolfbane, chumbo and madshi eac3to can recognize all the necessary filters. Now I have two questions.

1. I can decode and playback a demuxed DTS-HD audio track using Zoom Player and the Arcsoft AudioDecoderHD directshow filter. My question is, can I remux this audio track into MKV with demuxed video? Does MKVtoolNix support this? If this is possible, then there is no need to decode this track into FLAC to get lossless audio during playback.

2. When playing back the DTS-HD audio track using Zoom Player, the ArcSoft directshow filter seems to default to 2 channel and I have to pull up the filter's properties and select 5.1 or 7.1 channels. Is there a way to automatically enable this filter to output 5.1 or 7.1 as default?

Thanks in advance.

The_Keymaker

[madshi]

Originally Posted by mrr19121970

One of the the guys [...] is wanting to convert DTS-Express. there is nothing about it on page #1 here, so I just tried it. It doesn't work with libav/ffmpeg. i was wondering if this was by design ?

DTS-Express has no core. It consists of DTS-HD style blocks, only. Because of that libav can't decode it. I don't remember right now which decoders were able to decode it. I think ArcSoft could. Not sure about Sonic and Nero. You can search back in this thread for "Express". A few months ago I reported which decoder can do it and which not.

Originally Posted by The_Keymaker

1. I can decode and playback a demuxed DTS-HD audio track using Zoom Player and the Arcsoft AudioDecoderHD directshow filter. My question is, can I remux this audio track into MKV with demuxed video? Does MKVtoolNix support this? If this is possible, then there is no need to decode this track into FLAC to get lossless audio during playback.

The latest mkvtoolnix supports muxing DTS-HD. However, I've been told that when using the ArcSoft DirectShow filters, you get downgraded audio (max 48khz, 16bit). Haven't tested that myself yet, though.

Originally Posted by The_Keymaker

2. When playing back the DTS-HD audio track using Zoom Player, the ArcSoft directshow filter seems to default to 2 channel and I have to pull up the filter's properties and select 5.1 or 7.1 channels. Is there a way to automatically enable this filter to output 5.1 or 7.1 as default?

Not that I knew of.

[madshi]

Originally Posted by htpc@rischer.at

There ist a bug in 2.85 obviously: look at subtitle 19, the bolded lines

E:\..\EAC3TO>eac3to j: 3: oc3.vc1 4: oc3_eng.ac3 6: oc3_deu.ac3 19: oc3.sup
EVO, 1 video track, 4 audio tracks, 12 subtitle tracks, 2:02:09
"Feature Presentation"
8: Subtitle, English
19: Subtitle, German, "Forced"
Track 8 is used for destination file "oc3.sup".
[s08] Extracting subtitle track number 8...
[s08] Creating file "19oc3.sup"...

I can't reproduce this problem with the command line you've used. However, I can reproduce it if you do "19:oc3.sup" (without the space after the ":"). That is a bug and will be fixed in the next build.

[yesgrey]

Originally Posted by The_Keymaker

... then there is no need to decode this track into FLAC to get lossless audio during playback.

Originally Posted by madshi

However, I've been told that when using the ArcSoft DirectShow filters, you get downgraded audio (max 48khz, 16bit). Haven't tested that myself yet, though.

Also, if you decode the track into FLAC, I believe you will get a smaller file. At least, from the few tracks I have done it, the size was always smaller...

[Thunderbolt8]

what am I doing wrong, I try to make a 1 minute cut at 25 seconds of an audio track, but its not accepted
"eac3to bla.ac3 blub.ac3 -edit=0:00:25,-60000ms"

->invalid edit format "edit=0:00:25,-60000ms".

when I try it with 0:25:00 (=at 25mins) then its fine.

[madshi]

Originally Posted by Thunderbolt8

what am I doing wrong, I try to make a 1 minute cut at 25 seconds of an audio track, but its not accepted
"eac3to bla.ac3 blub.ac3 -edit=0:00:25,-60000ms"

->invalid edit format "edit=0:00:25,-60000ms".

when I try it with 0:25:00 (=at 25mins) then its fine.

Hmmmm... The cut works different than you seem to expect. You want the cut to *begin* at 0:00:25. eac3to's editing logic works in such a way that a positive edit *begins* at the specified runtime, while a negative edit *ends* at the specified runtime. This logic is inherited from the gap/overlap editing where it makes more sense this way. But I understand that for manual edits it would make more sense to always let the cut *begin* at the specified runtime, regardless of whether it's positive or negative? I'd have to change that, then. Your opinions?

For now cutting 60 seconds at 0:00:25 simply doesn't work, cause the begin of the cut would be at runtime -35 seconds. So use 0:01:25 and you'll get what you're looking for.

[rebkell]

Originally Posted by madshi

Hmmmm... The cut works different than you seem to expect. You want the cut to *begin* at 0:00:25. eac3to's editing logic works in such a way that a positive edit *begins* at the specified runtime, while a negative edit *ends* at the specified runtime. This logic is inherited from the gap/overlap editing where it makes more sense this way. But I understand that for manual edits it would make more sense to always let the cut *begin* at the specified runtime, regardless of whether it's positive or negative? I'd have to change that, then. Your opinions?

For now cutting 60 seconds at 0:00:25 simply doesn't work, cause the begin of the cut would be at runtime -35 seconds. So use 0:01:25 and you'll get what you're looking for.

I would vote for the cut to begin at the specified runtime, it would be more intuitive and seem more natural(IMO), to pick my start point and then determine I want to cut out so and so time from this point on. Of course that's just my opinion. One reason being that I can tell that the a/v sync is off at that point, and I can just nibble away until I get what I'm looking for.

[madshi]

eac3to v2.86 released

http://madshi.net/eac3to.zip

* fixed: "1:some.ac3" instead of "1: some.ac3" failed for 2 digit track numbers
* fixed: "eac3to source movie.mkv" demuxed video instead of muxing to MKV
* negative edit now begins at the specified runtime instead of ending there

v2.80 is no longer contained in the zip file.

[Greif]

Thanks again madshi for all your hard work.

[Snowknight26]

Originally Posted by madshi

Because of that libav can't decode it.

Even though libavcodec can decode DTS-HD?

http://svn.ffmpeg.org/ffmpeg/trunk/libavcodec/
r15951 of dca.c, dca.h, dca_parser.c -> 'Add support for parsing and decoding DCA-HD streams.'

[Thunderbolt8]

Originally Posted by madshi

eac3to v2.86 released

http://madshi.net/eac3to.zip

* fixed: "1:some.ac3" instead of "1: some.ac3" failed for 2 digit track numbers
* fixed: "eac3to source movie.mkv" demuxed video instead of muxing to MKV
* negative edit now begins at the specified runtime instead of ending there

v2.80 is no longer contained in the zip file.

[madshi]

Originally Posted by Snowknight26

Even though libavcodec can decode DTS-HD?

http://svn.ffmpeg.org/ffmpeg/trunk/libavcodec/
r15951 of dca.c, dca.h, dca_parser.c -> 'Add support for parsing and decoding DCA-HD streams.'

They mean decoding the core of DTS-HD streams.

[Thunderbolt8]

hm I not sure whether I understand the cutting process correctly. I want to cut 1 min away at 25 seconds at the beginning of the movie, so that the part, which actually begins at 1:25min now begins at 0:25 min. but when I tell eac3to to cut at 0:00:25, then the part till 1:25 is unchanged, but the part from 2:25 now begins at 1:25 :S

what am I doing wrong?

edit: nvm, got around my problem, was a part of a DVD repeating itself, apparently because of too long runtime, because of angles

[joerg-daniel]

eac3to v2.86
command line: "G:\bluray\eac3\eac3to.exe" "demuxed\english-comment2_11_audio.dts-es" "\test.ac3"
------------------------------------------------------------------------------
DTS Express, 1.0 channels, 1:26:10, 16 bits, 96kbps, 48khz
Decoding with ArcSoft DTS Decoder...
The ArcSoft DTS Decoder reported an error while decoding.
Aborted at file position 16384.
the file is 60 mb large

it was me asking for dts express

- how can I use the libavcodec with dts-hd support mentioned some posts ago?

[Thunderbolt8]

another question, when demuxing an audio track from a dvd with dvddecypter I got that line:
"0x81 - Audio - AC3 / 6ch / 48kHz / DRC / Deutsch / LBA: 19 / PTS: 00:00:00.287 / Delay: 0ms"

so does this mean now that DRC has been applied to that track? if yes, can eac3to remove it? from what I see is that dialnorm gets removed, but theres then only written that DRC is disabled for nero decoder so that it doesnt get applied when the nero decoder is processing the track. but does this mean as well that DRC, if already applied on that track before, can be removed again?

[madshi]

Originally Posted by Thunderbolt8

what am I doing wrong?

Nothing, it was my fault. Actually v2.85 did editing correctly, it just was too strict with the parameters. Then in v2.86 I fixed the parameter checking, but broke the editing. Argh... Another new build is coming...

Originally Posted by joerg-daniel

DTS Express, 1.0 channels, 1:26:10, 16 bits, 96kbps, 48khz
Decoding with ArcSoft DTS Decoder...
The ArcSoft DTS Decoder reported an error while decoding.
Aborted at file position 16384.

it was me asking for dts express

See this post:

http://forum.doom9.org/showthread.ph...06#post1154106

So seemingly Nero is the best decoder for DTS-Express.

Originally Posted by joerg-daniel

how can I use the libavcodec with dts-hd support mentioned some posts ago?

Not at all. libav CAN NOT decode DTS-HD. It can only decode the core of DTS-HD tracks. The only thing that has changed is that libav can now skip over DTS-HD data blocks, so that the core is properly decoded. That doesn't help with DTS-Express because DTS-Express consists of nothing but DTS-HD data blocks!

[madshi]

Originally Posted by Thunderbolt8

so does this mean now that DRC has been applied to that track?

I don't know exactly what DVDDecrypter means, but it definitely does not mean that DRC has already been applied. Probably it means that the AC3 track contains information about how to apply DRC, if you turn that feature in the decoder. So there's no reason to worry...

[Thunderbolt8]

Originally Posted by madshi

I don't know exactly what DVDDecrypter means, but it definitely does not mean that DRC has already been applied. Probably it means that the AC3 track contains information about how to apply DRC, if you turn that feature in the decoder. So there's no reason to worry...

DVD Decryper, that program to rip DVDs (development discontinued since a few years already)

[madshi]

Originally Posted by Thunderbolt8

DVD Decryper, that program to rip DVDs (development discontinued since a few years already)

I know what DVDDecrypter is. I meant that I don't know what DVDDecryptor wants to say by adding the text "DRC" to the demuxed audio file.

[madshi]

eac3to v2.87 released

http://madshi.net/eac3to.zip

* fixed: negative edit was done too late (introduced in v2.86)

[Thunderbolt8]

thanks!

[Kurtnoise]

Originally Posted by madshi

I meant that I don't know what DVDDecryptor wants to say by adding the text "DRC" to the demuxed audio file.

it's just an info from the IFO file...nothing is applied to the track itself.

[madshi]

Originally Posted by Kurtnoise13

it's just an info from the IFO file...nothing is applied to the track itself.

Ah, thanks.

[wolfbane5]

Originally Posted by madshi

Surcode is located by checking these registry values:

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

Again, if you have properly installed Surcode, these registry values should be set. Re-installing the programs with admin rights may solve the issue.

I reinstalled arcsoft via an update with admin rights and eac3to now recognizes it. It still doesn't recognize surcode is installed despite installing with admin rights. Any ideas?

[The_Keymaker]

Originally Posted by yesgrey3

Also, if you decode the track into FLAC, I believe you will get a smaller file. At least, from the few tracks I have done it, the size was always smaller...

I just tried this and can confirm the FLAC audio track file is smaller then the demuxed DTS-HD audio track file. Why is this? Is the DTS-HD container/compression scheme less efficient than FLAC? Or, is there something being lost in the FLAC conversion process?

Regards
The_keymaker

[nautilus7]

FLAC is more efficient. Also, don't forget that DTS-HD Master Audio can't be less than 1536kbps because of the DTS core inside, which reduces efficiency.

[The_Keymaker]

Originally Posted by nautilus7

FLAC is more efficient. Also, don't forget that DTS-HD Master Audio can't be less than 1536kbps because of the DTS core inside, which reduces efficiency.

Thanks Nautilus.

[rica]

Also my latest trials proved the only way of listening lossless audio formats as is (24 bit) over analog cables on conventional receivers is transcoding them to flac via eac3to.
Thanks to madshi again; hope you will make the madecoder filter someday which allows us to listen the lossless audio as is.

[The_Keymaker]

Based on suggestions from forum members, I've decided to mux (into MKV) the FLAC audio track instead of the DTS-HD audio track (which seemed to not playback very well in Zoom Player anyway).

One question though, in the past when I muxed my HD-DVD into MKV, I used a timecode file in MKVToolnix (23.976). Is it still necessary to do this?

If so should it be for the audio track or the video track or both?

Thanks!
The_Keymaker

[nautilus7]

If you let eac3to mux the video (and automatically remove pulldown) into mkv you don't need anything more. Just do a second manual pass with mkvmerge to mux audio, etc (again with no timecode).

[cavediver]

I'm working on the Jethro Tull Live in Montreux Blu-Ray. The movie is contained in two m2ts files one of which is only 00:00:01 in length. The movie contains one video file and three audio tracks: DTS-HD MA 5.1, AC3 5.1, and LPCM 2.0. The movie is h264/AVC, 1080i at 29.970 fps. It's not playing real smooth through the Popcorn Hour. If I use eac3to to change the frame rate of movie from 29.970 to 23.976 will I need to do anything to the audio tracks to make them sync with the new frame rate, or will they be ok native? Or do I just need to demux the movie to a raw h264 file and let eac3to fix whatever pulldown or stuff is going on with the video file?

[rica]

The best way i'd suggest is to:
1) Demux to elementary streams (including transcoding to flac) with eac3to.
2) Remux to mkv with MkvMergeGui 2.4.1.

Originally Posted by The_Keymaker

Based on suggestions from forum members, I've decided to mux (into MKV) the FLAC audio track instead of the DTS-HD audio track (which seemed to not playback very well in Zoom Player anyway).

[The_Keymaker]

Thanks Nautilus and Rica. That is what I'll do.

[rica]

Originally Posted by cavediver

I'm working on the Jethro Tull Live in Montreux Blu-Ray. The movie is contained in two m2ts files one of which is only 00:00:01 in length. The movie contains one video file and three audio tracks: DTS-HD MA 5.1, AC3 5.1, and LPCM 2.0. The movie is h264/AVC, 1080i at 29.970 fps. It's not playing real smooth through the Popcorn Hour. If I use eac3to to change the frame rate of movie from 29.970 to 23.976 will I need to do anything to the audio tracks to make them sync with the new frame rate, or will they be ok native? Or do I just need to demux the movie to a raw h264 file and let eac3to fix whatever pulldown or stuff is going on with the video file?

In my experience, especially concert BD Disks are natively interlaced; so you can't remove pulldown.
If there is a pulldown, eac3to will remove it.
So the best way; first demux the video via eac3to and check the frame rate with Mediainfo after demuxing.
If nothing has changed, it is clear the movie is nativelly interlaced and nothing to do with it.
If it returns to 23.96, that means you've removed the pulldown and no need to make any change on audio since the the original is 23.96 already.
But if you change it to 25 fps you have to use speedup option in both video and audio while demuxing.

[rack04]

Using eac3to v2.87 I'm having trouble with Ratatouille Blu-ray.

Using the command line:
eac3to "F:\Blu-ray\RATATOUILLE"

Gives me three seemless branches with the same duration. The first 1) is called angle 1, the second 2) is called angle 2, and the third 3) is called angle 3. When I play this disc using PowerDVD I don't have the option to change angles so I wonder if there really are angles on this disc.

Can anyone tell me what the difference is in the various angles?

[Snowknight26]

Play the different m2ts files and see for yourself?

[cavediver]

Originally Posted by rica

In my experience, especially concert BD Disks are natively interlaced; so you can't remove pulldown.
If there is a pulldown, eac3to will remove it.
So the best way; first demux the video via eac3to and check the frame rate with Mediainfo after demuxing.
If nothing has changed, it is clear the movie is nativelly interlaced and nothing to do with it.
If it returns to 23.96, that means you've removed the pulldown and no need to make any change on audio since the the original is 23.96 already.
But if you change it to 25 fps you have to use speedup option in both video and audio while demuxing.

Thanks Rica. You were right. It is natively interlaced at 1080i60. I fixed the problem by setting the output on the PCH to 1080i60.

[wolfbane5]

Originally Posted by rack04

Using eac3to v2.87 I'm having trouble with Ratatouille Blu-ray.

Using the command line:
eac3to "F:\Blu-ray\RATATOUILLE"

Gives me three seemless branches with the same duration. The first 1) is called angle 1, the second 2) is called angle 2, and the third 3) is called angle 3. When I play this disc using PowerDVD I don't have the option to change angles so I wonder if there really are angles on this disc.

Can anyone tell me what the difference is in the various angles?

I haven't seen Ratatouille on bluray so I can't say what each angle is. I'd recommend checking on the bluray itself. The best way to explain angling is this:

ex. If you've seen any of the Star Wars movies, in the beginning is the slanted yellow text. In the DVD Menu, you can choose to watch the movie in Eng/Fr/Spa and in doing so you also choose which version of the text is to be shown when starting the movie. When ripping DVD's to the computer with DVDDecrypter, you can choose which angle is ripped; I'm not as sure with bluray.

[joerg-daniel]

I tried to convert the dts express track

eac3to v2.86
command line: eac3to 2_11_audio.dts audio.wav -nero
------------------------------------------------------------------------------
DTS Express, 1.0 channels, 1:26:10, 16 bits, 96kbps, 48khz
Disabling DRC for Nero (E-)AC3 decoding...
Decoding with DirectShow (Nero Audio Decoder 2)...
The WAV writer didn't receive the format information.
Aborted at file position 62045184.

is this the right audio decoder?
where can i see if the nero hd coder is installed?

[madshi]

Originally Posted by The_Keymaker

I just tried this and can confirm the FLAC audio track file is smaller then the demuxed DTS-HD audio track file. Why is this? Is the DTS-HD container/compression scheme less efficient than FLAC?

It is always this way. FLAC is usually noticably smaller than DTS-HD Master Audio. The reason is simply that DTS wastes space by using a lossy core. That makes things less efficient. FLAC is a much simpler approach without any lossy components, which makes it much more effective. The DTS approach makes some sense, too, though, mainly because the lossy DTS core can be transported over SPDIF, while multichannel FLAC can't.

Originally Posted by The_Keymaker

One question though, in the past when I muxed my HD-DVD into MKV, I used a timecode file in MKVToolnix (23.976). Is it still necessary to do this?

No. Muxing HD DVD to MKV is 100% the same compared to Blu-Ray, when letting eac3to doing all the dirty work. No special tricks needed at all.

Originally Posted by rica

The best way i'd suggest is to:
1) Demux to elementary streams (including transcoding to flac) with eac3to.
2) Remux to mkv with MkvMergeGui 2.4.1.

That is NOT the best way. To my best knowledge mkvtoolnix' elementary stream muxing is inferior compared to eac3to. I've found several bugs in mkvtoolnix' elementary stream muxing, none of which are present in eac3to. If you want to end up with an MKV you should always let eac3to do the video muxing to MKV. The situation may be different when you want to end up with TS or m2ts. Then demuxing to elementary streams may make sense. But not when you want to get an MKV in the end. I've spent a lot of time comparing the different MKV video muxers and all of them have (different) problems. I'm not aware of any current problems with eac3to's MKV video muxing.

Originally Posted by joerg-daniel

I tried to convert the dts express track

eac3to v2.86
command line: eac3to 2_11_audio.dts audio.wav -nero
------------------------------------------------------------------------------
DTS Express, 1.0 channels, 1:26:10, 16 bits, 96kbps, 48khz
Disabling DRC for Nero (E-)AC3 decoding...
Decoding with DirectShow (Nero Audio Decoder 2)...
The WAV writer didn't receive the format information.
Aborted at file position 62045184.

is this the right audio decoder?

Looks like that to me. Can you upload a sample of the original m2ts source file, please? I guess 50MB should do. Alternatively you can upload 10MB of the demuxed DTS Express track. But the original m2ts file might help more...

[yesgrey]

Originally Posted by madshi

If you want to end up with an MKV you should always let eac3to do the video muxing to MKV.

madshi,
I am currently testing x264 encoding to find the best options for my encodings. I am creating h264 raw streams with x264 and then mux them into mkv with mkvmerge 2.4.1. After reading the above, I have tryed muxing with eac3to but it always says: "The format of the source file could not be detected".
I am using eac3to v2.87.

Here is the first 1MB of one of the files, it's a 23.976fps stream:
http://www.megaupload.com/?d=GX84ITRG

Could you take a look at it? Or eac3to does not support x264 h264 raw streams?

Thanks in advance

[madshi]

Originally Posted by yesgrey3

I am currently testing x264 encoding to find the best options for my encodings. I am creating h264 raw streams with x264 and then mux them into mkv with mkvmerge 2.4.1. After reading the above, I have tryed muxing with eac3to but it always says: "The format of the source file could not be detected".

eac3to needs AUDs to be in the stream. That is the case of all demuxed raw h264 streams from Blu-Ray, HD DVD and also from h264 broadcasts. Only self made h264 streams are usually without AUDs. I think x264 has an option for adding AUDs to the stream? If not, h264info can add AUDs, but h264info is quite slow...

[odin24]

Guys, this may seem like a silly question... Does Nero need to be installed in order to transcode an AAC track to AC3?

[Greif]

Originally Posted by madshi

That is NOT the best way. To my best knowledge mkvtoolnix' elementary stream muxing is inferior compared to eac3to. I've found several bugs in mkvtoolnix' elementary stream muxing, none of which are present in eac3to. If you want to end up with an MKV you should always let eac3to do the video muxing to MKV. The situation may be different when you want to end up with TS or m2ts. Then demuxing to elementary streams may make sense. But not when you want to get an MKV in the end. I've spent a lot of time comparing the different MKV video muxers and all of them have (different) problems. I'm not aware of any current problems with eac3to's MKV video muxing.

If I am re-encoding the video into an MP4 container, which should I do? Demux the raw video stream or mux into MKV? Is one option faster/better than the other?

[yesgrey]

Originally Posted by madshi

eac3to needs AUDs... I think x264 has an option for adding AUDs to the stream?

Yes, x264 has that option: --aud
I have encoded a new version and now eac3to recognizes it.

[nurbs]

Originally Posted by Greif

If I am re-encoding the video into an MP4 container, which should I do? Demux the raw video stream or mux into MKV? Is one option faster/better than the other?

Depends on how you do the reencoding. If you use neuron2s indexing software raw will be better because it doesn't support MKV IIRC. If you are using DirectShowSource directly or with graphs mkv will be better since I don't know of any source filters (splitter) that will handle raw vc1 and avc streams.

[rica]

Originally Posted by nurbs

since I don't know of any source filters (splitter) that will handle raw vc1 and avc streams.

One of them is here:


BTW, Happy New Year to you all.

best.

[rebkell]

Originally Posted by yesgrey3

Yes, x264 has that option: --aud
I have encoded a new version and now eac3to recognizes it.

I should know this, but what are AUD's?

[rack04]

Originally Posted by madshi

That is NOT the best way. To my best knowledge mkvtoolnix' elementary stream muxing is inferior compared to eac3to. I've found several bugs in mkvtoolnix' elementary stream muxing, none of which are present in eac3to. If you want to end up with an MKV you should always let eac3to do the video muxing to MKV. The situation may be different when you want to end up with TS or m2ts. Then demuxing to elementary streams may make sense. But not when you want to get an MKV in the end. I've spent a lot of time comparing the different MKV video muxers and all of them have (different) problems. I'm not aware of any current problems with eac3to's MKV video muxing

So using mkvtoolnix to merge the audio with the MKV from eac3to doesn't induce any errors?

[nurbs]

@rica:
I don't have Arcsoft, but good to know, thanks.

Happy new year.

[madshi]

Originally Posted by rebkell

I should know this, but what are AUD's?

Access Unit Delimiters. Bitstream "00 00 01 09 xx" or "00 00 00 01 09 xx". Usually only 6-7 bytes long and there's usually one between every video frame. They're ignored by the decoders, but are useful to find the right "split point" between two video frames.

Originally Posted by rack04

So using mkvtoolnix to merge the audio with the MKV from eac3to doesn't induce any errors?

No. mkvtoolnix leaves all important video muxing properties as they were created by eac3to.

[Chumbo]

Originally Posted by rack04

So using mkvtoolnix to merge the audio with the MKV from eac3to doesn't induce any errors?

Originally Posted by madshi

...No. mkvtoolnix leaves all important video muxing properties as they were created by eac3to.

So, actually the answer is yes, mkvtoolnix doesn't induce errors.

[rica]

So you should workaround to merge mkvs created by eac3to then

Originally Posted by madshi

No. mkvtoolnix leaves all important video muxing properties as they were created by eac3to.

[madshi]

Originally Posted by Chumbo

So, actually the answer is yes, mkvtoolnix doesn't induce errors.

That's one of those funny questions where you can say both "yes" and "no" and both can be interpreted both ways. If you ask me: "So it doesn't do that?" I'd answer: "No, it doesn't" while you wanted to hear "Yes, it doesn't". Probably the best answer would be "Correct, it doesn't". Anyway...

Originally Posted by rica

So you should workaround to merge mkvs created by eac3to then

What do you mean?

[rica]

Originally Posted by madshi

What do you mean?

Considering the best way of creating mkv files is eac3to; think i have more than one mkvs created by eac3to and i want to merge them.
If i try doing this via MkvMergeGui, i'll meet with the bugs.
So how shall i merge them; eac3to can not do this?

Edit: Sorry, i didn't suppose you used "leave" instead of "keep"; so i misundersood. Finally, mkvmerge gui can merge mkvs created by eac3to without any problem? I beg your pardon.

[Chumbo]

Originally Posted by madshi

That's one of those funny questions where you can say both "yes" and "no" and both can be interpreted both ways. If you ask me: "So it doesn't do that?" I'd answer: "No, it doesn't" while you wanted to hear "Yes, it doesn't". Probably the best answer would be "Correct, it doesn't". Anyway...

Yeah, just havin' a little fun. Happy New Year!

[rica]

Originally Posted by odin24

Guys, this may seem like a silly question... Does Nero need to be installed in order to transcode an AAC track to AC3?

Yes, afaik.

[rica]

Madshi,
What is your recipe:
Demux/remux to mkv video and demux audio/transcode to flac with eac3to at the same command line.
And remux to mkv container with MkvMerge Gui.
Is this correct?

[williewonton]

Madshi

In previous posts I was noting problems with the title Game Plan. Over the holiday break I have re-looked at this. M2TS files 00000 to 00091 are the correct files in linear sequence for the title. Files from 00092 and up seem to be part of some interactive features that I haven't looked at closely. Playlist 00139.mpls calls up a mixture of 0..91 plus those from 92 and up (these have the same audio track basically the start up music).

However playlist 00138.mpls does call up the correct linear sequence 0..91 (at least so far as TSMuxer confirms for me.)

Is it possible that eac3to picks the wrong mpls file? Or is it possible to specify the mpls file for eac3to to use?

[madshi]

Originally Posted by rica

What is your recipe:
Demux/remux to mkv video and demux audio/transcode to flac with eac3to at the same command line.
And remux to mkv container with MkvMerge Gui.
Is this correct?

Yes, that's exactly what I'd suggest.

Originally Posted by williewonton

In previous posts I was noting problems with the title Game Plan. Over the holiday break I have re-looked at this. M2TS files 00000 to 00091 are the correct files in linear sequence for the title. Files from 00092 and up seem to be part of some interactive features that I haven't looked at closely. Playlist 00139.mpls calls up a mixture of 0..91 plus those from 92 and up (these have the same audio track basically the start up music).

However playlist 00138.mpls does call up the correct linear sequence 0..91 (at least so far as TSMuxer confirms for me.)

Is it possible that eac3to picks the wrong mpls file? Or is it possible to specify the mpls file for eac3to to use?

If you do "eac3to GamePlanFolder" you'll get a list of titles. eac3to doesn't know which one is the movie title. This information is simply not available. eac3to just sorts all available titles. Titles with highest resolution are listed first. Titles with identical resolution are sorted so that the longest runtime title is listed first. The playlist used for each listed title is displayed by eac3to. So do "eac3to GamePlanFolder" and check which title uses playlist 00138.mpls. It might be 1) or maybe 2) or 3). Then do e.g. "eac3to GamePlanFolder 2) [otherOptions]" to select the wanted title/playlist.

[williewonton]

madshi

OK, thanks for the tips. The approach isn't clear from the documentation, but your advice has helped with clarity.

[Thunderbolt8]

does he -slowdown function actually remove the "EX" from ac3 tracks? I got an ac3 track here which is DD EX 5.1, but when I use -slowdown the track is only normal DD 5.1 after that. or does it maybe only remove the EX flag and the slowed down track is still EX? or was it never EX to begin with and only had the flag, which then correctly got removed at processing?

http://www.sendspace.com/file/13ese3

[nautilus7]

Proper EX decoding results in 6.1ch, but there's is no such decoder available. All we can get is 5.1. As a result, if you decode/slowdown/re-encode the track you can't get the original EX form back.

Happy New Year to everybody.

[madshi]

As far as I understand, the mixed in back channel ("EX") should not get lost when slowing down audio (someone correct me, please, if I'm wrong). However, currently eac3to doesn't *ever* set the EX flag when using Aften to encode an AC3 track. Which means that in every case where audio is reencoded, the EX flag is lost. I have on my to do list to transport the EX flag through to the final AC3 file, but this is not implemented yet...

[saint-francis]

Eac3to struggles with Universal Soldier. It says that there is only an 18 minute 480i track when in reality there is also a full length movie in there too. Here is the entire playlist folder.

[madshi]

Originally Posted by saint-francis

Eac3to struggles with Universal Soldier. It says that there is only an 18 minute 480i track when in reality there is also a full length movie in there too. Here is the entire playlist folder.

I need the CLIPINF folder, too. Thanks!

[saint-francis]

Originally Posted by madshi

I need the CLIPINF folder, too. Thanks!

Here you go.

[williewonton]

@madshi

Many thanks, all sorted

[asarian]

@madshi

I'm getting the following error on my Austin Powers 2 Blu-Ray: "Lossless check failed" (see below). Is that something I can ignore? eac3to seems to indicate success after all, but I'm not sure.

Thanks


--------
eac3to 00002.m2ts 4: c:\video\temp.pcm
M2TS, 1 video track, 3 audio tracks, 1 subtitle track, 1:29:35
1: Chapters, 25 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: AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB
6: Subtitle (PGS), English
[a04] Extracting audio track number 4...
[a04] Extracting TrueHD stream...
[a04] Removing TrueHD dialog normalization...
[a04] Decoding with libav/ffmpeg...
[a04] Swapping endian...
[a04] Remapping channels...
[a04] Creating file "c:\video\temp.24bit.pcm"...
[libav] Lossless check failed - expected 0, calculated 91
[a04] The original audio track has a constant bit depth of 16 bits.
[a04] The zero bytes were successfully removed.
Video track 2 contains 128879 frames.
eac3to processing took 12 minutes, 17 seconds.
Done.

[Thunderbolt8]

might be the same case I had, meaning its only that one instance inside the track which wont be lossless then, while the rest of all other million bytes are or something like that. in other words: you wont notice any difference.

[asarian]

Originally Posted by Thunderbolt8

might be the same case I had, meaning its only that one instance inside the track which wont be lossless then, while the rest of all other million bytes are or something like that. in other words: you wont notice any difference.

Thanks. It looks, indeed, rather benign. I thought that (presumably) the checksum fails because of leading/trailing zeroes (?). But it's good to know I can safely ignore it.

[Weirdo]

Please forgive some '-already-answered-but-can't-search-huge-thread' questions...
I want to convert a DTS-HD MA 5.1 24bit/48KHz track to 24/48 flac from a BD folder/m2ts file.

1. Source reports -8ms, does that mean I've got to add +8ms for any type of destination file, flac included?
2. The blu-ray remapping option (or any other option) isn't needed for this, right?

edit: 3. Are there known issues with different versions of the Arcsoft DTS decoder?

Thanks

[asarian]

Originally Posted by Weirdo

I want to convert a DTS-HD MA 5.1 24bit/48KHz track to 24/48 flac from a BD folder/m2ts file.

1. Source reports -8ms, does that mean I've got to add +8ms for any type of destination file, flac included?

No, that isn't necessary. Ere you'd expect to have to include "-8ms" also, but that isn't necessary either.

2. The blu-ray remapping option (or any other option) isn't needed for this, right?

Nope, don't need it.

N.B. Actually, never seen an occassion where I did need the "-blu-ray" option. I convert TrueHD/DTS-MA -> LPCM without it, at least.

3. Are there known issues with different versions of the Arcsoft DTS decoder?

If you bought an Arcsoft Product, like Total Media Theatre, your product will always upgrade to the latest and greatest anyway. But, far as I know, Arcsoft DTS decoder has no known issues.

[Thunderbolt8]

at least within eac3to it cannot decode DTS-HD 1.0 tracks

[The_Keymaker]

This has probably been answered somewhere in this thread, but Instead of plowing through 380+ pages I'm hoping someone can help me with a simple answer.

How or what do I need to do to the .sup subtitle files extracted by eac3to to get them in a format that I can mux with my audio and video files into .MKV? I believe MKVMerge can handle subtitles in the SRT, SSA, ASS and Vobsub formats (are there others MKVmerge can use?).

Thanks

[madshi]

Originally Posted by saint-francis

Eac3to struggles with Universal Soldier. It says that there is only an 18 minute 480i track when in reality there is also a full length movie in there too.

This is a troublesome playlist configuration. The main movie titles asks for this:

(1) play 00002.m2ts from start to end
(2) play the first second of 00002.m2ts another time

eac3to didn't like this configuration, because usually playlists which ask for the same m2ts part to be played more than once are menu titles.

The next eac3to will "fix" this problem.

[madshi]

Originally Posted by The_Keymaker

How or what do I need to do to the .sup subtitle files extracted by eac3to to get them in a format that I can mux with my audio and video files into .MKV? I believe MKVMerge can handle subtitles in the SRT, SSA, ASS and Vobsub formats (are there others MKVmerge can use?).

There are tools out there to OCR the subtitles and convert them to a text format like SRT. It can be a painful process, sometimes. Look for "SupRip".

[bmnot]

How come eac3to can't read from/work with .mkv files, even if it created it?

[dat720]

madshi

I just want to say Fan-Friggen-Tastic work with eac3to... it's plain awesome, never fails me, i was about to post a question regarding a tricky DTS file i had, it spits the error:
Can't downmix this channel configuration ($60f).
Internal error - unknown audio output format!

But a simple update from 2.77 to 2.87 fixed my probem.... awesome work!!! keep it up.

Do you have a paypal account that i could donate to say thanks for your time and effort?

[yesgrey]

Is this a bug?

eac3to v2.87
command line: eac3to f:\temp\fly_dxva_hd_hq.264 test.mkv
------------------------------------------------------------------------------
h264/AVC, 1920x816 24p /1.001 (40:17)
Muxing video to Matroska...
Added fps value to MKV header.
Video track 1 contains 300 frames.
eac3to processing took 1 second.
Done.

The "(40:17)" is the duration of the file? As you see, the file only has 300 frames.
Maybe it's a bug in x264 AUDs placement?...
You can download the file here:
http://www.megaupload.com/?d=6XTI9UH5

By the way, I have suggested MeGUI authors to automatically set the --aud option in x264 when outputting RAWAVC streams, and they agree do it in the next release.

[nurbs]

The 40:17 is the aspect ratio of the video.

[yesgrey]

Originally Posted by nurbs

The 40:17 is the aspect ratio of the video.

[piratburner]

First time to demux a HD-DVD

Is it ok to use this command
>eac3to h:\TEMP\MOVIE 1) -demux

And the take the files I want and add this to TsMuxer ?

Or have I forgot something :-)

EDIT

I want to do a Blu ray disc :-)

[Steel]

Originally Posted by The_Keymaker

This has probably been answered somewhere in this thread, but Instead of plowing through 380+ pages I'm hoping someone can help me with a simple answer.

How or what do I need to do to the .sup subtitle files extracted by eac3to to get them in a format that I can mux with my audio and video files into .MKV? I believe MKVMerge can handle subtitles in the SRT, SSA, ASS and Vobsub formats (are there others MKVmerge can use?).

Thanks

You can use suprip to read the .sup file.

[The_Keymaker]

Originally Posted by madshi

There are tools out there to OCR the subtitles and convert them to a text format like SRT. It can be a painful process, sometimes. Look for "SupRip".

Yes I'm aware of Supread and Suprip have used them before in the past and it was a PAIN IN THE A**!.

Seems like not much has changed with subtitling tools. It would be nice to just mux in the .sup file and have the appropriate Dshow filter demix and decode it.

Oh well...

[rica]

madshi,

eac3to doesn't recognize this:

http://www.sendspace.com/file/61lfv8

[Zwitterion]

Originally Posted by madshi

I've looked into this. The DIRAC documentation contains this text:


So I contacted the DIRAC company and asked about whether it would make any sense at all to use the free DIRAC version for movie tracks. Here's the reply I received:

"If you are planning on time stretching and pitch shifting 5.1 and 7.1 recordings relative phase is essential. You would need to use the PRO version of DIRAC in order to do this."

In other words: The free DIRAC version is useless for our needs, sadly.

Thank you for your efforts. However, I don't believe that relative phase is as essential as they claim for multi channel processing... IIRC the stereo mode of Waves SoundShifter just applies the mono algorithm on each channel.

Another thing: Is it possible to disable 2pass mode? I'm speeding up a large DTS file, and one pass takes nearly half an hour. I can accept a few clipped samples in this case.

Much better would be 32 or 64 bit floating point wave as output. Clipping would be a non-issue then.

[piratburner]

Originally Posted by piratburner

First time to demux a HD-DVD

Is it ok to use this command
>eac3to h:\TEMP\MOVIE 1) -demux

And the take the files I want and add this to TsMuxer ?

Or have I forgot something :-)

EDIT

I want to do a Blu ray disc :-)

One more thing :-)

I got a eac3 sound track (*.eac3)

Is it OK to make a DTS track of the eac3 sound ??

I have search but not got a answer :-(

[madshi]

Originally Posted by bmnot

How come eac3to can't read from/work with .mkv files, even if it created it?

Because reading is not the same as creating.

Originally Posted by dat720

I just want to say Fan-Friggen-Tastic work with eac3to... it's plain awesome, never fails me [...]

Do you have a paypal account that i could donate to say thanks for your time and effort?

Thank you. Right now there's no way to donate. Maybe I'll make that available later...

Originally Posted by piratburner

First time to demux a HD-DVD

Is it ok to use this command
>eac3to h:\TEMP\MOVIE 1) -demux

And the take the files I want and add this to TsMuxer ?

Generally yes. However, TrueHD tracks will need special treatment. Please ask in the tsMuxeR thread about this. They'll help you there. Also HD DVD style E-AC3 tracks must be transcoded to another format (e.g. AC3 or DTS, whatever you prefer) to be Blu-Ray compatible.

Originally Posted by Zwitterion

Thank you for your efforts. However, I don't believe that relative phase is as essential as they claim for multi channel processing... IIRC the stereo mode of Waves SoundShifter just applies the mono algorithm on each channel.

This might be true. But I'm not really feeling good implementing something which might output inferior results...

Originally Posted by Zwitterion

Another thing: Is it possible to disable 2pass mode? I'm speeding up a large DTS file, and one pass takes nearly half an hour. I can accept a few clipped samples in this case.

Much better would be 32 or 64 bit floating point wave as output. Clipping would be a non-issue then.

You can use the "-full" switch, that turns off downsampling. The WAV file will then be stored in the native processing bitdepth (which when speeding up is 64bit floating point). However, 2pass mode will still be active. It will be faster that way, though, because eac3to will automatically detect that it doesn't have to decode the track and speed it up another time. It will just do another pass WAV -> WAV to remove the clipping. You can get a similar effect by using the "-2pass" option. That will store a temporary 64bit floating point WAV file and then in a 2nd step simply remove clipping and convert the WAV to 24bit PCM. Just try either "-full" or "-2pass" and use whatever you like more. I think the 2nd pass should be fast enough in both cases to not bother you, anymore.

[piratburner]

Originally Posted by madshi

Generally yes. However, TrueHD tracks will need special treatment. Please ask in the tsMuxeR thread about this. They'll help you there. Also HD DVD style E-AC3 tracks must be transcoded to another format (e.g. AC3 or DTS, whatever you prefer) to be Blu-Ray compatible.

Thanks madshi :-) :-)

[madshi]

Originally Posted by rica

eac3to doesn't recognize this: [...]

That's because it's not a valid m2ts file. There are 128 garbage bytes at the beginning of the file. Or maybe it's half a TS block. Don't know. Remove the first 128 bytes with a hexeditor, afterwards eac3to will detect it.

[piratburner]

making DTS track

As default is Surcode DTS Encoder but I also have DTS Pro Encoder installed. I have searched but no luck , how to tell eac3to to use this instead ? or wich DTS encoder is the best to use for DTS?

[rica]

Originally Posted by madshi

That's because it's not a valid m2ts file. There are 128 garbage bytes at the beginning of the file. Or maybe it's half a TS block. Don't know. Remove the first 128 bytes with a hexeditor, afterwards eac3to will detect it.

No,

C:\>clip\clip C:\users\rica\desktop\test\CD_out.m2ts C:\users\rica\desktop\test\cd_out_cut.m2ts 130 74000000

C:\>eac3to\eac3to C:\users\rica\desktop\test\cd_out_cut.m2ts
The format of the source file could not be detected.

PDVD,TMT and MediaInfo works with both files.

[madshi]

Originally Posted by piratburner

how to tell eac3to to use this instead ? or wich DTS encoder is the best to use for DTS?

I guess the DTS Pro Encoder is probably better. eac3to can currently not automate it, but you can ask eac3to to produce a "something.agm" file. This is the file format the old DTS Pro Encoder needs. The latest DTS Master Audio Suite eats WAV just fine, AFAIK. So if you have that, just output WAV.

Originally Posted by rica

clip CD_out.m2ts cd_out_cut.m2ts 130 74000000

[rica]

Originally Posted by madshi

I

Me too

[madshi]

@rica, are you trying to annoy me? Do I really have to waste another post on telling you that "128" and "130" are different numbers?

[rica]

Thanks my friend, sorry for annoying you; forget it.

[laserfan]

Actually we were hoping to see from you: "I trimmed the front-off as you suggested madshi and it worked."

[rica]

Nobody offered me to cut the file bit by bit so as to make his soft work;
check this approach: (same test file)

http://forum.doom9.org/showthread.ph...96#post1232096

This was my last word here, thanks.

[DrNein]

When using the delay option to sync a new AC3 track, should it be entered as-is or rounded to the nearest block (32 divisor)? For instance, if the difference determined by visual comparison is +1064ms should that be entered or rather +1056?

That may seem trivial but I want to understand how it works and the ideal method.

[rebkell]

Originally Posted by DrNein

When using the delay option to sync a new AC3 track, should it be entered as-is or rounded to the nearest block (32 divisor)? For instance, if the difference determined by visual comparison is +1064ms should that be entered or rather +1056?

That may seem trivial but I want to understand how it works and the ideal method.

I've asked the same basic question in the past and answer is that Madshi will pick the closest match he can get with the audio frame he can, if you selected 100ms delay with a 48KHz AC3 he would actually insert 96ms if you selected a 50ms delay he would choose to add 64ms delay...

[DrNein]

Okay, so I may as well enter the nearest myself. However, is it invalid to use other delays and does the delaycut program allow them or also automatically adjust to the nearest?

[kartman_canada]

I'm trying to convert my HD-DVDs to something that is good for the PS3. I've been using EAC3TO to demux the VC-1 and demux the audio and convert to AC3 5.1 @ 640kbps. The final step is to convert the VC-1 to AVC and mux these into an M2TS.

I thought all was good until I tried watching a converted copy of Hot Fuzz. The video looks good and the audio is all there and in sync but I'm hearing some weirdness in the balance between the channels... dialog too quiet and fronts/sides too strong. I keep trying to turn up the volume to hear the dialog and then panic looking for the remote when an explosion happens.

Below is a listing from the filter test:

eac3to v2.84
command line: "E:\Video_Encoding\eac3to\eac3to.exe" -test
------------------------------------------------------------------------------
Nero Audio Decoder (Nero 7) works fine
ArcSoft DTS Decoder (1.1.0.0) works fine
Sonic Audio Decoder (2.87.0.0) doesn't seem to be installed
Haali Matroska Muxer (2008-03-29) is up to date
Nero AAC Encoder (1.1.34.2) is installed
  There's a new version (1.3.3.0) available
  http://www.nero.com/eng/downloads-nerodigital-nero-aac-codec.php
Surcode DTS Encoder (1.0.23.0) is installed
MkvToolnix (2.4.1.0, beta 2008-12-07) is installed

This particular movie is 5.1 E-AC3 EX. Are my filter versions the issue? Are there any EAC3TO audio switches that I should be using?

Here is an example of what I've been doing... in this case I'm only processing the audio and I experimented with not removing the dialog normalization. It didn't seem to help. I'd really appreciate any input the more experienced users out there might be able to offer!

Many thanx to Madshi too...

eac3to v2.84
command line: "E:\eac3to.exe" "E:\FEATURE_1.EVO"+"E:\FEATURE_2.EVO" 4: "E:\movie.ac3" -640 -keepDialnorm
------------------------------------------------------------------------------
EVO, 1 video track, 6 audio tracks, 7 subtitle tracks, 2:00:52
"Main Movie"
1: Joined EVO file
2: Chapters, 28 chapters with names
3: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
4: E-AC3 EX, English, 5.1 channels, 1536kbps, 48khz, dialnorm: -27dB, 133ms
5: E-AC3 EX, French, 5.1 channels, 768kbps, 48khz, dialnorm: -27dB, 133ms
6: E-AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, 133ms
   "Feature Commentary w/ Simon Pegg and Edgar Wright"
7: E-AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, 133ms
   "Feature Commentary w/ Sandford Police Service"
8: E-AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, 133ms
   "Feature Commentary w/ Sandford Village People"
9: E-AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, 133ms
   "Feature Commentary w/ The Real Fuzz"
10: Subtitle, English, "SDH"
11: Subtitle, French
12: Subtitle
13: Subtitle
14: Subtitle
15: Subtitle
16: Subtitle, English, "Fuzz-O-Meter: Trivia Track"
[a04] Extracting audio track number 4...
[a04] Decoding with DirectShow (Nero Audio Decoder 2)...
[a04] DirectShow reports 5.1 channels, 24 bits, 48khz
[a04] Applying RAW/PCM delay...
[a04] Encoding AC3 <640kbps> with libAften...
[a04] Creating file "E:\movie.ac3"...
Video track 3 contains 173880 frames.
eac3to processing took 15 minutes, 54 seconds.
Done.

[Nullity]

Originally Posted by kartman_canada

I'm trying to convert my HD-DVDs to something that is good for the PS3. I've been using EAC3TO to demux the VC-1 and demux the audio and convert to AC3 5.1 @ 640kbps. The final step is to convert the VC-1 to AVC and mux these into an M2TS.

I thought all was good until I tried watching a converted copy of Hot Fuzz. The video looks good and the audio is all there and in sync but I'm hearing some weirdness in the balance between the channels... dialog too quiet and fronts/sides too strong. I keep trying to turn up the volume to hear the dialog and then panic looking for the remote when an explosion happens.

Don't use the "-keepDialnorm" switch, it lowers the volume of the center channel. Or to be more correct, not using that switch allows eac3to to remove the dialog normalization which increases the volume of the center channel.

[tebasuna51]

Originally Posted by kartman_canada

... dialog too quiet and fronts/sides too strong. I keep trying to turn up the volume to hear the dialog and then panic looking for the remote when an explosion happens...

This is called 'Full Dynamic Range Preserved'.
Your command line are ok and -keepDialnorm don't help. This is the original sound.

By default many players apply a 'Dynamic Range Compression' than reduce the differences between low and high volume values (the quality is also reduced).

If you want this effect the receivers have a 'Night Mode' to use.

EDIT: @Nullity, the DialNorm value lower the volume of all channels not only center channel.

[Nullity]

Originally Posted by tebasuna51

@Nullity, the DialNorm value lower the volume of all channels not only center channel.

Ah, thanks for the clarification.

[kartman_canada]

Originally Posted by tebasuna51

This is called 'Full Dynamic Range Preserved'.
Your command line are ok and -keepDialnorm don't help. This is the original sound.

OK... The initial converted audio that I listened to (and found that I didn't like the level of the center channel) was processed using the command line below.

command line: "E:\eac3to.exe" "E:\FEATURE_1.EVO"+"E:\FEATURE_2.EVO" 4: "E:\movie.ac3" -640

I'm glad to hear that this is "correct" but I still think the center channel is too weak. Seeing as it's coming from an HD-DVD it's very hard for me to arrange a proper A/B comparison between the converted AC3 audio and the original E-AC3.

Is there anything else that might cause the level of the center to be reduced (or the other channels to be boosted)? It seems wrong but I guess that I could look to tweak the AC3 before muxing...

Any other comments or suggestions?

[ultratoto14]

I think that that was already asked but could it be possible to convert h264-dts-movie.mkv to h264-ac3-movie.mkv or just video-with-dts-movie.mkv to audio.ac3. I know that reading mkv is not the same as writing but it could be very interesting.

[MichaelAnders]

Mashi, found a bug in v2.87 with a BD, wrong play duration is displayed.

O:\bd\2>eac3to h:
1) 00021.mpls, 00005.m2ts+00006.m2ts, 1:53:44
- VC-1, 1080p24 /1.001 (16:9)
- DTS, German, multi-channel, 48khz
- DTS, English, multi-channel, 48khz
- RAW/PCM, German, multi-channel, 48khz
- RAW/PCM, English, multi-channel, 48khz

2) 00006.mpls, 00006.m2ts, 1:53:39
- VC-1, 1080p24 /1.001 (16:9)
- DTS Master Audio, German, multi-channel, 48khz
- DTS Master Audio, English, multi-channel, 48khz
- RAW/PCM, German, multi-channel, 48khz
- RAW/PCM, English, multi-channel, 48khz

3) 00014.mpls, 00014.m2ts, 0:21:15
- MPEG2, 480i60 /1.001 (16:9)
- DTS, English, stereo, 48khz

4) 00011.mpls, 00011.m2ts, 0:20:40
- MPEG2, 480i60 /1.001 (16:9)
- DTS, English, stereo, 48khz

O:\bd\2>eac3to h: 1)
M2TS, 1 video track, 4 audio tracks, 2 subtitle tracks, 0:00:05
1: Chapters, 22 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: DTS, German, 5.1 channels, 24 bits, 768kbps, 48khz
4: DTS, English, 5.1 channels, 24 bits, 768kbps, 48khz
5: RAW/PCM, German, 5.1 channels, 16 bits, 48khz
6: RAW/PCM, English, 5.1 channels, 16 bits, 48khz
7: Subtitle (PGS), German
8: Subtitle (PGS), German

Here the duration is now just 5 sec - seems this is the time for 00005.m2ts file which just displays "Senator"?

O:\bd\2>eac3to h: 2)
M2TS, 1 video track, 4 audio tracks, 2 subtitle tracks, 1:53:39
1: Chapters, 22 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: DTS Master Audio, German, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 768kbps, 48khz)
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 768kbps, 48khz)
5: RAW/PCM, German, 5.1 channels, 16 bits, 48khz
6: RAW/PCM, English, 5.1 channels, 16 bits, 48khz
7: Subtitle (PGS), German
8: Subtitle (PGS), German

Here the time is correct...

[madshi]

Originally Posted by DrNein

However, is it invalid to use other delays and does the delaycut program allow them or also automatically adjust to the nearest?

But eac3to and delaycut accept any number, but round it to the nearest possible value.

Originally Posted by kartman_canada

I'm glad to hear that this is "correct" but I still think the center channel is too weak.

Maybe the studio mix was bad? Or maybe there's something else going on with your playback chain? It's really hard to diagnose for us without having access to your PC and your playback hardware. What I can say is that you've used a Dolby reference decoder for decoding the E-AC3 track and you've used a well tested and usually very reliable AC3 encoder for encoding. So the tracks you created should be fine.

Originally Posted by kartman_canada

Is there anything else that might cause the level of the center to be reduced (or the other channels to be boosted)? It seems wrong but I guess that I could look to tweak the AC3 before muxing...

You could decode to WAV and then tweak the track in e.g. Audacity.

Originally Posted by ultratoto14

I think that that was already asked but could it be possible to convert h264-dts-movie.mkv to h264-ac3-movie.mkv or just video-with-dts-movie.mkv to audio.ac3. I know that reading mkv is not the same as writing but it could be very interesting.

It is not possible with the current eac3to version. You can use mkvextract (part of mkvtoolnix) to extract tracks from MKV container. Then you can use eac3to to convert the extracted audio tracks to whatever format you prefer. And then you can use mkvmerge (part of mkvtoolnix again) to mux the final track back into the MKV and to remove the tracks you don't like.

Originally Posted by MichaelAnders

Mashi, found a bug in v2.87 with a BD, wrong play duration is displayed.

Could you please zip the "CLIPINF" and "PLAYLIST" folders and upload them? Should be just a few KBs. Thanks!

[~bT~]

@ madshi

i'm sorry if this has been asked before but i was wondering if its poss to output celltimes.txt or not? cheers!

ps. if not, will it be poss to implement it?

[ultratoto14]

@madshi

Thanks for the answer.
I already do that.
I use a DLNA media server for the PS3 that can in real time send mkv remuxed to ts or m2ts. The server can transcode DTS to ac3 in real time during muxing as the PS3 does not handle DTS via streaming.
This is the only server i know that can do this and do not loose original mkv quality. But i prefer the sound quality of eac3to compared to the one used in this server (ffmpeg).

The writer told me that he could use eac3to if eac3to could be able to handle mkv or pipes as input/output.
Do you do some check on existing files before running ?
Can eac3to work with named pipes ?

[shambles]

eac3to seems not to strip zero bytes from truehd->flac anymore:

eac3to v2.87
command line: eac3to test 1) 3: test.flac
------------------------------------------------------------------------------
M2TS, 1 video track, 8 audio tracks, 18 subtitle tracks, 0:09:34
1: VC-1, 1080p24 /1.001 (16:9)
2: AC3, English, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
3: TrueHD/AC3, English, 5.1 channels, 48khz, dialnorm: -27dB
   (embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB)
-snip-
[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 "test.flac"...
[a03] [libav] End of stream indicated
[a03] [libav] Lossless check failed - expected 0, calculated fe
[a03] The original audio track has a constant bit depth of 16 bits.
Video track 1 contains 13762 frames.
eac3to processing took 1 minute, 55 seconds.
Done.

command line: eac3to test.flac test.wav
------------------------------------------------------------------------------
FLAC, 5.1 channels, 0:09:34, 16/24 bits, 2034kbps, 48khz
Decoding FLAC...
Stripping zero bytes...
Writing WAV...
Creating file "test.wav"...
The original audio track has a constant bit depth of 16 bits.
eac3to processing took 13 seconds.
Done.

command line: eac3to test.wav test.flac
------------------------------------------------------------------------------
WAV, 5.1 channels, 0:09:34, 16 bits, 4608kbps, 48khz
Reading WAV...
Encoding FLAC with libFlac...
Creating file "test.flac"...
The original audio track has a constant bit depth of 16 bits.
eac3to processing took 29 seconds.
Done.

eac3to test.flac
FLAC, 5.1 channels, 0:09:34, 16 bits, 2033kbps, 48khz

[MichaelAnders]

Originally Posted by madshi

Could you please zip the "CLIPINF" and "PLAYLIST" folders and upload them? Should be just a few KBs. Thanks!

http://rapidshare.com/files/180787649/BDMV.zip.html

[n0mag!c]

eac3to v2.87
command line: D:\PROGRAMS\AUDIO\EAC3TO\eac3to.EXE 5e.ac3 j:5e.wav
------------------------------------------------------------------------------
AC3, 2.0 channels, 0:28:46, 192kbps, 48khz, dialnorm: -27dB
The Nero decoder doesn't seem to work, will use libav instead.
Removing AC3 dialog normalization...
Decoding with libav/ffmpeg...
Reducing depth from 64 to 24 bits...
Writing WAV...
Creating file "j5e.wav"...
eac3to processing took 20 seconds.
Done.

Now "eac3to" creates file "j5e.wav" in current folder of current disk, but v.2.80 (I've upgraded from this version to 2.87) has created file "5e.wav" in current folder of disk "j", as it's supposed to be. Can you please fix it back?!

And second problem - "sound forge 8" can't open stereo wav-files, generated by "eac3to".
Headers for 16-bit wav-files:
sforge:

0000000000: 52 49 46 46 24 C8 C0 13 │ 57 41 56 45 66 6D 74 20  RIFF$ИА‼WAVEfmt
0000000010: 10 00 00 00 01 00 02 00 │ 80 BB 00 00 00 EE 02 00  ►   ☺ ☻ _>   о☻
0000000020: 04 00 10 00 64 61 74 61 │ 00 C8 C0 13 01 00 00 00  ♦ ► data ИА‼☺

eac3to:

0000000000: 52 49 46 46 3C C8 C0 13 │ 57 41 56 45 66 6D 74 20  RIFF<ИА‼WAVEfmt
0000000010: 28 00 00 00 FE FF 02 00 │ 80 BB 00 00 00 EE 02 00  (   юя☻ _>   о☻
0000000020: 04 00 10 00 16 00 10 00 │ 03 00 00 00 01 00 00 00  ♦ ► ■ ► ♥   ☺
0000000030: 00 00 10 00 80 00 00 AA │ 00 38 9B 71 64 61 74 61    ► _  Є 8>qdata

eac3to mono:

0000000000: 52 49 46 46 24 64 E0 09 │ 57 41 56 45 66 6D 74 20  RIFF$dа○WAVEfmt
0000000010: 10 00 00 00 01 00 01 00 │ 80 BB 00 00 00 77 01 00  ►   ☺ ☺ _>   w☺
0000000020: 02 00 10 00 64 61 74 61 │ 00 64 E0 09 01 00 00 00  ☻ ► data dа○☺

Is there special reason that "eac3to" creates different header for stereo file?

[tebasuna51]

@n0mag!c
About the wav headers:

v2.57
...
* parameter "-extensible" is no longer supported (it's default now)
* new parameter "-simple" can be used to disable the "-extensible" wav header

I obtain also mono with WAVE_FORMAT_EXTENSIBLE headers.

[Snowknight26]

Originally Posted by n0mag!c

Now "eac3to" creates file "j5e.wav" in current folder of current disk, but v.2.80 (I've upgraded from this version to 2.87) has created file "5e.wav" in current folder of disk "j", as it's supposed to be. Can you please fix it back?!

You should be using J:\5e.wav.

[n0mag!c]

"j:\5e.wav" means create file in ROOT folder of disk "j", but I want to create file in CURRENT folder of disk "j".
Obviously Madshi added/change parser, which removes ":" in file names now.

[n0mag!c]

Originally Posted by tebasuna51

* new parameter "-simple" can be used to disable the "-extensible" wav header

Thanks for the tip!
Can this switch be mentioned on the first page, as like as all other yet unmentioned switches? I forgot in time the switch, with which I can get float-wav. Can somebody please remind me? If I ain't wrong about its existing.

Originally Posted by tebasuna51

I obtain also mono with WAVE_FORMAT_EXTENSIBLE headers.

Why?


Another problem - "eac3to" crashes with this "m2ts"-file.
VC-1 stream was incorporated from WMV-file.

[nautilus7]

Originally Posted by n0mag!c

"j:\5e.wav" means create file in ROOT folder of disk "j", but I want to create file in CURRENT folder of disk "j".
Obviously Madshi added/change parser, which removes ":" in file names now.

The output file is written in the folder where eac3to.exe is called from or in whatever folder you set.

[Chumbo]

Originally Posted by n0mag!c

...Can this switch be mentioned on the first page, as like as all other yet unmentioned switches?...

Read your changelog.txt that's included with every release.

And for all those that don't use this file as a reference, it's the least you can do to show madshi you appreciate all his efforts.

[n0mag!c]

Originally Posted by nautilus7

The output file is written in the folder where eac3to.exe is called from or in whatever folder you set.

I fully understand the logic of creation files, don't worry about that. I state, that now "eac3to" removes ":" from file name like this "j:5e.wav" and creates file "j5e.wav" instead.

Originally Posted by Chumbo

Read your changelog.txt that's included with every release.

This is evil logic to keep description of application in "changelog.txt". Especially, while help screen exists.

And another little bug, as far as it goes.
When capturing output from "eac3to" like this: "eac3to>help.txt", file is filled with 08h characters, but they must have not be there.

Originally Posted by Chumbo

And for all those that don't use this file as a reference, it's the least you can do to show madshi you appreciate all his efforts.

P.S. Madshi, I appreciate your work very-very much, there's no doubt! Making world a little more perfect is great.

[rebkell]

I would guess the bug or change happened because of this problem:

http://forum.doom9.org/showpost.php?...postcount=7607

[alc0re]

madshi,

Could you add a command that forces dialog normalization...something like -forceDialnorm

Or even a command that would me me specify what level of dialog normalization an ac3 file has? Either way would work. Yet another option would be if eac3to detected a -0 dialog normalization level as having dialog normalization so that when eac3to extracted the file it would change it to -31. Reason I ask is so that I can "fix" my ac3 tracks with -0 dialog normalization without having to re-encode them. I believe that this wouldn't be too big of a feature to add since removing dialog normalization doesn't seem to require re-encoding (seems like its just changing the value from whatever the file was to -0 pre v2.85 or -31 post v2.85 .)

One other question : When I do an eac3to command on a bluray structure, it tells me my main movie track's video is something like 1080p24. lets say that was feature 1. So I do a eac3to 1) command and again is says the video track is 1080p24. When I check the m2ts file with mediainfo it says the video is 23.976 FPS. MeGUI also detects the video as 23.976 FPS. I transcode my video down to 720p and create a bluray structure with tsMuxer. When I play my avchd discs on my bluray player, the video stutters when there's movement...like if the camera is following the actor and the background is moving, the background seems to stutter. Now I can't figure out for the life of me why this is happening. Only difference I have noticed is the following. eac3to reports 1080p24 on both the regular eac3to command against the original bluray and when I look at the main movie feature (ie eac3to 1) ) When I do a eac3to command on my bluray structured transcoded bluray, it reports 720p24. But when I look at the main movie feature (eac3to 1)) it reports 720p23. I looked at my bluray structure with bdedit.exe on both the original bluray and my transcoded bluray and I don't see any difference. Where does eac3to get the information from that would be having eac3to say its 720p23 instead of 720p24? I think this might have something to do with why my video seems to stutter. PS. doesnt happen on my pc when I play the m2ts file directly. And yes when I transcode my video it is transcoded at 23.976 fps.

[n0mag!c]

Originally Posted by alc0re

When I play my avchd discs on my bluray player, the video stutters when there's movement...like if the camera is following the actor and the background is moving, the background seems to stutter. Now I can't figure out for the life of me why this is happening.

I guess it's just your TV issue. 24 fps is too slow for human eye. Size of your PC monitor is smaller and refresh rate is greater, that's why this stutter is less noticable on PC.
Modern TVs use different techniques to eliminate this effect, personally, I prefer interpolation of additional frames, similar to Philips' DNM (digital natural motion). But many others don't like it for the "theatre" look.

[nautilus7]

Originally Posted by n0mag!c

I fully understand the logic of creation files, don't worry about that. I state, that now "eac3to" removes ":" from file name like this "j:5e.wav" and creates file "j5e.wav" instead.

Have you ever tried using : for a filename in windows? Guess what happens.

[Chumbo]

Originally Posted by n0mag!c

...This is evil logic to keep description of application in "changelog.txt". Especially, while help screen exists....

The help screen is for existing switches and, at the developer's discretion, may not show hidden switches. It's not for "updates," i.e., switches may be added, removed, renamed, etc. That's why you need to check the changes log.

Originally Posted by n0mag!c

...And another little bug, as far as it goes.
When capturing output from "eac3to" like this: "eac3to>help.txt", file is filled with 08h characters, but they must have not be there.
...

Use the -log switch. There's no need to pipe the output since the -log now does everything and madshi was kind enough to allow multiple running instances to have their own logs.

It's not a bug, it's just what gets piped from the screen/console to the file.

All you need to do is run this: eac3to -log=help.txt

[n0mag!c]

Originally Posted by nautilus7

Have you ever tried using : for a filename in windows? Guess what happens.

Of course, constantly! I'm using "Far manager" as "GUI" for "DOS" environment. And using "j:5e.wav" as file name helps me to avoid using full path. So current paths for all disks live long lifes in this console application.

Originally Posted by Chumbo

It's not a bug, it's just what gets piped from the screen/console to the file.

But this happens only with "eac3to"...

[asarian]

@Madshi,

When I convert a DTS Hi-Res stream (from Predator 2) to LPCM, eac3to doesn't seem to list the final bit-depth:

eac3to 00013.m2ts 3: c:\video\temp.pcm
M2TS, 1 video track, 9 audio tracks, 24 subtitle tracks, 1:48:00
1: Chapters, 33 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: DTS Hi-Res, English, 5.1 channels, 24 bits, 3018kbps, 48khz
   (core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
...
[a03] Extracting audio track number 3...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Swapping endian...
[a03] Remapping channels...
[a03] Creating file "c:\video\temp.pcm"...
[a03] The last DTS frame is incomplete and thus gets skipped.
Video track 2 contains 155364 frames.
eac3to processing took 17 minutes, 40 seconds.
Done.

Normally, it gives me a message saying the final output file had 16/24 bit depth, etc. This time not. Is that a small bug?

Thanks

[alc0re]

I appreciate your reply n0magic but that's not taking into account that I said it doesn't happen when i'm watching non-transcoded retail bluray or hd-dvds on my tv...

[n0mag!c]

Originally Posted by alc0re

I appreciate your reply n0magic but that's not taking into account that I said it doesn't happen when i'm watching non-transcoded retail bluray or hd-dvds on my tv...

I'm sorry, all ordinary TVs seem like stuttering to me. Ok, then here's my second guess - bitrate overflow. Do you have ability to burn your transcodings on BD-RE which support higher transfer rate to test this?
Sometimes 23, 24, 23.976 are used as synonyms.

[alc0re]

no i dont have that ability. To me I see stuttering a lot too but not like this. I can take the same movie and watch it from the retail bluray or hd-dvd and then transcode it and see a big difference.

BD-RE is not an option for me.

I thought maybe the bitrate was too high also for a dvd, but I've been using Ryu77's profiles lately for avchd content. he's got a lower value in his profile. Ryu's profile : VBVBuffersize = 15000 and VBVMaxbirate = 17500. Those values are a lot lower than max 40000 for 4.1 (which is max supported.) I'm still curious why it says 1080p24 on the original bluray and not on the transcoded.

If I can't figure this out I'm going to try playing the avchd disc on another brand bluray player. Problem with that is that it won't be on my tv.

EDIT/UPDATE : This morning I transcoded The Matrix Revolutions using RipBot264 instead of MeGUI. My normal process is using MeGUI with Ryu77's AVCHD profile, with a target size of 8152 to fit on a dual layer dvd. I have been burning my movies to memorex dual layer dvds. I wanted to try something different so I transcoded the matrix 3 this morning using ripbot instead of megui. I also changed the target size to fit on a single layer dvd. I used ripbot's avchd/bluray profiles. I also burnt the movie to a Verbatim single layer dvd that I just bought cause I researched and found that Verbatim make better blank dvds. I'm watching that ripbot transcoded movie right now and I don't see the same stuttering issue. Unfortunately there were a few different things that are different now. RipBot did the transcoding, albeit with the same program that MeGUI does so that shouldn't be the difference. The bitrate is a lot lower than all my other transcodes since I made this one fit on a single layer dvd (dual layers are freakin expensive.) The x264 encoding profile is different. The level is different (4.0 as opposed to 4.1) I just now have to try different things to figure out what made the difference. Also, RipBot does the muxing through tsMuxer as opposed to me doing it manually. I need to figure out if RipBot changes any of the default muxing options in tsMuxer, like the "continually insert SPS/PPS or Add Picture Timing Info or or Use ASync I/O.

EDIT/UPDATE 2 : Demuxed a movie that had the stuttering issue. Remuxed the movie with tsRemux and on the video stream options, I changed the level from High@4.1 to High@4.0. Tested. Works perfectly. So for anyone interested, the panasonic bd35 does not play nice with video at level 4.1.

[idbirch2]

Originally Posted by asarian

When I convert a DTS Hi-Res stream (from Predator 2) to LPCM, eac3to doesn't seem to list the final bit-depth:

Normally, it gives me a message saying the final output file had 16/24 bit depth, etc. This time not. Is that a small bug?

Just run eac3to "c:\video\temp.pcm"

[shanghai2004]

Madshi,

Had finally time to make a sample to reproduce the issue from post #7225.
(issue: cannot extract EAC3 5.1 audio track from EVO, got endless 13ms audio overlap warnings, preventing eac3to from completing)

However, while testing the sample, eac3to doesn't like incomplete EAC3 packets anymore seems. Got crash.

Sample (part1.evo, 33MB):

http://www.mediafire.com/?c1sn23b2kxr

See also crash report send by email

[n0mag!c]

I've never got "Clipping detected" message on AC3-file before, is it normal?

eac3to v2.87
command line: D:\PROGRAMS\AUDIO\EAC3TO\eac3to.EXE en.ac3 J:\temp\trans\en.wavs
------------------------------------------------------------------------------
AC3, 5.1 channels, 2:23:27, 448kbps, 48khz, dialnorm: -27dB
The Nero decoder doesn't seem to work, will use libav instead.
Removing AC3 dialog normalization...
Decoding with libav/ffmpeg...
Remapping channels...
Reducing depth from 64 to 24 bits...
Writing WAVs...
Creating file "J:\temp\trans\en.C.wav"...
Creating file "J:\temp\trans\en.LFE.wav"...
Creating file "J:\temp\trans\en.L.wav"...
Creating file "J:\temp\trans\en.R.wav"...
Creating file "J:\temp\trans\en.SL.wav"...
Creating file "J:\temp\trans\en.SR.wav"...
Clipping detected, a 2nd pass will be necessary.
Starting 2nd pass...
Removing AC3 dialog normalization...
Decoding with libav/ffmpeg...
Remapping channels...
Reducing depth from 64 to 24 bits...
Writing WAVs...
Creating file "J:\temp\trans\en.L.wav"...
Creating file "J:\temp\trans\en.C.wav"...
Creating file "J:\temp\trans\en.R.wav"...
Creating file "J:\temp\trans\en.LFE.wav"...
Creating file "J:\temp\trans\en.SL.wav"...
Creating file "J:\temp\trans\en.SR.wav"...
eac3to processing took 9 minutes, 13 seconds.
Done.

"DelayCut" didn't find any problem:

====== INPUT FILE INFO ========================
File is          	ac3
Bitrate  (kbit/s) 	448
Act rate (kbit/s) 	448.000
File size (bytes)	481990656
Channels mode    	3/2: L+C+R+SL+SR
Sampling Frec    	48000
Low Frec Effects 	LFE: Present
Duration         	02:23:26.976
Frame length (ms)	32.000000
Frames/second    	31.250000
Num of frames    	268968
Bytes per Frame  	1792.0000
Size % Framesize 	0
CRC present: 	YES
=============================================
====== TARGET FILE INFO ======================
Start Frame   	0
End Frame     	268967
Num of Frames 	268968
Duration      	02:23:26.976
NotFixedDelay 	0.0000
=============================================
====== PROCESSING LOG ======================

Though "eac3to" created files with exactly the same length - 02:23:26.976

P.S. Gosh, I'd rather be using 2.80...

P.P.S. Here is that sample.

[Thunderbolt8]

is there actually any way to utilize DTS Express tracks somehow atm? using ac3filter, I only get garbage sound and muxing them with mkvmerge doesnt work as well yet. so is converting them to normal DTS or ac3 is the only choice I have atm?

[nautilus7]

Originally Posted by n0mag!c

I've never got "Clipping detected" message on AC3-file before, is it normal?

Yes, it is. eac3to can now detect audio peaks above "maximum level" and if necessary reduce the volume of the track so the clipping is gone.

(hope i explained this correctly, madshi)

Originally Posted by Thunderbolt8

is there actually any way to utilize DTS Express tracks somehow atm? using ac3filter, I only get garbage sound and muxing them with mkvmerge doesnt work as well yet. so is converting them to normal DTS or ac3 the only choice I have atm?

Can't you use nero decoder with your media player (mpc)? I think is the only dshow decoder available for dts express. Otherwise conversion is the only way.

[asarian]

Originally Posted by idbirch2

Just run eac3to "c:\video\temp.pcm"

Never knew you could do that. (since pcm/raw has no header and all). If at all possible, though, I'd like eac3to to include that information for DTS-HD tracks too, like it does for DTS-MA/TrueHD, etc.

[n0mag!c]

Originally Posted by nautilus7

eac3to can now detect audio peaks above "maximum level" and if necessary reduce the volume of the track so the clipping is gone.

Thanks! I incorrectly believed that this is equal to gap/overlap detection. (on coincidence, my two audio streams go out of sync at this point). I really don't like that the second pass is needed to eliminate clipping.

[Thunderbolt8]

Originally Posted by nautilus7

Yes, it is. eac3to can now detect audio peaks above "maximum level" and if necessary reduce the volume of the track so the clipping is gone.

(hope i explained this correctly, madshi)

Can't you use nero decoder with your media player (mpc)? I think is the only dshow decoder available for dts express. Otherwise conversion is the only way.

i deactivated all other audio filters, but still only got garbage noise with it. same for arcsoft and arcsoft HD decoder

[73ChargerFan]

Originally Posted by n0mag!c

When capturing output from "eac3to" like this: "eac3to>help.txt", file is filled with 08h characters, but they must have not be there.

Yup, funny.

Originally Posted by Chumbo

All you need to do is run this: eac3to -log=help.txt

By convention,
command > log.txt
should work with any console command line application. For some reason, eac3to prints 80 backspace characters at the beginning of each line. Wierd.

[rebkell]

Originally Posted by 73ChargerFan

Yup, funny.

By convention,
command > log.txt
should work with any console command line application. For some reason, eac3to prints 80 backspace characters at the beginning of each line. Wierd.

That's how he keeps the status line on the console in the same position, he writes 80 characters, then backspaces them out before he writes the next update line.

[Jaja1]

Originally Posted by n0mag!c

I really don't like that the second pass is needed to eliminate clipping.

Try to run eac3to while keeping dialnorm. Perhaps that prevents the clipping.

[asarian]

Originally Posted by Jaja1

Try to run eac3to while keeping dialnorm. Perhaps that prevents the clipping.

Hmm. So, are you saying that the normal procedure of removing dialog normalization causes/can cause the track to become damaged? (as in: parts of audio being cut off, above a certain threshold). That would not be good.

Here's to hoping I misunderstood.

[n0mag!c]

Originally Posted by rebkell

That's how he keeps the status line on the console in the same position, he writes 80 characters, then backspaces them out before he writes the next update line.

Common practice is to output "CR" (carriage return) character (0Dh) for that purpose. But in this case only 79 characters in line can be used to return back. But there is no need for returning back at help screen.

[n0mag!c]

Originally Posted by Jaja1

Try to run eac3to while keeping dialnorm. Perhaps that prevents the clipping.

It makes no difference. I provided a sample, you can try for yourself.

[tebasuna51]

Originally Posted by Jaja1

Try to run eac3to while keeping dialnorm. Perhaps that prevents the clipping.

Not at all. DialNorm is a field in header not used in this process.

[weedenbc]

Quick question - I'm trying to convert an m2ts files with a TrueHD audio track using eac3to. It keeps failing with error message "This audio conversion is not supported."

I thought the libav that shipped with eac3to was supposed to handle TrueHD? I also have Arcsoft TMT installed but that doesn't seem to be helping.

----
Brian

[Snowknight26]

Helps to post the command you're trying.

[asarian]

Originally Posted by weedenbc

Quick question - I'm trying to convert an m2ts files with a TrueHD audio track using eac3to. It keeps failing with error message "This audio conversion is not supported."

I thought the libav that shipped with eac3to was supposed to handle TrueHD? I also have Arcsoft TMT installed but that doesn't seem to be helping.

Two likely causes:

1) Wrong track number (you selected a video track, instead of audio, for instance).

2) Unrecognized extension for the audio output file.

So, show us what ya tried to do.

[weedenbc]

Yeah, you see it helps if I put a .mkv on the end of the second parameter telling eac3to where to write the video stream to.

Sorry

[bmnot]

Can anyone confirm that Nero Directshow AAC Decoder filter produce the same output as neroAacDec from NeroDigitalAudio.zip?

[tebasuna51]

Originally Posted by bmnot

Can anyone confirm that Nero Directshow AAC Decoder filter produce the same output as neroAacDec from NeroDigitalAudio.zip?

The (1) DirectShow decoder output 24 bits but (2) NeroAacDec only 16 bits.

Using a unique test file, the differences between the (1) downsampled to 16 bits and (2) are only in the last significant bit (-90.31 dB).

[Greif]

Since NeroAAcEnc now encodes 7.1, does eac3to support creating 7.1 channel AAC from DTS-HD/TrueHD/EAC3?

[saint-francis]

I am working with a BD that is several TV episodes. When I use -check it seems to only check one episode. Is this intentional? What if there is an error somewhere else in the rip?

[magic144]

Just been looking into some AC3 time-stretching using ffmpeg/sox/aften for the decode/stretch/encode steps (basically to shorten a movie's AC3 soundtrack by 160ms).

Just wondering if there are any hidden problems in using this toolchain/workflow.
I notice that the output .ac3 has the dialnorm gone (according to eac3to), presumably because when ffmpeg transforms the input .ac3 to multichannel .wav, it gets 'lost' (discarded?).

Not sure if ffmpeg is ignoring the dialnorm in doing this conversion, or if it would be worth a preliminary .ac3->.ac3 pass with eac3to to remove dialnorm?? What do those in the know think? Would a preliminary pass (.ac3->.ac3) with eac3to have any other side-effects (don't know too much about bit-depth or any other factors - would anything else be adjusted?)

Cheers in advance,
m

[DrNein]

I found that NeEacDec.dll (from a previous install or otherwise extracted) can be added to the DSFilter folder after installing Nero Micro (all options), thus resulting in Nero decode for eac3to and of course Nero Burning ROM while taking less space than installing only ShowTime (which is not much use anyway).

[DrNein]

magic144, I cannot imagine why it would be desirable to time-stretch a movie like that. More likely a delay/cut or edit is appropriate.

[rebkell]

Originally Posted by DrNein

magic144, I cannot imagine why it would be desirable to time-stretch a movie like that. More likely a delay/cut or edit is appropriate.

I was wondering what the purpose of stretching was myself, of course I'm only looking at it from my perspective, which usually involves dropped audio frames in my captures, which involves figuring out where it happens and adding in the right spots.

[magic144]

@DrNein - I cannot imagine why a movie rip would be so broken in this way, but I've done this on 2 films now, where the audio is gradually seen to go ~160ms out of sync by the end of the feature

I do know that rips I've done with eac3to do not suffer from this issue :-)

anyway, it is certainly possible that there may be 1 or only a handful of distinct points at which the audio might jump in sync (I can't pinpoint them), but from experience, applying a linear progressive delay in this way makes the entire film totally watchable and whatever sync errors might exist throughout are significantly harder to perceive with this approach

I didn't invent this by the way - I found it here:-
http://www.nabble.com/How-to-time-st...d12474612.html

But disregarding the reasons, can anyone answer my specific tech. queries wrt dialnorm?!

[sundansx]

Madshi,
Small request: Next time you are in the code, could you make it so the progressnumbers feature prints the progress percentage on one line instead of printing a new line each time. As you likely know, you can do this by printf-ing a few backspace (\b) escape sequences.

also I saw this tonight for the first time (it looked like the whole process was aborted when this happened):

eac3to v2.87
command line: eac3to.exe F:\VideoStore\DVD\Futurama_BendersGame_HD\FUTURAMA3_BENDERSGAME 1) 1: "F:\VideoStore\DVD\test\Futurama_BendersGame_HD\chapters.txt" 2: "F:\VideoStore\DVD\test\Futurama_BendersGame_HD\out.mkv" 4: "F:\VideoStore\DVD\test\Futurama_BendersGame_HD\out.dts" -arcsoft -core 7: "F:\VideoStore\DVD\test\Futurama_BendersGame_HD\subtitle.sup" -2pass
------------------------------------------------------------------------------
M2TS, 2 video tracks, 3 audio tracks, 6 subtitle tracks, 1:27:27
1: Chapters, 13 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, English, 2.0 channels, 224kbps, 48khz, dialnorm: -27dB
6: DTS Express, English, 2.0 channels, 24 bits, 192kbps, 48khz
7: Subtitle (PGS), English
8: Subtitle (PGS), Spanish
9: Subtitle (PGS), French
10: Subtitle (PGS), Chinese
11: Subtitle (PGS), English
12: Subtitle (PGS), English
Creating file "F:\VideoStore\DVD\test\Futurama_BendersGame_HD\chapters.txt"...
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[s07] Extracting subtitle track number 7...
[a04] Extracting DTS core...
[v02] Muxing video to Matroska...
[a04] Creating file "F:\VideoStore\DVD\test\Futurama_BendersGame_HD\out.dts"...
[s07] Creating file "F:\VideoStore\DVD\test\Futurama_BendersGame_HD\subtitle.sup"...
[a04] The last DTS frame is incomplete and thus gets skipped.
The temp file could not be interpreted correctly.

is this a bug or a bad disc?
thanks.

[tebasuna51]

Originally Posted by magic144

Just been looking into some AC3 time-stretching using ffmpeg/sox/aften for the decode/stretch/encode steps (basically to shorten a movie's AC3 soundtrack by 160ms).

Just wondering if there are any hidden problems in using this toolchain/workflow.
I notice that the output .ac3 has the dialnorm gone (according to eac3to), presumably because when ffmpeg transforms the input .ac3 to multichannel .wav, it gets 'lost' (discarded?).

Not sure if ffmpeg is ignoring the dialnorm in doing this conversion, or if it would be worth a preliminary .ac3->.ac3 pass with eac3to to remove dialnorm?? What do those in the know think? Would a preliminary pass (.ac3->.ac3) with eac3to have any other side-effects (don't know too much about bit-depth or any other factors - would anything else be adjusted?)

There are many ffmpeg versions, forks, ...
I can't know what your version do, maybe apply the DialNorm and the DRC.

Use eac3to instead ffmpeg to decode the ac3.

[idbirch2]

I can't seem to extract the video and dtshd audio from the King Kong BluRay usig 2.87. Here's the log:

eac3to v2.87
command line: eac3to  "I:\0Stuff\King Kong" 1) 2:F:\kk.mkv 5:kk.dtshd 10:F:\kk1.sup
------------------------------------------------------------------------------
M2TS, 3 video tracks, 5 audio tracks, 5 subtitle tracks, 3:20:07
1: Chapters, 54 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: VC-1, 480p24 /1.001 (3:2)
4: VC-1, 480p24 /1.001 (3:2)
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: DTS Express, English, 2.0 channels, 24 bits, 192kbps, 48khz
10: Subtitle (PGS), English
11: Subtitle (PGS), Spanish
12: Subtitle (PGS), French
13: Subtitle (PGS), Spanish
14: Subtitle (PGS), French
[a05] Extracting audio track number 5...
[v02] Extracting video track number 2...
[s10] Extracting subtitle track number 10...
[v02] Muxing video to Matroska...
[a05] Creating file "kk.dtshd"...
[s10] Creating file "F:\kk1.sup"...
[a05] The last DTS frame is incomplete and thus gets skipped.
[a05] Audio overlaps for 11ms at playtime 1:14:09.
[a05] Audio overlaps for 7ms at playtime 1:16:59.
[a05] Audio overlaps for 6ms at playtime 1:38:53.
[a05] Audio overlaps for 9ms at playtime 2:09:19.
[a05] Audio overlaps for 6ms at playtime 2:46:16.
[a05] Audio overlaps for 8ms at playtime 2:46:38.
[a05] Audio overlaps for 11ms at playtime 2:53:16.
[a05] Audio overlaps for 10ms at playtime 3:14:19.
[a05] Starting 2nd pass...
[a05] Realizing DTS gaps...
[a05] Creating file "kk.dtshd"...
[a05] Writing the destination file failed.
Aborted at file position 5681233920.

edit: I'm getting the same error with another disc, no seamless branching on this one though, used -demux switch instead and got everything I needed.

[73ChargerFan]

magic144,

160ms is very short... that seems to be a sync problem, not time-stretching.

eac3to can remove time at various places, like 30ms at 20%, 40%, 60%, 80% and 40ms at the end.

Or apply a negative delay?

Edit: sorry for getting to this discussion late...

[mbcd]

Hi

I think its a small bug:

eac3to crashes while extracting the core of an eac3-File

eac3-File is 640kbps and i didn`t wanted to reencode to get normal ac3.

so i tried to extract the core, but eac3to crashed using libav ...

eac3to-v2.87>eac3to.exe "XXXXX\bb - 6 - E-AC3, German, 5.1 channels, 640kbps, 48khz.eac3"
E-AC3, 5.1 channels, 2:20:03, 640kbps, 48khz, dialnorm: -27dB

eac3to-v2.87>eac3to.exe "XXXXXXX\bb - 6 - E-AC3, German, 5.1 channels, 640kbps, 48khz.eac3" g:\BB\ger.ac3 -core

E-AC3, 5.1 channels, 2:20:03, 640kbps, 48khz, dialnorm: -27dB
The Nero decoder doesn't seem to work, will use libav instead.
eac3to crashed...

Possible to get the core (as normal ac3) without reencoding ? That "should" normaly not be any problem, or ?

Regards

[madshi]

Originally Posted by ~bT~

i'm sorry if this has been asked before but i was wondering if its poss to output celltimes.txt or not?

I don't even know what that is right now. What is it exactly and what purpose does it have?

Originally Posted by ultratoto14

The writer told me that he could use eac3to if eac3to could be able to handle mkv or pipes as input/output.

Due to how eac3to is designed, it doesn't support input pipes, so that won't work, unfortunately...

Originally Posted by n0mag!c

And second problem - "sound forge 8" can't open stereo wav-files, generated by "eac3to".

That's most probably a bug/problem with sound forge 8 and not with eac3to. There are 2 different kinds of WAV files and sound forge 8 seems to only like the older one (which is outdated). Use the "-simple" switch to create WAV files in the older format with eac3to.

Originally Posted by n0mag!c

Is there special reason that "eac3to" creates different header for stereo file?

IIRC eac3to creates "simple" WAV files for mono files if you use the "something.wavs" output. The reason for that is that outputting "wavs" is usually useful for Surcode DTS encoding and Surcode only understands the "simple" WAV headers, too. Furthermore, with mono WAV files the channel mask information is not so important as with multichannel WAV files (the channel mask cannot be stored in the "simple" WAV format).

Originally Posted by n0mag!c

I forgot in time the switch, with which I can get float-wav. Can somebody please remind me?

"-full"

Originally Posted by n0mag!c

When capturing output from "eac3to" like this: "eac3to>help.txt", file is filled with 08h characters, but they must have not be there.

That's caused by the drawing of the progress bar. I don't think I can get rid of that.

Originally Posted by alc0re

Could you add a command that forces dialog normalization...something like -forceDialnorm

Or even a command that would me me specify what level of dialog normalization an ac3 file has? Either way would work. Yet another option would be if eac3to detected a -0 dialog normalization level as having dialog normalization so that when eac3to extracted the file it would change it to -31.

No option for forcing dialnorm right now. But at least the next version will automatically patch "-0db" dialnorm AC3 tracks to "-31db".

Originally Posted by n0mag!c

I really don't like that the second pass is needed to eliminate clipping.

There's no way to detect and fix clipping in one pass. However, you can manually lower volume by adding e.g. "-3db" to the command line. In most cases that should make sure that no clipping occurs, so no 2nd pass will be necessary.

Originally Posted by Greif

Since NeroAAcEnc now encodes 7.1, does eac3to support creating 7.1 channel AAC from DTS-HD/TrueHD/EAC3?

To be honest, I'm not sure myself right now. Why don't you simply give it a try and report back if it doesn't work.

Originally Posted by sundansx

Next time you are in the code, could you make it so the progressnumbers feature prints the progress percentage on one line instead of printing a new line each time. As you likely know, you can do this by printf-ing a few backspace (\b) escape sequences.

The progressnumbers feature is not meant for end users, it's only meant to be used by other programs which want to "remote control" eac3to. For such programs it's better if I don't use backspace.

Originally Posted by idbirch2

I can't seem to extract the video and dtshd audio from the King Kong BluRay usig 2.87. Here's the log:

[a05] Writing the destination file failed.
Aborted at file position 5681233920.

Probably your harddisk is full. Or the network connection got lost. Or your harddisk is broken? I don't see many other reasons for why writing to a file might fail. This doesn't look like an eac3to related error to me.

[madshi]

Originally Posted by sundansx

also I saw this tonight for the first time (it looked like the whole process was aborted when this happened):

eac3to v2.87
command line: eac3to.exe F:\VideoStore\DVD\Futurama_BendersGame_HD\FUTURAMA3_BENDERSGAME 1) 1: "F:\VideoStore\DVD\test\Futurama_BendersGame_HD\chapters.txt" 2: "F:\VideoStore\DVD\test\Futurama_BendersGame_HD\out.mkv" 4: "F:\VideoStore\DVD\test\Futurama_BendersGame_HD\out.dts" -arcsoft -core 7: "F:\VideoStore\DVD\test\Futurama_BendersGame_HD\subtitle.sup" -2pass

You're asking eac3to to process the subtitle in 2pass mode. That's not supported. The 2pass mode must be activated for every track you want it be active for. You should only use it for audio tracks.

[asarian]

Originally Posted by magic144

@DrNein - I cannot imagine why a movie rip would be so broken in this way, but I've done this on 2 films now, where the audio is gradually seen to go ~160ms out of sync by the end of the feature

I do know that rips I've done with eac3to do not suffer from this issue :-)

Never seen a commercial movie (the last ten years, at least) where the sound gradually goes out of sync, and if it's not caused by eac3to, as you seem to say, then why not just use eac3to?

My guess, in the words of eac3to itself, is that "The audio file was demuxed without making use of the gap/overlap information." (in a second pass).

[Jeff Flowerday]

I'm trying to repackage some of my SDs in to mkv containers. In many cases eac3to reports multiple video and audio tracks on the concatinated vob files. It also indicates 4 subtitle tracks when there aren't any in the file.

I'm not sure how much of a priority vob files are for you, but I've attached a sample.

Ultimately it doesn't matter which tool I use to extract the elementary streams from Pineapple Express the remuxed mkv has the audio slowly getting more and more out of sync as the movie progresses.

VOB, 2 video tracks, 2 audio tracks, 4 subtitle tracks, 0:04:59
1: MPEG2, 480p24 /1.001 (16:9) with pulldown flags, -173381ms
2: MPEG2, 480p24 /1.001 (16:9) with pulldown flags, -41ms
3: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB, -173381ms
4: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB, -41ms
5: Subtitle
6: Subtitle
7: Subtitle
8: Subtitle

Here is the clip:

http://www.flowerdayconsulting.com/Misc/Clip.zip

[setarip_old]

@Jeff Flowerday

For conversion of standard DVDs to MKVs, I'd suggest you try the recently introduced "MakeMKV" which, if need be will first very quickly "rip"/decrypt your DVD and then (also very quickly) convert them to MKV...

[madshi]

Originally Posted by mbcd

Hi

I think its a small bug:

eac3to crashes while extracting the core of an eac3-File

eac3-File is 640kbps and i didn`t wanted to reencode to get normal ac3.

so i tried to extract the core, but eac3to crashed using libav ...

eac3to-v2.87>eac3to.exe "XXXXX\bb - 6 - E-AC3, German, 5.1 channels, 640kbps, 48khz.eac3"
E-AC3, 5.1 channels, 2:20:03, 640kbps, 48khz, dialnorm: -27dB

eac3to-v2.87>eac3to.exe "XXXXXXX\bb - 6 - E-AC3, German, 5.1 channels, 640kbps, 48khz.eac3" g:\BB\ger.ac3 -core

E-AC3, 5.1 channels, 2:20:03, 640kbps, 48khz, dialnorm: -27dB
The Nero decoder doesn't seem to work, will use libav instead.
eac3to crashed...

Possible to get the core (as normal ac3) without reencoding ? That "should" normaly not be any problem, or ?

Please try again with the next build, once it's out (later today). If the crash still occurs, please send me the crash report via email (dear (at) madshi (dot) net).

Extracting the core doesn't really work since HD DVD style E-AC3 tracks don't have a core. But still there should be no crash, of course...

Originally Posted by Jeff Flowerday

I'm trying to repackage some of my SDs in to mkv containers. In many cases eac3to reports multiple video and audio tracks on the concatinated vob files. It also indicates 4 subtitle tracks when there aren't any in the file.

I'm not sure how much of a priority vob files are for you, but I've attached a sample.

Ultimately it doesn't matter which tool I use to extract the elementary streams from Pineapple Express the remuxed mkv has the audio slowly getting more and more out of sync as the movie progresses.

VOB, 2 video tracks, 2 audio tracks, 4 subtitle tracks, 0:04:59
1: MPEG2, 480p24 /1.001 (16:9) with pulldown flags, -173381ms
2: MPEG2, 480p24 /1.001 (16:9) with pulldown flags, -41ms
3: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB, -173381ms
4: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB, -41ms
5: Subtitle
6: Subtitle
7: Subtitle
8: Subtitle

Does that VOB come from a DVD? If so, it would probably be wise to use a tool like DVDDecrypter, because it makes use of the IFO files to properly extract the tracks. eac3to works fine for "simple" VOB files. But complicated DVDs (e.g. multi angle or seamless branching) are not yet properly supported. I'll have to add full IFO parsing etc in order to make that work...

However, with the next version the vob sample you uploaded is only listed with one video and audio track. So maybe the problem will already be fixed then.

The subtitles are shown because they're listed in the VOB information record. eac3to doesn't check whether there's actual subtitle data because sometimes with forced subtitles the first data comes in the middle of the movie.

Originally Posted by MichaelAnders

Mashi, found a bug in v2.87 with a BD, wrong play duration is displayed.

O:\bd\2>eac3to h:
1) 00021.mpls, 00005.m2ts+00006.m2ts, 1:53:44
- VC-1, 1080p24 /1.001 (16:9)
- DTS, German, multi-channel, 48khz
- DTS, English, multi-channel, 48khz
- RAW/PCM, German, multi-channel, 48khz
- RAW/PCM, English, multi-channel, 48khz

2) 00006.mpls, 00006.m2ts, 1:53:39
- VC-1, 1080p24 /1.001 (16:9)
- DTS Master Audio, German, multi-channel, 48khz
- DTS Master Audio, English, multi-channel, 48khz
- RAW/PCM, German, multi-channel, 48khz
- RAW/PCM, English, multi-channel, 48khz

O:\bd\2>eac3to h: 1)
M2TS, 1 video track, 4 audio tracks, 2 subtitle tracks, 0:00:05
1: Chapters, 22 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: DTS, German, 5.1 channels, 24 bits, 768kbps, 48khz
4: DTS, English, 5.1 channels, 24 bits, 768kbps, 48khz
5: RAW/PCM, German, 5.1 channels, 16 bits, 48khz
6: RAW/PCM, English, 5.1 channels, 16 bits, 48khz

Here the duration is now just 5 sec - seems this is the time for 00005.m2ts file which just displays "Senator"?

O:\bd\2>eac3to h: 2)
M2TS, 1 video track, 4 audio tracks, 2 subtitle tracks, 1:53:39
1: Chapters, 22 chapters
2: VC-1, 1080p24 /1.001 (16:9)
3: DTS Master Audio, German, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 768kbps, 48khz)
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 768kbps, 48khz)
5: RAW/PCM, German, 5.1 channels, 16 bits, 48khz
6: RAW/PCM, English, 5.1 channels, 16 bits, 48khz

Here the time is correct...

It seems that the two m2ts parts differ slightly. E.g. 00005.m2ts seems to only have DTS in it, while 00006.m2ts has full Master Audio. eac3to doesn't like that. An audio track should always stay in the same format.

I'd still like to find out why the runtime is only 5 seconds for the playlist 1), but unfortunately for that I'd need the full 00005.m2ts file and the first 50MB of the 00006.m2ts. But in the end all I may find is that the runtime is wrong because the audio track properties differ between the two m2ts parts. So it's your choice whether uploading samples is worth it for you or not.

Originally Posted by shanghai2004

Had finally time to make a sample to reproduce the issue from post #7225.
(issue: cannot extract EAC3 5.1 audio track from EVO, got endless 13ms audio overlap warnings, preventing eac3to from completing)

However, while testing the sample, eac3to doesn't like incomplete EAC3 packets anymore seems. Got crash.

Will be fixed in the next build.

Originally Posted by n0mag!c

Another problem - "eac3to" crashes with this "m2ts"-file.
VC-1 stream was incorporated from WMV-file.

That VC-1 stream does not have any framerate information in it. That made eac3to crash. The next build will handle this gracefully. But it will "guess" the framerate to be 23.976.

Originally Posted by shambles

eac3to seems not to strip zero bytes from truehd->flac anymore

Thanks for the note. Will be fixed in the next build.

Originally Posted by asarian

When I convert a DTS Hi-Res stream (from Predator 2) to LPCM, eac3to doesn't seem to list the final bit-depth.

Normally, it gives me a message saying the final output file had 16/24 bit depth, etc. This time not. Is that a small bug?

It seems to work with the next build. Don't know if I accidently fixed it?

[madshi]

Originally Posted by n0mag!c

Now "eac3to" creates file "j5e.wav" in current folder of current disk, but v.2.80 (I've upgraded from this version to 2.87) has created file "5e.wav" in current folder of disk "j", as it's supposed to be. Can you please fix it back?!

Will be fixed in the next build.

[zeropc]

question...

how do i use the "fix gaps" feature in cmd? currently i use the gui and check "fix gaps" but i'd like to know how to use it in cmd.

[madshi]

Originally Posted by zeropc

how do i use the "fix gaps" feature in cmd? currently i use the gui and check "fix gaps" but i'd like to know how to use it in cmd.

There's nothing you need to do. The current eac3to build will fix audio gaps automatically. Things are different with video gaps, but they are extremely rare. The check box in the GUI is probably intended for audio gaps and it's not needed, anymore. Some of the eac3to GUIs are kind of outdated, sadly...

[kurt]

small bug when using -test. eac3to reports a new version of haali matroska muxer (but there is none).

[zeropc]

does the auto gap fix work for video too or will become a feature in the future?

[madshi]

Originally Posted by kurt

small bug when using -test. eac3to reports a new version of haali matroska muxer (but there is none).

There *is* a new Haali version.

Originally Posted by zeropc

does the auto gap fix work for video too or will become a feature in the future?

It does not work for video and I don't plan to make it available because (1) video gaps are very rare and (2) there can be false alarms. So if video gaps are reported you need to do some manual work in any case...

[sundansx]

madshi - thanks for the help. thanks for your work on eac3to - excellent program.

[kurt]

Originally Posted by madshi

There *is* a new Haali version.

hm, I'm quite sure I have the latest haali media splitter (1.9.42.1) installed.

eac3to -test
...
Haali Matroska Muxer (2009-01-10) is installed
There's a new version (2009-01-11) available
http://haali.net/mkv
....

[Jeff Flowerday]

Originally Posted by kurt

hm, I'm quite sure I have the latest haali media splitter (1.9.42.1) installed.

I get the same message with the new Haali splitter.

[Thunderbolt8]

madshi, is there anything useful for eac3to with the new haali version? I mean any advantages for you and us? because just with the changelog I cannot do much, not sure how that
# Broken Matroska files with looped SeekHeads could cause a hang in Matroska Parser, the number of SeekHeads is now limited to 10
# Removed the workaround to find tags written by Matroska Shell Extension, this caused excessive file scanning when opening files created by recent MKVToolnix
cold affect remuxing haali files with mkvmerge and such

another question regarding the -bitrate switches for ac3 decoding: when slowing down a 25fps track to 23.976 then the default value is always 640 kbit. assuming the original track has a lower initial bitrate, can I use that switch then to adjust to that bitrate and still would have zero quality loss of the file I slowed down?

[madshi]

Originally Posted by kurt

hm, I'm quite sure I have the latest haali media splitter (1.9.42.1) installed.

Oh well. There's nothing I can do. The homepage says 11th January while the splitter itself says 10th. So eac3to thinks that there's a new version available. No way to fix that, unfortunately...

[Jeff Flowerday]

Originally Posted by madshi

Does that VOB come from a DVD? If so, it would probably be wise to use a tool like DVDDecrypter, because it makes use of the IFO files to properly extract the tracks. eac3to works fine for "simple" VOB files. But complicated DVDs (e.g. multi angle or seamless branching) are not yet properly supported. I'll have to add full IFO parsing etc in order to make that work...

Yes, it's from a DVD.

I used the MakeMKV and it worked, so I'm fine. Thanks again!

[magic144]

@all
thanks for the comments and responses re: audio stretching
just to say that this technique is merely a band-aid solution for an obviously broken rip - the stretching merely hides the problem(s) as far as general human perception is concerned - as for the tools, I will consider using eac3to to convert ac3 to wav in future, for a measure of certainty (for whatever the desired application/result)

the real solution is of course to go back to the source and rip the original using the latest and greatest version of eac3to!

cheers

[asarian]

Originally Posted by Thunderbolt8

another question regarding the -bitrate switches for ac3 decoding: when slowing down a 25fps track to 23.976 then the default value is always 640 kbit. assuming the original track has a lower initial bitrate, can I use that switch then to adjust to that bitrate and still would have zero quality loss of the file I slowed down?

I distinctly recall Madshi stating, earlier, that stretching always causes quality degradation at the same bitrate. I take it that's why it will use 640Kbps anyway (even though the source may have been at, say, 448Kbps).

[setarip_old]

@Jeff Flowerday

I used the MakeMKV and it worked, so I'm fine. Thanks again!

I'm glad to hear that my suggestion resolved your dilemma ;>}

[madshi]

First release in the year 2009. Since I didn't like the idea of reaching "v2.100" sometime this year, I decided to use the opportunity to jump to "v3.00". That's also the reason why there was no new build last week. I've "collected" some things, so that v3.00 really is a worthy upgrade. Enjoy!

-------------------------------------------

eac3to v3.00 released

http://madshi.net/eac3to.zip

* broken AC3, DTS, AAC and MPx streams are now automatically repaired
* errors in TS/m2ts files are now reported (with runtime) and ignored
* damaged first max 5MB and max 5% of a TS/m2ts file are automatically skipped
* video/audio tracks which can't be parsed, are now demuxed in raw form
* added support for "line 21" closed captions in ATSC/NTSC broadcasts and DVDs
* added reading of movie / network name from "line 21" XDS information
* for gaps, edits & repairs > 1000ms eac3to now inserts silence by default
* for gaps, edits & repairs < 1000ms eac3to now loops audio by default
* option "-silence" forces eac3to to insert silence instead of looping audio
* option "-loop" forces eac3to to loop audio instead of inserting silence
* newly encoded AC3 frame is now used for "silence" instead of file's 1st frame
* increased reading block size (might improve reading performance)
* optimized TS/m2ts demuxing performance
* optimized MPEG2, VC-1 and h264 parsing performance
* command line output is colored now (e.g. errors drawn in red)
* MPEG2 1920x1088 bitstream is now automatically patched/cropped to 1920x1080
* log file now contains "<WARNING>" and "<ERROR>" indicators
* workaround for movie playlists which want the same m2ts file played twice
* added version check for eac3to (doh!)
* when a read error occurs, reading is tried again up to 3 times
* (E-)AC3 frames with -0db dialnorm are now automatically patched to -31db
* updated to newer libAften build -> fixes 44.1khz encoding
* fixed: sometimes "The last DTS frame is incomplete" was a false alarm
* fixed: mkvtoolnix version check didn't work, anymore
* fixed: errors were meant to be output to stderr, but they weren't
* fixed: automatic gap/overlap fixing with AAC targets aborted processing
* fixed: positive edit began a bit too early
* fixed: two ID3 tags after each other made eac3to fail detecting the format
* fixed: some VOB files were not detected properly

Since there have been a lot of changes, some new bugs may appear (hopefully not).

[itsancho]

Thank You, madshi!

[Thunderbolt8]

looks huge, thanks!!!

edit: can you please explain how that automatic repair thing technically works? I guess this mainly refers to those .ts streams with audio frames errors?

[asarian]

@Madshi,

My, you've been busy!

[madshi]

Originally Posted by Thunderbolt8

can you please explain how that automatic repair thing technically works? I guess this mainly refers to those .ts streams with audio frames errors?

Corrupt frames are simply deleted from the audio streams. If the source is a container, the audio gap detection will automatically detect the resulting gaps in the audio stream and fill them again with either looping or silence (depending on how big the gap is). This way the audio tracks should stay in sync.

With demuxed audio tracks eac3to tries to keep the audio track in sync, too, but it's more difficult because the helpful container timestamps are missing. So keeping audio in sync when repairing demuxed audio tracks can be a lottery.

[Atak_Snajpera]

madshi your improvements are really amazing! Excellent JOB!!!

[itsancho]

oops... unfortunately i found the first bug, sorry...

eac3to v3.00
command line: z\eac3to "H:\Casino Royale (2006) Blu-ray AVC PCM" 1) 1: e:\Chapters.txt 2: e:\Casino.mkv 4: l:\Casino.wav 7: d:\Casino.sup 8: d:\Casino1.sup
------------------------------------------------------------------------------
M2TS, 1 video track, 4 audio tracks, 8 subtitle tracks, 2:24:13
1: Chapters, 16 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: AC3, English, 5.1 channels, 448kbps, 48khz
4: RAW/PCM, English, 5.1 channels, 16 bits, 48khz
5: AC3, French, 5.1 channels, 640kbps, 48khz
6: AC3, Spanish, 5.1 channels, 448kbps, 48khz, dialnorm: -29dB
7: Subtitle (PGS), English
8: Subtitle (PGS), English
9: Subtitle (PGS), French
10: Subtitle (PGS), Spanish
11: Subtitle (PGS), Chinese
12: Subtitle (PGS), Korean
13: Subtitle (PGS), Portuguese
14: Subtitle (PGS), Thai
Creating file "e:\Chapters.txt"...
[v02] Extracting video track number 2...
[a04] Extracting audio track number 4...
[a04] The ts pcm reader received a non-supported data format. <ERROR>
[s08] Extracting subtitle track number 8...
[s07] Extracting subtitle track number 7...
[v02] Muxing video to Matroska...
Aborted at file position 1048576. <ERROR>

with 2.87 audio is demuxing just fine.

[nautilus7]

Originally Posted by madshi

Corrupt frames are simply deleted from the audio streams. If the source is a container, the audio gap detection will automatically detect the resulting gaps in the audio stream and fill them again with either looping or silence (depending on how big the gap is). This way the audio tracks should stay in sync.

With demuxed audio tracks eac3to tries to keep the audio track in sync, too, but it's more difficult because the helpful container timestamps are missing. So keeping audio in sync when repairing demuxed audio tracks can be a lottery.

Was wondering about this, too. Thanks for that nice feature.

[magic144]

and thanks for the looping/silence audio edit options!

[sidekick2]

Thanks madshi!

I'm giving it a whirl right now.

Since you're using colors now, how about changing the "----" as it's converting to some bright color? The dark gray is hard to see with my old eyes!

I don't know if this a bug, or if I did something wrong with this: (It didn't like .thd+ac3 -- Bourne-Ultimatum hd-dvd)

C:\eac3to-ver3>eac3to.exe g: 1) 3: L:\bourne-ult\bu.vc1 2: L:\bourne-ult\bu.txt 5: L:\bourne-ult\bu.
thd+ac3 10: L:\bourne-ult\bu.sup
EVO, 2 video tracks, 5 audio tracks, 3 subtitle tracks, 1:55:18
"Main Movie"
1: Joined EVO file
2: Chapters, 20 chapters with names
3: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
4: VC-1, 480p24 /1.001 (3:2)
5: TrueHD, English, 5.1 channels, 48khz, dialnorm: -27dB
6: E-AC3, English, 5.1 channels, 384kbps, 48khz, dialnorm: -27dB, -17ms
7: E-AC3, French, 5.1 channels, 384kbps, 48khz, dialnorm: -27dB, -17ms
8: E-AC3 Surround, English, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, -17ms
"Commentary"
9: E-AC3, 2.0 channels, 192kbps, 48khz, dialnorm: -27dB, -4ms
10: Subtitle (VobSub), English, "SDH"
11: Subtitle (VobSub), French
12: Subtitle (VobSub), French, "Forced"
Creating file "L:\bourne-ult\bu.txt"...
v03 Extracting video track number 3...
v03 There's no valid framerate in this bitstream.
a05 Extracting audio track number 5...
a05 Extracting audio track number 5...
v03 Writing new framerate "24fps /1.001" to bitstream.
a05 Removing TrueHD dialog normalization...
a05 Removing TrueHD dialog normalization...
v03 Writing new framerate "24fps /1.001" to bitstream.
a05 Decoding with libav/ffmpeg...
v03 Removing VC-1 pulldown...
a05 Encoding AC3 <640kbps> with libAften...
a05 TrueHD overflow in the thd ac3 joiner.
Aborted at file position 1048576.

[madshi]

Originally Posted by itsancho

oops... unfortunately i found the first bug, sorry...
with 2.87 audio is demuxing just fine.

Originally Posted by sidekick2

I don't know if this a bug, or if I did something wrong with this: (It didn't like .thd+ac3 -- Bourne-Ultimatum hd-dvd)

These two bugs will be fixed in v3.01.

Originally Posted by sidekick2

Since you're using colors now, how about changing the "----" as it's converting to some bright color? The dark gray is hard to see with my old eyes!

I find the dark progress bar prettier. But if you don't like it, you can adjust the colors to your heart's content in the command prompt settings (right click on the top left corner of the command prompt's title bar, then choose "settings").

What is everyone's opinion? Is the dark gray progress bar good? Or do you prefer the old normal colored one?

[madshi]

eac3to v3.01 released

http://madshi.net/eac3to.zip

* fixed: m2ts LCPM demuxing didn't work with v3.00
* fixed: TrueHD -> TrueHD+AC3 conversion didn't work with v3.00

[bmnot]

Great update, thanks!

So when's this program going to be renamed to Madshi's Uber HD Tool?

[itsancho]

thanx, madshi!

well, about colours... i've ment not to border u with this, but since u've asked. ;-)
i prefer command prompt to look "normal" - white background/black screen text, but with v3.00/3.01 every time i hit "Enter" eac3to automaticly is chaning my custom made settings to black background/grey text...

[nautilus7]

I like the colors as they are...

madshi, what does this mean?

eac3to "F:\bd-sc-j" 1) 3: aaa.flac
M2TS, 1 video track, 4 audio tracks, 2:04:01
1: Chapters, 29 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), -13ms
4: TrueHD/AC3, Japanese, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB), -13ms
5: AC3, English, 2.0 channels, 256kbps, 48khz, dialnorm: -27dB, -13ms
6: AC3, English, 2.0 channels, 256kbps, 48khz, dialnorm: -27dB, -13ms
a03 Extracting audio track number 3...
a03 This track begins with a non-major frame.
a03 Extracting TrueHD stream...
a03 Removing TrueHD dialog normalization...
a03 Decoding with libav/ffmpeg...
a03 Applying RAW/PCM delay...
a03 Encoding FLAC with libFlac...
a03 Creating file "aaa.flac"...

It's Sin City JAP Blu-ray Disc. The procedure seems to continue normally after that.

[KevinMcPool]

--------------------------------------------------------------------------------

Using eac3to and its gui(which ive done successfully numerous times before on seamless branching titles-this one is the ext.cut of KK) ive encountered this error message whilst remuxing the dtshd to lpcm(for my pch a100)
"The temp file could not be interpreted correctly"..eac3to correctly identifies the audio overlaps..
[a05] Audio overlaps for 11ms at playtime 1:14:09.
[a05] Audio overlaps for 7ms at playtime 1:16:59.
[a05] Audio overlaps for 6ms at playtime 1:38:53.
[a05] Audio overlaps for 9ms at playtime 2:09:19.
[a05] Audio overlaps for 6ms at playtime 2:46:16.
[a05] Audio overlaps for 8ms at playtime 2:46:38.
[a05] Audio overlaps for 11ms at playtime 2:53:16.
[a05] Audio overlaps for 10ms at playtime 3:14:19
but then stops with the aforementioned error message and produces no .lpcm output just a log with the error meassage.
The same has also just happened with Close Encounters(seamless branching too)
Ive tried with eac3to latest and older versions also.
I use the sonic decoder.
Anyone help please??

[madshi]

Originally Posted by nautilus7

madshi, what does this mean?

a03 This track begins with a non-major frame.

It means that the TrueHD track doesn't begin as it should begin technically, but eac3to works around that problem. TrueHD tracks consist of major and non-major frames. Usually a track begins with a major frame.

Originally Posted by KevinMcPool

Using eac3to and its gui(which ive done successfully numerous times before on seamless branching titles-this one is the ext.cut of KK) ive encountered this error message whilst remuxing the dtshd to lpcm(for my pch a100)
"The temp file could not be interpreted correctly"..eac3to correctly identifies the audio overlaps..
[a05] Audio overlaps for 11ms at playtime 1:14:09.
[a05] Audio overlaps for 7ms at playtime 1:16:59.
[a05] Audio overlaps for 6ms at playtime 1:38:53.
[a05] Audio overlaps for 9ms at playtime 2:09:19.
[a05] Audio overlaps for 6ms at playtime 2:46:16.
[a05] Audio overlaps for 8ms at playtime 2:46:38.
[a05] Audio overlaps for 11ms at playtime 2:53:16.
[a05] Audio overlaps for 10ms at playtime 3:14:19
but then stops with the aforementioned error message and produces no .lpcm output just a log with the error meassage.
The same has also just happened with Close Encounters(seamless branching too)
Ive tried with eac3to latest and older versions also.
I use the sonic decoder.
Anyone help please??

Need to see the full eac3to log. From v3.01, please.

[KevinMcPool]

Originally Posted by madshi

Need to see the full eac3to log. From v3.01, please.

Sorry madshi unfortunately 3.01 doesnt produce a log when i ask it to convert dtshd to lpcm from the seamless branching titles(both Close Encounters and King Kong).It merely fails with the aforementioned error message though unlike eac3to 2.87(which did produce a log) it keeps the lpcm file with the extension of pass1.lpcm...thus to summarise..
2.87-builds the lpcm file from the dtshd but then deletes it at the point of the required second pass leaving only a log file with the error message "The temp file could not be interpreted correctly"
3.01-also builds the lpcm file but also fails at the point of the second pass leaving no log file but an lpcm file e.g. "CEDirCutdtshdto.pass1.lpcm"
Here is the log from 2.87..

eac3to v2.87
command line: "C:\Documents and Settings\My Documents\eac3to287\eac3to.exe" "D:\Close Encounters\CLOSE_ENCOUNTERS\BDMV\STREAM\" 1) 4: "D:\Close Encounters\CLOSE_ENCOUNTERS\BDMV\STREAM\closeencountersdcdtshdto.lpcm"
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 20 subtitle tracks, 2:17:13
1: Chapters, 20 chapters
2: h264/AVC, 1080p24 /1.001 (16:9)
3: TrueHD/AC3, English, 5.1 channels, 48khz
   (embedded: AC3, 5.1 channels, 448kbps, 48khz)
4: DTS Master Audio, English, 5.1 channels, 24 bits, 48khz
   (core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
5: Subtitle (PGS), English
6: Subtitle (PGS), English
7: Subtitle (PGS), Dutch
8: Subtitle (PGS), Arabic
9: Subtitle (PGS), Bulgarian
10: Subtitle (PGS), Croatian
11: Subtitle (PGS), Czech
12: Subtitle (PGS), Danish
13: Subtitle (PGS), Finnish
14: Subtitle (PGS), Modern Greek
15: Subtitle (PGS), Hebrew
16: Subtitle (PGS), Hindi
17: Subtitle (PGS), Hungarian
18: Subtitle (PGS), Icelandic
19: Subtitle (PGS), Norwegian
20: Subtitle (PGS), Polish
21: Subtitle (PGS), Romanian
22: Subtitle (PGS), Slovenian
23: Subtitle (PGS), Swedish
24: Subtitle (PGS), Turkish
[a04] The ArcSoft decoder doesn't seem to work, will use Sonic instead.
[a04] Extracting audio track number 4...
[a04] Decoding with DirectShow (Sonic Audio Decoder)...
[a04] DirectShow reports 5.1 channels, 24 bits, 48khz
[a04] Swapping endian...
[a04] Remapping channels...
[a04] Creating file "D:\Close Encounters\CLOSE_ENCOUNTERS\BDMV\STREAM\closeencountersdcdtshdto.lpcm"...
[a04] The last DTS frame is incomplete and thus gets skipped.
[a04] The original audio track has a constant bit depth of 24 bits.
[a04] Audio overlaps for 7ms at playtime 0:23:06.
[a04] Audio overlaps for 6ms at playtime 0:26:06.
[a04] Audio overlaps for 11ms at playtime 0:30:37.
[a04] Audio overlaps for 6ms at playtime 0:34:03.
[a04] Audio overlaps for 6ms at playtime 0:37:20.
[a04] Audio overlaps for 7ms at playtime 0:41:25.
[a04] Audio overlaps for 13ms at playtime 0:45:19.
[a04] Audio overlaps for 9ms at playtime 0:58:47.
[a04] Audio overlaps for 12ms at playtime 1:03:58.
[a04] Audio overlaps for 6ms at playtime 1:04:39.
[a04] Audio overlaps for 8ms at playtime 1:13:48.
[a04] Audio overlaps for 10ms at playtime 1:15:44.
[a04] Audio overlaps for 6ms at playtime 1:16:41.
[a04] Audio overlaps for 5ms at playtime 1:20:28.
[a04] Audio overlaps for 9ms at playtime 1:21:15.
[a04] Audio overlaps for 9ms at playtime 1:34:00.
[a04] Audio overlaps for 9ms at playtime 1:34:34.
[a04] Audio overlaps for 10ms at playtime 2:11:04.
The temp file could not be interpreted correctly.

Ive went back to my previously working method using your eac3to v2.52 and all is well..ive just to manually re-run the same command to ensure the RAW/PCM gaps are realized.

[canTsTop]

hello, i am new to eac3to

here is sample http://www.mediafire.com/file/kynimztnjwe/333_0_0.ts (~65mb), duration is 3mn 34s, but after i demux it with eac3to, its only few seconds
here is log http://www.paste.lt/paste/092a823690...e882c9b04966a3
it says The source file is encrypted. <WARNING>, but its not encrypted


here is another 2 samples with many errors http://www.mediafire.com/file/iqom3m...th_errors_2.ts (10mb) and http://www.mediafire.com/?zumjnyzywdy (24mb)
after i dumux them, audio is few seconds shorter then mediainfo reported on TS. is this normal?


can this tool repair video? thank you.

[mbcd]

Originally Posted by canTsTop

hello, i am new to eac3to

here is sample http://www.mediafire.com/file/kynimztnjwe/333_0_0.ts (~65mb), duration is 3mn 34s, but after i demux it with eac3to, its only few seconds
here is log http://www.paste.lt/paste/092a823690...e882c9b04966a3
it says The source file is encrypted. <WARNING>, but its not encrypted


here is another 2 samples with many errors http://www.mediafire.com/file/iqom3m...th_errors_2.ts (10mb) and http://www.mediafire.com/?zumjnyzywdy (24mb)
after i dumux them, audio is few seconds shorter then mediainfo reported on TS. is this normal?


can this tool repair video? thank you.

Your posted File (65mb-Sample) ist damaged during BAD-Signal or missing decryption for a short moment. Possibly recorded over DVB-S ?

Try to repair it with Project-X or something else.
Because of this error eac3to thinks its encrypted. Maybe there is realy a single Frame where decryption failed. I dont know if this TV-Broadcaster encypts.

Normaly something like this is not repairable. Best is to delete the damaged part and keep the rest.

eac3to can not repair such errors.

[Snowknight26]

It's not possible to go from DTSWAV -> DTS without some type of conversion, is it?

[nautilus7]

Do you mean w/o dts encoding? It is possible. Just do eac3to input.wav output.dts. If the wav is dtswav then the wav header will be stripped and you'll get raw dts file.

P.S maybe you need -768 if your dtswav is 768kbps, but i'm not sure.

[jj666]

Hello Madshi,

Found a strange error with 3.01 whilst remuxing two VC1 HD-DVDs I had, Shaun Of The Dead and Motorhead Stage Fright.

Demuxing the video stream (and removing pulldown) with 3.01 produces an un-usable file reported as 8168:6118p in TSMUXER. No problem at all with 2.87 and the same file/settings - reported as 1920:1080p in TSMUXER.

In case this was a problem with TSMUXER, I tried to play the VC1 stream with DGVC1DECNV which resulted in the video driver crashing. Again, no problems with the VC1 stream demuxed with 2.87.

Log files:

eac3to v3.01
command line: "E:\utils\encoding\eac3to\eac3to.exe"  "E:\bd temp\Motoerhead_Stage_Fright\" 1) 3: "E:\bd temp\Motoerhead_Stage_Fright\video3.01.vc1"
------------------------------------------------------------------------------
EVO, 2 video tracks, 3 audio tracks, 1:29:59
"MainFeature"
1: Joined EVO file
2: Chapters, 22 chapters without names
3: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
4: VC-1, 480i48 /1.001 (3:2)
5: DTS Hi-Res, English, 5.1 channels, 24 bits, 2046kbps, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
6: DTS Master Audio, English, 2.0 channels, 24 bits, 48khz
(core: DTS, 2.0 channels, 24 bits, 768kbps, 48khz)
7: DTS, English, 5.1 channels, 24 bits, 768kbps, 48khz
[v03] Extracting video track number 3...
[v03] There's no valid framerate in this bitstream.  <WARNING>
[v03] Writing new framerate "24fps /1.001" to bitstream.
[v03] Writing new framerate "24fps /1.001" to bitstream.
[v03] Removing VC-1 pulldown...
[v03] Creating file "E:\bd temp\Motoerhead_Stage_Fright\video3.01.vc1"...
Video track 3 contains 129453 frames.
Video track 4 contains 161802 frames.
eac3to processing took 14 minutes, 12 seconds.
Done.

eac3to v2.87
command line: "E:\utils\encoding\eac3to\eac3to2.87.exe"  "E:\bd temp\Motoerhead_Stage_Fright\" 1) 3: "E:\bd temp\Motoerhead_Stage_Fright\video.vc1"
------------------------------------------------------------------------------
EVO, 2 video tracks, 3 audio tracks, 1:29:59
"MainFeature"
1: Joined EVO file
2: Chapters, 22 chapters without names
3: VC-1, 1080p24 /1.001 (16:9) with pulldown flags
4: VC-1, 480i60 /1.001 (3:2)
5: DTS Hi-Res, English, 5.1 channels, 24 bits, 2046kbps, 48khz
   (core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
6: DTS Master Audio, English, 2.0 channels, 24 bits, 48khz
   (core: DTS, 2.0 channels, 24 bits, 768kbps, 48khz)
7: DTS, English, 5.1 channels, 24 bits, 768kbps, 48khz
[v03] Extracting video track number 3...
[v03] Removing VC-1 pulldown...
[v03] Creating file "E:\bd temp\Motoerhead_Stage_Fright\video.vc1"...
Video track 3 contains 129453 frames.
Video track 4 contains 161802 frames.
eac3to processing took 10 minutes, 58 seconds.
Done.

Cheers,

-jj-

[setarip_old]

@jj666

Hi!

I know nothing about this software but I see that line 4 of the two logs include different values - 480i48 and 480i60...

[ragboy]

Originally Posted by jj666

Hello Madshi,

Found a strange error with 3.01 whilst remuxing two VC1 HD-DVDs I had, Shaun Of The Dead and Motorhead Stage Fright.

Demuxing the video stream (and removing pulldown) with 3.01 produces an un-usable file reported as 8168:6118p in TSMUXER. No problem at all with 2.87 and the same file/settings - reported as 1920:1080p in TSMUXER.

In case this was a problem with TSMUXER, I tried to play the VC1 stream with DGVC1DECNV which resulted in the video driver crashing. Again, no problems with the VC1 stream demuxed with 2.87.

-jj-

I am having the same problem with Apollo 13, and another title, but I don't have 2.87 to test with that.

[Snowknight26]

Originally Posted by ragboy

I am having the same problem with Apollo 13

I can confirm it. After demuxing the VC-1 track muxing the first 100MB of the EVOs then muxing it with mkvmerge, libavcodec (from ffdshow r2594) crashes as soon as the file starts playing. MPC-HC r969 (internal VC-1 decoder with DXVA enabled) plays it until you stop the video, upon which it also crashes.

[sidekick2]

I can confirm that vc1 pulled from Bourne Ultimatum hd-dvd also shows up as 8168x6188 res, and won't play in anything when muxed in tsmuxer.

"Profile Advanced@3:Resolution: 8168:6118p Frame rate 23.976"

This is with version 3.01.

[madshi]

Originally Posted by KevinMcPool

Sorry madshi unfortunately 3.01 doesnt produce a log when i ask it to convert dtshd to lpcm from the seamless branching titles

My fault. Will be fixed in the next build.

Originally Posted by KevinMcPool

2.87-builds the lpcm file from the dtshd but then deletes it at the point of the required second pass leaving only a log file with the error message "The temp file could not be interpreted correctly"

Will be fixed in the next build.

Originally Posted by canTsTop

here is sample http://www.mediafire.com/file/kynimztnjwe/333_0_0.ts (~65mb), duration is 3mn 34s, but after i demux it with eac3to, its only few seconds
here is log http://www.paste.lt/paste/092a823690...e882c9b04966a3
it says The source file is encrypted. <WARNING>, but its not encrypted

here is another 2 samples with many errors http://www.mediafire.com/file/iqom3m...th_errors_2.ts (10mb) and http://www.mediafire.com/?zumjnyzywdy (24mb)
after i dumux them, audio is few seconds shorter then mediainfo reported on TS. is this normal?

can this tool repair video? thank you.

Thanks for the samples, but as mbcd already hinted, these files seem to be severely damaged, or maybe only partially decrypted. eac3to can not repair any such errors. It can only skip over errors. Which means that if the source is so much damaged, the output of eac3to can be much shorter than the source file claims to be. There's nothing I can do about that. eac3to's error tolerance just means that *minor* errors in a source file are properly ignored now (not repaired). If you have a really strongly damaged source file, just trash and rerecord.

Originally Posted by nautilus7

Do you mean w/o dts encoding? It is possible. Just do eac3to input.wav output.dts. If the wav is dtswav then the wav header will be stripped and you'll get raw dts file.

P.S maybe you need -768 if your dtswav is 768kbps, but i'm not sure.

The switch is not necessary. eac3to never reencodes, unless you ask it to.

Normally eac3to does detect dtswav automatically, but there are cases where the detection does not work. Maybe I can improve on that in a future version. For now you can use DtsParser to convert dtswav files to dts. And eac3to can then in any case decode them (if needed).

Originally Posted by jj666

Found a strange error with 3.01 whilst remuxing two VC1 HD-DVDs I had, Shaun Of The Dead and Motorhead Stage Fright.

Demuxing the video stream (and removing pulldown) with 3.01 produces an un-usable file reported as 8168:6118p in TSMUXER. No problem at all with 2.87 and the same file/settings - reported as 1920:1080p in TSMUXER.

Originally Posted by ragboy

I am having the same problem with Apollo 13, and another title

Originally Posted by Snowknight26

I can confirm it.

Originally Posted by sidekick2

I can confirm that vc1 pulled from Bourne Ultimatum hd-dvd also shows up as 8168x6188 res, and won't play in anything when muxed in tsmuxer.

Sorry guys, BAD bug. Will be fixed in next build.

[madshi]

eac3to v3.02 released

http://madshi.net/eac3to.zip

* fixed: VC-1 stream handling was broken
* fixed: destination file extension "*.lpcm" didn't work with 2pass processing
* fixed: MPEG2 1088 to 1080 cropping was incomplete
* fixed: no log was being created when "temp file could not be interpreted"

[madshi]

eac3to v3.03 released

http://madshi.net/eac3to.zip

* fixed: MPEG2 1088 to 1080 cropping was still incomplete

[ragboy]

Originally Posted by madshi

eac3to v3.02 released

http://madshi.net/eac3to.zip

* fixed: VC-1 stream handling was broken
* fixed: destination file extension "*.lpcm" didn't work with 2pass processing
* fixed: MPEG2 1088 to 1080 cropping was incomplete
* fixed: no log was being created when "temp file could not be interpreted"

Thanks for the quick update, great support.

[Chumbo]

@madshi,
This is an aesthetic issue so no hurry, but I noticed that v3 overrides the DOS window's color settings. Would you consider not overriding this please? Thank you.

[ragboy]

I am using eac3to to recode DTS or EAC3 and then tsmuxer to mux to m2ts and then to Handbrake. Handbrake no longer crashes, but handbrake sees the video as 2x the length it should be. If I open the bluray stream direct, I don't have the problem, but I can't read the audio. I have a small EVO file, 65 megs, unilogo.evo, that I use to test, and eac3to 3.03 doesn't seem to work with it. I can upload the file somewhere if you like.

[Momber]

Originally Posted by jj666

Found a strange error with 3.01 whilst remuxing two VC1 HD-DVDs I had, Shaun Of The Dead and Motorhead Stage Fright.

For starters, you need to use -keeppulldown with Motörhead Stagefright. It's not a 24p encoding.

[rack04]

I have a raw h264 stream that was encoded using x264 with the following avs script:

DirectShowSource("C:\Personal\Videos\sample.mkv", fps=23.976, audio=false)
Spline64Resize(1280,720)

When I take the raw h264 and mux to mkv using eac3to v3.03 it is reported as 23 fps.

http://www.mediafire.com/download.php?mj1tuomhmul

eac3to v3.03
command line: eac3to "C:\Personal\Videos\sample-output.h264" "C:\Personal\Videos\sample-output.mkv"
------------------------------------------------------------------------------
h264/AVC, 720p23 (16:9)
Muxing video to Matroska...
Added fps value to MKV header.
Video track 1 contains 256 frames.
eac3to processing took 1 second.
Done.

[n0mag!c]

very much for your efforts!

Originally Posted by madshi

That VC-1 stream does not have any framerate information in it. That made eac3to crash. The next build will handle this gracefully. But it will "guess" the framerate to be 23.976.

Could you please add force writting framerate information to such streams? It would be very useful!

Originally Posted by madshi

There's no way to detect and fix clipping in one pass. However, you can manually lower volume by adding e.g. "-3db" to the command line. In most cases that should make sure that no clipping occurs, so no 2nd pass will be necessary.

Lowering volume is not an option, at least for me.
How exactly elimination of clipping works? If in "eac3to" there will be write buffers for output files, then how big they must be to avoid 2nd pass running? Of course if you've decided to implement this.

[n0mag!c]

Originally Posted by madshi

-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", ...
-resampleTo48000 resample audio to "44100", "48000" or "96000" Hz

Is it possible to implement possibility to use ANY values for this switches? Then your tool will be ULTIMATE for resampling purposes.
P.S. Is "/" symbol present there by mistake?

[spida_singh]

Just demuxed dark knight blu ray (UK), using hd stream extractor. Video demuxed fine to mkv, audio saved as .thd (not THD+AC3). The resulting audio file is not accepted in tsmuxer,tsremux, or mkvmerge. I know Tsmuxer has issues with trued hd, but, it reads the m2ts I remuxed to an m2ts, direct from disc using TS4np 0.82. What i want to know is, does the true hd file needs to be in a container for it to be recognised in tsmuxer and ts4np (Tsremux), is it possible to take the mkv created from eac3to, and add the truehd track to make my final mkv within mvkmerge? I have deleted the log, but i know there were not any errors in the log, just need to know if anyone knows this from the top of their head, if not, ill redo the demux and post logs.

[madshi]

Originally Posted by Chumbo

This is an aesthetic issue so no hurry, but I noticed that v3 overrides the DOS window's color settings. Would you consider not overriding this please?

This is slightly problematic. eac3to uses colored output now. By doing so I have to overwrite the default settings, it seems. I could try to restore the original settings after eac3to has run through. Would that solve the problem? Or how else would you want to have the problem solved?

Originally Posted by ragboy

I am using eac3to to recode DTS or EAC3 and then tsmuxer to mux to m2ts and then to Handbrake. Handbrake no longer crashes, but handbrake sees the video as 2x the length it should be. If I open the bluray stream direct, I don't have the problem, but I can't read the audio. I have a small EVO file, 65 megs, unilogo.evo, that I use to test, and eac3to 3.03 doesn't seem to work with it. I can upload the file somewhere if you like.

I don't even know Handbrake, and I can't tell you where the problem is exactly. You should try to find out which of the software involved is guilty for the problem. Is it eac3to? I doubt it, but if you see any indication for that, just let me know.

Originally Posted by rack04

I have a raw h264 stream that was encoded using x264 with the following avs script:

DirectShowSource("C:\Personal\Videos\sample.mkv", fps=23.976, audio=false)
Spline64Resize(1280,720)

When I take the raw h264 and mux to mkv using eac3to v3.03 it is reported as 23 fps.

eac3to only accepts "24000/1001" as "24". If you use "23976/1000" that is displayed as "23". You should try encoding the h264 stream as "24000/1001" instead of 23.976. I can't really tell you how to do that, though, since I never used x264 myself yet...

Originally Posted by n0mag!c

Could you please add force writting framerate information to such streams?

Maybe.

Originally Posted by n0mag!c

Lowering volume is not an option, at least for me.

Well, lowering the volume is the ONLY option you have if there is clipping.

Originally Posted by n0mag!c

How exactly elimination of clipping works?

Volume is lowered by exactly the amount that is necessary to have no clipping, anymore.

Originally Posted by n0mag!c

If in "eac3to" there will be write buffers for output files, then how big they must be to avoid 2nd pass running?

The whole audio file!

Originally Posted by n0mag!c

Is it possible to implement possibility to use ANY values for this switches? Then your tool will be ULTIMATE for resampling purposes.
P.S. Is "/" symbol present there by mistake?

No, it's not possible, because the SSRC resampling library does not support "any" values. The "/..." in the help means that "23.976" is not the only supported value.

Originally Posted by spida_singh

Just demuxed dark knight blu ray (UK), using hd stream extractor. Video demuxed fine to mkv, audio saved as .thd (not THD+AC3). The resulting audio file is not accepted in tsmuxer,tsremux, or mkvmerge.

mkvmerge does not support TrueHD at all. tsMuxeR only supports "TrueHD+AC3", and even that only buggy. Don't know about TsRemux.

[shambles]

Originally Posted by madshi

eac3to only accepts "24000/1001" as "24". If you use "23976/1000" that is displayed as "23". You should try encoding the h264 stream as "24000/1001" instead of 23.976. I can't really tell you how to do that, though, since I never used x264 myself yet...

adding 'assumefps(24000,1001)' into the avs script gives exact 24000/1001 output

[n0mag!c]

Originally Posted by madshi

Volume is lowered by exactly the amount that is necessary to have no clipping, anymore.

The whole audio file!

Is this means that volume is lowered for whole audio file??? I guessed not!

[laserfan]

Originally Posted by rack04

I have a raw h264 stream that was encoded using x264 with the following avs script:

DirectShowSource("C:\Personal\Videos\sample.mkv", fps=23.976, audio=false)
Spline64Resize(1280,720)

Originally Posted by madshi

...eac3to only accepts "24000/1001" as "24". If you use "23976/1000" that is displayed as "23". You should try encoding the h264 stream as "24000/1001" instead of 23.976. I can't really tell you how to do that, though, since I never used x264 myself yet...

Originally Posted by shambles

adding 'assumefps(24000,1001)' into the avs script gives exact 24000/1001 output

I'm confused. Can someone explain the difference between:

1. Using the --24000/... option in eac3to vs.

2. the fps= option of DSS e.g. DirectShowSource(clip, fps=23.0976) vs.

3. assumefps e.g

DirectShowSource(clip)
AssumeFPS(24000,1001)

If the original raw h264 file contains info that IDs its framerate, do you need ANY of these 1, 2 or 3 when demuxing with eac3to and then re-encoding with x264?

[n0mag!c]

Originally Posted by laserfan

If the original raw h264 file contains info that IDs its framerate, do you need ANY of these 1, 2 or 3 when demuxing with eac3to and then re-encoding with x264?

Some manipulations will be needed only when muxing (to mkv) with eac3to.
When muxing with tsMuxer then better check "change fps".

[n0mag!c]

Originally Posted by rack04

When I take the raw h264 and mux to mkv using eac3to v3.03 it is reported as 23 fps.

You can use "demux" option in tsMuxer (yes, with raw stream), check "change fps" option with "24000/1001" and get another stream, which "eac3to" will suppose 24p.

[Jeff Flowerday]

Originally Posted by spida_singh

Just demuxed dark knight blu ray (UK), using hd stream extractor. Video demuxed fine to mkv, audio saved as .thd (not THD+AC3). The resulting audio file is not accepted in tsmuxer,tsremux, or mkvmerge. I know Tsmuxer has issues with trued hd, but, it reads the m2ts I remuxed to an m2ts, direct from disc using TS4np 0.82. What i want to know is, does the true hd file needs to be in a container for it to be recognised in tsmuxer and ts4np (Tsremux), is it possible to take the mkv created from eac3to, and add the truehd track to make my final mkv within mvkmerge? I have deleted the log, but i know there were not any errors in the log, just need to know if anyone knows this from the top of their head, if not, ill redo the demux and post logs.

eac3to d:\ 1) 1: chapter.txt 2: video.mkv 3: audio.thd+ac3

Of course put the appropriate track numbers in. The file extension thd isn't he same as thd+ac3

[madshi]

Originally Posted by n0mag!c

Is this means that volume is lowered for whole audio file???

Of course it is!

Originally Posted by laserfan

If the original raw h264 file contains info that IDs its framerate, do you need ANY of these 1, 2 or 3 when demuxing with eac3to and then re-encoding with x264?

This is not really the right thread to ask about how to reencode with x264. I can't help you with that at all.

[asarian]

Originally Posted by madshi

Well, lowering the volume is the ONLY option you have if there is clipping.

Pardon my ignorance in the matter, but I take it clipping is the result of a decoder chopping off data above a certain volume threshold? (Probably according to Dolby specs). So, does this happen in your amp? Or already in, say, the Arcsoft DTS decoder? (as in, effectively, losing data).

It just seems hard to imagine tracks would be released that an official decoder can't decode properly without having to chop off data.

[Chumbo]

Originally Posted by madshi

This is slightly problematic. eac3to uses colored output now. By doing so I have to overwrite the default settings, it seems. I could try to restore the original settings after eac3to has run through. Would that solve the problem? Or how else would you want to have the problem solved?

Restoring it would be nice as I can live with it being different during the execution. I'm curious though, can you not get that info prior and just use the defaults for the session? What about the background, i.e., is there a setting that allows you to use something like transparent? Any how, no big deal, restoring it would be great. Thank you.

[KevinMcPool]

for the update which (of course ) fixes the dtshd to lpcm conversion in seamless branching titles and produces a log file

[laserfan]

Originally Posted by madshi

This is not really the right thread to ask about how to reencode with x264. I can't help you with that at all.

Ok, there's nothing in the help file, and no readme with this program, so I had to search this thread to find out what your -fps option is really FOR:

Let me explain the new FPS changing options:

(1) If you have a source which contains FPS information, you don't need to tell eac3to which FPS the source has. E.g. if you feed eac3to a m2ts, TS, EVO or VOB file with a video track in it, eac3to will know which FPS the video and audio tracks have. So in this case you can just use "-slowdown" to convert both 24.000 and 25.000 movies to 23.976 fps. Or you can use "-speedup" to convert both 23.976 and 24.000 movies to 25.000 fps.

(2) If eac3to does not know the source FPS (which is usually the case if you feed eac3to with a demuxed audio track), you can still use the "-slowdown" and "-speedup" options. If you do that, eac3to will apply 25.000 -> 23.976 (slowdown) or 23.976 -> 25.000 (speedup) conversion.

(3) If you want to do any funny conversions. E.g. if you want to convert a 25.000 source to 24.000, you can use the option "-changeTo24.000". If eac3to knows the source FPS, that's all you need to do. If eac3to doesn't know the source FPS, you should do "-25.000 -changeTo24.000" to do a 25.000 -> 24.000 conversion.

(4) Generally, if the source is a container which contains both video and audio tracks, doing either "eac3to source -demux -anyFpsChangeOptions" or "eac3to source some.mkv -anyFpsChangeOptions" will result in eac3to doing FPS conversion for all audio and video tracks. Audio tracks will be transcoded in this situation. Lossless tracks will be transcoded to 24bit FLAC. Lossy tracks will be transcoded to 640kbps AC3.

(5) Video FPS changes will as usual not only result in adjusted MKV timestamps. eac3to will also automatically adjust the video bitstream itself to reflect the FPS value change.

So, at least wrt the fps options of eac3to, the ONLY TIME one needs to use the -23.976 is when the source file contains no fps information, and you want to CHANGE the frame rate. Correct?

I still don't understand your answer to rack04--I might've guessed it would just be "yes 23 is correct for your encoding".

[shanghai2004]

Madshi,

With v3.03 found minor issue with the handling of audio gaps/overlaps in EAC3 audio tracks.

1: h264/AVC, 1080i60 /1.001 (16:9)
2: E-AC3, 5.1 channels, 3024kbps, 48khz, -67ms
3: E-AC3, 2.0 channels, 448kbps, 48khz, -67ms
4: Subtitle (VobSub)

When giving just the command eac3to c:\movie e:\movie.mkv, normally, eac3to will process all audio tracks as well. The audio is output as .EAC3 files.

When there is a gap or overlap detected, eac3to wants to do a 2nd pass to correct the audio. This fails because eac3to seems not able to process gaps in EAC3 files.

This works properly (including 2nd pass):
eac3to c:\movie 1: e:\movie_v.mkv 2: movie_6ch.flac

[madshi]

Originally Posted by asarian

Pardon my ignorance in the matter, but I take it clipping is the result of a decoder chopping off data above a certain volume threshold? (Probably according to Dolby specs). So, does this happen in your amp? Or already in, say, the Arcsoft DTS decoder? (as in, effectively, losing data).

It just seems hard to imagine tracks would be released that an official decoder can't decode properly without having to chop off data.

Clipping means that the audio data gets over a specific threshold somewhere during its runtime. When that happens, the resulting audio file would either be invalid, or it the audio samples which are too "high" would have to be shortened ("clipped"). This mostly happens if you do processing. E.g. if you do "-slowdown", clipping can happen sometimes. The decoders themselves usually do not output clipped results. The ArcSoft/Nero decoders output integer samples, where clipping is not really possible technically. The libav decoders outputs floating point where clipping is possible. I have seen clipping from the libav decoders, but I think that's probably because it's not an "official" decoder and probably has output with too high volume. Anyway, eac3to can now fix clipping without any negative side effects.

Originally Posted by Chumbo

Restoring it would be nice as I can live with it being different during the execution. I'm curious though, can you not get that info prior and just use the defaults for the session? What about the background, i.e., is there a setting that allows you to use something like transparent?

Ok, so what if you configured your DOS box to show black font and white background? If I then wanted to draw white text on a transparent background, it would be invisible. You see, I can't use your defaults and then try multi-colored text output, because the number of colors possible is very limited and there'd be high danger of me accidently using a color which is invisible on your specific background color. That's why I have to redo the whole color scheme. Or I could think of a specific set of foreground colors for every possible background color you could have possibly specified. But I won't go there...

Originally Posted by shanghai2004

When there is a gap or overlap detected, eac3to wants to do a 2nd pass to correct the audio. This fails because eac3to seems not able to process gaps in EAC3 files.

eac3to should be able to fix gaps/overlaps in E-AC3 files just fine. Please post your full log.

[shanghai2004]

Originally Posted by madshi

eac3to should be able to fix gaps/overlaps in E-AC3 files just fine. Please post your full log.

No problem...

eac3to v3.03
command line: c:\eac3to\eac3to hv001t01.evo+hv001t02.evo+hv001t03.evo+hv001t04.evo+hv001t05.evo+hv001t06.evo+
hv001t07.evo+hv001t08.evo+hv001t09.evo+hv001t10.evo+hv001t11.evo+hv001t12.evo+
hv001t13.evo+hv001t14.evo+hv001t15.evo+hv001t16.evo+hv001t17.evo+hv001t18.evo+
hv001t19.evo+hv001t20.evo+hv001t21.evo+hv001t22.evo+hv001t23.evo+hv001t24.evo+
hv001t25.evo+hv001t26.evo e:\chicago.mkv
------------------------------------------------------------------------------
EVO, 1 video track, 2 audio tracks, 1 subtitle track, 14:47:35
1: Joined EVO file
2: h264/AVC, 1080i60 /1.001 (16:9)
3: E-AC3, 5.1 channels, 3024kbps, 48khz, -67ms
4: E-AC3, 2.0 channels, 448kbps, 48khz, -67ms
5: Subtitle (VobSub)
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a03] Applying (E-)AC3 delay...
[a04] Extracting audio track number 4...
[a04] Applying (E-)AC3 delay...
[v02] Muxing video to Matroska...
[a03] Creating file "e:\chicago - 3 - E-AC3, 5.1 channels, 3024kbps, 48khz.eac3"...
[a04] Creating file "e:\chicago - 4 - E-AC3, 2.0 channels, 448kbps, 48khz.eac3"...
[s05] Extracting subtitle track number 5...
[s05] Creating file "e:\chicago - 5 - Subtitle (VobSub).sup"...
[a03] Audio overlaps for 62ms at playtime 1:19:21. <WARNING>
[a04] Audio overlaps for 100ms at playtime 1:19:21. <WARNING>
[a03] Starting 2nd pass...
[a03] Creating silent AC3 frame failed. <WARNING>
[a03] Realizing (E-)AC3 gaps...
[a03] Creating file "e:\chicago - 3 - E-AC3, 5.1 channels, 3024kbps, 48khz.eac3"...
[a04] Starting 2nd pass...
[a04] Creating silent AC3 frame failed. <WARNING>
[a04] Realizing (E-)AC3 gaps...
[a04] Creating file "e:\chicago - 4 - E-AC3, 2.0 channels, 448kbps, 48khz.eac3"...
Added fps value to MKV header.
eac3to processing took 39 minutes, 39 seconds.
Done.

The 2nd pass took less then a second, so was not done properly probably.

[asarian]

Originally Posted by madshi

Clipping means that the audio data gets over a specific threshold somewhere during its runtime. When that happens, the resulting audio file would either be invalid, or it the audio samples which are too "high" would have to be shortened ("clipped"). This mostly happens if you do processing. E.g. if you do "-slowdown", clipping can happen sometimes. The decoders themselves usually do not output clipped results. The ArcSoft/Nero decoders output integer samples, where clipping is not really possible technically. The libav decoders outputs floating point where clipping is possible. I have seen clipping from the libav decoders, but I think that's probably because it's not an "official" decoder and probably has output with too high volume. Anyway, eac3to can now fix clipping without any negative side effects.

Thanks for the in-depth explanation. Much appreciated.

[ragboy]

I have a quick question. When I check a title, it shows the audio delay of a track, like say (7ms). When I transcode to AC3, it says it is applying the RAW/LPCM delay. Does this mean that I don't need to appy that 7ms when remuxing? Does eac3to apply that delay on transcoding, so I don't need to adjust when I mux?

[madshi]

Originally Posted by shanghai2004

[a03] Creating silent AC3 frame failed. <WARNING>

That's a bug, but it has no negative effect.

Originally Posted by shanghai2004

[a03] Realizing (E-)AC3 gaps...
[a03] Creating file "e:\chicago - 3 - E-AC3, 5.1 channels, 3024kbps, 48khz.eac3"...
[a04] Starting 2nd pass...
[a04] Creating silent AC3 frame failed.  <WARNING>
[a04] Realizing (E-)AC3 gaps...
[a04] Creating file "e:\chicago - 4 - E-AC3, 2.0 channels, 448kbps, 48khz.eac3"...
Added fps value to MKV header.
eac3to processing took 39 minutes, 39 seconds.
Done.

The 2nd pass took less then a second, so was not done properly probably.

2nd pass processing can be very fast, depending on the circumstances. If eac3to says 2nd pass processing succeeded, then it most probably did.

Originally Posted by ragboy

I have a quick question. When I check a title, it shows the audio delay of a track, like say (7ms). When I transcode to AC3, it says it is applying the RAW/LPCM delay. Does this mean that I don't need to appy that 7ms when remuxing? Does eac3to apply that delay on transcoding, so I don't need to adjust when I mux?

That's right. eac3to already applies the delay for you. Generally when using eac3to you usually don't need to worry about delays. The only exception is when eac3to creates a file for you with "DELAY" in the file name. In that case eac3to was not able to apply the delay. But the usually only happens for TrueHD tracks.

[rack04]

When muxing raw h264 stream to mkv using eac3to it doesn't appear to include the "muxing mode" information in the container. Verified using mediainfo. I understand this isn't anything important, just thought I'd pass it along.

For example:

Raw h264 stream to mkv using mkvmerge:

Video
Format                           : AVC
Format/Info                      : Advanced Video Codec
Format profile                   : High@L4.1
Format settings, CABAC           : Yes
Format settings, ReFrames        : 3 frames
Muxing mode                      : Container profile=Unknown@4.1
Codec ID                         : V_MPEG4/ISO/AVC
Duration                         : 1mn 59s
Bit rate                         : 3 182 Kbps
Nominal bit rate                 : 3 324 Kbps
Width                            : 1 280 pixels
Height                           : 688 pixels
Display aspect ratio             : 1.860
Frame rate                       : 23.976 fps
Resolution                       : 24 bits
Colorimetry                      : 4:2:0
Scan type                        : Progressive
Bits/(Pixel*Frame)               : 0.151
Writing library                  : x264 core 65 r1074M b6bb3d4

Raw h264 stream to mkv using eac3to:

Video
Format                           : AVC
Format/Info                      : Advanced Video Codec
Format profile                   : High@L4.1
Format settings, CABAC           : Yes
Format settings, ReFrames        : 3 frames
Muxing mode                      : Container profile=Unknown@0.0
Codec ID                         : V_MPEG4/ISO/AVC
Duration                         : 1mn 59s
Bit rate                         : 3 181 Kbps
Nominal bit rate                 : 3 324 Kbps
Width                            : 1 280 pixels
Height                           : 688 pixels
Display aspect ratio             : 1.860
Frame rate                       : 23.976 fps
Resolution                       : 24 bits
Colorimetry                      : 4:2:0
Scan type                        : Progressive
Bits/(Pixel*Frame)               : 0.151
Writing library                  : x264 core 65 r1074M b6bb3d4

[Nullity]

Originally Posted by madshi

That's right. eac3to already applies the delay for you. Generally when using eac3to you usually don't need to worry about delays. The only exception is when eac3to creates a file for you with "DELAY" in the file name. In that case eac3to was not able to apply the delay. But the usually only happens for TrueHD tracks.

But I thought eac3to could only apply delays in 32ms frames or "chunks"? So if his source audio stream had a delay of say +/- 7ms, what operation would be performed on that stream exactly? Would it not touch it, since 7ms is close to no delay, maybe only if it was +7? If -7, would it apply +32ms resulting in a final delay of +25ms? Are different audio types handled differently?

I have an HDDVD with an EAC3 track that has a delay of -1001ms. When I processed it (which was a few months ago, so an older version of eac3to, don't remember which), it said it took care of the delay, yet there is still a delay, just not -1001ms. I have not been able to tell exactly how much, but through trial and error, I have narrowed it down to somewhere around 100ms.

Could you please add a feature which always adds the correct delay amount to either the file name and/or the log, regardless of if it was processed or not? Even if eac3to was able to process the file correctly to a 0ms delay, that would still be helpful to see.

[rebkell]

Originally Posted by Nullity

But I thought eac3to could only apply delays in 32ms frames or "chunks"? So if his source audio stream had a delay of say +/- 7ms, what operation would be performed on that stream exactly? Would it not touch it, since 7ms is close to no delay, maybe only if it was +7? If -7, would it apply +32ms resulting in a final delay of +25ms? Are different audio types handled differently?

I have an HDDVD with an EAC3 track that has a delay of -1001ms. When I processed it (which was a few months ago, so an older version of eac3to, don't remember which), it said it took care of the delay, yet there is still a delay, just not -1001ms. I have not been able to tell exactly how much, but through trial and error, I have narrowed it down to somewhere around 100ms.

Could you please add a feature which always adds the correct delay amount to either the file name and/or the log, regardless of if it was processed or not? Even if eac3to was able to process the file correctly to a 0ms delay, that would still be helpful to see.

32ms is the length of an AC3 sample @48KHz, the various audio samples vary in length, 48KHz aac is 21.3333ms in length, Madshi can tell you the others, a lot of them are in the thread somewhere. The delay will be gotten as close as is possible to the nearest sample size.

I think I read somewhere that about 16ms is as close as anyone can determine a/v sync, could be even bigger, anyone know what the number is?

[Chumbo]

Originally Posted by madshi

Ok, so what if you configured your DOS box to show black font and white background? If I then wanted to draw white text on a transparent background, it would be invisible. You see, I can't use your defaults and then try multi-colored text output, because the number of colors possible is very limited and there'd be high danger of me accidently using a color which is invisible on your specific background color. That's why I have to redo the whole color scheme. Or I could think of a specific set of foreground colors for every possible background color you could have possibly specified. But I won't go there...

Yep, that's pretty much a given, i.e., that your colors may not match what a user may have configured, but it's the deal in the DOS world. I understand that I may need to change them if you go that route which is why if you just reset after eac3to runs, that should be fine...and less of a headache for you.

[madshi]

Originally Posted by rack04

When muxing raw h264 stream to mkv using eac3to it doesn't appear to include the "muxing mode" information in the container. Verified using mediainfo. I understand this isn't anything important, just thought I'd pass it along.

I have no control over that, I think. I'm using Haali's Matroska Muxer to do the muxing.

Originally Posted by Nullity

But I thought eac3to could only apply delays in 32ms frames or "chunks"? So if his source audio stream had a delay of say +/- 7ms, what operation would be performed on that stream exactly? Would it not touch it, since 7ms is close to no delay

When doing pure AC3 demuxing, a delay of +/- 7ms would simply be ignored. No normal human could possible notice such a delay error, anyway. However, if the track is decoded, delay can be done on a per sample base, which means delay exactness can be done in "1/48" ms steps.

Originally Posted by Nullity

I have an HDDVD with an EAC3 track that has a delay of -1001ms. When I processed it (which was a few months ago, so an older version of eac3to, don't remember which), it said it took care of the delay, yet there is still a delay, just not -1001ms. I have not been able to tell exactly how much, but through trial and error, I have narrowed it down to somewhere around 100ms.

There's no way on earth eac3to would leave 100ms delay in the stream without telling you. I'd guess that either the HD DVD was authored badly, or that for whatever reason eac3to calculated a slightly incorrect delay value. Timestamps in HD DVD are less exact as with Blu-Ray, so it's more or a guess in HD DVD handling compared to Blu-Ray.

Originally Posted by Nullity

Could you please add a feature which always adds the correct delay amount to either the file name and/or the log, regardless of if it was processed or not? Even if eac3to was able to process the file correctly to a 0ms delay, that would still be helpful to see.

IMO the file names are long enough as they are. Also I don't want to get people confused. I can already see getting flooded with questions like "do I have to take care about the delay now or not?" and "how can I get rid of those remaining 7ms delay?" etc. No thanks...

Originally Posted by rebkell

I think I read somewhere that about 16ms is as close as anyone can determine a/v sync

Technically with Blu-Ray at least you can calculate delay very exactly, better than 1ms. But whether the studio really authored the disc that well is another question. I often notice that when I compare English vs. German audio tracks in a WAV editor, that they don't match exactly in sync. There are often a few ms difference (*not* caused by eac3to).

A normal human can see a delay mismatch of maybe 30-50ms if he's really good. Many people only see a mismatch of 100ms or bigger.

The best eac3to can do with bitstream audio is half an audio frame, which in the worst case is a raster of 16ms correction for AC3 tracks. All other audio tracks have smaller audio frames than that. So 16ms mismatch is worst case. And as I said, if you ask eac3to to decode/transcode audio, delay can be corrected in 1/48 ms steps.

Originally Posted by Chumbo

Yep, that's pretty much a given, i.e., that your colors may not match what a user may have configured, but it's the deal in the DOS world. I understand that I may need to change them if you go that route which is why if you just reset after eac3to runs, that should be fine...and less of a headache for you.

Ok, will see if I can make that work...

[Nullity]

Originally Posted by madshi

IMO the file names are long enough as they are. Also I don't want to get people confused. I can already see getting flooded with questions like "do I have to take care about the delay now or not?" and "how can I get rid of those remaining 7ms delay?" etc. No thanks...

Fair enough, but could you please add the remaining delay to the log output? For instance, if my source AC3 stream has a delay of -112ms, and eac3to corrects it to -16ms (+3 32ms frames), could that be added to the log somewhere so I know what value to use when I remux the streams? That would be extremely helpful. I, and probably most people, do not know the frame size of all stream types, so doing the math ourselves is not really feasible. In my example case, I know a -16ms delay is not much and might not even be detectable, but it would be beneficial to know this information. eac3to already has to do the calculations, I would think it would be incredibly easy to just append that value to the log output. Even if you have to add a little note next to it saying "It is not necessary to correct a delay less than XXms" so you don't get a bunch of questions.

Originally Posted by madshi

A normal human can see a delay mismatch of maybe 30-50ms if he's really good. Many people only see a mismatch of 100ms or bigger.

I am one of those people who are extremely sensitive to audio delays.

[alc0re]

question about the delay issue...is the delay only fixed if actual transcoding is done? what if there's a delay detected but I'm only extracting the ac3 or dts file? is the delay fixed on an extract only operation?

[ExSport]

Hello all there
Please it is possible somehow to pipe eac3 with e.g. mkfifo or another way?
Many thanks for any info.
Also Madshi, thanks for great app

[Momber]

Originally Posted by madshi

A normal human can see a delay mismatch of maybe 30-50ms if he's really good. Many people only see a mismatch of 100ms or bigger.

That's right. 70ms is the threshold for people without special training.

[nautilus7]

Originally Posted by Nullity

Fair enough, but could you please add the remaining delay to the log output? For instance, if my source AC3 stream has a delay of -112ms, and eac3to corrects it to -16ms (+3 32ms frames), could that be added to the log somewhere so I know what value to use when I remux the streams? That would be extremely helpful. I, and probably most people, do not know the frame size of all stream types, so doing the math ourselves is not really feasible. In my example case, I know a -16ms delay is not much and might not even be detectable, but it would be beneficial to know this information. eac3to already has to do the calculations, I would think it would be incredibly easy to just append that value to the log output. Even if you have to add a little note next to it saying "It is not necessary to correct a delay less than XXms" so you don't get a bunch of questions.

What movie is that?

(Sorry, if you already answered that to any of your previous posts)

[bmnot]

What's the best way to redo 1088 mkvs I made long time ago with gdsmux? I wanna get rid of that gray line. Demux with mkvextractgui, then mux to mkv with eac3to, or mux to ts with tsmuxer and back to mkv with eac3to? or other?

[rebkell]

Originally Posted by Momber

That's right. 70ms is the threshold for people without special training.

Not intending to hijack this thread, but it seems to be a main subject today, I found this interesting and it seems that we're much more acceptable of audio arriving after video than the other way around, we can see it before we hear it and it's not as distracting as hearing it then seeing it

In 1998, ITU-R published BT.1359, recommending the relative timing of sound and vision for broadcasting. Studies by the ITU and others have suggested that the thresholds of timing detectability are about +45ms to -125ms, and the thresholds of acceptability are about +90ms to -185ms. (See Figure 1.)

Other research shows similar but not identical results — and being a function of human perception, we should expect the results to vary. The ATSC Implementation Subcommittee IS-191 has found that under all operational situations, the sound program should never lead the video program by more than 15ms and should never lag the video program by more than 45ms (±15ms). According to the IS, BT.1359 “was carefully considered and found inadequate for purposes of audio and video synchronization for DTV broadcasting.”

http://broadcastengineering.com/audi...ging_lip_sync/

[ggking7]

I'm able to 'eac3to /mnt/cdrom' with a mounted Blu-Ray disc, but 'eac3to /mnt/cdrom 1)' doesn't work. Does that work for anyone else?

[Nullity]

Originally Posted by nautilus7

What movie is that?

(Sorry, if you already answered that to any of your previous posts)

The post you quoted was just a made up example, but my original post asking about the -1001ms delay was from the King Kong HDDVD (US).

[nautilus7]

I have King Kong HD DVD. I think it's the US version (was there a different one?). I'll test and report about sync.

[Momber]

Originally Posted by rebkell

the thresholds of acceptability are about +90ms to -185ms.

Woooooaaah! Not to me they're not!

[laserfan]

Originally Posted by Momber

Woooooaaah! Not to me they're not!

Well then Momber you're obviously not "most people"!

(In fact I should probably add that many of the members here are not "most people" either!)

[Nullity]

Originally Posted by Momber

Woooooaaah! Not to me they're not!

I'm with you man, that would drive me crazy too.

[rebkell]

Originally Posted by Nullity

I'm with you man, that would drive me crazy too.

I agree completely, I would agree that the +15 to -45 would probably be acceptable, but no way would I not be irritated in a big way by nearly a tenth of a second early and nearly 2 tenths late.

[shanghai2004]

Originally Posted by madshi

2nd pass processing can be very fast, depending on the circumstances. If eac3to says 2nd pass processing succeeded, then it most probably did.

Madshi,

Thanks a lot for the reply!
After your reply I listened to the EAC3 tracks carefully, especially around the overlap time, they sound correct.

[madshi]

Originally Posted by Nullity

Fair enough, but could you please add the remaining delay to the log output?

Ok.

Originally Posted by alc0re

question about the delay issue...is the delay only fixed if actual transcoding is done?

Delay is always fixed by eac3to, as has been explained in about every 2nd page of this thread...

Originally Posted by ExSport

Please it is possible somehow to pipe eac3 with e.g. mkfifo or another way?

I don't know mkfifo, but most probably the answer is "no".

Originally Posted by bmnot

What's the best way to redo 1088 mkvs I made long time ago with gdsmux? I wanna get rid of that gray line. Demux with mkvextractgui, then mux to mkv with eac3to, or mux to ts with tsmuxer and back to mkv with eac3to?

Hmmmm... I guess both should work. Don't know which is better. Alternatively you could wait for an eac3to build which allows MKV reading. But I don't know if that feature will ever come...

Originally Posted by ggking7

I'm able to 'eac3to /mnt/cdrom' with a mounted Blu-Ray disc, but 'eac3to /mnt/cdrom 1)' doesn't work. Does that work for anyone else?

Is that Linux? I don't know why it doesn't work, I'm sorry...

[madshi]

[blatant advertising]

HDI has two new products which sound very awesome to me, namely:

- Dune BD Prime
- Dune HD Center



http://www.hdi.co.il/fullhd_players/

Both are full standalone Blu-Ray players, but they also support playing Blu-Ray disc rips from a network resource - including full menu and Java support. Also they support virtually any container / audio codec combination I can think of. Plus they can both decode and bitstream DTS-HD High Resolution, DTS-HD Master Audio, E-AC3 and TrueHD with full 7.1 channels.

E.g. they can play the video track of a MKV/TS/m2ts file together with external (demuxed) AC3, DTS and FLAC audio tracks. Or in other words: With these devices you don't need to mux the audio tracks into the container, anymore. You can just use "eac3to someMovie movie.mkv" and the Dune players will play the result just fine as it is. LOVE that.

Furthermore there's an option for a Gigabit network card, which works around the Sigma Designs chip's bad network speed, so that playing even highest bitrate Blu-Ray movies over (NFS) network connection is not a problem, anymore. Finally, the developers seem to be very open for improvement suggestions. Much more so than I've seen from any other media player company. See here:

http://mpcclub.com/modules.php?name=...wtopic&t=20060

This may finally be the product that can replace my HTPC for good. I'll order a HD Center with Gigabit card. The only thing I don't like is that there's a fan on the Sigma Designs chip...

[/blatant advertising]

Please don't discuss this topic further in this thread. I just wanted to show some support for HDI here, because I think such a promising product has earned it.

[Jom]

Hi.

Windows 7 has the ability to decode eac3 and ac3 natively using "Microsoft DTV-DVD Audio Decoder" aka msmpeg2adec.dll.

Could eac3to utilise this in place of nero / libav ?
I suppose this depends on whether or not it can deal with TrueHD?

I'm guessing these codecs might also make it into vista sp2.

[piratburner]

I have installed Windows 7, and tested eac3to but I can't get nero7 to work with eac3to. any advice to get it to work ?
everything else works, surcode, arcsoft ........

[Jom]

Originally Posted by piratburner

I have installed Windows 7, and tested eac3to but I can't get nero7 to work with eac3to. any advice to get it to work ?
everything else works, surcode, arcsoft ........

When you purchase the Nero HD pack they should email you a serial number you can enter in product license management.
The codecs are't active until you enter a valid serial.
Did you do that already ?

I just let eac3to use libav, the extra fluff installed with nero ain't worth it. Hopefully win7 users could avoid nero and arcsoft altogether if Windows 7 decodes Dolby* and DTS* out the box. (Which it appears to do at the moment - though i'm struggling to find samples of all the different formats in a container i can play)

[piratburner]

Originally Posted by Jom

When you purchase the Nero HD pack they should email you a serial number you can enter in product license management.
The codecs are't active until you enter a valid serial.
Did you do that already ?

Yes I have serial number for....
Blu-ray/HD DVD Video plugin
HD-DVD plugin
BluRay DVD plugin

Is something missing ?

[pbjr]

Originally Posted by ggking7

I'm able to 'eac3to /mnt/cdrom' with a mounted Blu-Ray disc, but 'eac3to /mnt/cdrom 1)' doesn't work. Does that work for anyone else?

whats the error? I do '1)'.

[ExSport]

Originally Posted by madshi

I don't know mkfifo, but most probably the answer is "no".

And do you plan it for the future versions if it is easy to implement it? I mean piping.
Your application is one of few which can recode DTS properly
It will be great if it will be possible to use your eac3to in PMS application(PS3 media streaming)
LINK
Anyway thanks for great app.

[Devrethman]

Originally Posted by ggking7

I'm able to 'eac3to /mnt/cdrom' with a mounted Blu-Ray disc, but 'eac3to /mnt/cdrom 1)' doesn't work. Does that work for anyone else?

Parenthesis are bash operators. Try eac3to /mnt/cdrom 1\)


For some reason when rip blu-rays with eac3to mplayer thinks the end of file is about 4 seconds in. If I run the files through mkvmerge, they work, but if I try to go straight from eac3to to mplayer/mencoder, it always stops around 4 seconds in. Is this normal?

[mrr19121970]

I made a nice GUI to use eac3to...

eacto, tsMuxer & ImgBurn Made Easy

Screen Shots

Select your default locations:


eac3to is called, and the playlists are presented to you. Select one and proceed:


eacto is called, and the video, audio and subtitle streams are presented to you. Select what you want and proceed.


eac3to is called and saves all the streams you selected. a meta & BAT file for tsMuxer are created. If you selected a .ISO image, then ImgBurn is called after after tsMuxer is completed.

Finally a sound is played - especially usefull if you leave you PC unattended, so you can hear it from another room.

[Thunderbolt8]

would be great if there was a similar program which included eac3to and mkvmerge for even easier creation of that final movie .mkv which includes all. madshi, why dont you ask mosu to merge your 2 tools

but good work for that one^^

that "H" symbol though is taken from some russian mouse button key binding tool, isn't it? :P

[sidekick2]

I'm not sure if this is a format issue, eac3to, or tsmuxer... I've extracted the eac3 audio from Bourne Identity HD-DVD, and also created a normal ac3 conversion from the eac3.

No errors in any applications, but my dd+ track doesn't play on the disk in a stand alone or ps3.

In tsmuxer, I have the vc1 file, .ac3 file, .eac3 file, and subtitles. It sees it no problem as an dolby digital+ file, and everything works as it should, creating the blu-ray disk structure.

Is dd+ broken in tsmuxer as well, or is eac3to not outputting a flavor of eac3 that tsmuxer likes, or???

I'm using eac3to 3.03, and tsmuxer 1.8.8b.

According to the wiki, blu-ray supports dolby digital+. I have the dd+ as a 2nd track, and when I select it, the video stutters and no audio is output. Reading the wiki, it says dd+ for blu-ray needs the 640k core ac3 file. My extract line on eac3to was just a 5: h:\bourne\bi.eac3 (for the audio track) Is there a toggle for eac3+ac3 so it includes the core for blu-rays sake, or is it in there already by using the .eac3 tag?

Or is this a tsmuxer issue (I know it has issues with thd) again, or am I just nuts trying to get dolby digital+ working on blu-ray?

Playing the folder in arcsoft works fine on the pc, with both audio tracks.

Madshi, so far 3.03 is great! Thanks!

** update ** I just tried using older version of eac3to, with the same results. I guess tsmuxer doesn't like dd+ either?!?

[asarian]

Madshi,

I'm not even entirely sure eac3to is causing it (though I'm fair certain it does); but when extracting the VC-1 stream of the American Gangster Blu-Ray, which is annoyingly split over dozens of m2ts files, I get the movie just fine, but the 'total time' index is off. Which is to say, it is set to the length of the first part (~13 mins). Even though the movie plays/streams in full, I can't seek (PS3 = 'Go To') beyond the 13 minutes marker.

So, is there a way eac3to could set the correct 'total time' when having fully recompiled all the parts? Obviously, eac3to already knows the correct runtime length, as it indicates so when starting.

Thanks

[idbirch2]

Originally Posted by piratburner

I have installed Windows 7, and tested eac3to but I can't get nero7 to work with eac3to. any advice to get it to work ?
everything else works, surcode, arcsoft ........

It didn't work for me either, in Nero licence management, it clearly shows the BluRay/HDDVD plugin being registered but eac3to just can't detect it. It detects Arcsoft, Sonic and Surcode just fine but something about the Windows 7 architecture is preventing eac3to from accurately checking Nero 7 status. To get around this, I am running eac3to in a WinXP virtual machine.

[~bT~]

^ could it be the environment variables thing again?

[DrNein]

I do not have experience with Windows 7 and Nero 7 however, confirm the following files exist and that the *.ax is registered (i.e. re-register it).

C:\Program Files\Common Files\Ahead\DSFilter\NeAudio2.ax
C:\Program Files\Common Files\Ahead\DSFilter\NeEacDec.dll

Also, the plugin license is not strictly required as other licenses encompass the audio functionality, such as Ultra Edition. As said previous, I think the best option is to install Nero 7 Micro with appropriate license and then add NeEacDec.dll as above. This achieves a minimal install of what remains a great disc writing program without the clutter.

[piratburner]

Originally Posted by DrNein

I do not have experience with Windows 7 and Nero 7 however, confirm the following files exist and that the *.ax is registered (i.e. re-register it).

C:\Program Files\Common Files\Ahead\DSFilter\NeAudio2.ax
C:\Program Files\Common Files\Ahead\DSFilter\NeEacDec.dll

Also, the plugin license is not strictly required as other licenses encompass the audio functionality, such as Ultra Edition. As said previous, I think the best option is to install Nero 7 Micro with appropriate license and then add NeEacDec.dll as above. This achieves a minimal install of what remains a great disc writing program without the clutter.

Checking my system..... and I don't have NeAudio2.ax on my windows 7 system disc ??

But NeEacDec.dll I have


???

[peterjcat]

Originally Posted by sidekick2

According to the wiki, blu-ray supports dolby digital+. I have the dd+ as a 2nd track, and when I select it, the video stutters and no audio is output. Reading the wiki, it says dd+ for blu-ray needs the 640k core ac3 file.

DD+ on Blu-ray is entirely different from DD+ on HD DVD and there is really no way to convert between them. You'll have to make do with the AC3 track, or else expand the HD DVD DD+ out to LPCM.

[Jom]

DrNein - Good shout.

Extracting NeEacDec.dll from the retail CD package was a pain, but it worked.

C:\Program Files (x86)\Common Files\Ahead\DSFilter>regsvr32 NeAudio2.ax

C:\>eac3to -test
eac3to (v3.03) is up to date
Nero Audio Decoder (Nero 7) works fine
ArcSoft DTS Decoder doesn't seem to be installed
... etc

On a side note, the dlls are signed, and the certificates expire on the 28th June 2009.
If the nero filters break in June, it should be easy to replace the signatures using signtool in the .NET SDK.

[Snowknight26]

Would it be possible to add a new command line option, say -plain, that outputs chapters with only the timestamps, one on each line?

[Jeff Flowerday]

Strat Pack Concert on Blu-Ray has the multi-channel AC3 and DTS MA being reported as 5.0 by eac3to. Apparently missing LFE channel.

Madshi: Thought it was a little weird so I've attached a 40 MB clip, just so you could confirm.


http://www.flowerdayconsulting.com/Misc/strat.zip

I got around it by forcing 6 channels. madflac didn't like the result with only 5 channels.

[piratburner]

Originally Posted by Jom

DrNein - Good shout.


C:\Program Files (x86)\Common Files\Ahead\DSFilter>regsvr32 NeAudio2.ax

C:\>eac3to -test
eac3to (v3.03) is up to date
Nero Audio Decoder (Nero 7) works fine
ArcSoft DTS Decoder doesn't seem to be installed
... etc

Thanks all for the tips ! :-) eac3to is up and running on windows 7

[Chumbo]

@madshi,
Just fyi, I just grabbed the latest Matroska splitter and, after the install, decided to run -test and go this:

Haali Matroska Muxer (2009-01-10) is installed
  There's a new version (2009-01-11) available  <WARNING>
  http://haali.net/mkv  <WARNING>

It may not even be an issue with eac3to, but wanted to let you know. Also, note the "<WARNING>" in the log. This is fine on-screen and only shows up in the log.

[LessThanJake]

madshi, does eac3to remove some for decoding unnecessary nal unit types, such as access unit delimiter, or multiple sps/pps nalus, when extracting an avc-stream from an BD-m2ts to a raw-stream, or is it a 1:1 extraction?

e.g.
eac3to src.m2ts 1: dest.h264

greets
LTJ

[Jeff Flowerday]

Originally Posted by Chumbo

@madshi,
Just fyi, I just grabbed the latest Matroska splitter and, after the install, decided to run -test and go this:

Haali Matroska Muxer (2009-01-10) is installed
  There's a new version (2009-01-11) available  <WARNING>
  http://haali.net/mkv  <WARNING>

It may not even be an issue with eac3to, but wanted to let you know. Also, note the "<WARNING>" in the log. This is fine on-screen and only shows up in the log.

Haali website and file dates are different. Nothing Madshi can do about this. It's been discussed a few pages back.

[n0mag!c]

Invalid edit format "edit=0:00:00,960ms".
Invalid edit format "edit=0:00:00.000,960ms".
Invalid edit format "edit=0:00:00.001,960ms".
Invalid edit format "edit=0:00:00.960,960ms".

command line: eac3to.exe g3.ac3 g4.ac3 -edit=0:00:00.961,960ms
----------------------------------------------------------------
AC3, 5.1 channels, 2:02:34, 448kbps, 48khz
Editing (E-)AC3...
Creating file "g4.ac3"...
eac3to processing took 11 seconds.
Done.

[Thunderbolt8]

perhaps you have to use '+' in that case, ',+960'

[Chumbo]

Originally Posted by Jeff Flowerday

Haali website and file dates are different. Nothing Madshi can do about this. It's been discussed a few pages back.

Gotcha, thanks for the info. Missed that one.

[sidekick2]

Originally Posted by peterjcat

DD+ on Blu-ray is entirely different from DD+ on HD DVD and there is really no way to convert between them. You'll have to make do with the AC3 track, or else expand the HD DVD DD+ out to LPCM.

Hmm... shouldn't this be a feature for the awesome eac3to?

Madshi, how about a conversion option from hd-dvd dd+ to bd version? I would assume it'd be similar to the thd+ac3, as it looks like dd+ for blu-ray needs the core as well. I would assume this would be loss less to convert from dd+ to dd+ blu-ray fmt.

Thanks!

[dat720]

madshi again awesome work, i cant praise eac3to enough, coupla quick questions:

Are you able to implement a version check? that at runtime checks your download site for a newer version and notifies with a beep or blue text or something...

And do you ever intend on implementing audio packing to mkv container with the video stream?

[peterjcat]

Originally Posted by sidekick2

Hmm... shouldn't this be a feature for the awesome eac3to?

Madshi, how about a conversion option from hd-dvd dd+ to bd version? I would assume it'd be similar to the thd+ac3, as it looks like dd+ for blu-ray needs the core as well. I would assume this would be loss less to convert from dd+ to dd+ blu-ray fmt.

Thanks!

eac3to is indeed awesome but it relies on external encoders and as far as I know there isn't a freely available encoder for Blu-ray DD+, like there is for AC3 (unless you have a handy $2900 for the official Dolby encoder). If we could encode DD+/TrueHD/DTS-HD then the world would be a much better place!

[TM2-Megatron]

The external software is what's getting me, as well. Is there any alternative to the Nero HD-DVD/Blu-Ray plug-in... that's not even for sale, anymore.

I've been contemplating the Surcode DTS plug-in for a while, now; but the actual advantage over AC-3 (which I can already encode through Vegas) isn't worth the money, for what I do.

I agree, though, TrueHD and DTS-HD encoders would be great.

[tebasuna51]

For what TrueHD and DTS-HD if we have FLAC?

[dat720]

What about the AV recievers that can natively decode the DTS and AC3 varients?

Not everyone watches movies on a PC....

[madshi]

Originally Posted by ExSport

And do you plan it for the future versions if it is easy to implement it? I mean piping.
Your application is one of few which can recode DTS properly

I don't plan that right now. Due to how eac3to is designed, it's difficult to implement input piping. Output is easy, but input is not.

Originally Posted by Devrethman

For some reason when rip blu-rays with eac3to mplayer thinks the end of file is about 4 seconds in. If I run the files through mkvmerge, they work, but if I try to go straight from eac3to to mplayer/mencoder, it always stops around 4 seconds in. Is this normal?

Please try playing the file with Haali's MKV splitter. Does that play fine? If both mkvtoolnix and Haali's MKV splitter can read the files fine then it's probably a bug in mplayer/mencoder.

Originally Posted by sidekick2

I'm not sure if this is a format issue, eac3to, or tsmuxer... I've extracted the eac3 audio from Bourne Identity HD-DVD, and also created a normal ac3 conversion from the eac3.

No errors in any applications, but my dd+ track doesn't play on the disk in a stand alone or ps3.

In tsmuxer, I have the vc1 file, .ac3 file, .eac3 file, and subtitles. It sees it no problem as an dolby digital+ file, and everything works as it should, creating the blu-ray disk structure.

Is dd+ broken in tsmuxer as well, or is eac3to not outputting a flavor of eac3 that tsmuxer likes, or???

I'm using eac3to 3.03, and tsmuxer 1.8.8b.

According to the wiki, blu-ray supports dolby digital+.

Blu-Ray does support DD+, but it's a whole different format compared to HD DVD. So HD DVD DD+ and Blu-Ray DD+ are not compatible and can not be converted into another.

Originally Posted by asarian

I'm not even entirely sure eac3to is causing it (though I'm fair certain it does); but when extracting the VC-1 stream of the American Gangster Blu-Ray, which is annoyingly split over dozens of m2ts files, I get the movie just fine, but the 'total time' index is off. Which is to say, it is set to the length of the first part (~13 mins). Even though the movie plays/streams in full, I can't seek (PS3 = 'Go To') beyond the 13 minutes marker.

There's not enough information for me to help. What exactly did you ask eac3to to do and what other tools did you use to create a PS3 compatible file?

Originally Posted by Snowknight26

Would it be possible to add a new command line option, say -plain, that outputs chapters with only the timestamps, one on each line?

What would that be good for?

Originally Posted by Jeff Flowerday

Strat Pack Concert on Blu-Ray has the multi-channel AC3 and DTS MA being reported as 5.0 by eac3to. Apparently missing LFE channel.

Madshi: Thought it was a little weird so I've attached a 40 MB clip, just so you could confirm.

http://www.flowerdayconsulting.com/Misc/strat.zip

I got around it by forcing 6 channels. madflac didn't like the result with only 5 channels.

What did madFlac do? It should play the file just fine - just as 5.0 and the audio renderer may not like this format. You can try chaining the ffdshow raw audio processor into your playback chain and letting is add an empty LFE channel or something like that...

Originally Posted by LessThanJake

madshi, does eac3to remove some for decoding unnecessary nal unit types, such as access unit delimiter, or multiple sps/pps nalus, when extracting an avc-stream from an BD-m2ts to a raw-stream, or is it a 1:1 extraction?

AUDs are not removed. The only thing removed by eac3to is padding. And I think I recently added some code to remove double sequence headers in the beginning of a h264 stream, but only there and it only occurred with a strange broadcast. I'd consider eac3to's h264 demuxing 1:1, except for removed padding.

Originally Posted by n0mag!c

Invalid edit format "edit=0:00:00,960ms".
Invalid edit format "edit=0:00:00.000,960ms".
Invalid edit format "edit=0:00:00.001,960ms".
Invalid edit format "edit=0:00:00.960,960ms".

Doing "edit=0:00:00.000,960ms" is a weird way of doing "+960ms", just use the latter instead. The other two edits don't work because eac3to wants to loop audio to do a (positive) edit. For that to work the runtime after which you do the edit must be bigger than the edit duration, or else the looping buffers can not be properly filled.

Originally Posted by dat720

Are you able to implement a version check?

It's already there, just do "eac3to -test".

Originally Posted by dat720

And do you ever intend on implementing audio packing to mkv container with the video stream?

Maybe. Maybe not.

[madshi]

eac3to v3.04 released

http://madshi.net/eac3to.zip

* video track framerates are now shown with up to 3 decimals, if necessary
* m2ts/TS framerate is determined by interpreting video track timestamps
* m2ts/TS framerate is displayed in the format description (if available)
* warning is shown if container timestamps don't match video framerate
* warning is shown if video bitstream has a non-standard framerate
* video without framerate information: container framerate is used
* video without framerate information: framerate can be set (e.g. "-23.976")
* video without framerate information: new framerate is written to bitstream
* remaining non-fixed audio delay is now shown in log
* command prompt colors are restored after eac3to has run through
* fixed: 2-pass processing for stripping zero bytes sometimes crashed
* fixed: CA (Conditional Access) tracks were shown as "Unknown audio track"

[starkhouse33]

would it be possible to add a switch into eac3to that can reset/change the video full range flag? some .ts files display the wrong black levels when played on nmt.
Thanks

[n0mag!c]

Originally Posted by Thunderbolt8

perhaps you have to use '+' in that case, ',+960'

Presence of "+" makes no difference.

Originally Posted by madshi

Doing "edit=0:00:00.000,960ms" is a weird way of doing "+960ms", just use the latter instead. The other two edits don't work because eac3to wants to loop audio to do a (positive) edit. For that to work the runtime after which you do the edit must be bigger than the edit duration, or else the looping buffers can not be properly filled.

Presence of "-silence" makes no difference. Is this a bug?

[Snowknight26]

Originally Posted by madshi

What would that be good for?

Can't recall fully (I'm just the messenger), but I think it had something to do with tsmuxer only accepting chapters in that format.

I know, 'stop being and edit the chapters yourself' is probably what you'd (or I'd) say, but I promised I'd ask.

[madshi]

Originally Posted by starkhouse33

would it be possible to add a switch into eac3to that can reset/change the video full range flag? some .ts files display the wrong black levels when played on nmt.

Can you give me more information about this? Which flag do you mean exactly and is it set wrong in those TS files?

Originally Posted by n0mag!c

Presence of "-silence" makes no difference. Is this a bug?

The behaviour is as intended by me, so I don't consider it a bug. Do you honestly have a real life situation where you need to add 960ms inside of the first second runtime of an audio track?

Originally Posted by Snowknight26

Can't recall fully (I'm just the messenger)

I refuse to add any features if I don't even know exactly what they're good for.

[n0mag!c]

Originally Posted by madshi

The behaviour is as intended by me, so I don't consider it a bug. Do you honestly have a real life situation where you need to add 960ms inside of the first second runtime of an audio track?

This IS a real life situation. I need to apply delay to the beginning of audio track. (I don't like delaycut interface )

[dat720]

Originally Posted by madshi

It's already there, just do "eac3to -test".

Nice one, i was using 2.87 and hadn't noticed that.... is it posible to make it do it during normal runtime? (i know it's possible but will you do it?) like show the file version in the normal output, kinda like mplayer does? that way you don't have to run -test to check the version....

[starkhouse33]

Originally Posted by madshi

Can you give me more information about this? Which flag do you mean exactly and is it set wrong in those TS files?

h.264 streams broadcast in Europe have the video flag set wrong in their encoder. using eac3to to slow down the video I would then have to run the file through ts4np and check change video full range flag in the options to have the video play correctly on my popcorn hour.

here is a quote that explains in detail what is wrong from a popcorn hour user.

Sky and probably other European companies have a bug in their H264 encoder. The sequence header indicates that they use full color range when in reality they are encoding using limited range. Their own STB ignores the flag so their normal customers are not affected.This is not a Sigma or NMT bug and they are doing the correct thing.
I suggest patching the original source files to correct the flag.

Broadcast digital video uses 16-235 levels, whilst PC video uses 0-255 levels. Therefore video black is at 16, and white at 235, whilst PC black is at 0 and white is at 255. Therefore video at 16 is black, and white at 235, whilst the PC is at 0 black and white is at 255th

If you get PC and Broadcast levels confused - or feed broadcast levels to a PC display or vice versa you can end up with crushed blacks, blown out whites, or nasty washed out pictures. If you get PC and broadcast levels confused - or broadcast feed levels to a PC display or vice versa you can end up with crushed blacks, whites blown out, or washed out nasty pictures.

This can be caused by the wrong format being flagged in a header - or a header being mis-interpreted.

All satellite receivers will assume 16-235 - but the video content could be flagged as 0-255 and ignored. All satellite receivers will assume 16-235 - but the video content could be flagged as 0-255 and ignored.

[Thunderbolt8]

Originally Posted by madshi

eac3to v3.04 released

[tebasuna51]

Originally Posted by dat720

What about the AV recievers that can natively decode the DTS and AC3 varients?

You always can play original BD or at least copy the original audio tracks.

Not everyone watches movies on a PC....

FLAC is supported by new HD players like PopCorn Hour and variants.

[jfcarbel]

A few questions:

- can extract AC3 stream from Blu-Ray TrueHD/AC3 tracks
- can extract DTS core from DTS-HD tracks

Do the above features require Nero or ArcSoft software? And is trial version of ArcSoft adequate to test it on my machine before I buy?

- can extract TrueHD stream from Blu-Ray TrueHD/AC3 tracks

Is this just mean extract the original HD audio stream with no changes? And does this require any 3rd party software?

[Jom]

Originally Posted by tebasuna51

You always can play original BD or at least copy the original audio tracks.

FLAC is supported by new HD players like PopCorn Hour and variants.

Does the PCH support multichannel FLAC now?
last i heard it only passed-through stereo

[madshi]

Originally Posted by n0mag!c

This IS a real life situation. I need to apply delay to the beginning of audio track. (I don't like delaycut interface )

Well, there is a separate way to do simple audio delays. Simply do e.g. "eac3to source.ac3 dest.ac3 +960ms". The "edit" option is not meant to be used for simple delays.

Originally Posted by dat720

Nice one, i was using 2.87 and hadn't noticed that.... is it posible to make it do it during normal runtime? (i know it's possible but will you do it?) like show the file version in the normal output, kinda like mplayer does? that way you don't have to run -test to check the version....

I don't know. Fetching the latest version number from the internet does cost a bit of time. Can be up to multiple seconds, if the doom9 server is slow. I don't like the idea of having every eac3to run be slowed down by trying to get online...

Originally Posted by starkhouse33

h.264 streams broadcast in Europe have the video flag set wrong in their encoder. using eac3to to slow down the video I would then have to run the file through ts4np and check change video full range flag in the options to have the video play correctly on my popcorn hour.

Ok, I see. I'll add "full range" patching to the next eac3to version. Also I'll post a warning whenever eac3to sees a h264 stream with "full range" set.

Originally Posted by jfcarbel

Do the above features require Nero or ArcSoft software?

Is this just mean extract the original HD audio stream with no changes? And does this require any 3rd party software?

All of this just extracts without changes (apart from eventually removing dialnorm information) and doesn't require any 3rd party software.

[tebasuna51]

Originally Posted by Jom

Does the PCH support multichannel FLAC now?
last i heard it only passed-through stereo

I don't have PCH to answer you but is a firmware question.
Of course can't be passed by SPDIF but can be send by HDMI or multichannel analogic output (Kaiboer)

[bold1342]

madshi

Excellent tool thank you.

I think I may have encountered a bug. I always speedup the blu ray audio 24 to 25fps.

The processed audio file is always higher bit depth than the original track. For example Harry Potter and the Prizoner of Azkaban blu ray has a 16 bit RAW audio track, when processed with eac3to the finished file reports that it is 32 bit?

Please see log file below

eac3to v3.04
command line: "C:\Users\dmb_laptop\Desktop\eac3to 3.04\eac3to.exe" "F:\POTTER 3\00103.m2ts" 3: "F:\POTTER 3\harry .dts" -speedup
------------------------------------------------------------------------------
M2TS, 1 video track, 14 audio tracks, 29 subtitle tracks, 2:21:42, 24p /1.001
1: VC-1, 1080p24 /1.001 (16:9)
2: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
3: RAW/PCM, 5.1 channels, 16 bits, 48khz
4: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
5: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
6: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
7: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
8: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
9: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
10: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
11: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB
12: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
13: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
14: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
15: AC3, 5.1 channels, 448kbps, 48khz, dialnorm: -27dB
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)
32: Subtitle (PGS)
33: Subtitle (PGS)
34: Subtitle (PGS)
35: Subtitle (PGS)
36: Subtitle (PGS)
37: Subtitle (PGS)
38: Subtitle (PGS)
39: Subtitle (PGS)
40: Subtitle (PGS)
41: Subtitle (PGS)
42: Subtitle (PGS)
43: Subtitle (PGS)
44: Subtitle (PGS)
[a03] Extracting audio track number 3...
[a03] Reading RAW/PCM...
[a03] Swapping endian...
[a03] Remapping channels...
[a03] Changing FPS from 23.976 to 25.000...
[a03] Reducing depth from 64 to 32 bits...
[a03] Writing WAVs...
[a03] Creating file "F:\POTTER 3\harry .SL.wav"...
[a03] Creating file "F:\POTTER 3\harry .R.wav"...
[a03] Creating file "F:\POTTER 3\harry .LFE.wav"...
[a03] Creating file "F:\POTTER 3\harry .SR.wav"...
[a03] Creating file "F:\POTTER 3\harry .L.wav"...
[a03] Creating file "F:\POTTER 3\harry .C.wav"...
[a03] The original audio track has a constant bit depth of 16 bits.
[a03] The processed audio track has a constant bit depth of 32 bits.
Encoding DTS <1536kbps> with Surcode...
Found Surcode DTS Encoder version 1.0.23.0.
Surcode encoding successfully started. Please wait...
Closing Surcode...


Any help would be appreciated.


Thanks


David

[madshi]

Originally Posted by bold1342

I think I may have encountered a bug. I always speedup the blu ray audio 24 to 25fps.

The processed audio file is always higher bit depth than the original track. For example Harry Potter and the Prizoner of Azkaban blu ray has a 16 bit RAW audio track, when processed with eac3to the finished file reports that it is 32 bit?

That's just fine. Speeding up audio properly requires a full floating point resampling process, which even ends up with 64bit data. eac3to then reduces the 64bit floating point data to whatever seems most appropriate. If you want to end up with only 16bit data, you can use "-down16". However, that doesn't make any sense if you want to use Surcode to encode to DTS, because when doing DTS encoding the higher the bitdepth you feed into the encoder, the better. That results in higher audio quality without any size increase...

[shambles]

i have quite an interesting lossless track where the bitdepth analyser doesn't tell the whole truth unless i convert it to separate waves and process those one by one.. i'm not sure though if anyone except me actually cares about that sort of thing :P

C:\Program Files\eac3to>eac3to E:\aviator.flac f:\aviator.wavs
FLAC, 5.1 channels, 2:50:07, 24 bits, 1691kbps, 48khz
Decoding FLAC...
Writing WAVs...
Creating file "f:\aviator.L.wav"...
Creating file "f:\aviator.LFE.wav"...
Creating file "f:\aviator.SR.wav"...
Creating file "f:\aviator.R.wav"...
Creating file "f:\aviator.C.wav"...
Creating file "f:\aviator.SL.wav"...
Original audio track: max 24 bits, average 18 bits, most common 16 bits.
eac3to processing took 4 minutes, 28 seconds.
Done.

C:\Program Files\eac3to>eac3to f:\aviator.L.wav aviator.L.wav
WAV, 1.0 channels, 2:50:07, 24 bits, 1152kbps, 48khz
Reading WAV...
Writing WAV...
Creating file "aviator.L.wav"...
Original audio track: max 24 bits, average 17 bits, most common 16 bits.
eac3to processing took 43 seconds.
Done.

C:\Program Files\eac3to>eac3to f:\aviator.LFE.wav aviator.LFE.wav
Original audio track: max 24 bits, average 16 bits, most common 16 bits.

C:\Program Files\eac3to>eac3to f:\aviator.SR.wav aviator.SR.wav
Original audio track: max 24 bits, average 18 bits, most common 16 bits.

C:\Program Files\eac3to>eac3to f:\aviator.R.wav aviator.R.wav
Original audio track: max 24 bits, average 17 bits, most common 16 bits.

C:\Program Files\eac3to>eac3to f:\aviator.C.wav aviator.C.wav
The original audio track has a constant bit depth of 24 bits.

C:\Program Files\eac3to>eac3to f:\aviator.SL.wav aviator.SL.wav
Original audio track: max 24 bits, average 18 bits, most common 16 bits.

it's from the french aviator bluray. what makes it even more interesting is how much bitrate it takes on the disc.

DTS-HD Master Audio English 5.1 / 48 kHz / 3545 kbps / 24-bit (DTS Core: 5.1 / 48 kHz / 1509 kbps / 24-bit)

i was really surprised how much lower the bitrate was when re-encoded to flac.. unless this is some sort of freak unique case, it would seem that dts-hd ma is really bad at compressing variable-bitdepth material.

[Jeff Flowerday]

Originally Posted by Jom

Does the PCH support multichannel FLAC now?
last i heard it only passed-through stereo

It doesn't support multichannel flac in mkv at all.

[Jeff Flowerday]

Originally Posted by madshi

What did madFlac do? It should play the file just fine - just as 5.0 and the audio renderer may not like this format. You can try chaining the ffdshow raw audio processor into your playback chain and letting is add an empty LFE channel or something like that...

Nothing really. If I drop the mkv in graphedit I get an error that it doesn't know which filter to use on the 5.0 flac. PS) I had the same issue with 6.1 flac I had to redoo the flac as 7.1

I guess I should post this kind of stuff in your madflac thread.

[alc0re]

I'm going to second the request for a -plain switch to chapters extraction. The reason is that tsMuxer likes just the timestamp in each chapter line when muxing. This would make muxing a lot quicker if we didn't have to edit out each non-timestamp line (ex Chapter1=)

[omf]

Hi everyone!

I'm trying to cut the first 28.7 seconds out of an audio file, so I'm doing this:

eac3to input.wav output.aac -edit=0:00:00,28700ms

eac3to complains that this is an "invalid edit format", though. Can anyone tell me what the correct syntax is? I'm using madshi's latest (3.04).

Thanks!

[Thunderbolt8]

as posted before, when you want to cut/insert something at the beginning, then just use the delay function and not the cut function.

[omf]

Originally Posted by Thunderbolt8

as posted before, when you want to cut/insert something at the beginning, then just use the delay function and not the cut function.

I see - I thought a "delay" was different from a "cut" (you normally wouldn't lose any audio with a "delay" - it would simply be moved to a later starting point within the stream).

I'll give it a try, thanks.

[rebkell]

Originally Posted by omf

I see - I thought a "delay" was different from a "cut" (you normally wouldn't lose any audio with a "delay" - it would simply be moved to a later starting point within the stream).

I'll give it a try, thanks.

Actually, a negative delay will cut off the first part of an audio stream and a positive delay will pad silence at the start of an audio file.

[omf]

Originally Posted by rebkell

Actually, a negative delay will cut off the first part of an audio stream and a positive delay will pad silence at the start of an audio file.

Makes sense, and the following command line had the desired result:

eac3to input.wav output.aac -28700ms

Thanks for the assist.

[Nullity]

Originally Posted by madshi

eac3to v3.04 released

http://madshi.net/eac3to.zip

* video track framerates are now shown with up to 3 decimals, if necessary
* m2ts/TS framerate is determined by interpreting video track timestamps
* m2ts/TS framerate is displayed in the format description (if available)
* warning is shown if container timestamps don't match video framerate
* warning is shown if video bitstream has a non-standard framerate
* video without framerate information: container framerate is used
* video without framerate information: framerate can be set (e.g. "-23.976")
* video without framerate information: new framerate is written to bitstream
* remaining non-fixed audio delay is now shown in log
* command prompt colors are restored after eac3to has run through
* fixed: 2-pass processing for stripping zero bytes sometimes crashed
* fixed: CA (Conditional Access) tracks were shown as "Unknown audio track"

Thanks madshi! I really appreciate all your work.

[n0mag!c]

Originally Posted by madshi

eac3to v3.04 released
* video without framerate information: new framerate is written to bitstream

Thanks a lot!

Originally Posted by madshi

Well, there is a separate way to do simple audio delays. Simply do e.g. "eac3to source.ac3 dest.ac3 +960ms". The "edit" option is not meant to be used for simple delays.

Thanks for the tip! Your programme is real multifunctional!

[peterjcat]

Originally Posted by Jeff Flowerday

It doesn't support multichannel flac in mkv at all.

Quite right. Furthermore, you can't put FLAC in a TS/M2TS, so even if the NMTs could play multichannel FLAC they could run into MKV overhead issues.

FLAC is not a solution for anything but a PC at the moment.

[ron spencer]

I just bought arcsoft....

when I try to have eac3to decode dts with arcsoft decoders I get:

"the arcsoft decoder does not seem to work"


I tried the fixes posted here (PATH, etc.). Version I have is 2.1.6.126.

any ideas with this?

[Chumbo]

Originally Posted by madshi

eac3to v3.04 released

http://madshi.net/eac3to.zip

...* command prompt colors are restored after eac3to has run through...

Appreciate you squeezing this one in. Thank you.

[madshi]

Originally Posted by peterjcat

Quite right. Furthermore, you can't put FLAC in a TS/M2TS, so even if the NMTs could play multichannel FLAC they could run into MKV overhead issues.

FLAC is not a solution for anything but a PC at the moment.

As far as I've been told, the new HDI Dune players can play MKV multichannel FLAC, also they're supposed to play external FLAC tracks in combination with TS/m2ts video. Also DivX7 has just been released and officially uses the MKV container now. DivX7 certification will require support of the MKV container. So there's hope for improved MKV and maybe also FLAC support.

Originally Posted by ron spencer

I just bought arcsoft....

when I try to have eac3to decode dts with arcsoft decoders I get:

"the arcsoft decoder does not seem to work"

I tried the fixes posted here (PATH, etc.). Version I have is 2.1.6.126.

There's been two tricks suggested to make it work. One was changing the path. The other one was manually registering some specific DirectShow filters. Search for "regsvr32" in this thread.

[williewonton]

madshi

Would be interested in your assessent of he Dune HD player you have ordered.

Let us all know the new thread you have opened for the device, and how you have found it's performance.

It looks good.

[nautilus7]

madshi, what about adding option to downmix to 6.1ch (-down7)?

[Paddy97]

Is it possible when retreiving the features of a bluray disc to get more then 8? I am trying to get a short film from Pixars Wall-E and since there are several playlists of the main feature film due to different languages the short film that is 7 minutes never makes it to the list when I run eac3to E: ( thats the bluray drive name ). It lists only the first 8 and the shortest feature is 24 minutes that it lists.

Any trick or switch I can use?

[nautilus7]

Load the .mpls file directly in eac3to.

[asarian]

Originally Posted by asarian
I'm not even entirely sure eac3to is causing it (though I'm fair certain it does); but when extracting the VC-1 stream of the American Gangster Blu-Ray, which is annoyingly split over dozens of m2ts files, I get the movie just fine, but the 'total time' index is off. Which is to say, it is set to the length of the first part (~13 mins). Even though the movie plays/streams in full, I can't seek (PS3 = 'Go To') beyond the 13 minutes marker.

Originally Posted by madshi

There's not enough information for me to help. What exactly did you ask eac3to to do and what other tools did you use to create a PS3 compatible file?

Simple. I asked eac3to to combine and extract a the main VC-1 stream, like:

eac3to 1) 2: c:\video\amg.vc1

I then remux with tsMuxeR, as I always do. Seems like each of the multipart m2ts files has its own duration index (or so it seems). As a result, the length of the entire movie is set to the length of the first part (~13 mins).

[asarian]

Originally Posted by Paddy97

Is it possible when retreiving the features of a bluray disc to get more then 8? I am trying to get a short film from Pixars Wall-E and since there are several playlists of the main feature film due to different languages the short film that is 7 minutes never makes it to the list when I run eac3to E: ( thats the bluray drive name ). It lists only the first 8 and the shortest feature is 24 minutes that it lists.

Any trick or switch I can use?

I usually just look at the length of the m2ts. The "Burn-E" short-movie (if that's the one you meant) resides all in 1 m2ts, and is ~3G in length.

[alc0re]

That's not what he meant. Burn-E is one of 2 shorts on the Wall-E bluray...there's a seperate Pixar Shorts bluray that has all of pixars short animated features on it...and it has more than 8 playlists on it.

[asc28]

Encoding issues w/ Sin City Nordic Blu-Ray DTS-MA?

eac3to v3.04
command line: eac3to 00009.m2ts 3:C:\sin.mp4
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 4 subtitle tracks, 2:03:59, 24p /1.001
1: h264/AVC, 1080p24 /1.001 (16:9)
2: AC3, 5.1 channels, 640kbps, 48khz
3: DTS Master Audio, 7.1 (strange setup) channels, 24 bits, 48khz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48khz)
4: Subtitle (PGS)
5: Subtitle (PGS)
6: Subtitle (PGS)
7: Subtitle (PGS)
CAUTION: Decoding this track with ArcSoft results in low volume. <WARNING>
[a03] Extracting audio track number 3...
[a03] Decoding with ArcSoft DTS Decoder...
[a03] Encoding AAC <0.50> with NeroAacEnc...
[a03] The original audio track has a constant bit depth of 24 bits.
Video track 1 contains 178368 frames.
eac3to processing took 1 hour, 32 minutes.
Done.

Is Arcsoft not doing the job right...should I use something different?

[nautilus7]

It's a known issue with the decoder.

Only thing thing you can do is send a sample to ArcSoft and ask them to fix their decoder.

btw, you should get the Japanese blu-ray of Sin City. It's way better.

[jfcarbel]

REMOVED, for some reason my post got done twice.

[jfcarbel]

Originally Posted by madshi

All of this just extracts without changes (apart from eventually removing dialnorm information) and doesn't require any 3rd party software.

The above was in response to a question about what 3rd party software is needed for:

- can extract AC3 stream from Blu-Ray TrueHD/AC3 tracks
- can extract DTS core from DTS-HD tracks

My followup question then, when would the ArcSoft DTS decoder be needed? Is this just when we need to re-encode the DTS to maybe a lower bitrate? And if so then I assume that task would then also require the SurCode DVD–DTS Encoder.

So if need to re-encode DTS for lower bitrate, then requires both: ArcSoft and SurCode - correct?

What is used for AC3 re-encoding? I am familiar with AC3 5.1 but what is (E-)AC3? From my reading it seems Nero is best for AC3 decode and second best is libav/ffmpeg - correct?

Can TrueHD be converted to AC3 5.1 and does this require any 3rd party software?

Sorry, just want to confirm I understand whats needed with this tool. I think I am almost there

UPDATE: Should I thought of WikiPedia on that (E-)AC3. I found the below:

(E-)AC3 = Dolby Digital Plus:
With the built in codec of Nero 7 Showtime 3.10.1.0 with the Blu-Ray/HD-DVD Video plugin, eac3to can successfully decode E-AC3 audio using the Showtime decoder and encode it as anything, FLAC, WAV, AC3, DTS, whatever.

[asc28]

Originally Posted by nautilus7

It's a known issue with the decoder.

Only thing thing you can do is send a sample to ArcSoft and ask them to fix their decoder.

btw, you should get the Japanese blu-ray of Sin City. It's way better.

So ArcSoft has always decoded ALL DTS-MA tracks w/ incorrect volume? Does Sonic (or anyone else) do it correctly?

The JAP Blu-Ray only has 5.1 sound, so the Nordic's audio is superior. I'll be combining it w/ video from JAP.

[jfcarbel]

Originally Posted by DrNein

I do not have experience with Windows 7 and Nero 7 however, confirm the following files exist and that the *.ax is registered (i.e. re-register it).

C:\Program Files\Common Files\Ahead\DSFilter\NeAudio2.ax
C:\Program Files\Common Files\Ahead\DSFilter\NeEacDec.dll

Also, the plugin license is not strictly required as other licenses encompass the audio functionality, such as Ultra Edition. As said previous, I think the best option is to install Nero 7 Micro with appropriate license and then add NeEacDec.dll as above. This achieves a minimal install of what remains a great disc writing program without the clutter.

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

[nautilus7]

Originally Posted by asc28

So ArcSoft has always decoded ALL DTS-MA tracks w/ incorrect volume? Does Sonic (or anyone else) do it correctly?

The JAP Blu-Ray only has 5.1 sound, so the Nordic's audio is superior. I'll be combining it w/ video from JAP.

Did you see any similar message/warning with any other track?

Don't worry, eac3to we'll warn you when there's a problem. This track is one of the very few cases where ArcSoft decoder doesn't work ok.

Wasn't aware of the audio difference between the 2 Blu-rays of Sin City. But since ArcSoft can't decode it correctly, i don't know if it worths it.

[Jom]

Regarding unregistering all the Arcsoft filters.

after installing TMT, I just opened up a command prompt, cd-ed to the directory that contained the filters and ran :

for /F %i in ('dir /B') do@regsvr32 /u /s %i

you then just regsvr32 the ax/dll's you need for DTS decoding.

as i recall, the names of the dlls are pretty obvious.

[madshi]

Originally Posted by nautilus7

madshi, what about adding option to downmix to 6.1ch (-down7)?

Which downmix method should I use for that? Should I just combine the both back channels to one? Or should the back channels also partially be mixed into the side channels? That's the main problem with downmixing: Figuring out which mixing matrix should be used exactly. I don't want to add something new, if I'm not sure that I'm doing it the right way...

Originally Posted by Paddy97

Is it possible when retreiving the features of a bluray disc to get more then 8?

eac3to is not limited to 8 titles. It lists 500 titles, if there are so many on the Blu-Ray disk. However, only titles with a runtime of 15 minutes or longer are listed. The reason for that is that there are quite a lot of Blu-Ray discs which have literally hundreds of titles. If I'd list all of them, the command prompt memory would run over so even if you scrolled up, you wouldn't see the movie's main title, anymore. So I *have* to set a runtime limit somewhere. Or else we'll be flooded with unneeded titles. Of course it's not nice if short films are missing in the title list. But with Blu-Ray there's no way for eac3to to know which titles are important and which aren't. So eac3to is strictly looking for a specific minimum runtime of 15 minutes.

Originally Posted by asarian

Simple. I asked eac3to to combine and extract a the main VC-1 stream, like:

eac3to 1) 2: c:\video\amg.vc1

I then remux with tsMuxeR, as I always do. Seems like each of the multipart m2ts files has its own duration index (or so it seems). As a result, the length of the entire movie is set to the length of the first part (~13 mins).

So the problem is most probably caused by tsMuxeR and not by eac3to. You can try playing the demuxed "amg.vc1" file, does it also stop at 13 min? Probably not. Or ask eac3to to mux the vc1 file to MKV. Does that MKV also stop playing at 13 min? Probably not. So it's probably a bug in tsMuxeR.

Originally Posted by jfcarbel

My followup question then, when would the ArcSoft DTS decoder be needed? Is this just when we need to re-encode the DTS to maybe a lower bitrate? And if so then I assume that task would then also require the SurCode DVD–DTS Encoder.

You need to separate decoding and encoding. You need a decoder for decoding and an encoder for encoding. The ArcSoft DTS decoder is needed for best quality DTS-HD decoding. It doesn't matter if you store the decoded data as WAV or if you reencode it to any other format.

Which encoders and decoders are recommended and which need to be supplied externally are documented on the first page of this thread.

[asc28]

Originally Posted by nautilus7

Did you see any similar message/warning with any other track?

Don't worry, eac3to we'll warn you when there's a problem. This track is one of the very few cases where ArcSoft decoder doesn't work ok.

Wasn't aware of the audio difference between the 2 Blu-rays of Sin City. But since ArcSoft can't decode it correctly, i don't know if it worths it.

Oh, I thought that message could've been included in the new eac3to versions. Is there any alternative to decoding this track (Sonic doesn't do 7.1, right)? Would love to have Sin City in beautiful 7.1.

Also, was this problem discussed in this thread/elsewhere? What specifically is wrong w/ the decoded track? My interpretation is that the "strange channel setup" messes w/ the volume? Thanks for the help

[madshi]

Originally Posted by asc28

Also, was this problem discussed in this thread/elsewhere?

Well, there's an easy way to find out, you know?

[asc28]

Originally Posted by madshi

Well, there's an easy way to find out, you know?

Unfortunately, I don't think it's possible to search a specific thread, only sections of the forum. I've searched for "strange setup" and other terms, but it just returns the eac3to thread, not specific posts/pages in the 400-page thread.

Please let me know if there's any info on this problem, or a way to search for it. Thanks!

[madshi]

Originally Posted by asc28

Unfortunately, I don't think it's possible to search a specific thread

Come on, it's not so hard. Why don't you simply click on "search this thread"?

[lexor]

Originally Posted by asc28

Unfortunately, I don't think it's possible to search a specific thread, only sections of the forum. I've searched for "strange setup" and other terms, but it just returns the eac3to thread, not specific posts/pages in the 400-page thread.

Please let me know if there's any info on this problem, or a way to search for it. Thanks!

Actually there is a way, scroll to the top of the page, there is a link saying "Search this Thread". Click it, a search box appears, type in search terms.

/edit: oops, madshi is too fast, but I gave more detailed instructions... probably 'couse I don't have to give them every page.

[asc28]

Originally Posted by madshi

Come on, it's not so hard. Why don't you simply click on "search this thread"?

Sorry, really retarded of me to miss that. Doh

I found this in an earlier post:

Originally Posted by madshi

If you want, you can report this problem to the ArcSoft guys (together with a small sample). Should be easy for them to fix. Just ask them to decode the sample as 5.1 and then as 7.1. The 7.1 decoding volume will be lower, which doesn't really make any sense.

By 5.1, I'm guessing you mean the DTS core? So, if I wanted to, I could compare the volume difference between the 5.1 and 7.1 and alter the decoded PCM's volume by that amount...

Why wouldn't this process be lossless? Wouldn't adding +Xdb and then subtracting the same -Xdb value to a PCM track return the same data? Sorry if I'm relatively uninformed on this matter.

And hopefully, I can assume, every channel in the decoded ArcSoft track is too soft by the same amount?

[madshi]

Originally Posted by asc28

Wouldn't adding +Xdb and then subtracting the same -Xdb value to a PCM track return the same data?

No. Because volume change is done in floating point, while PCM is integer. Conversion from floating point to integer is not lossless.

Originally Posted by asc28

And hopefully, I can assume, every channel in the decoded ArcSoft track is too soft by the same amount?

Yes. Furthermore with some tracks decoding doesn't work properly at all (distortion). If that is the case, you'll easily notice that on a quick check, though.

[asc28]

Originally Posted by madshi

No. Because volume change is done in floating point, while PCM is integer. Conversion from floating point to integer is not lossless.

Thanks for your help! I think I'll convert to floating-point PCM to apply the volume change, and then feed the floating-point directly to Nero AAC encoder. Hopefully, if I understand these concepts correctly, this should not result in much quality loss, had I just encoded an originally correct-volume PCM to AAC.

I'll also try to contact ArcSoft about the incorrect decoding.

[madshi]

Originally Posted by asc28

Thanks for your help! I think I'll convert to floating-point PCM to apply the volume change, and then feed the floating-point directly to Nero AAC encoder. Hopefully, if I understand these concepts correctly, this should not result in much quality loss, had I just encoded an originally correct-volume PCM to AAC.

The quality loss (compared to correct-volume PCM to AAC) should be minimal, most probably not audible for any normal person.

Originally Posted by asc28

I'll also try to contact ArcSoft about the incorrect decoding.

If you get any useful reply, please let us know.

[Boulder]

A (yet another) feature request: ability to use the SoundTouch library for audio speedup or slowdown without changing the audio pitch.

[asc28]

Originally Posted by asc28

I'll convert to floating-point PCM to apply the volume change, and then feed the floating-point directly to Nero AAC encoder. Hopefully, if I understand these concepts correctly, this should not result in much quality loss, had I just encoded an originally correct-volume PCM to AAC.

I actually might instead change the volume by applying gain on the AAC post-encode. I believe this would be a lossless process if the gain can be applied the same way mp3gain does it (by modifying the frame headers). Will eac3to do it this way (change volume losslessly by modifying ADTS frame headers on the AAC) if I do: eac3to input.aac output.aac -Xdb?

[edit] Also here are some tests I did w/ the latest WaveGain:

wavegain sin51.C.wav sin71.C.wav sin51.LFE.wav sin71.LFE.wav sin51.L.wav sin71.L.wav sin51.R.wav sin71.R.wav

Analyzing...

Gain | Peak | Scale | New Peak |Left DC|Right DC| Track
| | | |Offset | Offset |
--------------------------------------------------------------
-1.16 dB | 32767 | 0.87 | 28671 | 0 | 0 | sin51.C.wav
+2.19 dB | 21538 | 1.29 | 27715 | 0 | 0 | sin71.C.wav
-0.00 dB | 32767 | 1.00 | 32766 | 0 | 0 | sin51.LFE.wav
+3.64 dB | 21538 | 1.52 | 32765 | 0 | 0 | sin71.LFE.wav
+0.15 dB | 32209 | 1.02 | 32766 | 0 | 0 | sin51.L.wav
+3.64 dB | 21538 | 1.52 | 32765 | 0 | 0 | sin71.L.wav
+0.02 dB | 32681 | 1.00 | 32766 | 0 | 0 | sin51.R.wav
+3.64 dB | 21538 | 1.52 | 32765 | 0 | 0 | sin71.R.wav

WaveGain Processing completed normally

Looks like the volume difference is not exactly the same for every channel (varies from 3.35 to 3.64), though this might be explained by how the extra 2 channels are mapped / DTS compression.

[asarian]

Originally Posted by asc28

Oh, I thought that message could've been included in the new eac3to versions. Is there any alternative to decoding this track (Sonic doesn't do 7.1, right)? Would love to have Sin City in beautiful 7.1.

Also, was this problem discussed in this thread/elsewhere? What specifically is wrong w/ the decoded track? My interpretation is that the "strange channel setup" messes w/ the volume? Thanks for the help

I have had both Sin-City Blu-Rays. The Japanese actually has TrueHD 5.1 audio (instead of just DD 5.1), but the track is simply broken: if you try to extract it, you'll find it to be corrupted, and eac3to aborted at the time, so I could only salvage the AC3 core. The Scandinavian Blu-Ray comes with an odd reddish tint in the video, but has a flawsless DTS-HD MA 7.1 track. So, I finally remuxed the video from the Japanese disc with the audio from the Scandinavian one, threw out the Japanese disc, and now all is fine again. My longwinded point being: I had no trouble whatsoever converting the DTS-HD MA 7.1 track from the Nordic disc to LPCM, and experienced no volume issue or conversion hicup of any kind. This was done with an older version of eac3to at the time (eac3to v2.78), and, of course, the Arcsoft DTS Decoder.

[nautilus7]

What you're saying about flawless decoding of the DTS-HD track is simply not possible. If you're talking about the same disc, then eac3to (+arcsoft) should have produced the same warning (unless that eac3to version didn't warn for this).

What do you mean the TrueHD from the JAP disc is corrupted? Do you have a log or something. (if you don't have don't mind, i have the disc so i can test in a few days.)

[Paddy97]

Originally Posted by madshi

eac3to is not limited to 8 titles. It lists 500 titles, if there are so many on the Blu-Ray disk. However, only titles with a runtime of 15 minutes or longer are listed. The reason for that is that there are quite a lot of Blu-Ray discs which have literally hundreds of titles. If I'd list all of them, the command prompt memory would run over so even if you scrolled up, you wouldn't see the movie's main title, anymore. So I *have* to set a runtime limit somewhere. Or else we'll be flooded with unneeded titles.

Thanks for the update on how the logic works. I can relate to this design choice.

As allways there is a way around everything even though you need some more tools. What I do now is that I use BDInfo to pinpoint the correct playlist ( using the length as a way of picking the right one ) and then I go from there.

[asarian]

Originally Posted by nautilus7

What do you mean the TrueHD from the JAP disc is corrupted? Do you have a log or something. (if you don't have don't mind, i have the disc so i can test in a few days.)

Actually, I still have a log.

eac3to 1) 3: c:\video\sincity.pcm
M2TS, 1 video track, 4 audio tracks, 5 subtitle tracks, 2:04:01
1: Chapters, 29 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), -13ms
4: TrueHD/AC3, Japanese, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB), -13ms
5: AC3, English, 2.0 channels, 256kbps, 48khz, dialnorm: -27dB, -13ms
6: AC3, English, 2.0 channels, 256kbps, 48khz, dialnorm: -27dB, -13ms
7: Subtitle (PGS), Japanese
8: Subtitle (PGS), Japanese
9: Subtitle (PGS), English
10: Subtitle (PGS), Japanese
11: Subtitle (PGS), Japanese
[a03] Extracting audio track number 3...
[a03] This track is not clean.
[a03] Extracting TrueHD stream...
[a03] Decoding with libav/ffmpeg...
[libav] Stream parameters not seen; skipping frame
[a03] The libav decoder output an unexpected bitdepth (1).
Aborted at file position 49152.

The Nordic disc decoded without any warning or noticeable lower volume, near as I can tell.

[madshi]

Originally Posted by Boulder

A (yet another) feature request: ability to use the SoundTouch library for audio speedup or slowdown without changing the audio pitch.

As far as I know, SoundTouch is not the preferred choice for best quality processing. Commercial competing algorithms are supposed to sound noticeably better, right? I don't really like the idea to implement something and say: eac3to can do it, but better don't use it, cause it's quality isn't really good.

Originally Posted by asarian

Actually, I still have a log.

eac3to 1) 3: c:\video\sincity.pcm
M2TS, 1 video track, 4 audio tracks, 5 subtitle tracks, 2:04:01
1: Chapters, 29 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), -13ms
4: TrueHD/AC3, Japanese, 5.1 channels, 48khz, dialnorm: -27dB
(embedded: AC3, 5.1 channels, 640kbps, 48khz, dialnorm: -27dB), -13ms
5: AC3, English, 2.0 channels, 256kbps, 48khz, dialnorm: -27dB, -13ms
6: AC3, English, 2.0 channels, 256kbps, 48khz, dialnorm: -27dB, -13ms
7: Subtitle (PGS), Japanese
8: Subtitle (PGS), Japanese
9: Subtitle (PGS), English
10: Subtitle (PGS), Japanese
11: Subtitle (PGS), Japanese
[a03] Extracting audio track number 3...
[a03] This track is not clean.
[a03] Extracting TrueHD stream...
[a03] Decoding with libav/ffmpeg...
[libav] Stream parameters not seen; skipping frame
[a03] The libav decoder output an unexpected bitdepth (1).
Aborted at file position 49152.

This is probably with a rather old eac3to build. Try a new one.

[Boulder]

Originally Posted by madshi

As far as I know, SoundTouch is not the preferred choice for best quality processing. Commercial competing algorithms are supposed to sound noticeably better, right? I don't really like the idea to implement something and say: eac3to can do it, but better don't use it, cause it's quality isn't really good.

What program would you recommend for such operation? Preferably something that accepts 32bit float input

[ggking7]

I'm using eac3to-3.03 via wine and I'm trying to get info about a title before I rip the disc. 'eac3to /mnt/cdrom' displays the titles, but 'eac3to /mnt/cdrom "1)"' doesn't show info on the title. Instead I get:

"Please specify the source and dest files first and then the options."

If I switch it to 'eac3to "1)" /mnt/cdrom' I get:

"HD DVD / Blu-Ray disc structure not found."

Does anyone know how to get this to work?

[jj666]

The quotation marks around the 1 are not necessary i think.

-jj-

[J0nThn]

Originally Posted by ggking7

I'm using eac3to-3.03 via wine and I'm trying to get info about a title before I rip the disc. 'eac3to /mnt/cdrom' displays the titles, but 'eac3to /mnt/cdrom "1)"' doesn't show info on the title. Instead I get:

"Please specify the source and dest files first and then the options."

If I switch it to 'eac3to "1)" /mnt/cdrom' I get:

"HD DVD / Blu-Ray disc structure not found."

Does anyone know how to get this to work?

Try a path without a "slash in front". So for example if your current directory is /home/me

you have to use eac3to like this :

wine eac3to ../../mnt/cdrom "1)"

[ggking7]

The quotation marks around the 1 are not necessary i think.

I actually do need those quotes in order to use the ')' character in Linux.

Try a path without a "slash in front".

Weird, I get:

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

'eac3to /mnt/cdrom' and all other eac3to commands I've tried work perfectly.

[Esurnir]

Could we have an option to use a .mpls file as a source so that we get the -right- playlist (on blu-rays such as wall-e spreaded on tons of files with multiple files with video both in french english and spanish, the biggest mpls isnt the one containing the english video.

[lexor]

Originally Posted by Esurnir

Could we have an option to use a .mpls file as a source so that we get the -right- playlist (on blu-rays such as wall-e spreaded on tons of files with multiple files with video both in french english and spanish, the biggest mpls isnt the one containing the english video.

Did you try doing that? I'm pretty sure you can already do that.

[Esurnir]

Originally Posted by lexor

Did you try doing that? I'm pretty sure you can already do that.

*facepalm* it indeed does... (maybe it should be put in the instructions as a possibility).

[hubblec4]

hello madshi

i have a question to demuxing. the sups of some blurays are not correct demuxed. (the same result when i demux with TSMuxer).

the sup-file are very large and contains too much pictures (in this case over 2800).
subtitle 1 till 5 shows the same picture but subtitle 1 till 4 have a starttime only. subtitle 5 is ok. and than subtitle 6 till 8 have the same fault and so on...

if you open the file in supread0.52b2 you will see what i mean.

is it possible to fix that?


hubble

[ggking7]

Any other ideas with my issue above? Does it work for anyone else on Linux?

[kypec]

Originally Posted by ggking7

Any other ideas with my issue above? Does it work for anyone else on Linux?

Did you try aforementioned trick with escaping the parenthesis character by backslash?
eac3to /mnt/cdrom 1\)

[ggking7]

Did you try aforementioned trick with escaping the parenthesis character by backslash?

I get:

$ wine eac3to /mnt/cdrom 1\)
Please specify the source and dest files first and then the options.

I'm not having trouble getting 1) recognized. For example, the following works perfectly:

wine eac3to video "1)" 2: video.* 3: audio.*

It's just this one command:

wine eac3to /mnt/cdrom "1)"

Does that work for anyone else on Linux?

eac3to 3.04 crashes when I start processing a demuxed AVC stream.

http://www.sendspace.com/file/wwsa9t

[madshi]

Originally Posted by Boulder

What program would you recommend for such operation? Preferably something that accepts 32bit float input

Don't know, maybe TimeFactory?

Originally Posted by hubblec4

i have a question to demuxing. the sups of some blurays are not correct demuxed. (the same result when i demux with TSMuxer).

the sup-file are very large and contains too much pictures (in this case over 2800).
subtitle 1 till 5 shows the same picture but subtitle 1 till 4 have a starttime only. subtitle 5 is ok. and than subtitle 6 till 8 have the same fault and so on...

if you open the file in supread0.52b2 you will see what i mean.

is it possible to fix that?

I'll have a look at this. Can you try creating a 50MB sample and check whether the problem can be reproduced with that sample? If so, such a 50MB chunk would be helpful for me to figure out what's going on...

Originally Posted by honai

eac3to 3.04 crashes when I start processing a demuxed AVC stream.

http://www.sendspace.com/file/wwsa9t

That crash seems to be caused by a framerate format eac3to doesn't like. I'm confused why that crash happens - it shouldn't. Can I have the first 10MB of that AVC stream, please?

Thanks for looking into this.

http://www.sendspace.com/file/yy1tf6

[Thunderbolt8]

is it a stream demuxed by eac3to or was it demuxed by another program? in case of the latter it might be possible that this other program doesn't treat the stream the right for eac3to to continue working with it.

[madshi]

Originally Posted by honai

eac3to 3.04 crashes when I start processing a demuxed AVC stream.

The very first sequence header doesn't contain framerate information in this AVC stream, while all other sequence headers do. That made eac3to crash. This problem will be fixed in the next eac3to build. BTW, the "video_full_range_flag" is set in this stream, which is probably incorrect. So you should wait for the next eac3to build, anyway, which will then automatically clear that flag (and crop the stream to 1920x1080 and add timing information to the first sequence header).

Excellent, many thanks!

[ggking7]

Does dialog normalization and pulldown exist on DVD like it does on Blu-Ray? If so, can eac3to operate on a DVD or DVD image? If not, is there another program that can operate on a DVD or image and will remove dialog normalization and pulldown?

[lexor]

Originally Posted by ggking7

Does dialog normalization and pulldown exist on DVD like it does on Blu-Ray? If so, can eac3to operate on a DVD or DVD image? If not, is there another program that can operate on a DVD or image and will remove dialog normalization and pulldown?

DialNorm exists on DVD, since it's in the AC3 stream and it doesn't matter what medium that is on. eac3to should remove it, because it also doesn't really care what medium it is from. ac3 is ac3.

I don't know about pulldown.

[madshi]

Originally Posted by ggking7

Does dialog normalization and pulldown exist on DVD like it does on Blu-Ray? If so, can eac3to operate on a DVD or DVD image? If not, is there another program that can operate on a DVD or image and will remove dialog normalization and pulldown?

DVD video encoding is normally not stable. In other words: There may be some passages where pulldown could technically be removed, but most often there are other passages in the stream where it's not possible or where removing pulldown would even result in a worse stream. So eac3to does not even try to remove pulldown from any SD sources. If you see eac3to listing a DVD as "480p with pulldown flags" you can try using "-stripPulldown", but most probably processing will not run through successfully.

DVD file structure is not supported yet. You have to manually tell eac3to which VOBs to handle. And you shouldn't feed seamless branching or multi angle DVDs to eac3to cause those aren't handled correctly yet. Full DVD structure support might come in a later version (or not).

[hubblec4]

Originally Posted by madshi

I'll have a look at this. Can you try creating a 50MB sample and check whether the problem can be reproduced with that sample? If so, such a 50MB chunk would be helpful for me to figure out what's going on...

hi madshi

i have no idea how can i split the m2ts-file. with tsmuxer i can split it but after demuxing you can't load the sups.

is there another way so i can do that?
the movie is "NARC"

hubble

[madshi]

Originally Posted by hubblec4

i have no idea how can i split the m2ts-file. with tsmuxer i can split it but after demuxing you can't load the sups.

is there another way so i can do that?

You can use TsSplitter or a hexeditor.

[hubblec4]

Originally Posted by madshi

You can use TsSplitter or a hexeditor.

ok thanks. it works. after spliting and demuxing you can load the sup-file in supread and you see the mistake.


hubble

[J0nThn]

Originally Posted by ggking7

I get:

It's just this one command:

wine eac3to /mnt/cdrom "1)"

Does that work for anyone else on Linux?

Did you try this :

cd /mnt && wine eac3to cdrom "1)"

[ggking7]

cd /mnt && wine eac3to cdrom "1)"

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.

[Chumbo]

I wanted to report an issue I found. Looks like the number of frames is not reported when eac3to does a 2nd pass. I did not test this with 3.04 as I didn't see the problem until I no longer have the source. So if someone can verify if this problem still exists in 3.04 please. I'll see if I have anything else that may need a 2nd pass.

Here's a one pass:

eac3to v3.03
command line: eac3to BDSource 3) 1: "H:\Media\chapters1-3.txt" 2: "H:\Media\video1-3.mkv" 3: "H:\Media\audio1-3.ac3" -log="H:\Media\process1-3.txt"
------------------------------------------------------------------------------
M2TS, 1 video track, 1 audio track, 2 subtitle tracks, 0:29:33
1: Chapters, 2 chapters
2: MPEG2, 480i60 /1.001 (4:3)
3: AC3, English, 2.0 channels, 192kbps, 48khz
4: Subtitle (PGS), Spanish
5: Subtitle (PGS), Portuguese
Creating file "H:\Media\chapters1-3.txt"...
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[v02] Muxing video to Matroska...
[a03] Creating file "H:\Media\audio1-3.ac3"...
Added fps value to MKV header.
Video track 2 contains 53136 frames.
eac3to processing took 1 minute, 3 seconds.
Done.

This is the two-pass that does not contain the frame count

eac3to v3.03
command line: eac3to BDSource 2) 1: "H:\Media\chapters1-2.txt" 2: "H:\Media\video1-2.mkv" 3: "H:\Media\audio1-2.ac3" -log="H:\Media\process1-2.txt"
------------------------------------------------------------------------------
M2TS, 1 video track, 1 audio track, 2 subtitle tracks, 0:56:05
1: Chapters, 13 chapters
2: MPEG2, 480i60 /1.001 (4:3)
3: AC3, English, 2.0 channels, 192kbps, 48khz
4: Subtitle (PGS), Spanish
5: Subtitle (PGS), Portuguese
Creating file "H:\Media\chapters1-2.txt"...
[a03] Extracting audio track number 3...
[v02] Extracting video track number 2...
[v02] Muxing video to Matroska...
[a03] Creating file "H:\Media\audio1-2.ac3"...
[a03] Audio overlaps for 7ms at playtime 0:04:33.  <WARNING>
[a03] Audio overlaps for 18ms at playtime 0:09:52.  <WARNING>
[a03] Audio overlaps for 29ms at playtime 0:15:23.  <WARNING>
[a03] Audio overlaps for 16ms at playtime 0:19:55.  <WARNING>
[a03] Audio overlaps for 30ms at playtime 0:23:57.  <WARNING>
[a03] Audio overlaps for 6ms at playtime 0:28:56.  <WARNING>
[a03] Audio overlaps for 10ms at playtime 0:33:18.  <WARNING>
[a03] Audio overlaps for 9ms at playtime 0:38:05.  <WARNING>
[a03] Audio overlaps for 7ms at playtime 0:42:39.  <WARNING>
[a03] Audio overlaps for 23ms at playtime 0:46:56.  <WARNING>
[a03] Audio overlaps for 6ms at playtime 0:51:22.  <WARNING>
[a03] Starting 2nd pass...
[a03] Realizing (E-)AC3 gaps...
[a03] Creating file "H:\Media\audio1-2.ac3"...
Added fps value to MKV header.
number of frames found would normally go here
eac3to processing took 2 minutes, 9 seconds.
Done.

[madshi]

Originally Posted by ggking7

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.

Ehm, you are aware that eac3to does not remove copy protection? eac3to can only work on HD DVD and Blu-Ray discs which have no copy protection on them.

[ggking7]

DialNorm exists on DVD, since it's in the AC3 stream and it doesn't matter what medium that is on. eac3to should remove it, because it also doesn't really care what medium it is from. ac3 is ac3.

Thank you, does DialNorm only exist on ac3?

[ggking7]

Ehm, you are aware that eac3to does not remove copy protection? eac3to can only work on HD DVD and Blu-Ray discs which have no copy protection on them.

Of course. Does that mean it can't provide title details from the (encrypted) disc? I thought it might since 'wine eac3to /mnt/cdrom' works straight off the disc.

[jamos]

hello great tool. I am using this with scenarist and the dolby lossless profile (True-HD). I am extracting the .ac3 core track extracted from the original true-hd stream and using that as input then choosing Dolby Lossless from the pull down in scenarist then click create files. Then I choose the .thd extracted track instead of the mlp file. seems to work perfectly but am I doing it correct?

[madshi]

Originally Posted by Chumbo

I wanted to report an issue I found. Looks like the number of frames is not reported when eac3to does a 2nd pass.

Will check this out. Probably a bug.

Originally Posted by ggking7

Thank you, does DialNorm only exist on ac3?

AC3, TrueHD and DTS, but it's rarely used on DTS.

Originally Posted by ggking7

Of course. Does that mean it can't provide title details from the (encrypted) disc? I thought it might since 'wine eac3to /mnt/cdrom' works straight off the disc.

I'm surprised that even "wine eac3to /mnt/cdrom" works. eac3to was never intended to work with copy protected content at all and it never will do that.

Originally Posted by jamos

I am using this with scenarist and the dolby lossless profile (True-HD). I am extracting the .ac3 core track extracted from the original true-hd stream and using that as input then choosing Dolby Lossless from the pull down in scenarist then click create files. Then I choose the .thd extracted track instead of the mlp file. seems to work perfectly but am I doing it correct?

What do I know? This is not the Scenarist support thread.

[Kurtnoise]

A sample not detected properly with the latest build... Should be AC3-in-WAV

[madshi]

Originally Posted by Kurtnoise13

A sample not detected properly with the latest build... Should be AC3-in-WAV

That is very strange. I have two AC3WAV samples now in SPDIF format. And the SPDIF header "framesize" field is big endian in one of the files and small endian in the other. I don't really know how it's supposed to be. At least one of the AC3WAV samples I have is clearly incorrect.

Anyway, I've implemented a workaround that makes both samples be handled correctly. Will be fixed in the next build.

[madshi]

Originally Posted by hubblec4

i have a question to demuxing. the sups of some blurays are not correct demuxed. (the same result when i demux with TSMuxer).

the sup-file are very large and contains too much pictures (in this case over 2800).
subtitle 1 till 5 shows the same picture but subtitle 1 till 4 have a starttime only. subtitle 5 is ok. and than subtitle 6 till 8 have the same fault and so on...

That seems to be a bug in SUPread and not in eac3to or tsMuxeR. Try SupRip instead of SUPread. The next eac3to version will output how many subtitles are contained in the demuxed SUP file. It counts 7 subtitles for the sample you've uploaded. SupRip also shows 7 subtitles. However, SUPread shows 26 subtitles and many are duplicates. So SUPread is buggy...

[jamos]

Originally Posted by madshi

What do I know? This is not the Scenarist support thread.

well I thought you may have insight as previously you stated this:

Originally Posted by madshi

If you want to make a demuxed HD DVD TrueHD track (or a Blu-Ray TrueHD track demuxed by an older eac3to version) fit for Scenarist Blu-Ray muxing you can do this:

Code:
eac3to source.thd dest.thd+ac3This will automatically create an AC3 track and interweave it with the TrueHD track. Of course if your source is an HD DVD folder,
you can directly do e.g. "eac3to hddvdMovie 1) 1: video.mkv 3: audio.thd+ac3". That will demux the TrueHD track and interweave an AC3 track at the same time.

but anyways from scenarist forums it is the right way to do what I described. Extract the core ac3 using eac3to.exe from the original true hd stream and also the thd file from the same stream. .thd is the same as .mlp in scenarist. using dolby lossless in scenarist import the ac3 core then it will prompt you for the mlp file just use the .thd one instead. thanks for the great tool.

[mrr19121970]

@madshi, I just demuxed 1 vid & 2 DTS MA (with conversion to AC3):

eac3to v3.04
command line: "E:\TVIX\eac3to\eac3to.exe" "R:\" 1) 2: "D:\DEMUX\Tortured\Video_2.*" 3: "D:\DEMUX\Tortured\Audio_3_German.AC3" 4: "D:\DEMUX\Tortured\Audio_4_English.AC3" -LOG="D:\DEMUX\Tortured\eac3to_PASS3_LOG.LOG"
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 1 subtitle track, 1:47:09, 24p /1.001
1: Chapters, 12 chapters
2: VC-1, 1080p24 /1.001 (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), Undetermined
[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>
[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>
[v02] Extracting video track number 2...
[a03] Extracting audio track number 3...
[a04] Extracting audio track number 4...
[a03] Extracting DTS core...
[a04] Extracting DTS core...
[a03] Decoding with libav/ffmpeg...
[a04] Decoding with libav/ffmpeg...
[a03] Remapping channels...
[a04] Remapping channels...
[a03] Encoding AC3 <640kbps> with libAften...
[a04] Encoding AC3 <640kbps> with libAften...
[v02] Creating file "D:\DEMUX\Tortured\Video_2.vc1"...
[a03] Creating file "D:\DEMUX\Tortured\Audio_3_German.AC3"...
[a04] Creating file "D:\DEMUX\Tortured\Audio_4_English.AC3"...
[a03] Clipping detected, a 2nd pass will be necessary.  <WARNING>
[a03] The original audio track has a constant bit depth of 64 bits.
[a04] The original audio track has a constant bit depth of 64 bits.
[a03] The libav DTS decoder doesn't decode the full DTS-HD information.  <WARNING>
[a03] Starting 2nd pass...
[a03] Extracting audio track number 3...
[a03] Extracting DTS core...
[a03] Decoding with libav/ffmpeg...
[a03] Remapping channels...
[a03] Encoding AC3 <640kbps> with libAften...
[a03] Creating file "D:\DEMUX\Tortured\Audio_3_German.AC3"...
[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.
eac3to processing took 34 minutes, 43 seconds.
Done.

The 1st AC3 track was just fine (as always), but the second had that distinct 'underwater' bubble, crackle very loudly in the background.

I redid singly, and it worked ok.

eac3to v3.04
command line: "E:\TVIX\eac3to\eac3to.exe" "R:\" 1) 4: "D:\DEMUX\Tortured 2nd Try\Audio_4_English.AC3" -LOG="D:\DEMUX\Tortured 2nd Try\eac3to_PASS3_LOG.LOG"
------------------------------------------------------------------------------
M2TS, 1 video track, 2 audio tracks, 1 subtitle track, 1:47:09, 24p /1.001
1: Chapters, 12 chapters
2: VC-1, 1080p24 /1.001 (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), Undetermined
[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>
[a04] Extracting audio track number 4...
[a04] Extracting DTS core...
[a04] Decoding with libav/ffmpeg...
[a04] Remapping channels...
[a04] Encoding AC3 <640kbps> with libAften...
[a04] Creating file "D:\DEMUX\Tortured 2nd Try\Audio_4_English.AC3"...
[a04] The original audio track has a constant bit depth of 64 bits.
Video track 2 contains 154148 frames.
eac3to processing took 7 minutes, 14 seconds.
Done.

regards.

[hubblec4]

Originally Posted by madshi

That seems to be a bug in SUPread and not in eac3to or tsMuxeR. Try SupRip instead of SUPread. The next eac3to version will output how many subtitles are contained in the demuxed SUP file. It counts 7 subtitles for the sample you've uploaded. SupRip also shows 7 subtitles. However, SUPread shows 26 subtitles and many are duplicates. So SUPread is buggy...

thank you madshi.

thats really a bug of SupRead. damn. only this programm can convert bluray.sup to hd-dvd.sup.

hubble

[mbcd]

Originally Posted by hubblec4

thank you madshi.

thats really a bug of SupRead. damn. only this programm can convert bluray.sup to hd-dvd.sup.

hubble

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.

[Chumbo]

Originally Posted by madshi

Will check this out. Probably a bug.
...

Looks like it is. The log (top) in this post above done in 3.04 exhibits the same issue since it's a two-pass process too.

[madshi]

Originally Posted by mrr19121970

@madshi, I just demuxed 1 vid & 2 DTS MA (with conversion to AC3).

The 1st AC3 track was just fine (as always), but the second had that distinct 'underwater' bubble, crackle very loudly in the background.

I redid singly, and it worked ok.

Can I have a sample of the original m2ts file, please? Maybe 50MB or something like that? I'd like to find out why there is a problem when doing it all in one step. That is supposed to work just fine.