eac3to - audio conversion tool

[Thunderbolt8]

edited

[Thunderbolt8]

I've spent too much time on eac3to lately.
nah, not too much time, the final changes made it almost perfect ;)

[madshi]

By the way, do you have plans to implement STDIN streaming for the source?
No plans as of yet. What would that be good for?

[madshi]

eac3to v1.22 released

http://madshi.net/eac3to.zip

* 6.1 -> 7.1 channel doubling was sometimes incorrectly skipped
* OS speaker settings now don't have to be 7.1, anymore
* added detection of 5.1 output when 6.1 was expected
* DTS and DTS-ES files are now forcefully patched to 24 bit by eac3to (workaround for Sonic decoder)
* Sonic Audio Decoder is now always used by default for DTS decoding
See also updated first post in this thread.

[madshi]

I've noticed some strange things with DTS 6.1 tracks:

The Nero decoder always outputs 5.1, but the back channel seems to be mixed into the surround channels!

The Sonic decoder most of the time gives out proper 6.1. However, with I've one sample where it only outputs 5.1, although the track is definitely 6.1 Discrete. Very strange. With this one special track I can see the Nero has the back channel mixed into the surrounds again. With Sonic I only get 5.1 and the back channel is totally missing. Fortunately for 4 other sample Discrete tracks Sonic gives out proper 6.1.

[nautilus7]

eac3to v1.22 released
:thanks:

* OS speaker settings now don't have to be 7.1, anymore

Do i have to set OS speakers to 5.1 to get a 5.1 output, or stereo is just fine?

[madshi]

Here's a little goodie for all ya Windows FLAC users:

http://forum.doom9.org/showthread.php?t=130498

[madshi]

Do i have to set OS spekers to 5.1 to get a 5.1 output, or stereo is just fine?
Stereo should be fine. The OS speaker settings shouldn't matter at all, anymore.

[honai]

No plans as of yet. What would that be good for?

I'm writing a bridge tool between your tool and an EVOB demuxer. That way I could just invoke your tool with demuxed audio streams without storing them as separate files beforehand.

[madshi]

I'm writing a bridge tool between your tool and an EVOB demuxer. That way I could just invoke your tool with demuxed audio streams without storing them as separate files beforehand.
I see. The problem is that eac3to is using DirectShow filters and they want to load a file. So even if I added STDIN support, I'd have to store the files on harddisk, anyway...

[honai]

The problem is that eac3to is using DirectShow filters and they want to load a file.

For all audio stream types?

[nautilus7]

Stereo should be fine. The OS speaker settings shouldn't matter at all, anymore.I 've just decode a 5.1 dts having set OS speaker setting to headphones (stereo). It worked fine! Thanks!

[madshi]

For all audio stream types?
No, but for all compressed audio formats. Basically both decoding and encoding only works with files on harddisk. Things eac3to does itself (channel mapping, raw to wav(s) conversion, dts/ac3 core extraction, dialnorm removal, etc) could theoretically be done without hard disk. But the question is whether this all makes sense if only some parts of eac3to would get along without real files.

[superx]

E-AC3, 5.1 channels, 1:00:13, 640kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Decoding eac3 file to raw. Please wait...
Getting "Nero File Source / Splitter" and "DTS/AC3/DD+ Source" instances failed.


This is the error I get can anyone help me with it.

Thanks

[hristoff2]

E-AC3, 5.1 channels, 1:00:13, 640kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Decoding eac3 file to raw. Please wait...
Getting "Nero File Source / Splitter" and "DTS/AC3/DD+ Source" instances failed.


This is the error I get can anyone help me with it.

Thanks

Install Nero 7.x.x.x properly & register the HDDVD/Blu-Ray Plugin, or get the Sonic Decoders (beware, DRC applied on eac3 tracks) & ac3dts source (the new one, with eac3 support)

-> http://uploaded.to/?id=y73jnw ('dtsac3eac3' source)

[Beastie Boy]

Is it still possible to buy Nero 7? I see only ver 8 on the web site.

[superx]

Install Nero 7.x.x.x properly & register the HDDVD/Blu-Ray Plugin, or get the Sonic Decoders (beware, DRC applied on eac3 tracks) & ac3dts source (the new one, with eac3 support)

-> http://uploaded.to/?id=y73jnw ('dtsac3eac3' source)


how do I register the hdvd/blu-ray plugin, where do I get this from.

I just purchased nero 8 will that screw anything up

[The_Keymaker]

@ALL

I am still working on my WiFi remote control system. But will still try and support EAC3toGUI.

From what I can determine, -down6 and +/-100ms are the only options my GUI does NOT currently support. I will try and update EAC3toGUI with the new parameters this weekend.

Before I can update the program, I need someone to explain to me how to implement the +/-100ms command line parameter, i.e., is it +100 for advance and -100 for delay? and can it be any number from 0 to 100ms or is it fixed at 100ms?

Regards,
The_Keymaker

[madshi]

how do I register the hdvd/blu-ray plugin, where do I get this from.
You need to buy the plugin, too, in case you haven't yet. It's not part of the normal Nero package. Once you bought it you'll get an extra serial number for the plugin which you can then feed to the Nero Product Manager (or whatever it was called).

I just purchased nero 8 will that screw anything up
eac3to doesn't work with Nero 8. You need to install Nero 7 if you want to do TrueHD and E-AC3 decoding with eac3to.

[TheSof]

Can't imagine that there'd be a difference between v1.17 and v1.18. Except if you fed the EVO file into v1.17 instead of the demuxed file? Please check if the 2:47 FLAC stays in sync throughout the movie (after you applied the eventually necessary static delay). Maybe the 2:47 has some extra seconds of silence at the end or beginning of the movie? Don't know...

I'm using it on the demuxed thd.T he flac has about 10 mins extra runtime. The FLAC does not stay in sync, it will keep droping out and then coming back. Its very difficult to tell how far out of sync it is at the end but about 10 mins seems right. With 1.17 it is fine, albeit with the dial norm and thus 24bit.

Is anyone having success with titles such as Superman Returns and V for Vendetta?

[menlvd]

epp me to reencoding thd longer that reencoding eac3
thd from HD-DvD
eac3to.exe ver 1.22.0.0

[madshi]

I'm using it on the demuxed thd.T he flac has about 10 mins extra runtime. The FLAC does not stay in sync, it will keep droping out and then coming back. Its very difficult to tell how far out of sync it is at the end but about 10 mins seems right. With 1.17 it is fine, albeit with the dial norm and thus 24bit.

Is anyone having success with titles such as Superman Returns and V for Vendetta?
Can you please retry with v1.22? If the problem still exists in that version, I'll do some more checks.

[superx]

k thanks, where on the website can I get the hddvd/blu-ray plugin, they changed it around.

[Thunderbolt8]

Another problem: I tried to convert the 5.1 TrueHD 24-bit track of letters from iwo jima to FLAC (using eac3to v1.22; that track was the 2nd track on the disc btw, not the 1st). the thd track size is 1,32GB, the final FLAC has 3.2GB of size. movie length is 2h 20min. the strange thing is now that the flac file has a length of 3h 30min!
when playing this together with the .mkv movie, the fps of 23.976 are kept, but the sound is horribly slowed down (and of course, theres also no picture beyond the 2h 20 min mark). how can this be possible? is it possible that demuxed sources (and the remux to temp .evo) give some trouble because some information about playback, speed for example, can get lost? maybe it would help if you enable .evo input as source again, so we could test and compare this.

just converted the 2nd track on the disc, DD+ 5.1, to ac3 just to compare and in this case the original length was kept at 2h 20 mins (no muxing to temp .evo took place).
converted the DD+ 5.1 to flac, again to compare, length was 2h 20min (again no temp evo muxing took place).
(there was a length difference between the converted ac3 and flac track, both from the dd+ 5.1 source btw., the ac3 track is 32ms longer. how can this be possible with the same source file?)

on the other hand, that other DD+ 2.0 commentary track from another movie I converted to flac, there also didnt take place any temp evo muxing, length is fine, but still it speeds up the .mkv movie.



p.s. when rebuilding the video .evo, powerdvd wasnt able to jump into the 2nd half of the rebuilt evo, from whereon the 2nd evo begins. the time would still continue, but no picture is shown. after remuxing to .mkv with gdsmux & mkvmerge this was possible though with ffdshow, I guess this is normal for powerdvd and files up to a certain size?

[madshi]

eac3to v1.23 released

http://madshi.net/eac3to.zip

* bugfix: sometimes TrueHD decoding resulted in incorrect sampling rate
The above bug could result in e.g. FLAC files with 44,1khz instead of the correct 48khz. This results in lower than correct pitch and longer runtime. This is most probably what TheSof experienced and now also Thunderbolt8. Should be fixed now.

[Thunderbolt8]

lol thanks!
testing now, results in mb ~25 mins :P

[madshi]

From what I can determine, -down6 and +/-100ms are the only options my GUI does NOT currently support. I will try and update EAC3toGUI with the new parameters this weekend.

Before I can update the program, I need someone to explain to me how to implement the +/-100ms command line parameter, i.e., is it +100 for advance and -100 for delay? and can it be any number from 0 to 100ms or is it fixed at 100ms?
Thanks. The delays theoretically support any value betwee "-infinite ms" and "+infinite ms". E.g. "-17ms" is possible, or "+53862ms". I think you need to add an edit box. The delay works similar to how "delaycut" works or how you can specify a delay in MPC. So e.g. "+100ms" delays the audio track by 100ms (basically 100ms of silence is added at the beginning of the audio track). And e.g. "-200ms" cuts 200ms worth of audio data in the beginning of the audio track.

[madshi]

I tried to convert the 5.1 TrueHD 24-bit track of letters from iwo jima to FLAC (using eac3to v1.22; that track was the 2nd track on the disc btw, not the 1st). the thd track size is 1,32GB, the final FLAC has 3.2GB of size.
The TrueHD track is unlikely to be 24bit, if it's only 1,32GB. I think it's probably 16bit. The FLAC shouldn't be bigger than the TrueHD track. Usually FLAC files are a little bit smaller than the original TrueHD track. I can only guess that either the dialnorm removal didn't work properly, or the bug I just fixed in v1.23 messed up more than just the sampling rate.

p.s. when rebuilding the video .evo, powerdvd wasnt able to jump into the 2nd half of the rebuilt evo, from whereon the 2nd evo begins. the time would still continue, but no picture is shown. after remuxing to .mkv with gdsmux & mkvmerge this was possible though with ffdshow, I guess this is normal for powerdvd and files up to a certain size?
I don't think it depends on the size. I don't have any experience with how PowerDVD handles such files, but sometimes seeking doesn't work if the timecodes are not correct. You could try using OffsetPTS before joining the EVO files. If there's nothing to fix, OffsetPTS will simply do nothing but report that everything is fine. This is the case for most HD DVDs. However, sometimes the timecodes of the 2nd EVO are strange. This happens especially with HD DVDs from independent European studios. In that case OffsetPTS will correct the timecodes of the 2nd EVO file. This might make seeking possible in PowerDVD.

[Thunderbolt8]

did use offsetpts before any rebuilding & demuxing of course, but it reported everything to be fine. the only case it was needed for me was indeed so far mulholland drive from studio canal.

oh, now I see it, that truehd 5.1 flac created with v1.22 only had 32Khz :P

edit: ok tested it now with 1.23 and length is fine! thanks! guess only that commentary track video speedup remains for me atm :P

btw. it also seems like the delay it that tiny bit better with +30 ms, the +30 ms the ac3 track was longer than the flac track. how does it come delay modification is apparently needed only for flac tracks?

btw. dont know if that helps, but that normalization removal thing for all the tracks (trueHD and DD+) I recently converted always said -27db. no other value than this one. same for the bitrate for all the trueHD and dd+ tracks, always got the info that the current track contains more than 16-bit of information -> 24-bit. so far all the flac tracks I made, not only from trueHD, but also from DD+ were quite bigger than the original track sizes, so that normalization removal problem might be a more general one and not only restricted to the truehd tracks

[superx]

ok reinstalled my older version of Nero, I can't purchase Blu-ray/hd-dvd plugin, it keeps taking me to the product page of nero 8.

when I reinstalled nero 7 this is what I got.

E-AC3, 5.1 channels, 1:00:13, 640kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Decoding eac3 file to raw. Please wait...
Disabling DRC for Nero E-AC3 decoding...
The file size of the raw file doesn't seem to fit.
The expected file size for 16 bit is 1.93 GB.
The expected file size for 24 bit is 2.90 GB.
The real file size is 0 Bytes.

[Beastie Boy]

k thanks, where on the website can I get the hddvd/blu-ray plugin, they changed it around.

Here (http://www.nero.com/eng/store-blu-ray.html)

Cheers, Beastie.

[madshi]

how does it come delay modification is apparently needed only for flac tracks?
I don't know. But it is also needed for E-AC3 tracks sometimes (as I've already told you at least 10 times). It is probably also needed for DTS-HD tracks.

btw. dont know if that helps, but that normalization removal thing for all the tracks (trueHD and DD+) I recently converted always said -27db. no other value than this one.
Yeah, -27db is the default value. The studios should in theory adjust this value, but in practical life they rarely do.

same for the bitrate for all the trueHD and dd+ tracks, always got the info that the current track contains more than 16-bit of information -> 24-bit. so far all the flac tracks I made, not only from trueHD, but also from DD+ were quite bigger than the original track sizes, so that normalization removal problem might be a more general one and not only restricted to the truehd tracks
It is no surprise that DD+ tracks contain more than 16bit. There was never any doubt about that. It's also no surprise that DD+ 24bit FLAC files are MUCH bigger than the original DD+ track. It was very clear that this would happen.

However, it is not "right" that your FLAC tracks are bigger than the TrueHD tracks. There's definitely something wrong there! Since I added dialnorm removal for TrueHD tracks, I've never seen FLAC files getting bigger than the original TrueHD file on my PC yet!

What do the other eac3to users say? Are your FLAC files also sometimes (or often) bigger than the original TrueHD files?

[madshi]

ok reinstalled my older version of Nero, I can't purchase Blu-ray/hd-dvd plugin, it keeps taking me to the product page of nero 8.

when I reinstalled nero 7 this is what I got.

E-AC3, 5.1 channels, 1:00:13, 640kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Decoding eac3 file to raw. Please wait...
Disabling DRC for Nero E-AC3 decoding...
The file size of the raw file doesn't seem to fit.
The expected file size for 16 bit is 1.93 GB.
The expected file size for 24 bit is 2.90 GB.
The real file size is 0 Bytes.
This won't work without the plugin.

[tebasuna51]

(there was a length difference between the converted ac3 and flac track, both from the dd+ 5.1 source btw., the ac3 track is 32ms longer. how can this be possible with the same source file?)

Each ac3 frame (48 KHz) is 32 ms long, then an ac3 duration is always a multiple of 32 ms.

All ac3 encoders put a 5.33 ms. delay (256 samples at 48 KHz) with silence to preserve the initial value of uncompressed audio. This delay (only 5.33 sec) can force a new frame, at the end, 32 ms. long, the rest is filled with silence.

Using Aften parameter -pad 0 the delay disappear, but the first 5.33 ms. is not the same than uncompressed value.

[Thunderbolt8]

What do the other eac3to users say? Are your FLAC files also sometimes (or often) bigger than the original TrueHD files?maybe its just me and I make a mistake somewhere. but not sure in how far thats possible with eac3to's commandline any more.
what im doing is (for audio decoding)
1. checking both .evo files with offsetpts
2. unchecking all other audio/video/subtitle than the audio track(s) I want to use for my remuxes (evodemux).
3. giving them the correct endings (.thd for truehd tracks, .eac3 for dd+ tracks)
4. use them with eac3to with "eac3to source.thd/source.eac3 dest.flac"
(5. put them in .mka afterwards, but this doesnt matter for the flac filesize).

thats it basically. unless there some mistake in what im doing or all this occurs only because we have only used different movies so far, which apparently seem to have differences with their truehd tracks, there seems to be a difference in what we are doing or maybe also somehow with our eac3to versions (or maybe nero/sonic filters? is there any way how I can check if the correct filters are installed?)

[Thunderbolt8]

Each ac3 frame (48 KHz) is 32 ms long, then an ac3 duration is always a multiple of 32 ms.

All ac3 encoders put a 5.33 ms. delay (256 samples at 48 KHz) with silence to preserve the initial value of uncompressed audio. This delay (only 5.33 sec) can force a new frame, at the end, 32 ms. long, the rest is filled with silence.

Using Aften parameter -pad 0 the delay disappear, but the first 5.33 ms. is not the same than uncompressed value.
im not quite sure whether I understood this correctly, are you saying that converted eac3 -> ac3 track is now 32 or 5.33ms too long (which value is right)?

[madshi]

Each ac3 frame (48 KHz) is 32 ms long, then an ac3 duration is always a multiple of 32 ms.

All ac3 encoders put a 5.33 ms. delay (256 samples at 48 KHz) with silence to preserve the initial value of uncompressed audio. This delay (only 5.33 sec) can force a new frame, at the end, 32 ms. long, the rest is filled with silence.

Using Aften parameter -pad 0 the delay disappear, but the first 5.33 ms. is not the same than uncompressed value.
Thanks, that's interesting!

[madshi]

maybe its just me and I make a mistake somewhere. but not sure in how far thats possible with eac3to's commandline any more.
what im doing is (for audio decoding)
1. checking both .evo files with offsetpts
2. unchecking all other audio/video/subtitle than the audio track(s) I want to use for my remuxes (evodemux).
3. giving them the correct endings (.thd for truehd tracks, .eac3 for dd+ tracks)
4. use them with eac3to with "eac3to source.thd/source.eac3 dest.flac"
Are you rebuilding the audio files or demuxing them?

Can you send me a 10MB sample of a TrueHD file where the FLAC file ends up being larger than the TrueHD file? Before you send me that sample, please run the sample through eac3to and check if it reports that the audio data contains more than 16bit of information. Sending me the sample makes sense only if eac3to reports that. Thanks!

P.S: You could also name the movies you've tried so far. Maybe I own one of them, too.

[Thunderbolt8]

im just demuxing them (from the 2 original .evo files), as .evo as input doesnt work any more (of course, I could first rebuilt a big audio file only, load it into evodemux and demux from there then, but this takes more time of course. could try it though, if you think its worth testing and comparing). edit: did it, both files, the direclty demuxed one and the rebuilt evo -> demuxed from there one both have the same big file size.

im not really familiar with that cutting stuff, how can I cut samples from trueHD/flac/other audio tracks? does it work that easily that I could just stop the demuxing process after 10mb or would this sample then be invalid somehow?

movies ive tried so far (all hddvd):

fear and loathing in las vegas (truehd)
out of sight (dd+)
eternal sunshine of the spotless mind (dd+)
letters from iwo jima (truehd)

[madshi]

im not really familiar with that cutting stuff, how can I cut samples from trueHD/flac/other audio tracks? does it work that easily that I could just stop the demuxing process after 10mb
Yes, just stop demuxing after the audio file is 10MB big. Or else use a hexeditor to cut the file off after 10MB.

fear and loathing in las vegas (truehd)
out of sight (dd+)
eternal sunshine of the spotless mind (dd+)
letters from iwo jima (truehd)
The dd+ files are irrelevant. They're always 24bit. Only the TrueHD tracks are interesting. Fear and Loathing is really 24bit, I believe. The TrueHD track should be very big. Iwo Jima should be 16bit. So a sample of Iwo Jima might help, if the problem is reproducable.

[Thunderbolt8]

made a 21,4mb sample from the letters trueHD track (couldnt get to press cancel in evodemux any faster :P)
used eac3to conversion to test 'eac3to letters.thd letters.flac' and got the information that this track contains more than 16-bit of information.
flac size is 52,5mb

http://www.megaupload.com/?d=P5OV5GZP

btw. is the information for trueHD tracks that evodemux gives in the VTI window (quantization field) regarding their bit depth always accurate? or does it always show trueHD tracks with 24-bit, even though they only are 'real' 16-bit tracks sometimes?

[tebasuna51]

im not quite sure whether I understood this correctly, are you saying that converted eac3 -> ac3 track is now 32 or 5.33ms too long (which value is right)?

Always the ac3 is 5.33 ms. delayed, but long is any value between 5.33 and 37.33 ms.

Example: we have a source 59 ms. long, after the delay we have 64.33 ms., we need 3 frames (3 x 32 ms = 96 ms) then is 37 ms. too long.

[madshi]

made a 21,4mb sample from the letters trueHD track (couldnt get to press cancel in evodemux any faster :P)
used eac3to conversion to test 'eac3to letters.thd letters.flac' and got the information that this track contains more than 16-bit of information.
flac size is 52,5mb
Thanks for the sample. Here's what I'm getting on my PC:

C:\Desktop>c:\sources\eac3to\eac3to letters.thd letters.flac
TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
The following tasks are executed now. Please wait...
- remove Dialog Normalization information
- muxing TrueHD track into temp evo file
Decoding TrueHD track to raw. Please wait...
Checking raw data bitdepth. Please wait...
This TrueHD track contains only 16 bit of information.
The zero bytes were successfully removed from the raw file.
Encoding the raw file to FLAC. Please wait...

Done.
letters.thd: 21.2MB
letters.flac: 18.5MB

I've no idea why this doesn't seem to work on your PC!!! Maybe you have a different version of the TrueHD decoder compared to mine? Maybe yours is applying DRC?

btw. is the information for trueHD tracks that evodemux gives in the VTI window (quantization field) regarding their bit depth always accurate? or does it always show trueHD tracks with 24-bit, even though they only are 'real' 16-bit tracks sometimes?
I'm not sure. The best hint is the file size. If the TrueHD file is smaller than 2GB, it's most probably a 16bit track.

[Thunderbolt8]

trueHD filesize is 1.32GB

allright, again how can I check all my filters (nero, sonic) such to be absolutely sure they are the same/right ones you also use?
theres should be at least no mistake in the eac3to directory, as I always overwrite the old files with the new ones.
and eac3to is also not complaining that a filter is missing or something like that. but maybe I got an old version of something somwhere :S

LOL I got it
checked the nero product setup and beside that HDDVD/blue-ray plugin there were also a BD test and HDDVD test plugin installed. I remember once having installed them few days ago, because I wasnt sure if that hddvd/blue-ray plugin which was already registered there worked properly, due to other eac3to bugs at that time.

"This TrueHD track contains only 16 bit of information.
The zero bytes were successfully removed from the raw file."
flac file size is 18,5mb now -.-

shit, it was my own fault. sorry for taking up so much of your time with that :/

[TheSof]

Can you please retry with v1.22? If the problem still exists in that version, I'll do some more checks.

Retried with latest version (1.23) and all is welll again with regard to runtimes, although I will not have the chance to hear the results until later.

True HD file size is 1.57GB

TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
The following tasks are executed now. Please wait...
- remove Dialog Normalization information
- muxing TrueHD track into temp evo file
Decoding TrueHD track to raw. Please wait...
Checking raw data bitdepth. Please wait...
This TrueHD track contains only 16 bit of information.
The zero bytes were successfully removed from the raw file.
Encoding the raw file to FLAC. Please wait...

Done.

FLAC file size is 1.39 GB.

Runtime now matches, being 2:34:19. This is with Superman Returns. When I get a chance to play it I will post the delay info.

Thanks madshi.

EDIT - comparing the file properties of the 1.18 flac and the 1.23 flac, the difference was the sampling rate, 44 vs 48, so you were correct, that was the issue. Thanks for fixing it.

[madshi]

shit, it was my own fault. sorry for taking up so much of your time with that :/
No problem. Glad it's solved now.

[madshi]

Runtime now matches, being 2:34:19. This is with Superman Returns. When I get a chance to play it I will post the delay info.
Thanks. Please also post the EvoDemux log of that TrueHD track. That goes to everyone: If you convert your TrueHD files to FLAC (or anything else) with the latest eac3to version, and if you need to apply a manual delay, please post your final delay value and the log from EvoDemux. Collecting this kind of information helps other people (they don't need to figure out the delay themselves) and it might also help finding a common base on how to calculate the delay automatically for a future eac3to version.

[TheSof]

Quick question, where is the best place to add the delay for flac - with eac3to or when muxing the mkv?

[madshi]

Quick question, where is the best place to add the delay for flac - with eac3to or when muxing the mkv?
I don't know if mkvtoolnix can properly delay FLAC files. Can it? If it does, it might be the faster way to handle the delay. However, if you demux and remux the FLAC file, the delay might be gone (I'm not sure). When applying the delay with eac3to it's an everlasting fix/change.

[TheSof]

I don't know if mkvtoolnix can properly delay FLAC files. Can it? If it does, it might be the faster way to handle the delay. However, if you demux and remux the FLAC file, the delay might be gone (I'm not sure). When applying the delay with eac3to it's an everlasting fix/change.

I'll stick to eac3to then.

[Thunderbolt8]

ill demux the truehd track again with evodemux for the log file, once that conversion here is done (that bitdepth checking takes up lot of time now, was over before in 1 second :P)

should I post the complete log with all infos from all tracks or only certain parts or only the part when he begins to work with the demuxing?

[madshi]

ill demux the truehd track again with evodemux for the log file, once that conversion here is done (that bitdepth checking takes up lot of time now, was over before in 1 second :P)

should I post the complete log with all infos from all tracks or only certain parts or only the part when he begins to work with the demuxing?
I don't need the demuxing log. I just need some parts of the information EvoDemux posts when you load the EVO file. These infos should be enough:

PTM of first video frame = 00000123
VC-1 video stream 0 found!
First PTS = 00000123
Dolby TrueHD audio stream 0 found!
First PTS = 00000123
Bitdepth checking takes longer now because it aborts as soon as more than 16bits are found, while it runs through until the end of the raw file if it never finds more than 16bit. The reason for that is that the bitdepth checking already strips the zero bytes in a temporary file so that after checking there doesn't need to be an additional step for removing the zero bytes. So the proper name would be "bitdepth checking & zero byte removal". It's done in one step.

[Thunderbolt8]

I guess in cases we would have to apply offsetpts we should do this then before posting these infos, right?


for TrueHD 5.1 of letters from iwo jima (1st EVO)

PTM of first video frame = 6736B4B1
VC-1 video stream 0 found!
First PTS = 2736B4B1 (+35791394ms)
Dolby TrueHD audio stream 1 found!
First PTS = 2736B4B1 (+35791394ms)

Dolby Digital Plus audio stream 0 found!
First PTS = 6736B4B1


2h 20mn 37s 322ms = length of truehd -> flac
2h 20mn 37s 472ms = length of dd+ -> ac3
2h 20mn 37s 440ms = length of dd+ -> flac

normal truehd -> flac length seemed to need a little delay, which was apparently frome some scenes. decided again to try the mathematical way and to take the difference between the truehd and eac3 track. chose the length of the eac3 -> flac track, because it might be a little more accurate due to those additional 32ms for ac3 frames. just seemed to make most sense to me to compare both of the flac conversions to each other. delay needed is 118ms then. I cant say that this is 100% exact the same delay now as on the source, but I would guess it might be, one just cant be sure 100% without being able to calculate it. at least it was the most accurate value for me so far, so I decided take this one.

118ms is btw also exact the same value I took as delay for the fear & loathing trueHD track. cant say if this is coincidence, just did (almost) the same here, converted the dd+ track to ac3 (and flac? cant remember any more if I converted the dd+ to both) and compared its length to the truehd -> flac track. the difference I got from there was also 118ms, same as here with letters from iwo jima.

values for fear & loathing TrueHD 5.1

Opening file FEATURE_1.EVO

PTM of first video frame = 00000D8E
Dolby TrueHD audio stream 0 found!
First PTS = 00000D8E

edit: on the other hand the delay for "letters..." calculated from the eac3 -> ac3 compared to truehd -> clac track of 150ms then could fit as well. too hard to compare 118ms and 150ms with only a difference of 32ms only with the eyes -.-

[madshi]

I guess in cases we would have to apply offsetpts we should do this then before posting these infos, right?
OffsetPTS only changes the timestamps of the 2nd EVO file and those are not important for our needs. So OffsetPTS doesn't make any difference for audio delay.

for TrueHD 5.1 of letters from iwo jima (1st EVO)
PTM of first video frame = 6736B4B1
VC-1 video stream 0 found!
First PTS = 2736B4B1 (+35791394ms)
Dolby TrueHD audio stream 1 found!
First PTS = 2736B4B1 (+35791394ms)
Dolby Digital Plus audio stream 0 found!
First PTS = 6736B4B1
Wow, those are quite crazy!

118ms is btw also exact the same value I took as delay for the fear & loathing trueHD track.
I've had 200ms with two TrueHD tracks as of yet.

[TheSof]

I'm having some problems getting mpc to play the flac as a dub alongside an mkv to find the delay. It just won't open the flac using madflac or any others. Graphedit plays them fine. Anyone know what I'm doing wrong?

[madshi]

I'm having some problems getting mpc to play the flac as a dub alongside an mkv to find the delay. It just won't open the flac using madflac or any others. Graphedit plays them fine. Anyone know what I'm doing wrong?
You need to tell MPC to load the FLAC file by starting MPC with the "dub" parameter via command line.

[TheSof]

Let me start off by saying that I hate finding delays - I'm terrible at it and it takes me ages.

Having said that, I thought that superman returns stayed insync through out, without any delay. I'm going to try V for Vendetta and Harry Potter next.

[madshi]

Let me start off by saying that I hate finding delays - I'm terrible at it and it takes me ages.
As with everything, you'll improve with some practising... :p

Having said that, I thought that superman returns stayed insync through out, without any delay. I'm going to try V for Vendetta and Harry Potter next.
That's fine.

[Thunderbolt8]

should we also post a log for cases we believe there is no delay needed at all?

[madshi]

should we also post a log for cases we believe there is no delay needed at all?
Well, it might be helpful. At least it won't harm.

[Thunderbolt8]

after all its only us who believe(!) we dont need any delay :P

[honai]

If you convert your TrueHD files to FLAC (or anything else) with the latest eac3to version, and if you need to apply a manual delay, please post your final delay value and the log from EvoDemux. Collecting this kind of information helps other people

Personally, I don't think that's such a good idea. For one thing - and I really don't mean to be rude - a great many people just don't know what they're doing. But the biggest problem is that, while the effort may be well-intentioned, specific setups might yield a delay that's only valid for the environment of that specific user.

What I mean is this:
- the user might have a filter in the audio chain that causes a small delay, e.g. for downconverting multi-channel to stereo setups, or applying inherent effects on the sound card (many Creative card exhibit this)
- the user might be using an LCD display with implicit video delay (most overdrive/PVA panels and most panels with 10-bit LUTs exhibit a delay of up to 50ms caused by the display's DSP circuitry)
- if running audio through a receiver that might also introduce another delay

These factors could quickly sum up into the 100ms range, i.e. a/v is perfectly in sync to the user, but only for his specific setup, while the actual a/v sync is off by up to 100ms!

[TheSof]

I still think it's a good idea, even if it's just to know how many need delay and how many don't

We have to assume that the user is already adding a global delay via receiver, soundcard or ffdshow etc. if it's needed in that particular environment.

I also don't think they should be taken as concrete proof, but just as a guideline, perhaps as a starting point.

[madshi]

Personally, I don't think that's such a good idea. For one thing - and I really don't mean to be rude - a great many people just don't know what they're doing. But the biggest problem is that, while the effort may be well-intentioned, specific setups might yield a delay that's only valid for the environment of that specific user.

What I mean is this:
- the user might have a filter in the audio chain that causes a small delay, e.g. for downconverting multi-channel to stereo setups, or applying inherent effects on the sound card (many Creative card exhibit this)
- the user might be using an LCD display with implicit video delay (most overdrive/PVA panels and most panels with 10-bit LUTs exhibit a delay of up to 50ms caused by the display's DSP circuitry)
- if running audio through a receiver that might also introduce another delay

These factors could quickly sum up into the 100ms range, i.e. a/v is perfectly in sync to the user, but only for his specific setup, while the actual a/v sync is off by up to 100ms!
I see what you mean but I think users which need such a setup related delay will hopefully have dialed that in in the receiver. I mean otherwise they'd have to apply a delay to every SD-DVD they rip, too! And to any movie they might download.

[honai]

I see what you mean but I think users which need such a setup related delay will hopefully have dialed that in in the receiver. I mean otherwise they'd have to apply a delay to every SD-DVD they rip, too! And to any movie they might download.

Well, many users have really, really low expectations, and while some may notice that "something" is wrong they couldn't nail it where and what went wrong.

I'm just saying that there are too many factors in the equation to come up with a reliable way of determining the actual delay.

[Thunderbolt8]

could anyone already tell please how blue ray tracks behave with the current eac3to version? was it said somewhere that the LPCM tracks wont suffer from delay when converting to flac or do I have kept something wrong in my mind regarding that aspect?
has anyone already tried converting blueray trueHD tracks to flac, if yes do they also need delay like (most of) those on HDDVDs ?

[madshi]

could anyone already tell please how blue ray tracks behave with the current eac3to version? was it said somewhere that the LPCM tracks wont suffer from delay when converting to flac or do I have kept something wrong in my mind regarding that aspect?
has anyone already tried converting blueray trueHD tracks to flac, if yes do they also need delay like (most of) those on HDDVDs ?
The conversion to FLAC is usually not what is creating any delays. It's the demuxing. If you encode a decoded TrueHD or LPCM track to FLAC or AC3 or DTS shouldn't matter much.

About LPCM tracks: I'm not sure myself. I have converted some LPCM tracks to FLAC, but I don't remember if I had to apply a delay or not. I don't consider it a big problem, so that's probably why I don't remember...

[honai]

On the last few Blu-Rays that I converted - 300, The Prestige, The Lives of Others, Pirates of the Caribbean - no delay was needed.

[Thunderbolt8]

did you try it with both 300 tracks, LPCM and trueHD, or only with the LPCM one? could be interesting to know

[madshi]

did you try it with both 300 tracks, LPCM and trueHD, or only with the LPCM one? could be interesting to know
Just for my interest I compared different tracks from the 300 Blu-Ray disc today. It's an interesting disc because several tracks are available in two different formats:

(1) There's a 16bit LPCM track.
(2) There's a 16bit TrueHD track.
(3) There's a 640kbps AC3 track interweaved with the TrueHD track.
(4) There's a separate 640kbps AC3 track.

I've compared (3) and (4) first. The AC3 track from (3) was extracted by using eac3to. Both AC3 tracks are 100% identical - except the last two AC3 frames, which are different for whatever reason. That's 64ms worth of audio data at the end of the movie. Everything else is bit perfectly the same. So also the delay is the same.

Then I've compared (1) and (2). Here it gets a bit funny. There are the following differences between the two tracks:

- there's 105ms of additional audio data in the beginning of the LPCM track
- there's 98ms of additional audio data at the end of the LPCM track
- at timecode 0:00:35 there's additional 20ms audio data in the LPCM track

Apart from these differences the tracks are bit perfectly identical. That means we have the proof now that TrueHD decoding with eac3to and Nero 7 is bit perfect. However, those 20ms of additional audio data in the LPCM track at timecode 0:00:35 are confusing me. I'm not sure where this is coming from. Maybe there's a TrueHD frame which was not decoded for whatever reason? It's really strange...

[Thunderbolt8]

did you compare the original LPCM and trueHD tracks, or both in their converted to flac versions?
when converting both, the LPCM track and the trueHD track to flac, did you have to apply delay for one or both to fit for a remux to .mkv with 23.9760239 fps? if needed for both, was the delay value needed different from both tracks, apart from that stuff that was different anyway due to different additional audio data at beginning/in between/end, which could somehow result from the demux process or such?

[madshi]

did you compare the original LPCM and trueHD tracks, or both in their converted to flac versions?
I compared the raw audio data (after correct channel remapping).

when converting both, the LPCM track and the trueHD track to flac, did you have to apply delay for one or both to fit for a remux to .mkv with 23.9760239 fps?
I haven't remuxed the video yet, so I can't say. However, since the LPCM track has 105ms more audio data in the beginning, the TrueHD decoded track will need 105ms more delay than the LPCM track. If the LPCM track is in sync without any manual delay, the TrueHD track will need exactly 105ms.

[calinb]

Let me start off by saying that I hate finding delays - I'm terrible at it and it takes me ages. <snip>
As mentioned here, MPC is a good tool for finding the delay values because you can quickly change them "on the fly." I remapped the MPC audio delay increase/decrease keys to a more convenient key pair for this purpose.

First, find the audio delay (+/-) necessary to obtain sync early in the file. Next find the delay required late in the file. If they are the same, it's easy--you only need to remux the file with this delay. If they are different, you must remux the file with this delay "offset" but also "stretch/shrink" the audio or adjust the framerate. I use the mkvmerge audio "stretch by" feature, though the tool tips advises the feature doesn't work with all audio formats. If your format doesn't work with mkvmerge, try using another tool like BeSweet to stretch/shrink the audio in the wav files before encoding. The stretch factor is the difference between the early delay value and the late delay value compared to the timestamp (seek position) time difference between them. For example:

1st sync check (ideally at the start of the file):

0:03:04 requires -200ms audio delay


2nd sync check (ideally, much later in the file):

2:02:04 requires +300ms audio delay


The sync checks are located 119 minutes, 0 seconds apart (7140 sec) so the audio must be stretched by 500ms (300ms - (-200ms) = 0.5 sec) over this period.

Enter -200 delay and 7140.5/7140 stretch values into mkvmerge (the GUI accepts fractions).

Of course, this only works if the audio/video "drift" is constant in the file.

[Thunderbolt8]

I'd rather say try to work on the fps settings and alter them a little from 25 or 23.976 fps instead of stretching the audio, because this alters the quality a little.

[calinb]

I'd rather say try to work on the fps settings and alter them a little from 25 or 23.976 fps instead of stretching the audio, because this alters the quality a little.Yes--for no quality loss, I think a timecodes.txt file could be applied to the video stream in mkvmerge:

# timecode format v1
assume 23.976

Just change the the framerate in the 2nd line above before muxing. A more sophisticated timecodes.txt file could probably be used to correct sync where the A/V drift is not constant.

[Thunderbolt8]

I just tried another way to come close to the delay for trueHD for HDDVD remuxes. I rebuilt the .evo of 'letters from iwo jima' with the video and the trueHD track and opened that rebuilt .evo with mediainfo. the length it gave me is 2h 20mn 37s 479ms, 157ms longer than the demuxed & converted trueHD -> flac track. tried the delay and it looked really very well, maybe the best result so far (well of course I wont see any difference between this and the 7ms different value I got from comparism with the converted ac3 track :P )
there are still some questions for me, though.

1. does the delay/length of that trueHD track remains exactly the same when rebuilding it into .evo (either together with the video track or alone, if theres a difference) as it is when just playing back the whole movie normally via its original files, or could already some changes occur in that rebuilding stage with evodemux?

2. mediainfo is not yet (fully) compatible for those new kind of audio tracks yet, for example it wont give you any info when just trying to open a demuxed trueHD or eac3 track with it. however, when I told it to open that rebuilt video & audio .evo file it gave me for this file with only 1 rebuilt audio stream:

General : blablabla\rebuiltVideoAudio.EVO
Format : MPEG-2PS
Length : 19 GiB for 2h 20mn 37s 479ms

Video #0 : VC-1 at 18 Mbps
Aspect : 1920 x 1080 (1.778)

Audio #0 : at 256 Kbps
Infos : , 44 KHz

Audio #1 : at 128 Kbps
Infos : , 44 KHz

the audio info displayed there is beyond reality, theres of course only the 1 trueHD track in it. but the length seems to be correct, when substracting the converted flac track length from it you get the value of 157ms which seems to be perfect for that movie!
maybe we could test if this works for other movies too, madshi (or others too, of course ;) ), if you could just do the same with the trueHD HDDVD movies you have.
if this should be a method how to find the exact delay for those tracks, you could use these accurate delay values and maybe find out how they originate and a way how they can be calculated and delay applied automatically then ;)


edit: well how to proceed in a similar case for blue rays then? mediainfo doesnt give any length on m2ts files, I also tried to use tsremux (even though I dont know if it will stay 100% original sync after that) to make a remux with video and lpcm audio only, but this new m2ts file didnt have any audio in it afterwards, or at least powerdvd and ffdshow both didnt find any.

[Thunderbolt8]

hm I just tried myself on the terminator blue ray, but I got 3 discontinuity errors while trying to demux the LPCM track (2nd audio track on the blue ray) out of the main m2ts file. I used xport 0.98, tried it both ways, one time without any video demuxing and 2nd together with video demuxing but I got the errors both times (both audio streams also had same length at the end). therefore I cant be sure if the converted LPCM -> flac stream is now 100% in sync with the remuxed video to .mkv. maybe im just imagining a tiny delay now, because I know about the discontinuity errors; if there really was a tiny error then it would be only very very small (I'd say <50ms).

any other idea what I could do to prevent this audio demux error?

[madshi]

hm I just tried myself on the terminator blue ray, but I got 3 discontinuity errors while trying to demux the LPCM track (2nd audio track on the blue ray) out of the main m2ts file. I used xport 0.98, tried it both ways, one time without any video demuxing and 2nd together with video demuxing but I got the errors both times (both audio streams also had same length at the end). therefore I cant be sure if the converted LPCM -> flac stream is now 100% in sync with the remuxed video to .mkv. maybe im just imagining a tiny delay now, because I know about the discontinuity errors; if there really was a tiny error then it would be only very very small (I'd say <50ms).

any other idea what I could do to prevent this audio demux error?
I'm having discontinuity errors with many Blu-Rays. I'm just ignoring them and didn't have any problems with that yet.

[Thunderbolt8]

do you mean you dont have any problems with delay then, as the audio track still syncs perfectly or not 'any problems' regarding that video and audio are still playable (but with some small delay needed then)?

[madshi]

do you mean you dont have any problems with delay then, as the audio track still syncs perfectly or not 'any problems' regarding that video and audio are still playable (but with some small delay needed then)?
I've not had any problems with delay running out of sync in the middle of a movie yet, if you mean that.

[Thunderbolt8]

yes, this and how many blue rays movies with LPCM tracks you've tested already did need some initial delay modification? did they all sync perfectly without any delay needed, despite all those discontinuity errors?

[madshi]

yes, this and how many blue rays movies with LPCM tracks you've tested already did need some initial delay modification? did they all sync perfectly without any delay needed, despite all those discontinuity errors?
I've already answered this question in post 1067.

[Thunderbolt8]

well not exactly this question but a quite similar one. must have forgotten to checkmark it on my big question paper. thanks!

[Rectal Prolapse]

The discontinuity errors on Blu-rays *might* be caused by a bad rip - there have been issues in the past where if you use Windows Explorer interface to decrypt blu-ray files with AnyDVD you may get corrupted files.

Best bet is to re-rip them using AnyDVD's tool directly and not use Windows Explorer at all.

[序列人]

c:\audio\eac3to audiodtshd
.thd audiodtshd.flac -24
TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
The following tasks are executed now. Please wait...
- remove Dialog Normalization information
- muxing TrueHD track into temp evo file
Decoding TrueHD track to raw. Please wait...
Checking raw data bitdepth. Please wait...
This TrueHD track contains more than 16 bit of information.
Encoding the raw file to FLAC. Please wait...
지정된 경로를 찾을 수 없습니다.

"flac" reported error code "1". A valid *.raw file was created sucessfully, thou
gh.

Why have i got an error message after I use EAC3 V1.11? Thanks in advanced.

[madshi]

Why have i got an error message after I use EAC3 V1.11? Thanks in advanced.
eac3to v1.11? That's a very old version. Try the latest!

Also: Are there really "?" chars in the eac3to output? Do you have a "Flac" folder in your eac3to folder and is there a "flac.exe" inside?

[DeepBeepMeep]

Great tool!

Is there a way to keep the dial norm when extracting an AC3 soundtrack encapsulated in a Dolby True HD track?

Since I don't plan to reencode the AC3 track I would be quite happy to keep the dial norm in this particular case.

I thought the -keepdialnorm option would work here but I lose the dialnorm anyway.

Thanks for your help

[LJstc]

hey madshi!

Thx for your work!

But i try to convert eac3 track from Mission Imp. 3 to ac3 track and also get the 0 Byte error!

Have Nero and codecs correctly installed!

Can you tell me which chain i can use in graph edit to test if it works there?

Which decoders and muxers i need to connect in graph edit?

thx!

[madshi]

I thought the -keepdialnorm option would work here but I lose the dialnorm anyway.
I think there's a bug in the eac3to output text. I think it does say that it removes dialnorm even though you used the "-keepdialnorm" switch, but it doesn't really remove dialnorm. Please try running the extracted AC3 file through eac3to again. If dialnorm is still in there, eac3to will say so.

Another thing to note is that Sony doesn't use dialnorm to begin with. So if it's a Sony Blu-Ray you're talking about, dialnorm was never there.

[madshi]

hey madshi!

Thx for your work!

But i try to convert eac3 track from Mission Imp. 3 to ac3 track and also get the 0 Byte error!

Have Nero and codecs correctly installed!

Can you tell me which chain i can use in graph edit to test if it works there?

Which decoders and muxers i need to connect in graph edit?
You need to rename GraphEdit.exe to Recode.exe. Then you can use the Nero Splitter (or something like that) and the Nero Audio Decoder 2.

Please note that the usual cause of the "0 byte" problem is that you didn't properly buy/license the HD DVD plugin.

[LJstc]

Thank you for your fast answer!

but i registered the HDDVD plugin with the Nero registrator!

i pressed the ADD button in Nero registrator and filled in my HDDVD Plugin serial! Under expire tab stands NONE at HDDVD_TEST feature.


and in graphedit i do have all filters Nero Filesource/Splitter, Nero Audio Decoder 2 properly installed.

i connected Nero File Source/Splitter ---> Nero Audio Decoder2 ---> Dump (test.ac3)

my source file is a demuxed DD+ file with EVODEMUX can that be a problem? because i dont know exactly the ending of the file? EAC3to writes me in dos prompt its a E-AC3 file! so the right extension is EAC3 right? or DDP ?

Recode.exe an setting up the graph as above dont work too i get 0 byte file.

maybe you can tell me some hints with this informations?

[madshi]

but i registered the HDDVD plugin with the Nero registrator!

i pressed the ADD button in Nero registrator and filled in my HDDVD Plugin serial! Under expire tab stands NONE at HDDVD_TEST feature.
Did you correctly buy the plugin?

[LJstc]

Lets say i got a working serial!

Now and its registered correctly in the Nero Register Tab

There Stands "Nero BlueRay/HD-DVD Video plugin" Serial Number and then Expiration NONE

I know i will buy i license this is first only for testing if it works!

[LJstc]

Hey! thanks!

It works now!

Plugin wasnt correctly registered!


Thx for hint!

[TheSof]

Probably answered somewhere before but I couldn't find it.

Superman Returns 1.57GB 16bit THD > FLAC, 16bit 1.39 GB.

V 4 Vendetta 1.27GB 16bitTHD > FLAC, 16bit 1GB

Harry Potter 1.64GB 16bit THD > FLAC 24bit 1.4GB

So I guess I'm asking why some 16bit THD files come out as 16bit FLAC, but others at 24bit when clearly they are really 16bit from looking at the file size?

[madshi]

Probably answered somewhere before but I couldn't find it.

Superman Returns 1.57GB 16bit THD > FLAC, 16bit 1.39 GB.

V 4 Vendetta 1.27GB 16bitTHD > FLAC, 16bit 1GB

Harry Potter 1.64GB 16bit THD > FLAC 24bit 1.4GB

So I guess I'm asking why some 16bit THD files come out as 16bit FLAC, but others at 24bit when clearly they are really 16bit from looking at the file size?
I've really no idea. Maybe the Harry Potter THD file is corrupt? Please try reripping the HD DVD with the latest version of DumpHD and then demux again with the latest version of EvoDemux. Does that fix the problem?

[TheSof]

I've really no idea. Maybe the Harry Potter THD file is corrupt? Please try reripping the HD DVD with the latest version of DumpHD and then demux again with the latest version of EvoDemux. Does that fix the problem?

That could be the reason, but I'm using the latest EvoDemux, and the evo, mkv and flac plays perfect. Ripping the disc again could be a problem as I have sold it on, and it was tough getting my hands on it as it is hard to source.

Indeed, apart from the fact that the flac is 24bit, the file size and runtime are fine. Strange.

[OVEBILL]

Here is some link for XXXXXXXXXXXXXXXXXXXXXXXXXX

[madshi]

That could be the reason, but I'm using the latest EvoDemux, and the evo, mkv and flac plays perfect. Ripping the disc again could be a problem as I have sold it on, and it was tough getting my hands on it as it is hard to source.

Indeed, apart from the fact that the flac is 24bit, the file size and runtime are fine. Strange.
Well, you could try creating a small test sample (10MB). It you can reproduce the problem with that small test sample, you can send it to me. Maybe I can find out the reason why it behaves strange.

[humax]

Hello
I have a problem mith eac3to.

I always geht this message :

http://img159.imageshack.us/img159/6666/soundxq4.th.jpg (http://img159.imageshack.us/my.php?image=soundxq4.jpg)

Is the Gui wrong an i shall use the .exe command file ??
Sonic is installed .
Whats wrong ?

Thx for Help

[Thunderbolt8]

you are missing the 'dump' filter, which comes along with graphedit

btw. im not really into all that programming stuff and such, but I have to say the commandline from eac3to is really easy to use. I'd say the GUI gives me more trouble somehow :P

[Rectal Prolapse]

Harry Potter 1.64GB 16bit THD > FLAC 24bit 1.4GB

I am sure a properly ripped version will NOT have this issue. This is another way of saying: If you had bought the disc, and you get the above result, it is because you ripped it using Windows Explorer and AnyDVD instead of AnyDVD's built-in tools...ripping the disc within Windows Explorer will give you corrupted video and/or audio - which you may not even see or hear if the glitch is small enough - but will cause eac3to to see garbage data that goes beyond the 16 bit boundary.

Harry Potter 4's soundtrack is truly 16 bit. Re-rip (from the original disc of course) and try again. :)

[humax]

GraphEdit build 011008 is installed - btw it was a zip file . Do i have to register the dump filter ?

[madshi]

GraphEdit build 011008 is installed - btw it was a zip file . Do i have to register the dump filter ?
Yep, you need to run "regsvr32 dump.ax".

[humax]

Well same problem ...
I ll reboot to see if it works then

[nautilus7]

@ madshi

Which program do you use to compare if 2 files are bit-to-bit identical?

[nautilus7]

I am trying to convert a dts file to mono wavs, but i get the following error. Does it have to do with thes free space on my hdd (only 4,79 GB - raw is 4,79 GB - 9,58GB sum)

C:\Tools>eac3to kung.dts kung.wavs
DTS, 5.1 channels, 1:39:46, 24 bits, 768kbit/s, 48khz
Decoding DTS track to raw. Please wait...
Find sync word: 7ffe8001
Find sync extension: 3f
Warning: alias sync word (0x7ffe8001) occurred in right alignment.
Warning: alias sync word (0x7ffe8001) occurred in right alignment.
Warning: alias sync word (0x7ffe8001) occurred in left alignment.
Warning: alias sync word (0x7ffe8001) occurred in right alignment.
Warning: alias sync word (0x7ffe8001) occurred in left alignment.
Warning: alias sync word (0x7ffe8001) occurred in left alignment.
Audio Decoder output was 24 bit this time.
Convert the multi channel raw file to mono WAV files. Please wait...
Writing to the mono WAV files failed.
Splitting the multi channel raw file failed. The multi channel raw file is valid, though.

[VempX]

OK
please tell me how to encode the 5.1 LPCM file which demuxed from BD to 5.1 FLAC~~
m(_ _)m

[calinb]

I have a .thd file produced with Evodemux.

Dolby TrueHD audio stream 1 found!
First PTS = 0000921B
Substream id = B1
TrueHD stream (up to 8 channels)
Sampling frequency = 48 kHz
2 ch. decoder
channel modifier = Stereo
6 ch. decoder
channel modifier = Stereo
channel arrangement = Main (Left,Right), Centre, LFE, Surrounds (Ls/Rs)
8 ch. decoder
channel modifier = Stereo
channel arrangement = Main (Left,Right), Centre, LFE, Surrounds (Ls/Rs)
Dynamic range control = -10.6371 dB .. 8.4282 dB

eac3to v1.22 writes out the wav files as 32KHz. Sure enough--when I play them, they are about 32/48 pitch reduced and 48/32 duration lengthened.

eac3to v1.16 correctly sets the sampling rate at 48KHz when operating on a .evo file containing only this track.

Please let me know if you need a small sample, madshi

[madshi]

Well same problem ...
I ll reboot to see if it works then
If it still doesn't work, try running regsvr32 again. It should show a little message box. What does the text say?

[madshi]

Which program do you use to compare if 2 files are bit-to-bit identical?
Personally, I'm using Beyond Compare (commercial) or the latest beta of this free hexeditor:

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

[madshi]

I am trying to convert a dts file to mono wavs, but i get the following error. Does it have to do with thes free space on my hdd (only 4,79 GB - raw is 4,79 GB - 9,58GB sum)

C:\Tools>eac3to kung.dts kung.wavs
DTS, 5.1 channels, 1:39:46, 24 bits, 768kbit/s, 48khz
Decoding DTS track to raw. Please wait...
Find sync word: 7ffe8001
Find sync extension: 3f
Warning: alias sync word (0x7ffe8001) occurred in right alignment.
Warning: alias sync word (0x7ffe8001) occurred in right alignment.
Warning: alias sync word (0x7ffe8001) occurred in left alignment.
Warning: alias sync word (0x7ffe8001) occurred in right alignment.
Warning: alias sync word (0x7ffe8001) occurred in left alignment.
Warning: alias sync word (0x7ffe8001) occurred in left alignment.
Audio Decoder output was 24 bit this time.
Convert the multi channel raw file to mono WAV files. Please wait...
Writing to the mono WAV files failed.
Splitting the multi channel raw file failed. The multi channel raw file is valid, though.
Yeah, it's quite probable that there's not enough space on your harddisk. Never saw mono WAV file creation fail for any other reason yet.

[madshi]

OK
please tell me how to encode the 5.1 LPCM file which demuxed from BD to 5.1 FLAC~~
m(_ _)m
(1) Demux the LPCM track by using xport (not TsRemux!).
(2) Run "eac3to source.pcm dst.flac".

Usually that's it. Sometimes eac3to cannot automatically detect which bitdepth the LPCM track has. In that case you have to specify the bitdepth by using the "-16" or "-24" parameters. So you'd do e.g. "eac3to source.pcm dst.flac -16".

[madshi]

I have a .thd file produced with Evodemux.

Dolby TrueHD audio stream 1 found!
First PTS = 0000921B
Substream id = B1
TrueHD stream (up to 8 channels)
Sampling frequency = 48 kHz
2 ch. decoder
channel modifier = Stereo
6 ch. decoder
channel modifier = Stereo
channel arrangement = Main (Left,Right), Centre, LFE, Surrounds (Ls/Rs)
8 ch. decoder
channel modifier = Stereo
channel arrangement = Main (Left,Right), Centre, LFE, Surrounds (Ls/Rs)
Dynamic range control = -10.6371 dB .. 8.4282 dB

eac3to v1.22 writes out the wav files as 32KHz. Sure enough--when I play them, they are about 32/48 pitch reduced and 48/32 duration lengthened.

eac3to v1.16 correctly sets the sampling rate at 48KHz when operating on a .evo file containing only this track.
That's a bug in v1.22 which was already fixed in v1.23. Simply download and use v1.23, please.

[nautilus7]

Yeah, it's quite probable that there's not enough space on your harddisk. Never saw mono WAV file creation fail for any other reason yet.Yeah, that was the problem. I deleted some stuff and it worked ok. Thanks!

[nautilus7]

I am trying to remove the DialNorm from an ac3 2.0 track.

Eac3to says that DialNorm is removed successfully (-27db), but the track looks exactly the same (before and after the "removal") in adobe audition.

The track is a director's commentary. Do these tracks usually have DialNorm or there is a bug in eac3to?

The hex editor you suggested me above, shows some minor differences between the two tracks, so i guess something was removed.

If you need a sample, please tell me how to cut a small piece from the track.

[madshi]

I am trying to remove the DiagNorm from an ac3 2.0 track.

Eac3to says that DiagNorm is removed successfully (-27db), but the track looks exactly the same (before and after the "removal") in adobe audition.

The track is a director's commentary. Do these tracks usually have DiagNorm or there is a bug in eac3to?

The hex editor you suggested me above, shows some minor differences between the two tracks, so i guess something was removed.

If you need a sample, please tell me how to cut a small piece from the track.
I don't know what adobe audition does. Maybe it ignores dialnorm? I've no idea. But if you see a difference in the hex editor, then obviously eac3to has done something. My best guess is that dialnorm removal worked just fine and adobe audition is not the right tool to see a difference. Try using GraphEdit + ffdshow + WavDest + WavWriter on a 10MB sample. Then compare the result in Audacity. That should show a difference.

[nautilus7]

I'm afraid i wasn't clear...

The files i loaded to audition ware wavs (of the ac3s) made by tranzcode.

To sum up, what i 've done was:
1. ac3 --> remove DialNorm with eac3to --> convert to wav via tranzcode --> load in audition
2. ac3 --> convert to wav via tranzcode --> load in audition

Both look the same. Does tranzocde remove DialNorm by default?

Sample: http://rapidshare.com/files/61882717/Track15.ac3.html
http://img227.imageshack.us/img227/4874/dntestsh6.th.png (http://img227.imageshack.us/my.php?image=dntestsh6.png)

[madshi]

I think tranzcode ignores dialnorm.

[nautilus7]

That could be a good explanation. Thanks!

Any update about implementing ac3 decoding to eac3to?

[madshi]

Any update about implementing ac3 decoding to eac3to?
http://forum.doom9.org/showthread.php?p=1051536#post1051536

[nautilus7]

I meant beyond this. But as i can understand, the answer is no.

[madshi]

Well, I've posted that comment just a good week ago and said in there that it would be "a while before I invest time into that". A while is more than just a week. I have a real job to do besides developing eac3to...

[nautilus7]

Sorry... I didn't want to hear like that. Take your time. Eac3to is great anyway.

[superx]

whoohoo finally got stuff working, anyway how can convert a DD+ to DTS 1.5 mbbits

what programs would I need for this and how would I do it, thanks in advance.

[madshi]

whoohoo finally got stuff working, anyway how can convert a DD+ to DTS 1.5 mbbits

what programs would I need for this and how would I do it, thanks in advance.
Check out the first post of this thread.

[Roscoe62]

Hey Madshi,

Just wanted to say a big Thank You for this awesome app. To me it started out as an app I probably wouldn't use, and it's turned into an app that I could NOT do without. I'm still trying to figure out what I did before I found it. :)

Anyway, thanks for spending the time developing this wonderful app and supporting it here.

Much appreciated!

[madshi]

@Roscoe62, thanks for the nice feedback... :)

[nmeli15]

Like @Rosco62 I feel the same way. GREAT work Madshi

[honai]

@madshi

When and why was the option removed to convert TrueHD tracks from within an EVO? Right now I have an EVO with only one TrueHD track, and when I want to convert it to WAVs eac3to tells me to extract the TrueHD first ... only to tell me then that it's remuxing the TrueHD into an EVO before converting it.

This doesn't make much sense to me. Am I missing something here?

EDIT:

Is it because eac3to can't remove Dial.Norm. on-the-fly?

[madshi]

Is it because eac3to can't remove Dial.Norm. on-the-fly?
Yep, that's the reason. eac3to isn't clever enough to remove the dialnorm inside of the EVO container. It can only remove dialnorm from demuxed TrueHD files. This may eventually change in the future. But right now you have to demux first.

[VempX]

(1) Demux the LPCM track by using xport (not TsRemux!).
(2) Run "eac3to source.pcm dst.flac".

Usually that's it. Sometimes eac3to cannot automatically detect which bitdepth the LPCM track has. In that case you have to specify the bitdepth by using the "-16" or "-24" parameters. So you'd do e.g. "eac3to source.pcm dst.flac -16".


Please tell me how to use xport~

I had demux the 5.1 LPCM from m2ts,but I get a .mpa file......

why?

[ACrowley]

Please tell me how to use xport~

I had demux the 5.1 LPCM from m2ts,but I get a .mpa file......

why?

Yes ?...correct
simply rename it to raw or pcm....its only the File eextension

[VempX]

Yes ?...correct
simply rename it to raw or pcm....its only the File eextension

What??:eek:
I just rename the mpa to pcm?
Orz

[madshi]

What??:eek:
I just rename the mpa to pcm?
Yes. Of course that only works if you demuxed the right track.

[sportcenter]

on a 1.5Mbps E-AC3 source, what would be the difference between a flack track and a DTS one....both @ 1.5Mbps?

[madshi]

on a 1.5Mbps E-AC3 source, what would be the difference between a flack track and a DTS one....both @ 1.5Mbps?
You can't tell FLAC which bitrate it should use. A lossless decoder *always* has a variable bitrate. Decoding an E-AC3 source will result in 24bit of audio data. If you compress that with FLAC the file will be much larger than 1.5Mbps. Of course you could reduce the bitdepth to 16bit. In that case the size of the FLAC and the DTS tracks should be roughly similar. Don't ask me which sounds better then, though.

[sportcenter]

You can't tell FLAC which bitrate it should use. A lossless decoder *always* has a variable bitrate. Decoding an E-AC3 source will result in 24bit of audio data. If you compress that with FLAC the file will be much larger than 1.5Mbps. Of course you could reduce the bitdepth to 16bit. In that case the size of the FLAC and the DTS tracks should be roughly similar. Don't ask me which sounds better then, though.

so will the FLAC @ 24bit sound better than the DTS @ 1.5Mbps...noticeably at least?

[madshi]

so will the FLAC @ 24bit sound better than the DTS @ 1.5Mbps...noticeably at least?
It should sound a bit better. But how much depends on your equipment (receiver, speakers etc) and also on your ears. Some people may not hear a difference. Maybe more than "some". I don't know.

[ACrowley]

on a 1.5Mbps E-AC3 source, what would be the difference between a flack track and a DTS one....both @ 1.5Mbps?


As is said in my Ears and on my Setup EAC3 to AC3 640 sounds very close to the Source.
And youve perfect Encoder Options like the latest certified Dolby Digital v7 Pro Encoder from Sony Vegas .
And SPDIF Output works perfect to a DD Receiver.

I pers encode to FLAC only from losslesss Sources like TrueHD/LPCM/DTS HD MAS to Back it up with less Discspace.
And AC3 640 Output sound OK from AC3 Filter or ffdshow in Combi with madFlac/ffdshow

SPDIF is not possible with FLAC so you must output FLAC as AC3 640 .
EAC3 to FLAC to AC3 makes no Sense for me
So i prefer nice Dolby Digital Encoding from EAC3.
Ofcourse EAC3 1.5Mbps to DTS 1.5 Mbps is a Option too...but imho AC3 640 reencode sounds more close compared with a Surcode DTS ,also because Surcode is not the best Encoder :)
I encode to dts only from lossless Sources and not from "pre" compressed EAC3

However...decide by yourself what sound best for you

[Thunderbolt8]

remux of terminator 2 DC studio canal with DTS-HD MA -> flac:

Opening file L0_mainMovie.EVO
PTM of first video frame = 0000AFC8
VC-1 video stream 0 found!
First PTS = 0000AFC8
DTS HD (DTS) audio stream 0 found!
First PTS = 0000AFC8

the rebuilt evo in evodemux gave me as duration: 2:33:25.229
the converted FLAC track length is 2h 33mn 25s 258ms.
the difference is ~30ms and I really cant say whether I need that -30ms of delay or not, its too hard to find out.

just to compare, the length of the ac3 track (converted from the dtshd) is 2h 33mn 25s 280ms.

[honai]

The .229 and 258 are not necessarily the same base, one could be frames of any other fraction. Also, there is probably no delay but simply a difference at the end of the track, so nothing to worry about.

[Thunderbolt8]

I got +1ms from muxing the .flac into .mka this time (usually this does not happen though), so the original flac file had afaik .228ms

[Coolpplse]

Hi Madshi,

I have Nero 7.5.7.0 Installed with the Blu-Ray/HD DVD Playback Plug-in Installed and registered and I am having trouble connecting the filters for converting Dolby TrueHD to FLAC

Based on your method

Nero File Source (Async) -> Nero Splitter -> Nero Audio Decoder 2 -> Dump

When I use the TrueHD file that is demuxed I can get to here

Nero File Source (Async) -> "Error Message"

When I use the TrueHD file rebuilt with the TrueHD Audio only I get to here

Nero File Source (Async) -> Nero Splitter -> "Error Message"

Using the latest eac3to v1.23
If I can get this to work I am going to try to make a step by step guide for everyone with pictures

[madshi]

Hi Madshi,

I have Nero 7.5.7.0 Installed with the Blu-Ray/HD DVD Playback Plug-in Installed and registered and I am having trouble connecting the filters for converting Dolby TrueHD to FLAC

Based on your method

Nero File Source (Async) -> Nero Splitter -> Nero Audio Decoder 2 -> Dump

When I use the TrueHD file that is demuxed I can get to here

Nero File Source (Async) -> "Error Message"

When I use the TrueHD file rebuilt with the TrueHD Audio only I get to here

Nero File Source (Async) -> Nero Splitter -> "Error Message"

Using the latest eac3to v1.23
If I can get this to work I am going to try to make a step by step guide for everyone with pictures
Why are you trying to do this manually in GraphEdit? eac3to should do all that for you. Anyway, if you want to use GraphEdit, you need to rename it to Recode.exe. Furthermore please note that pirated Nero keys are known to make problems.

[honai]

If I can get this to work I am going to try to make a step by step guide for everyone with pictures

I'm not sure what problem you'd be solving with that. As madshi pointed out, eac3to already does all that is needed. Going the manual route will most likely introduce problems between keyboard and chair.

[Thunderbolt8]

matrix remux with 16-bit trueHD -> flac

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

delay=0 or 110 ms

rebuilt the audio track only and loaded that evo in evodemux: length is 2:16:17.786
length of the flac: 2:16:17:680

so I guess the delay is either ~110ms or no delay at all. cant decide yet.

[Coolpplse]

Why are you trying to do this manually in GraphEdit? eac3to should do all that for you. Anyway, if you want to use GraphEdit, you need to rename it to Recode.exe. Furthermore please note that pirated Nero keys are known to make problems.

I dont know whether my Nero is correctly registered but it does look like this:

http://img87.imageshack.us/img87/5695/neroag8.th.jpg (http://img87.imageshack.us/my.php?image=neroag8.jpg)

The keys arent there as I erased it on the screenshot......

[honai]

delay=0 or 110 ms

Where do you get the 110ms from? It's 0ms.

[TruckChase!]

Has anyone been able to get this working with a demo key of Nero and a paid key for the HD-DVD plugin? I wanted to give this a try before I shell out the 99 bux for Nero... I figured that if I could get it working with just spending the 25 dollars for the plugin I could try out the results, but unfortunately I just get "Getting "Nero File Source Async" instance failed." when trying to extract ac3 frames from a TrueHD file. (Blood Diamond) Running Vista 32bit.

[madshi]

I dont know whether my Nero is correctly registered
I don't know, either. But I know that nobody reported any problems yet with a properly bought HD DVD plugin. So did you buy the plugin or did you not? If you didn't please don't ask in this thread for help.

[madshi]

Has anyone been able to get this working with a demo key of Nero and a paid key for the HD-DVD plugin? I wanted to give this a try before I shell out the 99 bux for Nero...
Nice idea. But I don't know if that works.

unfortunately I just get "Getting "Nero File Source Async" instance failed." when trying to extract ac3 frames from a TrueHD file. (Blood Diamond) Running Vista 32bit.
Just extracting the AC3 frames from a Blu-Ray TrueHD stream shouldn't result in this error because eac3to shouldn't even try to get a "Nero File Source Async" instance in that case. Can I please see the command line you called eac3to with plus the full eac3to output?

P.S: Or is it a HD DVD TrueHD track? In that case there are no AC3 frames in the stream and your only chance is to reencode the audio track.

[Darth Pinous]

I dont know whether my Nero is correctly registered but it does look like this:
If the plugin appears on that screen, I think it is correctly registered. Wrong serial shouldn't appear on that screen. What's strange is that it appears as "Blu-ray/HD DVD Playback Plug-in". Mine appears as "Blu-ray/HD DVD Video Plug-in" and it appears like that on Nero site, too...

delay=0 or 110 ms

rebuilt the audio track only and loaded that evo in evodemux: length is 2:16:17.786
length of the flac: 2:16:17:680

so I guess the delay is either ~110ms or no delay at all. cant decide yet.

Audio length doesn't matter. I've seen perfectly synced audio tracks that were a second shorter than the video. What matters is the difference between the first PTS of the video and the first PTS of the audio. In your case :

VC-1 video stream 0 found!
First PTS = 00000DC7
Dolby TrueHD audio stream 1 found!
First PTS = 00000DC7

Both have same first PTS, so no audio delay.

P.S: Or is it a HD DVD TrueHD track? In that case there are no AC3 frames in the stream and your only chance is to reencode the audio track.

Blood Diamond should be PCM only on BluRay and TrueHD only on HDDVD, I think your guess is good. Only way is to reencode.

By the way, thanks for your tool, madshi. I've just noticed that even DTS-HD MA can be treated with it, waow !

[Thunderbolt8]

Where do you get the 110ms from? It's 0ms.
approx difference of length of the flac track and the trueHD track in the .evo

[madshi]

By the way, thanks for your tool, madshi. I've just noticed that even DTS-HD MA can be treated with it, waow !
Yep! :) But only with the help of the Sonic Audio Decoder 4.3.

[Thunderbolt8]

Audio length doesn't matter. I've seen perfectly synced audio tracks that were a second shorter than the video. What matters is the difference between the first PTS of the video and the first PTS of the audio. In your case :

VC-1 video stream 0 found!
First PTS = 00000DC7
Dolby TrueHD audio stream 1 found!
First PTS = 00000DC7

Both have same first PTS, so no audio delay.
well thats the situation for the audio inside the .evo. but apparently this changes slightly when the audio is demuxed in most cases. check the info on page 27 for letter from iwo jima. both, audio (truehd) and video were in sync in the evo, had same beginning PTS, but after the conversion the flac track needed ~120ms of delay.

[Thunderbolt8]

some questions:
1. when opening a rebuilt .evo of a movie (1xaudio+video) with evodemux, to which stream does the duration then refer? to the video stream? to the audio stream? to the one whos longest of both? (this is mostly void for movies where both stream have the same length, but this does not account for all movies)
2. when demuxing or rebuilding a video only .evo of a movie could there also be length alternation so that for example the original duration is changed a bit like when demuxing a trueHD audio track, which might need a delay then afterwards to sync with the video? or will the video track always stay the same, regardless of rebuilding or demuxing?
3. can the audio delay for trueHD tracks (or also dts-hd) also be altered by rebuilding the .evo with only this track in it? or does this only happen when demuxing the audio and via rebuilding the audio remains unaltered, compared to its original state regarding delay?

[madshi]

1. when opening a rebuilt .evo of a movie (1xaudio+video) with evodemux, to which stream does the duration then refer?
I think this duration is taken from an "info" block in the EVO structure. I don't think it's directly related to video/audio timestamps.

2. when demuxing or rebuilding a video only .evo of a movie could there also be length alternation so that for example the original duration is changed a bit like when demuxing a trueHD audio track, which might need a delay then afterwards to sync with the video? or will the video track always stay the same, regardless of rebuilding or demuxing?
3. can the audio delay for trueHD tracks (or also dts-hd) also be altered by rebuilding the .evo with only this track in it? or does this only happen when demuxing the audio and via rebuilding the audio remains unaltered, compared to its original state regarding delay?
Rebuilding the EVO shouldn't change anything. But demuxing might change things. Remuxing might also change things. But I'm not sure about the exact effects. It might also depends on which DirectShow filters are used... :(

[Thunderbolt8]

hm if rebuilding doesnt change things then it might work to rebuilt an audio .evo only and take the length from there and compare it with the length of the demuxed&converted flac file in the end. maybe this will always give the correct delay.

[Thunderbolt8]

I put all 3 tracks for matrix (the rebuilt evo with video+audio, video only, audi only) into evodemux and got the same length for all of them. since first PTS of video and audio was exactly the same (00000DC7) I guess we can assume that both tracks beging at the same time in the original movie, right? so whats needed to be done afterwards (after remuxing & converting) is establishing this again: the flac track must have the same duration as the video, because according to evo demux both have the same length and start at the same point. so could a way to determine the correct delay be look at the remuxed video with corrected timestamps with mkvmerge and the converted audio, and then try to have the audio delayed so that it will fit to the video duration? you said that remuxing might change things so I guess it makes most sense to look at both tracks, video and audio, when as much re- and demuxing has be done as possible already, before the final merge and not when both tracks are still in the .evo where the world is fine?

2:16:17.786 = duration of rebuilt audio&video, rebuilt audio, rebuilt video
VC-1 video stream 0 found!
First PTS = 00000DC7
Dolby TrueHD audio stream 0 found!
First PTS = 00000DC7

2:16:17:680 = converted flac track
2:16:17:753 = remuxed video .evo with -> gdsmux -> mkvmerge with 23.9760239 fps

so 753-680 = 73 <--- the correct delay needed for the audio?

[madshi]

I've no idea if that works. I can very well imagine that some discs have audio which is simply shorter or longer than the video - without having any effect on the delay.

[TruckChase!]

Nice idea. But I don't know if that works.


Just extracting the AC3 frames from a Blu-Ray TrueHD stream shouldn't result in this error because eac3to shouldn't even try to get a "Nero File Source Async" instance in that case. Can I please see the command line you called eac3to with plus the full eac3to output?

P.S: Or is it a HD DVD TrueHD track? In that case there are no AC3 frames in the stream and your only chance is to reencode the audio track.

Sorry, my bad. It's HDDVD TrueHD. I was confused. Here are some other attempts:

D:\encoding\video\eac3to>eac3to.exe "e:\video\work\HDDVD\GOOD_SHEPHERD\THEGOODSH
EPHERD\HVDVD_TS\WORK\FEATURE_1_MERGED.DD+.stream.00.ec3" d:\temp\try.ac3 -640
E-AC3, 5.1 channels, 2:47:30, 1536kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Decoding eac3 file to raw. Please wait...
Getting "Nero File Source / Splitter" and "DTS/AC3/DD+ Source" instances failed.

D:\encoding\video\eac3to>eac3to.exe "e:\video\work\HDDVD\work\BLOOD_DIAMOND_HDDV
D\feature_MERGED.DD+.stream.00.ec3" d:\pleasework.ac3 -640
E-AC3, 5.1 channels, 2:23:22, 640kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Decoding eac3 file to raw. Please wait...
Getting "Nero File Source / Splitter" and "DTS/AC3/DD+ Source" instances failed.

:confused:

I purchased the Sonic decoders first (before I fully read the cons), and those work. Doh! Perhaps I need to get the Nero 7.8.x trial or something. If anyone has any ideas I'd appreciate it.. I'd really like to evaluate the results before I spend $100. I'm very strapped for cash right now.

[honai]

You'll need the Nero HD-DVD/Blu-ray plugin, and there is no trial "or something". Not sure if you can just buy the plugin w/o the Burning ROM package.

[TruckChase!]

You'll need the Nero HD-DVD/Blu-ray plugin, and there is no trial "or something". Not sure if you can just buy the plugin w/o the Burning ROM package.

You can, and I did :D

[Thunderbolt8]

I've no idea if that works. I can very well imagine that some discs have audio which is simply shorter or longer than the video - without having any effect on the delay.
hm yes. but you should still have the information at which point both tracks begin, via the first PTS comparism and of the overall duration, when you rebuilt both tracks seperately in .evos and their lengths should differ. with these 2 options it should be possible to match the correct delay then at the muxing stage. for example, when knowing the audio is 150ms shorter at all, but later at the muxing stage its suddenly 200ms shorter you know you will have to add 50ms for delay to establish the old difference again.
this should work, unless the duration info given is not the actual info of each track, but some number written inside the .evo, which is always the same. this would be a problem then.

another thing, I just tried to test the same with the oldboy HD DVD (eac3 5.1), but after rebuilding audio and video seperately in evos the duration evodemux showed to me it only that of the 1st of the two original .evo files for both, audio and video each. the duration of a rebuilt evo including video AND audio though is fine. the file sizes are fine too, big enough to see that the complete rebuilt process actually went fine. its just the numbers are displayed wrong by evodemux. could this be possible, because I had to rename the 2 evo files, because otherwise it wouldnt have continued automatically with the 2nd evo, and by that somehow connection to other files got lost, which would be needed to have evodemux display that info correctly?

[honai]

For questions regarding EVOdemux please use the EVOdemux thread.

(And yes, there are still bugs in EVOdemux.)

[Thunderbolt8]

For questions regarding EVOdemux please use the EVOdemux thread.

(And yes, there are still bugs in EVOdemux.)
nvm, Ive just redone it with the audio and the duration is displayed correctly now.

[Coolpplse]

If the plugin appears on that screen, I think it is correctly registered. Wrong serial shouldn't appear on that screen. What's strange is that it appears as "Blu-ray/HD DVD Playback Plug-in". Mine appears as "Blu-ray/HD DVD Video Plug-in" and it appears like that on Nero site, too...


Is it also referred to as "Blu-ray/HD DVD Video Plugin in Nero 7? Cause all versions Nero 8 and above seem to be using that name.......

and yes I did buy the plugin....

[woah!]

hm yes. but you should still have the information at which point both tracks begin, via the first PTS comparism and of the overall duration, when you rebuilt both tracks seperately in .evos and their lengths should differ. with these 2 options it should be possible to match the correct delay then at the muxing stage. for example, when knowing the audio is 150ms shorter at all, but later at the muxing stage its suddenly 200ms shorter you know you will have to add 50ms for delay to establish the old difference again.
this should work, unless the duration info given is not the actual info of each track, but some number written inside the .evo, which is always the same. this would be a problem then.

another thing, I just tried to test the same with the oldboy HD DVD (eac3 5.1), but after rebuilding audio and video seperately in evos the duration evodemux showed to me it only that of the 1st of the two original .evo files for both, audio and video each. the duration of a rebuilt evo including video AND audio though is fine. the file sizes are fine too, big enough to see that the complete rebuilt process actually went fine. its just the numbers are displayed wrong by evodemux. could this be possible, because I had to rename the 2 evo files, because otherwise it wouldnt have continued automatically with the 2nd evo, and by that somehow connection to other files got lost, which would be needed to have evodemux display that info correctly?

this could be a evo demux bug/error so it probably has nothing to do with eacto at all. you are talking like evo demux is perfect but it isnt yet... good/great yes but not perfect..

[woah!]

i just remembered that there is a 73ms delay on the 2nd PEVOB if you check.

VC-1 video stream 0 found!
First PTS = 15D0DB4F

Dolby Digital Plus audio stream 0 found!
First PTS = 15D0C187

video = 4066.7685 secs
audio = 4066.6951 secs

delay = 0.0734 ms which is 73ms. i have argued this with a few people about alot of movies having delays on the 2nd EVO but gave up.

also the TrueHD track is 106ms delayed...

[madshi]

Sorry, my bad. It's HDDVD TrueHD. I was confused. Here are some other attempts:

D:\encoding\video\eac3to>eac3to.exe "e:\video\work\HDDVD\GOOD_SHEPHERD\THEGOODSH
EPHERD\HVDVD_TS\WORK\FEATURE_1_MERGED.DD+.stream.00.ec3" d:\temp\try.ac3 -640
E-AC3, 5.1 channels, 2:47:30, 1536kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Decoding eac3 file to raw. Please wait...
Getting "Nero File Source / Splitter" and "DTS/AC3/DD+ Source" instances failed.

D:\encoding\video\eac3to>eac3to.exe "e:\video\work\HDDVD\work\BLOOD_DIAMOND_HDDV
D\feature_MERGED.DD+.stream.00.ec3" d:\pleasework.ac3 -640
E-AC3, 5.1 channels, 2:23:22, 640kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Decoding eac3 file to raw. Please wait...
Getting "Nero File Source / Splitter" and "DTS/AC3/DD+ Source" instances failed.

:confused:

I purchased the Sonic decoders first (before I fully read the cons), and those work. Doh! Perhaps I need to get the Nero 7.8.x trial or something. If anyone has any ideas I'd appreciate it.. I'd really like to evaluate the results before I spend $100. I'm very strapped for cash right now.
Sonic is no good for TrueHD decoding (see first post of this thread). You do need at least Nero 7 Essentials plus the HD DVD plugin.

You can get Nero Essentials 7 very cheap on Ebay. So it's not $100. It's much much lower than that.

[madshi]

You can, and I did :D
But the plugin doesn't work on its own. It needs at least Nero Essentials 7 to be installed, as far as I can say.

[madshi]

this should work, unless the duration info given is not the actual info of each track, but some number written inside the .evo, which is always the same.
As I said before, I think that the duration info is a number taken from an info block in the EVO file which has nothing to do with the audio/video data.

[madshi]

Is it also referred to as "Blu-ray/HD DVD Video Plugin in Nero 7? Cause all versions Nero 8 and above seem to be using that name.......
Are you using Nero 8? That won't work. Nero 8 forcefully blocks Nero's Audio Decoder from being used outside of Nero ShowTime. You need to install Nero 7.

[madshi]

i just remembered that there is a 73ms delay on the 2nd PEVOB if you check.
Timecodes/delays of the 2nd EVO don't have any meaning for overall audio delay of the whole movie. You need to think of the two EVO files as being one. They just happen to be split somewhere. The way they are split often results in timecode differences in the 2nd EVO file. But that doesn't really help us to find the delay needed for the whole movie.

[Thunderbolt8]

As I said before, I think that the duration info is a number taken from an info block in the EVO file which has nothing to do with the audio/video data.is there any way to determine whether that is always the case?
when I open these rebuilt evos with mediainfo the delay is mostly slightly different compared to what evodemux tells me. so could it be possible that mediainfo maybe reads the actual length of the tracks (at least when it 'seems' to display the info correctly, sometimes I just get durations like 3 hours something for a rebuild video evo for whatever reason. in this case the info is not reliable of course).

[TruckChase!]

But the plugin doesn't work on its own. It needs at least Nero Essentials 7 to be installed, as far as I can say.

Ahh, I didn't think of trying a cheaper, older version. I'll see if I can locate and purchase a copy of 7.8.5 (I'm running Vista) and give it a try tonight. At least then people wouldn't have to shell out a $125 to use the decoder. :D

Edit: It's ALIVE! It's ALIVE! (evil laugh) Looks like Nero 7.8.5 is the way to go. Fortunately it can be had for pretty cheap if one looks on Froogle. All seems well now, thanks!

[XolocoTuxmaster]

First of all, thanks madshi for your program, is awesome, I never thought deconding HD tracks would be "so simple" nor we'd have it so early. I have read all your comments in these 59 pages, but I still have some doubts:

As I understand, except for DTS HD-HR and 7.1+Dialnorm TrueHD, this program allow us to extract all HD tracks information 100% lossless, right?

I mean, if I have a 24 bit 5.1 TrueHD from a Sony movie, writing this

eac3to src.thd dst.wav

I'd get the bit per bit original uncompress track packed in a wav file, with no lose of information nor unsync or whatever? So I wouldn't have to apply delay, raise volume or something, right?

I also wanted to know in which is the lower Scenearist version that contains 4.3

[madshi]

As I understand, except for DTS HD-HR and 7.1+Dialnorm TrueHD, this program allow us to extract all HD tracks information 100% lossless, right?
That's right. You can also decode DTS-HD-HR, btw. You just cannot transcode it to anything else without either losing audio quality or increasing file size.

I mean, if I have a 24 bit 5.1 TrueHD from a Sony movie, writing this

eac3to src.thd dst.wav

I'd get the bit per bit original uncompress track packed in a wav file, with no lose of information nor unsync or whatever? So I wouldn't have to apply delay, raise volume or something, right?
Correct - with one exception: You may have to apply a delay. I've not figured out yet how to automatically apply the correct delay programmatically. However, applying a delay just costs you time. It doesn't reduce audio quality in any way.

[Coolpplse]

Are you using Nero 8? That won't work. Nero 8 forcefully blocks Nero's Audio Decoder from being used outside of Nero ShowTime. You need to install Nero 7.

I was using Nero 7.5.7.0, I got so frustrated with this thing I called Nero's Tech Support and told them about the problem, apparently they gave me the wrong key so they gave me a new one and told me to upgrade to 7.10.1.0. Now it displays the Blu-Ray/HD DVD Video Plug-in and eac3to works!!!!!!!!!!

:thanks: for the help Madshi and everyone else in this thread :)

[XolocoTuxmaster]

OK thanks. Is there a way to know what delay I have to apply, like reading the .thd header or something? If so, what program can I use?

[Chumbo]

@madshi,
Thank you for this utility. :) If I may make a "wish list" suggestion for future versions please:

add an option to use a different "working" location. For example, when it builds the evo container right now, it uses the source location. It would be nice to have it use a specific alternate location to help speed things up, for example, pointing to another physical drive. :)

Thank you.

[Thunderbolt8]

OK thanks. Is there a way to know what delay I have to apply, like reading the .thd header or something? If so, what program can I use?
unfortunately not :/ (yet)

[XolocoTuxmaster]

OK:thanks:

[Chumbo]

OK thanks. Is there a way to know what delay I have to apply, like reading the .thd header or something? If so, what program can I use?
If your source is an EVO, then EVODemux will tell you if the audio is skewed and by how many milliseconds.

If your source is something else, then use a small clip to determine the delay.

[madshi]

If I may make a "wish list" suggestion for future versions please:

add an option to use a different "working" location. For example, when it builds the evo container right now, it uses the source location. It would be nice to have it use a specific alternate location to help speed things up, for example, pointing to another physical drive. :)
Your wish will be moot soon. Don't ask me why, I'm not telling... :devil:

[madshi]

If your source is an EVO, then EVODemux will tell you if the audio is skewed and by how many milliseconds.
That was my original guess, too, but for whatever reasons that doesn't seem to work properly. Still haven't found out why yet.

[Chumbo]

Your wish will be moot soon. Don't ask me why, I'm not telling... :devil:
Don't you mean "your wish is my..." :p I promise not to ask. ;)

[madshi]

Don't you mean "your wish is my..." :p
No, I mean you will stop seeing a need for that wish soon... ;)

[TruckChase!]

So, i should be able to use eac3to without the nero filters ?

Because it does not work on my system.

add -sonic to the end of the command line.

[Thunderbolt8]

That was my original guess, too, but for whatever reasons that doesn't seem to work properly. Still haven't found out why yet.
maybe its just a bug in evodemux? maybe pelican could update and bugfix evodemux further

[Chumbo]

No, I mean you will stop seeing a need for that wish soon... ;)
I can't wait to see what you have up your sleeve you little devil. :D

[Chumbo]

maybe its just a bug in evodemux? maybe pelican could update and bugfix evodemux further
Actually the numbers from evodemux are quite close. I use MPC on small clips to get the lip sync just right and that number is usually very close, within 30-60ms. At least that's been the case so far.

[idbirch2]

I've got a BluRay with a really weird DTS audio format. When I try and run eac3to on it I get:

DTS Master Audio, 3/1 channels, 24 bits, 48khz
This channel format is currently not supported.

Is there any pre-processing I can do to get a compatible stream or is there anything I can provide/upload which will help add support for this audio type?

[madshi]

I've got a BluRay with a really weird DTS audio format.
Can you please make a 5MB sample available for me? Thanks.

[idbirch2]

Sure can, can you just point me in the right direction for a tool that will let me cut 5MB of the .dts file? And do you want the first 5MB or a clip from the middle?

[madshi]

Sure can, can you just point me in the right direction for a tool that will let me cut 5MB of the .dts file? And do you want the first 5MB or a clip from the middle?
I'm using this free hexeditor for such tasks:

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

The first 5MB, if you please...

[idbirch2]

Thanks for the tip, here's the first 5000KB of the file:

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

It was demuxed using TSRemux if that is relevant.

[Thunderbolt8]

I got a eac3to error for the FLAC conversion of The Graduate's (Studio Canal) 2nd and 3rd DTS-HD track. for the first track though everything seems to be fine, the track was converted normally. but right after having pressed enter for the other 2 tracks eac3to & command line just crashed and gave me a windows error message (which I couldnt copy&paste). I've sent you 3x5MB samples of all 3 streams on that HD DVD, the errors happens for streams 01 and 02.

evodemux info:

Opening file L0_mainMovie.EVO
Reading...
File size: 11388 Mbytes.
VOB number 1 contains 1 video and 3 audio streams.
PTM of first video frame = 0000AFC8
PTM of last video frame = 164ED025
Duration = 1:09:17.987
VC-1 video stream 0 found!
First PTS = 0000AFC8
Substream id = 55
Profile = Advanced
Level = 3
Chroma Format = 4:2:0
Size = 1920x1080
Display size = 1920x1080
Aspect ratio = 1:1 (square samples)
Frame Rate = 23,976 (24000/1001)
DTS HD (DTS) audio stream 0 found!
First PTS = 0000AFC8
Substream id = 88
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 2011
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = 1536 kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
DTS HD (DTS) audio stream 1 found!
First PTS = 0000AFC8
Substream id = 89
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
DTS HD (DTS) audio stream 2 found!
First PTS = 0000AFC8
Substream id = 8A
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
.
Opening file L1_mainMovie.EVO
Reading...
File size: 6039 Mbytes.
VOB number 1 contains 1 video and 3 audio streams.
PTM of first video frame = 164ED025
PTM of last video frame = 2219CE14
Duration = 0:36:38.363
VC-1 video stream 0 found!
First PTS = 164ED025 (+4157987ms)
Substream id = 55
Profile = Advanced
Level = 3
Chroma Format = 4:2:0
Size = 1920x1080
Display size = 1920x1080
Aspect ratio = 1:1 (square samples)
Frame Rate = 23,976 (24000/1001)
DTS HD (DTS) audio stream 0 found!
First PTS = 164E3188 (+4157536ms)
Substream id = 88
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 2011
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = 1536 kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
DTS HD (DTS) audio stream 1 found!
First PTS = 164E4BC8 (+4157610ms)
Substream id = 89
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
DTS HD (DTS) audio stream 2 found!
First PTS = 164E5AC8 (+4157653ms)
Substream id = 8A
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
Done.

5401 counted frames (0:03:45.267) in video stream 0.
152400 calculated frames in video stream 0.

Reading HVA00001.VTI
8 VTS_EVOB_ATRs found.
13 VTS_EVOBIs found.
Done.

http://www.megaupload.com/de/?d=E2VW6NH3

[The_Keymaker]

Hello forum members,

I have released a new version of EAC3toGUI (v1.32) for your use.

Link: http://www.sendspace.com/file/gtb38a

Changes to the new version include:

1. - Added -down6 option
2. - Added Audio Delay tab and option. You can now specifiy
audio delay/advance in ms
3. - Added -sonic filter option

Remember, as with previous version of EAC3toGUI, you should place the program in same directory as eac3to.exe. If you decide to place EAC3toGUI in a different directory, you should specify the location of eac3to.exe using the settings menu option.

Please report any bugs and feel free to post comments.

@madshi, please upload the updated program to your link on page 1.

Regards,
The_Keymaker

[ACrowley]

Hello forum members,

I have released a new version of EAC3toGUI (v1.32) for your use.

Link: http://www.sendspace.com/file/gtb38a

Changes to the new version include:

1. - Added -down6 option
2. - Added Audio Delay tab and option. You can now specifiy
audio delay/advance in ms
3. - Added -sonic filter option

Remember, as with previous version of EAC3toGUI, you should place the program in same directory as eac3to.exe. If you decide to place EAC3toGUI in a different directory, you should specify the location of eac3to.exe using the settings menu option.

Please report any bugs and feel free to post comments.

@madshi, please upload the updated program to your link on page 1.

Regards,
The_Keymaker

THX

Important :

- add Input Extension .thd (demuxed TrueHD) and .dtshd /.dts for dtshd and dts Input (eac3 with Sonic now decodes full DTSHD and standard dts Audio)

.flac ( to set a Delay for FLAC Input)

Because dts and dtshd decoding is most important new Feature. Also eac3 now supports demuxed Truehd (.thd)

Andy maybe you can add the trueHD/dts Core extraction. Should be enough to add the Input Output Extensions to remove Dialnorm from .ac3 and core extraction
from .dts and .thd

[The_Keymaker]

@Acrowley

Please try v1.36, available here..: http://www.sendspace.com/file/1zrqp8
v1.36 changelog:

- Added *.dts, *.dtshd, *.flac and *.thd input extensions.

Regards,
keymaker

[Thunderbolt8]

I got a eac3to error for the FLAC conversion of The Graduate's (Studio Canal) 2nd and 3rd DTS-HD track. for the first track though everything seems to be fine, the track was converted normally. but right after having pressed enter for the other 2 tracks eac3to & command line just crashed and gave me a windows error message (which I couldnt copy&paste). I've sent you 3x5MB samples of all 3 streams on that HD DVD, the errors happens for streams 01 and 02.

evodemux info:

Opening file L0_mainMovie.EVO
Reading...
File size: 11388 Mbytes.
VOB number 1 contains 1 video and 3 audio streams.
PTM of first video frame = 0000AFC8
PTM of last video frame = 164ED025
Duration = 1:09:17.987
VC-1 video stream 0 found!
First PTS = 0000AFC8
Substream id = 55
Profile = Advanced
Level = 3
Chroma Format = 4:2:0
Size = 1920x1080
Display size = 1920x1080
Aspect ratio = 1:1 (square samples)
Frame Rate = 23,976 (24000/1001)
DTS HD (DTS) audio stream 0 found!
First PTS = 0000AFC8
Substream id = 88
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 2011
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = 1536 kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
DTS HD (DTS) audio stream 1 found!
First PTS = 0000AFC8
Substream id = 89
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
DTS HD (DTS) audio stream 2 found!
First PTS = 0000AFC8
Substream id = 8A
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
.
Opening file L1_mainMovie.EVO
Reading...
File size: 6039 Mbytes.
VOB number 1 contains 1 video and 3 audio streams.
PTM of first video frame = 164ED025
PTM of last video frame = 2219CE14
Duration = 0:36:38.363
VC-1 video stream 0 found!
First PTS = 164ED025 (+4157987ms)
Substream id = 55
Profile = Advanced
Level = 3
Chroma Format = 4:2:0
Size = 1920x1080
Display size = 1920x1080
Aspect ratio = 1:1 (square samples)
Frame Rate = 23,976 (24000/1001)
DTS HD (DTS) audio stream 0 found!
First PTS = 164E3188 (+4157536ms)
Substream id = 88
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 2011
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = 1536 kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
DTS HD (DTS) audio stream 1 found!
First PTS = 164E4BC8 (+4157610ms)
Substream id = 89
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
DTS HD (DTS) audio stream 2 found!
First PTS = 164E5AC8 (+4157653ms)
Substream id = 8A
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
PCM core samples = 32
PCM sample blocks = 16
Primary frame byte size = 595
Channel arrangement = L + R (stereo)
Sampling frequency = 48 kHz
Transmission bitrate = open kbit/s
LFE channel = not present
Encoder software rev. = 7, Current
Source PCM resolution = 16 bits
Done.

5401 counted frames (0:03:45.267) in video stream 0.
152400 calculated frames in video stream 0.

Reading HVA00001.VTI
8 VTS_EVOB_ATRs found.
13 VTS_EVOBIs found.
Done.

http://www.megaupload.com/de/?d=E2VW6NH3
could the error maybe be caused, because those 2 tracks are only normal dts tracks? at least mediainfo tells so. while evodemux names them as DTS-HD on the main page -.-

[Chumbo]

@Acrowley

Please try v1.36, available here..: http://www.sendspace.com/file/1zrqp8
v1.36 changelog:

- Added *.dts, *.dtshd, *.flac and *.thd input extensions.

Regards,
keymaker
Thank you for your continued efforts to improve this program. It's much appreciated.

I wanted to report one bug I found:
- the Source file and Destination file text boxes do not scroll horizontally. When the path exceeds the physical length of the box, the text box doesn't scroll. I think it's just a control setting, so it should be realy easy to fix. :) Thanks.

@madshi,
I forgot one other "wish" item in my earlier post.
- If possible to add an option, -keep, or something that would allow us to keep the working files created to achieve an end result. Right now, those files get deleted when the process is done. Thanks again.

[ACrowley]

@Acrowley

Please try v1.36, available here..: http://www.sendspace.com/file/1zrqp8
v1.36 changelog:

- Added *.dts, *.dtshd, *.flac and *.thd input extensions.

Regards,
keymaker

THX

Yes,with these new Extensions you can use all Functions from latest eac3to :)

But:
To set a Delay you must set the Value for positive Delay without + (correct= 100ms , not +100ms)

Your Gui 1.36 use a + :)

[Thunderbolt8]

300 HD DVD with TrueHD -> FLAC

PTM of first video frame = 6E191637
PTM of last video frame = 800037B3
Duration = 0:55:37.334
VC-1 video stream 0 found!
First PTS = 2E191637 (+35791394ms)
Dolby TrueHD audio stream 1 found!
First PTS = 2E191637 (+35791394ms)

did the mathematical way again, according to the PTS both streams started at the same time in the original movie. compared length of remuxed video and FLAC track with mediainfo, the difference was 116ms. applied that delay, looks good. 116ms is also quite close to the 105ms you calculated for the blueray, which the trueHD track might need there, in case the LPCM track is in sync with no delay at all.

[The_Keymaker]

@Acrowley,

When setting the delay, it is not necessary to input a "+" sign. The EAC3toGUI program will automatically assign and precede the amount specified by a "+" in the command line for values greater than zero (0).

If this is incorrect, please give a clear example of how the delay parameter (both for delay and advance) is specified on the command line and I will correct the EAC3toGUI program.

For instance, all command line options are preceded by "-". So is +100ms implemented as -100ms on the command line? and is -100ms implemented as --100ms?

Please confirm.

Regards,
The_Keymaker

[ACrowley]

@Acrowley,

When setting the delay, it is not necessary to input a "+" sign. The EAC3toGUI program will automatically assign and precede the amount specified by a "+" in the command line for values greater than zero (0).

If this is incorrect, please give a clear example of how the delay parameter (both for delay and advance) is specified on the command line and I will correct the EAC3toGUI program.

For instance, all command line options are preceded by "-". So is +100ms implemented as -100ms on the command line? and is -100ms implemented as --100ms?

Please confirm.

Regards,
The_Keymaker

Positive Delay :

Incorrect :
eac3to xxx.flac xxx1.flac +1000ms

Correct :
eac3to xxx.flac xxx1.flac 1000ms

Youre using +1000ms what is wrong

Negative Delay :
eac3to xxx.flac xxx1.flac -1000ms

eac3to use always -1000ms :)


Everything clear now ?

[ed_co]

I have a mkv file from the movie "300"
[4,29 GB (4.615.924.964 bytes)].

I want to convert the audio eac3 to ac3 (and next to mp3).

I extract the audio with mkvextract.

Then I try to convert the eac3 file to ac3, with eac3to, but the message that I see is the next:
C:\>"C:\Documents and Settings\Administrator\Desktop\EAC3toGUI\eac3to.exe" "C:\a
udio.eac3" "C:\audio.ac3" -640 -sonic
E-AC3, 5.1 channels, 1:56:33, 640kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
This track is not clean. Processing aborted.
Please clean the track with delaycut and then retry eac3to."

I try to clean the eac3 file with delaycut, but I obtain a 53 mb file (aprox), with different methods (ignore, fix, silence, skip), but I haven't got any file succesfully.

I don't know how to make it. I hope that someone help me.

Thank you very much.

[ACrowley]

When youve no Succes to fix your eac3, you can disable the DialNorm Removing

In this Case eac3to should work because the 100% clean EAC3 is required for the DialNorm removement.

Also you can try to demux the eac3 with avimuxgui (extract binary). Maybe its caused by mkvextract

[Chumbo]

@Acrowley,

When setting the delay, it is not necessary to input a "+" sign. The EAC3toGUI program will automatically assign and precede the amount specified by a "+" in the command line for values greater than zero (0).

If this is incorrect, please give a clear example of how the delay parameter (both for delay and advance) is specified on the command line and I will correct the EAC3toGUI program.

For instance, all command line options are preceded by "-". So is +100ms implemented as -100ms on the command line? and is -100ms implemented as --100ms?

Please confirm.

Regards,
The_Keymaker
I just wanted to add that my command line looks fine when the delay is a positive value whether I enter it or use the up arrow to increment it. When I execute the command with the delay at 100, the command fails because 100 is written as !00 with the exclamation point. I have no idea why that is, but it's strange. It's always the left most 1 that is replaced with an !, e.g.,
1 = !
11 = !1
111 = !11

And the same goes for the 2s and 3s and so on. The pattern is that the characters shifted character is what shows up when the line is executed and it's always the first, e.g.,
2112 = @112
400 = $00
and so on.

I'm on Windows XP Pro with latest sps. Anyone else exhibiting this behavior?

[The_Keymaker]

I have uploaded the latest version of EAC3toGUI, v1.37.

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

Change log:
- Fixed audio delay in command line (no "+" sign)
- Added Horizontal scroll to input and destination File

Please report any bugs or improvements.

Regards,
The_Keymaker

[Chumbo]

I have uploaded the latest version of EAC3toGUI, v1.37.

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

Change log:
- Fixed audio delay in command line (no "+" sign)
- Added Horizontal scroll to input and destination File

Please report any bugs or improvements.

Regards,
The_Keymaker
Wow, that was fast. :) Thank you. A few things to report:
- Positive delay number issue that I was running into is fixed. whew. ;)
- Source horizontal scroll works, but not Destination
- The first time I ran it and pointed to the location of eac3to.exe, it ran it but without the drive letter, i.e., "\avutils\eac3to.exe" and so on. After I repointed to the location, it then ran it as "c:\avutils\eac3to.exe" and so on.
- I noticed that when the command line is executed, it uses the location of eac3to to run from. Can this be changed to the location of the source instead? Thoughts on this?

More wish list items:
- Any chance of adding a drag-and-drop feature for the known file types?
- Since eac3to supports .ac3 files for removing dialog normalization, any chance of adding this to the UI?
- After you close the app and reopen it, the Source file Browse takes you to the eac3to.exe location rather than "remembering" the last Source location. It would be nice if it "remembered" the last Source location.
Thanks for continuing to solidify this program. :D

[ed_co]

When youve no Succes to fix your eac3, you can disable the DialNorm Removing

In this Case eac3to should work because the 100% clean EAC3 is required for the DialNorm removement.

Also you can try to demux the eac3 with avimuxgui (extract binary). Maybe its caused by mkvextract

Thanks ACrowley for your reply.

OK, I trid it with the next results:
First don't found the nero file source or dts\ac3\dd+, I download a dtsac3source.ax (in VistaCodecs_v451.exe), and I registered it.

Then the error, were the next:
"C:\>"C:\Documents and Settings\Administrator\Desktop\EAC3toGUI\eac3to.exe" "C:\audio.eac3" "C:\audio.ac3" -keepDialnorm
E-AC3, 5.1 channels, 1:56:33, 640kbit/s, 48khz, dialnorm: -27dB
Decoding eac3 file to raw. Please wait...
Getting "Nero Audio Decoder 2" instance failed."

But I select the sonic decoder (I had it, and I haven't got nero) with the next result:

"C:\>"C:\Documents and Settings\Administrator\Desktop\EAC3toGUI\eac3to.exe" "C:\audio.eac3" "C:\audio.ac3" -sonic -keepDialnorm
E-AC3, 5.1 channels, 1:56:33, 640kbit/s, 48khz, dialnorm: -27dB
Decoding eac3 file to raw. Please wait..."

It made two things:
1) rename the audio.eac3 file to audio.tmp.ddp
2) made a small file called audio.raw (107mb) and I can't reproduce it (I suppose that it's normal).

I don't know, what can I do next.

Thank you.

[Thunderbolt8]

an observation I recently made regarding the delay of DTS-HD and TrueHD converted FLAC files: I've only made FLAC out of 3 DTS-HD movies so far, but none of them seemed to need any delay.

All the TrueHD flacs I've made needed some delay, mostly ~100-200ms. im so far to say that, when both streams, video and audio, begin at the same PTS in the evo, then I can work mathematically. I compare the length of each track, remuxed video to .mkv and length of the FLAC with mediainfo and use the difference as delay. So far all of the tracks I did that with looked good.

But the same thing just doesnt work for the DTS-HD flacs. the delay I get here from mediainfo is mostly <100ms difference and therefore hard to spot anyway, whether its needed or not. But when looking extremely closely it has always looked better when I applied no delay at all for those files instead of relying on the mathematical way, as I can do for the TrueHD FLACs. I dont know if thats maybe due to a read error of mediainfo, that it reads wrong information about the dts-hd tracks, but once its in FLAC format it shouldnt matter anyway any more, whether it was DTS-HD or TrueHD before. Since I've only had 3 DTS-HD tracks so far I cant say if all such tracks wont need any delay, but lets see what the future will bring.

So to sum it up: What worked for me for finding delay for TrueHD and DTS-HD is

1. no delay at all needed for DTS-HD so far
2. TrueHD delay calculated by length of .mkv video (23.9760239fps) - length of converted FLAC track.

the result in both cases looked well and COULD(!) maybe be completely identical with the original source status.

[morphx2]

Hmm..maybe you can help me out with my audio question

VC-1 video stream 0 found!
First PTS = 00000D45
Dolby Digital Plus audio stream 0 found!
First PTS = 00001105 (+10ms)

So in the audio delay feature, do I do +10ms like it says or -10ms to correct it? I am confused :) Or is there a mathematical way to figure out the correct delay (if evodemux is not the true delay)

Thanks!

[ACrowley]

Thanks ACrowley for your reply.

OK, I trid it with the next results:
First don't found the nero file source or dts\ac3\dd+, I download a dtsac3source.ax (in VistaCodecs_v451.exe), and I registered it.

Then the error, were the next:
"C:\>"C:\Documents and Settings\Administrator\Desktop\EAC3toGUI\eac3to.exe" "C:\audio.eac3" "C:\audio.ac3" -keepDialnorm
E-AC3, 5.1 channels, 1:56:33, 640kbit/s, 48khz, dialnorm: -27dB
Decoding eac3 file to raw. Please wait...
Getting "Nero Audio Decoder 2" instance failed."

But I select the sonic decoder (I had it, and I haven't got nero) with the next result:

"C:\>"C:\Documents and Settings\Administrator\Desktop\EAC3toGUI\eac3to.exe" "C:\audio.eac3" "C:\audio.ac3" -sonic -keepDialnorm
E-AC3, 5.1 channels, 1:56:33, 640kbit/s, 48khz, dialnorm: -27dB
Decoding eac3 file to raw. Please wait..."

It made two things:
1) rename the audio.eac3 file to audio.tmp.ddp
2) made a small file called audio.raw (107mb) and I can't reproduce it (I suppose that it's normal).

I don't know, what can I do next.

Thank you.

You dont need single Nero Filters
When youve your Nero7 with HDDVD/BR Plugin installed, all Filters/Decoders are registred.
So for decoding eac3 with Nero youve already all what you need.

Your Error "Nero Audio Decoder 2" instance failed."
Sound sfor me youve worng Nero Version...You know, you can only use Nero 7 ? Nero8 will not work with eac3to !!
And/Or the HDDVD/BluRAy Plugin isnt properly installed

AH, dont use Sonic Decoder! You get full DRC/DialoNorm/Gain=BAD

Try :
1 As i say, demux eac3 from mkv with AvimuxGui!
2. Check Nero installation

@TheKeymaker
THX! It works

It would like to see this Features :

- When eac3to Process is complet, a simple "Complete/Done" Message from the Gui itself would be nice.
Because eac3to Command Promt Window closes autom

-Yes, drag&drop would be nice

-.ac3 Input/Output Extension , to use the DialogNorm Remove Feature : eac3to src.ac3 dest.ac3

[ed_co]

You dont need single Nero Filters
When youve your Nero7 with HDDVD/BR Plugin installed, all Filters/Decoders are registred.
So for decoding eac3 with Nero youve already all what you need.

Your Error "Nero Audio Decoder 2" instance failed."
Sound sfor me youve worng Nero Version...You know, you can only use Nero 7 ? Nero8 will not work with eac3to !!
And/Or the HDDVD/BluRAy Plugin isnt properly installed

AH, dont use Sonic Decoder! You get full DRC/DialoNorm/Gain=BAD

Try :
1 As i say, demux eac3 from mkv with AvimuxGui!
2. Check Nero installation

@TheKeymaker
THX!

Thank you again, ACrowley.

I will try it, when I come of a travel (in 25 minutes, I must go to the airport and I'll come back on friday), and I'll post the results.

Thank you.

Greetings.

[Chumbo]

Hmm..maybe you can help me out with my audio question

VC-1 video stream 0 found!
First PTS = 00000D45
Dolby Digital Plus audio stream 0 found!
First PTS = 00001105 (+10ms)

So in the audio delay feature, do I do +10ms like it says or -10ms to correct it? I am confused :) Or is there a mathematical way to figure out the correct delay (if evodemux is not the true delay)

Thanks!
You do what it says, e.g., if it's +100 then you use 100 in delaycut for example and if it's -70 then you'd use -70 in delaycut.

[morphx2]

wonderful, thanks sir!

[Sephiroth0000]

Hello everyone. I do not know if anyone remembers me but I was attempting EVO to WMV with 5.1 surround sound. And I recieved alot of help from this particular forum for who without their help I would not of been successful. I know its been awhile but I have completed my walkthrough (which is a more thorough guide of OliverDB's) I do apologise for the delay but being a father of 3 and working makes it hard to get anything finished lol. All I would like to know now is how is the best way to upload my walkthrough?

[Sephiroth0000]

http://rapidshare.com/files/66007512/EVO_to_WMV_HD_with_5.1.docx.html

This is my link to my EVO to WMV with 5.1 support. Oh and ACrowley that problem you are having with your Audio is a problem I once had and can be corrected very easily. If you look at pages 44 and above on this thread you should find the answer as it was the answe someone gave me with regards to the same problem. Either that or read me guide :P

[madshi]

Thanks for the tip, here's the first 5000KB of the file:

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

It was demuxed using TSRemux if that is relevant.
Thanks, I'll have a look at this later. Might take a few days, though.

[madshi]

I got a eac3to error for the FLAC conversion of The Graduate's (Studio Canal) 2nd and 3rd DTS-HD track. for the first track though everything seems to be fine, the track was converted normally. but right after having pressed enter for the other 2 tracks eac3to & command line just crashed and gave me a windows error message (which I couldnt copy&paste). I've sent you 3x5MB samples of all 3 streams on that HD DVD, the errors happens for streams 01 and 02.
Thanks. Just in case: Could you please also make a sample of the EVO file available to me? EvoDemux has had problems with demuxing DTS-HD before (The Perfume), so in order to rule an EvoDemux bug out, having a little EVO sample would be helpful. The first 50MB of the first EVO file should hopefully be enough. Please do not touch the EVO file with EvoDemux at all, just to be safe.

[madshi]

could the error maybe be caused, because those 2 tracks are only normal dts tracks? at least mediainfo tells so. while evodemux names them as DTS-HD on the main page -.-
Nope, there's no problem if a DTS-HD flagged stream is only conventional DTS.

[Carpo]

if mem serves me right isnt docx - office 2007 if so any chance you could save the file as a 97/200/2003 doc so people using open office could use them

Thanks

[madshi]

Positive Delay :

Incorrect :
eac3to xxx.flac xxx1.flac +1000ms

Correct :
eac3to xxx.flac xxx1.flac 1000ms

Youre using +1000ms what is wrong

Negative Delay :
eac3to xxx.flac xxx1.flac -1000ms

eac3to use always -1000ms :)


Everything clear now ?
IIRC eac3to accepts both "+1000ms" and "1000ms". It's handled identically.

[madshi]

@madshi, please upload the updated program to your link on page 1.
Thanks for the new updated GUI! v1.37 uploaded to the link on page 1.

[madshi]

- If possible to add an option, -keep, or something that would allow us to keep the working files created to achieve an end result. Right now, those files get deleted when the process is done.
Which files exactly are you interested in? It would be no big problem to add such a feature with the current eac3to version. But things are different with the next bigger version. So I need to know exactly why you need this and which intermediate files you need exactly.

[baconharvest]

I'm having a problem converting eac3 to ac3 using eac3to. Here's the list of the stuff I have in the command line:

"C:\Documents and Settings\Name\Desktop\eac3to\eac3to.exe" "C:Shared\audio.eac3" "C:\Shared\audio.ac3" -640 -nero
E-AC3, 5.1 channels, 1:08:59, 640kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Decoding eac3 file to raw. Please wait...
The file size of the raw file doesn't seem to fit.
The expected file size for 16 bit is 2.21 GB.
The expected file size for 24 bit is 3.31 GB.
The real file size is 0 Bytes.

Additional information: I extracted the eac3 audio file (315mb) from an mkv using mkvextract (by way of command line), I also have an installed and registered version of Nero 7 Ultimate Edition and Graphedit from doom9 with the dump filter.

Any ideas on how to fix the error above and get an ac3 file? And is there a way to tell how big the ac3 file will be?

[Thunderbolt8]

IIRC eac3to accepts both "+1000ms" and "1000ms". It's handled identically.
yes, I can confirm that. used both already, same effect
Thanks. Just in case: Could you please also make a sample of the EVO file available to me? EvoDemux has had problems with demuxing DTS-HD before (The Perfume), so in order to rule an EvoDemux bug out, having a little EVO sample would be helpful. The first 50MB of the first EVO file should hopefully be enough. Please do not touch the EVO file with EvoDemux at all, just to be safe.
done.
http://www.megaupload.com/de/?d=VBMNCSVR

[Chumbo]

Which files exactly are you interested in? It would be no big problem to add such a feature with the current eac3to version. But things are different with the next bigger version. So I need to know exactly why you need this and which intermediate files you need exactly.
No biggie, I can wait for the next bigger version. :) I basically was messing with eac3to just to check out all its features and noticed it moves audio into an evo container, for example, and then creates a raw file and then eventually produces the result. I just thought it would be convenient to keep those "working" evo and raw files out of curiosity. So no big deal. ;)

[Chumbo]

I'm having a problem converting eac3 to ac3 using eac3to. Here's the list of the stuff I have in the command line:

"C:\Documents and Settings\Name\Desktop\eac3to\eac3to.exe" "C:Shared\audio.eac3" "C:\Shared\audio.ac3" -640 -nero
E-AC3, 5.1 channels, 1:08:59, 640kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Decoding eac3 file to raw. Please wait...
The file size of the raw file doesn't seem to fit.
The expected file size for 16 bit is 2.21 GB.
The expected file size for 24 bit is 3.31 GB.
The real file size is 0 Bytes.

Additional information: I extracted the eac3 audio file (315mb) from an mkv using mkvextract (by way of command line), I also have an installed and registered version of Nero 7 Ultimate Edition and Graphedit from doom9 with the dump filter.

Any ideas on how to fix the error above and get an ac3 file? And is there a way to tell how big the ac3 file will be?
Do you have enough disk space where the raw file is being created?

[Sephiroth0000]

CARPO you asked for a 1997/2000/2003 word file instead so here is the new link. Enjoy it everyone :)

http://rapidshare.com/files/66107349/EVO_to_WMV_HD_with_5.1.doc.html

[baconharvest]

Do you have enough disk space where the raw file is being created?

Yes, I have 5 gigs of free space on my computer.

[Phorze]

I'm having a problem converting eac3 to ac3 using eac3to. Here's the list of the stuff I have in the command line:

"C:\Documents and Settings\Name\Desktop\eac3to\eac3to.exe" "C:Shared\audio.eac3" "C:\Shared\audio.ac3" -640 -nero
E-AC3, 5.1 channels, 1:08:59, 640kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Decoding eac3 file to raw. Please wait...
The file size of the raw file doesn't seem to fit.
The expected file size for 16 bit is 2.21 GB.
The expected file size for 24 bit is 3.31 GB.
The real file size is 0 Bytes.

Additional information: I extracted the eac3 audio file (315mb) from an mkv using mkvextract (by way of command line), I also have an installed and registered version of Nero 7 Ultimate Edition and Graphedit from doom9 with the dump filter.

Any ideas on how to fix the error above and get an ac3 file? And is there a way to tell how big the ac3 file will be?

I'm having the exact same problem, no matter what i try, i've come across this problem more often in the thread.
I have it with all options, -nero -sonic -orbitlee.
It just fails to write the raw file or something. There have been pointers on how to test using graphedit however im not sure how (it includes renaming to recode.exe , but dunno what i should rename)

In my case i'm trying to recode a dd+ track to dts.

H:\PROGGIES\eac3to>eac3to F:\Films\test\test.eac3 F:\Films\test\test.dts -1536 -nero
E-AC3, 5.1 channels, 2:23:27, 1536kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Decoding eac3 file to raw. Please wait...
The file size of the raw file doesn't seem to fit.
The expected file size for 16 bit is 4,61 GB.
The expected file size for 24 bit is 6,92 GB.
The real file size is 0 Bytes.

I've got nero 7.8 with the hddvd filter activated, and installed the sonic package.

[madshi]

@Phorze and baconharvest:

(1) Make sure you're using the latest available Nero 7 version.
(2) Have you bought the Nero HD DVD plugin?

[madshi]

is there a way to tell how big the ac3 file will be?
The size of the AC3 file only depends on the runtime of the movie and the bitrate used for encoding. Your E-AC3 is 640kbit/s. If you use the eac3to defaults for AC3 encoding, 640kbit/s will be used again. So the final AC3 file should have the same size as the original E-AC3 file.

[Carpo]

CARPO you asked for a 1997/2000/2003 word file instead so here is the new link. Enjoy it everyone :)

http://rapidshare.com/files/66107349/EVO_to_WMV_HD_with_5.1.doc.html

Thanks

[nautilus7]

Yes, I have 5 gigs of free space on my computer.

I don't think this is enough...
Only .raw file could be bigger than 5GB!

[Sephiroth0000]

This post may prove of use to some of you that keep on getting this audio raw expected file size error. I had the problem once when I was doing EVO to WMV with 5.1. Now I will assume that the people having these errors are doing it with 5.1 audio. The way I got around this problem was making sure that EAC3TO, SOX and WAVAVIMUX was installed to the same folder as your audio and video sources that you are transcoding from. I was using Nero 7.10.0.1 with the BluRay HD DVD Plugin aswell which is crucial. Make sure that the file name of the audio file that it is you are trying to transcode is not a long file name and change it to something small and simple if you can.

But remember DO NOT INSTALL WAVAVIMUX to the Windows Media Encoder components folder and instead to the folder you are working from along with all your other files.

[Phorze]

This post may prove of use to some of you that keep on getting this audio raw expected file size error. I had the problem once when I was doing EVO to WMV with 5.1. Now I will assume that the people having these errors are doing it with 5.1 audio. The way I got around this problem was making sure that EAC3TO, SOX and WAVAVIMUX was installed to the same folder as your audio and video sources that you are transcoding from. I was using Nero 7.10.0.1 with the BluRay HD DVD Plugin aswell which is crucial. Make sure that the file name of the audio file that it is you are trying to transcode is not a long file name and change it to something small and simple if you can.

But remember DO NOT INSTALL WAVAVIMUX to the Windows Media Encoder components folder and instead to the folder you are working from along with all your other files.

i don't get this, i don't need wavavimux right?
And eac3to refers to it's own subfolders to sox or aften when needed right?

[Sephiroth0000]

i don't get this, i don't need wavavimux right?
And eac3to refers to it's own subfolders to sox or aften when needed right?

I am not too sure about the runnings and how it all connects but whatever 5.1 audio file I am doing with EACH3TO all I do is make sure that EAC3TO, EAC3TO GUI, SOX and WAVAVIMUX is all based in the same folder as the files im working on. I am not saying this method may nesscarily work for you but it did for me when I got that error. Also make sure you have the Nero BluRay HD DVD plugin (must be geniune) Also please note that EAC3TO does not work or function properly with Nero 8.

[Phorze]

So i kinda gave up on nero, i just can't get it to work, showtime plays HD content fine tho. And yes it's a legit version.

using -orbitlee -sonic is working now, but as mentioned in startpost it automagically applies DRC to the track which is catastrophic ofcourse.

I found this site:
http://codec.kiev.ua/mpeg2.htm

somewhere near the bottom :
Nero Suite ShowTime v.5.2.1.0 CE

Would this be usable? (and free!)

[Sephiroth0000]

What is it you are trying to do anyway Phorze? I might be able to help more if I know what it is you are doing.

[madshi]

So i kinda gave up on nero, i just can't get it to work, showtime plays HD content fine tho. And yes it's a legit version.
Have you bought the HD DVD / Blu-Ray plugin?

Would this be usable? (and free!)
No, it's not usable. And I don't think that that site is legal.

[Sephiroth0000]

If you use the EAC3TO GUI then you may be able to stop the DRC entirely. Im sure it has something about DRC in there.

[baconharvest]

How can I tell whether or not I have the hd-dvd plugin? All I see under my product license is Nero 7 Ultra Edition (I was under the impression that the hd-dvd plugin came with it).

[Sephiroth0000]

How can I tell whether or not I have the hd-dvd plugin? All I see under my product license is Nero 7 Ultra Edition (I was under the impression that the hd-dvd plugin came with it).

Unfortunately not. You do have to purchase the BluRay HD DVD Plugin seperately and it comes in the form of a serial that you enter at NeroProductSetup. The PlugIn cost me £14.99 when I purchased it and that was 2 months ago. You will have to purchase it.

[ACrowley]

Unfortunately not. You do have to purchase the BluRay HD DVD Plugin seperately and it comes in the form of a serial that you enter at NeroProductSetup. The PlugIn cost me £14.99 when I purchased it and that was 2 months ago. You will have to purchase it.

When youve the Plugin installed you can see it under your Licenses in NeroStartSmart!
The must be Nero HDDVD/Bluray Plugin

When its not there ...youve no Plugin installed :)

Yes, it never come with a standard Ultra/premium Version.
You must by it seperatly

[baconharvest]

If I purchase the Bluray/HDDVD Plugin, will Nero ShowTime be able to play the mkv file with eac3 audio?

[madshi]

If I purchase the Bluray/HDDVD Plugin, will Nero ShowTime be able to play the mkv file with eac3 audio?
I don't know exactly what Nero ShowTime will or will not be able to do. I'm only using the Nero Audio Decoder. But I'm planning to (still didn't find the time to do so) release a little tweak that allows using the Nero Audio Decoder inside of Media Player Classic and ZoomPlayer etc with DRC turned off. This tweak will only work for Nero 7, though.

[Thunderbolt8]

remuxed the blu-ray of casino royale. after remuxing the video with 23,9760239 fps and converting the LPCM track to flac both tracks had exactly(!) the same length. no delay needed at all.

[nautilus7]

@ Madshi

I don't understand why the trueHD track has to muxed into an .evo container before converted to wav/flac.
I know that the initial .evo has to demuxed in order to remove the dialog norm, but i don't get why it has to muxed again.

Thanks!

[madshi]

I don't understand why the trueHD track has to muxed into an .evo container before converted to wav/flac.
I know that the initial .evo has to demuxed in order to remove the dialog norm, but i don't get why it has to muxed again.
There's no other way (that I know) to feed the Nero Audio Decoder with the TrueHD data. It only accepts input from the Nero Splitter and the Nero Splitter doesn't handle demuxed TrueHD files.

[nautilus7]

It just looked odd to me. I see now. Thanks for the clarification.

[nautilus7]

These pics are from the "Letters from Iwo Jima" HD DVD.
It contains an eac3 and a trueHD track.
I demuxed both with latest version of eac3to using nero filters.

eac3 track
http://img475.imageshack.us/img475/7480/jimaeac3xx6.th.png (http://img475.imageshack.us/my.php?image=jimaeac3xx6.png)

trueHD track
http://img458.imageshack.us/img458/1931/jimathdry0.th.png (http://img458.imageshack.us/my.php?image=jimathdry0.png)

How do you explain that eac3 has higher picks than trueHD? Is this random (has to do with the specific movie)?

About calculating the delay: Have you ever find an eac3 track that needs to be delayed?

[madshi]

These pics are from the "Letters from Iwo Jima" HD DVD. It contains an eac3 and a trueHD track. I demuxed both with latest version of eac3to using nero filters.

How do you explain that eac3 has higher picks than trueHD? Is this random (has to do with the specific movie)?
I'm not totally sure about that. I think I did compare TrueHD vs. E-AC3 and they had identical peaks to me, but it's some time ago so I'm not really sure. Maybe you can check again with the next movie? I don't think this is caused by a bug in eac3to, but I'm not sure 100% about that, either...

About calculating the delay: Have you ever find an eac3 track that needs to be delayed?
I think I've had some E-AC3 tracks which needed a delay. It's rather the exception than the rule, though.

[nautilus7]

I 'll check with another movie, like you said, and i 'll let you know.

[Thunderbolt8]

what is actually needed for TrueHD 7.1 support for eac3to ?
does eac3to actually differentiate between dts-hd hr and MA and that 16-bit downsampling applies only to HR, or does it apply to both?

[madshi]

what is actually needed for TrueHD 7.1 support for eac3to ?
A decoder which can decode 7.1. Maybe the new open source decoder in ffmpeg can do that. We'll have to wait a bit for that. Do you have a movie with 7.1 TrueHD? Which one?

does eac3to actually differentiate between dts-hd hr and MA and that 16-bit downsampling applies only to HR, or does it apply to both?
I'm not sure what you mean exactly. Which 16bit downsampling do you mean?

[Taktaal]

Now that Nero 7 isn't sold anymore, there's no way to decode DD+ tracks, isn't there?

I sure hope an Open Source DD+ codec comes soon, that's pretty much the holy grail of free HD playback right now.

[madshi]

Now that Nero 7 isn't sold anymore
Nero 7 Essentials can be had for very cheap on Ebay. And I think it works together with the HD DVD/Blu-Ray plugin.

[Wilbert]

I'm not sure what the difference is between Blu-Ray TrueHD and TrueHD (anyone?), but the creator of the MLP/TrueHD patch is asking for some Blu-Ray TrueHD.

http://lists.mplayerhq.hu/pipermail/ffmpeg-devel/2007-October/thread.html#36628

So join the ffmpeg mailinglist, and send some samples.

Btw, Kurtnoise added MLP/TrueHD/E-AC3 support to the AviSynth plugin FFmpegSource:

http://forum.doom9.org/showthread.php?t=127037&page=9

You might want to go over there and start testing.

[madshi]

I'm not sure what the difference is between Blu-Ray TrueHD and TrueHD (anyone?)
Blu-Ray TrueHD has some AC3 frames added in the TrueHD stream. Basically you just need to remove/skip them and you end up with a HD DVD style TrueHD stream.

I don't have a sample at hand right now. Can anybody else help out?

[honai]

Sure, here you go:

http://myfreefilehosting.com/f/b8cfd37f97_42.65MB

[madshi]

Thanks, honai.

@Wilbert, could you post that to the mailing list, as I don't have access to it right now (and currently no time to join).

[nautilus7]

The delay option is only working on flac conversion, right?
Is it possible to apply a delay on ac3/dts also?

[madshi]

The delay option is only working on flac conversion, right?
The delay works for any conversion which has a PCM/RAW stage in it. So if there's real "decoding" and/or "encoding".

Is it possible to apply a delay on ac3/dts also?
That's currently not supported. You can use delaycut for that purpose. Maybe I'll add this functionality in a future version of eac3to.

[nautilus7]

Thanks for the quick answer.

[Wilbert]

@honai/mashi,

I've send a mail to Ian (the guy who made the patch). I will let you know if he needs other samples.

[hristoff2]

Hi,

eac3to won't read this pcm 5.1 16bit file properly, the output is only noise.

eac3to reports it can't determine whether the file is 16bit or 24bit, so I add -16 to the cli -> output is noise but I can play the file when I mux it to matroska (the pcm source) and play it then with ac3filter, so I guess there's something broken

http://rapidshare.com/files/67172503/sample.pcm.html

greets :)

[ed_co]

You dont need single Nero Filters
When youve your Nero7 with HDDVD/BR Plugin installed, all Filters/Decoders are registred.
So for decoding eac3 with Nero youve already all what you need.

Your Error "Nero Audio Decoder 2" instance failed."
Sound sfor me youve worng Nero Version...You know, you can only use Nero 7 ? Nero8 will not work with eac3to !!
And/Or the HDDVD/BluRAy Plugin isnt properly installed

AH, dont use Sonic Decoder! You get full DRC/DialoNorm/Gain=BAD

Try :
1 As i say, demux eac3 from mkv with AvimuxGui!
2. Check Nero installation

@TheKeymaker
THX! It works

It would like to see this Features :

- When eac3to Process is complet, a simple "Complete/Done" Message from the Gui itself would be nice.
Because eac3to Command Promt Window closes autom

-Yes, drag&drop would be nice

-.ac3 Input/Output Extension , to use the DialogNorm Remove Feature : eac3to src.ac3 dest.ac3

I can't install nero 7.10.1.0, because I have the next error "neroag_registercomex", and I try to clean with nero general clean tool and "your uninstaller 2006" (because I found it like solution in forums, but didn't work).
I don't understand.

[ACrowley]

@ed_co
Simply use the NeroCleaner tool

@hristof

Try to change the Endian Type.
Did you demux it with TSremux ? All LPCM Tracks i demuxed with TSRemux where garabage ,with a loud noise from eac3to
So i use only Xport (Big Endian always)

[hristoff2]

Used Xport in the first place (but the output was corrupted, 300mb too small), so I used gdsmux.exe from haali dir. which worked like a charm.

For the time being I used audacity to export the audio tracks (mono) and corrected the channel mapping manually :)

[ed_co]

@ed_co
Simply use the NeroCleaner tool


Like I said. Don't work.
I have windows xp x64 (for your information).

[Thunderbolt8]

A decoder which can decode 7.1. Maybe the new open source decoder in ffmpeg can do that. We'll have to wait a bit for that. Do you have a movie with 7.1 TrueHD? Which one?
batman begins HD DVD is 7.1 TrueHD.

I'm not sure what you mean exactly. Which 16bit downsampling do you mean?
I was refering to this:

known problems:

(2) DTS-HD High Resolution tracks encoded from 16bit masters are forcefully downconverted to 16bit.

So does that mean that DTS-HD MA 24-bit tracks also get downconverted to 16bit or does that only apply to the DTS-HD HR tracks?

[ACrowley]

@natilus7

Wheres your Problem ? Its fine.
I dont think the differences in your eac3/thd Test is out of Norm, i cant see anything what is not right, no differences which are relevant (if any ,theyre marginal only)
We compared (and had a long discusion)a lot of Tracks for a while and eac3to/Nero(tweaked) Decoder applies 100% clean no DRC and no DialNorm
Both waves in your pics are obviously without DRC/DialNorm and Ok
So it "surely" not a eac3to/Nero Problem and surely no Problem at all :)

@hristoff
Ah Ok.. so it must be caused from gdsmux .
Never had Problems with xport..it was always correct/Sync/Size

What works too is via Graphedit :
Filesource(Async)-SonicHDDemuxer-Dump

@Thunderbolt8
Theres no 16 bit downsampling from eac3to on dts hd. You get always the correct output ,when the dts hd has 24bit its 24bit and reversed.
Standard DTS output is "always" 24bit for some good reasons
So we have no "16bit downsampling" on dts (hd) in any case

[madshi]

So does that mean that DTS-HD MA 24-bit tracks also get downconverted to 16bit or does that only apply to the DTS-HD HR tracks?
With DTS-HD Master Audio the situation is quite reversed: If the master was only 16bit then we *WANT* the decoding to be only 16bit, too. That's the point of "lossless".

[madshi]

Theres no 16 bit downsampling from eac3to on dts hd. You get always the correct output ,when the dts hd has 24bit its 24bit and reversed.
Standard DTS output is "always" 24bit for some good reasons
So we have no "16bit downsampling" on dts (hd) in any case
I believe DTS-HD High Resolution is downconverted to 16bit by the Sonic decoder, if the original master was only 16bit.

[madshi]

eac3to won't read this pcm 5.1 16bit file properly, the output is only noise.

eac3to reports it can't determine whether the file is 16bit or 24bit, so I add -16 to the cli -> output is noise but I can play the file when I mux it to matroska (the pcm source) and play it then with ac3filter, so I guess there's something broken
The sample file you uploaded is a WAV file and not a PCM file. Just rename it to WAV and you're done.

[madshi]

batman begins HD DVD is 7.1 TrueHD.
Could you upload a small sample? That would be helpful. 10MB should be enough. Thanks!

[honai]

Just noticed something odd: When converting an EAC3 track to WAVs (and possibly with other operations as well) eac3to seems to pull (or instantiate, not sure what exactly it does) the Sonic Audio Decoder's control even though my system has the Nero decoder installed. Isn't eac3to supposed to check for Nero first? Might this be an indication that eac3to actually uses Sonic even though it tried to instantiate Nero's decoder?

[nautilus7]

@natilus7

Wheres your Problem ? Its fine.
I dont think the differences in your eac3/thd Test is out of Norm, i cant see anything what is not right, no differences which are relevant (if any ,theyre marginal only)
We compared (and had a long discusion)a lot of Tracks for a while and eac3to/Nero(tweaked) Decoder applies 100% clean no DRC and no DialNorm
Both waves in your pics are obviously without DRC/DialNorm and Ok
So it "surely" not a eac3to/Nero Problem and surely no Problem at all :)

I didn't say there is drc applied to the thd track. I just asked why there were higher (even that little) peaks in the thd. I am glad it's nothing wrong. Thanks!

[KillTonic]

I've demuxed the m2ts with xport.
But the TrueHD file i've tried to convert to 5.1 wav (for further Nero AAC Encoding) is just crappy noise.
btw, whats the correct file extension on a TrueHD Audio File..PCM?
Can anyone please help me about this? :)

[nautilus7]

TrueHD extension that eac3to accepts is .thd

Have in mind also that wav max size is 4GB. The size of a 2 hours 5.1 wav file is more than this, so you gonna have problem. You can use .wavs as output to convert the thd to 6 mono wavs, but i don't know how your gonna encode them to aac.

[Chumbo]

I've demuxed the m2ts with xport.
But the TrueHD file i've tried to convert to 5.1 wav (for further Nero AAC Encoding) is just crappy noise.
btw, whats the correct file extension on a TrueHD Audio File..PCM?
Can anyone please help me about this? :)
The first post of a tool thread usually will answer the type of question you have and so does the command line help in most cases.

[madshi]

Just noticed something odd: When converting an EAC3 track to WAVs (and possibly with other operations as well) eac3to seems to pull (or instantiate, not sure what exactly it does) the Sonic Audio Decoder's control even though my system has the Nero decoder installed. Isn't eac3to supposed to check for Nero first? Might this be an indication that eac3to actually uses Sonic even though it tried to instantiate Nero's decoder?
How do you know that eac3to wants to "pull" the Sonic Audio Decoder? I'm not sure exactly what you mean. Maybe posting the eac3to output text would help?

[Thunderbolt8]

Could you upload a small sample? That would be helpful. 10MB should be enough. Thanks!
I could, but I dont have much time atm, so I dont know when this will be. might as well take 1-2 weeks :S

btw. I just read that flac.exe cant handle 7.1 audio anyway, is this true? if yes, then this might be pointless anyway :/

[ACrowley]

I didn't say there is drc applied to the thd track. I just asked why there were higher (even that little) peaks in the thd. I am glad it's nothing wrong. Thanks!

yes, i know waht you mean :) i only want to tell you theres no Problem.
No DRC/DialNorm from eac3to/tweaked NeroDecoder and your waves are correct and almost identical
=Null Problemo

[honai]

How do you know that eac3to wants to "pull" the Sonic Audio Decoder?

That's what the registry monitor of Kaspersky told me.

[depechefan]

I get this message when trying to convert the .THD to .WAV

Getting "Nero File Source Async" instance failed

any suggestions?

Here was my command line:
D:\star fighter\2>eac3to FEATURE_1_MERGED.MLP.stream.00.thd output.wav
TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
The following tasks are executed now. Please wait...
- remove Dialog Normalization information
- muxing TrueHD track into temp evo file
Decoding TrueHD track to raw. Please wait...
Getting "Nero File Source Async" instance failed.

[madshi]

I could, but I dont have much time atm, so I dont know when this will be. might as well take 1-2 weeks :S
No problem, I'm not in a hurry.

btw. I just read that flac.exe cant handle 7.1 audio anyway, is this true? if yes, then this might be pointless anyway :/
FLAC handles 7.1 just fine.

[madshi]

That's what the registry monitor of Kaspersky told me.
I think you can safely ignore that.

[madshi]

I get this message when trying to convert the .THD to .WAV

Getting "Nero File Source Async" instance failed

any suggestions?

Here was my command line:
D:\star fighter\2>eac3to FEATURE_1_MERGED.MLP.stream.00.thd output.wav
TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
The following tasks are executed now. Please wait...
- remove Dialog Normalization information
- muxing TrueHD track into temp evo file
Decoding TrueHD track to raw. Please wait...
Getting "Nero File Source Async" instance failed.
Most probably you don't have Nero installed properly. Try uninstalling and reinstalling Nero. Please note that Nero 8 won't work. You need Nero 7.

[depechefan]

Most probably you don't have Nero installed properly. Try uninstalling and reinstalling Nero. Please note that Nero 8 won't work. You need Nero 7.

Oh! Well, that's why then! I had Nero 8! I'll try it. Thank you!

[nautilus7]

I guess that i can replace the sox.exe and aften.exe that come with eac3to with newer ones.
Which version of aften does eac3to work with? SSE, SSE2, SSE3, "simple" one or all?

The sox link in the fisrt post doesn't point to the latest version (14).

[nautilus7]

And something else i forgot to ask...

TrueHD from BluRay has ac3 frames in it, right?
So, if i run eac3to bluray.thd output.ac3 it will extract those frames. How can i encode a new ac3 from this trueHD track (using aften)?


Also in the first post, it says:
extraction of AC3 frames from AC3/TrueHD tracks:
eac3to src.thd dst.ac3

Shouldn't this be extraction of AC3 frames from EAC3/TrueHD tracks?

:thanks:

[honai]

So, if i run eac3to bluray.thd output.ac3 it will extract those frames. How can i encode a new ac3 from this trueHD track (using aften)?

Why would you want to do that? It makes no sense. The embedded AC3 is already at 640kbps, you can't get better quality or higher bitrate with aften.

Shouldn't this be extraction of AC3 frames from EAC3/TrueHD tracks?

No.

[madshi]

I guess that i can replace the sox.exe and aften.exe that come with eac3to with newer ones.
Theoretically yes. Practically I haven't tested it and I'm not sure if you gain anything by doing that. But of course you can try.

Which version of aften does eac3to work with? SSE, SSE2, SSE3, "simple" one or all?
I'm using the x86 build which internally has branches for SSE*, IIRC.

The sox link in the fisrt post doesn't point to the latest version (14).
That's probably because I didn't test that version yet. And I won't ever test it, because the next bigger eac3to version will not use sox at all, anymore. The only thing eac3to uses sox for right now is for converting 24bit tracks down to 16bit. The next bigger eac3to version will do that internally - and in a better quality compared to sox. Sox 13 at least is just rounding each sample - which is a bad idea for audio quality. eac3to will use proper TPDF ("Triangular Probability Density Function") dithering.

[madshi]

TrueHD from BluRay has ac3 frames in it, right?
Correct.

So, if i run eac3to bluray.thd output.ac3 it will extract those frames. How can i encode a new ac3 from this trueHD track (using aften)?
You could theoretically run "eac3to bluray.thd bluray.wav" to decode the TrueHD track to wav and then you could convert the wav file to AC3 by using aften.exe (don't know right now if eac3to supports "WAV -> AC3").

But as honai already said, this doesn't really make sense. The AC3 frames in the combined AC3/TrueHD track should always sound better than any AC3 encoding you can do with Aften.

[idbirch2]

Hi madshi, any joy with that weird DTS file?

I've got a BluRay with a really weird DTS audio format. When I try and run eac3to on it I get:

DTS Master Audio, 3/1 channels, 24 bits, 48khz
This channel format is currently not supported.

[nautilus7]

The next bigger eac3to version will do that internally - and in a better quality compared to sox. Sox 13 at least is just rounding each sample - which is a bad idea for audio quality. eac3to will use proper TPDF ("Triangular Probability Density Function") dithering.Nice!

Why would you want to do that? It makes no sense. The embedded AC3 is already at 640kbps, you can't get better quality or higher bitrate with aften.

But as honai already said, this doesn't really make sense. The AC3 frames in the combined AC3/TrueHD track should always sound better than any AC3 encoding you can do with Aften.

Maybe i want to get an ac3 with lower bitrate than 640. So, i have to do it in two steps.

I understood why it's AC3/TrueHD: you mean trueHD with ac3 in it, where i thought that both EAC3 and TrueHD have AC3 frames.

[madshi]

Hi madshi, any joy with that weird DTS file?
I'm sorry that it takes so long. I haven't forgotten your weird DTS file. It's still lying around here waiting to be fixed. The problem is that I'm busy with the next bigger eac3to version. Most of eac3to will be totally rewritten. But my time is very limited. So I'd rather delay checking your weird DTS file until the new version is ready. Otherwise I'd had to fix the problem twice - once in the old/current version and once in the new version.

[madshi]

i thought that both EAC3 and TrueHD have AC3 frames.
E-AC3 does not have AC3 frames in it. It's a different format. So you cannot simply extract AC3 frames from E-AC3. Theoretically there is a way to partially decode E-AC3 and reencode as AC3 (which saves CPU time and also improves audio quality a bit), but only Dolby knows how that works. So for us there is no other way than to fully decode E-AC3 to PCM/RAW and then reencode to AC3.

[idbirch2]

No apology necessary, I hadn't been keeping up with the thread as much recently so wasn't aware of the complete re-write, best of luck with this, I'd be lost without eac3to!

[madshi]

wasn't aware of the complete re-write
Well, I didn't talk about it yet... ;)

[Thunderbolt8]

what will be the benefits of the re-writing? less work for yourself with future versions? will it be faster? or require less hardware requirements?

[madshi]

what will be the benefits of the re-writing?
The old eac3to source code began with about 100 lines of code and grew over time to about 3500 lines of code. Often, when a little simple tool grows over time into something much bigger and more complex, the source code mutates into some kind of monster. This is what happened to eac3to. So the main purpose of the rewrite is to clean up the source code. Doesn't sound very impressive, does it?

Well, fortunately, the rewrite will also have several performance and feature improvements. I feel safe to predict that you will not be disappointed! ;)

However, it will take at least 2 weeks until it's ready to go, maybe even longer. I only have a limited time available for work on eac3to.

[nautilus7]

I guess the ac3 decode feature wont be added... Have you done any progress in it?

Looking forward for the new version!

[madshi]

I guess the ac3 decode feature wont be added...
Sooner or later it might get added. But I don't know when.

Have you done any progress in it?
No. I'm investing all my time into the rewrite.

[Thunderbolt8]

However, it will take at least 2 weeks until it's ready to go, maybe even longer. I only have a limited time available for work on eac3to.
doesnt matter whether its 2 weeks or a little longer. looking forward for the improvements :)

[Thunderbolt8]

is it possible that eac3 tracks behave different from trueHD tracks delaywise?
what I mean is that I made a remux from the game HD DVD and kept the original eac3 audio track, because I can play it with sonic decoder 4.3 (even if it does apply drc or norm or what it was, doesnt matter). the track requires no delay at all. the funny thing is just, when I tried to apply the mathematical way and see it this also works for eac3 tracks, which had resulted in -60ms of delay that you could see it everytime (!) when someone speaks in that movie that this delay is too much and that the original state of no delay at all is correct. for other movies its often hard up to very hard to find out whether a movie needs +/- 100ms of delay or more or not. depending on the movie it can be hell. and in this case here now in really every scene it shows that the -60ms is not needed. each time someone opens his mouth it can be seen, whereas you mostly need to look for specific scenes for other movies delays.
is this just the movie or has it something to do with the attributes of eac3 tracks?

[madshi]

is it possible that eac3 tracks behave different from trueHD tracks delaywise?
what I mean is that I made a remux from the game HD DVD and kept the original eac3 audio track, because I can play it with sonic decoder 4.3 (even if it does apply drc or norm or what it was, doesnt matter). the track requires no delay at all. the funny thing is just, when I tried to apply the mathematical way and see it this also works for eac3 tracks, which had resulted in -60ms of delay that you could see it everytime (!) when someone speaks in that movie that this delay is too much and that the original state of no delay at all is correct. for other movies its often hard up to very hard to find out whether a movie needs +/- 100ms of delay or more or not. depending on the movie it can be hell. and in this case here now in really every scene it shows that the -60ms is not needed. each time someone opens his mouth it can be seen, whereas you mostly need to look for specific scenes for other movies delays.
is this just the movie or has it something to do with the attributes of eac3 tracks?
With the "mathematical" way you mean your way of comparing the runtime of the different tracks? I don't think that this is the "right" way to calculate the delay. Maybe it works for most current TrueHD discs, but IMHO that could change sooner or later. The "mathematical" correct would be the just compare the timestamps of the first audio and video packets. For most movies the first eac3 frame and the first video frame have the same timestamp. That means, no delay is needed at all. With many movies TrueHD seems to behave the same - but still it does need delay most of the time!! I don't know yet why this is the case.

[honai]

Probably mastering errors. I remember them from the "olden" days of DVD discs.

Seriously, most friends of mine couldn't even spot a 100ms delay, or care about it.

There are also cases of A-list titles (like Underworld from Concorde) where video is mastered at 23.976fps but audio at 24fps .

Like I said in another thread, most "engineers" at those studios don't really know what they're doing.

[madshi]

There are also cases of A-list titles (like Underworld from Concorde) where video is mastered at 23.976fps but audio at 24fps .
Are you serious!?!? Are you talking about the German Underworld Blu-Ray? Do you mean audio slowly drifts out of sync on that disc?

:eek:

[honai]

Yes, the German HD-DVD (not Blu-ray). Funny enough, it doesn't drift, but if you demux audio you'll notice that it's 7 seconds shorter than the indicated video length in the XPL and demuxed video. 7 seconds = difference between 24fps and 23.976fps for 2 hours. You won't notice the error, though, because basically the timestamps are correct and thus audio frames will be rendered at the right time.

[madshi]

Yes, the German HD-DVD (not Blu-ray). Funny enough, it doesn't drift, but if you demux audio you'll notice that it's 7 seconds shorter than the indicated video length in the XPL and demuxed video. 7 seconds = difference between 24fps and 23.976fps for 2 hours. You won't notice the error, though, because basically the timestamps are correct and thus audio frames will be rendered at the right time.
So if you demux audio and then remux audio and video into MKV audio drifts out of sync then?

[honai]

Well, I've used mkvmerge for remuxing and supplied a 23.976fps timecode file, and the result was in sync. I'm not exactly sure why, I've never fully understood the concept of how mkvmerge syncs v/a streams, but my guess is that either for video or audio some frames are simply skipped or repeated.

[madshi]

That doesn't sound to me like the audio was really mastered at 24.000. Maybe it's simply 7 seconds shorter than the video for whatever reason, but still mastered at 23.976?

[honai]

Nope, it really plays until the end of video.

Maybe I mixed video/audio up, and video is actually 7 seconds shorter than what the XPL and audio length indicate, I will look it up later.

[ACrowley]

Yes, the German HD-DVD (not Blu-ray). Funny enough, it doesn't drift, but if you demux audio you'll notice that it's 7 seconds shorter than the indicated video length in the XPL and demuxed video. 7 seconds = difference between 24fps and 23.976fps for 2 hours. You won't notice the error, though, because basically the timestamps are correct and thus audio frames will be rendered at the right time.


No honai...So far i know the HDDVD is "surely" 23.976FPs. Video and Audio

Maybe you mean the demuxed dtscore ?
Remember the dtscore is not sync (demuxed outside Container ) and with incorrect Runtime with the standard dts_ac3 Source.ax Filter.
And the difference was ~ 7sec . That would exactly explain your 7 sec difference on the demuxed dtscore :)

With Madshis Source Filter Mod its Ok..the "demuxed" dtscore plays at correct Runtime and its sync
So use Madshis modified DTS_AC3_DD+ Source Filter and take a new look, im pretty sure its 7 sec longer

[The_Keymaker]

Hello forum members,

I have released a new version of EAC3toGUI (v1.38) for your use.

Link: http://www.sendspace.com/file/v0fg69

Changes to the new version include:

1. - Added Horizontal scroll to destination File
2. - Added *.ac3 source file extension
3. - Source path now "remembered" after program closes

Remember, as with previous version of EAC3toGUI, you should place the program in same directory as eac3to.exe. If you decide to place EAC3toGUI in a different directory, you should specify the location of eac3to.exe using the settings menu option.

Please report any bugs and feel free to post comments.

@madshi, please upload the updated program to your link on page 1.

Regards,
The_Keymaker

[Nikos]

So far i know the HDDVD subtitles and chapters points is "surely" 24 fps.
Any explain?

[honai]

Ok, I looked it up again.

The video stream on the disc says runtime is 02:13:25:00.

<Title id="mainMovie" titleNumber="4" titleDuration="02:13:25:00" onEnd="black" displayName="Underworld" description="mainMovie">

The runtime of the extracted audio stream is 02:13:25:00.

The runtime of the extracted video stream at 23.976fps is 02:13:32:00.

The runtime of the remuxed MKV with timecode file at 23.976fps is 02:13:32:00.

Obviously something is wrong here, don't you think? ;)

[Chumbo]

Hello forum members,

I have released a new version of EAC3toGUI (v1.38) for your use.

Link: http://www.sendspace.com/file/v0fg69

Changes to the new version include:

1. - Added Horizontal scroll to destination File
2. - Added *.ac3 source file extension
3. - Source path now "remembered" after program closes

Remember, as with previous version of EAC3toGUI, you should place the program in same directory as eac3to.exe. If you decide to place EAC3toGUI in a different directory, you should specify the location of eac3to.exe using the settings menu option.

Please report any bugs and feel free to post comments.

@madshi, please upload the updated program to your link on page 1.

Regards,
The_Keymaker
Thank you for the changes. Here are my reports/comments:
1. - Added Horizontal scroll to destination File
Verified that it works. :)

2. - Added *.ac3 source file extension
Verified that it works. :)

3. - Source path now "remembered" after program closes
This one not working as expected even though you went beyond what I was asking. The "remembered" I was referring to is to just remember where the last folder was when you hit the Browse button, i.e., it would take you back to the last location you loaded a file from. Although what you came up with is even better, however, it's kinda buggy right now.

When I launched it the first time, it entered the path plus eac3to.exe, i.e., "d:\avutils\eac3to.exe" which immediately caused the invalid file message to pop up. Once I loaded a file successfully, closed it down and reopened it and it did fill the fields, however, when I clicked Browse for the source, it still took me to the eac3to folder rather than the last location. When I clicked Cancel on the "Select the Source File" dialog, it then removed the Source file entry. You should be able to duplicate this, I think.

It seems to do this only after you initially open the app. Once you Browse and select a source file, you can open the Browse again and hit Cancel and it will not affect the source.

I hope I've explained this well and not confused everyone. :cool: Please let me know if you need any clarification. Thanks again.

[ACrowley]

Ok, I looked it up again.

The video stream on the disc says runtime is 02:13:25:00.



The runtime of the extracted audio stream is 02:13:25:00.

The runtime of the extracted video stream at 23.976fps is 02:13:32:00.

The runtime of the remuxed MKV with timecode file at 23.976fps is 02:13:32:00.

Obviously something is wrong here, don't you think? ;)


What i know is , the untouched 23.976 HDDVD should have 133min , (2h:13m:25s):)

But strange, i know a few x264 mkv reencodes , and they all have different Runtime

I have the HDDVD here and i will tak a closer Look into it

[Thunderbolt8]

The "mathematical" correct would be the just compare the timestamps of the first audio and video packets.
I would like to try this out with future remuxes, could you please give me a little instruction how I can do that, with remuxed .mkv video and flac/.mka audio? please take into consideration that I dont really have a clue of such progs and steps I have to do :P
if I encouter a timestamp difference between video and audio, how can I calculate that back to ms of delay?
thanks!

[The_Keymaker]

@Chumbo,

The issue you encountered will only happen the FIRST TIME time you open v1.38 of EAC3toGUI. Just hit "OK" when the "unsupported filetype" dialog pops up and continue as normal. Everything will be fine after that.

Thanks for the feedback.

Regards,
The_Keymaker

[madshi]

I would like to try this out with future remuxes, could you please give me a little instruction how I can do that, with remuxed .mkv video and flac/.mka audio? please take into consideration that I dont really have a clue of such progs and steps I have to do :P
if I encouter a timestamp difference between video and audio, how can I calculate that back to ms of delay?
thanks!
Check out posts 929 und 930 of this thread.

[The_Keymaker]

New version (v1.39) uploaded which fixes the bug noted by Chumbo.

Link: http://www.sendspace.com/file/423w9l

Change log v1.39:

1 - Fixes default source file from a previous EAC3toGUI session.

Please try this new version and report any bugs.

Thanks
The_Keymaker

[Thunderbolt8]

Check out posts 929 und 930 of this thread.
the problem is that this only accounts for the rebuilt source video and audio evos, but not for the remuxed and demuxed versions in .mkv or flac/mka in the end. we (or better said you :P ) already presumed the sync problem occurs when the streams are dragged out of their native state in the container. so I guess this wont help me much. as far as I can remember all my last movies had the same identical first pts/ptm values for audio and video in evodemux (regardless of eac3, truehd or dts-hd), but some of them needed a delay, some didnt.

apart from that this would only account for HD DVD anyway, as we only have a HD DVD program that gives us such info as evodemux. what could I do in case of blu-ray? (well, xport gives info about the first video pts when demuxing video (and audio), but no info about audio, so it doenst help me much)

[Chumbo]

New version (v1.39) uploaded which fixes the bug noted by Chumbo.

Link: http://www.sendspace.com/file/423w9l

Change log v1.39:

1 - Fixes default source file from a previous EAC3toGUI session.

Please try this new version and report any bugs.

Thanks
The_Keymaker
Cool, thanks. Here are my findings:
- deleted ini file and brought it up. Came up fine w/out filling in the Source file and thus not causing the error to come up.
- clicked source browse and selected a source file then closed the app
- reopened the app with no issues
- clicked the source Browse and, woohoo, it opened the last location navigated to. I selected my source file and clicked Open with no issues.
- Immediately after I clicked my Source browse again, then I clicked Cancel which brought up the Unsupported Filetype dialog. When I clicked OK, it cleared out the Source and Destination. It also did not clear the Command Line Preview correctly as it kept the destination string even though the Destination box was cleared out.

Almost there. :)

[madshi]

the problem is that this only accounts for the rebuilt source video and audio evos, but not for the remuxed and demuxed versions in .mkv or flac/mka in the end.
Well, if you legally own the movies then there's no problem checking the timestamps with the original EVO files, right? :)

we (or better said you :P ) already presumed the sync problem occurs when the streams are dragged out of their native state in the container. so I guess this wont help me much.
Of course it will, cause the first timestamps of the different video/audio tracks are the single most important facts when thinking about audio delay. If I ever find a correct way to calculate the proper audio delay of all audio tracks then it MUST involve the pts/ptm values for the audio and video tracks. There's no other way to get it right in all situations.

as far as I can remember all my last movies had the same identical first pts/ptm values for audio and video in evodemux
I have several movies where this is not the case.

(regardless of eac3, truehd or dts-hd), but some of them needed a delay, some didnt.
Normally if the first pts/ptm values are identical, no track should need a delay. The big question is why the TrueHD tracks still do sometimes. But that doesn't mean that the pts/ptm values are in any way useless. They're extremely important. We just need to solve the mystery about why the TrueHD tracks need a delay when they really shouldn't. There must be a reason and we'll probably find it sooner or later...

apart from that this would only account for HD DVD anyway, as we only have a HD DVD program that gives us such info as evodemux. what could I do in case of blu-ray? (well, xport gives info about the first video pts when demuxing video (and audio), but no info about audio, so it doenst help me much)
So what do you want me to do? Should I write a tool which analyzes the video images with artificial intelligence to find moving mouths and then analyze the audio data to find spoken words in them which match the lip movements?

There's no way to "automatically" find the right delay value if you have no timestamp information and if the tracks are out of sync. Period. Sometimes ugly tricks like checking the length of the audio/video track might help (what you're doing), but it's not really a good solution and it will not always work.

[Thunderbolt8]

Well, if you legally own the movies then there's no problem checking the timestamps with the original EVO files, right? :)
thats not what I mean. of course I have the original evo files, what I mean is that when I compare them of course they are mostly in sync, audio and video there. but when de- and remuxing the tracks something obviously changes and then the original values only help me that much as it indicates the tracks on the original disc begin completely in sync, so when I can establish this state with the remuxed audio and video too then they should be in sync as well.
but as you already said with comparing the length it might just not be the right way each time thats why I need something like a way to compare the pts or something equal in their remuxed counterparts, like the point when the first real audio and video data occurs in those files. this should be a bit different in those cases where movies still need a delay even though, according to timestamps, it shouldnt.

im basically asking for a way to compare the final streams that will go into the mux, not the source files. maybe its possible to spot differences there with a hex editor or something like that.

[madshi]

im basically asking for a way to compare the final streams that will go into the mux, not the source files. maybe its possible to spot differences there with a hex editor or something like that.
I'm not sure why we keep on arguing about this. There's only one proper way to calculate the audio delays automatically. I see no sense in talking about any other solution.

Yes, the only proper way currently fails to work with TrueHD streams. But instead of trying to find other proper solutions (which don't exist) we should really try to find out *why* the only proper way fails to work with TrueHD streams.

[Thunderbolt8]

it was only an idea. I didnt know that having a look on the files which will go into the mux wont work so I had to ask.

btw. what was again with the DTS-HD streams? did they all generally need a delay or not? are there differences between MA and HR ?
ive already told that all the streams Ive had so far with remuxes didnt need delay, regardless of any numbers. but then again I noticed those were all studio canal titles so far and I hope theres no difference between tracks from that studio or tracks from other studios or blu-ray for example.

[madshi]

The method I explained in post 930 should work for any and all audio streams, doesn't matter which type they are.

(Obviously TrueHD makes problems, but that must have a very specific reason.)

[honai]

It's probably because TrueHD defines alignment block sizes, and when audio and video are not aligned to a specific block there is an implicit offset calculated from some schema. Unfortunately I don't have the specs.

Maybe Alex Zambelli or Ben Waggoner can help, they seem to have a lot of experience with HD disc container formats and buffer sizes from the specs.

[madshi]

It's probably because TrueHD defines alignment block sizes, and when audio and video are not aligned to a specific block there is an implicit offset calculated from some schema. Unfortunately I don't have the specs.

Maybe Alex Zambelli or Ben Waggoner can help, they seem to have a lot of experience with HD disc container formats and buffer sizes from the specs.
That does make some sense. I'll see of those two are willing to help. Thanks for the suggestion.

[The_Keymaker]

@Chumbo (and others using EAC3toGUI),

I believe this version (v1.40) should finally fix any lingering source file bugs. It certainly resolves the bug reported by Chumbo.

Link: http://www.sendspace.com/file/hzdz90

Please test and report any bugs.

I apologize for any inconvenience these bugs may cause and I appreciate everyone's patience.

Regards,
The_Keymaker

[Thunderbolt8]

The method I explained in post 930 should work for any and all audio streams, doesn't matter which type they are.

(Obviously TrueHD makes problems, but that must have a very specific reason.)
yes i know that, but then noticed that all the dts-hd titles I had so far were all from studio canal and then I suddenly was afraid it could be only due to that :P

but I recently remuxed the fly BD, which has DTS-HD MA, and it also looked fine without any delay at all there.

[Chumbo]

@Chumbo (and others using EAC3toGUI),

I believe this version (v1.40) should finally fix any lingering source file bugs. It certainly resolves the bug reported by Chumbo.

Link: http://www.sendspace.com/file/hzdz90

Please test and report any bugs.

I apologize for any inconvenience these bugs may cause and I appreciate everyone's patience.

Regards,
The_Keymaker
No need to apologize. Best way to get the bugs out is by people actually using this stuff and giving feedback. ;)

Confirmed that the last bug I reported is now fixed. So thank you.

If I may ask for a couple "wish list" items at your convenience:
- allow the Source entry to be edited. It looks like, currently, the edit box is read-only
- a queue or batch type feature where jobs can be added and then executed as a batch

Again, thanks for your hard work on this.

[The_Keymaker]

@madshi,

If you desire, please post the latest version (v1.40) of EAC3toGUI to your link on the first page.

Link to latest version: http://www.sendspace.com/file/hzdz90

Regards,
The_Keymaker

[Chumbo]

@The_Keymaker,
Small "bug" to report. Under the Conversion Options tab, "TrueHD" is misspelled in "PCM and TreuHD Options."

There also seems to be a bug in the repainting of parts of the window. For example, drag half of the app off the screen and then drag it back. You'll notice that it doesn't repaint correctly. It only seems to happen off the left side of the screen though.

http://img215.imageshack.us/img215/4126/paintissueik2.th.jpg (http://img215.imageshack.us/my.php?image=paintissueik2.jpg)

[The_Keymaker]

Will fix misspelling AND input text box-edit next release.

I can not duplicate the repainting problem on either my Vista32 system or my legacy XP system. It seems to be system specific.

Thanks for the feedback.

Regards,
The_Keymaker

[madshi]

@madshi,

If you desire, please post the latest version (v1.40) of EAC3toGUI to your link on the first page.
Done - thanks!

[hgmeier]

Hi all,

I am new in this forum and I have a question. After I have demuxed an HD-DVD with Evodemux an converted the DDPlus track to DD by using eac3to. Now I would like to put the vdeo- and audiostream back together. Which program do you recommend for that purpose?
It would be very nice if someone could make a suggestion.

Thanks,

Chris

[madshi]

I am new in this forum and I have a question. After I have demuxed an HD-DVD with Evodemux an converted the DDPlus track to DD by using eac3to. Now I would like to put the vdeo- and audiostream back together. Which program do you recommend for that purpose?
It would be very nice if someone could make a suggestion.
This is not really the right thread to discuss this as this thread is about eac3to (audio converting) only. Please search through the other threads. If you can't find a solution please post a new thread and ask there. Thanks.

[Sephiroth0000]

This forum has helped me out before so im back with more problems lol. I have a DDP audio file from EVODemux and when I run it through EAC3TO it does come up at the full time it should (that being 2hours23minutes) but yet when it converts it to WAV it is only coming out at 1hour6minutes long and yet the WAV file size is massive (6GB to be exact) I have never had this problem before and have tried numerous times now. Any suggestions anyone?

[madshi]

This forum has helped me out before so im back with more problems lol. I have a DDP audio file from EVODemux and when I run it through EAC3TO it does come up at the full time it should (that being 2hours23minutes) but yet when it converts it to WAV it is only coming out at 1hour6minutes long and yet the WAV file size is massive (6GB to be exact) I have never had this problem before and have tried numerous times now. Any suggestions anyone?
There are problems with WAV files above 2GB or 4GB. There's a "file size" field in the WAV file header which cannot hold more than 4GB. As a result this field "overflows" for too big WAV files. Some audio tools work around that by ignoring this header field if the WAV file is bigger than 2GB or 4GB. Others don't work around the problem. What you're seeing is most probably a WAV interpreter which doesn't handle the situation well. The WAV file is probably alright. It's just the tool or DirectShow filter which you're using which shows a wrong runtime.

[Sephiroth0000]

There are problems with WAV files above 2GB or 4GB. There's a "file size" field in the WAV file header which cannot hold more than 4GB. As a result this field "overflows" for too big WAV files. Some audio tools work around that by ignoring this header field if the WAV file is bigger than 2GB or 4GB. Others don't work around the problem. What you're seeing is most probably a WAV interpreter which doesn't handle the situation well. The WAV file is probably alright. It's just the tool or DirectShow filter which you're using which shows a wrong runtime.

MADSHI as always you are the man of wisodm but I have played the file and it only does let me play the 1 hour and 6 minutes worth of audio. Aftet that the file finishes. Are you saying when I actually apply it to the video it will be ok? How do I correct this problem?

[tebasuna51]

These wav's >4 GB can be encoded to ac3 with Aften (need the parameter: -readtoeof 1) or to aac with NeroAacEnc (parameter: -ignorelength).

You can also split in monowav's with WaveWizard and use others commercial encoders than accept monowav's.

[Sephiroth0000]

These wav's >4 GB can be encoded to ac3 with Aften (need the parameter: -readtoeof 1) or to aac with NeroAacEnc (parameter: -ignorelength).

You can also split in monowav's with WaveWizard and use others commercial encoders than accept monowav's.

Ironically enough that is what I am trying to do (making the WAV into AC3) however I am not sure on how to do it. I mean normally I do my HD DVD's into WMV with 5.1 which is easy but beyond that I do not really know a great deal. Could you go into more detail for me at all please? I do appreciate it :)

[madshi]

Ironically enough that is what I am trying to do (making the WAV into AC3)
Then why did you ask eac3to to give you a WAV file in the first place? Just use "eac3to source.ddp dest.ac3" and you're done.

[Sephiroth0000]

Then why did you ask eac3to to give you a WAV file in the first place? Just use "eac3to source.ddp dest.ac3" and you're done.

I did not ask it to make me a WAV file. I gave it the input file (ddp file) with an output file beaing a ac3 extensions and it comes up with some error about AC3 failing for some reason (not at computer so cannot state) but a WAV file was created successfully. I normally only have ever done WAV files before but this is stumping me bigtime. With regards to that command you put in your quote how would I put that in the GUI of your EAC3TO?

[DVDCake]

New at this, I'm getting "Getting Dump instance failure" after the temp file is created.

Any suggestions?

~DC

[Chumbo]

New at this, I'm getting "Getting Dump instance failure" after the temp file is created.

Any suggestions?

~DC
Did you bother to use the Search? I think this question's been asked a hundred times on this forum.

[Chumbo]

...With regards to that command you put in your quote how would I put that in the GUI of your EAC3TO?
OMG, I hope you're kidding...

[Sephiroth0000]

New at this, I'm getting "Getting Dump instance failure" after the temp file is created.

Any suggestions?

~DC

You are going to have to install the DUMP.AX filter in your Windows SYSTEM32 folder and then register it via COMMAND PROMPT with the regsvr32 command (please note that should you be running Windows Vista please run COMMAND PROMPT as Administrator) You can get the dump.ax filter with the program GRAPHEDIT which is a free download.

[Sephiroth0000]

OMG, I hope you're kidding...


I wish I was Chumbo but unfortunately not. I am still new to this and was only successful with doing EVO to WMV with 5.1 because of some of the geniuses and their input from this exact thread for who I thankyou very much (Afterwards I wrote up the guide and made it simplier covering every problem I had when attepting EVO to WMV HD with 5.1). Im not going to pretend to be a mastermind with this as im not. I am making everything I do available to everyone else so should anyone else get stuck they won't be anymore.

The more we all share our information the faster we move along as a whole.

[Chumbo]

I wish I was Chumbo but unfortunately not. I am still new to this and was only successful with doing EVO to WMV with 5.1 because of some of the geniuses and their input from this exact thread for who I thankyou very much (Afterwards I wrote up the guide and made it simplier covering every problem I had when attepting EVO to WMV HD with 5.1). Im not going to pretend to be a mastermind with this as im not. I am making everything I do available to everyone else so should anyone else get stuck they won't be anymore.

The more we all share our information the faster we move along as a whole.
Did you note what part I was responding to? If you use eac3togui, all you have to do is open the file and voila! It even shows you the command line it constructs. That's why I only included that part of the quote. Otherwise, what GUI are you talking about?

Thanks for your contributions. :)

[madshi]

I did not ask it to make me a WAV file. I gave it the input file (ddp file) with an output file beaing a ac3 extensions and it comes up with some error about AC3 failing for some reason (not at computer so cannot state)
Then please post the error text here so we can find out why AC3 encoding failed.

[Thunderbolt8]

is it normal that eac3to sometimes cant detect the bitdepth of a pcm track and therefore we need to put "-24" there manually? or is this considered to be a bug?

[ACrowley]

is it normal that eac3to sometimes cant detect the bitdepth of a pcm track and therefore we need to put "-24" there manually? or is this considered to be a bug?

tell eac3to the Bitdepth manuall and youve no Problmes

I Think Madshi can tell you why

[madshi]

is it normal that eac3to sometimes cant detect the bitdepth of a pcm track and therefore we need to put "-24" there manually? or is this considered to be a bug?
It is kind of normal. The current implementation just guesses with a rather simple algorithm. The algorithm only succeeds if it's really sure which bitdepth is the correct one to avoid a false classification.

The next eac3to version will have a much more sophisticated guessing algorithm. It will not only be able to detect 8bit, 16bit and 24bit bitdepths reliably, but it will also automatically detect number of channels (anything between 1 - 8) and big/little endian. It will even detect if the PCM/RAW file is really PCM/RAW or not.

Oooops, there goes my plan to not reveal any details about the next eac3to version yet... :eek: Fortunately there are still some improvements in store that I haven't mentioned yet... ;)

[Thunderbolt8]

tell eac3to the Bitdepth manuall and youve no Problmes

I Think Madshi can tell you why
I already did that of course, I just wanted to know if this happening is considered to be normal in the current version and whether he might need a sample from that file :P

[honai]

The next eac3to version will have a much more sophisticated guessing algorithm. It will not only be able to detect 8bit, 16bit and 24bit bitdepths reliably, but it will also automatically detect number of channels (anything between 1 - 8) and big/little endian. It will even detect if the PCM/RAW file is really PCM/RAW or not.

Oooops, there goes my plan to not reveal any details about the next eac3to version yet... Fortunately there are still some improvements in store that I haven't mentioned yet...

That and the previous announcements sound like quite a large feature set. Have you considered putting up a donation link to support your work? I'd be more than happy to do it.

[madshi]

That and the previous announcements sound like quite a large feature set. Have you considered putting up a donation link to support your work? I'd be more than happy to do it.
Thanks, I'll consider it... :D

[madshi]

Good news: Found the reason why the TrueHD decoded files often need a delay. The problem should be fixed in the next eac3to version.

[nautilus7]

YEAAAAAAAAAAAAAAAAAAAAAH

:eek::eek::eek::eek::eek::eek::eek::eek::eek::eek::eek::eek::eek:

Very good news indeed!

[Thunderbolt8]

Good news: Found the reason why the TrueHD decoded files often need a delay. The problem should be fixed in the next eac3to version.
this is really great. :thanks:
ive saved the original trueHD tracks in rebuilt .evos from the movies I remuxed & converted the sound to flac. can you already tell if I demux the TrueHD track again from that .evo and then use the new eac3to version (when its done), then the delay issue will be fixed automatically? or would I need the complete evos which include both, original audio and video, for that?

[madshi]

this is really great. :thanks:
ive saved the original trueHD tracks in rebuilt .evos from the movies I remuxed & converted the sound to flac. can you already tell if I demux the TrueHD track again from that .evo and then use the new eac3to version (when its done), then the delay issue will be fixed automatically? or would I need the complete evos which include both, original audio and video, for that?
It's how I explained earlier: If the first timestamps of the audio and video tracks are identical, you should not need any delay, but audio sync should be perfect automatically. Basically with the next eac3to version TrueHD files should behave the same way as E-AC3 files do with the current version. If however you have some movies where the first timestamp of the TrueHD track differs from the first timestamp of the video track, you will need to apply a delay. This delay can be directly seen in EvoDemux. I think the majority of the HD DVD movies will probably need no delay at all. But there are some movies which do need a delay. EvoDemux should tell us that.

[Thunderbolt8]

darn, should have saved the timestamp infos, too :P

[shambles]

maybe this should be in the madflac thread as it could just be a madflac bug but..

i converted a 7.1 blu-ray lpcm track to 7.1 flac. when playing it with madflac with the 'limit to 5.1' option on, the bl/br channels are used instead of sl/sr. when converting the 8ch blu-ray pcm to 8ch wave or flac, the channel order used seems to be l,r,c,lfe,bl,br,sl,sr. using -down6 results in sl/sr being used correctly.

is it eac3to mapping the channels wrong or madflac using/scrapping the wrong channels?

[madshi]

maybe this should be in the madflac thread as it could just be a madflac bug but..

i converted a 7.1 blu-ray lpcm track to 7.1 flac. when playing it with madflac with the 'limit to 5.1' option on, the bl/br channels are used instead of sl/sr. when converting the 8ch blu-ray pcm to 8ch wave or flac, the channel order used seems to be l,r,c,lfe,bl,br,sl,sr. using -down6 results in sl/sr being used correctly.

is it eac3to mapping the channels wrong or madflac using/scrapping the wrong channels?
Good catch! It's a bug in madFlac. Will be fixed with the next build.

[shambles]

holy fast reply, batman!

looking forward to new builds of both madflac and eac3to then :D

if i could request a feature for eac3to, i'd love to be able to specify 2 (or more) input files and have eac3to join them while decoding/converting. it's a pain in the ass with blu-ray having the movie spread over several m2ts's..

if you could just do 'eac3to input1.pcm+input2.pcm output.flac', it'd make the whole process much more pleasant ;)

[madshi]

if i could request a feature for eac3to, i'd love to be able to specify 2 (or more) input files and have eac3to join them while decoding/converting. it's a pain in the ass with blu-ray having the movie spread over several m2ts's..

if you could just do 'eac3to input1.pcm+input2.pcm output.flac', it'd make the whole process much more pleasant ;)
Hmmmm... I'm not 100% sure if simply joining the audio files is the mathematically correct way to handle this. I'm not sure how m2ts files are joined technically. I think simple joining would do for HD DVD audio. But I'm not sure about Blu-Ray. Probably it would work, but maybe someone with more m2ts knowledge can confirm?

[Inventive Software]

holy fast reply, batman!

looking forward to new builds of both madflac and eac3to then :D

if i could request a feature for eac3to, i'd love to be able to specify 2 (or more) input files and have eac3to join them while decoding/converting. it's a pain in the ass with blu-ray having the movie spread over several m2ts's..

if you could just do 'eac3to input1.pcm+input2.pcm output.flac', it'd make the whole process much more pleasant ;)

I believe that simply joining the TS files will suffice, from what I've read.

[Thunderbolt8]

when I understand it correctly the way the trueHD delay issue in cases where there shouldnt be any due to correct timestamps will be fixed only for the flac files, because eac3to will correct that problem, right? what about the demuxed trueHD tracks then, they will remain with this issue? wouldnt it be possible to talk with pelican and maybe try to implement it in evodemux, so that both, the normal trueHD tracks and the resulting flac tracks, will be fine?

[shambles]

I believe that simply joining the TS files will suffice, from what I've read.

but joining them takes a lot of time and disc space.. implementing the correct way to join the audio tracks to eac3to would make working with blu-ray audio so much easier.

[madshi]

when I understand it correctly the way the trueHD delay issue in cases where there shouldnt be any due to correct timestamps will be fixed only for the flac files, because eac3to will correct that problem, right? what about the demuxed trueHD tracks then, they will remain with this issue? wouldnt it be possible to talk with pelican and maybe try to implement it in evodemux, so that both, the normal trueHD tracks and the resulting flac tracks, will be fine?
There never was a problem with the demuxed TrueHD files. The problem was in how eac3to muxed those TrueHD files into a temporary EVO container. So no need to change anything in EvoDemux.

[madshi]

but joining them takes a lot of time and disc space.. implementing the correct way to join the audio tracks to eac3to would make working with blu-ray audio so much easier.
Well, I guess I could add an audio join feature. If it doesn't work then it doesn't work. But maybe it does.

[Thunderbolt8]

There never was a problem with the demuxed TrueHD files. The problem was in how eac3to muxed those TrueHD files into a temporary EVO container. So no need to change anything in EvoDemux.
oh. so you say that basically all demuxed trueHD tracks (with same video & audio starting pts) are fine and only the conversion to flac with current eac3to versions caused that problem?

[madshi]

oh. so you say that basically all demuxed trueHD tracks (with same video & audio starting pts) are fine and only the conversion [...] with current eac3to versions caused that problem?
Yes.

only the conversion to flac with current eac3to versions caused that problem?
The problem has nothing to do with FLAC. The bug is in eac3to's decoding of TrueHD. If you do "TrueHD -> AC3" with the current eac3to version, you'll have the same delay problems you're having with "TrueHD -> FLAC".

[nautilus7]

Just curious... Can you explain what was wrong with the remuxed .evo that cause the delay problem (if it's not too complicated to explain)?

[honai]

Probably just that the offset of the source EVO was not applied to the temp EVO that eac3to created when removing Dial.Norm.

[madshi]

eac3to searched for the first occurrence of a TrueHD header in the TrueHD stream and removed all bytes before the first header. I thought those bytes were garbage. But now I found out that if I don't remove those bytes, the decoded audio data is longer. Tested it with one sample TrueHD file. Got additional 106ms worth of audio data in the beginning of the file. I don't fully understand what those "garbage" bytes are for, but in the end it doesn't really matter. If I stop deleting those bytes, the delay problems should be gone.

[nautilus7]

I see. Thanks.

[killa_kid]

eac3to searched for the first occurrence of a TrueHD header in the TrueHD stream and removed all bytes before the first header. I thought those bytes were garbage. But now I found out that if I don't remove those bytes, the decoded audio data is longer. Tested it with one sample TrueHD file. Got additional 106ms worth of audio data in the beginning of the file. I don't fully understand what those "garbage" bytes are for, but in the end it doesn't really matter. If I stop deleting those bytes, the delay problems should be gone.

i just ripped The Departed from HD DVD which has TrueHD audio and it was approx 100ms out of sync, so I'm guessing its true or most files. Batman Begins didn't seem to have much of a problem though. I am currently doing Blood Diamond and lets see if it happens this time.

And thanks madshi, the tool is fantastic and we'd all be screwed without it!

[honai]

so I'm guessing its true or most files

You are guessing wrong.

[Murleen]

eac3to searched for the first occurrence of a TrueHD header in the TrueHD stream and removed all bytes before the first header. I thought those bytes were garbage. But now I found out that if I don't remove those bytes, the decoded audio data is longer. Tested it with one sample TrueHD file. Got additional 106ms worth of audio data in the beginning of the file. I don't fully understand what those "garbage" bytes are for, but in the end it doesn't really matter. If I stop deleting those bytes, the delay problems should be gone.

How many bytes were you seeing before the F8726FBA sync word?

[killa_kid]

You are guessing wrong.

I mistyped that, I meant to say some files, not most.

[madshi]

How many bytes were you seeing before the F8726FBA sync word?
That differs between tracks. However, it seems that the 4 bytes directly before the sync dword are important (for whatever reason). If I remove only those 4 bytes, I'm already losing about 100ms worth of audio data. Seemingly the audio decoder wants to have those 4 bytes and skips the whole frame if those bytes are not there. I'm not sure, maybe those 4 bytes in front of the sync dword are all that was missing. But sometimes there's more "garbage" (or not garbage) in front of the first sync dword. Not sure whether those bytes have any meaning. Maybe, maybe not...

[Thunderbolt8]

could there actually be a problem in the future, when I remove dialnorm from eac3 tracks, which I intend to use further? what I mean is could something happen like for example that some hardware decoders would try to remove it at playback, but since its already removed, it would result in errors, or wouldnt play at all or some other loss of quality? so would it be best to keep that track 100% in its original state or could the removal of dialnorm via eac3to not harm the tracks quality at all somehow?


btw. are dialnorm and DRC actually also applied for normal AC3 and DTS tracks (not eac3 and DTS-HD) from DVDs (DRC was restricted to dolby only?) ? if yes, is it actually possible then also to improve all those normal DVD audio tracks and get rid of those 2 things by converting these tracks to FLAC ?

[honai]

could there actually be a problem in the future, when I remove dialnorm from eac3 tracks, which I intend to use further?

I admire your ability to invent problems and post fictitous scenarios in this thread.

what I mean is could something happen like for example that some hardware decoders would try to remove it at playback, but since its already removed, it would result in errors, or wouldnt play at all or some other loss of quality?

Answer: No.

[madshi]

could there actually be a problem in the future, when I remove dialnorm from eac3 tracks, which I intend to use further? what I mean is could something happen like for example that some hardware decoders would try to remove it at playback, but since its already removed, it would result in errors, or wouldnt play at all or some other loss of quality? so would it be best to keep that track 100% in its original state or could the removal of dialnorm via eac3to not harm the tracks quality at all somehow?
The studio can decide which dialnorm value to use. It's perfectly legit for a studio to encode an (E)-AC3 or TrueHD track with dialnorm set to "off". So no decoder can possible stumble over such tracks. Removing dialnorm does not harm audio quality any bit. Not now and not in the future. So there's no need to keep the original track. The only thing you'll lose by removing dialnorm is ... the dialnorm feature.

Btw, Sony has dialnorm turned off on all their newer Blu-Ray discs.

btw. are dialnorm and DRC actually also applied for normal AC3 and DTS tracks (not eac3 and DTS-HD) from DVDs (DRC was restricted to dolby only?) ? if yes, is it actually possible then also to improve all those normal DVD audio tracks and get rid of those 2 things by converting these tracks to FLAC ?
AC3: Most tracks have dialnorm activated. The reference decoders (Nero, Sonic) apply both dialnorm and DRC on those tracks. AC3Filter applies neither. So when using AC3Filter you should be fine.
DTS: Most tracks do not have dialnorm activated. Having dialnorm "off" seems to be the default value in the DTS encoder. No decoder that I know applies DRC on DTS decoding - unless you specifically ask it to. So DTS seems to be rather problem free in this aspect.
TrueHD: Most tracks have dialnorm activated (with the exception of those coming from Sony). The reference decoders (Nero, Sonic) apply dialnorm on those tracks. Theoretically they should also apply DRC. But only Sonic does that. Nero doesn't - for whatever reason.

What I said above about E-AC3 dialnorm removal is also valid for all other formats: Turning dialnorm off doesn't harm audio quality in any way.

Converting AC3 tracks to FLAC to avoid DRC and dialnorm is surely possible. But if you playback through the PC, you don't really need to do that. Just use AC3Filter and you're done. Of course AC3Filter doesn't use the Dolby reference AC3 decoder code, though. So it's possible that maybe the reference decoders could have slightly better audio quality (don't know). But when using the reference decoders we're back with at least DRC.

[nautilus7]

I am trying to convert an E-AC3 track (from Smokin Aces HD DVD) to AC3, but i get the following error message. What does it mean?

C:\Tools>eac3to aces.eac3 aces.ac3
E-AC3, 5.1 channels, 1:48:56, 1536kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
This track is not clean. Processing aborted.
Please clean the track with delaycut and then retry eac3to.

EDIT: I run delaycut in ignore mode. Here's the log:

====== INPUT FILE INFO ========================
File is eac3
Bitrate (kbit/s) 1536
Act rate (kbit/s) 1536.000
File size (bytes) 1254904832
Channels mode 3/2: L+C+R+SL+SR
Sampling Frec 48000
Low Frec Effects LFE: Present
Duration 01:48:55.962
Frame length (ms) 5.333333
Frames/second 187.500000
Num of frames 1225493
Bytes per Frame 1024.0000
Size % Framesize 0
CRC present: YES
=============================================
====== TARGET FILE INFO ======================
Start Frame 0
End Frame 1225492
Num of Frames 1225493
Duration 01:48:55.962
NotFixedDelay 0.0000
=============================================
====== PROCESSING LOG ======================
Time 00:20:39.722; Frame#= 232449. Unsynchronized frame...SKIPPED 1024 bytes. Found new synch word
Time 01:06:05.269; Frame#= 743489. Unsynchronized frame...SKIPPED 1024 bytes. Found new synch word

[honai]

Probably CRC errors. Delaycut has an option to fix those.

[Murleen]

That differs between tracks. However, it seems that the 4 bytes directly before the sync dword are important (for whatever reason). If I remove only those 4 bytes, I'm already losing about 100ms worth of audio data. Seemingly the audio decoder wants to have those 4 bytes and skips the whole frame if those bytes are not there. I'm not sure, maybe those 4 bytes in front of the sync dword are all that was missing. But sometimes there's more "garbage" (or not garbage) in front of the first sync dword. Not sure whether those bytes have any meaning. Maybe, maybe not...

The 4 bytes beforehand are actually part of the same frame - the first two bytes indicate the frame length, the next two are a form of DTS - these are then followed by the sync word. Without the bytes, 128 packets of 40 samples will be dropped - about 107ms by my reckoning.

If you've got any samples with extra junk on the front I'd like to get a look at them - PM me if that's possible.

[madshi]

The 4 bytes beforehand are actually part of the same frame - the first two bytes indicate the frame length, the next two are a form of DTS - these are then followed by the sync word. Without the bytes, 128 packets of 40 samples will be dropped - about 107ms by my reckoning.
That's interesting - thanks for the information!

Is it possible to get the runtime from those DTS bytes? I mean could I just search in the file for the first and last sync word, substract the DTS values and convert them to a runtime somehow? That would be nice.

If you've got any samples with extra junk on the front I'd like to get a look at them - PM me if that's possible.
Right now I don't have any at hand. But I'm sure I'll run across them again sooner or later. Will PM you then.

[shambles]

it seems weird that .raw is not a valid input extension since eac3to decodes to just that. renaming it to .pcm works but then eac3to seems to assume it's a blu-ray lpcm track and wants to remap the channels.

i wanted to use -down6 on a raw file that was decoded from flac, but i just couldn't get it to work..

'eac3to bla.pcm bla.flac -8 -down6' makes a 6ch flac but channels are mapped wrong

'eac3to bla.pcm bla.flac -8 -0,1,2,3,4,5,6,7 -down6' makes an 8ch flac

flac to flac with -down6 doesn't work either.. 'eac3to bla.flac bla2.flac -down6' creates an 8ch file

it of course works just fine with the orig blu-ray lpcm track but not anymore once the channels have been remapped.

also .wav input seems to be broken.. even wavs made by eac3to give the error message "The raw file has 0 channels, which is not supported by eac3to."

[madshi]

it seems weird that .raw is not a valid input extension since eac3to decodes to just that. renaming it to .pcm works but then eac3to seems to assume it's a blu-ray lpcm track and wants to remap the channels.

i wanted to use -down6 on a raw file that was decoded from flac, but i just couldn't get it to work..

'eac3to bla.pcm bla.flac -8 -down6' makes a 6ch flac but channels are mapped wrong

'eac3to bla.pcm bla.flac -8 -0,1,2,3,4,5,6,7 -down6' makes an 8ch flac

flac to flac with -down6 doesn't work either.. 'eac3to bla.flac bla2.flac -down6' creates an 8ch file

it of course works just fine with the orig blu-ray lpcm track but not anymore once the channels have been remapped.

also .wav input seems to be broken.. even wavs made by eac3to give the error message "The raw file has 0 channels, which is not supported by eac3to."
Probably those are all bugs. And probably they are all already fixed with eac3to v2.0.

[madshi]

eac3to v2.0 released

http://madshi.net/eac3to.zip

Please see first post of this thread for the full help text of the new version. Here I'll post a list of the changes instead:

totally new features
* AC3 decoding support (Nero's decoder without DRC/dialnorm)
* resampling to 44.1/48/96 kHz (by using "r8brain")
* apply/reverse PAL speedup (by using "r8brain")
* "eac3to sourceFile" will print out source file details

strongly enhanced features
* dramatically improved performance (no intermediate files, anymore!)
* proper 6.1/7.1 downmixing to 5.1 instead of just dropping the back surround channels
* RAW/PCM file detection now auto detects channels, bitdepth and endian
* WAV is now fully supported as source file format
* destination file extension "PCM" creates Blu-Ray style LPCM tracks
* bitdepth can be reduced to anything between 14 bits and 23 bits

DTS related improvements/changes
* DTS-96/24 support added
* "open bitrate" support added
* strange channel configuration support added
* removal of zero padding from DTS files added
* eac3to can fix broken DTS-ES files (they decode to 5.1 instead of 6.1 without the fix)
* dialog normalization can be removed without removing the additional DTS-HD data now
* core extraction must be specifically asked for now (see "-core" switch)

AC3 related improvements
* did I mention that eac3to can decode AC3 now?
* strange channel configuration support added

TrueHD related improvements
* delay problem (hopefully) solved
* fixed: sometimes some audio data in the middle of a track was lost
* TrueHD/AC3 interweaved file can be stripped to TrueHD only now

various minor improvements/changes
* progress bar added
* eac3to detects file format independently of file extension
* multiple input files can be treated as one big file
* "sox" is not needed, anymore
* "dump" filter not needed, anymore
* "aften.exe" replaced by "libAften.dll"
* "flac.exe" replaced by "libFlac.dll"
* DTS/DD+/AC3 source filter ships with eac3to now
* 8bit support added
* various bug fixes
Probably I've forgotten to mention some changes. There's one thing I should note, though: This is a total rewrite! That means, there may be tons of new bugs. So we'll need to do a lot of testing before we can consider it really stable. One thing I'm not sure about yet is how well the new version will work in Vista, especially in Vista64. I've implemented a number of ugly hacks and tested them only in XP. So eventually I'll need to fix some things in Vista.

Just to be sure, here's the previous eac3to v1.23 for download:

http://madshi.net/eac3to.v1.23.zip

[Thunderbolt8]

:thanks:

edit: hows the quality of PAL speedup removal compared to an original NTSC track, is it of same quality? when slowing down to NTSC with besweet for example there was still some noticeable difference

[laserfan]

Thanks madshi, I just started playing w/eac3to today, and here pops your post! Like the sourcefile printout, and already noticed it prints my "Nero missing" error instantly this time! ;)

I see that Nero 7 is dirt-cheap, and the plugins are $25; but I have not used Nero for years as it's bloated with stuff I neither need nor want on my computer.

Has anyone here figured-out how to minimally install the audio decoder from Nero 7, and the Plugins without all that other junk, er, good stuff?

[Thunderbolt8]

i will remux letters from iwo jima again later tonight, will be able to say then if that truehd delay thing is fixed now

btw. could you please explain what this means:
* destination file extension "PCM" creates Blu-Ray style LPCM tracks ?

when I want to convert blue-ray movie LPCM tracks to flac, do I still have to rename the tracks to .pcm before I put them into eac3to, or to .lpcm or doesnt it matter which of both?

[madshi]

hows the quality of PAL speedup removal compared to an original NTSC track, is it of same quality? when slowing down to NTSC with besweet for example there was still some noticeable difference
Just try it out and report back! :) Quality should be a bit better than besweet. But the original NTSC track is of course hard to beat.

[madshi]

Has anyone here figured-out how to minimally install the audio decoder from Nero 7, and the Plugins without all that other junk, er, good stuff?
Not sure. But the installer of Nero 7 Essentials should allow you to choose which packages to install. Maybe you can even disable "everything". Haven't tried that yet...

[madshi]

btw. could you please explain what this means:
* destination file extension "PCM" creates Blu-Ray style LPCM tracks ?
That means that you can e.g. do "eac3to source.ac3 dest.pcm" and you'll get a PCM file which is formatted the same way as Blu-Ray PCM files are. Not sure if it's useful for anything, but there were some people who were asking for that. So...

when I want to convert blue-ray movie LPCM tracks to flac, do I still have to rename the tracks to .pcm before I put them into eac3to, or to .lpcm or doesnt it matter which of both?
The new RAW/PCM detection algorithm costs some time. So I'm using it only if the file extension is "raw" or if there's a "PCM" somewhere in the file name. Not sure but IIRC xport puts a "PCM" in the file name? If it does, you can just convert the file without having to rename it. Otherwise rename it to "*.pcm" or "*.raw". That's not too hard - is it!?

[Thunderbolt8]

xport puts .mpa at the end.

[madshi]

xport puts .mpa at the end.
I said "if there's a "PCM" somewhere in the file name".

[Thunderbolt8]

you didnt get my question at all, I asked if have to rename the file to .pcm or .lpcm for blue-ray flac tracks, since the the source input option now is (L)PCM and not only .pcm as in previous versions.
found out it actually doesnt matter.

[nautilus7]

I have to make an announcement: MADSHI IS A GREAT LIER!!!

...and i explain:
1. He was telling that new eac3to won't be ready till next week,
2. that he hadn't make any progress to AC3 decoding,
3. that PAL<--> NTSC conversion won't be ready for the next version.

3 big lies :D:D:D


Thank you very much!!!

:thanks::thanks::thanks:

[madshi]

I have to make an announcement: MADSHI IS A GREAT LIER!!!

...and i explain:
1. He was telling that new eac3to won't be ready till next week,
2. that he hadn't make any progress to AC3 decoding,
3. that PAL<--> NTSC conversion won't be ready for the next version.

3 big lies :D:D:D


Thank you very much!!!
Sorry for lying... :D

No honestly, I didn't really expect AC3 decoding and PAL<->NTSC in the first version. The reason why AC3 decoding made it into this version is very simple: I noticed that my E-AC3 Nero related hacks worked for AC3, too, without any changes. So it was really easy to add AC3 decoding.

[nautilus7]

I am currently installing windows on my server... After that i 'll do some conversions. Get ready for feedback.

[Thunderbolt8]

remuxed letters from iwo jima again, both audio and video had same first timestamps. therefore I didnt apply any delay to the truehd flac and everything is in sync. seems to work perfertly now, thanks again!

[nautilus7]

I found out some more features that you didn't tell.
A very good bug report. Yeah, i had a crash. It's the same file i mentioned a few post above. Can you tell anything about this?

[shambles]

fantastic. i don't know how i could live without this tool and now it's a tenfold better..

you're my hero.

by the way, i'm guessing noise-shaping is used with the TPDF dithering? any idea how the quality is compared to ssrc/wavewizard? really interested to see what kind of results 24->20bit conversion will bring.. :D

[nautilus7]

C:\Tools>eac3to ice.greek.ac3 ice.greek.new.ac3 -slowdown -resampleto48000
AC3, 5.1 channels, 1:26:53, 384kbit/s, 48khz
Creating/writing file "ice.greek.new.ac3"...
Done.

Tried something different this time.
The process finished almost in no time and the output file is identical to the source.
Did i do something wrong?

[madshi]

I found out some more features that you didn't tell.
A very good bug report. Yeah, i had a crash. It's the same file i mentioned a few post above. Can you tell anything about this?
Not sure. Can you please mail that bug report to me? A sample of that file would help, too (if the crash also occurs with the sample).

[madshi]

by the way, i'm guessing noise-shaping is used with the TPDF dithering?
No noise shaping. It's "simple" TPDF dithering. Should be good enough, as long as you don't go under 16bit. Noise shaping only reduces the noise that you hear. I don't think TPDF noise is audible in 16bit.

any idea how the quality is compared to ssrc/wavewizard?
No idea. The best idea would be: Try it out and report it here!

[madshi]

C:\Tools>eac3to ice.greek.ac3 ice.greek.new.ac3 -slowdown -resampleto48000
AC3, 5.1 channels, 1:26:53, 384kbit/s, 48khz
Creating/writing file "ice.greek.new.ac3"...
Done.

Tried something different this time.
The process finished almost in no time and the output file is identical to the source.
Did i do something wrong?
Ah, that's a bug. Here's what eac3to thinks:

"Source file is ac3, destination file is ac3, and no different bitrate was specified. So I'm not supposed to reencode. I'm only supposed to remove dialnorm (if even necessary)."

You can work around that by doing it in 2 steps:

(1) eac3to ice.greek.ac3 ice.greek.wav -slowdown
(2) eac3to ice.greek.wav ice.greek.new.ac3

Please note that the "-resampleto48000" is probably superfluous in your case. AC3 files coming from DVDs are usually already 48000. And "-slowdown" does not change the samplerate.

[nautilus7]

I 'll send you the bug report, but how? I don't use an email client. Only webmail (gmail). And a sample too.

For the 2nd problem, i 'll do what you suggested.

[madshi]

really interested to see what kind of results 24->20bit conversion will bring.. :D
Ah yes, that's one thing I forgot to mention: Bitdepth can be reduced to anything between 14 bits and 23 bits now. And for FLAC encoding every bit helps getting the file size down. Roger Dressler (Dolby) recently stated on AVSForum that in his opinion 20 bits may be the sweet spot. So personally I might now reduce all my 24 bit tracks to 20 bit. Haven't fully decided yet...

[hristoff2]

You certainly should re-think the name 'eac3to' :D

We're so reliant on you and your work, thanks a million. :)

[Mtz]

Can somebody post a command line for an eac3 source and the desired target to be AC3 640kbps using the sonic filter?
This is not working anymore for me:
eac3to stream1.eac3 stream2.ac3 -48 -640 -sonic

enjoy,
Mtz

[madshi]

I 'll send you the bug report, but how? I don't use an email client. Only webmail (gmail). And a sample too.
You can PM me the bug report or you can post it into gmail. Once the bug report window is visible on screen, the bug report is automatically stored into the clipboard. So just open PM or gmail and press Ctrl+V.

[madshi]

You certainly should re-think the name 'eac3to' :D
Yeah, that's true. I was already thinking about alternative names. But the name "eac3to" is well known now, even if it doesn't really fit, anymore. So I'm not sure if I should change the name...

[madshi]

Can somebody post a command line for an eac3 source and the desired target to be AC3 640kbps using the sonic filter?
This is not working anymore for me:
eac3to stream1.eac3 stream2.ac3 -48 -640 -sonic
What is that "-48" supposed to mean? Just remove that. It should work then. The "-640" is probably not necessary, either, cause it's the default value used by eac3to for multichannel audio tracks. So just "eac3to stream1.eac3 stream2.ac3 -sonic" should do.

[hristoff2]

Yeah, that's true. I was already thinking about alternative names. But the name "eac3to" is well known now, even if it doesn't really fit, anymore. So I'm not sure if I should change the name...

Indeed it's well known, you've got a point there. Guess that's the same reason why 'avimux' is still named avimux although it supports matroska etc etc. :)

[Mtz]

Still not working. Error log sent via PM. The problem appear after: Encoding AC3...
Sonic decoder version 4.2.

enjoy,
Mtz

[Thunderbolt8]

I have a problem with ratatouille remuxing (AVC, LPCM).
the movie has seamless branching so I joined each .m2ts part via "copy /b..". after that I used xport to extract video and audio from the joined .m2ts file and used eac3to to convert the LPCM stream (24-bit) to FLAC (everything fine).
but after I remuxed both together with mkvmerge I get an audio problem at playback (ffdshow, coreflac): audio suddenly disappeares at ~53 seconds movietime and also doesnt come back later on, shortly after that mpc crashes.
im not quite sure whether this could be a problem of the new eac3to version or because of seamless branching joining; however the error occurs before the first switch from the 1st to 2nd .m2ts file took place (length of 1st .m2ts file ~1:30 mins).

im currently using v1.23 to convert to flac again and will mux it all again together to see whether it might still crash then.

[madshi]

I have a problem with ratatouille remuxing (AVC, LPCM).
the movie has seamless branching so I joined each .m2ts part via "copy /b..". after that I used xport to extract video and audio from the joined .m2ts file and used eac3to to convert the LPCM stream (24-bit) to FLAC (everything fine).
but after I remuxed both together with mkvmerge I get an audio problem at playback (ffdshow, coreflac): audio suddenly disappeares at ~53 seconds movietime and also doesnt come back later on, shortly after that mpc crashes.
im not quite sure whether this could be a problem of the new eac3to version or because of seamless branching joining; however the error occurs before the first switch from the 1st to 2nd .m2ts file took place (length of 1st .m2ts file ~1:30 mins).

im currently using v1.23 to convert to flac again and will mux it all again together to see whether it might still crash then.
What happens if you play the external FLAC file? Do you get audio after those 53 seconds? Does madFlac also fail to work? Why are you using ffdshow/coreflac? madFlac should be better...

Usually after having joined TS files you need to run TsRemux over the result. However, I'm not sure if that does anything to audio. Maybe as a result of the seamless branching the joined audio data is corrupt. Would be strange but not impossible. E.g. if you add one byte in the middle of a valid PCM file you'll get totally trash sound after that byte.

[nautilus7]

You can PM me the bug report or you can post it into gmail. Once the bug report window is visible on screen, the bug report is automatically stored into the clipboard. So just open PM or gmail and press Ctrl+V.OK, i got it.

I also think a rename of eac3to is needed.

[madshi]

Still not working. Error log sent via PM. The problem appear after: Encoding AC3...
Sonic decoder version 4.2.
It seems that libAften is crashing for you. It's probably a multithreaded related bug. I'll switch libAften.dll to single threading in the next build, just to be safe.

[madshi]

New eac3to name suggestions:

- madAvConvert(er) - (AV for AudioVideo)
- madHcConvert(er) - (Hc for HomeCinema)
- madConvert(er)

The AudioVideo just in case. If I name it "AudioConverter" now, I'll have to change again in case I should ever decide to add some kind of video processing.

Your opinions? Any other suggestions?

[Thunderbolt8]

What happens if you play the external FLAC file? Do you get audio after those 53 seconds? Does madFlac also fail to work? Why are you using ffdshow/coreflac? madFlac should be better...

Usually after having joined TS files you need to run TsRemux over the result. However, I'm not sure if that does anything to audio. Maybe as a result of the seamless branching the joined audio data is corrupt. Would be strange but not impossible. E.g. if you add one byte in the middle of a valid PCM file you'll get totally trash sound after that byte.
afaik madflac was only for external flac files, but my file was inside the .mkv
I tried to mux the flac file into .mka and wanted to play it outside, but I got an error directly before at the beginning of the parsing stage (and strangely enough this error was not reported or didnt stop the muxing of the previous .mkv with video and audio together).

[madshi]

afaik madflac was only for external flac files, but my file was inside the .mkv
I tried to mux the flac file into .mka and wanted to play it outside, but I got an error directly before at the beginning of the parsing stage (and strangely enough this error was not reported or didnt stop the muxing of the previous .mkv with video and audio together).
The newer versions of madFlac support both external files and also files inside of MKV or other containers.

[Thunderbolt8]

The newer versions of madFlac support both external files and also files inside of MKV or other containers.
tried to use madflac instead of coreflac for the remuxed audio/video file, but it gave this grey filter screen directly at the beginning and no sound filter at all was loaded then.

the mkvmerge remux with the v1.23 flac file just finished parsing successfully, will soon find out whether it will work together with the video.

[nautilus7]

New eac3to name suggestions:

- madAvConvert(er) - (AV for AudioVideo)
- madHcConvert(er) - (Hc for HomeCinema)
- madConvert(er)

The AudioVideo just in case. If I name it "AudioConverter" now, I'll have to change again in case I should ever decide to add some kind of video processing.

Your opinions? Any other suggestions?I like the fisrt the most, but not very very much. I 'll propose one later.

The ctrl+V doesn't work. It doesn't copy the error report. Do you want only the first tab of it?
I found also that the error report window is always in front of the other windows, like task manager. It's little annoying.


EDIT: I found out the log.txt in the eac3to folder, so no problem.

[Thunderbolt8]

the audio is fine when using eac3to v1.23 (apart from some delay, cause by the way of joining the files together I guess), the overall filesize is also bigger, because of the successful flac parsing.

the filesize of both tracks is also different, depending on the use of which version:

FLAC v.123: 2.77 GB
FLAC v2.0: 2.73 GB

I made a 10mb sample of the 3 tracks (source .pcm, flac v1.23, flac v2.0), maybe it helps.
http://www.megaupload.com/de/?d=09KGU6AN

[nautilus7]

Ah, that's a bug. Here's what eac3to thinks:

"Source file is ac3, destination file is ac3, and no different bitrate was specified. So I'm not supposed to reencode. I'm only supposed to remove dialnorm (if even necessary)."

You can work around that by doing it in 2 steps:

(1) eac3to ice.greek.ac3 ice.greek.wav -slowdown
(2) eac3to ice.greek.wav ice.greek.new.ac3

Please note that the "-resampleto48000" is probably superfluous in your case. AC3 files coming from DVDs are usually already 48000. And "-slowdown" does not change the samplerate.It just came to me...

The output wav would exceeds the 4GB limit. It's a 1,5 hour movie. So, is there any other method?

[nautilus7]

About the eac3 to ac3 conversion i am trying to do... Here's the log and a sample, but i think it's a libaften related problem, because eac3 to wavs works.

http://rapidshare.com/files/70467275/sample.zip.html

[Thunderbolt8]

the audio is fine when using eac3to v1.23 (apart from some delay, cause by the way of joining the files together I guess), the overall filesize is also bigger, because of the successful flac parsing.

the filesize of both tracks is also different, depending on the use of which version:

FLAC v.123: 2.77 GB
FLAC v2.0: 2.73 GB

I made a 10mb sample of the 3 tracks (source .pcm, flac v1.23, flac v2.0), maybe it helps.
http://www.megaupload.com/de/?d=09KGU6AN
did another remux try, the first attempt was only with copy /b and then xport use. did now run tsremux over the with copy /b merged .m2ts before using xport, but the result was the same: audio still out of sync in the big .mkv during the movie (flac with v1.23) and the FLAC file created with v20 still cant be put into .mka, again parsing error direct at the beginning. might really be a little bug of the new version :P

[ACrowley]

@Madshi excellent work

I like the AC3 decoding Feature with a Reference Decoder (nero) so the AC3 to X Quality should be optimal.
Also the Timestrech is very nice..no need for me to use Behappy anymore..now i will use eac3to for "ALL" Codecs/Formats to .wavs

Hopefull Keymaker create a new gui because gui 1.5 fails from the beginning on...Filepath in ,", quotes will not work with eac3to 2.0 etc etc etc

[Sephiroth0000]

Hiya eveeryone! Madshi loving the new EAC3TO. Although on that note when I attempted to make a DTS file from it does not finish up. It does all the WAV but yet at the end it saids something like SURCODE DTS ENCODER DOES'NT SEEM TO BE INSTALLED

Any idea's anyone?

[menlvd]

Hiya eveeryone! Madshi loving the new EAC3TO. Although on that note when I attempted to make a DTS file from it does not finish up. It does all the WAV but yet at the end it saids something like SURCODE DTS ENCODER DOES'NT SEEM TO BE INSTALLED

Any idea's anyone?

have same problem! but in my sys installed 1.0.29 ver of SurCode DVD-DTS

[Sephiroth0000]

I did not even know I was supposed to install anything else. What else am I supposed to get to make it work?

[madshi]

have same problem! but in my sys installed 1.0.29 ver of SurCode DVD-DTS
Please check your registry. If Surcode is properly installed for you there should be a registry value named:

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

eac3to is checking if that value exists. If it doesn't exist eac3to complains that Surcode is not installed.

[madshi]

I did not even know I was supposed to install anything else. What else am I supposed to get to make it work?
See first post of this thread. Surcode is a commercial software and must be bought and installed for DTS encoding to work. If you don't want to spend money, you can use AC3 encoding instead.

[Sephiroth0000]

See first post of this thread. Surcode is a commercial software and must be bought and installed for DTS encoding to work. If you don't want to spend money, you can use AC3 encoding instead.

I went to their website and there are loads of different products. Which one do I actually need Madshi? (please not the expensive ones lol)

[menlvd]

thnx madshi for great tool and good apply/reverse PAL speedup
maybe U can do in future ver with output multimono waves :)

[madshi]

I went to their website and there are loads of different products. Which one do I actually need Madshi? (please not the expensive ones lol)
Again read the first post in this thread. The product name is listed there. And unfortunately it *is* an expensive one.

[madshi]

thnx madshi for great tool and good apply/reverse PAL speedup
maybe U can do in future ver with output multimono waves :)
What do you mean with multimono waves? You mean 6 mono wav files? That's already supported! Just do "eac3to sourceFile dest.wavs".

[menlvd]

What do you mean with multimono waves? You mean 6 mono wav files? That's already supported! Just do "eac3to sourceFile dest.wavs".

thx my mistake
and how about an a percentage progress bar of job
such as a completed xx%

[madshi]

eac3to v2.01 released

http://madshi.net/eac3to.zip

* fixed: AC3 encoding sometimes crashed when being fed 24 bit audio data
* fixed: AC3 encoded files were invalid when being fed 24 bit audio data
* eac3toGUI didn't work with eac3to v2.0
* "eac3to source.ac3 dest.ac3 -slowdown" didn't do anything useful
* when a crash occurs, the bug report is automatically copied to clipboard now
* some minor cosmetic improvements

[madshi]

thx my mistake
and how about an a percentage progress bar of job
such as a completed xx%
There is already a progress bar. It's not with %, but the conversion is fully done when the progress bar reaches the right side of the window.

[madshi]

did another remux try, the first attempt was only with copy /b and then xport use. did now run tsremux over the with copy /b merged .m2ts before using xport, but the result was the same: audio still out of sync in the big .mkv during the movie (flac with v1.23) and the FLAC file created with v20 still cant be put into .mka, again parsing error direct at the beginning. might really be a little bug of the new version :P
Can't reproduce the problem here. Your samples you sent me can be put into mka just fine. Does putting them into mka really fail for you?

Again: Please test the external standalone v20 created FLAC file with madFlac. Does that play properly?

I think I found a bug in mkvtoolnix. Just converted (as a test) Pirates of the Caribbean 24bit PCM to FLAC and tried to mux that into mkvtoolnix. mkvtoolnix claimed conversion was done ok, but the output file was only 5MB big! I think mkvtoolnix stumbled over the FLAC file size (> 2GB) or maybe about the number of samples (> 4GB). Don't know...

[shambles]

did the new version break eac3 decoding? i get a crash if dtsac3source is not registered when i run eac3to and this if it is:

E:\Program Files\eac3to>eac3to 00.eac3 00.wav
E-AC3, 5.1 channels, 1:39:57, 640kbit/s, 48khz, dialnorm: -27dB
Decoding with DirectShow (Nero Audio Decoder 2)...
Removing dialog normalization...
Loading DirectShow source file failed.

i tried un/re-registering dtsac3source and even reinstalling nero and re-registering the hddvd/bluray plugin, but none of that helped. i sent you the bug report via pm..

[Thunderbolt8]

Can't reproduce the problem here. Your samples you sent me can be put into mka just fine. Does putting them into mka really fail for you?

Again: Please test the external standalone v20 created FLAC file with madFlac. Does that play properly?

I think I found a bug in mkvtoolnix. Just converted (as a test) Pirates of the Caribbean 24bit PCM to FLAC and tried to mux that into mkvtoolnix. mkvtoolnix claimed conversion was done ok, but the output file was only 5MB big! I think mkvtoolnix stumbled over the FLAC file size (> 2GB) or maybe about the number of samples (> 4GB). Don't know...
I cant put that v2.0 24-bit flac file into .mka with mkvtoolnix, it gives a parsing error immediately, while its possible with the one created with version v.123

error msg:
mkvmerge v2.1.0 ('Another Place To Fall') built on Aug 19 2007 13:39:56
'...\rataCOPYB.flac': Using the FLAC demultiplexer.
+-> Parsing the FLAC file. This can take a LONG time.
+-> Pre-parsing FLAC file: 0%
+-> Pre-parsing FLAC file: 100%
'...\rataCOPYB.flac' track 0: Using the FLAC output module.
The file '...\rataCOPYB.mka' has been opened for writing.
The cue entries (the index) are being written...
Muxing took 0 seconds.

[Mtz]

eac3to v2.01 released

This version solved my previous problems and the GUI is working too. Thank you, madshi.
Regarding new names I think madconverter from that list si the best.
Regarding GUI: at destination file tab, after pressing Browse to inculde the dts extension, and Save button instead of Open. Will be nice at the "Conversions Options" tab to have checkboxes for all formats which can be saved by eac3to, the "DTS Rate" to be just "Rate" because of checkboxes.
Also more settings like "keep intermediate files" or "Temporary folder".

enjoy,
Mtz

[nautilus7]

From the 1st post: -quality=4 slowdown/speedup/resampling quality (0 = low; 4 = very high)

What is the default value if not used?
What does it mean/do or how it works?
How does it affect the processing time?

[nautilus7]

The -slowdown "thing" seems to work, but i just want to clarify something: input.ac3 output.ac3 -slowdown doesn't leave the bitrate same as the input but re-encodes the new ac3 in 640 kbps, right?

[madshi]

I cant put that v2.0 24-bit flac file into .mka with mkvtoolnix, it gives a parsing error immediately, while its possible with the one created with version v.123

error msg:
mkvmerge v2.1.0 ('Another Place To Fall') built on Aug 19 2007 13:39:56
'...\rataCOPYB.flac': Using the FLAC demultiplexer.
+-> Parsing the FLAC file. This can take a LONG time.
+-> Pre-parsing FLAC file: 0%
+-> Pre-parsing FLAC file: 100%
'...\rataCOPYB.flac' track 0: Using the FLAC output module.
The file '...\rataCOPYB.mka' has been opened for writing.
The cue entries (the index) are being written...
Muxing took 0 seconds.
Does mkvtoolnix really say that an error occurred? Where does it say that? I have the same result you're having (log looks the same), but for me mkvtoolnix claims everything is fine. Obviously not everything is fine because the resulting mka file is much too short. But mkvtoolnix believes everything would be fine. Is it the same for you?

And now I'm asking you for the 3rd time: Does the external standalone FLAC file play fine with madFlac? If you want me to fix the problem, it would help a lot if you could answer the questions I'm asking you. Thanks...

[madshi]

did the new version break eac3 decoding? i get a crash if dtsac3source is not registered when i run eac3to and this if it is:

E:\Program Files\eac3to>eac3to 00.eac3 00.wav
E-AC3, 5.1 channels, 1:39:57, 640kbit/s, 48khz, dialnorm: -27dB
Decoding with DirectShow (Nero Audio Decoder 2)...
Removing dialog normalization...
Loading DirectShow source file failed.

i tried un/re-registering dtsac3source and even reinstalling nero and re-registering the hddvd/bluray plugin, but none of that helped. i sent you the bug report via pm..
Thanks for the report. eac3to tried to automatically register the dtsac3source filter if it isn't registered yet. There was a bug in this automatic registering which made eac3to crash. This is fixed in v2.02. However, I don't really understand why manually registering dtsac3source didn't help. It should have helped. And it did in my test. Anyway, the crash should be fixed now.

[madshi]

From the 1st post: -quality=4 slowdown/speedup/resampling quality (0 = low; 4 = very high)

What is the default value if not used?
What does it mean/do or how it works?
How does it affect the processing time?
The default value is "4" (highest quality). The higher the quality the higher the processing time, of course. I wouldn't recommend to go lower than quality "2". Actually I recommend to keep using "4". I think it's better to wait a few seconds longer than having to live forever with a lower than possible audio quality... :)

The -slowdown "thing" seems to work, but i just want to clarify something: input.ac3 output.ac3 -slowdown doesn't leave the bitrate same as the input but re-encodes the new ac3 in 640 kbps, right?
Yes. eac3to always uses 640kbps encoding for multichannel tracks by default, regardless of which bitrate the source file had. This is as intended/designed and I won't change that. If you want to encode with lower bitrate, you can manually tell eac3to to do that.

[madshi]

eac3to v2.02 released

http://madshi.net/eac3to.zip

* fixed: automatic registering of the dtsac3source filter crashed

[Sephiroth0000]

Madshi I have made DTS, AVI, WAV and DDP audio files yet Nero Recode 2 does not register any of them at all and will not let me add them (it does the video just fine) but not the audio....any suggestions please?

[madshi]

Madshi I have made DTS, AVI, WAV and DDP audio files yet Nero Recode 2 does not register any of them at all and will not let me add them (it does the video just fine) but not the audio....any suggestions please?
I've never used Nero Recode yet, so I can't really help you there. Is this problem in any way related to eac3to?

[Sephiroth0000]

I've never used Nero Recode yet, so I can't really help you there. Is this problem in any way related to eac3to?

No not really EAC3TO works extremely well and seems to run faster actually on my system. I just thought you would have a heads up doing all this audio and stuff....no problem

[shambles]

no crash now but i'm still getting "Loading DirectShow source file failed." :confused:

dtsac3source -> nero audio decoder 2 works fine in graphedit (renamed to recode.exe obviously)

[madshi]

no crash now but i'm still getting "Loading DirectShow source file failed." :confused:
Which OS is that? Did that work with eac3to v2.0? That would be strange because I've no done any changes in this area. Do you have more than just one eac3 test files? Do they all fail to work? Or is it just one?

[madshi]

I cant put that v2.0 24-bit flac file into .mka with mkvtoolnix, it gives a parsing error immediately, while its possible with the one created with version v.123

error msg:
mkvmerge v2.1.0 ('Another Place To Fall') built on Aug 19 2007 13:39:56
'...\rataCOPYB.flac': Using the FLAC demultiplexer.
+-> Parsing the FLAC file. This can take a LONG time.
+-> Pre-parsing FLAC file: 0%
+-> Pre-parsing FLAC file: 100%
'...\rataCOPYB.flac' track 0: Using the FLAC output module.
The file '...\rataCOPYB.mka' has been opened for writing.
The cue entries (the index) are being written...
Muxing took 0 seconds.
I've found out that this is a bug in mkvtoolnix. mkvtoolnix doesn't seem to like *any* FLAC files created by FLAC v1.2.1. But FLAC files created by FLAC v1.2.0 (which eac3to v1.23 used) work fine with mkvtoolnix. I'll contact Mosu and ask for a fix. In the meanwhile I've reverted back to FLAC v1.2.0. You can redownload the latest eac3to build:

http://madshi.net/eac3to.zip

I didn't change eac3to, I just swapped the libFLAC.dll with on older version. This fixed the mkvtoolnix problem for me.

[shambles]

xp sp2, didn't work in 2.0, all eac3 tracks fail to work..

they work just fine in 1.23 though.

[Thunderbolt8]

will give it a try

[shambles]

also the source1+source2 input is broken atleast with the 7.1 blu-ray pcm track i have, it's fine until the join point but from then on the channels are mapped wrong (same problem i had when trying to join the files by copy /b blah1+blah2 blah)

[madshi]

xp sp2, didn't work in 2.0, all eac3 tracks fail to work..

they work just fine in 1.23 though.
xpsp2 here, too.

That's kind of strange. Do eac3to tracks still work with 1.23 right now? I mean if you start 2.02 eac3 conversion fails. And if you start 1.23 right next, it works? Please try 1.23 with the "-orbitlee" switch because that is comparable to what 2.02 is doing internally. Thanks...

[madshi]

also the source1+source2 input is broken atleast with the 7.1 blu-ray pcm track i have, it's fine until the join point but from then on the channels are mapped wrong (same problem i had when trying to join the files by copy /b blah1+blah2 blah)
There's nothing I can do here. If the channels are mapped wrong after the join point that means that there are a few bytes too much or too few at the join point. It's nearly impossible for eac3to to know/detect that.

Are you demuxing with xport or with TsRemux? Have you tried running TsRemux over the joined TS file? Does that fix the problem?

[honai]

Wow. That's a huge improvement, thanks a lot for your work and dedication! Now I won't ever have to use any other tool for all my audio needs.

madEierlegendeWollmilchsau might be worth pondering (Germans are known for cranky names anyway). ;)

I'm wondering - how did you avoid having to create temp files? Do you use junction points?

[shambles]

xpsp2 here, too.

That's kind of strange. Do eac3to tracks still work with 1.23 right now? I mean if you start 2.02 eac3 conversion fails. And if you start 1.23 right next, it works? Please try 1.23 with the "-orbitlee" switch because that is comparable to what 2.02 is doing internally. Thanks...

yes, 1.23 still works now. and it does work with -orbitlee too

There's nothing I can do here. If the channels are mapped wrong after the join point that means that there are a few bytes too much or too few at the join point. It's nearly impossible for eac3to to know/detect that.

Are you demuxing with xport or with TsRemux? Have you tried running TsRemux over the joined TS file? Does that fix the problem?

demuxing with xport. and i mean i had the channel mapping problem when i tried to join the demuxed pcm files. the mapping is fine when joining the ts files

[madshi]

madEierlegendeWollmilchsau might be worth pondering
:D

I'm wondering - how did you avoid having to create temp files? Do you use junction points?
Not sure what you mean with "junction points". eac3to v2 internally is built somewhat similiar to how DirectShow works. There are decoder "modules", encoder modules and processing modules. They pass data from one to another in RAM. So no temp files need to be stored. There were a few problems:

(1) I had to replace the well known "Dump" filter with my own real DirectShow filter which grabs the DirectShow output. Of course instead of dumping the output to a file (that's what the original dump filter does) my custom dump filter passes the data directly to me in RAM. Furthermore my dump filter tells me which data format the data has (how many channels etc). The original dump filter didn't store this information so the old eac3to version had to guess.
(2) I'm hooking the file read accesses of the DirectShow source filters (e.g. DTS/DD+/AC3 source). Instead of reading a real file I'm simulating a virtual file. This way I can feed data to the DirectShow source filters via RAM and don't need to create a temp file. This hooking is a bit hacky, so I'm a bit afraid that it might make some problems, but so far it seems to work well.

The only situation where I need to create temp files now is for Surcode encoding. Surcode is another process so it'd be harder to hook file reading there. It might still be possible but I don't think that I'll invest the time for that.

[madshi]

yes, 1.23 still works now. and it does work with -orbitlee too
Hmmmm... Does AC3 and DTS decoding work with eac3to 2.0x? Is it just E-AC3 decoding which fails? Or does AC3 and DTS decoding also fail?

demuxing with xport. and i mean i had the channel mapping problem when i tried to join the demuxed pcm files. the mapping is fine when joining the ts files
Too bad. I don't think I can work around that. You could ask the xport author if he could correct it. He might not be able to demux perfectly the same way as if you joined the TS files first. But at least it should be possible for him to demux in that way that the channel mapping stays correct.

[shambles]

Hmmmm... Does AC3 and DTS decoding work with eac3to 2.0x? Is it just E-AC3 decoding which fails? Or does AC3 and DTS decoding also fail?

ac3 and dts decoding also fail indeed


Too bad. I don't think I can work around that. You could ask the xport author if he could correct it. He might not be able to demux perfectly the same way as if you joined the TS files first. But at least it should be possible for him to demux in that way that the channel mapping stays correct.

well, there's nothing wrong with the channel mapping with the individual pcm files, it only gets screwed when they're joined..

[Thunderbolt8]

And now I'm asking you for the 3rd time: Does the external standalone FLAC file play fine with madFlac? If you want me to fix the problem, it would help a lot if you could answer the questions I'm asking you. Thanks...
should I still try that out, is it still needed ?
I already tried that, but the windows complains about the pathname thing, when I add /dubb + flacpath at the end of the commandline.

putting the flac into .mka worked now btw.

[madshi]

ac3 and dts decoding also fail indeed
Hmmmmmmmmm... That means that the DTS/DD+/AC3 source filter fails to work correctly with v2.0x on your PC. I'll have to think about a way how we can find out what's going wrong. Do you have another PC you can try this on? It seems that this seems to work for most other people. So I guess there must be something special about your PC...

well, there's nothing wrong with the channel mapping with the individual pcm files, it only gets screwed when they're joined..
Well, that's a good hint. It means that the demuxed audio data of the first part is a few bytes too long (or too short). Basically the size of each demuxed PCM file must be dividable through "channelno * bitdepth / 8". E.g. if you have a 24bit 6-channel movie and the file size of a PCM file is 18002 bytes long, there's something wrong. If you calculate "18002 / (channelno * 24 / 8) = 1000.111" you end up with a decimal number. This is the reason why the channel mapping is all wrong. You can manually fix that by reducing file size so that the file size is dividable correctly. E.g. if you reduce the file size in the above example to 18000 we get "18000 / (6 * 24 / 8) = 1000". This will correct the channel mapping problem.

xport should be able to do this automatically, but seemingly it doesn't. As I said before this is something the xport author could probably easily fix.

[madshi]

should I still try that out, is it still needed ?
It's not needed if eac3to v2.02 with libFlac 1.2.0 works perfectly fine for you. Then we can all be happy... :)

[nautilus7]

The default value is "4" (highest quality). The higher the quality the higher the processing time, of course. I wouldn't recommend to go lower than quality "2". Actually I recommend to keep using "4". I think it's better to wait a few seconds longer than having to live forever with a lower than possible audio quality... :)Yes, i agree. I like 4. :p


Yes. eac3to always uses 640kbps encoding for multichannel tracks by default, regardless of which bitrate the source file had. This is as intended/designed and I won't change that. If you want to encode with lower bitrate, you can manually tell eac3to to do that.I knew that 640 was the default bitrate, but i thought the -slowdown would produced a same bitrate track as the input file.

[Thunderbolt8]

Hmmmmmmmmm... That means that the DTS/DD+/AC3 source filter fails to work correctly with v2.0x on your PC. I'll have to think about a way how we can find out what's going wrong. Do you have another PC you can try this on? It seems that this seems to work for most other people. So I guess there must be something special about your PC...


Well, that's a good hint. It means that the demuxed audio data of the first part is a few bytes too long (or too short). Basically the size of each demuxed PCM file must be dividable through "channelno * bitdepth / 8". E.g. if you have a 24bit 6-channel movie and the file size of a PCM file is 18002 bytes long, there's something wrong. If you calculate "18002 / (channelno * 24 / 8) = 1000.111" you end up with a decimal number. This is the reason why the channel mapping is all wrong. You can manually fix that by reducing file size so that the file size is dividable correctly. E.g. if you reduce the file size in the above example to 18000 we get "18000 / (6 * 24 / 8) = 1000". This will correct the channel mapping problem.

xport should be able to do this automatically, but seemingly it doesn't. As I said before this is something the xport author could probably easily fix.
not sure if I understood this correctly now, but it basically means that demuxed audio tracks (by xport) from joined .m2ts tracks are not 100% in sync as they should?
maybe this also explains my problems with that seamless branching of ratatouille. no matter how I join the files (copy /b, tsremux, tssplitter), in the end the remuxed file (avc, flac) loses sync sooner or later.
and when I just try to play that joined .m2ts file with powerdvd then I dont have any sound at all :S

[madshi]

not sure if I understood this correctly now, but it basically means that demuxed audio tracks (by xport) from joined .m2ts tracks are not 100% in sync as they should?
If there's only one m2ts file then it should be in sync. Problems start when there are multiple m2ts files. I'm not sure if there's a problem with xport. Maybe some audio data is stored twice (in the end of part1.m2ts and in the beginning of part2.m2ts)? In that case xport can't really do much. There'd have to be a demuxer which looks at all m2ts files and demux the audio while taking into account the exact timestamps etc. Probably quite difficult to do.

[Thunderbolt8]

hm...:D
would it actually be much work to enhance the offsetpts function also to work on blu-ray .m2ts files? :P

[madshi]

ac3 and dts decoding also fail indeed
Can you please retry with v2.03? It will most likely still fail. But if you use the new "-debug" switch, eac3to will write a "log.txt" file. Can you please PM me that file? Thanks...

[madshi]

hm...:D
would it actually be much work to enhance the offsetpts function also to work on blu-ray .m2ts files? :P
It would just change the timestamps. I don't think it would improve the audio demuxing with xport. It would probably not have any effect on xport.

[madshi]

eac3to v2.03 released

http://madshi.net/eac3to.zip

* new "-debug" switch added

[Thunderbolt8]

It would just change the timestamps. I don't think it would improve the audio demuxing with xport. It would probably not have any effect on xport.
hm in how far are .m2ts files then different in this regard to .evo files? evodemux seems to handle them correctly after this change, but xport obviously somehow ignores this info?

[madshi]

hm in how far are .m2ts files then different in this regard to .evo files? evodemux seems to handle them correctly after this change, but xport obviously somehow ignores this info?
OffsetPTS has no effect whatsoever on demuxing audio tracks with EvoDemux. OffsetPTS helps only for joining EVO files. If all you're doing is demuxing audio with EvoDemux you'd never need OffsetPTS.

[shambles]

Well, that's a good hint. It means that the demuxed audio data of the first part is a few bytes too long (or too short). Basically the size of each demuxed PCM file must be dividable through "channelno * bitdepth / 8". E.g. if you have a 24bit 6-channel movie and the file size of a PCM file is 18002 bytes long, there's something wrong. If you calculate "18002 / (channelno * 24 / 8) = 1000.111" you end up with a decimal number. This is the reason why the channel mapping is all wrong. You can manually fix that by reducing file size so that the file size is dividable correctly. E.g. if you reduce the file size in the above example to 18000 we get "18000 / (6 * 24 / 8) = 1000". This will correct the channel mapping problem.

xport should be able to do this automatically, but seemingly it doesn't. As I said before this is something the xport author could probably easily fix.

demuxing the pcm files again now, will see what the filesizes are. if it is some bytes too long, how would i go about reducing the size?

Hmmmmmmmmm... That means that the DTS/DD+/AC3 source filter fails to work correctly with v2.0x on your PC. I'll have to think about a way how we can find out what's going wrong. Do you have another PC you can try this on? It seems that this seems to work for most other people. So I guess there must be something special about your PC...

it does indeed work on the other computer.. really strange.

Can you please retry with v2.03? It will most likely still fail. But if you use the new "-debug" switch, eac3to will write a "log.txt" file. Can you please PM me that file? Thanks...

yes it still fails but it doesn't seem to write log.txt with -debug

E:\Program Files\eac3to>eac3to 00.tmp.ddp 00.wav -debug
E-AC3, 5.1 channels, 1:39:57, 640kbit/s, 48khz
Decoding with DirectShow (Nero Audio Decoder 2)...
Loading DirectShow source file failed.

no log.txt in eac3to dir..

[shambles]

the pcm file from the first m2ts = 2 294 179 654 bytes

2294179654 / (8 * 16 / 8) = 143386228,375 so yes, indeed, therein lies our bug.

the file from the second m2ts is correctly dividable

[Thunderbolt8]

so xport is currently not recommended to use for joined .m2ts files?

are there other, more reliable demuxers at present?

[madshi]

yes it still fails but it doesn't seem to write log.txt with -debug

E:\Program Files\eac3to>eac3to 00.tmp.ddp 00.wav -debug
E-AC3, 5.1 channels, 1:39:57, 640kbit/s, 48khz
Decoding with DirectShow (Nero Audio Decoder 2)...
Loading DirectShow source file failed.

no log.txt in eac3to dir..
The log.txt is written to the "current" folder. The one where the command prompt is at. E.g. if the command prompt sais: "C:>" the log file it written to "C:\log.txt".

[madshi]

the pcm file from the first m2ts = 2 294 179 654 bytes

2294179654 / (8 * 16 / 8) = 143386228,375 so yes, indeed, therein lies our bug.

the file from the second m2ts is correctly dividable
Ok, now you can use e.g. a hexedit to cut off the last 6 bytes. Afterwards the channel mapping should be correct when the files are joined. E.g. personally, I'm using this free hexeditor:

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

Also you might want to contact the xport author and ask for a fix. Should be really easy for him to fix.

[madshi]

so xport is currently not recommended to use for joined .m2ts files?
I don't know. It usually works for me. But seamless branching can be problematic.

are there other, more reliable demuxers at present?
You could try Haali Media Splitter -> Dump. Don't know if that's more reliable or not.

[shambles]

The log.txt is written to the "current" folder. The one where the command prompt is at. E.g. if the command prompt sais: "C:>" the log file it written to "C:\log.txt".

you can actually see in my post that the eac3to dir was the current dir.. no log file anywhere :confused:

[Thunderbolt8]

hm use the windows file search "log.txt" :P

[madshi]

you can actually see in my post that the eac3to dir was the current dir.. no log file anywhere :confused:
I'm confused. Is it really 2.03? :confused:

[Thunderbolt8]

new haali version is out btw :D (doesnt really belong here, but its so universal :P)

[shambles]

http://img88.imageshack.us/img88/592/eac3tobh0.png

[shambles]

Ok, now you can use e.g. a hexedit to cut off the last 6 bytes. Afterwards the channel mapping should be correct when the files are joined. E.g. personally, I'm using this free hexeditor:

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

Also you might want to contact the xport author and ask for a fix. Should be really easy for him to fix.

using bla1+bla2 in eac3to after cutting off the last 6 bytes works perfectly. channel mapping and everything perfect. :D

i'll report the bug to the xport author asap

[madshi]

new haali version is out btw :D (doesnt really belong here, but its so universal :P)
This is actually a VERY important release. I'll post details in a few minutes in the Haali thread.

[madshi]

http://img88.imageshack.us/img88/592/eac3tobh0.png
There really IS something strange with your PC. Can you please test logging on that other PC, too? :D

[drmpeg]

Ok, now you can use e.g. a hexedit to cut off the last 6 bytes. Afterwards the channel mapping should be correct when the files are joined. E.g. personally, I'm using this free hexeditor:

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

Also you might want to contact the xport author and ask for a fix. Should be really easy for him to fix.
I have a fix for the LPCM channel mapping issue with concatenated .m2ts files.

Version 0.99

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

I wasn't able to fully test it, since I don't have any full movie .m2ts files. However, I did regress it with the samples I do have, so it shouldn't be any worse.

The error message "LPCM sample resync, adding x samples" will occur when xport detects a mismatch (it looks for a match at the beginning of every PES packet). It just adds enough samples to pad out to the expected number of channels. A/V sync should not be affected too much, since it's only 1/48000 of a second (20 microseconds).

The error message should only occur at concatenation points. xport will not pad the end of a demuxed LPCM file.

Ron

[Thunderbolt8]

thanks!

[Rectal Prolapse]

When combining Ratatouille files you need to run the resultant large m2ts file through tsremux again - rewriting the timestamps. This is NEEDED to fix playback in Haali and many other players, and will probably fix demux issues when using xport on the big file.

[madshi]

I have a fix for the LPCM channel mapping issue with concatenated .m2ts files.

Version 0.99

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

I wasn't able to fully test it, since I don't have any full movie .m2ts files. However, I did regress it with the samples I do have, so it shouldn't be any worse.

The error message "LPCM sample resync, adding x samples" will occur when xport detects a mismatch (it looks for a match at the beginning of every PES packet). It just adds enough samples to pad out to the expected number of channels. A/V sync should not be affected too much, since it's only 1/48000 of a second (20 microseconds).

The error message should only occur at concatenation points. xport will not pad the end of a demuxed LPCM file.

Ron
Thanks, Ron! :)

[序列人]

There really IS something strange with your PC. Can you please test logging on that other PC, too? :D

Me too, everything goes well before v2.0!

[shambles]

There really IS something strange with your PC. Can you please test logging on that other PC, too? :D

logging works on the other pc, except it seems to just print out the log in the command prompt window, it doesn't make a file.

[Murleen]

Is it possible to get the runtime from those DTS bytes? I mean could I just search in the file for the first and last sync word, substract the DTS values and convert them to a runtime somehow? That would be nice.

Alas no - there's only two bytes, which tend to wrap every 1.4s - I think the value is used to perform bitrate smoothing, telling a decoder to decode some samples and buffer them earlier...

[madshi]

Me too, everything goes well before v2.0!
And it does not work for you with v2.0? Can you give me a bit more details, please?

[madshi]

Alas no - there's only two bytes, which tend to wrap every 1.4s - I think the value is used to perform bitrate smoothing, telling a decoder to decode some samples and buffer them earlier...
Ah ok, thanks. Do you happen to be the ffmpeg TrueHD decoder author? :)

[madshi]

logging works on the other pc, except it seems to just print out the log in the command prompt window, it doesn't make a file.
Don't understand that. But I'll make a new build which will write the log file to the eac3to folder.

[killa_kid]

I'm trying to convert the DTS-ES audio from my HD DVD of Face/Off and i get the following:

DTS-ES, 6 channels, 2:19:25, 24 bits, 1536kbit/s, 48khz
AC3 encoding only supports 6 channels. Will mix the surround channels.
Decoding with DirectShow (Sonic Audio Decoder)...
DirectShow reports 5.1 channels, 24 bits, 48khz
Encoding AC3...
Mixing surround channels...
The channel modder was started with incorrect parameters.

I tried messing around with a few options (including -downmix and the remapping) but it did not work. I do have Sonic installed, so I am not sure why this is happening. Unless, you can only pull the core DTS out of this? I looked for log.txt but it was not in the directory I ran eac3to in.

The program is working fantastically for my other audio files. This is my only DTS-ES audio, so I can't even try any others.

Any help is appreciated!

[madshi]

I'm trying to convert the DTS-ES audio from my HD DVD of Face/Off and i get the following:

DTS-ES, 6 channels, 2:19:25, 24 bits, 1536kbit/s, 48khz
AC3 encoding only supports 6 channels. Will mix the surround channels.
Decoding with DirectShow (Sonic Audio Decoder)...
DirectShow reports 5.1 channels, 24 bits, 48khz
Encoding AC3...
Mixing surround channels...
The channel modder was started with incorrect parameters.

I tried messing around with a few options (including -downmix and the remapping) but it did not work. I do have Sonic installed, so I am not sure why this is happening. Unless, you can only pull the core DTS out of this? I looked for log.txt but it was not in the directory I ran eac3to in.

The program is working fantastically for my other audio files. This is my only DTS-ES audio, so I can't even try any others.

Any help is appreciated!
Can you send me a small sample of that DTS file? 5MB should be plenty.

P.S: Just as a short explanation: eac3to fails because it expects 6.1, but DirectShow only delivers 5.1. I'll work around that problem. But I'll also like to check out that DTS-ES file. There must be something special about it.

[Murleen]

Ah ok, thanks. Do you happen to be the ffmpeg TrueHD decoder author? :)

Yeah - now know all too much about MLP/TrueHD! Hopefully should be able to get the decoder merged into FFmpeg soon.

[madshi]

Yeah - now know all too much about MLP/TrueHD! Hopefully should be able to get the decoder merged into FFmpeg soon.
Yeah, you must be the TrueHD guru now... :) I had spent several hours on finding out the CRC algorithm for the TrueHD header myself, so I could patch dialog normalization away. I don't want to know how many hours you've spent!! :scared:

Btw, I had tried your decoder patches and got it to compile, but the decoding didn't really work for a test TrueHD file I tried. It always decoded one frame and then aborted, complaining about something (don't remember the exact message). I tried to cut the first few TrueHD frames but that didn't really help much. If there's anything I can help you with, just let me know. I don't like C++, though... :D

[Inventive Software]

@madshi: Do you ever sleep? :D Good work BTW. :)

[idbirch2]

When combining Ratatouille files you need to run the resultant large m2ts file through tsremux again - rewriting the timestamps. This is NEEDED to fix playback in Haali and many other players, and will probably fix demux issues when using xport on the big file.Unfortunately, not so. TSRemux does not fix the sync issues encoutered when dealing with multiple .m2ts movies. I have used TSRemux for both Spiderman 2 and Ratatouille and in both cases, a great deal of time was required to manually fix sync in VideoRedo after re-encoding. I really hope someone figures out a proper way to join .m2ts file soon, these multi-file releases are really annoying!

[Thunderbolt8]

i also experienced that, when I put that tsremux file in tsremux again after that the displayed video length is 2 seconds too short compared to the orginal movie length. and this 2 missing seconds result in 2 seconds audio delay throughout the movie.

someone suggested that overclocking could be an issue because of silent data corruption at the joining stage. I already clocked by system down again, but the result is still the same. will do a reset to bios defaults and test again.

[idbirch2]

My experience is slightly different because the audio skew is not a constant throughout the whole movie. If it was, this would be very easy to correct.

In the movies I have done, the audio is knocked foward of backwards at certain join points. Not all join points of course, that would make sense, no, this is "random".

[killa_kid]

Can you send me a small sample of that DTS file? 5MB should be plenty.

P.S: Just as a short explanation: eac3to fails because it expects 6.1, but DirectShow only delivers 5.1. I'll work around that problem. But I'll also like to check out that DTS-ES file. There must be something special about it.

Here it is: http://www.sendspace.com/file/mzzmzp. I hope it helps, I'd prefer to use the DTS as my source!

The tool is amazing thanks a lot madshi! The fact you're willing to do things like this, is great.

[Thunderbolt8]

i also experienced that, when I put that tsremux file in tsremux again after that the displayed video length is 2 seconds too short compared to the orginal movie length. and this 2 missing seconds result in 2 seconds audio delay throughout the movie.

someone suggested that overclocking could be an issue because of silent data corruption at the joining stage. I already clocked by system down again, but the result is still the same. will do a reset to bios defaults and test again.
tried it all, but still same results. just wont sync :/
ffs how did some people manage to join that successfully then?

[nautilus7]

I have a problematic eac3 track.
I am triyng to convert it to ac3 and eac3to stops saying that i should use delaycut to clean the track.
I load the eac3 track to the latest version of delaycut that supports eac3 and it recognize 2 frames as problematic and drops them whatever mode i choose (fix, ignore, silence). Is there a bug there?

I can't cut a sample with these frames in it, because delaycut drops them whatever i do.

[Thunderbolt8]

take a hex editor and cut out the rest :D
e.g. if its a the beginning, then cut away the rest after 5 mb or something like that. this sample you can send madshi then

[nautilus7]

Yes, you 're right. Didn't think of that.

Here's a sample: http://rapidshare.com/files/70928211/sample.zip.html

It contains only 1 of the 2 problematic frames. The other was way too far, so i had to make another sample, or a very big one.

[nautilus7]

@ madshi

You haven't update the first post to latest version and also in the ac3 bitrate options you don't mention 384 kbps.

[xkodi]

Btw, I had tried your decoder patches and got it to compile, but the decoding didn't really work for a test TrueHD file I tried. It always decoded one frame and then aborted, complaining about something (don't remember the exact message).

i tested the open source MLP/TrueHD decoder with "V for Vendetta" MLP track and the output is byte by byte identical to the output of eac3to 2.x, with previous eac3to 1.23 the output was different, but that was because of the first 4 bytes cut issue, so that is a huge prove for both eac3to and the open-source MLP/TrueHD decoder, because they use so different approaches and produce the same output.

[menlvd]

try to reencode dts-hd and got that

D:\minep\me_gui\tools\eac3_to>eac3to.exe i:\hair.dtshd g:\hair.ac3
DTS Hi-Res, 5.1 channels, 5:44:36, 24 bits, 1914kbit/s, 48khz
Decoding with DirectShow (Sonic Audio Decoder)...

but its only 1:56:06. From hairspay BD movie

[Thunderbolt8]

hairspray is Blu-ray 7.1 DTS-HD MA afaik, not yes implemented in eac3to.
please cut a 50mb sample from the beginning of the demuxed track and send it to madshi, then this is probably also supported soon.

[menlvd]

50mb of hairspray dts-hd ma 7.1
hairspray (http://rapidshare.com/files/71122156/hairspray.dtshd)

[Thunderbolt8]

gogogo madshi :>

with a bit luck the next version might feature nice 7.1 truehd and dts-hd support ;)

[Blue_MiSfit]

I'm having problems converting E-AC3 to AC3. The source is Black Snake Moan from HD DVD. I'm using Nero decoders.

Basically, it starts to decode the E-AC3, but never starts encoding with Aften - so I get no output file.


eac3.exe "i:\feature.ddp" "feature.ac3"


Running Vista x64... latest version of eac3to and Nero 7.

Suggestions?

~MiSfit

[Falcon4]

I'm having an impossible (LITERALLY) time trying to figure out how to get past the first major hurdle with eac3to. Finding a compatible decoder. It seems literally impossible. The links in the first post are completely useless and it doesn't seem to be compatible with the only two Blu-Ray compatible "DVD" player programs I know of - WinDVD HD and PowerDVD. There's mention of Nero, which burns discs, and Sonic, which who knows what it does (and the website is far from helpful). All over the place here I'm seeing mention of "sonic demuxer" and "sonic decoder" but yet there's not a single useful Google result for such terms. Sonic doesn't even seem to have a player of any form available to buy or otherwise.

How the hell is everyone else here seeming to have so many further problems when the post introducing eac3to isn't helpful at all in even setting the software up?

All I want is to decode the mysterious format that my blu-ray movie's audio came in (DTS Master Audio, 5.1 channels, 24 bits, 1536kbits/s, 48khz), into something I can put to VirtualDub and encode into a more universally compatible XviD format with 2 channel MP3 audio (hopefully I can convince it to do some dynamic range compression too, because let me tell you how much I HATE that "loud gunshots and quiet voices" bullcrap).

And before anyone even says it, there's no way in heck I'm buying a $500 authoring suite just to get a stinkin' DTS decoder. I just want to know what program people here are even USING, because "sonic.com" is a hell of a worthless link to be putting up to say "here's what I'm talking about".

Other than that, hey, thanks. :P

edit: Here's what I have so far...
DTS Master Audio, 5.1 channels, 24 bits, 1536kbit/s, 48khz
Decoding with DirectShow (Sonic Audio Decoder)...
Getting "DTS/AC3/DD+ Source" instance failed.
Whatever that means.

[Beastie Boy]

The easiest way is to install Nero 7 (I believe the Essentials version is OK) and purchase the HD-DVD/BluRay plugin.

The sonic site can be hard to navigate, so I would suggest a Google search as this will also search inside a domain name. Try something like "Sonic.CinePlayer.HD.DVD.Decoder.v4.3" and see if it turns up something on their site.

Cheers, Beastie.

[Falcon4]

But, according to eac3to...
The Nero Audio Decoder doesn't support DTS-HD decoding.

So, fortunately I saw that error before I went on a small spending spree... and the version of CinePlayer you can buy on Sonic's site doesn't claim to support HD discs at all. So as for the "buying" thing, I'm up sh*t creek (still). But with the pointing-in-the-right-direction of Cineplayer, I may be "acquiring" a CinePlayer of some kind that may just do the job I need.

But damn if it hasn't been the past 5 hours of looking.

[honai]

How the hell is everyone else here seeming to have so many further problems when the post introducing eac3to isn't helpful at all in even setting the software up?

You know, it's a fantastic way to introduce yourself to this forum by bashing a free product (which you intend to use, anyway) in your very first post.

Whatever that means.

Post #1 of this thread tells you exactly what it means.

I'm having an impossible (LITERALLY) time trying to figure out how to get past the first major hurdle with eac3to.

The actual problem is between keyboard and chair.

[Falcon4]

Post #1 tells me absolutely nothing more than "here's the puzzle box full of assorted pieces, have fun seeing if they're all there and no, I won't help you find the others". It gives me a command line program that spits out unhelpful error messages that could mean pretty much anything from the codec being installed but not working, to the codec not even being installed. It tells me "Install Sonic" which is like telling someone to install "Microsoft" when they really need Microsoft SQL Server. It dangles this beautiful carrot called "decoding this piece of crap DTS file" in front of me and then teases me with not being able to actually DO it.

Oh, and as for the "free product" thing? Earth to honai: MOST software IS free. While the author has no obligation to create a program for free that actually works, it's definitely in their best interest if they don't want a little thing called a "bad reputation" following them around. Now I do believe that eac3to does actually work, but it's the documentation that's sorely lacking - especially for it not to work on a PC (such as mine) that's actually designed from the ground up, by Sony, to play Blu-Ray movies. I'm not trying to get around any hardware limitations with this software - I'm trying to make it do what it already does in a different context.

Now if you continue to think that PEBKAC, then you need to STFU and GTFO the internets. Srsly.

[honai]

Oh, and as for the "free product" thing? Earth to honai: MOST software IS free. While the author has no obligation to create a program for free that actually works, it's definitely in their best interest if they don't want a little thing called a "bad reputation" following them around.

The only bad reputation eac3to is getting from the tens of thousands of people who downloaded it so far is ... from you.

And let me refer you to this post, written by the maintainer of this forum:

http://forum.doom9.org/showthread.php?t=52597

"General conduct - or how to accept a gift properly"

Those who only aim to take will eventually be singled out and will most likely run into a strike or two along the way. Not knowing the value of community and the rules governing it they will not use the proper channels to lodge a complaint but just start throwing insults around on this board, other boards or their own personal websites and they will eventually find themselves on a suspension.

So far I fail to see any kind words from you for the author of eac3to, or any polite request for help from the community that supports the program. It's especially rich because obviously you did download eac3to ... without paying a single dime for it, while in actual fact the author spent a great deal of unpaid time making it the fantastic product that it is.

[Falcon4]

Right, now how about you actually address the points of my posts instead of just whining about my complaining about the author's lack of coherent documentation?

[honai]

At this point I don't feel inclined to help you out. Hurling insults and then asking for help, that's rich.

The answer to all of your problems lies in the pages 1 to 78. Happy reading.

[Beastie Boy]

Seriously Falcon4, I gave you the benefit of doubt as it was your first post, but with your post-count now at 4, I think they may be the worst and most offensive first posts I have seen. I agree with Honai 100%. Perhaps you should just move along and find some other software to solve your problem. Oh wait, there isn't any.

Perhaps if you eventually get this working, you could write the guide that you think is sorely lacking. Or better still, dedicate hours writing your own piece of software that you could share with the rest of the world.

[Falcon4]

And just to compare pots with kettles...

So far I fail to see any kind words from you for the author of eac3to, or any polite request for help from the community that supports the program. It's especially rich because obviously you did download eac3to ... without paying a single dime for it, while in actual fact the author spent a great deal of unpaid time making it the fantastic product that it is.

Software authors should be expected to provide their bits of code for free. How can I be such an asshole to say this? Because I'm a developer myself - you can see my "portfolio" of current work at falconfour.com (http://falconfour.com) or just go straight to my best site, hostfile.org (http://hostfile.org). I don't charge a damn dime for anything I provide, and I never will either. I think if people really like it, and I've done a good enough job on it, people will use the shiny "donate" link. Now, I may be way off here, but I'd have to say that I put a freaking shitload of a lot more time into Hostfile than the author of eac3to put into that program, considering eac3to is practically a front-end for two audio decoders and a handful of encoders, and my site is a complex system of inter-working scripts and databases using almost entirely my own code. If you'd like to keep going on about the whole "free" crap, then the spam is on you.

In the meantime I'd like to know what crap people have installed on their computers that allows them to "magically" use this program without any further problems beyond just "waah, it introduced a 20ms delay that's making my nose hairs tingle". If I could get it to actually spit out a single byte of audio data, or even begin decoding the original audio, I'd have no complaints with eac3to. Meanwhile... it simply does not work.

edit: And yes, as you can see from the previous statement above, I have spent hours going through, searching, and reading all ridiculous 78 pages of this thread trying to find out how the heck to make it work.

[Falcon4]

Well, without the help of this thread, and with the marginal help of Google (pointing me to a package), I solved the problem and eac3to is now happily crunching away my audio.

The problem was that my DTS/AC3/DD+ Source DirectShow filter was corrupted - specifically, it was registered to a file that was in my "temp" folder and got deleted. I wonder who did that (no, really - I doubt it was eac3to). I also found a slew of other odd filters and plugins that no longer existed and were causing problems. Ironically enough, the K-Lite Codec Pack installer had a neat utility, the Codec Tweak Tool, that was able to detect and repair these problems. It also had a working copy of the DTS/AC3/DD+ Source tool.

So by downloading the K-Lite Codec Pack, setting it to full Custom (deselect all), choosing only the DTS/AC3 Source filter and the Codec Tweak Tool, then running the Codec Tweak Tool to scan for broken filters, I ended up solving the problem.

It sure is taking a while to decode that 30mb audio clip though... o_O

So hey. Thanks to the author of eac3to for creating it, and no thanks to the fine members of this forum for pointing out the obvious.

[madshi]

Software authors should be expected to provide their bits of code for free.
Sure. Shame on every software author who dares to make a living. Of course all software authors were borne with an unlimited and neverending debt to the community. They rightfully owe their lifetime's work to the general public. That's a binding curse which must be the direct consequence of misdeeds in their previous lifes.

Oh wait! I don't believe in reincarnation! Hmmm... :confused:

[honai]

Congratulations for post #1000, madshi. I admire your neverending patience. ;)

[Xorcist]

I am using eac3to 2.03 and I'm trying to convert a DTS-HD track to AC3. However, the process just freezes and never finishes. I read about similar problems a few pages earlier in this thread, and they were resolved with v2.01, but I still have this problem.

Any ideas?

Here's the output:

H:\encoding\tools>eac3to bits0001.dtshd bits0001.ac3 -384 -debug
DTS Master Audio, 5.1 channels, 24 bits, 1536kbit/s, 48khz
Decoding with DirectShow (Sonic Audio Decoder)...
CreateFileW "c:\virtual.dtsma" hooked.
ReadFile (4) hooked.
ReadFile -> read: 4; newPos: 4.
ReadFile (8) hooked.
ReadFile -> read: 8; newPos: 12.
GetFileSize hooked -> 0.
SetFilePointer(0, 0) hooked.
ReadFile (12072) hooked.
ReadFile -> read: 12072; newPos: 12072.
DirectShow reports 5.1 channels, 24 bits, 48khz
Encoding AC3...
SetFilePointer(12072, 0) hooked.
ReadFile (12072) hooked.
ReadFile -> read: 12072; newPos: 24144.

[nautilus7]

@ madshi

Did you have a look at the sample i uploaded? Any ideas?

[Falcon4]

I am using eac3to 2.03 and I'm trying to convert a DTS-HD track to AC3. However, the process just freezes and never finishes. I read about similar problems a few pages earlier in this thread, and they were resolved with v2.01, but I still have this problem.

Any ideas?

As I kinda hinted at in my last post, mine locked up too (requiring a Ctrl+C). Turned out that xport.exe split out a botched up audio stream somehow, but a quick "eac3to ... -core" output a corrected file, which decoded properly into a mixed-up WAV file (like I asked it to) the second run-around. So my suggestion would be to do -core on it first, then decode it. Hope it helps! :)

[Xorcist]

As I kinda hinted at in my last post, mine locked up too (requiring a Ctrl+C). Turned out that xport.exe split out a botched up audio stream somehow, but a quick "eac3to ... -core" output a corrected file, which decoded properly into a mixed-up WAV file (like I asked it to) the second run-around. So my suggestion would be to do -core on it first, then decode it. Hope it helps! :)

Yeah, thanks. Worked out well to get the DTS core and then transcode the DTS to AC3 :)

[madshi]

@ madshi

Did you have a look at the sample i uploaded? Any ideas?
No time yet. Will have time on Sunday, I think...

[Furiousflea]

Hi there, thanks for this unique program first of all :)

been having success with DTS\DTSHD tracks but EAC3 isn't working for me....

I get this message....

C:\eac3to>eac3to shepard.eac3 output.wavs
E-AC3, 5.1 channels, 2:47:30, 1536kbit/s, 48khz, dialnorm: -27dB
Removing dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
Disabling DRC for Nero (E-)AC3 decoding...
The WAV writer didn't receive the format information.

C:\eac3to>

Any ideas what that means, I have purchased the HDDVD plugin for Nero 7 for this specifically so really hope it works

Thanks in advance for any help, Rob

[nautilus7]

No time yet. Will have time on Sunday, I think...OK, i hope you rest a little bit. You deserve it.

[menlvd]

and how about dts-hd 7.1 any chance to get worked in future?
i think dts-hd 7.1 have same as dts-hd 5.1 core of dts 1536 kbps
when i play the m2ts file Nero audio 2 showing that
bitrate: 1509
framesize: 2012
Frequency:48000
Bits: 24
Channels: 6 mask: 0x3f

** DTS Deteils:
Input Stream: DTS Digital Surround
Output Stream: DTS Digital Surround

[nautilus7]

eac3to crashes when trying input.thd output.wavs with nero filters.

Here's the debug log:
date/time : 2007-11-23, 02:17:16, 296ms
computer name : NAUTILUS7
user name : Spyros <admin>
registered owner : Nautilus7
operating system : Windows XP Service Pack 2 build 2600
system language : Greek
system up time : 6 hours 39 minutes
program up time : 53 milliseconds
processors : 2x Intel(R) Core(TM)2 CPU 6600 @ 2.40GHz
physical memory : 512/1023 MB (free/total)
free disk space : (C:) 23,88 GB
display mode : 1680x1050, 32 bit
process id : $f6c
allocated memory : 3,78 MB
command line : eac3to matrix.1.thd qweqwe.wavs
executable : eac3to.exe
exec. date/time : 2007-11-18 19:28
version : 2.0.3.0
compiled with : Delphi 7
madExcept version : 3.0f beta 1
callstack crc : $7fd81968, $1b46de15, $1b46de15
count : 2
exception number : 1
exception class : EAccessViolation
exception message : Zugriffsverletzung bei Adresse 004D6D7D in Modul 'eac3to.exe'. Lesen von Adresse 00000000.

main thread ($5c4):
004d6d7d +45d eac3to.exe eac3to 797 +31 Process
004d77b7 +15f eac3to.exe eac3to 874 +17 initialization

modules:
00330000 libFlac.dll C:\Tools
00400000 eac3to.exe 2.0.3.0 C:\Tools
10000000 libAften.dll C:\Tools
10100000 lgscroll.dll 4.0.121.0 C:\Program Files\Logitech\SetPoint
5ad70000 uxtheme.dll 6.0.2900.2845 C:\WINDOWS\system32
5d090000 comctl32.dll 5.82.2900.2982 C:\WINDOWS\system32
71aa0000 WS2HELP.dll 5.1.2600.2180 C:\WINDOWS\system32
71ab0000 WS2_32.dll 5.1.2600.2180 C:\WINDOWS\system32
71ad0000 wsock32.dll 5.1.2600.2180 C:\WINDOWS\system32
71bf0000 SAMLIB.dll 5.1.2600.2180 C:\WINDOWS\system32
74720000 MSCTF.dll 5.1.2600.2809 C:\WINDOWS\system32
755c0000 msctfime.ime 5.1.2600.2180 C:\WINDOWS\system32
76390000 IMM32.DLL 5.1.2600.2180 C:\WINDOWS\system32
763b0000 comdlg32.dll 6.0.2900.2180 C:\WINDOWS\system32
76f60000 WLDAP32.dll 5.1.2600.2180 C:\WINDOWS\system32
77120000 oleaut32.dll 5.1.2600.3139 C:\WINDOWS\system32
773d0000 comctl32.dll 6.0.2900.2982 C:\WINDOWS\WinSxS\x86_Microsoft.Windows.Common-Controls_6595b64144ccf1df_6.0.2600.2982_x-ww_ac3f9c03
774e0000 ole32.dll 5.1.2600.2948 C:\WINDOWS\system32
77690000 NTMARTA.DLL 5.1.2600.2180 C:\WINDOWS\system32
77c00000 version.dll 5.1.2600.2180 C:\WINDOWS\system32
77c10000 msvcrt.dll 7.0.2600.2180 C:\WINDOWS\system32
77dd0000 advapi32.dll 5.1.2600.2180 C:\WINDOWS\system32
77e70000 RPCRT4.dll 5.1.2600.3173 C:\WINDOWS\system32
77f10000 GDI32.dll 5.1.2600.3159 C:\WINDOWS\system32
77f60000 SHLWAPI.dll 6.0.2900.3086 C:\WINDOWS\system32
77fe0000 Secur32.dll 5.1.2600.2180 C:\WINDOWS\system32
78130000 MSVCR80.dll 8.0.50727.762 C:\WINDOWS\WinSxS\x86_Microsoft.VC80.CRT_1fc8b3b9a1e18e3b_8.0.50727.762_x-ww_6b128700
7c420000 MSVCP80.dll 8.0.50727.762 C:\WINDOWS\WinSxS\x86_Microsoft.VC80.CRT_1fc8b3b9a1e18e3b_8.0.50727.762_x-ww_6b128700
7c800000 kernel32.dll 5.1.2600.3119 C:\WINDOWS\system32
7c900000 ntdll.dll 5.1.2600.2180 C:\WINDOWS\system32
7c9c0000 shell32.dll 6.0.2900.3241 C:\WINDOWS\system32
7e410000 user32.dll 5.1.2600.3099 C:\WINDOWS\system32

processes:
000 Idle
004 System normal
264 smss.exe normal C:\WINDOWS\system32
2ac csrss.exe normal C:\WINDOWS\system32
2c4 winlogon.exe high C:\WINDOWS\system32
2f0 services.exe normal C:\WINDOWS\system32
2fc lsass.exe normal C:\WINDOWS\system32
3ac svchost.exe normal C:\WINDOWS\system32
3fc svchost.exe normal C:\WINDOWS\system32
45c svchost.exe normal C:\WINDOWS\System32
498 svchost.exe normal C:\WINDOWS\system32
520 svchost.exe normal C:\WINDOWS\system32
568 svchost.exe normal C:\WINDOWS\system32
660 spoolsv.exe normal C:\WINDOWS\system32
690 explorer.exe normal C:\WINDOWS
240 nod32kui.exe normal C:\Program Files\Eset
28c ctfmon.exe normal C:\WINDOWS\system32
3b8 TeaTimer.exe idle C:\Program Files\Spybot - Search & Destroy
514 nod32krn.exe normal C:\Program Files\Eset
33c nvsvc32.exe normal C:\WINDOWS\system32
54c PnkBstrA.exe normal C:\WINDOWS\system32
5a4 SetPoint.exe normal C:\Program Files\Logitech\SetPoint
5ec svchost.exe normal C:\WINDOWS\system32
6b8 WLService.exe normal C:\Program Files\Linksys Wireless-G PCI Wireless Network Monitor
134 KHALMNPR.EXE normal C:\Program Files\Common Files\Logitech\KhalShared
8b0 alg.exe normal C:\WINDOWS\System32
b14 WMP54Gv4.exe high C:\Program Files\Linksys Wireless-G PCI Wireless Network Monitor
af0 uTorrent.exe normal C:\Program Files\uTorrent
a58 firefox.exe normal C:\Program Files\Mozilla Firefox
fd8 cmd.exe normal C:\WINDOWS\system32
f6c eac3to.exe normal C:\Tools

hardware:
+ Bluetooth Device
- Bluetooth HID Manager (driver 1.0.10.0)
- Bluetooth VComm Manager (driver 1.4.8.1)
+ Computer
- ACPI Multiprocessor PC
+ Disk drives
- ST3320620AS
+ Display adapters
- NVIDIA GeForce 7600 GT (driver 6.14.10.9424)
+ DVD/CD-ROM drives
- QV4471C BHK234U SCSI CdRom Device
- TSSTcorp CDDVDW SH-S203B
+ Human Interface Devices
- Bluetooth HID Enum Device (driver 1.4.8.1)
+ IDE ATA/ATAPI controllers
- Intel(R) ICH8 2 port Serial ATA Storage Controller - 2825 (driver 8.0.0.1008)
- Intel(R) ICH8 4 port Serial ATA Storage Controller - 2820 (driver 8.0.0.1008)
- Primary IDE Channel
- Primary IDE Channel
- Primary IDE Channel
- Secondary IDE Channel
- Secondary IDE Channel
- Secondary IDE Channel
- Standard Dual Channel PCI IDE Controller
+ IEEE 1394 Bus host controllers
- VIA OHCI Compliant IEEE 1394 Host Controller
+ Keyboards
- PS/2 Keyboard (driver 4.0.0.0)
+ Mice and other pointing devices
- PS/2 Compatible Mouse (driver 4.0.0.0)
+ Modems
- Bluetooth DUN Modem (driver 1.0.6.0)
- Bluetooth Fax Modem (driver 1.0.6.0)
- Bluetooth LAP Modem (driver 1.2.1.0)
- Bluetooth LAP Modem #2 (driver 1.2.1.0)
+ Monitors
- SyncMaster 206BW,SyncMaster Magic CX206BW(Digital) (driver 3.0.0.0)
+ Network adapters
- 1394 Net Adapter
- Attansic L1 Gigabit Ethernet 10/100/1000Base-T Controller (driver 1.7.3790.7)
- Bluetooth PAN Network Adapter (driver 1.0.0.0)
- Linksys Wireless-G PCI Adapter (driver 1.0.3.0)
+ Portable Devices
- Nokia 6131 (driver 6.83.31.1)
- Phone (driver 6.83.31.1)
+ Ports (COM & LPT)
- Bluetooth Serial Port (COM10) (driver 2.0.1.1)
- Bluetooth Serial Port (COM11) (driver 2.0.1.1)
- Bluetooth Serial Port (COM12) (driver 2.0.1.1)
- Bluetooth Serial Port (COM13) (driver 2.0.1.1)
- Bluetooth Serial Port (COM3) (driver 2.0.1.1)
- Bluetooth Serial Port (COM4) (driver 2.0.1.1)
- Bluetooth Serial Port (COM5) (driver 2.0.1.1)
- Bluetooth Serial Port (COM6) (driver 2.0.1.1)
- Bluetooth Serial Port (COM7) (driver 2.0.1.1)
- Bluetooth Serial Port (COM8) (driver 2.0.1.1)
- Bluetooth Serial Port (COM9) (driver 2.0.1.1)
- Communications Port (COM1)
- ECP Printer Port (LPT1)
+ Processors
- Intel(R) Core(TM)2 CPU 6600 @ 2.40GHz
- Intel(R) Core(TM)2 CPU 6600 @ 2.40GHz
+ SCSI and RAID controllers
- SCSI/RAID Host Controller
+ Sound, video and game controllers
- Audio Codecs
- Bluetooth AV Audio (driver 1.1.2.0)
- Bluetooth SCO Audio (driver 1.1.2.0)
- FlyTV Express X1 MST-T2A2 (driver 1.30.5.1)
- Legacy Audio Drivers
- Legacy Video Capture Devices
- Media Control Devices
- SoundMAX Integrated Digital HD Audio (driver 5.10.1.6240)
- Video Codecs
+ System devices
- ACPI Fixed Feature Button
- ACPI Power Button
- ATK0110 ACPI UTILITY (driver 1043.2.15.37)
- Direct memory access controller
- Extended IO Bus
- High Precision Event Timer (driver 7.0.0.1011)
- Intel(R) 82801 PCI Bridge - 244E
- Intel(R) ICH8 Family PCI Express Root Port 1 - 283F (driver 8.0.0.1010)
- Intel(R) ICH8 Family PCI Express Root Port 2 - 2841 (driver 8.0.0.1010)
- Intel(R) ICH8 Family PCI Express Root Port 4 - 2845 (driver 8.0.0.1010)
- Intel(R) ICH8 Family PCI Express Root Port 5 - 2847 (driver 8.0.0.1010)
- Intel(R) ICH8 Family SMBus Controller - 283E (driver 8.0.0.1008)
- Intel(R) ICH8/ICH8R Family LPC Interface Controller - 2810 (driver 8.0.0.1010)
- Intel(R) P965/G965 PCI Express Root Port - 29A1 (driver 8.0.0.1010)
- Intel(R) P965/G965 Processor to I/O Controller - 29A0 (driver 8.0.0.1010)
- ISAPNP Read Data Port
- Logical Disk Manager
- Microcode Update Device
- Microsoft ACPI-Compliant System
- Microsoft System Management BIOS Driver
- Microsoft UAA Bus Driver for High Definition Audio
- Motherboard resources
- Motherboard resources
- Motherboard resources
- Motherboard resources
- Numeric data processor
- PCI bus
- Plug and Play Software Device Enumerator
- Printer Port Logical Interface
- Programmable interrupt controller
- System board
- System board
- System CMOS/real time clock
- System speaker
- System timer
- Terminal Server Device Redirector
- Terminal Server Keyboard Driver
- Terminal Server Mouse Driver
- Volume Manager
+ Universal Serial Bus controllers
- Intel(R) ICH8 Family USB Universal Host Controller - 2830 (driver 8.0.0.1008)
- Intel(R) ICH8 Family USB Universal Host Controller - 2831 (driver 8.0.0.1008)
- Intel(R) ICH8 Family USB Universal Host Controller - 2832 (driver 8.0.0.1008)
- Intel(R) ICH8 Family USB2 Enhanced Host Controller - 2836 (driver 8.0.0.1008)
- Standard Universal PCI to USB Host Controller
- USB Root Hub
- USB Root Hub
- USB Root Hub
- USB Root Hub
- USB Root Hub

cpu registers:
eax = 00000000
ebx = 004bde01
ecx = 004cca5c
edx = 00000001
esi = 004df67c
edi = 00a41cc8
eip = 004d6d7d
esp = 0012febc
ebp = 0012ff74

stack dump:
0012febc 84 ff 12 00 ac 3f 40 00 - 74 ff 12 00 14 00 00 00 .....?@.t.......
0012fecc 00 00 00 00 00 70 fd 7f - 00 00 00 00 00 00 00 00 .....p..........
0012fedc 00 00 00 00 00 00 00 00 - 00 00 00 00 00 00 00 00 ................
0012feec 00 00 00 00 00 00 00 00 - 00 00 00 00 00 00 00 00 ................
0012fefc 00 00 00 00 00 00 00 00 - 00 00 00 00 00 00 00 00 ................
0012ff0c 00 00 00 00 00 00 00 00 - 00 00 00 00 00 00 00 00 ................
0012ff1c 00 00 00 00 00 00 00 00 - 00 00 00 00 00 00 00 00 ................
0012ff2c 00 00 00 00 00 00 00 00 - 00 00 00 00 00 00 00 00 ................
0012ff3c 00 00 00 00 00 00 00 00 - 00 00 00 00 00 00 00 00 ................
0012ff4c 00 00 00 00 00 00 00 00 - 00 00 00 00 00 00 00 00 ................
0012ff5c 00 00 01 01 00 00 00 00 - 00 00 00 00 00 00 00 00 ................
0012ff6c c8 f6 4d 00 d0 2b a4 00 - c0 ff 12 00 bc 77 4d 00 ..M..+.......wM.
0012ff7c cc f6 4d 00 c4 f6 4d 00 - b4 ff 12 00 ac 3f 40 00 ..M...M......?@.
0012ff8c c0 ff 12 00 00 00 00 00 - 00 70 fd 7f 00 00 00 00 .........p......
0012ff9c 00 00 00 00 00 00 00 00 - 00 00 00 00 00 00 00 00 ................
0012ffac 00 00 00 00 00 00 00 00 - e0 ff 12 00 dc 41 40 00 .............A@.
0012ffbc c0 ff 12 00 f0 ff 12 00 - f7 6f 81 7c 14 00 00 00 .........o.|....
0012ffcc 00 00 00 00 00 70 fd 7f - ed a6 54 80 c8 ff 12 00 .....p....T.....
0012ffdc c0 97 5b 83 ff ff ff ff - 30 9a 83 7c 00 70 81 7c ..[.....0..|.p.|
0012ffec 00 00 00 00 00 00 00 00 - 00 00 00 00 58 76 4d 00 ............XvM.

disassembling:
[...]
004d6d6d mov eax, [eax+4]
004d6d70 mov ecx, [eax]
004d6d72 call dword ptr [ecx+$2c]
004d6d75 mov eax, [ebp-$a0]
004d6d7b mov dl, 1
004d6d7d > mov ecx, [eax]
004d6d7f call dword ptr [ecx+$44]
004d6d82 jmp loc_4d6da9
004d6d84 799 lea eax, [ebp-$14]
004d6d87 call -$d08c4 ($4064c8) ; System.@IntfClear
004d6d8c 800 mov al, [ebp-$15]
[...]

[Furiousflea]

Hi there, thanks for this unique program first of all

been having success with DTS\DTSHD tracks but EAC3 isn't working for me....

I get this message....

C:\eac3to>eac3to shepard.eac3 output.wavs
E-AC3, 5.1 channels, 2:47:30, 1536kbit/s, 48khz, dialnorm: -27dB
Removing dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
Disabling DRC for Nero (E-)AC3 decoding...
The WAV writer didn't receive the format information.

C:\eac3to>

Any ideas what that means, I have purchased the HDDVD plugin for Nero 7 for this specifically so really hope it works


To update my previous post, I tried doing the same conversion with an older version of EAC3TO, to see what happened and it came up with this error instead of the previous one (that was using the latest version of EAC3TO

Microsoft Windows XP [Version 5.1.2600]
(C) Copyright 1985-2001 Microsoft Corp.

C:\Documents and Settings\Rob>cd\wmvhd

C:\WMVHD>eac3to shepard.eac3 shepherd.wavs
E-AC3, 5.1 channels, 2:47:30, 1536kbit/s, 48khz, dialnorm: -27dB
Remove Dialog Normalization information. Please wait...
Decoding eac3 file to raw. Please wait...
Disabling DRC for Nero E-AC3 decoding...
The file size of the raw file doesn't seem to fit.
The expected file size for 16 bit is 5.39 GB.
The expected file size for 24 bit is 8.08 GB.
The real file size is 0 Bytes.

C:\WMVHD>

So.....hmmmm....I have no idea what that means, be greatful for any advice\pointer. Thanks, Rob

[nautilus7]

How much free space do you have on your hdd? But i don't think the new eac3to needs lots of it...

[hristoff2]

mmm missing dump filter?
- http://uploaded.to/?id=v40yi1

(obviously everything but writing/dumping the file works, as stated by the log)

[nautilus7]

eac3to 2.xx version doesn't need dump filter. madshi wrote his own filter for this purpose.

[Furiousflea]

How much free space do you have on your hdd? But i don't think the new eac3to needs lots of it...

Hi, Over a 70GB of space on the HD remaining so defo aint to do with that, for the old version dump.ax was "successfully registered".

?:( thanx

[nautilus7]

I think we all have to wait till Sunday. Madshi will fix our "issues" then.

[madshi]

I have a problematic eac3 track.
I am triyng to convert it to ac3 and eac3to stops saying that i should use delaycut to clean the track.
I load the eac3 track to the latest version of delaycut that supports eac3 and it recognize 2 frames as problematic and drops them whatever mode i choose (fix, ignore, silence). Is there a bug there?
I've checked. It's intended behaviour. The problem is that there's not a CRC error in those 2 frames. Instead there's not even a valid header for those frames. So delaycut doesn't know what to do and sees no other chance than to skip forward to the next valid header. delaycut was implemented to behave this way. Those modes work only if the header (actually only the first 2 bytes!) is valid.

in the ac3 bitrate options you don't mention 384 kbps.
There are many more supported bitrates which I didn't mention. I don't flood the help with every single supported bitrate, so I've only listed the most common ones.

[madshi]

i tested the open source MLP/TrueHD decoder with "V for Vendetta" MLP track and the output is byte by byte identical to the output of eac3to 2.x, with previous eac3to 1.23 the output was different, but that was because of the first 4 bytes cut issue, so that is a huge prove for both eac3to and the open-source MLP/TrueHD decoder, because they use so different approaches and produce the same output.
Interesting! I couldn't get the open source TrueHD decoder to work, but it seems that I did something wrong.

[madshi]

try to reencode dts-hd and got that

D:\minep\me_gui\tools\eac3_to>eac3to.exe i:\hair.dtshd g:\hair.ac3
DTS Hi-Res, 5.1 channels, 5:44:36, 24 bits, 1914kbit/s, 48khz
Decoding with DirectShow (Sonic Audio Decoder)...

but its only 1:56:06. From hairspay BD movie
That's an interesting track! There's a new flag combination in the DTS-HD header which eac3to didn't know yet. It got interpreted as DTS Hi-Res while actually it's DTS Master Audio 7.1. I've added support for this in the latest eac3to build.

[madshi]

with a bit luck the next version might feature nice 7.1 truehd and dts-hd support ;)
Unfortunately I'm limited by what the decoders support. The Nero TrueHD decoder doesn't support 7.1 decoding. And the Sonic Audio Decoder seems to decode only 6.1, but not 7.1. With that Hairspray 7.1 DTS-HD Master Audio track Sonic only outputs 5.1, sadly. Once the open source TrueHD decoder is ready to roll, we might get 7.1 TrueHD decoding, at least.

[madshi]

I'm having problems converting E-AC3 to AC3. The source is Black Snake Moan from HD DVD. I'm using Nero decoders.

Basically, it starts to decode the E-AC3, but never starts encoding with Aften - so I get no output file.


eac3.exe "i:\feature.ddp" "feature.ac3"


Running Vista x64... latest version of eac3to and Nero 7.
I can only guess that this is caused by eac3to using dirty hacks which might not work that well in Vista64. Please try the next eac3to build. I've replaced the dirty hacks with a cleaner approach. That might work in Vista64.

[madshi]

eac3to crashes when trying input.thd output.wavs with nero filters.
Fixed in next build.

[madshi]

The real file size is 0 Bytes.
There seems to be something wrong with your Nero installation. I'd suggest that you uninstall and reinstall Nero from scratch. If that still doesn't work please contact Nero support.

[madshi]

eac3to v2.04 released

http://madshi.net/eac3to.zip

* don't need dtsac3source.ax, anymore
* don't need Nero Splitter, anymore
* don't need Sonic HD Demuxer, anymore
* replaced hacked DirectShow feeding with a cleaner approach
* added support for DTS-HD Master Audio 7.1 tracks (only 5.1 decoding)
* little performance boost for PAL speedup/down on DualCore CPUs
* fixed some bugs

[Beastie Boy]

Excellent update! Thank you.

[ACrowley]

eac3to v2.04 released

http://madshi.net/eac3to.zip

* don't need dtsac3source.ax, anymore
* don't need Nero Splitter, anymore
* don't need Sonic HD Demuxer, anymore
* replaced hacked DirectShow feeding with a cleaner approach
* added support for DTS-HD Master Audio 7.1 tracks (only 5.1 decoding)
* little performance boost for PAL speedup/down on DualCore CPUs
* fixed some bugs

NICE ! THX

[Thunderbolt8]

thanks for the next update :)

just to clarify again, truehd 7.1 input isnt supported at all currently and dts-hd ma is, but the outcome will be 5.1?

[madshi]

just to clarify again, truehd 7.1 input isnt supported at all currently and dts-hd ma is, but the outcome will be 5.1?
Everything is supported. But output is sometimes only 5.1.

[Thunderbolt8]

is it actually possible that the TrueHD track from a clockwork orange is only a 16-bit track, despite whats said everywhere? I encoded it to flac with this version, the former version and also with 1.23 and each time the rest 8-bit were only zero information bits.

[madshi]

is it actually possible that the TrueHD track from a clockwork orange is only a 16-bit track, despite whats said everywhere? I encoded it to flac with this version, the former version and also with 1.23 and each time the rest 8-bit were only zero information bits.
Sure it's possible. How big is the TrueHD file and what is the runtime?

[shambles]

the new version fixes the problem i had.. thank you so much :)

[madshi]

the new version fixes the problem i had.. thank you so much :)
Good to hear! :)

[Thunderbolt8]

Sure it's possible. How big is the TrueHD file and what is the runtime?
1.32 gb for 2hours 16min

[madshi]

1.32 gb for 2hours 16min
Definitely 16 bit.

[madshi]

eac3to v2.05 released

http://madshi.net/eac3to.zip

* added support for libav/ffmpeg decoding of TrueHD/MLP and E-AC3
* added "-libav" switch to force libav decoding
The libav TrueHD decoder seems to work well. In my quick tests it gave out bit perfect results compared to Nero's decoder. However, with a 24bit TrueHD sample the libav decoder complained about a failing "lossless check". Still the data was bit perfect. But the complaint shows that it might still make sense to keep using the Nero TrueHD decoder for a while - at least for comparison purposes. However, we might have a chance to get 7.1 TrueHD decoding now with the libav decoder. Does anybody have a sample of a 7.1 TrueHD track?

The libav E-AC3 decoder is not useable yet IMHO. It sometimes gives out distorted sound for me. I've still enabled it so that you can play with it, if you like. Another problem of the libav E-AC3 decoder is that it always forcefully shortens the data to 16 bit.

[Thunderbolt8]

However, we might have a chance to get 7.1 TrueHD decoding now with the libav decoder. Does anybody have a sample of a 7.1 TrueHD track?
I sent you that batman begins 7.1 truehd sample that day
http://www.megaupload.com/de/?d=ZGIL8AF6

[madshi]

I sent you that batman begins 7.1 truehd sample that day
http://www.megaupload.com/de/?d=ZGIL8AF6
Unfortunately it's not 7.1 but only 5.1... :(

[Thunderbolt8]

ew really? according to evodemux or the real data inside?
maybe its not possible for evodemux to demux or display the tracks properties properly? maybe pelican9 needs to update :S

this is what evodemux says:

mainwindow:
Dolby TrueHD audio stream 1 found!
First PTS = 000053C3 (+200ms)
Substream id = B1
TrueHD stream (up to 8 channels)
Sampling frequency = 48 kHz
2 ch. decoder
channel modifier = Stereo
6 ch. decoder
channel modifier = Lt/Rt
channel arrangement = Main (Left,Right), Centre, LFE, Surrounds (Ls/Rs)
8 ch. decoder
channel modifier = Lt/Rt
channel arrangement = Main (Left,Right), Centre, LFE, Surrounds (Ls/Rs)
Dynamic range control = -10,6371 dB .. 8,4282 dB


VTI:
Channels: 6

[Wilbert]

The libav TrueHD decoder seems to work well. In my quick tests it gave out bit perfect results compared to Nero's decoder. However, with a 24bit TrueHD sample the libav decoder complained about a failing "lossless check". Still the data was bit perfect.
Afaik 24 bit output is not supported in libav, but it could be put in 32 bit. Did you get that working?

Kurtnoise13 couldn't get it to work in ffmpegsource: http://www.mediafire.com/?1we0uonvjm0

[Thunderbolt8]

32-bit should be really big I guess :S

[nautilus7]

I've checked. It's intended behaviour. The problem is that there's not a CRC error in those 2 frames. Instead there's not even a valid header for those frames. So delaycut doesn't know what to do and sees no other chance than to skip forward to the next valid header. delaycut was implemented to behave this way. Those modes work only if the header (actually only the first 2 bytes!) is valid.
So, is there anything i can do? Can i fix the header?

Thanks for the nice updates you 've made!

[Blue_MiSfit]

Confirmed. Using ffmpeg to decode EAC3 I get all sorts of nasty high frequency artifacts. Not good! :)

I hope this matures!

~MiSfit

[madshi]

ew really? according to evodemux or the real data inside?
According to the real data inside.

maybe its not possible for evodemux to demux or display the tracks properties properly? maybe pelican9 needs to update :S

this is what evodemux says:

mainwindow:
Dolby TrueHD audio stream 1 found!
First PTS = 000053C3 (+200ms)
Substream id = B1
TrueHD stream (up to 8 channels)
Sampling frequency = 48 kHz
2 ch. decoder
channel modifier = Stereo
6 ch. decoder
channel modifier = Lt/Rt
channel arrangement = Main (Left,Right), Centre, LFE, Surrounds (Ls/Rs)
8 ch. decoder
channel modifier = Lt/Rt
channel arrangement = Main (Left,Right), Centre, LFE, Surrounds (Ls/Rs)
Dynamic range control = -10,6371 dB .. 8,4282 dB

VTI:
Channels: 6
Where does EvoDemux say that it's a 7.1 track? The 8 channel decoder settings say "Main, Centre, LFE, Surrounds" which is 5.1. So it's a 5.1 track.

[madshi]

Afaik 24 bit output is not supported in libav, but it could be put in 32 bit. Did you get that working?

Kurtnoise13 couldn't get it to work in ffmpegsource: http://www.mediafire.com/?1we0uonvjm0
I got 32 bit output working (only from the TrueHD decoder, not from the E-AC3 decoder) and internally converted it down to 24 bit. That gives me bit perfect results compared to Nero's decoder for both 16 bit and 24 bit TrueHD tracks.

[madshi]

32-bit should be really big I guess :S
Well, there's only 24 bits of real data (or even only 16 bits), so those 32 bit can be cut down to 24 bit without any loss of "real" data. And that's what eac3to does with the output of the libav TrueHD decoder.

[madshi]

So, is there anything i can do? Can i fix the header?
Well, you could check the frame size (in bytes) in delaycut. Then with a hex editor you can search for the frame where the first two bytes of the frame are not correct and manually correct them. Of course this only works if the "garbage" in your track is exactly one frame long. If it's real added garbage, it'd be better to skip that data. If it's not real garbage, but just a corrupted frame then it might be better to silence the frame.

[MichalHabart]

Hello Madshi,

thanks for your great tool. I have only one question, option -bluray is needed alway when i use as source PCM file xported from BluRay Disk? Because remember that option was in some early version of EAC3 tool but then it dissapeared and was done automatically. Now it is again manual and nothing is done automatically?

[madshi]

thanks for your great tool. I have only one question, option -bluray is needed alway when i use as source PCM file xported from BluRay Disk? Because remember that option was in some early version of EAC3 tool but then it dissapeared and was done automatically. Now it is again manual and nothing is done automatically?
Good question. I should have explained it.

eac3to doesn't really know (and has no way to find out) in which order the channels are, if you feed it with a RAW or PCM file. The previous eac3to version made it simple: RAW input files were not supported. Only PCM files (files with the extension "pcm") were supported. Basically eac3to interpreted any PCM file input as being Blu-Ray and thus automatically remapped the channels accordingly.

But now the new eac3to version supports both PCM and RAW files (which are really one and the same). It's impossible to know whether the file came from Blu-Ray or not. So it's also impossible to know whether the channels need to be remapped or not. eac3to v2.x now does an educated guess: PCM tracks from Blu-Ray are usually big endian. RAW tracks decoded from any DirectShow (or other) decoder are usually little endian. So eac3to v2.x handles big endian RAW/PCM files as coming from Blu-Ray and little endian RAW/PCM files as not coming from Blu-Ray. As a result channels are remapped according to Blu-Ray specs when the source file is big endian. And channels are NOT remapped at all when the source file is little endian. If the channels are remapped eac3to v2.x sais so (something like "this is probably a Blu-Ray PCM file, so I'll remap the channels").

If you demux Blu-Ray (L)PCM tracks with either xport or TsRemux the demuxed files will be big endian. In this case you don't need to tell eac3to anything. Just call it as usual with "eac3to source.pcm dest.whatever" and the channels will be automatically remapped correctly. However, if you demux Blu-Ray (L)PCM tracks by using GraphEdit -> SomeSplitter -> Dump, the resulting file will be little endian. As a result eac3to will think that this file is not coming from Blu-Ray. In this case you can use the reintroduced switch "-blu-ray" to force eac3to to remap the channels. But you need to be cautious: The DirectShow splitter might already have remapped the channels - or not.

Short version: If you use xport or TsRemux for demuxing Blu-Ray (L)PCM tracks, you don't need the "-blu-ray" switch. eac3to v2.x will still automatically remap the channels correctly for you.

[tebasuna51]

So, is there anything i can do? Can i fix the header?

Like Madshi say you can use a hex editor (WinHex or the free HxD (http://www.mh-nexus.de)) to try to correct them.

With your sample you can correct the problem if you:

1) Open test.eac3 in HxD and go to Offset(d) 2840576 (or Offset(h) 2B5800). The offset is (Frame reported by DelayCut -1) * FrameSize = (2775 - 1) * 1024 = 2840576

2) Replace the values in the sample: DB 00 0A D0
by the correct values: 0B 77 01 FF
and save the file.

3) Now you can check test.eac3 with DelayCut with no errors, then the CRC are ok.

[honai]

@madshi

I've extracted the PCM from POTC3 with xport (-h) and used the -bluray switch to convert it to WAVs. The channel mapping in the end is correct. But shouldn't it be messed up now according to your explanation?

[menlvd]

i try to reencode ddp track with -libav option and get bad sound
but when reencode with nero have a normal sound
I'll upload short cut off resulting file

2 minutes (http://rapidshare.com/files/71931453/pay-libav_bad.ac3)

[MichalHabart]

Good question. I should have explained it.

eac3to doesn't really know (and has no way to find out) in which order the channels are, if you feed it with a RAW or PCM file. The previous eac3to version made it simple: RAW input files were not supported. Only PCM files (files with the extension "pcm") were supported. Basically eac3to interpreted any PCM file input as being Blu-Ray and thus automatically remapped the channels accordingly.

But now the new eac3to version supports both PCM and RAW files (which are really one and the same). It's impossible to know whether the file came from Blu-Ray or not. So it's also impossible to know whether the channels need to be remapped or not. eac3to v2.x now does an educated guess: PCM tracks from Blu-Ray are usually big endian. RAW tracks decoded from any DirectShow (or other) decoder are usually little endian. So eac3to v2.x handles big endian RAW/PCM files as coming from Blu-Ray and little endian RAW/PCM files as not coming from Blu-Ray. As a result channels are remapped according to Blu-Ray specs when the source file is big endian. And channels are NOT remapped at all when the source file is little endian. If the channels are remapped eac3to v2.x sais so (something like "this is probably a Blu-Ray PCM file, so I'll remap the channels").

If you demux Blu-Ray (L)PCM tracks with either xport or TsRemux the demuxed files will be big endian. In this case you don't need to tell eac3to anything. Just call it as usual with "eac3to source.pcm dest.whatever" and the channels will be automatically remapped correctly. However, if you demux Blu-Ray (L)PCM tracks by using GraphEdit -> SomeSplitter -> Dump, the resulting file will be little endian. As a result eac3to will think that this file is not coming from Blu-Ray. In this case you can use the reintroduced switch "-blu-ray" to force eac3to to remap the channels. But you need to be cautious: The DirectShow splitter might already have remapped the channels - or not.

Short version: If you use xport or TsRemux for demuxing Blu-Ray (L)PCM tracks, you don't need the "-blu-ray" switch. eac3to v2.x will still automatically remap the channels correctly for you.

Thank you for very good explanation.

[nautilus7]

Well, you could check the frame size (in bytes) in delaycut. Then with a hex editor you can search for the frame where the first two bytes of the frame are not correct and manually correct them. Of course this only works if the "garbage" in your track is exactly one frame long. If it's real added garbage, it'd be better to skip that data. If it's not real garbage, but just a corrupted frame then it might be better to silence the frame.I believe the garbage is 1 frame long only, 2 frames with problem. Silence doesn't work and the frames are always skipped as you told me before because they are not valid.

Like Madshi say you can use a hex editor (WinHex or the free HxD (http://www.mh-nexus.de)) to try to correct them.

With your sample you can correct the problem if you:

1) Open test.eac3 in HxD and go to Offset(d) 2840576 (or Offset(h) 2B5800). The offset is (Frame reported by DelayCut -1) * FrameSize = (2775 - 1) * 1024 = 2840576

2) Replace the values in the sample: DB 00 0A D0
by the correct values: 0B 77 01 FF
and save the file.

3) Now you can check test.eac3 with DelayCut with no errors, then the CRC are ok.

The correct value for the frame header is the same for every frame? I mean do i put the same value for the other bad frame in the track?

I 'll try it and report back. Hope it works!

EDIT: I corrected the 2 frames and now eac3to seems to accept the track properly! Thank you both very much!

[madshi]

I've extracted the PCM from POTC3 with xport (-h) and used the -bluray switch to convert it to WAVs. The channel mapping in the end is correct. But shouldn't it be messed up now according to your explanation?
No, why? "xport"ed PCM tracks are big endian. So eac3to enabled the "-blu-ray" switch automatically. You manually enabled that switch, too, which had no effect, since it was already automatically enabled by eac3to.

[madshi]

i try to reencode ddp track with -libav option and get bad sound
I'm not in a position to fix the sound. Have you read what I wrote in the eac3to v2.05 release notes?

The libav E-AC3 decoder is not useable yet IMHO. It sometimes gives out distorted sound for me. I've still enabled it so that you can play with it, if you like.

[honai]

No, why?

It seemed from your first explanation that the -blu-ray switch also forcefully remaps channels (in which case it'd have messed up the order because it was correct to start with). Obviously that's not the case.

[madshi]

It seemed from your first explanation that the -blu-ray switch also forcefully remaps channels (in which case it'd have messed up the order because it was correct to start with). Obviously that's not the case.
No, the order was not correct to start with. It needed to be corrected. But of course eac3to is not that stupid that it executes the Blu-Ray channel remapping twice on the same data... :)

[honai]

Hmm. How exactly does eac3to determine the channel order in the source PCM? By some heuristic on the presumed LFE channel? Or is -blu-ray always enabled for all .pcm content? This is really confusing.

[madshi]

Hmm. How exactly does eac3to determine the channel order in the source PCM? By some heuristic on the presumed LFE channel? Or is -blu-ray always enabled for all .pcm content? This is really confusing.
Please read my very long explanation I posted as a reply to MichalHabart's question. It explains what happens fairly extensively. Basically "-blu-ray" is automatically enabled for big endian raw/pcm files.

[honai]

I did that, and from what I've read so far it seems that eac3to will remap channels if the source is big-endian, regardless of the actual channel order. Which means that if the source PCM happens to be big-endian but already has the correct channel order, eac3to will mess it up implicitly. I don't think that's desirable behavior. The endianness and channel ordering should be decoupled.

This is all assuming that you still want to develop eac3to as a general audio tool, in which case the above educated guesses about the format of the source could very well be wrong for a great number of audio sources.

Concrete suggestion: Change the "-blu-ray" option into "-xport" (or something to that effect) to indicate that the source is from xport and is big-endian *and* has Blu-ray channel order.

[madshi]

I did that, and from what I've read so far it seems that eac3to will remap channels if the source is big-endian, regardless of the actual channel order. Which means that if the source PCM happens to be big-endian but already has the correct channel order, eac3to will mess it up implicitly. I don't think that's desirable behavior. The endianness and channel ordering should be decoupled.

This is all assuming that you still want to develop eac3to as a general audio tool, in which case the above educated guesses about the format of the source could very well be wrong for a great number of audio sources.

Concrete suggestion: Change the "-blu-ray" option into "-xport" (or something to that effect) to indicate that the source is from xport and is big-endian *and* has Blu-ray channel order.
Well, I respect your point of view, but mine happens to differ. eac3to was written mainly as a tool for HD DVD and Blu-Ray tracks. Ok, it does more than that right now, but the primary focus is still to make converting HD DVD and Blu-Ray tracks as easy as possible. I don't want people to remember having to add a "-blu-ray" or "-xport" switch everytime they convert a Blu-Ray (L)PCM track.

Please name a few multichannel sources which are big endian and do *not* have the Blu-Ray channel ordering. Are there any? I'm not aware of any such sources. Because of that I don't think there's any danger in automatically activating the "-blu-ray" switch for big endian PCM files. You can try to convince me otherwise, but you'll need good arguments... :) E.g. it would help if you could say where a big endian PCM track with WAV channel order could come from.

Of course you can force eac3to to not remap channels by overwriting the channel mapping matrix (e.g. "eac3to source.pcm dest.whatever -0,1,2,3,4,5").

[Thunderbolt8]

According to the real data inside.
ok, is this now 100% sure? or could it be because maybe there wasnt any data for the other 2 channels in that sample I gave = would it help if I provided another sample?

[madshi]

ok, is this now 100% sure? or could it be because maybe there wasnt any data for the other 2 channels in that sample I gave = would it help if I provided another sample?
It's 100% sure, as far as I can say.

Besides, I've searched and didn't find a single HD DVD or Blu-Ray disc with a 7.1 TrueHD. It seems such a disc doesn't exist yet. I also didn't find any disc with a 7.1 E-AC3 track.

[Thunderbolt8]

good, then trueHd decoding seems to be complete :D

[Snowknight26]

So I tried converting the Poseidon TrueHD track (24bit) to DTS and FLAC, but I noticed different lengths. I then compared it to my previous conversions, and the DTS length was different yet again.

eac3to v2.05:
DTS, 5.1 channels, 1:38:44, 24 bits, 1536kbit/s, 48khz
FLAC, 5.1 channels, 1:38:13, 16 bits, 48khz (-down16 was used)

eac3to v1.23:
DTS, 5.1 channels, 1:40:18, 24 bits, 1536kbit/s, 48khz
FLAC, 5.1 channels, 1:38:13, 16 bits, 48khz (-down16 was used)

How do I know which one is the proper one (if any), and why do the DTS transcodes have varying length?

Edit:
I also compared a different DD+ track to various encodes. Source is "E-AC3, 5.1 channels, 1:47:35, 1536kbit/s, 48khz, dialnorm: -27dB" according to eac3to v2.05. Then just for the heck of it, I converted that to AC3 using v2.05.

eac3to v2.05:
DTS, 5.1 channels, 1:48:09, 24 bits, 1536kbit/s, 48khz
AC3, 5.1 channels, 1:47:35, 640kbit/s, 48khz

eac3to v1.23:
DTS, 5.1 channels, 1:49:52, 24 bits, 1536kbit/s, 48khz

:\

[Thunderbolt8]

a question regarding that trueHD syncing, could it be possible that the track (flac from truehd) is always in sync now, even if evodemux states a delay for it regarding different first pts?
im trying to remux batman begins and according to evodemux the track should need -200ms of delay. but frankly said some scenes are like this and some (actually most) fit better without that -200. thats why Im asking if with that sync fix something like this could have happened as (unintended?) side effect. or could it possibly be that the delay is different for the 1st and 2nd evo in this special case? some situations definately look a bit differently regarding sync compared to playback of the disc with powerdvd.

Opening file PEVOB_1.EVO
Reading...
File size: 11205 Mbytes.
VOB number 4 contains 2 video , 5 audio and 4 subpicture streams.
PTM of first video frame = 00000D61
PTM of last video frame = 19B28220
Duration = 1:19:50.286
VC-1 video stream 0 found!
First PTS = 00000D61
Dolby TrueHD audio stream 1 found!
First PTS = 000053C3 (+200ms)

Opening file PEVOB_2.EVO
Reading...
File size: 7732 Mbytes.
VOB number 5 contains 2 video , 5 audio and 4 subpicture streams.
PTM of first video frame = 19B28220
PTM of last video frame = 2D139A61
Duration = 1:00:12.559
VC-1 video stream 0 found!
First PTS = 19B28220 (+4790285ms)
Dolby TrueHD audio stream 1 found!
First PTS = 19B2659A (+4790204ms)

ive just noticed that compared to powerdvd the playtime is 8 seconds longer, but theres no other delay specific problem other than the one I related to (meaning the delay is not off by 8 seconds). still, could it be possible that the framerate is slightly different which could result to those slight sync differences?

[Snowknight26]

I think the delay between the audio and video stream is different between the 1st and 2nd EVO. That might be causing my problems too, as the 2nd test I did had EVOs like what I just described.

According to your post:

1st EVO's delay is +200. (200-0=+200)
2nd EVO's delay is -81. (4790204-4790285=-81)

What it seems is that during the middle of the movie, the audio would be off by 119ms (or 281ms, can't think, heads spinning :p).

But then again, maybe I'm totally wrong.

[Thunderbolt8]

I guess if then it would just change at once, with the beginning of the 2nd evo, but there wouldnt be nothing like an average value of both evos. I know that madshi said the value of the 2nd .evo actually dont matter. but maybe during the last updates something happened unintentionally, dont know :S or that disc is just different

[moshmothma]

Any chance eac3to could support mlp from DVD-A? I get an error when I try to decode it. thanks

[killa_kid]

from what i understand of .evo (could be wrong), they are actualy 1 file that is then split (for whatever reason).

Since they are in fact 1 file, that is played as necessary, it is possible a portion of the audio (less then a second) is actually in the 1st file (provided the delay causes this to happen). i have ripped 11 of my HD-DVDs by using DumpHD (AnyDVD for transformers) and then using EVOdemux to demux them into a video and audio file. Then I use only the delay from the 1st evo file. This has resulted in no problems in sync.

When I tried to do them in 2 parts or other methods I had many issues mostly involving audio drift. This method has worked for me, and my understanding could be wrong, but you can always give it a shot and see what happens

PS: I have ripped Batman Begins with TrueHD and got a -200ms delay not +200ms and I have no sync errors. Since you have to subtract the audio delay from the video (since a positive delay means starting to audio a slight bit later, very prominent in The Bourne Supremacy it is 1001ms late)

[Thunderbolt8]

i also applied -200ms, not +200 of course. this would be way off

still, even though it should be mathematically correct, its still a bit off in quite some scenes, noticeably (otherwise I wouldnt have noticed of course). it sometimes differes from the way it looks when playing the movie via powerdvd (yes, I have selected the right track).

[killa_kid]

interesting....i had issues of a similar nature the way i resolved them was by encoding the movie at the 30000/1001 frame rate (the disc default), but when I mux it to mkv i set the frame rate to 24000/1001 (the actual frame rate). may not be the optimal solution, but it worked for me.

[madshi]

So I tried converting the Poseidon TrueHD track (24bit) to DTS and FLAC, but I noticed different lengths. I then compared it to my previous conversions, and the DTS length was different yet again.

eac3to v2.05:
DTS, 5.1 channels, 1:38:44, 24 bits, 1536kbit/s, 48khz
FLAC, 5.1 channels, 1:38:13, 16 bits, 48khz (-down16 was used)

eac3to v1.23:
DTS, 5.1 channels, 1:40:18, 24 bits, 1536kbit/s, 48khz
FLAC, 5.1 channels, 1:38:13, 16 bits, 48khz (-down16 was used)

How do I know which one is the proper one (if any), and why do the DTS transcodes have varying length?

Edit:
I also compared a different DD+ track to various encodes. Source is "E-AC3, 5.1 channels, 1:47:35, 1536kbit/s, 48khz, dialnorm: -27dB" according to eac3to v2.05. Then just for the heck of it, I converted that to AC3 using v2.05.

eac3to v2.05:
DTS, 5.1 channels, 1:48:09, 24 bits, 1536kbit/s, 48khz
AC3, 5.1 channels, 1:47:35, 640kbit/s, 48khz

eac3to v1.23:
DTS, 5.1 channels, 1:49:52, 24 bits, 1536kbit/s, 48khz

:\
That's interesting! Are the DTS tracks in sync when playing them with the video? Not sure where the problem comes from. Maybe Surcode does something wrong?

Wait... Have you encoded the DTS tracks with eac3to's Surcode automation? Or have you manually started Surcode? Or did you use a different DTS encoder?

Another question: Where did you get the runtime information from? Did you throw the v2.05 and v1.23 created DTS files into eac3to v2.05 to get the runtime information? Or did you feed the same DTS file into both v2.05 and v1.23 and they report different lengths for the same track?

[madshi]

a question regarding that trueHD syncing, could it be possible that the track (flac from truehd) is always in sync now, even if evodemux states a delay for it regarding different first pts?
I don't think so. At least that didn't work for me with Bourne Supremecy. Interestingly, with Bourne Supremecy I had to use a delay of 1200-1250ms instead of 1001ms to get good audio sync! I'm wondering whether we need to recalculate the delay by doing "realDelay = evoDemuxDelay * 29.97 / 23.976"?

im trying to remux batman begins and according to evodemux the track should need -200ms of delay.
No, that should be +200ms, just like EvoDemux sais!

[madshi]

I know that madshi said the value of the 2nd .evo actually dont matter.
Right.

It may be different if you reencode both video and audio fo the first and second EVO files separately and then join the reencoded audio and video files. But that'd be a bad idea to start with!

[madshi]

Any chance eac3to could support mlp from DVD-A? I get an error when I try to decode it. thanks
Can you upload a 5MB sample, please?

[shambles]

when removing dialnorm from ac3 tracks, eac3to sets the value to -0db.. shouldn't the value be -31db?

[Thunderbolt8]

No, that should be +200ms, just like EvoDemux sais!
its definately not +200. i applied -200 in the flac and it was fine in some situations when I left it that way in mpc with 0 (=-200) or set +200 (=0, compared to the original state) for other situations.

[xkodi]

Can you upload a 5MB sample, please?

http://samples.mplayerhq.hu/mlp/

http://samples.mplayerhq.hu/DVD-Audio/

http://www.rarewares.org/rja/samples/

[killa_kid]

I'm wondering whether we need to recalculate the delay by doing "realDelay = evoDemuxDelay * 29.97 / 23.976"?

well i checked EVOdemux, and some of the movies are at different frame rates then the actual movie (eg: 59.94) so if you factor that in to the occasion (obviously using the correct source and actual frame rate) you can then do the math.

Out of my 16 HD DVDs, I only have 5 which need a delay (positive or negative) and with doing the math this way I only found 1 where this had an effect (Transformers). Later today I'll try ripping with my previously determined and newly calculated delays and see if it makes a difference or not.

[drmpeg]

I've updated xport to remove the sequence end codes from the demuxed video elementary stream. This avoids having to use TSRemux just for this purpose on concatenated Blu-ray .m2ts files.

Version 1.00

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

I've also looked at the A/V sync issues with seamless branching titles (where multiple .m2ts files are joined together). xport is working correctly, but there are issues with the way the segments are authored. Basically, there's too much audio at the end of each segment. Unfortunately, it can't be avoided, since each segment must end with a full PES packet for audio.

This means there can be as much as 5 milliseconds of extra LPCM audio and as much as 32 milliseconds of extra AC3 audio per segment.

I've also cleaned up the PTS/DTS dumping, so that it can be used to show the extra audio. Here's an example:

C:\xfer>xport -havs 00017tail.m2ts 1 1 1
xport Transport Stream Demuxer 1.00
program = 1, video channel = 1, audio channel = 1
Transport Sync Error, skipped 64 bytes, at 0
Program Number = 0 (0x0000), Program Map PID = 31 (0x001f)
Program Number = 1 (0x0001), Program Map PID = 256 (0x0100)
program descriptor = 0x05, 0x04, 0x48, 0x44, 0x4d, 0x56
program descriptor = 0x88, 0x04, 0x0f, 0xff, 0xff, 0xfc
Video PID = 4113 <0x1011>, type = 0x1b
ES descriptor for stream type 0x1b = 0x05, 0x08, 0x48, 0x44, 0x4d, 0x56, 0xff, 0x1b, 0x61, 0x3f
Audio PID = 4352 <0x1100>, type = 0x80
ES descriptor for stream type 0x80 = 0x05, 0x08, 0x48, 0x44, 0x4d, 0x56, 0xff, 0x80, 0x61, 0xff
ES descriptor for stream type 0x81 = 0x05, 0x04, 0x41, 0x43, 0x2d, 0x33
ES descriptor for stream type 0x81 = 0x81, 0x04, 0x06, 0x48, 0x0e, 0x00
ES descriptor for stream type 0x81 = 0x05, 0x04, 0x41, 0x43, 0x2d, 0x33
ES descriptor for stream type 0x81 = 0x81, 0x04, 0x06, 0x48, 0x0e, 0x00
ES descriptor for stream type 0x81 = 0x05, 0x04, 0x41, 0x43, 0x2d, 0x33
ES descriptor for stream type 0x81 = 0x81, 0x04, 0x06, 0x48, 0x0e, 0x00
ES descriptor for stream type 0x81 = 0x05, 0x04, 0x41, 0x43, 0x2d, 0x33
ES descriptor for stream type 0x81 = 0x81, 0x04, 0x06, 0x29, 0x04, 0x00
Audio PTS = 2529960, 2529960
LPCM Audio Mode = 3/2+lfe
LPCM Audio Bits/sample = 24
LPCM Audio Sample Rate = 48000
Audio PTS = 2530410, 450
Audio PTS = 2530860, 450
Audio PTS = 2531310, 450
Audio PTS = 2531760, 450
Audio PTS = 2532210, 450
Audio PTS = 2532660, 450
Audio PTS = 2533110, 450
Audio PTS = 2533560, 450
Audio PTS = 2534010, 450
Audio PTS = 2534460, 450
Audio PTS = 2534910, 450
Video PTS(B) = 2553813
Audio PTS = 2535360, 450
Audio PTS = 2535810, 450
Audio PTS = 2536260, 450
Video PTS(B) = 2557567, 3754
Audio PTS = 2536710, 450
Audio PTS = 2537160, 450
Audio PTS = 2537610, 450
Audio PTS = 2538060, 450
Audio PTS = 2538510, 450
Audio PTS = 2538960, 450
Audio PTS = 2539410, 450
Audio PTS = 2539860, 450
Audio PTS = 2540310, 450
Audio PTS = 2540760, 450
Audio PTS = 2541210, 450
Audio PTS = 2541660, 450
Audio PTS = 2542110, 450
Audio PTS = 2542560, 450
Audio PTS = 2543010, 450
Audio PTS = 2543460, 450
Audio PTS = 2543910, 450
Video PTS(P) = 2572582, DTS(P) = 2561321, 3754
Audio PTS = 2544360, 450
Audio PTS = 2544810, 450
Audio PTS = 2545260, 450
Audio PTS = 2545710, 450
Audio PTS = 2546160, 450
Audio PTS = 2546610, 450
Audio PTS = 2547060, 450
Video PTS(B) = 2565075, 3754
Audio PTS = 2547510, 450
Audio PTS = 2547960, 450
Audio PTS = 2548410, 450
Audio PTS = 2548860, 450
Video PTS(B) = 2568828, 3753
Audio PTS = 2549310, 450
Video PTS(P) = 2583843, DTS(P) = 2572582, 3754
Audio PTS = 2549760, 450
Audio PTS = 2550210, 450
Audio PTS = 2550660, 450
<big snip>
.
.
Video PTS(P) = 2737747, DTS(P) = 2726486, 3754
Audio PTS = 2701410, 450
Audio PTS = 2701860, 450
Audio PTS = 2702310, 450
Audio PTS = 2702760, 450
Audio PTS = 2703210, 450
Audio PTS = 2703660, 450
Audio PTS = 2704110, 450
Audio PTS = 2704560, 450
Video PTS(B) = 2730240, 3754
Audio PTS = 2705010, 450
Audio PTS = 2705460, 450
Audio PTS = 2705910, 450
Audio PTS = 2706360, 450
Audio PTS = 2706810, 450
Audio PTS = 2707260, 450
Audio PTS = 2707710, 450
Audio PTS = 2708160, 450
Audio PTS = 2708610, 450
Video PTS(B) = 2733993, 3753
Audio PTS = 2709060, 450
Audio PTS = 2709510, 450
Audio PTS = 2709960, 450
Audio PTS = 2710410, 450
Audio PTS = 2710860, 450
Audio PTS = 2711310, 450
Audio PTS = 2711760, 450
Audio PTS = 2712210, 450
Video PTS(P) = 2749008, DTS(P) = 2737747, 3754 <-- last video PTS
Audio PTS = 2712660, 450
Audio PTS = 2713110, 450
Audio PTS = 2713560, 450
Audio PTS = 2714010, 450
Audio PTS = 2714460, 450
Audio PTS = 2714910, 450
Audio PTS = 2715360, 450
Audio PTS = 2715810, 450
Audio PTS = 2716260, 450
Video PTS(B) = 2741501, 3754
Audio PTS = 2716710, 450
Audio PTS = 2717160, 450
Audio PTS = 2717610, 450
Audio PTS = 2718060, 450
Audio PTS = 2718510, 450
Audio PTS = 2718960, 450
Audio PTS = 2719410, 450
Audio PTS = 2719860, 450
Video PTS(B) = 2745255, 3754
Audio PTS = 2720310, 450
Audio PTS = 2720760, 450
Audio PTS = 2721210, 450
Audio PTS = 2721660, 450
Audio PTS = 2722110, 450
Audio PTS = 2722560, 450
Audio PTS = 2723010, 450
Audio PTS = 2723460, 450
Audio PTS = 2723910, 450
Audio PTS = 2724360, 450
Audio PTS = 2724810, 450
Audio PTS = 2725260, 450
Audio PTS = 2725710, 450
Audio PTS = 2726160, 450
Audio PTS = 2726610, 450
Audio PTS = 2727060, 450
Audio PTS = 2727510, 450
Audio PTS = 2727960, 450
Audio PTS = 2728410, 450
Audio PTS = 2728860, 450
Audio PTS = 2729310, 450
Audio PTS = 2729760, 450
Audio PTS = 2730210, 450
Audio PTS = 2730660, 450
Audio PTS = 2731110, 450
Audio PTS = 2731560, 450
Audio PTS = 2732010, 450
Audio PTS = 2732460, 450
Audio PTS = 2732910, 450
Audio PTS = 2733360, 450
Audio PTS = 2733810, 450
Audio PTS = 2734260, 450
Audio PTS = 2734710, 450
Audio PTS = 2735160, 450
Audio PTS = 2735610, 450
Audio PTS = 2736060, 450
Audio PTS = 2736510, 450
Audio PTS = 2736960, 450
Audio PTS = 2737410, 450
Audio PTS = 2737860, 450
Audio PTS = 2738310, 450
Audio PTS = 2738760, 450
Audio PTS = 2739210, 450
Audio PTS = 2739660, 450
Audio PTS = 2740110, 450
Audio PTS = 2740560, 450
Audio PTS = 2741010, 450
Audio PTS = 2741460, 450
Audio PTS = 2741910, 450
Audio PTS = 2742360, 450
Audio PTS = 2742810, 450
Audio PTS = 2743260, 450
Audio PTS = 2743710, 450
Audio PTS = 2744160, 450
Audio PTS = 2744610, 450
Audio PTS = 2745060, 450
Audio PTS = 2745510, 450
Audio PTS = 2745960, 450
Audio PTS = 2746410, 450
Audio PTS = 2746860, 450
Audio PTS = 2747310, 450
Audio PTS = 2747760, 450
Audio PTS = 2748210, 450
Audio PTS = 2748660, 450
Audio PTS = 2749110, 450
Audio PTS = 2749560, 450
Audio PTS = 2750010, 450
Audio PTS = 2750460, 450
Audio PTS = 2750910, 450
Audio PTS = 2751360, 450
Audio PTS = 2751810, 450
Audio PTS = 2752260, 450
Audio PTS = 2752710, 450 <--- last audio PTS

packets for pid 0 <0x0000> = 31, first = 1209, last = 52013
packets for pid 31 <0x001f> = 3, first = 8231, last = 47155
packets for pid 256 <0x0100> = 31, first = 1210, last = 52014
packets for pid 4097 <0x1001> = 32, first = 2486, last = 52045
packets for pid 4113 <0x1011> = 33190, first = 2, last = 49311
packets for pid 4352 <0x1100> = 12096, first = 1, last = 51538
packets for pid 4353 <0x1101> = 1362, first = 243, last = 52079
packets for pid 4354 <0x1102> = 1362, first = 245, last = 52080
packets for pid 4355 <0x1103> = 1362, first = 247, last = 52081
packets for pid 4356 <0x1104> = 372, first = 249, last = 50634
packets for pid 4611 <0x1203> = 2240, first = 3807, last = 48772
packets for pid 8191 <0x1fff> = 2, first = 52082, last = 52083
coded pictures = 52, video fields = 0

C:\xfer>

This is a demux of the last 10 Megabytes of a segment from a seamless branching movie. The last audio PTS is 2752710 and the last video PTS is 2749008. The video ends at 2749008 + 3754 = 2752762. The audio ends at 2752710 + 450 = 2753160. 2753160 - 2752762 = 398. 398 / 90000 = 4.42 milliseconds too much audio.

Here's the AC3 track:

C:\xfer>xport -havs 00017tail.m2ts 1 1 2
xport Transport Stream Demuxer 1.00
program = 1, video channel = 1, audio channel = 2
Transport Sync Error, skipped 64 bytes, at 0
Program Number = 0 (0x0000), Program Map PID = 31 (0x001f)
Program Number = 1 (0x0001), Program Map PID = 256 (0x0100)
program descriptor = 0x05, 0x04, 0x48, 0x44, 0x4d, 0x56
program descriptor = 0x88, 0x04, 0x0f, 0xff, 0xff, 0xfc
Video PID = 4113 <0x1011>, type = 0x1b
ES descriptor for stream type 0x1b = 0x05, 0x08, 0x48, 0x44, 0x4d, 0x56, 0xff, 0x1b, 0x61, 0x3f
ES descriptor for stream type 0x80 = 0x05, 0x08, 0x48, 0x44, 0x4d, 0x56, 0xff, 0x80, 0x61, 0xff
Audio PID = 4353 <0x1101>, type = 0x81
ES descriptor for stream type 0x81 = 0x05, 0x04, 0x41, 0x43, 0x2d, 0x33
ES descriptor for stream type 0x81 = 0x81, 0x04, 0x06, 0x48, 0x0e, 0x00
ES descriptor for stream type 0x81 = 0x05, 0x04, 0x41, 0x43, 0x2d, 0x33
ES descriptor for stream type 0x81 = 0x81, 0x04, 0x06, 0x48, 0x0e, 0x00
ES descriptor for stream type 0x81 = 0x05, 0x04, 0x41, 0x43, 0x2d, 0x33
ES descriptor for stream type 0x81 = 0x81, 0x04, 0x06, 0x48, 0x0e, 0x00
ES descriptor for stream type 0x81 = 0x05, 0x04, 0x41, 0x43, 0x2d, 0x33
ES descriptor for stream type 0x81 = 0x81, 0x04, 0x06, 0x29, 0x04, 0x00
Audio PTS = 2497200, 2497200
Audio PTS = 2500080, 2880
Audio Bitrate = 640000, Audio Sampling Rate = 48000
Audio Mode = 3/2, bsid = 6, bsmod = 0
Video PTS(B) = 2553813
Video PTS(B) = 2557567, 3754
Audio PTS = 2502960, 2880
Audio PTS = 2505840, 2880
Audio PTS = 2508720, 2880
Video PTS(P) = 2572582, DTS(P) = 2561321, 3754
Audio PTS = 2511600, 2880
Video PTS(B) = 2565075, 3754
Audio PTS = 2514480, 2880
Video PTS(B) = 2568828, 3753
Video PTS(P) = 2583843, DTS(P) = 2572582, 3754
Audio PTS = 2517360, 2880
Video PTS(B) = 2576336, 3754
Audio PTS = 2520240, 2880
Video PTS(B) = 2580090, 3754
Audio PTS = 2523120, 2880
Video PTS(P) = 2595105, DTS(P) = 2583843, 3753
Audio PTS = 2526000, 2880
Audio PTS = 2528880, 2880
Video PTS(B) = 2587597, 3754
Audio PTS = 2531760, 2880
Video PTS(B) = 2591351, 3754
Audio PTS = 2534640, 2880
Video PTS(P) = 2606366, DTS(P) = 2595105, 3754
Audio PTS = 2537520, 2880
Video PTS(B) = 2598858, 3753
Audio PTS = 2540400, 2880
Audio PTS = 2543280, 2880
Video PTS(B) = 2602612, 3754
Audio PTS = 2546160, 2880
Video PTS(P) = 2617627, DTS(P) = 2606366, 3754
Audio PTS = 2549040, 2880
Video PTS(B) = 2610120, 3754
Audio PTS = 2551920, 2880
Audio PTS = 2554800, 2880
Video PTS(B) = 2613873, 3753
Audio PTS = 2557680, 2880
Video PTS(P) = 2621381, DTS(P) = 2617627, 3754
17 frames before first I-frame
High Profile
Level = 4.1
Audio PTS = 2560560, 2880
Audio PTS = 2563440, 2880
Audio PTS = 2566320, 2880
Video PTS(P) = 2628888, DTS(P) = 2621381, 3754
First Video PTS = 0x0027ffc5
Audio PTS = 2569200, 2880
Video PTS(B) = 2625135, 3754
Video PTS(P) = 2640150, DTS(P) = 2628888, 3753
Audio PTS = 2572080, 2880
Video PTS(B) = 2632642, 3754
Audio PTS = 2574960, 2880
Video PTS(B) = 2636396, 3754
Audio PTS = 2577840, 2880
Audio PTS = 2580720, 2880
Video PTS(P) = 2651411, DTS(P) = 2640150, 3754
Audio PTS = 2583600, 2880
Video PTS(B) = 2643903, 3753
Audio PTS = 2586480, 2880
Video PTS(B) = 2647657, 3754
Audio PTS = 2589360, 2880
Audio PTS = 2592240, 2880
Video PTS(P) = 2662672, DTS(P) = 2651411, 3754
Audio PTS = 2595120, 2880
Video PTS(B) = 2655165, 3754
Audio PTS = 2598000, 2880
Video PTS(B) = 2658918, 3753
Audio PTS = 2600880, 2880
Audio PTS = 2603760, 2880
Video PTS(P) = 2673933, DTS(P) = 2662672, 3754
Audio PTS = 2606640, 2880
Video PTS(B) = 2666426, 3754
Audio PTS = 2609520, 2880
Video PTS(B) = 2670180, 3754
Audio PTS = 2612400, 2880
Video PTS(P) = 2685195, DTS(P) = 2673933, 3753
Audio PTS = 2615280, 2880
Audio PTS = 2618160, 2880
Video PTS(B) = 2677687, 3754
Audio PTS = 2621040, 2880
Video PTS(B) = 2681441, 3754
Audio PTS = 2623920, 2880
First Audio PTS = 0x0027fe70, -341
Video PTS(P) = 2696456, DTS(P) = 2685195, 3754
Audio PTS = 2626800, 2880
Audio PTS = 2629680, 2880
Audio PTS = 2632560, 2880
Video PTS(B) = 2688948, 3753
Audio PTS = 2635440, 2880
Video PTS(B) = 2692702, 3754
Video PTS(P) = 2703963, DTS(P) = 2696456, 3754
Audio PTS = 2638320, 2880
Audio PTS = 2641200, 2880
Video PTS(B) = 2700210, 3754
Audio PTS = 2644080, 2880
Video PTS(P) = 2715225, DTS(P) = 2703963, 3753
Audio PTS = 2646960, 2880
Video PTS(B) = 2707717, 3754
Audio PTS = 2649840, 2880
Audio PTS = 2652720, 2880
Video PTS(B) = 2711471, 3754
Audio PTS = 2655600, 2880
Video PTS(P) = 2726486, DTS(P) = 2715225, 3754
Audio PTS = 2658480, 2880
Video PTS(B) = 2718978, 3753
Audio PTS = 2661360, 2880
Video PTS(B) = 2722732, 3754
Audio PTS = 2664240, 2880
Audio PTS = 2667120, 2880
Video PTS(P) = 2737747, DTS(P) = 2726486, 3754
Audio PTS = 2670000, 2880
Video PTS(B) = 2730240, 3754
Audio PTS = 2672880, 2880
Video PTS(B) = 2733993, 3753
Audio PTS = 2675760, 2880
Audio PTS = 2678640, 2880
Video PTS(P) = 2749008, DTS(P) = 2737747, 3754 <-- last video PTS
Audio PTS = 2681520, 2880
Video PTS(B) = 2741501, 3754
Audio PTS = 2684400, 2880
Video PTS(B) = 2745255, 3754
Audio PTS = 2687280, 2880
Audio PTS = 2690160, 2880
Audio PTS = 2693040, 2880
Audio PTS = 2695920, 2880
Audio PTS = 2698800, 2880
Audio PTS = 2701680, 2880
Audio PTS = 2704560, 2880
Audio PTS = 2707440, 2880
Audio PTS = 2710320, 2880
Audio PTS = 2713200, 2880
Audio PTS = 2716080, 2880
Audio PTS = 2718960, 2880
Audio PTS = 2721840, 2880
Audio PTS = 2724720, 2880
Audio PTS = 2727600, 2880
Audio PTS = 2730480, 2880
Audio PTS = 2733360, 2880
Audio PTS = 2736240, 2880
Audio PTS = 2739120, 2880
Audio PTS = 2742000, 2880
Audio PTS = 2744880, 2880
Audio PTS = 2747760, 2880
Audio PTS = 2750640, 2880 <-- last audio PTS

packets for pid 0 <0x0000> = 31, first = 1209, last = 52013
packets for pid 31 <0x001f> = 3, first = 8231, last = 47155
packets for pid 256 <0x0100> = 31, first = 1210, last = 52014
packets for pid 4097 <0x1001> = 32, first = 2486, last = 52045
packets for pid 4113 <0x1011> = 33190, first = 2, last = 49311
packets for pid 4352 <0x1100> = 12096, first = 1, last = 51538
packets for pid 4353 <0x1101> = 1362, first = 243, last = 52079
packets for pid 4354 <0x1102> = 1362, first = 245, last = 52080
packets for pid 4355 <0x1103> = 1362, first = 247, last = 52081
packets for pid 4356 <0x1104> = 372, first = 249, last = 50634
packets for pid 4611 <0x1203> = 2240, first = 3807, last = 48772
packets for pid 8191 <0x1fff> = 2, first = 52082, last = 52083
coded pictures = 52, video fields = 0

C:\xfer>

The last audio PTS is 2750640 and the last video PTS is again 2749008. The video ends at 2749008 + 3754 = 2752762. The audio ends at 2750640 + 2880 = 2753520. 2753520 - 2752762 = 758. 758 / 90000 = 8.42 milliseconds too much audio.

The problem is, this extra audio adds up over the course of the many segments required for seamless branching. For a 30 segment movie, there could be up to 150 milliseconds too much LPCM and 960 milliseconds too much AC3. Of course, the average will be about half that.

When the clip is being played with A/V sync, the extra audio is taken care of when the decoder starts the next clip. If the timestamps are far enough apart, the video decoder will repeat a frame to maintain A/V sync.

BTW, you can download a utility to cut the end off of .m2ts files so that you can get the ending PTS's faster than demuxing the whole segment.

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

tail <infile> <outfile> <length>

For example:

tail 00017.m2ts 00017tail.m2ts 10000000

Ron

[Thunderbolt8]

is this the same version as the test version you gave me, ron, or is this again another update?

so this was the reason why xport and mencoder demuxed 1 frame more for each joined part, to have the audio stay in sync? I assume, even if the audio is too long then compared to its original state we dont need to alter it (or the video), because of that additional 1 frame per part everything stays in sync throughout the movie?

thanks for your effort btw.! :)

[madshi]

when removing dialnorm from ac3 tracks, eac3to sets the value to -0db.. shouldn't the value be -31db?
It's not really "-0db", it's zeroed out, which according to the specification has the same effect as setting it to "-31db". Sony doesn't use "-31db" for their Blu-Ray audio tracks. Sony zeroes out the dialnorm fields. So I decided to let eac3to disable dialnorm similar to how Sony is doing it.

[madshi]

its definately not +200. i applied -200 in the flac and it was fine in some situations when I left it that way in mpc with 0 (=-200) or set +200 (=0, compared to the original state) for other situations.
As far as I understand it, the mathematics say it should be +200. Have you tried with +200? Please try it before saying it must be wrong.

[madshi]

well i checked EVOdemux, and some of the movies are at different frame rates then the actual movie (eg: 59.94) so if you factor that in to the occasion (obviously using the correct source and actual frame rate) you can then do the math.

Out of my 16 HD DVDs, I only have 5 which need a delay (positive or negative) and with doing the math this way I only found 1 where this had an effect (Transformers). Later today I'll try ripping with my previously determined and newly calculated delays and see if it makes a difference or not.
Will be interesting to hear your results. I'd really like to find a way to automatically calculate the perfect delay value.

[Snowknight26]

That's interesting! Are the DTS tracks in sync when playing them with the video? Not sure where the problem comes from. Maybe Surcode does something wrong?

Wait... Have you encoded the DTS tracks with eac3to's Surcode automation? Or have you manually started Surcode? Or did you use a different DTS encoder?

Another question: Where did you get the runtime information from? Did you throw the v2.05 and v1.23 created DTS files into eac3to v2.05 to get the runtime information? Or did you feed the same DTS file into both v2.05 and v1.23 and they report different lengths for the same track?

No they aren't, unfortunately.
Yes, used eac3to's Surcode automation.
Runtime for all files is from v2.05. I was comparing the files made by v2.05 and v1.23, but getting the info strictly from v2.05.

[Thunderbolt8]

As far as I understand it, the mathematics say it should be +200. Have you tried with +200? Please try it before saying it must be wrong.
of course I did, but its way off then

[madshi]

http://samples.mplayerhq.hu/mlp/

http://samples.mplayerhq.hu/DVD-Audio/

http://www.rarewares.org/rja/samples/
Thanks, will have a look.

[madshi]

I've updated xport to remove the sequence end codes from the demuxed video elementary stream. This avoids having to use TSRemux just for this purpose on concatenated Blu-ray .m2ts files.
Thanks!! :)

I guess this is mainly meant for reencoding? Wouldn't it be even better to remove the sequence end codes while still leaving the TS structure in place - so that we can just join the TS parts into one big TS file? Or would the timecodes also need to be rewritten to make simple file joining work?

I've also looked at the A/V sync issues with seamless branching titles (where multiple .m2ts files are joined together). xport is working correctly, but there are issues with the way the segments are authored. Basically, there's too much audio at the end of each segment. Unfortunately, it can't be avoided, since each segment must end with a full PES packet for audio.

This means there can be as much as 5 milliseconds of extra LPCM audio and as much as 32 milliseconds of extra AC3 audio per segment.

[...]

The problem is, this extra audio adds up over the course of the many segments required for seamless branching. For a 30 segment movie, there could be up to 150 milliseconds too much LPCM and 960 milliseconds too much AC3. Of course, the average will be about half that.

When the clip is being played with A/V sync, the extra audio is taken care of when the decoder starts the next clip. If the timestamps are far enough apart, the video decoder will repeat a frame to maintain A/V sync.
That's very interesting!!

I'm just thinking about ProjectX. I believe ProjectX checks the timecodes and adds/removes AC3 frames when necessary to keep audio sync perfect when joining multiple VOB files. xport currently always only handles one TS file at a time, right? Would it be much work to enhance xport so that it can handle a specified sequence of TS files? That should allow xport to double check timecodes and remove or duplicate AC3 frames if necessary to avoid cumulated audio delays. What do you think?

Maybe I'm asking too much... :) I don't know how much time you have for doing this kind of stuff. Please don't feel urged to spend hours on this. But of course many of us would greatly welcome if you were able to solve the seamless branching audio sync problem!

--------

BTW, I'm not sure but I think with some movies there may be more than just 5ms LPCM data too much! E.g. I remember when I tried to remux "Ghost in the Shell 2 - Innocence" I had quite some trouble with the joining of the two TS parts. At the join point video stuttered quite a lot while audio was playing fluidly. Only after letting mkvtoolnix rewriting the video timestamps I got it solved more or less. But at the join point I now have about half a second of video where the frames are shown slowed down. I think there was at least half a second of too much LPCM data at the end of the first TS part!

[Thunderbolt8]

btw. any reason why remuxing ratatouille (AVC) from .m2ts into .mkv results into video glitches, which get the audio of sync?

someone reported its the same with meet the robinsons btw., also AVC, seamless branching and an animated movie. has it something to do how animated movies are produced?


p.s. ron, is this improved function for seamless branching movies only restricted to avc or does it work with vc-1 and mpeg2 as well ? (even though I dont know any such seamless branching movie, does 1 already exist?)

[madshi]

No they aren't, unfortunately.
Yes, used eac3to's Surcode automation.
Runtime for all files is from v2.05. I was comparing the files made by v2.05 and v1.23, but getting the info strictly from v2.05.
Hmmmm... One reason why the runtimes differ could be that some of those tracks have zero padding in them while others don't. Can you please run all DTS tracks through v2.05 ("eac3to source.dts dest.dts") and check if v2.05 claims that it removes zero padding for some DTS files? Maybe the runtime of the v1.23 tracks will change once you run the DTS tracks through v2.05?

[Thunderbolt8]

the galapagos HD DVD is 1080i and also has a different framerate (29.97) btw.

[madshi]

btw. any reason why remuxing ratatouille (AVC) from .m2ts into .mkv results into video glitches, which get the audio of sync?
Which remuxing method are you using? I've always said that demuxing h264 and then muxing it with mp4box/creator/mkvtoolnix is a bad idea. But somehow nobody listens to me. I strongly recommend using Haali's filters to do any h264 remuxing. Too bad that mkvtoolnix can't handle Blu-Ray h264 remuxes done by the latest Haali filters, but hopefully Mosu will fix that soon.

[Thunderbolt8]

I did also try that as well, with gdsmux. I packed the whole .m2ts file into gdsmux. but when only selecting the video stream it just stops (the nice stop error), for both version, the old and new one. so I always have to add at least one audio stream.
when its finished the file looks nice, it has the correct length and is also in sync. but you cant jump back and forth in the stream at all. and you also cant remux the file with mkvmerge again, the final file will result in a neverending processing when double clicking, but no picture or sound will occur. so I cant really use gdsmux.

and these were the only 2 movies with glitches I had so far. no other avc/h264 hddvd, blu-ray or hdtv source resulted in glitches so far that way.

[Snowknight26]

Hmmmm... One reason why the runtimes differ could be that some of those tracks have zero padding in them while others don't. Can you please run all DTS tracks through v2.05 ("eac3to source.dts dest.dts") and check if v2.05 claims that it removes zero padding for some DTS files? Maybe the runtime of the v1.23 tracks will change once you run the DTS tracks through v2.05?


eac3to.exe "G:\Encoding Tools\temp\poseidon.new.dts" "G:\Encoding Tools\temp\poseidon.new.redone.dts"
DTS, 5.1 channels, 1:38:44, 24 bits, 1536kbit/s, 48khz
Creating/writing file "G:\Encoding Tools\temp\poseidon.new.redone.dts"...
Removing DTS zero padding...
Done.

eac3to.exe "G:\Encoding Tools\temp\poseidon.old.dts" "G:\Encoding Tools\temp\poseidon.old.redone.dts"
DTS, 5.1 channels, 1:40:18, 24 bits, 1536kbit/s, 48khz
Creating/writing file "G:\Encoding Tools\temp\poseidon.old.redone.dts"...
Removing DTS zero padding...
Done.

eac3to.exe "G:\Encoding Tools\temp\poseidon.new.redone.dts"
DTS, 5.1 channels, 1:38:36, 24 bits, 1536kbit/s, 48khz

eac3to.exe "G:\Encoding Tools\temp\poseidon.old.redone.dts"
DTS, 5.1 channels, 1:38:35, 24 bits, 1536kbit/s, 48khz

[honai]

I've always said that demuxing h264 and then muxing it with mp4box/creator/mkvtoolnix is a bad idea.

Why is that? Done this on all my Blu-ray discs, never had any problems, video plays smooth and in sync.

[madshi]

I did also try that as well, with gdsmux. I packed the whole .m2ts file into gdsmux. but when only selecting the video stream it just stops (the nice stop error), for both version, the old and new one. so I always have to add at least one audio stream.
when its finished the file looks nice, it has the correct length and is also in sync. but you cant jump back and forth in the stream at all. and you also cant remux the file with mkvmerge again, the final file will result in a neverending processing when double clicking, but no picture or sound will occur. so I cant really use gdsmux.
Yeah, as I said, mkvtoolnix doesn't like the Haali/gdsmux remuxed MKVs. But I hope Mosu will fix that soon. Hopefully running the MKV through a fixed mkvtoolnix with rewriting the timestamps will also give us seeking back.

[madshi]

Why is that? Done this on all my Blu-ray discs, never had any problems, video plays smooth and in sync.
Try it with Equilibrium, then you know why. The same problem occurs with some German h264 broadcasts. Anyway, that's out of topic here...

[madshi]

eac3to.exe "G:\Encoding Tools\temp\poseidon.new.dts" "G:\Encoding Tools\temp\poseidon.new.redone.dts"
DTS, 5.1 channels, 1:38:44, 24 bits, 1536kbit/s, 48khz
Creating/writing file "G:\Encoding Tools\temp\poseidon.new.redone.dts"...
Removing DTS zero padding...
Done.

eac3to.exe "G:\Encoding Tools\temp\poseidon.old.dts" "G:\Encoding Tools\temp\poseidon.old.redone.dts"
DTS, 5.1 channels, 1:40:18, 24 bits, 1536kbit/s, 48khz
Creating/writing file "G:\Encoding Tools\temp\poseidon.old.redone.dts"...
Removing DTS zero padding...
Done.

eac3to.exe "G:\Encoding Tools\temp\poseidon.new.redone.dts"
DTS, 5.1 channels, 1:38:36, 24 bits, 1536kbit/s, 48khz

eac3to.exe "G:\Encoding Tools\temp\poseidon.old.redone.dts"
DTS, 5.1 channels, 1:38:35, 24 bits, 1536kbit/s, 48khz
Interesting! Why is there zero padding in your DTS files? Which Surcode version are you using? eac3to normally disables zero padding when automating Surcode.

Are the files with removed padding in sync with the video now?

[Snowknight26]

Interesting! Why is there zero padding in your DTS files? Which Surcode version are you using? eac3to normally disables zero padding when automating Surcode.

Are the files with removed padding in sync with the video now?

No idea. 1.0.21 according to the help menu.

Edit: poseidon.new.redone.dts synced perfectly. Should I even bother with the other 3 dts files?

[TripleH]

Hi friends,

I've demuxed the DTS-HD MA track of X-Men: The Last Stand with TSRemux, and here is what eac3to reports:

DTS Master Audio, 6.1 channels, 24 bits, 1536kbit/s, 48khz

Am I getting the full bitrate DTS-HD MA track or only the core ?

As far as I know the 1536kbps is the core.

[madshi]

poseidon.new.redone.dts synced perfectly. Should I even bother with the other 3 dts files?
Not necessary. If you have problems again in the future, just let me know.

[madshi]

Hi friends,

I've demuxed the DTS-HD MA track of X-Men: The Last Stand with TSRemux, and here is what eac3to reports:

DTS Master Audio, 6.1 channels, 24 bits, 1536kbit/s, 48khz

Am I getting the full bitrate DTS-HD MA track or only the core ?

As far as I know the 1536kbps is the core.
If you have the Sonic Audio Decoder (v4.3 or newer) installed you'll get the full bitrate DTS-HD MA track decoded.

[madshi]

eac3to v2.06 released

http://madshi.net/eac3to.zip

* doing FLAC -> FLAC now copies metadata from source to destination file
* MLP files are correctly decoded now (by both Nero and libav/ffmpeg)
* runtime for padded DTS files is shown correctly now

[Snowknight26]

What was wrong with decoding MLP files?

[killa_kid]

Will be interesting to hear your results. I'd really like to find a way to automatically calculate the perfect delay value.

well it did seem to have an improvement the problem is that the change was 21ms, so it was barely noticeable.

I have a quick little script I made in JavaScript. You enter the PTM of the EVO, first PTS of video, first PTS of audio, the frame rate that EVOdemux says the video is (in the case of 60 its 30, 59.94 is 29.976,etc) and what the true frame rate is.

So you obviously need to know a bit about the movie before hand, so its not perfect, but I've only found movies with a 23.976 output...granted all of my movies are by American companies and if they were DVD they would be NTSC (in fact I own a couple of them in NTSC DVD :P).

[Thunderbolt8]

What was wrong with decoding MLP files?
did this explain my batman delay issue?

and which versions affected this? after the new 200 i redecoded all my trueHD tracks again, will I have to do that again now?

[TripleH]

If you have the Sonic Audio Decoder (v4.3 or newer) installed you'll get the full bitrate DTS-HD MA track decoded.


Although it reports 1536kbps (I have the Sonic Audio Decoder 4.3 installed) ?

Thanks.

[madshi]

What was wrong with decoding MLP files?
Nothing, I just forgot to add support for that.

did this explain my batman delay issue?
Not at all. We were talking about MLP and not about TrueHD.

[madshi]

Although it reports 1536kbps (I have the Sonic Audio Decoder 4.3 installed) ?
Ooops, that's a bug in eac3to. But don't worry, it's just a cosmetic issue. Basically eac3to shouldn't show the bitrate at all for DTS-HD Master Audio tracks. You'll still get the full DTS-HD Master Audio track decoded just fine. Just ignore that "1536kbit/s" text.

[Snowknight26]

Doesn't TrueHD use MLP?

[Thunderbolt8]

wasnt both the same ? 0o
at least the normal evodemux window tells me truehd, while at the stream window theres always mlp displayed :S

[madshi]

Dolby bought MLP from Meridian, modified MLP a bit to allow more than 5.1 and named their modificated MLP "TrueHD". It's related to MLP but not the same. The headers differ noticably. HD DVD and Blu-Ray use TrueHD. DVD-Audio used MLP. Maybe HD DVD and Blu-Ray could also use MLP, but so far all tracks I've seen were TrueHD and not MLP. eac3to v2.06 fixed/added MLP decoding. This change had nothing to do with TrueHD decoding.

[Snowknight26]

Still leaves us with the sync-up-until-the-half-way-mark issues that Thunder (and I) experience. Could it be an EVOdemux problem?

[killa_kid]

Still leaves us with the sync-up-until-the-half-way-mark issues that Thunder (and I) experience. Could it be an EVOdemux problem?

are you merging (or demuxing) both EVO files into 1? I had the same problems with I didn't make a single file out of them

[Thunderbolt8]

rebuilt the truehd track as 1 seperate .evo and then demuxed the track from there

[killa_kid]

what about the video?

[Thunderbolt8]

rebuilt (vc1)

[madshi]

Those movies where there are still delay problems, does EvoDemux report timestamp differences there? Can we at least confirm that those movies with identical timestamps for video and audio have all delay problems sorted out?

If only those movies are now problematic where first timestamps differ, maybe we should then collect new information? I mean, everybody who meets such a movie, do a manual sync and report here the delay EvoDemux reported and the manual delay you ended up using. Maybe we can find a pattern... But please report only if you're sure that the delay you're using is near perfect. Posting information where the delay might be 100ms off (or even more) won't help.

[madshi]

Bourne Supremecy English E-AC3
EvoDemux said: +1001ms
ended up using: +1250ms

[Thunderbolt8]

yes, timestamps differences. might be the 1st one with timestamps differences for truehd, but am not exactly sure. all the movies with timestamp differences for DD+ were fine with the calculated delay.

batman begins:

Opening file PEVOB_1.EVO
Reading...
File size: 11205 Mbytes.
VOB number 4 contains 2 video , 5 audio and 4 subpicture streams.
PTM of first video frame = 00000D61
PTM of last video frame = 19B28220
Duration = 1:19:50.286
VC-1 video stream 0 found!
First PTS = 00000D61
Dolby TrueHD audio stream 1 found!
First PTS = 000053C3 (+200ms)

[killa_kid]

Transformers Eng E-AC3
EvoDemux said: 0 (but the video had a delay of 83ms)
ended up using: -104ms

All of my other rips used no delay or the same as EVOdemux. The only difference between Transformers is that the frame rate in EVOdemux is different from the final. The rest have the same EVOdemux frame rate and final frame rate.

[madshi]

Please only post the information about the first EVO file. The information about the 2nd EVO file is useless. Also which delay did you end up using and how sure are you that the final delay is near perfect? If you're not sure if -200ms or 0ms is correct then the information doesn't help us much. Finally: What are you doing with the video? Are you demuxing and/or reencoding?

[Thunderbolt8]

hm it definately wont work out with +200 for me :S

I didnt decide for anything yet. was only an information post so far :P

[madshi]

Transformers Eng E-AC3
EvoDemux said: 0 (but the video had a delay of 83ms)
ended up using: -104ms
Ah, interesting! "-83ms" for the audio would have been the mathematical delay if the video file had a delay of "+83ms". So it seems that for you the mathematical way always works correctly.

Strange enough in your original (not edited) post you listed Bourne Supremecy as working with +1001ms delay. But for me that clearly didn't work. I needed to use +1250ms. That's still confusing the heck out of me.

[killa_kid]

Ah, interesting! "-83ms" for the audio would have been the mathematical delay if the video file had a delay of "+83ms". So it seems that for you the mathematical way always works correctly.

Strange enough in your original (not edited) post you listed Bourne Supremecy as working with +1001ms delay. But for me that clearly didn't work. I needed to use +1250ms. That's still confusing the heck out of me.

I got -83ms originally but I tried the idea you put in my head earlier today, and it seems to help. The problem is the difference is only 21ms, so its nothing substantial to really tell how its working.

For The Bourne Supremacy, I'm guessing its a dubbed track? If so it could be off for some other reason (bad mastering, etc).

PS: I realized you just wanted to see how we fixed these things, and my original post had things that i didn't need to fix :P

[madshi]

For The Bourne Supremacy, I'm guessing its a dubbed track? If so it could be off for some other reason (bad mastering, etc).
No, it's the original English track. The German dubbed track had the same timestamps as the video and automatically had perfect sync.

[Snowknight26]

2 Fast 2 Furious, English DD+ track.
Opening file FEATURE_1.EVO
Reading...
File size: 6569 Mbytes.
VOB number 1 contains 1 video , 4 audio and 4 subpicture streams.
PTM of first video frame = 00000D8E
PTM of last video frame = 0CEB6176
Duration = 0:40:08.323
VC-1 video stream 0 found!
First PTS = 00000D8E
Dolby Digital Plus audio stream 0 found!
First PTS = 00016D78 (+1001ms)

+1001 works for the beginning, but buy the end, its off. Would need +1350 at the end.
Since I'm reencoding to x264 as 23.976, could that be whats causing it? If so, why aren't my other encodes off.. like Poseidon?

[killa_kid]

No, it's the original English track. The German dubbed track had the same timestamps as the video and automatically had perfect sync.

hmmmm...the first time I watched my rip of The Bourne Identity, it was not my computer, lying on a couch, now that I am watching it from my monitor up close I am seeing some sync issues. I tried adding a 150ms delay as it was playing, and it seems to be fixing the issues. With my already present 1001ms + 150ms = 1250ms delay. It looks like there may be more to this then I can see. I'm gonna check all of my rips now much much closer

[madshi]

+1001 works for the beginning, but buy the end, its off. Would need +1350 at the end.
Since I'm reencoding to x264 as 23.976, could that be whats causing it? If so, why aren't my other encodes off.. like Poseidon?
If +1001 works for the beginning that's all what is important for eac3to. If the audio is out of sync at the end of the movie that probably has something to do with how you do the reencoding. As was said before, you should join the EVO files before you do the reencoding. Don't know if you already did that.

[madshi]

hmmmm...the first time I watched my rip of The Bourne Identity, it was not my computer, lying on a couch, now that I am watching it from my monitor up close I am seeing some sync issues. I tried adding a 150ms delay as it was playing, and it seems to be fixing the issues. With my already present 1001ms + 150ms = 1250ms delay. It looks like there may be more to this then I can see. I'm gonna check all of my rips now much much closer
1001ms + 150ms = 1250ms? Ehm...... :eek:

[killa_kid]

1001ms + 150ms = 1250ms? Ehm...... :eek:

250ms* happy? ;)

[Snowknight26]

If +1001 works for the beginning that's all what is important for eac3to. If the audio is out of sync at the end of the movie that probably has something to do with how you do the reencoding. As was said before, you should join the EVO files before you do the reencoding. Don't know if you already did that.

Would that affect the video, audio, or both.. and when, demuxing or reencoding?


Edit:
Also:
eac3to.exe "G:\Encoding Tools\temp\2fast.dd+.mpa" 2fast.dts
E-AC3, 5.1 channels, 1:47:35, 1536kbit/s, 48khz, dialnorm: -27dB
Removing dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
Disabling DRC for Nero (E-)AC3 decoding...
DirectShow reports 5.1 channels, 24 bits, 48khz
Writing WAVs...
Creating/writing file "2fast.L.wav"...
Creating/writing file "2fast.R.wav"...
Creating/writing file "2fast.C.wav"...
Creating/writing file "2fast.LFE.wav"...
Creating/writing file "2fast.SL.wav"...
Creating/writing file "2fast.SR.wav"...
Surcode encoding successfully started. Please wait...
Closing Surcode...
Done.

eac3to.exe 2fast.dts
DTS, 5.1 channels, 1:47:35, 24 bits, 1536kbit/s, 48khz

eac3to.exe 2fast.dts 2fast.redone.dts
DTS, 5.1 channels, 1:47:35, 24 bits, 1536kbit/s, 48khz
Creating/writing file "2fast.redone.dts"...
Removing DTS zero padding...
Done.

eac3to.exe 2fast.redone.dts
DTS, 5.1 channels, 1:48:00, 24 bits, 1536kbit/s, 48khz

DTS runtimes still mismatching.

[nautilus7]

I am trying to decode a trueHD track with ffmpeg. I get this message:

C:\Tools>eac3to matrix.3.thd matrix.3.ffmpeg.wavs -libav
TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
Writing WAVs...
Removing dialog normalization...
Creating/writing file "matrix.3.lib.24bit.L.wav"...
Creating/writing file "matrix.3.lib.24bit.R.wav"...
Creating/writing file "matrix.3.lib.24bit.C.wav"...
Creating/writing file "matrix.3.lib.24bit.LFE.wav"...
Creating/writing file "matrix.3.lib.24bit.SL.wav"...
Creating/writing file "matrix.3.lib.24bit.SR.wav"...
-------------------------------------------------------------------------------[
mlp @ 68A442E0]End of stream indicated
[mlp @ 68A442E0]Substream 1 parity check failed
[mlp @ 68A442E0]Substream 1 checksum failed
[mlp @ 68A442E0]Substream 1 length mismatch.
This audio track contains only 16 bit of information.
The zero bytes were successfully removed.
Done.

The output file looks ok, but it is 1 ms sorter than the one decoded with nero decoder. This ms is missing from the end of the track.

There are also some different bytes at the start of the decoded tracks (nero vs ffmpeg).

Apart from these 2 points both decoded tracks are identical.

Here's a sample: http://rapidshare.com/files/72300511/sample.zip.html

[nautilus7]

I have dts runtime displayed wrong here too.

C:\Tools>eac3to matrix.3.dts
DTS, 5.1 channels, 2:09:46, 24 bits, 1536kbit/s, 48khz

C:\Tools>eac3to matrix.3.dts matrix.3.padded.dts
DTS, 5.1 channels, 2:09:46, 24 bits, 1536kbit/s, 48khz
Creating/writing file "matrix.3.padded.dts"...
Done.

C:\Tools>eac3to matrix.3.padded.dts
DTS, 5.1 channels, 2:09:46, 24 bits, 1536kbit/s, 48khz

C:\Tools>eac3to matrix.3.c.wav
WAV, 1.0 channels, 2:09:16, 24 bits, 48khz

The correct runtime is 2:09:16. The dts was already padded, so nothing was changed.

[Thunderbolt8]

1ms shouldnt be that bad ;)

[nautilus7]

I know and actually is less than 1 ms.

It's all about development of the ffmpeg/eac3to. :D

[Thunderbolt8]

there could even be 1 ms difference when muxing flac into mka. so I guess this is just normal

[nautilus7]

I wouldn't care about this at all, but i got an error (?) message from eac3to when decoding. So i thought why not make a sample for madshi?

[Thunderbolt8]

I guess we can cancel that batman thing from the list.
I just redid the flac track with 205 and its fine now with 200ms delay. dunno what happened before, maybe I just saw ghosts :S

[puppydg68]

I'm trying to get eac3to running, I'm running nero 7
and I keep getting the message
Disabling DRC for Nero (E-)AC3 decoding... then lines go across the screen and then it exits. It does not continue.
is there a specific version of nero 7 I need?

I tried a nero lite version and it said getting nero audio decoder 2 instance failed.

[xkodi]

@madshi

several things about the MLP (DVD-Audio) support:

1) there are 3 types of channel order for 5.1 MLP files:

ID 12: Group 1 (Lf, Rf), Group 2 (C, LFE, Ls, Rs)
ID 17: Group 1 (Lf, Rf, C), Group 2 ( LFE, Ls, Rs)
ID 20: Group 1 (Lf, Rf, Ls, Rs), Group 2 (C, LFE)

and they are all valid channel order for 5.1 MLP, decoded to WAV, they all produce the same WAV file with the valid channel order for 5.1 WAV files: Lf, Rf, C, LFE, Ls, Rs.

but eac3to 2.06 reports:

C:\eac3to206>eac3to.exe c:\mlp51_id12.mlp
MLP, 5.1 channels, 16 bits, 44.1khz

C:\eac3to206>eac3to.exe c:\mlp51_id17.mlp
MLP, 5.1 channels, 16 bits, 44.1khz

C:\eac3to206>eac3to.exe c:\mlp51_id20.mlp
MLP, 5.1 wrong order channels, 16 bits, 44.1khz

ID 20 order is wrong for WAV files, but not for MLP, so "5.1 wrong order channels" doesn't seem right to me.

2) with Nero decoder all of the samples: mlp51_id12.mlp, mlp51_id17.mlp, mlp51_id20.mlp are decoded, but the output is different for mlp51_id20.mlp, probably it produces WAV with channel order Lf, Rf, Ls, Rs, C, LFE instead of the correct one Lf, Rf, C, LFE, Ls, Rs. it's important, because most of the DVD-Audio discs use ID20, ID12 and ID17 are very rare.

3) with -libav switch all of the above samples produce:

mlp @ XXXXXXXX]Lossless check failed - expected 0, calculated 7
The libav decoder output an unexpected bitdepth.
The WAV writer didn't receive any audio data.

and no output at all, i don't know why because i have ffmpeg patched with one of the first MLP patches and it decodes all of the samples without complaining about the checksums, so maybe with the newest ffmpeg MLP patches, something is broken and previous versions are better to use at the moment, because there is no problem with the checksums according to Surecode MLP: Verify: stream "C:\mlp51_id20.mlp" passed.

here are the samples: http://xkodi.svobodno.com/mlp51/

[TripleH]

Another problem here.

I converted X-Men: The Last Stand DTS-HD Master Audio track to FLAC, and the playing didn't go well, so I put the FLAC file in mka container and now it is playable (If I play the flac file as it is I get nothing but static rain/snow noise).

Anyway, now FFDShow decodes and it is fine, but FFDShow reports 16bit while it should be 24bit:

Input Description: 48000 Hz, 8 channels flac (libavcodec)
Output Description: 48000 Hz, 6 channels 16-bit integer

I checked the output section and the support for 24bit is checked.

How can I make it do right ?

Edit: I tried to uncheck the 16bit option and only leave the 24bit option checked and now it reports 24-bit integer.

Do you think it is upsampling the audio by itself instead of decoding it as truly 24bit ?

[xkodi]

about The libav decoder output an unexpected bitdepth. problem, that i mentioned in my previous post:

when i do:

./configure --disable-static --enable-shared
make

to build avcodec.dll, the newly build avcodec.dll still doesn't work with the samples above and eac3to produces the same error, but ffmpeg works without complaining.

when i just do:

./configure
make


avcodec.dll is not built and again ffmpeg works with the samples without complaining.

so, ffmpeg works with the samples in both cases: when it's a static build and when uses avcodec.dll, but the same avcodec.dll, that works with ffmpeg and the samples, doesn't work with eac3to and gives The libav decoder output an unexpected bitdepth. error, so hope madshi will find a solution to that problem.

p.s.
the problem is with all 16bit MLP files, when the file is 24bit, then eac3to works.

[madshi]

Would that affect the video, audio, or both.. and when, demuxing or reencoding?
I've never done reencoding, so I can't say anything about that.

DTS runtimes still mismatching.
:scared:

[madshi]

I am trying to decode a trueHD track with ffmpeg. I get this message:

C:\Tools>eac3to matrix.3.thd matrix.3.ffmpeg.wavs -libav
TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
Writing WAVs...
Removing dialog normalization...
Creating/writing file "matrix.3.lib.24bit.L.wav"...
Creating/writing file "matrix.3.lib.24bit.R.wav"...
Creating/writing file "matrix.3.lib.24bit.C.wav"...
Creating/writing file "matrix.3.lib.24bit.LFE.wav"...
Creating/writing file "matrix.3.lib.24bit.SL.wav"...
Creating/writing file "matrix.3.lib.24bit.SR.wav"...
-------------------------------------------------------------------------------[
mlp @ 68A442E0]End of stream indicated
[mlp @ 68A442E0]Substream 1 parity check failed
[mlp @ 68A442E0]Substream 1 checksum failed
[mlp @ 68A442E0]Substream 1 length mismatch.
This audio track contains only 16 bit of information.
The zero bytes were successfully removed.
Done.

The output file looks ok, but it is 1 ms sorter than the one decoded with nero decoder. This ms is missing from the end of the track.

There are also some different bytes at the start of the decoded tracks (nero vs ffmpeg).

Apart from these 2 points both decoded tracks are identical.

Here's a sample: http://rapidshare.com/files/72300511/sample.zip.html
Thanks, I'll forward this to the MLP/TrueHD decoder author.

I can only reproduce the problem with the end of the file, though. Your "start.thd" sample decodes identically with Nero/ffmpeg for me. Can you recheck the start, please?

[madshi]

I guess we can cancel that batman thing from the list.
I just redid the flac track with 205 and its fine now with 200ms delay. dunno what happened before, maybe I just saw ghosts :S
That's good to know, thanks. So it seems that for now the mathematical delay calculation (based on first timestamps) works alright. The only remaining question is whether we need to multiply that value by "29.97 / 23.976" or not.

[nautilus7]

Concerning trueHD sync issues...

I am very confused with The Matrix REV HD DVD. I have the original HD DVD and an encode of it which comes with dts sound. I didn't make this encode, but i know that the dts track was made from lossless trueHD track, with an eac3to version prior to trueHD Dialog Normalization fix. So i thought of making a flac track in order to replace the dts track.

The result is very confusing... The flac track is in sync in the 1st half (i guess the part that came from the first evo file), but de-sync (needs about 150 ms of delay) in the 2nd part. The really strange thing is that when i decoded both the dts and the flac tracks to wav and opened them in audacity, they had a constant 10 ms offset from beginning to end!!! Of course the dts track that was original included in the encode is in sync with the video.

Can you give me a logical explanation, please?

[madshi]

I'm trying to get eac3to running, I'm running nero 7 and I keep getting the message
Disabling DRC for Nero (E-)AC3 decoding... then lines go across the screen and then it exits. It does not continue.
is there a specific version of nero 7 I need?

I tried a nero lite version and it said getting nero audio decoder 2 instance failed.
Can you please give me the full eac3to output with the Nero 7 (non lite) version? Did you properly buy the HD DVD plugin?

[madshi]

1) there are 3 types of channel order for 5.1 MLP files:

ID 12: Group 1 (Lf, Rf), Group 2 (C, LFE, Ls, Rs)
ID 17: Group 1 (Lf, Rf, C), Group 2 ( LFE, Ls, Rs)
ID 20: Group 1 (Lf, Rf, Ls, Rs), Group 2 (C, LFE)

and they are all valid channel order for 5.1 MLP, decoded to WAV, they all produce the same WAV file with the valid channel order for 5.1 WAV files: Lf, Rf, C, LFE, Ls, Rs.
I know that there are multiple channel ordering variations for MLP which are all "correct". I didn't know the details, though.

but eac3to 2.06 reports:

ID 20 order is wrong for WAV files, but not for MLP, so "5.1 wrong order channels" doesn't seem right to me.
:) I know. I just wanted to post something other than "5.1" to indicate that this is not the normal 5.1 channel order, but something different. I didn't expect that this other ordering would really be used in real life, so I just named it "wrong order" just to give it a name.

2) with Nero decoder all of the samples: mlp51_id12.mlp, mlp51_id17.mlp, mlp51_id20.mlp are decoded, but the output is different for mlp51_id20.mlp, probably it produces WAV with channel order Lf, Rf, Ls, Rs, C, LFE instead of the correct one Lf, Rf, C, LFE, Ls, Rs. it's important, because most of the DVD-Audio discs use ID20, ID12 and ID17 are very rare.

3) with -libav switch all of the above samples produce:

and no output at all, i don't know why because i have ffmpeg patched with one of the first MLP patches and it decodes all of the samples without complaining about the checksums, so maybe with the newest ffmpeg MLP patches, something is broken and previous versions are better to use at the moment, because there is no problem with the checksums according to Surecode MLP:

here are the samples: http://xkodi.svobodno.com/mlp51/
Thanks for the samples! Will check this out. May take a few days, though.

[madshi]

Another problem here.

I converted X-Men: The Last Stand DTS-HD Master Audio track to FLAC, and the playing didn't go well
The reason for that is that you most probably don't have a "FLAC source" filter installed which could handle standalone FLAC files. ffdshow can only handle FLAC files which are part of a container. ffdshow cannot handle standalone FLAC files.

so I put the FLAC file in mka container and now it is playable (If I play the flac file as it is I get nothing but static rain/snow noise).

Anyway, now FFDShow decodes and it is fine, but FFDShow reports 16bit while it should be 24bit
That's another problem of ffdshow. Actually most of ffmpeg (which ffdshow is based on) doesn't support 24bit audio at all. I strongly do not recommend using ffdshow for FLAC decoding. I have an alternative suggestion for you which should fix all the problems you had. See here:

http://forum.doom9.org/showthread.php?t=130498

[madshi]

to build avcodec.dll, the newly build avcodec.dll still doesn't work with the samples above and eac3to produces the same error, but ffmpeg works without complaining.
Ah, you little :devil: you've recompiled avcodec.dll and replaced the version which is shipping with eac3to. Did I allow you to do that? :angry:

Just joking, of course.

When compiling ffmpeg with the default options, you won't get 24bit TrueHD tracks decoded properly. Instead you'll get them only as 16bit. Because of that I've compiled ffmpeg with special config flags. Basically after ./configure you have to edit config.h and add the following two switches:

#define USE_HIGHPRECISION 1
#define CONFIG_AUDIO_NONSHORT 1

Afterwards you can use "make" as usual. This config change makes sure that the MLP/TrueHD decoder outputs the data as 32bit instead of as 16bit. This is necessary to get full 24bit from 24bit TrueHD tracks.

BTW, if you do recompile ffmpeg once in a while with always the latest MLP/E-AC3 decoder patches, I'd be thankful if you could send me the dlls, so that I can update them in my eac3to distribution. I hate recompiling C++ projects, so I'll only do it rarely.

[nautilus7]

Thanks, I'll forward this to the MLP/TrueHD decoder author.

I can only reproduce the problem with the end of the file, though. Your "start.thd" sample decodes identically with Nero/ffmpeg for me. Can you recheck the start, please?
Thanks. I 'll check again and make another sample if needed.

[puppydg68]

Can you please give me the full eac3to output with the Nero 7 (non lite) version? Did you properly buy the HD DVD plugin?

C:\HDTV-Tools\eac3to>eac3to.exe "C:\FEATURE_1_MERGED.mpa" "C:\Feature1.ac3" -640

E-AC3, 5.1 channels, 2:23:27, 1536kbit/s, 48khz, dialnorm: -27dB
Removing dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
Disabling DRC for Nero (E-)AC3 decoding...

C:\HDTV-Tools\eac3to>

I did not buy the plug-in. I thought it only needed the nero audio decoder 2, to go from a demuxed file to ac3. Maybe this is the problem?

[madshi]

Thanks. I 'll check again and make another sample if needed.
That would be good. Thanks.

[madshi]

C:\HDTV-Tools\eac3to>eac3to.exe "C:\FEATURE_1_MERGED.mpa" "C:\Feature1.ac3" -640

E-AC3, 5.1 channels, 2:23:27, 1536kbit/s, 48khz, dialnorm: -27dB
Removing dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
Disabling DRC for Nero (E-)AC3 decoding...

C:\HDTV-Tools\eac3to>
Well, I don't see any complaint by eac3to. But there's also no "Done". That's a bit strange...

I did not buy the plug-in. I thought it only needed the nero audio decoder 2, to go from a demuxed file to ac3. Maybe this is the problem?
Yes, that's most probably the problem. You do need to buy the plugin and register it properly.

[nautilus7]

That would be good. Thanks.I can't reproduce the problem with a sample. Only the original (uncut) thd gives differences in the beginning of each wav channel. There are 4 different bytes at the first 48 bytes of each channel.

Ah, it just came to me!
It has to do with other problem (at the end of the file), i think. I believe these different bytes in the beginning have to do with the track runtime, which is different when i decode the original (uncut) .thd (by 1 ms) and exactly the same when i decode the sample.

So, nothing to worry for now, at least. If it's not fixed when you fix the "end" problem, then i 'll bring it back. :D

[madshi]

I can't reproduce the problem with a sample. Only the original (uncut) thd gives differences in the beginning of each wav channel. There are 4 different bytes at the first 48 bytes of each channel.

Ah, it just came to me!
It has to do with other problem (at the end of the file), i think. I believe these different bytes in the beginning have to do with the track runtime, which is different when i decode the original (uncut) .thd (by 1 ms) and exactly the same when i decode the sample.

So, nothing to worry for now, at least. If it's not fixed when you fix the "end" problem, then i 'll bring it back. :D
Yes, these are the "WAV data chunk size" and "WAV total file size" fields in the WAV file header.

[nautilus7]

2 hints

1. Feature request: I would like you to add an option to choose which channels to decode, when the output format is .wavs.

2. Shouldn't that be the opposite order (the bold letters), like with nero decoding?

C:\Tools>eac3to input.thd output.wavs -libav
TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
Writing WAVs...
Removing dialog normalization...
...

C:\Tools>eac3to input.thd output.wavs
TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
Removing dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
DirectShow reports 5.1 channels, 24 bits, 48khz
Writing WAVs...
...

[madshi]

1. Feature request: I would like you to add an option to choose which channels to decode, when the output format is .wavs.
What would that be good for? You can throw away the channels WAVs you don't like. I don't really like the idea to add an option for this because it would just make the interface more complicated without too much real life benefit. The only thing you'd gain is a tiny little bit of performance. Or am I missing an important advantage of such an option?

2. Shouldn't that be the opposite order (the bold letters), like with nero decoding?

C:\Tools>eac3to input.thd output.wavs -libav
TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
Writing WAVs...
Removing dialog normalization...
...

C:\Tools>eac3to input.thd output.wavs
TrueHD, 5.1 channels, 48khz, dialnorm: -27dB
Removing dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
DirectShow reports 5.1 channels, 24 bits, 48khz
Writing WAVs...
...
Yeah, it should probably be the other way round. But dialog normalization removal, WAVs writing, DirectShow decoding etc are all separate modules in eac3to v2.x which work more or less next to each other. Because of that the order of the log messages is sometimes "funny". No need to worry about that, though...

[nautilus7]

What would that be good for? You can throw away the channels WAVs you don't like. I don't really like the idea to add an option for this because it would just make the interface more complicated without too much real life benefit. The only thing you'd gain is a tiny little bit of performance. Or am I missing an important advantage of such an option?
Almost every time i do a conversion, i like to decode the input and output files to wavs, and compare them in a wave editor like audacity. One channel is enough for this job and my hdd will write it faster. That's the only reason. :p


Yeah, it should probably be the other way round. But dialog normalization removal, WAVs writing, DirectShow decoding etc are all separate modules in eac3to v2.x which work more or less next to each other. Because of that the order of the log messages is sometimes "funny". No need to worry about that, though...
Yeah, it was obvious...

[Furiousflea]

Quick question...

Finally got everything working for EAC3 decoding, however all my HD DVD EAC3 tracks that I'm using are recognised as 24bit, is this normal? Or is possibly Nero HD DVD plugin doing something to them that it shouldn't?

Thanks...:)

[nautilus7]

Normal. All e-ac3 tracks are 24bit.

[madshi]

Almost every time i do a conversion, i like to decode the input and output files to wavs, and compare them in a wave editor like audacity. One channel is enough for this job and my hdd will write it faster. That's the only reason. :p
Maybe I can add an undocumented option for that... :) Which channel are you using? Center?

[nautilus7]

Yes, center. That would be great!

[madshi]

Quick question...

Finally got everything working for EAC3 decoding, however all my HD DVD EAC3 tracks that I'm using are recognised as 24bit, is this normal? Or is possibly Nero HD DVD plugin doing something to them that it shouldn't?

Thanks...:)
E-AC3 tracks don't really have a defined bitdepth. They are stored in a way where bitdepth doesn't have much meaning, I think. The decoder usually always outputs 24 bit or even floating point.

[nautilus7]

Madshi, i 've kept you busy with some minor issues and you probably missed post #1700, which is burning me!

[hristoff2]

Is there a proper way to make a 7.1 -> 6.1 downmix (for DTS-ES 6.1 Discrete encoding)?
(got the encoder, so that's not the problem)

If nobody 'complains' I'm gonna make the Cs [Center Sorround] channel by mixing BL & BR channel.

/edit
6.1 PCM: BL = BR = CS :)

[Thunderbolt8]

That's good to know, thanks. So it seems that for now the mathematical delay calculation (based on first timestamps) works alright. The only remaining question is whether we need to multiply that value by "29.97 / 23.976" or not.
woudlnt this only apply in case the movie would typically be 1080i with its 29.97 fps?

[madshi]

Is there a proper way to make a 7.1 -> 6.1 downmix (for DTS-ES 6.1 Discrete encoding)?
(got the encoder, so that's not the problem)

If nobody 'complains' I'm gonna make the Cs [Center Sorround] channel by mixing BL & BR channel.

/edit
6.1 PCM: BL = BR = CS :)
I'm not sure how many PCM tracks are 6.1 and how many are 7.1. 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. Maybe PCM 6.1 storage is not allowed? Anyway, as long as BL and BR are identical, of course you don't need to mix them... :) You can ask eac3to to give you mono wavs. You can then use a file compare tool to check whether BL and BR are identical or not.

[madshi]

woudlnt this only apply in case the movie would typically be 1080i with its 29.97 fps?
I had to do "*29.97/23.976" for Bourne Supremecy which clearly is not 1080i. So I'm not sure yet whether Bourne Supremecy was a strange case or whether we really always have to do "*29.97/23.976" or not.

[madshi]

Concerning trueHD sync issues...

I am very confused with The Matrix REV HD DVD. I have the original HD DVD and an encode of it which comes with dts sound. I didn't make this encode, but i know that the dts track was made from lossless trueHD track, with an eac3to version prior to trueHD Dialog Normalization fix. So i thought of making a flac track in order to replace the dts track.

The result is very confusing... The flac track is in sync in the 1st half (i guess the part that came from the first evo file), but de-sync (needs about 150 ms of delay) in the 2nd part. The really strange thing is that when i decoded both the dts and the flac tracks to wav and opened them in audacity, they had a constant 10 ms offset from beginning to end!!! Of course the dts track that was original included in the encode is in sync with the video.

Can you give me a logical explanation, please?
Sorry, missed this post.

Not sure what happened there. Let me guess: Probably the reencoder reencoded both EVO files separately. And then he probably joined the resulting fully reencoded parts into one big piece. I could imagine that with such a joined movie there are some "funny" things going on at the join point. E.g. it could be possible that the timecodes let the DTS decoder skip or repeat some frames at the join point. Of course I'm only guessing...

[nautilus7]

Not exactly what i waited to hear. :p
I 'll do more tests.

[TripleH]

The reason for that is that you most probably don't have a "FLAC source" filter installed which could handle standalone FLAC files. ffdshow can only handle FLAC files which are part of a container. ffdshow cannot handle standalone FLAC files.

Actually, I did have it installed while getting the noise.


That's another problem of ffdshow. Actually most of ffmpeg (which ffdshow is based on) doesn't support 24bit audio at all. I strongly do not recommend using ffdshow for FLAC decoding. I have an alternative suggestion for you which should fix all the problems you had. See here:

http://forum.doom9.org/showthread.php?t=130498

Anyway, I tried your flac decoder and works like a charm.

[Furiousflea]

Normal. All e-ac3 tracks are 24bit.

Thanks for the quick answer :)

However now got another problem...bearing in mind that the program has worked fine on a good 10 or so HD DVDs.

I'm now having a problem with a DTS Master Audio track on "Rambo First Blood Part II"

I'm using this....

eac3to rambo.dtshd rambo.wavs

Its reporting everything correctly, except when the decoding starts it goes very slow and gets gradually slower until it stops about an eigth way through the the track.

Its just on this track, I can decode other MA stuff fine?

Thanks, Rob

[nautilus7]

I want to remove Dialog Norm. from an ac3 track.
I must do input.ac3 output.ac3. Right?

C:\Tools>eac3to input.ac3 output.ac3
AC3, 5.1 channels, 1:48:51, 640kbit/s, 48khz
Creating/writing file "output.ac3"...
Done.

Shouldn't i get a message whether the track had Dialog Norm on/off?

[madshi]

Actually, I did have it installed while getting the noise.
Then most probably (for whatever reason) your media player has not used madFlac but another (bad) source filter instead.

[madshi]

However now got another problem...bearing in mind that the program has worked fine on a good 10 or so HD DVDs.

I'm now having a problem with a DTS Master Audio track on "Rambo First Blood Part II"

I'm using this....

eac3to rambo.dtshd rambo.wavs

Its reporting everything correctly, except when the decoding starts it goes very slow and gets gradually slower until it stops about an eigth way through the the track.

Its just on this track, I can decode other MA stuff fine?
Not sure where the problem comes from. Please try using "eac3to rambo.dtshd rambo.ac3", just to make sure that it's not eac3to's mono wav output which is stumbling. However, I rather guess that the Sonic Audio Decoder doesn't like this specific track for whatever reason. You could try only decoding the core, but obviously that'd be bad for audio quality... :(

[madshi]

I want to remove Dialog Norm. from an ac3 track.
I must do input.ac3 output.ac3. Right?

C:\Tools>eac3to input.ac3 output.ac3
AC3, 5.1 channels, 1:48:51, 640kbit/s, 48khz
Creating/writing file "output.ac3"...
Done.

Shouldn't i get a message whether the track had Dialog Norm on/off?
eac3to sais: "AC3, 5.1 channels, 1:48:51, 640kbit/s, 48khz". There's no dialnorm listed there, so this AC3 track is already dialnorm free. Let me guess: Is it from a Sony Blu-Ray disc? Sony doesn't use dialnorm.

[Furiousflea]

Not sure where the problem comes from. Please try using "eac3to rambo.dtshd rambo.ac3", just to make sure that it's not eac3to's mono wav output which is stumbling. However, I rather guess that the Sonic Audio Decoder doesn't like this specific track for whatever reason. You could try only decoding the core, but obviously that'd be bad for audio quality... :(


C:\EAC3To>eac3to audio2.dtshd rambo.wavs
DTS Master Audio, 5.1 channels, 16 bits, 1536kbit/s, 48khz
Decoding with DirectShow (Sonic Audio Decoder)...
DirectShow reports 5.1 channels, 16 bits, 48khz
Writing WAVs...
Creating/writing file "rambo.L.wav"...
Creating/writing file "rambo.R.wav"...
Creating/writing file "rambo.C.wav"...
Creating/writing file "rambo.LFE.wav"...
Creating/writing file "rambo.SL.wav"...
Creating/writing file "rambo.SR.wav"...
------------- <- stops here

Thanks for the reply, here's a copy of what gets reported mate :)

I extracted the core with no probs and that decoded to 6 mono wavs no probs....hmmmm

[nautilus7]

eac3to sais: "AC3, 5.1 channels, 1:48:51, 640kbit/s, 48khz". There's no dialnorm listed there, so this AC3 track is already dialnorm free. Let me guess: Is it from a Sony Blu-Ray disc? Sony doesn't use dialnorm.
But eac3to wrote a new file, which has some changed bytes (periodically). So...
It's not from a blu-ray. It's an hd dvd e-ac3 --> ac3 track, done with eac3to (a very very old version, prior to DRC fix).

[tebasuna51]

Ah, you little :devil: you've recompiled avcodec.dll and replaced the version which is shipping with eac3to. Did I allow you to do that? :angry:

Just joking, of course.

But the problem decoding the mlp samples from xkodi is using your avcodec.dll. I make test using samples download from this thread:
eac3to 440hz.mlp 440hz.wav -libav

The libav decoder output an unexpected bitdepth.
---[mlp @ 68A442E0]Lossless check failed - expected 0, calculated d2
...
The WAV writer didn't receive any audio data.
====================================================================
eac3to mlp51_id12.mlp mlp51_id12.wav -libav

The libav decoder output an unexpected bitdepth.
---[mlp @ 68A442E0]Lossless check failed - expected 0, calculated 7
...
The WAV writer didn't receive any audio data.
====================================================================
eac3to Acapela1.mlp Acapela1.wav -libav

The libav decoder output an unexpected bitdepth.
...
The WAV writer didn't receive any audio data.
====================================================================
eac3to "God Save The Queen.mlp" godgood.wav -libav -0,1,4,5,2,3

OK

Only last sample work (with the needed remap for info).

Where:
440hz.mlp stereo, 16 bits, 44100 Hz
Acapela1.mlp stereo, 16 bits, 44100 Hz
mlp51_id12.mlp 6chan , 16 bits, 44100 Hz
God Save The Queen.mlp 6chan , 24 bits, 96000 Hz

The four samples (and the other samples from xkodi) are decoded well by ffmpeg_eac3_mlp_r11045 from Kurtnoise with the know issues:
- Bad channelmapping with multichanel ID 20
- 0.6 ms added at end of files in samples from xkodi, if sources included and mlp encode are right.

BTW, the message for ID 20:
"5.1 wrong order channels"
I think must be replaced by a correct remapping, you are supplying now a incorrect wav and the user don't know what is wrong.
A decoder must accept and know the input formats and supply the output in correct format.

To see if the remap is make when output ac3 I try:
eac3to "God Save The Queen.mlp" god.ac3 -libav -resampleTo48000
MLP, 5.1 wrong order channels, 24 bits, 96khz
Resampling to 48khz...
Encoding AC3...
invalid sample rate
without success.

EDIT: If can help you to solve this last problem:
after the -resampleTo48000 I have a $temp subfolder with
$a23440.dll 219.136
$a232d4.dll 219.136
$a23168.dll 219.136
$a22efc.dll 219.136
All the same and with Properties: r8b.dll 4.0.3.17

[madshi]

But eac3to wrote a new file, which has some changed bytes (periodically). So...
It's not from a blu-ray. It's an hd dvd e-ac3 --> ac3 track, done with eac3to (a very very old version, prior to DRC fix).
AC3 files encoded by eac3to/Aften don't have dialnorm activated. Not sure exactly what bytes eac3to is changing. I guess maybe it rewrites dialnorm from -31dB to 0dB (which is basically the same thing).

[madshi]

But the problem decoding the mlp samples from xkodi is using your avcodec.dll.
Oh, then I misunderstood him.

I make test using samples download from this thread:
eac3to 440hz.mlp 440hz.wav -libav

The libav decoder output an unexpected bitdepth.
---[mlp @ 68A442E0]Lossless check failed - expected 0, calculated d2
...
The WAV writer didn't receive any audio data.
====================================================================
eac3to mlp51_id12.mlp mlp51_id12.wav -libav

The libav decoder output an unexpected bitdepth.
---[mlp @ 68A442E0]Lossless check failed - expected 0, calculated 7
...
The WAV writer didn't receive any audio data.
====================================================================
eac3to Acapela1.mlp Acapela1.wav -libav

The libav decoder output an unexpected bitdepth.
...
The WAV writer didn't receive any audio data.
====================================================================
eac3to "God Save The Queen.mlp" godgood.wav -libav -0,1,4,5,2,3

OK
Ok, thanks for the report. Looks like a "simple" bug in eac3to.

BTW, the message for ID 20:
"5.1 wrong order channels"
I think must be replaced by a correct remapping, you are supplying now a incorrect wav and the user don't know what is wrong.
A decoder must accept and know the input formats and supply the output in correct format.
Yes, I agree. I'll fix that in one of the next builds.

To see if the remap is make when output ac3 I try:
eac3to "God Save The Queen.mlp" god.ac3 -libav -resampleTo48000
MLP, 5.1 wrong order channels, 24 bits, 96khz
Resampling to 48khz...
Encoding AC3...
invalid sample rate
without success.
Looks like another "simple" bug... :)

EDIT: If can help you to solve this last problem:
after the -resampleTo48000 I have a $temp subfolder with
$a23440.dll 219.136
$a232d4.dll 219.136
$a23168.dll 219.136
$a22efc.dll 219.136
All the same and with Properties: r8b.dll 4.0.3.17
That's normal. The r8brain dll I'm using for resampling is not thread safe. So I'm loading multiple copies of that dll. That's the only way I can use r8brain from multiple threads at the same time without getting crashes. The file name of each dll copy is the id of the thread it's being called from... ;)

However, the $temp subfolder should be automatically deleted. Will check why that didn't work. Maybe it's only deleted after a successfull action? Probably...

[Beastie Boy]

Regarding movies where the audio drifts out of synch throughout the movie:
I originally had this problem with Batman Begin HD-DVD (vc1 video). The problem was caused by the Sonic video filter in the Graphedit chain. When I replaced this with with WMV Decoder DMO (or similar, I'm not at home at the moment), everything was OK.

There are a few posts on the forum regarding synch issues caused by the Sonic video decoder. Maybe this is the cause of some of these problems.

Cheers, Beastie.

[nautilus7]

AC3 files encoded by eac3to/Aften don't have dialnorm activated. Not sure exactly what bytes eac3to is changing. I guess maybe it rewrites dialnorm from -31dB to 0dB (which is basically the same thing).
It's possible because they both look the same in audacity.

[nautilus7]

I am doing an eac3 --> ac3 conversion using version 2.06 and the aften version that comes with eac3to (input .eac3 output.ac3).

Why the output ac3 file is 10-11 ms delayed compared to the input eac3?

Here's a sample: http://rapidshare.com/files/72646698/sample.eac3.html

[Icemaan]

Hallo to all

With the new Version of eac3to i get following error when I want encode one DD+ track into Dts or Ac3

The Format of the Source File Could not be detected.This Error I get when I want make
1. Ac3
2. Dts

With Old Version and aften/sox I can convert to ac3
But I want to encode to Dts

Whats going wrong .Please Help
Thanks
icemaan

[shambles]

i've got a truehd track that's 2.13gb and using nero filters, the flac i encoded from it turned out to be 3.26gb. using -libav, the flac is 1.88gb

i guess it's a 20bit track.. here's a 5mb sample http://rapidshare.com/files/72692697/001.thd.html

[tebasuna51]

I am doing an eac3 --> ac3 conversion using version 2.06 and the aften version that comes with eac3to.

Why the output ac3 file is 10-11 ms delayed compared to the input eac3?

How do you determine this delay?

I decoded your sample to wav and encoded to ac3.

The ac3 decoded to a new wav and is 5.333 ms delayed over the previous wav like was predictable.
Always Aften (and other commercial ac3 encoders) delay the output 5.333 ms, unless you use the parameter -pad 0.

[nautilus7]

How do you determine this delay?

I decoded your sample to wav and encoded to ac3.

The ac3 decoded to a new wav and is 5.333 ms delayed over the previous wav like was predictable.
Always Aften (and other commercial ac3 encoders) delay the output 5.333 ms, unless you use the parameter -pad 0.eac3 --> ac3 --> wav
eac3 --> wav

Then i compared the 2 wavs.

[tebasuna51]

eac3 --> ac3 --> wav
eac3 --> wav

Then i compared the 2 wavs.

Same procedure using:
eac3to sample.eac3 sample.ac3 -libav
eac3to sample.eac3 sample1.wav -libav
azid -d3/2 -L0 -l1 -ol,r,c,lfe,sl,sr sample.ac3 sample2.wav

Sample2.wav delayed 5.333 ms

If:
eac3to sample.eac3 sample1.wav -libav
aften -pad 0 sample1.wav sample.ac3
azid -d3/2 -L0 -l1 -ol,r,c,lfe,sl,sr sample.ac3 sample2.wav

Delay 0 ms.

Maybe Nero decoder (I haven't) introduce a new delay?

[nautilus7]

Possible. I don't know what to say.

What is the frame length that delaycut reports for every stream? I get 16 ms for eac3 and 32 ms for ac3. I got 5.333 ms for another eac3 track with a 1536 kbps bitrate.

[mutha88]

Please excuse my stupidity, but how can i encode to DTS? I have buyed and installed SurCode, but when I launch "eac3to" i can't see the DTS rate, filters and so one. I can't select them... where can i read a FAQ/guide for this wonderfull tool?

:stupid::scared:

[nautilus7]

What you need is in 1st post.

You can use: eac3to input.track output.dts -your bitrate

[mutha88]

What you need is in 1st post.

You can use: eac3to input.track output.dts -your bitrate


Well... this is a little light in the dark... but i will try...

By the way... i am using the GUI :rolleyes:

[shambles]

madshi, earlier in the thread you posted that you converted pirates of the caribbean 24bit pcm to flac and that the filesize was bigger than 2gb.. was that dead man's chest or curse of the black pearl?

the reason i'm asking is because i just converted the curse of the black pearl pcm track and the bitrate is 1395 kbps, filesize 1.39gb despite both the input pcm and the output flac track being 24bit. i sort of hope something has gone wrong at my end because having a 16 bit track in a 24 bit file would be amazingly stupid when it's pcm...

sample http://www.sendspace.com/file/4730s7

[mutha88]

Well... this is a little light in the dark... but i will try...

By the way... i am using the GUI :rolleyes:

Nothing happens. How can i make the GUI to find the surcodedvd.exe ?!:eek:

[nautilus7]

Don't know. I use the cli. The command i wrote to you, doesn't work?

[mutha88]

Don't know. I use the cli. The command i wrote to you, doesn't work?

Well... how can i use eac3to.exe? When i opened it a MS-DOS window flashes and that's all... how to open it? If i can open it, then i will read and understand the 1st post, but i dunno how to start the exe... im so lame :stupid: :rolleyes:

[nautilus7]

Navigate to the folder that eac3to is located in the "ms-dos" window. It is called cmd (command prompt) actually. Then type eac3to bla bla bla.

[Furiousflea]

...Just tried another DTS Master Audio Lossless track and am getting same results as before (Progress bar stops part way through, but always at same point for each track)

:(

Any ideas anyone what I can do, surely 2 tracks in a row with this happening, there must be something wrong....Just don't have a clue what to try...

:(

[tebasuna51]

Possible. I don't know what to say.

What is the frame length that delaycut reports for every stream? I get 16 ms for eac3 and 32 ms for ac3. I got 5.333 ms for another eac3 track with a 1536 kbps bitrate.

The frame length is function of bitrate and samplerate.

Really the most elemental atom is the Block (256 samples or 5.333 ms if 48 KHz (256/48000)).

With ac3 a frame is always 6 blocks (1536 samples or 32 ms if 48 KHz).

The new eac3 can have 1, 2, 3 or 6 blocks per frame then each eac3 48 KHz frame can be of 5.333, 10.667, 16 or 32 ms.

If you have 5.333 ms each frame at 1536 kbps the FrameLength must be: 1536000*0.005333/8 = 1024 bytes.

[nautilus7]

Yeah, that's right, but don't understand how the 5,3333 ms delay is introduced in my case (640 kbps eac3, 640 kbps ac3).

[Thunderbolt8]

does the eac3to PAL / NTSC slowdown function slow down to 23.976fps or to EXACT 24000/1001 = 23.9760239...fps?

[nautilus7]

Look at the 1st post. It says 24000/1001.

[moshmothma]

eac3to v2.06 released

http://madshi.net/eac3to.zip

* doing FLAC -> FLAC now copies metadata from source to destination file
* MLP files are correctly decoded now (by both Nero and libav/ffmpeg)
* runtime for padded DTS files is shown correctly now

thanks. THe new version allows me to convert my dvda mlp files to wav. The channels are mapped perfectly with some manual tweaking (-0,1,4,5,2,3). Maybe a -dvda switch would be good. Thanks again

[Furiousflea]

.....:( Just tried running EAC3To on a different computer in the house after installing the sonic audio decoder, now its just saying...

C:\EAC3To>eac3to recall1.dtshd 1.wavs
DTS Master Audio, 5.1 channels, 16 bits, 1536kbit/s, 48khz
Decoding with DirectShow (Sonic Audio Decoder)...
The DirectShow audio decoder didn't accept the input stream.

...I despair, would really appreciate some help, before it ran it did say that "msvcr71.dll" was missing? I assume the Sonic decoder needs this, I downloaded that dll and placed it in my system32 folder but then it just ran until the point with the quotes

[mutha88]

Hello it is me again. I have encoded a *.wav DTS file into *.dts file.

The input *.wav is 988 MB. The output *.dts is 972 MB. Is this a well encoded DTS ?! :stupid:

[ACrowley]

Is there a proper way to make a 7.1 -> 6.1 downmix (for DTS-ES 6.1 Discrete encoding)?
(got the encoder, so that's not the problem)

If nobody 'complains' I'm gonna make the Cs [Center Sorround] channel by mixing BL & BR channel.

/edit
6.1 PCM: BL = BR = CS :)

I think a 7.1 dts capable decoder should autom downmix 7.1 into 6.1 by blendig the Channels ?!
Thats the way 6.1 to 5.1 works because those dts Streams are downwards compatible. The BC is blended into BL / BR for 5.1 Output

But Sonic Decoder isnt 7.1 capable and no other Decoder...Sonic maximum is 6.1

So you wont get 8 Mono Channles from Sonic/eac3to
Output from 7.1 dtshd is 5.1

3) DTS decoding fully supports 6.1, but 7.1 tracks are decoded as 5.1 only.

@mutha88
Hä ? No need to reencode a dts wave to dts :) Simply "write" the dts wave to dts lossless (besliced etc) Reencoding in this case is uselsss.

[mutha88]

I think a 7.1 dts capable decoder should autom downmix 7.1 into 6.1 by blendig the Channels ?!

Thats the way 6.1 to 5.1 works because those dts Streams are downwards compatible. The BC is blended into BL / BR for 5.1 Output

But Sonic Decoder isnt 7.1 capable and no other Decoder...Sonic maximum is 6.1

So you wont get 8 Mono Channles from Sonic/eac3to
Output from 7.1 dtshd is 5.1



@mutha88
Hä ? No need to reencode a dts wave to dts :) Simply "write" the dts wave to dts lossless (besliced etc) Reencoding in this case is uselsss.

I have a question. I've extracted the *.wav file from a movie... but this extracted *.wav file doesn't have the Center audio.... just left and right... i can't heard the voices in the *.wav file... how can i fix it? With which program should i exctract the *.wav file from the movie ?! :stupid:

[madshi]

The libav decoder output an unexpected bitdepth.
Fixed in eac3to v2.07.

BTW, the message for ID 20:
"5.1 wrong order channels"
I think must be replaced by a correct remapping
Implemented in eac3to v2.07.

To see if the remap is make when output ac3 I try:
eac3to "God Save The Queen.mlp" god.ac3 -libav -resampleTo48000
MLP, 5.1 wrong order channels, 24 bits, 96khz
Resampling to 48khz...
Encoding AC3...
invalid sample rate
without success.
This worked for me with v2.07. So I guess one of the other fixes also fixed this. Could you please retry, just to be sure?

[madshi]

With the new Version of eac3to i get following error when I want encode one DD+ track into Dts or Ac3

The Format of the Source File Could not be detected.
That means that your E-AC3 file is probably not clean. Probably there are some garbage bytes in front of the real E-AC3 data. Please try fixing the stream with delaycut.

[madshi]

...Just tried another DTS Master Audio Lossless track and am getting same results as before (Progress bar stops part way through, but always at same point for each track)

:(

Any ideas anyone what I can do, surely 2 tracks in a row with this happening, there must be something wrong....Just don't have a clue what to try...
Please try what I already suggested in my previous reply to you, namely encoding to ac3 to make sure that the problem is the decoding and not the WAVs file writing.

.....:( Just tried running EAC3To on a different computer in the house after installing the sonic audio decoder, now its just saying...
The Sonic decoder doesn't seem to be installed correctly on that different computer.

[madshi]

thanks. THe new version allows me to convert my dvda mlp files to wav. The channels are mapped perfectly with some manual tweaking (-0,1,4,5,2,3). Maybe a -dvda switch would be good. Thanks again
Channel mapping should be automatically corrected by v2.07. If you find that it's not corrected correctly, please let me know.

[madshi]

Please excuse my stupidity, but how can i encode to DTS? I have buyed and installed SurCode, but when I launch "eac3to" i can't see the DTS rate, filters and so one. I can't select them... where can i read a FAQ/guide for this wonderfull tool?
Try the latest eac3to version v2.07. The older version didn't recognize the latest Surcode version while v2.07 should do.

Hello it is me again. I have encoded a *.wav DTS file into *.dts file.
Hmmmmm... I don't think eac3to supports "wav dts" files... :confused:

I have a question. I've extracted the *.wav file from a movie... but this extracted *.wav file doesn't have the Center audio.... just left and right... i can't heard the voices in the *.wav file... how can i fix it? With which program should i exctract the *.wav file from the movie ?!
Depends on the movie format. Is it DVD? Or HD DVD? Or Blu-Ray? Or something else?

[madshi]

i've got a truehd track that's 2.13gb and using nero filters, the flac i encoded from it turned out to be 3.26gb. using -libav, the flac is 1.88gb

i guess it's a 20bit track.. here's a 5mb sample http://rapidshare.com/files/72692697/001.thd.html
Ouch - that's a good sample! Compression ratio of the TrueHD file seems is about 2.87:1. The Nero FLAC has 1.83:1 while the libAV FLAC ends up with 3.17:1. This is the beginning of the movie so compression ratio is probably slightly better here than later in the movie. Overall I agree with you that this is likely to be neither 16bit nor 24bit. I'm not sure if it's 20bit. Maybe. Or maybe 18bit. Anyway, from the FLAC compression ratios it seems to me that libAV decodes it correctly while Nero seems to apply some processing. Maybe dialnorm removal doesn't work for this track for whatever strange reason? I'll have to invest some more time into checking this sample out. But it will probably take some days cause I'm short on time right now...

P.S: Probably the libAV FLAC is alright. So I'd suggest to use that for now. But better keep the TrueHD track until I've found out why Nero fails to work correctly here.

[madshi]

eac3to v2.07 released

http://madshi.net/eac3to.zip

* fixed libAV MLP decoding support
* added automatic MLP ID20 channel remapping
* Surcode 1.0.29 (or newer) home directory detection added

[mutha88]

Try the latest eac3to version v2.07. The older version didn't recognize the latest Surcode version while v2.07 should do.


Hmmmmm... I don't think eac3to supports "wav dts" files... :confused:


Depends on the movie format. Is it DVD? Or HD DVD? Or Blu-Ray? Or something else?

It is a HD-DVD DTS movie... KMPlayer shows me that the encoded *.dts WAV file is DTS-HD. :stupid:

[nautilus7]

Can you check the ac3 delay issues i have please? I have uploaded a sample.

Thanks for the update!

[madshi]

Can you check the ac3 delay issues i have please? I have uploaded a sample.
As tebasuna51 has already explained, every Aften encoding adds a 5.333ms delay - unless you switch Aften to a special (non recommended) mode to avoid this delay. This special mode is currently not supported by eac3to. I think the Dolby encoders behave similarly to Aften, at least I've been told so.

Are you really worried about 5.333ms of delay? Some people don't notice a delay of 100ms. Personally, I believe that I notice incorrect delays as low as 40ms, but I'm not totally sure. I'm very sure that I wouldn't notice a delay of 5.333ms, even if my life depended on it.

[tebasuna51]

Fixed in eac3to v2.07.
...
Implemented in eac3to v2.07.
...
This worked for me with v2.07. So I guess one of the other fixes also fixed this. Could you please retry, just to be sure?

Yes, all work fine now with v2.07.

Thanks.

[nautilus7]

As tebasuna51 has already explained, every Aften encoding adds a 5.333ms delay - unless you switch Aften to a special (non recommended) mode to avoid this delay. This special mode is currently not supported by eac3to. I think the Dolby encoders behave similarly to Aften, at least I've been told so.

Are you really worried about 5.333ms of delay? Some people don't notice a delay of 100ms. Personally, I believe that I notice incorrect delays as low as 40ms, but I'm not totally sure. I'm very sure that I wouldn't notice a delay of 5.333ms, even if my life depended on it.
OK, but this didn't happen to another eac3 to ac3 encode i did a few days ago.

[madshi]

Yes, all work fine now with v2.07.
Thanks for confirming.

[madshi]

OK, but this didn't happen to another eac3 to ac3 encode i did a few days ago.
It must have happened to that other eac3 to ac3 encode, too... :D

[nautilus7]

Are you trying to drive me nuts? :eek:
It didn't happen. And it's not 5,33 ms delay. It's 10,5 ms.

Oh, man... I 'll have to check the tracks again. :mad:

[Thunderbolt8]

does TrueHD conversion of eac3to currently also support files with 96kHz ?

[Snowknight26]

eac3to "G:\Encoding Tools\temp\FEATURE_1_MERGED.DD+.stream.00.mpa" "..\temp\audio\2fast.done.dts"
E-AC3, 5.1 channels, 1:47:35, 1536kbit/s, 48khz, dialnorm: -27dB
Removing dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
Disabling DRC for Nero (E-)AC3 decoding...
DirectShow reports 5.1 channels, 24 bits, 48khz
Writing WAVs...
Creating/writing file "..\temp\audio\2fast.done.L.wav"...
Creating/writing file "..\temp\audio\2fast.done.R.wav"...
Creating/writing file "..\temp\audio\2fast.done.C.wav"...
Creating/writing file "..\temp\audio\2fast.done.LFE.wav"...
Creating/writing file "..\temp\audio\2fast.done.SL.wav"...
Creating/writing file "..\temp\audio\2fast.done.SR.wav"...
Surcode sais/asks: "Invalid Wave File ..\temp\audio\2fast.done.L.wav.".
Surcode sais/asks: "Invalid Wave File ..\temp\audio\2fast.done.R.wav.".
Surcode sais/asks: "Invalid Wave File ..\temp\audio\2fast.done.SL.wav.".
Surcode sais/asks: "Invalid Wave File ..\temp\audio\2fast.done.SR.wav.".
Surcode sais/asks: "Invalid Wave File ..\temp\audio\2fast.done.C.wav.".
Surcode sais/asks: "Invalid Wave File ..\temp\audio\2fast.done.LFE.wav.".
Surcode sais/asks: "At least one valid source file must be specified to encode.".
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...

Main problem is the file names not being accepted, 2nd is that "sais" should be "says".
Would switching to v1.0.29.0 fix the path problems?

[Chumbo]

eac3to "G:\Encoding Tools\temp\FEATURE_1_MERGED.DD+.stream.00.mpa" "..\temp\audio\2fast.done.dts"
E-AC3, 5.1 channels, 1:47:35, 1536kbit/s, 48khz, dialnorm: -27dB
Removing dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
Disabling DRC for Nero (E-)AC3 decoding...
DirectShow reports 5.1 channels, 24 bits, 48khz
Writing WAVs...
Creating/writing file "..\temp\audio\2fast.done.L.wav"...
Creating/writing file "..\temp\audio\2fast.done.R.wav"...
Creating/writing file "..\temp\audio\2fast.done.C.wav"...
Creating/writing file "..\temp\audio\2fast.done.LFE.wav"...
Creating/writing file "..\temp\audio\2fast.done.SL.wav"...
Creating/writing file "..\temp\audio\2fast.done.SR.wav"...
Surcode sais/asks: "Invalid Wave File ..\temp\audio\2fast.done.L.wav.".
Surcode sais/asks: "Invalid Wave File ..\temp\audio\2fast.done.R.wav.".
Surcode sais/asks: "Invalid Wave File ..\temp\audio\2fast.done.SL.wav.".
Surcode sais/asks: "Invalid Wave File ..\temp\audio\2fast.done.SR.wav.".
Surcode sais/asks: "Invalid Wave File ..\temp\audio\2fast.done.C.wav.".
Surcode sais/asks: "Invalid Wave File ..\temp\audio\2fast.done.LFE.wav.".
Surcode sais/asks: "At least one valid source file must be specified to encode.".
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...

Main problem is the file names not being accepted, 2nd is that "sais" should be "says".
Would switching to v1.0.29.0 fix the path problems?
You shouldn't use relative paths like that since you're using an external program in this case. The surcode settings are registry-based and "..\" means nothing to SurCode and if it did, it would be relative to surcode's path and not what you intend. Stick to a fully-qualified path and you'll be fine.

If you're too lazy to type in the full path, then map a drive to your long-and-often-used location. ;)

[Snowknight26]

eac3to is in the same folder as SurCode is, but oh well, I had a problem anyway that would have prevented it from getting the file names to begin with. Still the issue of sais. :p

[MichalHabart]

Hello Madshi,
i found strange thing. When i try to convert TrueHD to DTS, it correctz detects bit depth of TrueHD but it still creates all wav files twice, one for 24 bit and the second for 16. Can i somehow prevent this behaviour? Because it requires twice space on HDD and decode is then twice slower.

[Falcon4]

.....:( Just tried running EAC3To on a different computer in the house after installing the sonic audio decoder, now its just saying...



...I despair, would really appreciate some help, before it ran it did say that "msvcr71.dll" was missing? I assume the Sonic decoder needs this, I downloaded that dll and placed it in my system32 folder but then it just ran until the point with the quotes

Same problem I was having after moving to another computer to do my encoding dirty work. I knew it was because I didn't have the Sonic decoder installed, so I tried copying the DirectShow filters to the new computer and registering them. That got the filter to be identified, but it still wouldn't decode. I thought it was part of bundled software with the Vaio I ripped the Blu-Rays on, but I remembered it was a Sonic CinePlayer decoder pack that I installed which gave me the DTS functionality. So I just installed the pack (without the player) and voila, it decodes again.

I'll just point you in the general direction (http://hostfile.org/viewalbum.php?id=209) to get you started on how to find that decoder... lemme tell you, it comes in very handy.

Now, as for my most recent problem, which, again has no reference at all in this thread... (grr)... I've used eac3to to happily convert/extract the audio for 5 movies so far... the inconvenience of having to use Audition to downmix the tracks to stereo is nothing compared to the convenience of having it work at all. Well... until most recently. I have a 1.5gb DTS file (extracted using eac3to's DTS Core function) that plays fine in Media Player, but when asked to decode to WAV with eac3to, it results in a 6gb WAV (if 24-bit, otherwise 4gb in 16-bit) that sometimes comes out to around 15 minutes, sometimes to 55 minutes, etc... I can't tell if it's a problem with the reader/player or the file itself, but I can't get this damn thing to decode. If I use avisynth to do DirectShowSource on the DTS file and a 1x1px fake video, I can use VirtualDub to extract a full length 2 channel WAV file, properly downmixed and everything, but it ends up being glitchy - it syncs right in one position but skips a few MS and gets further and further off-sync every minute or so. A very curious DTS file indeed. Any ideas?

edit: It has to be a WAV formatter problem... I mean, if I calculate 48000(hz) * 2 (bytes; 16 bits per sample) * 840 (seconds; 14 minutes) * 6 (channels), I get 483,840,000, or about 461 MB. My dbPowerAmp properties tab shows that the file size should be (actually, it claims it "is") 475.5mb - where back on the "general" tab, it clearly says the file is 4.46gb. What the heck gives?

[madshi]

does TrueHD conversion of eac3to currently also support files with 96kHz ?
MLP decoding does. So I guess TrueHD decoding does, too.

[madshi]

Hello Madshi,
i found strange thing. When i try to convert TrueHD to DTS, it correctz detects bit depth of TrueHD but it still creates all wav files twice, one for 24 bit and the second for 16. Can i somehow prevent this behaviour? Because it requires twice space on HDD and decode is then twice slower.
Well, it doesn't make sense to even check the TrueHD bitdepth when encoding to AC3 or DTS, so I'll remove the bitdepth check in those special cases.

[madshi]

I have a 1.5gb DTS file (extracted using eac3to's DTS Core function) that plays fine in Media Player, but when asked to decode to WAV with eac3to, it results in a 6gb WAV (if 24-bit, otherwise 4gb in 16-bit) that sometimes comes out to around 15 minutes, sometimes to 55 minutes, etc... I can't tell if it's a problem with the reader/player or the file itself, but I can't get this damn thing to decode. If I use avisynth to do DirectShowSource on the DTS file and a 1x1px fake video, I can use VirtualDub to extract a full length 2 channel WAV file, properly downmixed and everything, but it ends up being glitchy - it syncs right in one position but skips a few MS and gets further and further off-sync every minute or so. A very curious DTS file indeed. Any ideas?
This has been discussed multiple times before. The WAV file header can't really handle files over 4GB. There's no proper way to store a file size of 4GB (or more) in the WAV file header. As a result if the WAV reading program/filter doesn't add special handling for such big files it will only read a part of the whole file. Some programs have special handling for such big WAV files. E.g. eac3to can read WAV files of more than 4GB. Many programs can't do that.

I mean, if I calculate 48000(hz) * 2 (bytes; 16 bits per sample) * 840 (seconds; 14 minutes) * 6 (channels), I get 483,840,000, or about 461 MB. My dbPowerAmp properties tab shows that the file size should be (actually, it claims it "is") 475.5mb - where back on the "general" tab, it clearly says the file is 4.46gb. What the heck gives?
There's no way a 1.5GB DTS file can be 14 minutes. So your calculation doesn't make any sense to me.

[Falcon4]

Since eac3to is the only program to understand DTS files, dbPowerAmp couldn't possibly have read the file, and hence there's no way I could have been referring to the DTS file in those calculations... while you came to the correct conclusion that a 1.5gb DTS file couldn't possibly be 14 minutes, you oh so happily pinned it on my calculation's fault. I was calculating the size of the raw WAV data based on the returned length (14 minutes) of the WAV file. And it turned out to be 475 MB. Beh...

Anyway, thanks for the info about the WAV files. That explains why I was able to do 5 other movies - they were all under 4gb for the WAV output. I didn't think WAV files would store their length info like that. But hey, Microsoft is Microsoft, no?

And, no, I did searches in this thread on words like "incomplete" or "partial", and both words returned no results. So I can't see how it's been discussed before unless you were speaking a different language. It's sure nice to have such quick replies though... even if a little rude.

I guess I'll try finding a way around this problem since Audition can't import FLACs and I can't split the file into chunks to process in parts... =\

edit: Is it possible to force eac3to to output individual tracks as WAVs like I've seen other outputs in this thread do? I think I'm going to try "-0", "-1", etc to get it to output single tracks while I wait for a reply though... the "-0,1,2,3,4,5" command line option may help here if it allows me to specify less than 6 channels...
editedit: Nope, it doesn't. Ugh. Okay, I'm stuck here... maybe I'll try to find a WAV splitter in the meantime.

[madshi]

And, no, I did searches in this thread on words like "incomplete" or "partial", and both words returned no results. So I can't see how it's been discussed before unless you were speaking a different language.
Search for "4GB" in this thread and you'll find plenty of posts.

even if a little rude.
Says the most polite poster in this thread... :p

edit: Is it possible to force eac3to to output individual tracks as WAVs like I've seen other outputs in this thread do? I think I'm going to try "-0", "-1", etc to get it to output single tracks while I wait for a reply though... the "-0,1,2,3,4,5" command line option may help here if it allows me to specify less than 6 channels...
editedit: Nope, it doesn't. Ugh. Okay, I'm stuck here... maybe I'll try to find a WAV splitter in the meantime.
eac3to source.dts dest.wavs

[Falcon4]

Search for "4GB" in this thread and you'll find plenty of posts.
Couldn't have known to search for that if I didn't even know that was a limitation to begin with ;)
(Maybe it would be a good idea to put a warning in the program itself that almost no programs will be able to read a 4+gb WAV file...?)

Says the most polite poster in this thread... :p
Yeah, sorry, I'm just a little ticked that the only program in existence to be able to decode a much-in-demand format is so hard to use properly... I mean, seriously, the new generation of high-definition formats are nothing but consumer-unfriendly and there are thousands of people probably dying to do this. Eac3to (and xport.exe) is(/are) the only thing that can manage the audio portion of it. And I'm even having an impossible time using it. But don't get me wrong - not being a desktop programmer, I am thankful that it exists at all. ;)

eac3to source.dts dest.wavs
Ahh -- to the documentation-editor with you then! :)
edit: That worked great!
C:\Users\Falcon\Desktop\eac3to>eac3to.exe f:\audio.wav c:\audio1.wavs
WAV, 5.1 channels, 2:18:42, 16 bits, 48khz
Reading WAV...
Writing WAVs...
Creating/writing file "c:\audio1.L.wav"...
Creating/writing file "c:\audio1.R.wav"...
Creating/writing file "c:\audio1.C.wav"...
Creating/writing file "c:\audio1.LFE.wav"...
Creating/writing file "c:\audio1.SL.wav"...
Creating/writing file "c:\audio1.SR.wav"...
-----
(and no, that's not Vista.)

edit edit: I really only wanted the L, R, C, and LFE channels (don't need, and won't be encoding/reading, the SL and SR), so maybe an option to simply not decode/write them is possible? At any rate, I seem to now have the problem solved, so thanks!!

[menlvd]

how can it work? using surcode 1.0.29

D:\minep\me_gui\tools\eac3_to>eac3to.exe 12.ac3 test.dts -1536
AC3, 5.1 channels, 0:01:51, 640kbit/s, 48khz
Decoding with DirectShow (Nero Audio Decoder 2)...
DirectShow reports 5.1 channels, 24 bits, 48khz
Writing WAVs...
Creating/writing file "test.L.wav"...
Creating/writing file "test.R.wav"...
Creating/writing file "test.C.wav"...
Creating/writing file "test.LFE.wav"...
Creating/writing file "test.SR.wav"...
Creating/writing file "test.SL.wav"...
Surcode sais/asks: "At least one valid source file must be specified to encode."
.
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...

[tebasuna51]

... the inconvenience of having to use Audition to downmix the tracks to stereo is nothing compared to the convenience of having it work at all. Well... until most recently. I have a 1.5gb DTS file (extracted using eac3to's DTS Core function) that plays fine in Media Player, but when asked to decode to WAV with eac3to, it results in a 6gb WAV (if 24-bit, otherwise 4gb in 16-bit) that sometimes comes out to around 15 minutes, sometimes to 55 minutes, etc... I can't tell if it's a problem with the reader/player or the file itself, but I can't get this damn thing to decode.
There are a limit of 4GB for wav files because a field, in the header, can't support a value greater than 2^32. You have all the data in your wav file but only a few soft can accept them.

If I use avisynth to do DirectShowSource on the DTS file and a 1x1px fake video, I can use VirtualDub to extract a full length 2 channel WAV file, properly downmixed and everything, but it ends up being glitchy - it syncs right in one position but skips a few MS and gets further and further off-sync every minute or so. A very curious DTS file indeed. Any ideas?

If you are an AviSynth-VirtualDub user I recommend you use some audio dedicated plugins instead the DirectShow-VirtualDub method.

You need an audio decoder like NicAudio.dll (http://avisynth2.sourceforge.net/NicAudio_20070821.zip) (ac3, dts, mp2, mp3, lpcm), or a plugin to manage wav >4GB RaWav.dll (http://www.mytempdir.com/2070490), and SoundOut (http://forum.doom9.org/showthread.php?t=120025), GUI to encode to flac, ape, mp2, mp3, ac3, ogg, ...

Then you can use an avs script like:
#a = RaWavSource("G:\YourPath\big.wav", 2)
a = NicDtsSource("G:\YourPath\input.dts")
# A 5 -> 2 downmix Dolby ProLogic II compatible
flr = GetChannel(a, 1, 2)
fcc = GetChannel(a, 3, 3)
lrc = MixAudio(flr, fcc, 0.3254, 0.2301)
bl = GetChannel(a, 4)
br = GetChannel(a, 5)
sl = MixAudio(bl, br, 0.2818, 0.1627)
sr = MixAudio(bl, br, -0.1627, -0.2818)
blr = MergeChannels(sl, sr)
MixAudio(lrc, blr, 1.0, 1.0)
# if you want (recommended after a downmix):
Normalize(0.95)
# And now the GUI to encode
SoundOut()
You can use NicAudio decoder for dts or let Sonic decoder and use RaWav for the big wav. You can use also other downmix method.

And open the avs in VirtualDub.

[Falcon4]

Holy Jesus, that's impressive. You didn't just come up with that on-the-fly, did you? That's cool.

I don't think I could really do a single-step process like that though, because I'm encoding with a dual-core system that I'm trying to split the audio and video up into. At the moment I process the video in two simultaneous VirtualDubMod instances for two passes (with XviD), process the audio with eac3to and Adobe Audition (which the new "wavs" option will greatly assist since Audition has to re-split and scan the multichannel WAV on opening), manually apply DRC to the center channel to my ears' content (cue "You Bastard!" remarks), save as 192k CBR MP3 (again with the "You Bastard!"), then merge the two parts and mux the single audio file in one single pass (yay). It all actually works rather nicely, albeit taking a long time for the VDubMod MPEG2 parsing of 20+gb video files and whatnot. Although if I time it just right and only have a single video file to work with, I can open (parse) the same file in two VDubMod instances at the same time and they'll just fight over Windows' disk cache to read the file only once. But the result is a pristine quality 4.37gb 1080p AVI file that fits perfectly on a DVD-R. :)

Maybe I'll write a guide to Blu-Ray to XviD conversion. I imagine that would help a good number of people... :)

edit: To clean up a little double-talk, I'm saying here that I like using Audition as opposed to a single-step process, where above I said Audition was a pain in the rear - yes, that was true, before a certain few minutes revealed that I could split the DTS decoding into individual WAVs (removing Audition's tedious split-and-rewrite time AND the scan time if I hit "cancel" and zoom in with multitrack view), as well as have Audition apply DRC to make voices more audible with just a few clicks. Yay for extra steps ensuring quality :)

[tebasuna51]

edit edit: I really only wanted the L, R, C, and LFE channels (don't need, and won't be encoding/reading, the SL and SR), so maybe an option to simply not decode/write them is possible?

You are free to delete the files but I think is a minority request.

And warning, mix the LFE channel with front channels can destroy info, Dolby don't recommend use the LFE channel in downmix, only if you are sure the info in LFE is not present in front channels.

You still can use the AviSynth method with monowavs to do the downmix more easy than Audition method:

fl = WavSource("c:\audio1.L.wav")
fr = WavSource("c:\audio1.R.wav")
fc = WavSource("c:\audio1.C.wav")
...
and mix and merge like you want.

[shambles]

madshi, earlier in the thread you posted that you converted pirates of the caribbean 24bit pcm to flac and that the filesize was bigger than 2gb.. was that dead man's chest or curse of the black pearl?

the reason i'm asking is because i just converted the curse of the black pearl pcm track and the bitrate is 1395 kbps, filesize 1.39gb despite both the input pcm and the output flac track being 24bit. i sort of hope something has gone wrong at my end because having a 16 bit track in a 24 bit file would be amazingly stupid when it's pcm...

sample http://www.sendspace.com/file/4730s7

any comment on this? maybe you could have bit depth checking for pcm tracks also..

[Furiousflea]

Please try what I already suggested in my previous reply to you, namely encoding to ac3 to make sure that the problem is the decoding and not the WAVs file writing.


The Sonic decoder doesn't seem to be installed correctly on that different computer.

Hi madshi, sorry I didn't let you know if it was working with AC3 encoding. It doesn't, I get exactly the same results with that (progress bar stops half way through)....hmmm thanks for any help :)

[The_Keymaker]

Hello Forum,

Based on suggestions from madshi and ACrowley, I will try and update EAC3toGUI to accommodate the newest version of eac3to.

I hope to have something ready early next week.

Regards,
The_Keymaker

[Chumbo]

Hello Forum,

Based on suggestions from madshi and ACrowley, I will try and update EAC3toGUI to accommodate the newest version of eac3to.

I hope to have something ready early next week.

Regards,
The_Keymaker
Thanks for the continued effort. :)

[DreckSoft]

Fist let me say thanks for the great tool you're providing.

Unfortunately the downconvert option doesn't seem to work properly. I tried to convert a PCM Track (5.1 24bit) to FLAC (20bit - 16 doesn't work either).


eac3to.exe in.pcm out_20.pcm -down20
This might be a RAW/PCM file. Trying to figure out the details.
This will probably take a while. Please be patient...
The RAW/PCM file seems to be big endian.
The RAW/PCM file seems to have a bitdepth of 24 bits.
The RAW/PCM file seems to have 6 channels.
RAW/PCM, 5.1 channels, 1:50:33, 24 bits, 48khz
This is probably a Blu-ray PCM track. Will remap channels accordingly.
Reading RAW/PCM...
Swapping endian...
Remapping channels...
Reducing depth from 24 to 20 bits...
Swapping endian...
Remapping channels...
Loading white noise (needed for dithering)...
Creating/writing file "in.pcm"...
The endian swapper crashed.


Btw: Which compression level do you use when encoding flac? I assume it's 5 but I would prefer 8. Could you offer a command line switch for this?

It's strange that
Swapping endian...
Remapping channels...
seems to be done twice.

[nautilus7]

The dts encoding (with surcode) doesn't work. I 've used both 1.0.21 and 1.0.29 but i get the same error message:

Surcode sais/asks: "At least one valid source file must be specified to encode.".
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...

I don't type relative paths (like someone above).

Is there anyone that can actually use surcode to encode a dts file?

[idbirch2]

Can anyone offer any tips on converting PCM? I've always managed to avoid it so far but I now have a BluRay (Open Season) where the only English audio is lossless PCM.

eac3to accepts the .pcm input and completes the conversion but the resulting ac3 file has a really loud buzzing noise. I've tried forcing both Nero and Sonic filters with no change. Here's the eac3to output:

H:\>"C:\Program Files\Audio\eAC3to\eac3to.exe" "H:\os.pcm" "H:\os.ac3"
This might be a RAW/PCM file. Trying to figure out the details.
This will probably take a while. Please be patient...
The RAW/PCM file seems to be big endian.
The RAW/PCM file seems to have a bitdepth of 16 bits.
The RAW/PCM file seems to have 6 channels.
RAW/PCM, 5.1 channels, 0:03:00, 16 bits, 48khz
This is probably a Blu-ray PCM track. Will remap channels accordingly.
Reading RAW/PCM...
Swapping endian...
Remapping channels...
Encoding AC3...
Creating/writing file "H:\os.ac3"...

The short duration is due to the movie being split over 3 .m2ts files, this example is the first 2 joined together.

[Thunderbolt8]

why dont you convert to flac?

[idbirch2]

Erm, because I don't want to? Back to the problem...

[Thunderbolt8]

well you didnt mention that, you only spoke of pcm conversion in general. and afaik conversion to flac doesnt cause these problems (at least not for me yet) thats why I mentioned it.


I wanted to convert the DTS-HD hi-res track from basinc instinct blu-ray to flac, but im not sure if everything went ok. as advertised the track is supposed to be 5.1, but eac3to said something of 7 channels and made 6.1 out of it or something like that.

heres a 10mb sample: http://www.sendspace.com/file/132bk5

the original track size (24-bit) is: 2.72 GB, the flac size is 3.75 GB

[madshi]

Maybe it would be a good idea to put a warning in the program itself that almost no programs will be able to read a 4+gb WAV file...?
Added.

Ahh -- to the documentation-editor with you then! :)
Added "WAVs" info to the eac3to help output.

[madshi]

how can it work? using surcode 1.0.29

Surcode sais/asks: "At least one valid source file must be specified to encode."
.
Pressing the Surcode "Encode" button didn't seem to work...
Closing Surcode...
Fixed in v2.08.

[madshi]

any comment on this? maybe you could have bit depth checking for pcm tracks also..
I've checked Curse of the Black Pearl. Most of the track is really only 16bit. However, there are some very few parts of the track which contain more than 24bit. E.g. the first 700ms of the track are 24bit. Then there is half an hour of only 16bit. Then suddenly there are some samples with more than 24bit again. Fortunately FLAC is a very effective compressor. Most of the track is only 16bit, so the FLAC file ends up with a 16bit comparable file size.

Interestingly, on my German Curse of the Black Pearl the English PCM track is only 16bit! I've compared the two tracks. The 16bit track from my German disc was downconverted from the 24bit track by using some kind of dithering, maybe even noise shaping. Normally this is a good method to do the downconversion to 16bit. But since 99% of the track were only 16bit to begin with, the dithering harms more than it helps. I've compared the FLAC files of the 2 tracks. And actually the 24bit FLAC file from the English 24bit PCM track is *smaller* by about 10% compared to the 16bit FLAC file from the English 16bt PCM track from my German disc. Very strange. Seemingly the dithering noise makes compression more difficult.

Version v2.08 now has an (undocumented) new switch "-check16bit" with which you can force eac3to to check whether a 24bit decoded datastream only contains 16bit of real information. You can use it on PCM and WAV input tracks.

[madshi]

The dts encoding (with surcode) doesn't work. I 've used both 1.0.21 and 1.0.29 but i get the same error message
Conversion with 1.0.21 should still work, but only if you don't have 1.0.29 installed yet. eac3to v2.08 should also work with 1.0.29 now.

[madshi]

Can anyone offer any tips on converting PCM? I've always managed to avoid it so far but I now have a BluRay (Open Season) where the only English audio is lossless PCM.

eac3to accepts the .pcm input and completes the conversion but the resulting ac3 file has a really loud buzzing noise. I've tried forcing both Nero and Sonic filters with no change. Here's the eac3to output:

H:\>"C:\Program Files\Audio\eAC3to\eac3to.exe" "H:\os.pcm" "H:\os.ac3"
This might be a RAW/PCM file. Trying to figure out the details.
This will probably take a while. Please be patient...
The RAW/PCM file seems to be big endian.
The RAW/PCM file seems to have a bitdepth of 16 bits.
The RAW/PCM file seems to have 6 channels.
RAW/PCM, 5.1 channels, 0:03:00, 16 bits, 48khz
This is probably a Blu-ray PCM track. Will remap channels accordingly.
Reading RAW/PCM...
Swapping endian...
Remapping channels...
Encoding AC3...
Creating/writing file "H:\os.ac3"...

The short duration is due to the movie being split over 3 .m2ts files, this example is the first 2 joined together.
eac3to's output looks alright to me. Which tool did you use to demux the audio? Does conversion work if you convert the 3 m2ts PCM files separately to AC3? It would help if you could upload a little sample of the PCM file. Of course that will help only if the noise problem can be reproduced with that sample.

You might also want to test WAV or FLAC writing instead of AC3. I'm not trying to convince you not to use AC3. But trying WAV or FLAC would belp us finding out where the problem is located. If the AC3 file is noisy but the WAV and FLAC files sound correct then the problem occurs during AC3 encoding. If the WAV and FLAC files have the same noise as the AC3 encoding, then the problem must be in the reading or interpretation of the PCM data.

[madshi]

I wanted to convert the DTS-HD hi-res track from basinc instinct blu-ray to flac, but im not sure if everything went ok. as advertised the track is supposed to be 5.1, but eac3to said something of 7 channels and made 6.1 out of it or something like that.

heres a 10mb sample: http://www.sendspace.com/file/132bk5

the original track size (24-bit) is: 2.72 GB, the flac size is 3.75 GB
This really seems to be a 6.1 DTS-HD High Resolution track. I'm not sure, maybe the back surround channel is all empty? In the sample you sent me the back surround channel seems to be idle, but the sample is very short. So it's hard to say for me. Anyway, eac3to definitely behaves correctly. The DTS-HD file is flagged as 6.1 and decodes as such.

[madshi]

Unfortunately the downconvert option doesn't seem to work properly. I tried to convert a PCM Track (5.1 24bit) to FLAC (20bit - 16 doesn't work either).

eac3to.exe in.pcm out_20.pcm -down20
Why are you doing "in.pcm out_20.pcm"? Didn't you just say you wanted to convert to FLAC? :confused:

[QUOTE=DreckSoft;1071257]Btw: Which compression level do you use when encoding flac? I assume it's 5 but I would prefer 8. Could you offer a command line switch for this?
I'm using strongest compression (8). Does anybody want lighter compression? Don't see a need for that. But if there are enough voices who want lighter compression, I might add a switch.

It's strange that
Swapping endian...
Remapping channels...
seems to be done twice.
That's correct. Every input "module" converts the raw audio data to little endian and the default WAV channel order. Since you chose "PCM" as output format, eac3to has to convert endian and reswap channels for output again.

[DreckSoft]

[QUOTE=DreckSoft;1071257]eac3to.exe in.pcm out_20.pcm -down20
Why are you doing "in.pcm out_20.pcm"? Didn't you just say you wanted to convert to FLAC? :confused:


I changed the filenames and made a typo.

But the problem also exists when doing PCM => PCM.

[Penecho]

Hi, i have laso a problem with my audio file. My Source file is a 5.1 DTSHD Track from a BluRay Disc.

xxxxxxxxxx\eac3to.exe" "file.dtshd" "file.ac3" -640
DTS Master Audio, 5.1 channels, 24 bits, 48khz
Decoding with DirectShow (Sonic Audio Decoder)...
DirectShow reports 5.1 channels, 24 bits, 48khz
Encoding AC3...
Creating/writing file "file.ac3"...


than he creates a 163kb *.ac3 big file and nothing else happens :( ( I waited about 1 hour, but filesize did not change)

I am using the latest version of eac3to


Cu

[nautilus7]

Conversion with 1.0.21 should still work, but only if you don't have 1.0.29 installed yet. eac3to v2.08 should also work with 1.0.29 now.
No it didn't that's why i installed 1.0.29, but no luck either. I am waiting for 2.08. ;)

[madshi]

Hi madshi, sorry I didn't let you know if it was working with AC3 encoding. It doesn't, I get exactly the same results with that (progress bar stops half way through)....hmmm thanks for any help :)
Can't reproduce any problems with DTS-HD decoding here. I've just today converted a Blu-Ray 24bit DTS-HD Master Audio track (Eragon) and a HD DVD 16bit DTS-HD Master Audio track (Total Recall) to FLAC without any problems.

I can only guess that your DTS-HD file is faulty. Which movies does it come from? With which tool did you extract it?

[madshi]

Hi, i have laso a problem with my audio file. My Source file is a 5.1 DTSHD Track from a BluRay Disc.

xxxxxxxxxx\eac3to.exe" "file.dtshd" "file.ac3" -640
DTS Master Audio, 5.1 channels, 24 bits, 48khz
Decoding with DirectShow (Sonic Audio Decoder)...
DirectShow reports 5.1 channels, 24 bits, 48khz
Encoding AC3...
Creating/writing file "file.ac3"...

than he creates a 163kb *.ac3 big file and nothing else happens :( ( I waited about 1 hour, but filesize did not change)
Hmmmm... Same as with Furiousflea's problem: I can't reproduce the problem here. Probably your DTS-HD file is faulty. Which movie does it come from? With which tool did you extract it?

[Penecho]

Live free or Die Hard, demuxed with TSremux



Cu

[DreckSoft]

Live free or Die Hard, demuxed with TSremux



Cu

That movie has BD+ protection, so the content should be still encrypted. Try to play the file (or the core), it shouldn't work either.

[Penecho]

No that works, i also demuxed the other audio tracks and used eac3to to make *.ac3 files.



Cu

[nautilus7]

Live free or die hard is BD+, that's true. No encoding is possible, yet.

[Thunderbolt8]

This really seems to be a 6.1 DTS-HD High Resolution track. I'm not sure, maybe the back surround channel is all empty? In the sample you sent me the back surround channel seems to be idle, but the sample is very short. So it's hard to say for me. Anyway, eac3to definitely behaves correctly. The DTS-HD file is flagged as 6.1 and decodes as such.
since the first one was maybe a bit too short, I made another 50mb sample: http://www.sendspace.com/file/dk0bcz
I just cant test channel distribution myself, because I only have a stereo system :P

[DreckSoft]

No that works, i also demuxed the other audio tracks and used eac3to to make *.ac3 files.


The stream is only partially encrypted. Try to play the file in ZoomPlayer using CoreAVC codec (CyberLink decoder skipps all frames with decoding errors so you'll see a "jumping" video). You'll see a partially distorted video. Audio seems to be fine also I've heard a few clicks. Lets say random 10% of the stream is encrypted using BD+. So most of the encryption will be in the video stream. Less in a PCM stream and maybe none in the AC3 stream. Even if there are some encrypted frames in the extracted AC3 you'll only find them if you listen to the entire stream (or maybe check it using delaycur or similar).

[madshi]

Agreed. BD+ movies have some "random" garbage bytes added over the video/audio data. That's most probably the reason why DTS-HD conversion fails.

[idbirch2]

eac3to's output looks alright to me. Which tool did you use to demux the audio? Does conversion work if you convert the 3 m2ts PCM files separately to AC3? It would help if you could upload a little sample of the PCM file. Of course that will help only if the noise problem can be reproduced with that sample.

You might also want to test WAV or FLAC writing instead of AC3. I'm not trying to convince you not to use AC3. But trying WAV or FLAC would belp us finding out where the problem is located. If the AC3 file is noisy but the WAV and FLAC files sound correct then the problem occurs during AC3 encoding. If the WAV and FLAC files have the same noise as the AC3 encoding, then the problem must be in the reading or interpretation of the PCM data.
Hi madshi, I demuxed the .pcm file using TSRemux and I have discovered I have the same problem when converting to plain WAV. I have uploaded the PCM file here (http://rapidshare.com/files/73804314/OpenSeason.pcm.html).

[madshi]

i've got a truehd track that's 2.13gb and using nero filters, the flac i encoded from it turned out to be 3.26gb. using -libav, the flac is 1.88gb

i guess it's a 20bit track.. here's a 5mb sample http://rapidshare.com/files/72692697/001.thd.html
Fixed in v2.08.

[madshi]

Almost every time i do a conversion, i like to decode the input and output files to wavs, and compare them in a wave editor like audacity. One channel is enough for this job and my hdd will write it faster.
Added the undocumented new switch "-mono" for this purpose in v2.08. It extracts the center channel from multichannel sources respectively the front left channel from stereo sources.

[nautilus7]

Added the undocumented new switch "-mono" for this purpose in v2.08. It extracts the center channel from multichannel sources respectively the front left channel from stereo sources.
Thank you so much. I am looking forward to 2.08.

[madshi]

eac3to v2.08 released

http://madshi.net/eac3to.zip

* fixed: bitdepth reducer sometimes crashed when being fed a PCM file
* fixed: FLAC encoder sometimes crashed when delay was applied
* fixed: some TrueHD files were dithered/processed by Nero when they shouldn't
* fixed: Surcode 1.0.29 encoding automation
* fixed: source file was deleted when source and dest file names were identical
* eac3to output is now always written to "log.txt"
* when a crash occurs, "log.txt" is added to the bug report
* improved help text + hints slightly
* undocumented switch "-check16bit" added
* undocumented switch "-mono" added

[madshi]

Hi madshi, I demuxed the .pcm file using TSRemux and I have discovered I have the same problem when converting to plain WAV. I have uploaded the PCM file here (http://rapidshare.com/files/73804314/OpenSeason.pcm.html).
That PCM file is definitely corrupt. If you check it with a hex editor, the file begins with the following 4 bytes:

0B 40 91 40
Furthermore you'll find those bytes regularly throughout the whole PCM file. These bytes are not PCM audio data. Instead they're part of the m2ts file structure. This is what causes the noise. Please report this bug in the TsRemux thread and use xport in the meanwhile for demuxing PCM.

[Thunderbolt8]

btw. I also have a madflac problem (?) with that dtshd file I sent you. when I remux with video + flac, the video gives me already enough trouble anyway with glitches (AVC), but together with this flac file the file wont open at all. when I only mux the video then it takes some time to load and I also get crashes when jumping back and forth, but as already said it wont start at all together with a flac file from that dtshd track.

[idbirch2]

You were right madshi, using xport to demux worked a treat, thanks for the advice. Have you thought any more about setting up a donations page?

[shambles]

I've checked Curse of the Black Pearl. Most of the track is really only 16bit. However, there are some very few parts of the track which contain more than 24bit. E.g. the first 700ms of the track are 24bit. Then there is half an hour of only 16bit. Then suddenly there are some samples with more than 24bit again. Fortunately FLAC is a very effective compressor. Most of the track is only 16bit, so the FLAC file ends up with a 16bit comparable file size.

Interestingly, on my German Curse of the Black Pearl the English PCM track is only 16bit! I've compared the two tracks. The 16bit track from my German disc was downconverted from the 24bit track by using some kind of dithering, maybe even noise shaping. Normally this is a good method to do the downconversion to 16bit. But since 99% of the track were only 16bit to begin with, the dithering harms more than it helps. I've compared the FLAC files of the 2 tracks. And actually the 24bit FLAC file from the English 24bit PCM track is *smaller* by about 10% compared to the 16bit FLAC file from the English 16bt PCM track from my German disc. Very strange. Seemingly the dithering noise makes compression more difficult.

Version v2.08 now has an (undocumented) new switch "-check16bit" with which you can force eac3to to check whether a 24bit decoded datastream only contains 16bit of real information. You can use it on PCM and WAV input tracks.

that's curious, 16 and 24bit data in the same track. wouldn't have expected that.

i've also noticed that dithering hurts compressibility, which is interesting indeed.. anyways.

eac3to v2.08 released

thank you very very much for all your continuing amazing work :D

just out of interest, did you find out whether that truehd sample was 18 or 20 bit?

[Thunderbolt8]

is there a way to detect which truehd tracks could possibly have been dithered/processed by nero, when they are inside .mkv with video ?

[madshi]

btw. I also have a madflac problem (?) with that dtshd file I sent you. when I remux with video + flac, the video gives me already enough trouble anyway with glitches (AVC), but together with this flac file the file wont open at all. when I only mux the video then it takes some time to load and I also get crashes when jumping back and forth, but as already said it wont start at all together with a flac file from that dtshd track.
Have you demuxed video with TsRemux or with xport?

I don't think this is a madFlac problem. It's more likely to be a problem related to the video part which reflects back to the FLAC audio track somehow. Your DTS-HD sample definitely is 6.1 and all 7 channels have real information in them.

[madshi]

Have you thought any more about setting up a donations page?
Didn't have time for that yet. Developing eac3to is more important, right? :D

[madshi]

just out of interest, did you find out whether that truehd sample was 18 or 20 bit?
Didn't really check. I just hacked around with the TrueHD headers until Nero finally gave out the same result as libav.

[madshi]

is there a way to detect which truehd tracks could possibly have been dithered/processed by nero, when they are inside .mkv with video ?
Not really.

[idbirch2]

Hi madshi, well I went back to that strange DTS file to see if v2 can handle it and it kind of does as it now accepts the input. Trouble is, if I choose to convert to AC3, I get:

DTS Master Audio, 3/1 channels, 24 bits, 48khz
Decoding with DirectShow (Sonic Audio Decoder)...
DirectShow reports 4 channels, 24 bits, 48khz
The AC3 encoder received a non-supported data format.

If I try and convert to WAV I just get:

DTS Master Audio, 3/1 channels, 24 bits, 48khz
Decoding with DirectShow (Sonic Audio Decoder)...
DirectShow reports 4 channels, 24 bits, 48khz
Writing WAV...
Creating/writing file "H:\ed.wav"...

And then nothing, eac3to maxes one of my CPU cores but nothing happens, the .wav never grows and its been running for over 20 mins now with no HDD activity. Any thoughts?

I've got a BluRay with a really weird DTS audio format. When I try and run eac3to on it I get:

DTS Master Audio, 3/1 channels, 24 bits, 48khz
This channel format is currently not supported.

Is there any pre-processing I can do to get a compatible stream or is there anything I can provide/upload which will help add support for this audio type?

[nautilus7]

The new -mono switch works fine.
DTS encoding also works now (surcode version 1.0.29).

@ madshi
I have another request: I would like you to inform us which version is every decoder/encoder (aften, flac, etc) that comes with eac3to. Or at least tell it when you update them.

[madshi]

Hi madshi, well I went back to that strange DTS file to see if v2 can handle it and it kind of does as it now accepts the input. Trouble is, if I choose to convert to AC3, I get:

DTS Master Audio, 3/1 channels, 24 bits, 48khz
Decoding with DirectShow (Sonic Audio Decoder)...
DirectShow reports 4 channels, 24 bits, 48khz
The AC3 encoder received a non-supported data format.
Currently eac3to's Aften encoder only supports 1, 2 and 6 channels. I'm not sure if Aften itself can encode 3/1 channels. Will have to check that.

If I try and convert to WAV I just get:

DTS Master Audio, 3/1 channels, 24 bits, 48khz
Decoding with DirectShow (Sonic Audio Decoder)...
DirectShow reports 4 channels, 24 bits, 48khz
Writing WAV...
Creating/writing file "H:\ed.wav"...

And then nothing, eac3to maxes one of my CPU cores but nothing happens, the .wav never grows and its been running for over 20 mins now with no HDD activity. Any thoughts?
It works for me with the sample you sent me. Maybe you can make a new sample which reproduces this new problem?

[madshi]

I have another request: I would like you to inform us which version is every decoder/encoder (aften, flac, etc) that comes with eac3to. Or at least tell it when you update them.
Yeah, makes sense. Will have to check myself, don't know right now. I do know that FLAC is currently 1.2.0 because mkvtoolnix doesn't like 1.2.1 encoded FLAC files.

[nautilus7]

Thanks.

I remember saying that about flac 1.2.1. Do you mean 1.2.1b, because i know someone who had success muxing an 1.2.1b flac track into .mkv with mkvtoolnix. I am checking it myself right now though.

[TheSof]

Madshi,

Sometime ago you spoke of the possibility of perhaps making the nero decoder bypass drc in realtime within any dshow player. Is this still something you are still looking into? If not thats fine, but would hate to eac3to a huge stack of material and then find out it was unnecessary.

Cheers.

[madshi]

I remember saying that about flac 1.2.1. Do you mean 1.2.1b, because i know someone who had success muxing an 1.2.1b flac track into .mkv with mkvtoolnix. I am checking it myself right now though.
Not sure.

[madshi]

Sometime ago you spoke of the possibility of perhaps making the nero decoder bypass drc in realtime within any dshow player. Is this still something you are still looking into? If not thats fine, but would hate to eac3to a huge stack of material and then find out it was unnecessary.
That's on my to do list. Actually I was hoping to get it done yesterday, but time was too short. Might take another 1-2 weeks.

[nautilus7]

madshi,

I just used this (http://downloads.sourceforge.net/flac/flac-1.2.1-devel-win.zip) to replace the one that comes with eac3to. Works perfectly with mkvtoolnix.

[madshi]

Ok, I'll recheck FLAC 1.2.1.

[DreckSoft]

I retried converting the problematic PCM file with V2.08 but there are still problems.

PCM => DTS

D:\>eac3to.exe EN.pcm EN.dts -1536
This might be a RAW/PCM file. Trying to figure out the details.
This will probably take a while. Please be patient...
The RAW/PCM file seems to be big endian.
The RAW/PCM file seems to have a bitdepth of 24 bits.
The RAW/PCM file seems to have 6 channels.
RAW/PCM, 5.1 channels, 1:50:33, 24 bits, 48khz
This is probably a Blu-ray PCM track. Will remap channels accordingly.
Reading RAW/PCM...
Swapping endian...
Remapping channels...
Writing WAVs...
Creating/writing file "EN.L.wav"...
Creating/writing file "EN.R.wav"...
Creating/writing file "EN.C.wav"...
Creating/writing file "EN.LFE.wav"...
Creating/writing file "EN.SR.wav"...
Creating/writing file "EN.SL.wav"...

Stops after creating 6 wavs with 3kb each

PCM (24bit) => FLAC (20bit)

D:\>eac3to.exe EN.pcm EN.flac -down20
This might be a RAW/PCM file. Trying to figure out the details.
This will probably take a while. Please be patient...
The RAW/PCM file seems to be big endian.
The RAW/PCM file seems to have a bitdepth of 24 bits.
The RAW/PCM file seems to have 6 channels.
RAW/PCM, 5.1 channels, 1:50:33, 24 bits, 48khz
This is probably a Blu-ray PCM track. Will remap channels accordingly.
Reading RAW/PCM...
Swapping endian...
Remapping channels...
Reducing depth from 24 to 20 bits...
Encoding FLAC...
Loading white noise (needed for dithering)...
The endian swapper crashed.
Creating/writing file "EN.flac"...


Without -down20 it works. -down16 crashes too.

The Source is not BD+ protected, the pcm file (and the video) plays fine.

[Wilbert]

Something is wrong with 1.2.1b (1.2.1a works fine though), but i forgot the problem (24 bit output broken?) :)

[madshi]

I retried converting the problematic PCM file with V2.08 but there are still problems.

PCM => DTS

D:\>eac3to.exe EN.pcm EN.dts -1536
This might be a RAW/PCM file. Trying to figure out the details.
This will probably take a while. Please be patient...
The RAW/PCM file seems to be big endian.
The RAW/PCM file seems to have a bitdepth of 24 bits.
The RAW/PCM file seems to have 6 channels.
RAW/PCM, 5.1 channels, 1:50:33, 24 bits, 48khz
This is probably a Blu-ray PCM track. Will remap channels accordingly.
Reading RAW/PCM...
Swapping endian...
Remapping channels...
Writing WAVs...
Creating/writing file "EN.L.wav"...
Creating/writing file "EN.R.wav"...
Creating/writing file "EN.C.wav"...
Creating/writing file "EN.LFE.wav"...
Creating/writing file "EN.SR.wav"...
Creating/writing file "EN.SL.wav"...

Stops after creating 6 wavs with 3kb each

PCM (24bit) => FLAC (20bit)

D:\>eac3to.exe EN.pcm EN.flac -down20
This might be a RAW/PCM file. Trying to figure out the details.
This will probably take a while. Please be patient...
The RAW/PCM file seems to be big endian.
The RAW/PCM file seems to have a bitdepth of 24 bits.
The RAW/PCM file seems to have 6 channels.
RAW/PCM, 5.1 channels, 1:50:33, 24 bits, 48khz
This is probably a Blu-ray PCM track. Will remap channels accordingly.
Reading RAW/PCM...
Swapping endian...
Remapping channels...
Reducing depth from 24 to 20 bits...
Encoding FLAC...
Loading white noise (needed for dithering)...
The endian swapper crashed.
Creating/writing file "EN.flac"...


Without -down20 it works. -down16 crashes too.

The Source is not BD+ protected, the pcm file (and the video) plays fine.
Are you sure that this is really v2.08? :confused: I was sure that I had fixed the endian swapper crash... :(

[madshi]

Something is wrong with 1.2.1b (1.2.1a works fine though), but i forgot the problem (24 bit output broken?) :)
Ah - thanks!

[DreckSoft]

Aaaargh.
I've downloaded the file shortly after you posted the release of 2.08. It seems there was still the old zip online (or some browser cache problem). It was 2.07.

2.08 should work in both cases. Encoding is still in progress but it looks good.

Thanks!

[Snowknight26]

eac3to.exe "W:\Encoding Tools\temp\eragon.mpa" eragon.c2q.flac -down16 -libav
DTS Master Audio, 5.1 channels, 24 bits, 48khz
Decoding with DirectShow (Sonic Audio Decoder)...
Getting "Sonic Audio Decoder" instance failed.

Shouldn't be using Sonic, should it?

Edit: Ahh, nevermind, still need Sonic for DTS decoding it seems. I wonder if you'll ever implement DTS decoding using libdts or something similar.

[ACrowley]

eac3to.exe "W:\Encoding Tools\temp\eragon.mpa" eragon.c2q.flac -down16 -libav
DTS Master Audio, 5.1 channels, 24 bits, 48khz
Decoding with DirectShow (Sonic Audio Decoder)...
Getting "Sonic Audio Decoder" instance failed.

Shouldn't be using Sonic, should it?

Edit: Ahh, nevermind, still need Sonic for DTS decoding it seems. I wonder if you'll ever implement DTS decoding using libdts or something similar.

Why ? Sonic decodes DTS perfect. Including DTS HD and DTS ES 6.1
And its a Reference Decoder which should deliver better Quality compared to ffmpeg

[madshi]

Why ? Sonic decodes DTS perfect. Including DTS HD and DTS ES 6.1
And its a Reference Decoder which should deliver better Quality compared to ffmpeg
Exactly. Well, I could probably still add libav DTS decoding support. But I'm not sure if I really should do that since it can only be of worse quality compared to Sonic's decoder...

[Inventive Software]

Exactly. Well, I could probably still add libav DTS decoding support. But I'm not sure if I really should do that since it can only be of worse quality compared to Sonic's decoder...

Hold it... if it's a perfect implementation, it should be bit-identical to Sonic's version...

[madshi]

Hold it... if it's a perfect implementation, it should be bit-identical to Sonic's version...
But it isn't.

[Inventive Software]

What about using libdca, aka libdts?

[madshi]

Well, I'd have to spend precious time to support another totally new library without any real benefit - apart from not needing the Sonic decoder for conventional DTS tracks. And what about DTS-HD? No decoder apart from Sonic can decode DTS-HD! But if you want you can test that libdca/libdts decoder. If you find that it gives out results that are bit identical to Sonic for conventional DTS tracks then I might consider adding support for that.

[Inventive Software]

I actually don't use your software, but merely wanted to point out the other solutions. libdca's been around for a while, but is still in it's infancy. libdts is what's used in AC3Filter. And, I don't have Sonic's decoder so can't make the comparison! :(

[madshi]

I've compared AC3Filter's DTS decoding to Sonic's and Nero's DTS decoding. AC3Filter's decoding is too loud and definitely not bit identical.

Look, I appreciate suggestions (I really do!). It's just that I want to offer only the highest quality solutions. Furthermore my programming time is quite limited. Because of that I don't really want to spend time to add further decoders which are of worse quality than what eac3to already supports right now.

[nautilus7]

It's just that I want to offer only the highest quality solutions.That's the spirit!!! :D

[Inventive Software]

I've compared AC3Filter's DTS decoding to Sonic's and Nero's DTS decoding. AC3Filter's decoding is too loud and definitely not bit identical.

Look, I appreciate suggestions (I really do!). It's just that I want to offer only the highest quality solutions. Furthermore my programming time is quite limited. Because of that I don't really want to spend time to add further decoders which are of worse quality than what eac3to already supports right now.

I wondered what was wrong with Die Hard 4.0.... :D

I appreciate your lack of time, and didn't know libdts had deficiencies.

[madshi]

didn't know libdts had deficiencies.
Well, it's a general thing with open source lossy decoders (AC3 and DTS). I believe they're decoding "correctly", but the decoding code works differently than the reference decoder (used by Nero and Sonic) does. So the result is not bit identical. Since AC3 and (conventional) DTS are lossy codecs, it's probably not correct to say that the reference decoder is "right" and the open source decoders are "wrong". But I guess that the reference decoder should at least not be worse than the open source decoders, maybe slightly better. Using the reference decoders is the "safe" way IMHO. And when talking about DTS, Sonic is the only decoder that can decode DTS-HD. So IMO Sonic is absolutely the way to go for DTS decoding right now.

Not sure whether there's a problem with libdts or whether AC3Filter uses libdts incorrectly.

[maxpower2078]

I searched this post and couldn't seem to find this answer, so here I am.

I am trying to use this great program to extract the core DTS or AC3 tracks out of DTS-HD or DD+ tracks after demuxing them with EVOdemux and using the mpa file as the input.

with the following command:
eac3to sourcefile destfile -core

source is e-ac3, 5.1 ......640....

I get this operation is not permitted or something similar. Sorry, I am not in front of the machine right now.

I then tried to just downconvert it to see if that worked

eac3to sourcefile destfile -192


source is e-ac3..5.1....640....

I get the same message as well

I have the sonic 4.3 decoder installed as well as powerdvd ultra. I am wondering if I need the nero decoder or something?


My goal in this is to have the streams separate so that I can play then in media player classic and output over SPDIF in their full multi channel glory as I haven't been able to do so with the original format.

[nautilus7]

Only DTS-HD tracks have a simple DTS core in them. Extract it using:

eac3to input.dtshd output.dts -core

E-AC3 tracks don't have any AC3 core in them. Only Blu-ray TrueHD tracks might have AC3 frames (track) in them, but that's another thing.
To convert (encode) an E-AC3 track to AC3 you have to type:

eac3to input.eac3 output.ac3 -your desired bitrate

As you see in input file i set the extension to what is the real one. You don't have to do this. You can leave it as it is (.mpa). eac3to can automatically determine the input extension.

[maxpower2078]

Only DTS-HD tracks have a simple DTS core in them. Extract it using:

eac3to input.dtshd output.dts -core

E-AC3 tracks don't have any AC3 core in them. Only Blu-ray TrueHD tracks might have AC3 frames (track) in them, but that's another thing.
To convert (encode) an E-AC3 track to AC3 you have to type:

eac3to input.eac3 output.ac3 -your desired bitrate

As you see in input file i set the extension to what is the real one. You don't have to do this. You can leave it as it is (.mpa). eac3to can automatically determine the input extension.

Great, I'll give it a try when I get home, but the only thing different from your code is that I didn't put an .ac3 extension on my output file and kept the mpa extension too. Will this cause it to fail?

Also once I get past this step, how do I determine if there is a delay needed and what it would be?

Also, if I need to get the nero audio decoder, what version is the best to use? I know the changelog says to not use 8 and to use 7, but does it come in the bundle package or something?

[nautilus7]

Great, I'll give it a try when I get home, but the only thing different from your code is that I didn't put an .ac3 extension on my output file and kept the mpa extension too. Will this cause it to fail?
Sure it does!
.mpa doesn't mean anything to eac3to either it's input or output. It's just the extension that evodemux gives to all audio tracks (don't know where it comes from).
Input extension doesn't matter to be set correctly because, as i said before, eac3to can find out what is the input file by its structure, without have to look at the extension.
Output is a different thing. eac3to understands what has to do by looking to the output extension. If it's .ac3 understands that has to do ac3 encoding, if it's .flac understands that has to do flac encoding, etc.

Also once I get past this step, how do I determine if there is a delay needed and what it would be? You can get this value using evodemux. The track comes from an HD DVD, right? You have to load the first .evo of the movie and look at the Firts PTS value of the video and audio stream. There you can find the delay needed, if any.

Also, if I need to get the nero audio decoder, what version is the best to use? I know the changelog says to not use 8 and to use 7, but does it come in the bundle package or something?Only Nero 7 works and the decoders are included.

[maxpower2078]

Sure it does!
.mpa doesn't mean anything to eac3to either it's input or output. It's just the extension that evodemux gives to all audio tracks (don't know where it comes from).
Input extension doesn't matter to be set correctly because, as i said before, eac3to can find out what is the input file by its structure, without have to look at the extension.
Output is a different thing. eac3to understands what has to do by looking to the output extension. If it's .ac3 understands that has to do ac3 encoding, if it's .flac understands that has to do flac encoding, etc.


That is what I thought, ok, I'll try tonight.

The whole reason I am doing this as I said before was to be able to output the multichannel sound to a receiver via SPDIF which I can do with regular DVDs via ac3filter.

the HD-dvds are a little harder though. I am hoping if I can get these HD tracks down to good ol' AC3 or DTS I can mux them back together with the video and have it work via Media player classic or something.

The weird thing I ran into and haven't checked out yet is that the video .mpv elemental stream from the EVOdemux output plays with a pixalated green screen where the original plays just fine.

Has anyone run into this? Will it be find when I mux them back together?

[nautilus7]

Demuxing the video with evodemux doesn't always work.
The sasfest way for that is to use haali media splitter --> haali matroska muxer in graphedit, but it's not the right thread to discuss it.
Use search to find what you need. There are plenty of posts around.

[nautilus7]

@ madshi

Features request: I think it would be nice to add an option like press that key to cancel encoding. Also a message telling encoding took xxx seconds would be useful. These, if you have time and don't know how to spend it. :D

Finally, i updated the libaften.dll with the latest available (R703) and everything is working fine.

[madshi]

Only Nero 7 works and the decoders are included.
Just for completeness sake: Nero 7 is needed plus the Nero HD DVD / Blu-Ray plugin.

[madshi]

Features request: I think it would be nice to add an option like press that key to cancel encoding. Also a message telling encoding took xxx seconds would be useful. These, if you have time and don't know how to spend it. :D
As with every command line tool, you can press Ctrl+C to cancel encoding. That's Windows default behaviour for command line tools. Giving out encoding time is easy to add.

Finally, i updated the libaften.dll with the latest available (R703) and everything is working fine.
Good to know. According to jruggle the latest build should sound every so slightly better than the older builds. So I'll update to the latest libAften build with the next eac3to build.

[nautilus7]

As with every command line tool, you can press Ctrl+C to cancel encoding. That's Windows default behaviour for command line tools.Didn't know that. Thanks!

According to jruggle the latest build should sound every so slightly better than the older builds.
That's why i swapped it in the first place. :p Btw, i used the SSE3 enabled cause i have a core 2 duo.

[idbirch2]

Hi madshi, here's (http://forum.doom9.org/showthread.php?p=1072182#post1072182) the post I'm following up on, looks like quotes can't be quoted.

Currently eac3to's Aften encoder only supports 1, 2 and 6 channels. I'm not sure if Aften itself can encode 3/1 channels. Will have to check that.

It works for me with the sample you sent me. Maybe you can make a new sample which reproduces this new problem?

I have split the dtshd file into 10mb chunks with the hex editor you mentioned earlier but I still can't get it to work. Here's (http://rapidshare.com/files/74662017/esh001.dtshd.html) a 10MB chunk, if I try and convert this to wav, I just get:

F:\>"C:\Program Files\Audio\eAC3to\eac3to.exe" "H:\es\esh001.dtshd" "F:\esh1.wa
v"
DTS Master Audio, 3/1 channels, 24 bits, 48khz
Decoding with DirectShow (Sonic Audio Decoder)...
DirectShow reports 4 channels, 24 bits, 48khz
Writing WAV...
Creating/writing file "F:\esh1.wav"...
----------------------------------------------------------- How long should it take to do a 10MB chunk? So far I've been waiting 10 mins and it's not got any further.

[shambles]

i have a problem with a dts-hd master audio track.

DTS Master Audio, 5.1 channels, 24 bits, 48khz
Decoding with DirectShow (Sonic Audio Decoder)...
DirectShow reports 5.1 channels, 24 bits, 48khz
Encoding FLAC...
Creating/writing file "2.flac"...
This track is not clean. Processing aborted.
Please clean the track with delaycut and then retry eac3to.

it encodes fine until it gets to the very end. delaycut of course doesn't work with dts-hd tracks, but i tried trimming off a little bit from the end with HxD and saving the rest to a new file but then i get "The format of the source file could not be detected." i also tried running it through tsremux first and then demuxing but i get the same 'track is not clean' error.

when the processing is aborted, the output file is also deleted. is there any switch that would keep the output file? seeing as the error only occurs at the very end, i'm not sure there would be any actual audio data missing..

or better yet would be to fix the original file.. but how could that be done?

[nautilus7]

I had the exact same problem, but with an e-ac3 track. Delaycut couldn't correct it, so i used it just to get the frame that had the problem. Then i corrected it in a hex editor.
So, if you somehow find where exactly is the problem, you might be able to correct it in a hex editor. Can you upload the final part of the track? Try to include the erroneous part.

EDIT: That's a very good reason why madshi should update delaycut with dts-hd support.

[shambles]

with the split function in file tools in HxD i was able to produce a working dtshd file, and i believe i split it from exactly the spot where the bad frame is, seeing as now eac3to gets to the end fine with even the last frame being fully decoded (when trying to split it before, i got it wrong and eac3to would drop the last frame as it was incomplete)

http://rapidshare.com/files/74760031/broken.dtshd.html sample of the end of the track beginning with what i believe is the bad frame

[nautilus7]

Maybe i'm tottaly wrong, but it think the only problem with your sample is that the last frame isn't complete.
You can remove this frame by deleting the last 1455 (5AF in hex) bytes of the track. The sample is the end of the track, right?

@ madshi

Can you tell me the dts-hd frame structure/size please?

[shambles]

yes it's the end of the track. but if the last frame is incomplete, eac3to just says this "The last DTS frame is incomplete and thus gets skipped." and doesn't give any errors..

[G_M_C]

Am trying to get AC3 decoding to work with libav (doing a NTSC2PAL speedup, and demuxing into separate WAV's), but the -libav switch doesnt seem to work; EAC3To keeps wanting to use Nero, despite the swich).

(/me is not a fan of nero ....)

[nautilus7]

libav is only for e-ac3 and mpl/trueHD decoding.

[kakomu]

I've extracted an EAC3 track from a Matroska sample source using command line and mkvextract. I've also reinstalled Nero 7.9. When I used EAC3to with the EAC3 file, I receive this message when I try to encode into AC3:

E-AC3, 5.1 channels, 0:01:01, 640kbit/s, 48khz, dialnorm: -27dB
Removing dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
Disabling DRC for Nero (E-)AC3 decoding...

When I try to encode into Wav, I get:

E-AC3, 5.1 channels, 0:01:01, 640kbit/s, 48khz, dialnorm: -27dB
Removing dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
Disabling DRC for Nero (E-)AC3 decoding...
The WAV writer didn't receive the format information.

I must be missing something here. Is there some sort of guide on installation or am I SOL?

[nautilus7]

Do you have the HD DVD/Blu-ray plug-in for Nero?

[saint-francis]

I'm getting this error with every attempt to use eas3to now:

G:\mymovie>"C:\eac3to.exe" "G:\mymovie\featureXX.ddp" "G:\mymovie\featureXX.ac3" -640 -nero
source file "G:\mymovie\featureXX.ddp" not found.

I can assure you that the files are indeed there.

[nautilus7]

I think it has to do with the "" you put in the paths. Remove them to see what happens.

The -640 and -nero switches you specify are not needed because these are the parameters that eac3to uses by default.

[saint-francis]

This is just what eac3togui has given to eac3to.exe. I'll give it a shot from the command line though.

[kakomu]

Do you have the HD DVD/Blu-ray plug-in for Nero?

No, I do not. I take it that's required?

Is there a list of required software that is necessary to run this program successfully?

[nautilus7]

This is just what eac3togui has given to eac3to.exe. I'll give it a shot from the command line though.
You didn't tell it. Don't know cause i don't use the gui. Sorry.

[nautilus7]

No, I do not. I take it that's required?

Is there a list of required software that is necessary to run this program successfully?
Yes, it's required.

Have a look at the 1st post of this thread. I believe you 'll find what you need.

[The_Keymaker]

Hello fellow forum members,

I am in the process of upgrading EAC3toGUI to reflect the options in Madshi's latest version of eac3to.

In the meantime I am releasing an interim upgrade, v1.45. This does not have all the latest option switches, but it WILL let you type these option switches into the Command Line Preview box so they can be executed.

Remember to use the "Settings" menu to show EAC3toGUI where eac3to is located.

LINK: http://www.sendspace.com/file/9k2byp

I should have a full upgrade ready soon.

Thanks!
The_Keynmaker

[Thunderbolt8]

have some trouble with some audio pcm (?) files from the bruce springsteen - live at hammersmith odeon london '75 DVD. the DVD has 2 audio tracks, a 5.1 DTS track and a 2.0 LPCM track (the track of my choice)
this dvd is originally a video concert dvd, but dvd audio extractor lets me extract/convert the audio tracks to flac, wav or demux (pcm). I can choose the bitrate manually here in dvd audio extractor, but I dont know whether the original audio is 16- or 24-bit. thats why I tried the "demux" option and wanted to put those pcm files it gave me in eac3to, hoping it would run the 16 & 24-bit detection test and in the end remove possibly zero bites, if there were any.
but eac3to cant detect those types here so I made a sample. the 2 files are the 1st and 2nd track from that disc. the 1st (intro) being only ~15 seconds and from the 2nd track I cut a 10mb sample. the eac3to detection stops right at the beginning for the 2nd track, I guess thats because the track begins at once, without any audible break after the 1st track if they are in the playlist in that order.

http://www.sendspace.com/file/1bcba7

[saint-francis]

I'm getting this error with every attempt to use eas3to now:

G:\mymovie>"C:\eac3to.exe" "G:\mymovie\featureXX.ddp" "G:\mymovie\featureXX.ac3" -640 -nero
source file "G:\mymovie\featureXX.ddp" not found.

I can assure you that the files are indeed there.

The exact commands that the GUI gives work fine when I don't use the GUI. An issue with the GUI?

[madshi]

Here's a 10MB chunk, if I try and convert this to wav, I just get:

F:\>"C:\Program Files\Audio\eAC3to\eac3to.exe" "H:\es\esh001.dtshd" "F:\esh1.wa
v"
DTS Master Audio, 3/1 channels, 24 bits, 48khz
Decoding with DirectShow (Sonic Audio Decoder)...
DirectShow reports 4 channels, 24 bits, 48khz
Writing WAV...
Creating/writing file "F:\esh1.wav"...
----------------------------------------------------------- How long should it take to do a 10MB chunk? So far I've been waiting 10 mins and it's not got any further.
This sample decodes just fine for me (decoding takes about 5 seconds). Maybe your Sonic installation is not good? Try uninstalling and reinstalling Sonic.

[madshi]

with the split function in file tools in HxD i was able to produce a working dtshd file, and i believe i split it from exactly the spot where the bad frame is, seeing as now eac3to gets to the end fine with even the last frame being fully decoded (when trying to split it before, i got it wrong and eac3to would drop the last frame as it was incomplete)

http://rapidshare.com/files/74760031/broken.dtshd.html sample of the end of the track beginning with what i believe is the bad frame
Thanks. That was kind of a funny sample. The broken frame had one core block and two DTS-HD blocks. Basically the DTS-HD block was repeated twice. Very strange. The next build of eac3to will handle this situation. The Sonic decoder doesn't seem to care about such double DTS-HD blocks, so eac3to will allow that, too.

[madshi]

@ madshi

Can you tell me the dts-hd frame structure/size please?
Basically there's always a core frame and then a DTS-HD frame. The DTS-HD frame begins with "64 58 20 25". Directly after this DTS-HD "sign" there's the format indicator (High Resolution or Master Audio) and then the length of the DTS-HD block. The length is constant for High Resolution tracks. But for Master Audio every DTS-HD block can have a different length.

[madshi]

I'm getting this error with every attempt to use eas3to now:

G:\mymovie>"C:\eac3to.exe" "G:\mymovie\featureXX.ddp" "G:\mymovie\featureXX.ac3" -640 -nero
source file "G:\mymovie\featureXX.ddp" not found.

I can assure you that the files are indeed there.
Can't reproduce the problem here. Seems to work for me.

[madshi]

have some trouble with some audio pcm (?) files from the bruce springsteen - live at hammersmith odeon london '75 DVD. the DVD has 2 audio tracks, a 5.1 DTS track and a 2.0 LPCM track (the track of my choice)
this dvd is originally a video concert dvd, but dvd audio extractor lets me extract/convert the audio tracks to flac, wav or demux (pcm). I can choose the bitrate manually here in dvd audio extractor, but I dont know whether the original audio is 16- or 24-bit. thats why I tried the "demux" option and wanted to put those pcm files it gave me in eac3to, hoping it would run the 16 & 24-bit detection test and in the end remove possibly zero bites, if there were any.
but eac3to cant detect those types here so I made a sample. the 2 files are the 1st and 2nd track from that disc. the 1st (intro) being only ~15 seconds and from the 2nd track I cut a 10mb sample. the eac3to detection stops right at the beginning for the 2nd track, I guess thats because the track begins at once, without any audible break after the 1st track if they are in the playlist in that order.

http://www.sendspace.com/file/1bcba7
I'm not sure what these tracks are. They seem to be 16bit signed PCM. But converting them to either big or little endian both results in garbage. Maybe the demuxing didn't work correctly?

[nautilus7]

Basically there's always a core frame and then a DTS-HD frame. The DTS-HD frame begins with "64 58 20 25". Directly after this DTS-HD "sign" there's the format indicator (High Resolution or Master Audio) and then the length of the DTS-HD block. The length is constant for High Resolution tracks. But for Master Audio every DTS-HD block can have a different length.
The dts core is always 2013 bytes i guess.

I had a look at the sample and i saw that one frame was bigger than the others, but thought it was ok because it's a Master Audio track.
In that sample the last frame was ok? It's too sort i think.

[Thunderbolt8]

I'm not sure what these tracks are. They seem to be 16bit signed PCM. But converting them to either big or little endian both results in garbage. Maybe the demuxing didn't work correctly?
the option I selected was direct stream copy and he gave me these .pcm files for that. I also could have chosen output as waves, but eac3to wont alter bitdepth for these at flac conversion.

[madshi]

The dts core is always 2013 bytes i guess.
The DTS-HD core is usually 2012 bytes. Don't know why it's one byte shorter than usual...

[madshi]

the option I selected was direct stream copy and he gave me these .pcm files for that. I also could have chosen output as waves, but eac3to wont alter bitdepth for these at flac conversion.
What do you mean with "alter"? You mean it doesn't detect if it's 16bit packaged as 24bit (and then strip the zero bytes)? You can force eac3to to do that by using the undocumented "-check16bit" option.

[Thunderbolt8]

yes, thats what I meant. when I choose output as wave then, will this also be lossless as the direct steam copy to .pcm is?

another problem, I just demuxed the eac3 track from dazed & confused HD DVD and wanted to process it with eac3to for dialnorm removal, but it told me the track is not clean, I should use delaycut first. but when I load the track into delaycut (v1212) then it only says file type: unknown and when I press process it only says "finished with error!" :S and I get a 0 byte file :S

[nautilus7]

another problem, I just demuxed the eac3 track from dazed & confused HD DVD and wanted to process it with eac3to for dialnorm removal, but it told me the track is not clean, I should use delaycut first. but when I load the track into delaycut (v1212) then it only says file type: unknown and when I press process it only says "finished with error!" :S and I get a 0 byte file :S
This is the latest version made by jsoto. Then madshi update delaycut to v1.3.0.0 with e-ac3 support. It's in the delaycut thread.

[Thunderbolt8]

ok thanks

btw. when I want to maintain the exact delay, which option do I have to choose, silence or fix?
evodemux told me +23m for that track, this needs to stay :S

I choose 'silence' and now the duration is .373ms, while it was .560 before
with 'fix' its .379/.378

do I have to take those differences to .560 into account as general delay at the beginning of the track or is this difference divided up at different places in the whole track?

[nautilus7]

madshi, i found a bug which i think we had discovered some time ago and you said you fixed it.

If i type

eac3to input.ac3 output/ac3 -a delay

it just says writing file and the process finishes very fast (obviously no encoding takes place) but no delay is applied.

If i type

eac3to input.ac3 output.ac3 -a delay -same bitrate as input

it does encode and probably applies the delay (didn't let it finish).

If i remember correctly, before you "fixed" it, eac3to didn't encode whether you specified the same bitrate or not.

The proper behaviour for the 1st case is just to apply the dealy, right? Maybe you should add a switch like -no encode or something and tell eac3to always to encode unless it's activated.

[nautilus7]

ok thanks

btw. when I want to maintain the exact delay, which option do I have to choose, silence or fix?
evodemux told me +23m for that track, this needs to stay :S

I choose 'silence' and now the duration is .373ms, while it was .560 before
with 'fix' its .379/.378

do I have to take those differences to .560 into account as general delay at the beginning of the track or is this difference divided up at different places in the whole track?

fix tries to correct the crc errors.
silence replaces the erroneous frame with a silent one.

If you search a few pages back in the delaycut thread you 'll find the explanation of these options.

But, you always can open the file in a hex editor, and try to correct the frames manually if delaycut doesn't work.
I had a little different problem and here's (http://forum.doom9.org/showpost.php?p=1069064&postcount=1601) how it was fixed.

[madshi]

yes, thats what I meant. when I choose output as wave then, will this also be lossless as the direct steam copy to .pcm is?
I don't know what the other program does. WAV handling by eac3to is lossless, of course.

[madshi]

madshi, i found a bug which i think we had discovered some time ago and you said you fixed it.

If i type

eac3to input.ac3 output/ac3 -a delay

it just says writing file and the process finishes very fast (obviously no encoding takes place) but no delay is applied.
It's not a bug, it's a feature... :D The reason for this behaviour is that applying a delay to AC3 tracks should be made by using delaycut. What you want eac3to to do would harm audio quality cause the AC3 track would need to be decoded and reencoded again. delaycut can apply delays without decoding/reencoding which is MUCH better.

A future version of eac3to will probably also be able to apply delays to AC3 and DTS tracks without having to reencode.

[nautilus7]

Yes, that is what i was trying to tell. To apply the delay like delaycut is doing when input.ac3 output.ac3 -delay is used.

[Thunderbolt8]

fix tries to correct the crc errors.
silence replaces the erroneous frame with a silent one.

If you search a few pages back in the delaycut thread you 'll find the explanation of these options.

But, you always can open the file in a hex editor, and try to correct the frames manually if delaycut doesn't work.
I had a little different problem and here's (http://forum.doom9.org/showpost.php?p=1069064&postcount=1601) how it was fixed.
sorry, but thats too complicated for me :S I just dont get how he gets the frame numbers Ill have to edit with hex editor and to replace which which values

this is what I got from the delaycut log file (chosen 'fix' option):

[Input info]
Bitrate=1536
Actual rate=1536.000000
Sampling Frec=48000
TotalFrames=1150980
Bytesperframe=1024.0000
Filesize=1178603520
FrameDuration= 5.3333
Framespersecond=187.5000
Duration=01:42:18.560
Channels mode=3/2: L+C+R+SL+SR
LFE=LFE: Present
[Target info]
StartFrame=0
EndFrame=1150979
NotFixedDelay= 0.0000
Duration=01:42:18.560
====== PROCESSING LOG ======================
Time 00:00:00.490; Frame#= 93. Crc error FIXED: read = ECA9; calculated=1E46
Time 00:00:00.496; Frame#= 94. Unsynchronized frame...SKIPPED 3072 bytes. Found new synch word
Time 00:00:00.501; Frame#= 95. Crc error FIXED: read = E05D; calculated=C20D
Time 00:00:00.506; Frame#= 96. Unsynchronized frame...SKIPPED 1024 bytes. Found new synch word
Time 00:00:00.512; Frame#= 97. Crc error FIXED: read = 5E0F; calculated=5E18
Time 00:00:00.517; Frame#= 98. Unsynchronized frame...SKIPPED 1024 bytes. Found new synch word
Time 00:00:00.522; Frame#= 99. Crc error FIXED: read = CFBE; calculated=43B4
Time 00:00:00.528; Frame#= 100. Unsynchronized frame...SKIPPED 25087 bytes. Found new synch word
Time 00:00:00.533; Frame#= 101. Some basic parameters changed between Frame #1 and this frame
Time 00:00:00.533; Frame#= 101. Crc error FIXED: read = F00F; calculated=871C
Time 00:00:00.538; Frame#= 102. Unsynchronized frame...SKIPPED 367 bytes. Found new synch word
Time 00:00:00.544; Frame#= 103. Some basic parameters changed between Frame #101 and this frame
Time 00:03:41.024; Frame#= 41443. Crc error FIXED: read = 8BA0; calculated=6C80
Time 00:03:41.029; Frame#= 41444. Unsynchronized frame...SKIPPED 4096 bytes. Found new synch word
Number of written frames = 1150945
Number of Errors= 14

duration afterwards (input file info): 1:42:18.379

[nautilus7]

What option did you use in delaycut (fix, silence, ignore, etc)?

Anyway... You can upload a sample (or 2) that contains these frames. I think a sample from the start to the 4th minute would not be too big.

EDIT: I just culculated it... The first 44 MB should contain all erroneous frames. Upload them please.

[Thunderbolt8]

I chose fix.
from which track should I upload those 1st part, from the originally demuxed one or from that one I ran through delaycut + fix already?

[nautilus7]

The original, of course. :p

Don't bet your money that i would fix though. :D

[Thunderbolt8]

ok, here it is
http://www.sendspace.com/file/9gh0kf

[ACrowley]

Hello fellow forum members,

I am in the process of upgrading EAC3toGUI to reflect the options in Madshi's latest version of eac3to.

In the meantime I am releasing an interim upgrade, v1.45. This does not have all the latest option switches, but it WILL let you type these option switches into the Command Line Preview box so they can be executed.

Remember to use the "Settings" menu to show EAC3toGUI where eac3to is located.

LINK: http://www.sendspace.com/file/9k2byp

I should have a full upgrade ready soon.

Thanks!
The_Keynmaker

THX
Nice that you implement my small suggesteion with the additional Parameters in preview Box

But, maybe you can make it more obvious.
A small "custom commandline Parameters Window" where you can type in -switches ,would be nice

[nautilus7]

ok, here it is
http://www.sendspace.com/file/9gh0kf

I spent some time on your sample. These frames (with the errors) are completely wrong, i think. Trying some stuff right now, maybe i'll upload a fixed one later so ea3to can process it at least.

[Thunderbolt8]

would be nice, thanks!

[nautilus7]

would be nice, thanks!
Here (http://rapidshare.com/files/75625426/fixed.eac3.html) you are. I believe i have corrected the frames so they are not always skipped in delaycut.

Replace the first 51.200 KB of your original eac3 track with fixed part i uploaded (do not delete the original track of course - this may still not work). Then pass the eac3 track through delaycut, so it can fix the crc errors. eac3to will process the track now, i think.

Note: the frames i fixed may not contain valid audio information. They are just valid e-ac3 frames now. That's all.

[Thunderbolt8]

thanks, will try!

[Thunderbolt8]

what do I have to do again in case the PTS value of the video was higher than the value of the 1st Audio PTS?

e.g. here for casablanca:

Opening file Feature_1.EVO
Reading...
File size: 6886 Mbytes.
VOB number 4 contains 1 video , 5 audio and 4 subpicture streams.
PTM of first video frame = 00000D91
PTM of last video frame = 105CFFB1
Duration = 0:50:50.297
VC-1 video stream 0 found!
First PTS = 00000D91
Dolby Digital Plus audio stream 0 found!
First PTS = 00000A67 (+47721849ms)

+47721849 can hardly be the correct delay, so in this case I have to subtract the audio value from the video value?

D91 - A67 = 810 : 90 = 9
so the correct delay value will be -9ms ?

[madshi]

-9ms is the correct value. EvoDemux doesn't show negative delays correctly.

[nautilus7]

D91 - A67 = 810 : 80 = 9
so the correct delay value will be -9ms ?
You should divide by 90, but the result is right. :D

[nautilus7]

@ madshi

When i type eac3to filename.thd to get the file info, the duration of the TrueHD track isn't shown, while it does for other extensions.

[madshi]

@ madshi

When i type eac3to filename.thd to get the file info, the duration of the TrueHD track isn't shown, while it does for other extensions.
TrueHD tracks have a variable bitrate because of which it's impossible for eac3to to retrieve the correct runtime - except by scanning the whole file (which eac3to doesn't do for performance reasons). Same thing with DTS-HD Master Audio tracks.

[nautilus7]

I thought this was a possible explanation. Thanks!

[Thunderbolt8]

You should divide by 90, but the result is right. :D
typo -.-
but the result was already right, according to what the correct delay is supposed to be :P

[Phantomas_X]

eac3to.exe "K:\FEATURE_1_MERGED.DD+.stream.00.mpa" "K:\Trans.eng.wavs"

E-AC3, 5.1 channels, 2:23:23, 1536kbit/s, 48khz, dialnorm: -27dB
Removing dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
Disabling DRC for Nero (E-)AC3 decoding...
DirectShow reports 2.0 channels, 24 bits, 48khz
Writing WAVs...
The channel modder was started with incorrect parameters.
Creating/writing file "K:\Trans.eng.L.wav"...

Creating/writing file "K:\Trans.eng.R.wav"...

The channel modder didn't receive the format information.
The channel modder didn't receive the format information.
The channel modder didn't receive the format information.

I have Nero 7, I think with Nero HD. I looked at nero.com and they writing that the HD Plugin is included in Nero7... so on, how can i convert it ?

[madshi]

@Phantomas_X, the HD DVD/Blu-Ray plugin is not included in Nero by default. You need to buy it separately, as far as I'm aware. If you're not sure about this, I'd suggest that you contact Nero support. Maybe in your country there are combined packages for Nero + plugins available. I think in my country there aren't.

[Phantomas_X]

yup, thats right is not included...

so on, there is no other way, i think, otherwise there would be a update, right ? :)

[madshi]

so on, there is no other way, i think, otherwise there would be a update, right ? :)
No other way for what? For decoding DD+? Sure, you can use the "-libav" switch. But the quality is rather bad right now. The open source libav DD+ decoder is a work in progress. If you want good quality decoding Nero is the only choice right now.

[G_M_C]

@Phantomas_X, the HD DVD/Blu-Ray plugin is not included in Nero by default. You need to buy it separately, as far as I'm aware. If you're not sure about this, I'd suggest that you contact Nero support. Maybe in your country there are combined packages for Nero + plugins available. I think in my country there aren't.

You need to buy any part of Nero, and that's why i'm all for alternatives. But EAC3To only supports Nero for "normal" AC3. Could you make it so that the -libav switch works for regular DD too ?

Anyway; The test i did on the DD encoding part of EAC3To didnt reveal the problem i had with DVDLab; It is probably a problem with DVDLAb. Encoding separately with Aften (Gui) has the advantage that you have more control/options, but that is logical. I think i'll keep doining it that way because of that (mainly the pre-filtering options add some quality to the AC3 file).

[Phantomas_X]

ahh, alright, working....

but now

This track is not clean. Processing aborted.
Please clean the track with delaycut and then retry eac3to.

when i try to repair/clean up with delaycut, it reads it as an mpeg 1 audio file. After rename to .ac3 it will fix it, but only the half file size and only silence.

one other thing, i tried to play the ac3 wit nero showtime, after this the original e-ac3, and it worked and show "Digital Plus" in Display... Are there internal Decoder in Showtime ?

[scarbrtj]

Having trouble with the eac3 --> ac3 conversion process for The Matrix. This is my first attempt at this so please bear with me... I have all the right decoders, etc.

Basically what is happening is that my converted .ac3 file is "jumpy". It sounds like it is flipping back and forth in the timeline or something, every 3 to 4 seconds. Weird ,I know. Otherwise, it sounds great and gets passed through S/PDIF great! The nero filter is set for no downmixing... I don't know what else parameter I can check.

Thanks, especially Madshi, hopefully someone will understand my problem and offer a solution.

[Mtz]

I should have a full upgrade ready soon.
Using the delay buttons and going up to 2000ms, the Gui showed 2.000ms which is not correct for eac3to. Correct is 2000ms.
I hope to include the dts extension in Save As (and have the Save button instead of Open) and the settings to remember the eac3to.exe location.

enjoy,
Mtz

[madshi]

Could you make it so that the -libav switch works for regular DD too ?
I think so. But I don't really recommend libav for DD decoding because:

(1) The libav DD decoder is not based on the reference decoder. So quality might be ever so slightly lower compared to the reference decoder (or not, I don't really know).
(2) The libav DD decoder always only outputs 16 bit while the reference decoder outputs 24 bit.

Anyway; The test i did on the DD encoding part of EAC3To didnt reveal the problem i had with DVDLab; It is probably a problem with DVDLAb.
Good to know. Thanks.

[madshi]

when i try to repair/clean up with delaycut, it reads it as an mpeg 1 audio file. After rename to .ac3 it will fix it, but only the half file size and only silence.
Have you used the latest delaycut v1.3? See delaycut thread here in doom9.

one other thing, i tried to play the ac3 wit nero showtime, after this the original e-ac3, and it worked and show "Digital Plus" in Display... Are there internal Decoder in Showtime ?
Obviously if you buy the Nero HD DVD/Blu-Ray plugin, Nero installs a decoder for E-AC3 on your PC. This decoder is used by eac3to and by Showtime.

[madshi]

Basically what is happening is that my converted .ac3 file is "jumpy". It sounds like it is flipping back and forth in the timeline or something, every 3 to 4 seconds.
That's quite strange. Is the runtime of the AC3 correct? Please try converting to another format, e.g. FLAC, DTS or WAV. This way you can find out where the problem comes from decoding E-AC3 or from encoding to AC3.

[G_M_C]

I think so. But I don't really recommend libav for DD decoding because:

(1) The libav DD decoder is not based on the reference decoder. So quality might be ever so slightly lower compared to the reference decoder (or not, I don't really know).
(2) The libav DD decoder always only outputs 16 bit while the reference decoder outputs 24 bit.


Isnt LibAV based on/part of FFDShow_Tryouts (i really am no expert, so its only a stab in the dark for me here ;) ) ?

I can set 24 bits on that (and 16 or 32).

[madshi]

Isnt LibAV based on/part of FFDShow_Tryouts (i really am no expert, so its only a stab in the dark for me here ;) ) ?

I can set 24 bits on that (and 16 or 32).
ffdshow supports various bitdepths, but most libav decoders don't. ffdshow makes use of several decoder packages. libav is one of them. Of course ffdshow can take the libav data and convert it to 24bit. But that won't change the fact that the real decoded data is only 16bit to begin with when using libav. Well, theoretically it's possible that ffdshow have modified libav to allow more than 16bit decoding output. Don't know if they did that, but I rather think not.

[G_M_C]

Ahh, dont know if they have adapted the decoder(s) either.

But the reason i ask about this is that the Nero package is such bloatware these days. For instance the premium-version is such bloatware, it can hog resources and is virtually impossible to get rid off.

I'd like to prevent problems by installing separate open-source tools, that give only the features i need, without causing all that mess.

[madshi]

But the reason i ask about this is that the Nero package is such bloatware these days. For instance the premium-version is such bloatware, it can hog resources and is virtually impossible to get rid off.

I'd like to prevent problems by installing separate open-source tools, that give only the features i need, without causing all that mess.
You can limit the Nero installation to Nero Showtime. Of course you can also choose open source software instead. But honestly: Do you want the best quality or not? There's not really a usable alternative to Nero's E-AC3 decoder available right now, as far as I can see. So I don't really see a way around Nero. I do expect the open source E-AC3 decoder to mature over time. But personally, I'll probably continue to use Nero's decoder, anyway. Simply because I think that the official Dolby reference decoder should give me the best possible quality.

[Penecho]

Hi, i got the following message, when i try to convert e-ac3 to ac3:


E-AC3, 5.1 channels, 1:25:12, 1536kbit/s, 48khz, dialnorm: -27dB
Removing dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
Disabling DRC for Nero (E-)AC3 decoding...
This track is not clean. Processing aborted.
Please clean the track with delaycut and then retry eac3to.

What do i have to do now?

Dolby Digital Plus audio stream 1 found!
First PTS = 11DC2AFF (+3329312ms)

What delay is that?



Cu

[nautilus7]

Wtf is wrong with the e-ac3 tracks? Lot of people have the same problem!!!

Hi, i got the following message, when i try to convert e-ac3 to ac3:


E-AC3, 5.1 channels, 1:25:12, 1536kbit/s, 48khz, dialnorm: -27dB
Removing dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
Disabling DRC for Nero (E-)AC3 decoding...
This track is not clean. Processing aborted.
Please clean the track with delaycut and then retry eac3to.

What do i have to do now?

Dolby Digital Plus audio stream 1 found!
First PTS = 11DC2AFF (+3329312ms)

What delay is that?



Cu
Use latest delaycut (v1.3.0.0) to find out what/where is the problem. Use ignore option or what ever you want (to start with) and post the log here afterwards. If there are only crc errors, use fix to correct them and you 'll be ok.

[scarbrtj]

That's quite strange. Is the runtime of the AC3 correct? Please try converting to another format, e.g. FLAC, DTS or WAV. This way you can find out where the problem comes from decoding E-AC3 or from encoding to AC3.

Madshi...can you give me commandline syntax for converting my trueHD file to DTS or WAV?

One other thing I should note. When forcing the Sonic decoder, I get a "good" AC3 file...but... it only decodes with the Sonic decoder (plays back well, though, via that decoder ONLY). I use the NVIDIA audio decoder for almost all my AC3 playback needs as it passes through SPDIF well. When I play back these eac3 files converted to ac3 via Sonic, they do not get decoded by NVIDIA (can't even force it)...only playable via Sonic Audio Decoders.

Oh, one other thing. It doesn't seem to matter whether I take the DD+ stream or the "MLP" (trueHD?) stream for decoding. Still have the "weird" ac3 file generation via Nero.

[madshi]

Wtf is wrong with the e-ac3 tracks? Lot of people have the same problem!!!
Don't know. Either EvoDemux has a bug which produces incorrect E-AC3 tracks. Or the ripping was faulty. Can't really believe that the HD DVDs itself are that faulty. Personally, I've not had any such trouble yet.

[madshi]

Madshi...can you give me commandline syntax for converting my trueHD file to DTS or WAV?
It's as simple as can be:

eac3to source.thd dest.dts
eac3to source.thd dest.wav

One other thing I should note. When forcing the Sonic decoder, I get a "good" AC3 file...but... it only decodes with the Sonic decoder (plays back well, though, via that decoder ONLY). I use the NVIDIA audio decoder for almost all my AC3 playback needs as it passes through SPDIF well. When I play back these eac3 files converted to ac3 via Sonic, they do not get decoded by NVIDIA (can't even force it)...only playable via Sonic Audio Decoders.

Oh, one other thing. It doesn't seem to matter whether I take the DD+ stream or the "MLP" (trueHD?) stream for decoding. Still have the "weird" ac3 file generation via Nero.
Maybe the NVidia audio decoder doesn't like 640kbps AC3 files? Try feeding eac3to with an additional "-448" parameter ("eac3to sourcefile dest.ac3 -448"). That will produce an AC3 file with 448kbps bitrate instead of the default 640kbps. A compliant AC3 decoder should be able to decode 640kbps, though!

[Penecho]

Wtf is wrong with the e-ac3 tracks? Lot of people have the same problem!!!


Use latest delaycut (v1.3.0.0) to find out what/where is the problem. Use ignore option or what ever you want (to start with) and post the log here afterwards. If there are only crc errors, use fix to correct them and you 'll be ok.

It tells me too many errors, stop logging...

and i cant change the options, what to do with crc-errors...


[Input info]
Bitrate=128
Actual rate=128.000000
Sampling Frec=32000
TotalFrames=20449685
Bytesperframe= 48.0000
Filesize=981584864
FrameDuration= 3.0000
Framespersecond=333.3333
Duration=17:02:29.054
Channels mode=Stereo
LFE=LFE: Not present
[Target info]
StartFrame=0
EndFrame=20449684
NotFixedDelay= 0.0000
Duration=17:02:29.055
====== PROCESSING LOG ======================
Time 00:00:00.000; Frame#= 1. Unsynchronized frame...SKIPPED 6 bytes. Found new synch word
Time 00:00:00.006; Frame#= 3. Unsynchronized frame...SKIPPED 306 bytes. Found new synch word
Time 00:00:00.009; Frame#= 4. Uncomplete frame...SKIPPED
Time 00:00:00.012; Frame#= 5. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.015; Frame#= 6. Uncomplete frame...SKIPPED
Time 00:00:00.018; Frame#= 7. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.021; Frame#= 8. Uncomplete frame...SKIPPED
Time 00:00:00.024; Frame#= 9. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.027; Frame#= 10. Uncomplete frame...SKIPPED
Time 00:00:00.030; Frame#= 11. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.033; Frame#= 12. Uncomplete frame...SKIPPED
Time 00:00:00.036; Frame#= 13. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.039; Frame#= 14. Uncomplete frame...SKIPPED
Time 00:00:00.042; Frame#= 15. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.045; Frame#= 16. Uncomplete frame...SKIPPED
Time 00:00:00.048; Frame#= 17. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.051; Frame#= 18. Uncomplete frame...SKIPPED
Time 00:00:00.054; Frame#= 19. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.057; Frame#= 20. Uncomplete frame...SKIPPED
Time 00:00:00.060; Frame#= 21. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.063; Frame#= 22. Uncomplete frame...SKIPPED
Time 00:00:00.066; Frame#= 23. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.069; Frame#= 24. Uncomplete frame...SKIPPED
Time 00:00:00.072; Frame#= 25. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.075; Frame#= 26. Uncomplete frame...SKIPPED
Time 00:00:00.078; Frame#= 27. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.081; Frame#= 28. Uncomplete frame...SKIPPED
Time 00:00:00.084; Frame#= 29. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.087; Frame#= 30. Uncomplete frame...SKIPPED
Time 00:00:00.090; Frame#= 31. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.093; Frame#= 32. Uncomplete frame...SKIPPED
Time 00:00:00.096; Frame#= 33. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.099; Frame#= 34. Uncomplete frame...SKIPPED
Time 00:00:00.102; Frame#= 35. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.105; Frame#= 36. Uncomplete frame...SKIPPED
Time 00:00:00.108; Frame#= 37. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.111; Frame#= 38. Uncomplete frame...SKIPPED
Time 00:00:00.114; Frame#= 39. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.117; Frame#= 40. Uncomplete frame...SKIPPED
Time 00:00:00.120; Frame#= 41. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.123; Frame#= 42. Uncomplete frame...SKIPPED
Time 00:00:00.126; Frame#= 43. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.129; Frame#= 44. Uncomplete frame...SKIPPED
Time 00:00:00.132; Frame#= 45. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.135; Frame#= 46. Uncomplete frame...SKIPPED
Time 00:00:00.138; Frame#= 47. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.141; Frame#= 48. Uncomplete frame...SKIPPED
Time 00:00:00.144; Frame#= 49. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.147; Frame#= 50. Uncomplete frame...SKIPPED
Time 00:00:00.150; Frame#= 51. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.153; Frame#= 52. Uncomplete frame...SKIPPED
Time 00:00:00.156; Frame#= 53. Unsynchronized frame...REWINDED 0 bytes. Found new synch word
Time 00:00:00.159; Frame#= 54. Uncomplete frame...SKIPPED
Time 00:00:00.162; Frame#= 55. Unsynch
Too Many Errors. Stop Logging.


Cu


Penecho

[madshi]

It tells me too many errors, stop logging...

and i cant change the options, what to do with crc-errors...
Please put this E-AC3 track into the trash bin and rerip the disc (you own it, don't you?). Then use the latest EvoDemux version for demuxing the audio tracks. I think there must have been something wrong when you first ripped and demuxed. Maybe your PC is overclocked? Try to clock it down.

[Penecho]

Yea i own the disc. I used EvoDemux v0.627 b7



Cu



Penecho

[nautilus7]

Oh, that eac3 track is f@cked up!

Which program did you use for ripping. I 've head that if you use Anydvd to copy files though windows explorer, sometimes the files become corrupted. Don't know exactly.

[Penecho]

Do i have to merge the 2 *.evo files first perhaps?

I used anydvd and the function it has, rip dics to HDD...



Cu



Penecho

[nautilus7]

Do i have to merge the 2 *.evo files first perhaps?

I used anydvd and the function it has, rip dics to HDD...



Cu



Penecho
No, it's not necessary.

[madshi]

I used anydvd and the function it has, rip dics to HDD...
Please try doing the whole process again. Maybe it will work on 2nd try? If all else fails, I have working EVO demuxing code lying around here. I could cook up a little utility as an alternative to EvoDemux, in case EvoDemux does something wrong with this specific movie. Which movie is that, btw?

[madshi]

@Penecho, have you renamed the file to "*.eac3" or "*.dpp" before feeding it to delaycut? You have to do that. Furthermore please make sure you're using delaycut 1.3.0.0 and not an older version. The audio track properties your delaycut shows look VERY wrong. Maybe you named the file "*.ac3"? delaycut doesn't like that. You must use the correct file extension.

[Penecho]

Now it seems to work :) The name was still .mpa! But after the rerip, the Audio Track was also bigger (400MB) so I guess something went wrong the first time...


But he does not write the output file...

Looks like that now (he does something an writes ------- and than stops (when the --- reach the end of the line in my dos window) and has not written the outputfile):

E-AC3, 5.1 channels, 1:49:16, 1536kbit/s, 48khz, dialnorm: -27dB
Removing dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
Disabling DRC for Nero (E-)AC3 decoding...

C:\

Cu

[madshi]

he does something an writes ------- and than stops (when the --- reach the end of the line in my dos window) and has not written the outputfile
The "---" should reach the end of the line (all but the last column) and then conversion should be done. There's no output file at all? Are you sure that the Nero decoder is correctly installed?

[scarbrtj]

Madshi...

Everything seems to be working awesomely, now, with sonic 4.3. I think I was adding decoders, encoders, etc., like crazy...sometimes I have found it takes a PC reboot to make everything work nice. The 640K AC3 files play back fine via the NVIDIA audio decoder.

For now, FWIW, I am going from HD-DVD VC1 files, demuxing those into .mpv and .mpa (DD+ usually) files. I set 23.97 frame rate with vc1conv. Then I do vc12avi. Next, I convert my DD+ .mpa file to 640K AC3 file using Madshi's eac3to. Finally, I use the Haali muxer to make an .mkv file which plays back directly in WMP 11 or Media Center. I render the .avi file, through the AVI splitter, into the Haali muxer; I render the .ac3 file, through the AC3 parser (where did that come from?), into the Haali muxer, simultaneously.

This is of course a .mkv file. I rename it with an .mpg extension which does nothing except make the file "recognizable" by Media Center.

Hey Madshi... THANKS :D

[ACrowley]

@scarbrtj

I would not recommend SonicDecoder for EAC3 decoing!
You get full DRC+DialNorm +gain = very bad!

[Penecho]

The "---" should reach the end of the line (all but the last column) and then conversion should be done. There's no output file at all? Are you sure that the Nero decoder is correctly installed?

I would say yes, but how can i see that its correct installed? or can i use another encoder instead of nero?

or is my command line wrong (i am using the eac3to 1.4 gui)?

eac3to.exe stream.00.eac3 testoutput.ac3 -640
E-AC3, 5.1 channels, 1:49:16, 1536kbit/s, 48khz, dialnorm: -27dB
Removing dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
Disabling DRC for Nero (E-)AC3 decoding...
----------------------------------------------------------------------------

Cu

[scarbrtj]

@scarbrtj

I would not recommend SonicDecoder for EAC3 decoing!
You get full DRC+DialNorm +gain = very bad!


My problem is I could not get Nero to give me an AC3 file that sounded right. It skipped all over and I could not fix that. On the other hand, despite the issues you guys point out with using Sonic to decode DD+ --> AC3, the AC3 file sounds very good using an external high-end decoder played through big speakers!

Hey... what do you need to decode the .wav files eac3to generates if decoding to WAV? These do not play for me in WMP so that I can test them/listen to them.

[Penecho]

The "---" should reach the end of the line (all but the last column) and then conversion should be done. There's no output file at all? Are you sure that the Nero decoder is correctly installed?

It works with the sonic decoder...

So i guess you were right! How can i reinstall the nero decoder?

Edit: i reinstalled nero 7 premium, but it did not help..
I am also noticed, that there is no "writing outoputfile" when i use nero, with sonic, there is one....



Cu

[SvT]

How can i reinstall the nero decoder?

Cu

You need to buy the HD plugin serial seperately from Nero. You put in your serial at "ProductSetup/Licence/Add" and you're good to go !

No SPAM just for help:
http://www.nero.com/eng/bluray-hddvd-video-plugin.html

Goodluck.

[Penecho]

I think i had that already, coz if not it would look like this (i uninstalled my nero and tested):


E-AC3, 5.1 channels, 1:49:16, 1536kbit/s, 48khz, dialnorm: -27dB
Removing dialog normalization...
Decoding with DirectShow (Nero Audio Decoder 2)...
Getting "Nero Audio Decoder 2" instance failed.


Cu


Penecho

[SvT]

Decoding with DirectShow (Nero Audio Decoder 2)...
Getting "Nero Audio Decoder 2" instance failed.

Cu


Just look at ProductSetup/Licence if your plugin serial is there ! (Not just the Nero serial). If so you're installed.

(I think your error shows if the plugin isn't properly registered).

[The_Keymaker]

Fellow Forum members,

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

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

Changes and features in this version include:

- Removed comma in audio delay output for values over 999ms.
- Added "Other Options" tab to allow custom or non-integrated options.
- Added *.dts as valid extension in the destination file.
- Changed command line output box to READ only.

This is an interim revision and I hope to release a version with all current options integrated.

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

Please report any problems or feature requests.

Regards,
The_Keymaker

[Thunderbolt8]

can anyone explain what the current status with xport 1.00 and seamless branching movies regarding audio sync is? link with info here: http://forum.doom9.org/showthread.php?p=1069463#post1069463

I dont understand what we have to do now, or if we have to do something other than just demuxing with xport to keep audio in sync. will any joined part with LPCM audio now have +5ms delay and we have to fix that manually, either with altered fps rate or with that 'tail' procedure, that well have to cut off the endings and adjust both audio and video pts together at the end? or was this only an info he posted there and xport takes care of all that automatically?

[Nestorix]

Sorry to say, it is not working, tried all but cannot convert, not even automatic without adjusting something.....

[drmpeg]

can anyone explain what the current status with xport 1.00 and seamless branching movies regarding audio sync is? link with info here: http://forum.doom9.org/showthread.php?p=1069463#post1069463

I dont understand what we have to do now, or if we have to do something other than just demuxing with xport to keep audio in sync. will any joined part with LPCM audio now have +5ms delay and we have to fix that manually, either with altered fps rate or with that 'tail' procedure, that well have to cut off the endings and adjust both audio and video pts together at the end? or was this only an info he posted there and xport takes care of all that automatically?
xport doesn't do anything automatically. Chopping off audio samples would be much too heavy handed (and annoyingly audible).

I'm not sure what the best remedy is. If you're re-encoding, then an extra frame (strategically placed) when necessary would be the best solution. If you're just re-muxing, why? ;)

There is one positive note. The PES granularity of Dolby Lossless is 0.83 milliseconds. So for movies with Dolby Lossless audio, the A/V sync should be very close over many segments.

Ron

[Thunderbolt8]

Im remuxing, because I want the best possible quality at smallest possible size :P

which track count as dolby lossless, only TrueHD tracks, right? 0.83 is like nothing then. is this the same for DTS-HD (MA) tracks?? would it be possible for you to make a quick list of all tracks granularity for each .m2ts file, this should be helpful then when having movies with many parts to figure out how the delay has to be at the end. so for LPCM it was 5ms and for DD+ something like 32ms? trueHD 0.83 then? if you still need info about other audio tracks, then tell me, maybe I will be able to cut more samples, just to complete this.

LPCM: 5ms
DD+: 32ms
TrueHD: 0.83ms
DTS-HD MA: ?
DTS-HD HiRes: ? (same as MA?)

(I sent you both samples, MA with close encounters and DTS-HD HiRes with basic instict)


just assuming it is also 0.83 for DTS-HD MA for example, so when I remuxed close encounters with dts-hd converted to flac and this movie consists of 30 pieces, then the delay at the end would be ~25ms for the last part of the movie?
and in case of ratatouille with 31 pieces of LPCM the delay at the end would then be 155ms? and a 2 piece .m2ts movie with lpcm only ending up with 5ms, so this would be not worth correcting at all




btw. you mean positive delay, right (meaning we would have to slow down the fps rate of the video a little to make the video stream gradually longer) ?

btw² since you mentioned PES, do we have to use the -z option in xport now additionally for such files? (=e.g. xport -hz bla.m2ts 1 1 1) or still only -h ?

[Chumbo]

Im remuxing, because I want the best possible quality at smallest possible size :P

which track count as dolby lossless, only TrueHD tracks, right? 0.83 is like nothing then. is this the same for DTS-HD (MA) tracks?? would it be possible for you to make a quick list of all tracks granularity for each .m2ts file, this should be helpful then when having movies with many parts to figure out how the delay has to be at the end. so for LPCM it was 5ms and for DD+ something like 32ms? trueHD 0.83 then? if you still need info about other audio tracks, then tell me, maybe I will be able to cut more samples, just to complete this.

LPCM: 5ms
DD+: 32ms
TrueHD: 0.83ms
DTS-HD MA: ?
DTS-HD HiRes: ? (same as MA?)

(I sent you both samples, MA with close encounters and DTS-HD HiRes with basic instict)


just assuming it is also 0.83 for DTS-HD MA for example, so when I remuxed close encounters with dts-hd converted to flac and this movie consists of 30 pieces, then the delay at the end would be ~25ms for the last part of the movie?
and in case of ratatouille with 31 pieces of LPCM the delay at the end would then be 155ms? and a 2 piece .m2ts movie with lpcm only ending up with 5ms, so this would be not worth correcting at all




btw. you mean positive delay, right (meaning we would have to slow down the fps rate of the video a little to make the video stream gradually longer) ?
Humans don't notice a sync issue until the audio/video are off by 70ms or greater. Anything less is a non-issue, so I wouldn't waste so much time on it otherwise.

The real problem is when the small amounts per segment amount to a total offset that's greater than 70ms.

[Thunderbolt8]

Humans don't notice a sync issue until the audio/video are off by 70ms or greater. Anything less is a non-issue, so I wouldn't waste so much time on it otherwise.

The real problem is when the small amounts per segment amount to a total offset that's greater than 70ms.
the problem is when knowing theres a little delay, even when it would possibly not be perceivable otherwise. it would just make me myself mad, I would look at the sync all the time and this would spoil the fun for me :S So I'll have to correct :S

[drmpeg]

LPCM: 5ms
DD+: 32ms
TrueHD: 0.83ms
DTS-HD MA: 10.666ms
DTS-HD HiRes: 10.666ms

Note that these are the maximum amounts of extra audio. On average, it will be half of the maximums. And yes, there's too much audio, so the video needs to slow down.

Don't use the -z option. That's for demuxing to PES packets, which almost no tool uses. Also, I think it's a little buggy (and never fixed since nobody uses it).

Here's an article on lip-sync tolerance.

http://www.tvtechnology.com/features/audio_notes/f-TC-Keeping_it_all.shtml

"The International Telecommunications Union (ITU) released a specification called BT.1359-1 in 1998. It was based on research that showed the reliable detection of A/V sync errors was between 45 msec audio leads video to 125 msec audio lags video. Remember, this is just the detectability region; the acceptability region is an even wider +90 to -185 msec."

Ron

[Thunderbolt8]

thanks for completing the list

hm the difference between max and average shouldnt make too much difference, at least not for trueHD and LPCM. but with movies with ~30 .m2ts files then it can already make a little difference (~50ms) for dts-hd and surely will in case of DD+ (~500ms). guess theres some experimenting needed :S

[Snowknight26]

When I join 3 m2ts files and demux the LPCM stream with xport, eac3to comes up saying that the format isn't recognized, though a graphedit graph of File -> ffdshow audio decoder shows that it is indeed (L)PCM.
Need a sample?

[shambles]

Chopping off audio samples would be much too heavy handed (and annoyingly audible).

how would it be audible? if you calculate the amount of samples you need to chop off at the end of each audio segment, wouldn't you just be restoring it to its original length so it matches perfectly with the video?

for 48khz there's 48 samples per ms so you can get pretty damned accurate if you just have the patience to do it right.

(from earlier in this thread)

The last audio PTS is 2752710 and the last video PTS is 2749008. The video ends at 2749008 + 3754 = 2752762. The audio ends at 2752710 + 450 = 2753160. 2753160 - 2752762 = 398. 398 / 90000 = 4.42 milliseconds too much audio.

from your example, take the 398 / 90000, multiply by 1000 = 4.4222..., multiply by 48 (48 samples/ms), then multiply by 18 (18 bytes/sample for 24bit/6channels) = 3820.8. but of course you can't chop off 3820.8 bytes off a file, and what we need to chop off needs to be dividable by 18 for 24bit/6ch files to keep the channel order correct, so divide it by 18 again = 212.2666..., then multiply 212 by 18 = 3816 which is the correct amount of bytes to chop off from the end of the pcm file. and if you want to be extra anal about it, like i do, you can subtract the 3816 from the 3820.8 and see that 4.8 bytes would count toward the next pcm file. :p

it takes a lot of time but i feel it's worth it. what would make it a bit easier would be if xport could be updated with a switch that displays only the last video and audio pts

[madshi]

When I join 3 m2ts files and demux the LPCM stream with xport, eac3to comes up saying that the format isn't recognized, though a graphedit graph of File -> ffdshow audio decoder shows that it is indeed (L)PCM.
Need a sample?
Yes, a sample would be helpful. But before uploading the sample please check whether you can reproduce the problem with the sample. If eac3to works fine with the sample, the sample won't help.

[madshi]

Chopping off audio samples would be much too heavy handed (and annoyingly audible).
I'm wondering what the studio muxing software does when creating the multiple TS parts. I mean the studio master is one big video and audio file. Now if we join the TS parts and demux audio and video, audio is probably a few milliseconds too long. So obviously the TS muxing software must have added some audio data somewhere, right? How did it do that? Isn't it the most probably thing that the audio data is ever so slightly overlapping at the end of the first TS part and at the beginning of the 2nd TS part? So if you just join the PCM samples, a few milliseconds worth of audio data are played twice? Wouldn't chopping off the audio samples probably result in that we get the original audio track again? I don't really know, I'm just guessing...

Obviously things are much more complicated with AC3 compared to LPCM because we can't just chop off half an AC3 frame. So for AC3 demuxing the only reasonable solution would be to keep track of the "too much audio milliseconds". Once the delay sums up to over the length of an AC3 frame, we could then chop off one full AC3 frame (or several, if necessary). This way we could keep delay under the length of one AC3 frame.

Thoughts?

[rickardk]

Can someone please help me with this I would be very very thankful.

Just bought Bourne Ultimatum and I would like to remux it into TS and put it on my server so I can have it playable in Vista Media Center.


The edition I bought have a TrueHD track. And as there are no decoders that works with vista media center I will (if possible) convert it to flac. I don't really understand about sync problems and how to fix them.

Also the pulldown (29.97fps issue) must be removed from the VC-1 stream before it's muxed into a ts.

I know in theory what needs to be done. But not how (tools and exec) to do it.

HD DVD disc -> Demux VC-1 es and TrueHD -> remove pulldown from VC-1 es -> a perfect convert TrueHD to FLAC -> sync everything 100% -> mux into ts

If someone could write a step-by-step guide to get a perfect in Vista Media Center playable TS with the same exact sq as the truehd track (no dynamic range compression and stuff like that). I will just need one (TrueHD->FLAC) audio stream. And also a perfect VC-1 es without stutter or speed up. It would be the best christmas gift ever!



Sorry If this post belongs in another thread...

[madshi]

Can someone please help me with this I would be very very thankful.

Just bought Bourne Ultimatum and I would like to remux it into TS and put it on my server so I can have it playable in Vista Media Center.

The edition I bought have a TrueHD track. And as there are no decoders that works with vista media center I will (if possible) convert it to flac. I don't really understand about sync problems and how to fix them.

Also the pulldown (29.97fps issue) must be removed from the VC-1 stream before it's muxed into a ts.

I know in theory what needs to be done. But not how (tools and exec) to do it.

HD DVD disc -> Demux VC-1 es and TrueHD -> remove pulldown from VC-1 es -> a perfect convert TrueHD to FLAC -> sync everything 100% -> mux into ts

If someone could write a step-by-step guide to get a perfect in Vista Media Center playable TS with the same exact sq as the truehd track (no dynamic range compression and stuff like that). I will just need one (TrueHD->FLAC) audio stream. And also a perfect VC-1 es without stutter or speed up. It would be the best christmas gift ever!

Sorry If this post belongs in another thread...
Discussion about how to remux video doesn't really belong here. This thread is only about audio processing. For TrueHD -> FLAC decoding just demux the TrueHD audio track with EvoDemux and then use eac3to to recode that to FLAC. You'll get perfect quality, as long as you use the Nero or libav decoder. EvoDemux will tell you the delay value you need to use for the TrueHD track. You can feed that delay value into eac3to.

[idbirch2]

Also the pulldown (29.97fps issue) must be removed from the VC-1 stream before it's muxed into a ts.
I was under the impression that all HD-DVDs were 23.976p...

[madshi]

Please no video discussion in this thread. The thread is already long enough without that. Thanks.

[TheSof]

I've used eac3to alot, but all of a sudden I'm getting a flac data mismatch error. Sounds like an overclocking issue, I know, but it has been stable for years now and has no problems using eac3to with other files - it's just Batman Begins I'm getting this error with.

[Phantomas_X]

can anybody post the link to delycut 1.3 ? I can't find the tread...:thanks:

[bobasp1]

Can any1 let me know if theres a way around this problem? I have the nero decoder installed, but it doesn't seem to want to work.

http://img517.imageshack.us/img517/5960/hlpyw5.jpg

[Thunderbolt8]

you cant use .evo files as input any more, you need to demux the audio stream out of it first.

[hristoff2]

can anybody post the link to delycut 1.3 ? I can't find the tread...:thanks:

--> http://uploaded.to/?id=ym2ck7 :)

[bobasp1]

you cant use .evo files as input any more, you need to demux the audio stream out of it first.

I did it saved it as a mpa file and doesn't seem to work.. ether

What im trying to do is basicly get the TrueHD to Flac audio but yeah I'm kind of lost because it when i use evodemux to take out the truehd audio its in a .mpa extension and it doesn't work either.

I was testing to see if the nero plugin was working *maybe theres more than one but* when I was doing a high bit rate file to a low one it did it just fine.

@Thunderbolt8 I did demux the audio stream and like I said it didn't work because its in mpa extension *changed it to a few others and it didn't work* What would you suggest.

[Thunderbolt8]

im getting confused again with that first pts thing of video and audio. here in this case the delay is given for the video stream and not for the audio :S

PTM of first video frame = 00001269
PTM of last video frame = 10BE4675
Duration = 0:52:01.118
H.264 (AVC) video stream 0 found!
First PTS = 00002FBD (+83ms)
DTS HD (DTS) audio stream 0 found!
First PTS = 00001269

so which would the delay then be for the audio, +83 or -83 ?
and since the values of first video frame is the same as of the first PTS of the audio, but not the same as the first PTS of the video is there a delay needed at all in this case? :S

[Snowknight26]

Yes, a sample would be helpful. But before uploading the sample please check whether you can reproduce the problem with the sample. If eac3to works fine with the sample, the sample won't help.

It doesn't work.

Here is a sample of the beginning 20MB of the 1st m2ts file, then a sample of the demuxed (with xport) LPCM stream from the sample, respectively.

http://www.stfcc.org/misc/therock.m2ts
http://www.stfcc.org/misc/therock.mpa

I've sent the same samples to drmpeg to see if he can find any problems.

[madshi]

I've used eac3to alot, but all of a sudden I'm getting a flac data mismatch error. Sounds like an overclocking issue, I know, but it has been stable for years now and has no problems using eac3to with other files - it's just Batman Begins I'm getting this error with.
Can you reproduce the problem with a little sample? If so, please upload that sample somewhere, please. Thanks...

[madshi]

I did it saved it as a mpa file and doesn't seem to work.. ether

What im trying to do is basicly get the TrueHD to Flac audio but yeah I'm kind of lost because it when i use evodemux to take out the truehd audio its in a .mpa extension and it doesn't work either.

I was testing to see if the nero plugin was working *maybe theres more than one but* when I was doing a high bit rate file to a low one it did it just fine.

@Thunderbolt8 I did demux the audio stream and like I said it didn't work because its in mpa extension *changed it to a few others and it didn't work* What would you suggest.
Please use EvoDemux to demux the TrueHD stream and then let eac3to do its work. If that still doesn't work, please let us see what eac3to outputs. Also please try the "-libav" switch. Does that one work?

[madshi]

im getting confused again with that first pts thing of video and audio. here in this case the delay is given for the video stream and not for the audio :S

PTM of first video frame = 00001269
PTM of last video frame = 10BE4675
Duration = 0:52:01.118
H.264 (AVC) video stream 0 found!
First PTS = 00002FBD (+83ms)
DTS HD (DTS) audio stream 0 found!
First PTS = 00001269

so which would the delay then be for the audio, +83 or -83 ?
and since the values of first video frame is the same as of the first PTS of the audio, but not the same as the first PTS of the video is there a delay needed at all in this case? :S
That's a good question. A day or two ago I'd have said you need to delay audio by -83ms. But right now I'm not sure. Right now I think that it depends on what you do with the video. If you remux the video with gdsmux I guess that no audio delay is needed in this case. But if you demux video and remux it with mkvtoolnix (or mp4creator/mp4box) then probably you'll have to apply -83ms delay to the audio track. But I'm not totally sure. I might have more information about this later...

[bobasp1]

Please use EvoDemux to demux the TrueHD stream and then let eac3to do its work. If that still doesn't work, please let us see what eac3to outputs. Also please try the "-libav" switch. Does that one work?

I just gave up used FFmpeg the one posted on the site with eac3 or something and i just ripped it straight from the evo tested it and there wasn't any problems. ripped it in Flac, and ac3 to see if there were any problems and there wasn't. So yeah.. good night every1.

[Thunderbolt8]

That's a good question. A day or two ago I'd have said you need to delay audio by -83ms. But right now I'm not sure. Right now I think that it depends on what you do with the video. If you remux the video with gdsmux I guess that no audio delay is needed in this case. But if you demux video and remux it with mkvtoolnix (or mp4creator/mp4box) then probably you'll have to apply -83ms delay to the audio track. But I'm not totally sure. I might have more information about this later...
I will definately have to use mkvmerge sooner or later.

[madshi]

It doesn't work.

Here is a sample of the beginning 20MB of the 1st m2ts file, then a sample of the demuxed (with xport) LPCM stream from the sample, respectively.

http://www.stfcc.org/misc/therock.m2ts
http://www.stfcc.org/misc/therock.mpa

I've sent the same samples to drmpeg to see if he can find any problems.
Works for me. eac3to only checks for RAW/PCM files if the file extension is "*.raw" or if the text "pcm" appears somewhere in the file name or file extension. Otherwise eac3to doesn't test for RAW/PCM because the test for RAW/PCM can consume a lot of time. So please rename "therock.mpa" to "therock.pcm" or "therock PCM.mpa" or something like that. Then eac3to should do its work.

[madshi]

I will definately have to use mkvmerge sooner or later.
That is not important. Import is if there's a step in your processing chain which loses the timestamps of the EVO container. If you use gdsmux + mkvmerge, the timestamps are preserved. However, as soon as you *de*mux the video to a raw file, the timestamps are lost. When the original timestamps of the EVO file are lost, you will have to apply -83ms to the audio file. If the timestamps are not lost, I think you don't need to apply a delay in this specific case, because the first audio timestamp matches the "PTM of first video frame" timestamp of the EVO file. I'm not fully sure yet, though.

[Thunderbolt8]

what I did is I demuxed both streams, video and audio, with evodemux and then muxed it together in mkvmerge. im not 100% sure yet, but it might indeed be the way that it looks best with no delay at all applied. the source is the 1st band of brothers HD DVD and since I already looked into the 2nd disc too I can say these timestamps for video and audio are the same for all 4 episodes so far.
I might give it a try and do both, also remux with gdsmux or haali filters or h264tsto (should all be the same, right? btw. will it be ok if I only remux the video stream in the .mkv file for the timestamps instead of having 1 audio stream in it too?) and then try to compare the lip sync of all 3 episodes (the 1st EP gives me funny rainbow frames :S)

[madshi]

what I did is I demuxed both streams, video and audio, with evodemux and then muxed it together in mkvmerge. im not 100% sure yet, but it might indeed be the way that it looks best with no delay at all applied.
You got my explanation the wrong way. If you demux video, you *do* need to apply a delay in this specific case.

should all be the same, right? btw. will it be ok if I only remux the video stream in the .mkv file for the timestamps instead of having 1 audio stream in it too?
Yes, should be the same. Only remuxing video is ok.

[Chumbo]

im getting confused again with that first pts thing of video and audio. here in this case the delay is given for the video stream and not for the audio :S

PTM of first video frame = 00001269
PTM of last video frame = 10BE4675
Duration = 0:52:01.118
H.264 (AVC) video stream 0 found!
First PTS = 00002FBD (+83ms)
DTS HD (DTS) audio stream 0 found!
First PTS = 00001269

so which would the delay then be for the audio, +83 or -83 ?
and since the values of first video frame is the same as of the first PTS of the audio, but not the same as the first PTS of the video is there a delay needed at all in this case? :S
From my experience, it's been consistently correct to do the following.
- if the delay is on the video, then apply the opposite to the audio, e.g., if it's +83ms then apply -83ms to the audio. I do use the mkv tools on occasion and have not had any issues

- if the delay is listed on the audio, then apply it as is to the audio, i.e., if it shows +83ms, then apply 83ms to the audio.

But as madshi mentioned, it's possible other issues may crop under the right conditions. Just because I haven't run into them doesn't mean they're not there. ;)