eac3to - audio conversion tool

[n0mag!c]

-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I guess 1st message description is outdated?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.php?p=1211516#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]

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/Tais__sample_.dts.html

[madshi]

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?

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).

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...

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...

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]

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]

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]

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]

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.php?p=111181&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]

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]

* 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]

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]

eac3to v2.76 released

http://madshi.net/eac3to.zip



Another revolutionary update, thanks.

_ _ _

[madshi]

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.

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?

Another revolutionary update, thanks.
Pleasure!

[n0mag!c]

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]

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]

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 :devil:

[gillie]

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]

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

[madshi]

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.

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]

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.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.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]

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]

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]

As additional info, you can find (by searching this trhead or the wiki) ...
Madshi,
I didn't even remember there was a wiki (http://en.wikibooks.org/wiki/Eac3to) under development.
Would you please add a link to it in the first post?

[rica]

Madshi,
I didn't even remember there was a wiki (http://en.wikibooks.org/wiki/Eac3to) 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]

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]

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 (http://rapidshare.com/files/165110840/sony_mts.rar.html)

[nwg]

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]

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]

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]

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]

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.

:thanks:

[rica]

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]

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]

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]

what app is SOX?
This is the Sox Homepage (http://sox.sourceforge.net/)

[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... :thanks:
dant3s

[tebasuna51]

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 (http://forum.doom9.org/showthread.php?p=1214986#post1214986) ...

[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]

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]

Your answer in DelayCut thread (http://forum.doom9.org/showthread.php?p=1214986#post1214986) ...
thanks ;)

[Daodan]

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]

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]

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]

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]

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]

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]

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]

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]

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]

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.:confused:

[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.
Will see what I can do...

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.

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...

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.

Is there a similar command to view the properties of TrueHD track as there is for DTS (-logdts)?
No.

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.

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.

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.

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.

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.

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.

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.

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...

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?

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]

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 (http://rapidshare.com/files/166647703/closed_gop.rar.html) 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]

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]

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...

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]

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]

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]

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]

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?

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]

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]

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]

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]

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" (https://www.ip.philips.com/download_attachment/2498/2498.pdf) 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]

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.

5000$ + Non Disclosure Agreement....

But look in "Philips DVD Verifier" (https://www.ip.philips.com/download_attachment/2498/2498.pdf) documentation page 127
Thanks!! Would you mind uploading the first part of that "3ms" VOB sample?

[yesgrey]

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 (http://rapidshare.com/files/166912346/VTS_01_1.rar.html) the sample

[madshi]

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:

http://madshi.net/SSRC/EAC3TO.pnghttp://madshi.net/SSRC/EAC3TO_tone.png
http://madshi.net/SSRC/EAC3TO_passband.pnghttp://madshi.net/SSRC/EAC3TO_transition.png
http://madshi.net/SSRC/EAC3TO_phase.pnghttp://madshi.net/SSRC/EAC3TO_pulse.png

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

[madshi]

Here (http://rapidshare.com/files/166912346/VTS_01_1.rar.html) 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]

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...

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!...:D

[madshi]

Would that be a correct assumption?
Yes.

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:

http://i11.photobucket.com/albums/a199/rack04/1.jpg

http://i11.photobucket.com/albums/a199/rack04/2.jpg

[yesgrey]

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]

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]

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.

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...

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...

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.

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 (http://forum.doom9.org/showthread.php?p=1214448#post1214448)

[madshi]

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?

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]

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]

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]

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?)Which discs do have DD+ commentary tracks without DD core?
Transformers blu-ray.

[evdberg]

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.
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]

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]

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]

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 (http://rapidshare.com/files/167564714/00062.rar.html) is the sample (75 mb)

[madshi]

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 (http://rapidshare.com/files/167564714/00062.rar.html) 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]

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]

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.

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".

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]

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]

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]

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.

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.

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]

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]

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]

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 :p

[madshi]

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]

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]

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]

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]

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]

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]

MKV doesn't support TrueHD (yet?).
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]

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...

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.

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.

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]

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]

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.

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 (http://forum.doom9.org/showthread.php?p=1218003#post1218003)

[n0mag!c]

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]

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]

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]

..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]

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]

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.

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...

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...

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]

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]

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]

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]

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]

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...

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]

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.

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]

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]

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.....:D
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]

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]

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]

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.

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=fc577931a2d88b41d2db6fb9a8902bda

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]

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]

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.

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]

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]

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]

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]

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]

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]

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]

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]

Will be fixed in the next build.

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

[Thunderbolt8]

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]

Just out of curiosity, what was the issue? Bad FLAC track possibly?
No runtime information in the FLAC track. That threw eac3to off.

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]

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]

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

:search:

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

[tebasuna51]

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]

Another idea:
DIRAC time stretching (http://www.dspdimension.com/technology-licensing/dirac/). 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]

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]

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]

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...

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]

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.php?file=./jul.2007/Digiturk.HD.Sirius.2.4.8E.10.jul.2007.ts

[madshi]

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]

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]

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]

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]

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.