EVOB De/Multiplexers

[madshi]

First of all try not to use the GUI but rather the cli. This way you have more feedback. Also for me tranzcode fails to do 24>16 bit conversion. But that's not a problem. Just interpose in the chain in graph (between sonic and wavdest) ffdshow or ac3filter (set to 16 bit output) and you get 16 bit wav that tranzcode doesn't have problems with. This was tried on a small dd+, one thing that can occur is that windows and >2 gb wavs is a problematic issue, most clis won't take it as input (I remember that from the times I used to transcode big wav to aac 5.1).
Thank you. Adding ac3filter in between crashed. But adding ffdshow (tryouts) in between with output set to 16bit worked great. Now also all 6 channels have correct information in them! Yippie!!

However, there's still one problem left: Tranzcode still doesn't work. Here's what it sais:

D:\>tranzcode test.wav tmp
Input filename: [test.wav] (Multichannel Wave)
Input attribs: [Ext wav fmt, 6 ch, 16 bit, 48.0 kHz]
Output attribs: [PCM wav fmt, (6) x 1 ch, 16 bit, 48.0 kHz]
Output file(s): tmp-FL.wav
tmp-FR.wav
tmp-C.wav
tmp-LFE.wav
tmp-SL.wav
tmp-SR.wav
Processing: Completed. [in 1 sec]
Again all files are 44 Bytes long.

Edit: Asking tranzcode to resample to 44100 doesn't help, either.

[kornesque]

hey guys. unlike most of you i'm actually trying to derive a simple stereo file from the demuxed dd+ stream. i've tried as many variations of filters as i can stand, but the only one that seems to get me anywhere (won't crash) is the dd+ source filter made by orbitlee (cheers man!). when i slap my file in the graph riding that filter and try putting it to the intervideo audio decoder, the only way i can find to render it in any way is to connect inteligent, which produces 4 acm wrappers, roxio mp3 encoder, and ffdshow decoder in the circuit. using filewriter at the end, the result is a useless file, unrecognized by anything i have.

can anybody help me build a simple dd+>stereo .wav graph? i've been following this thread forever, and have learned alot, but i'm still stabbing in the dark with most of this stuff. any help would be greatly appreciated.

[Daodan]

@madshi, if wav file > 2gb it may be that the problem. Try splitting the wav and then try again. Also there are other programs that can make 6 mono wavs, BeLight for example (GUI for besweet).

@kornesque, getting 2 channel was never a problem.
A graph like evo>sonic demuxer>intervideo audio decoder>wavdest>filewriter gives 2.0 output (that is all intervideo seems to give, if you set it to 5.1 it gives extra empty channels).

[kornesque]

thanks for the assist Daodan. i checked out that graph, and i can't get the pins from sonicdemux>iviaudio to connect. is there something i might be forgetting? also, i tried the sonic and ffdshow audio decoders in lieu of intervideos, but still get no useable file.

[juhu]

sorry to interrupt the audio discussion, but I'd like to point out I'm not sure anymore Hd-dvd run @ 23.976 fps

let's take a concrete example : Casablanca
in xpl file, we have the correct timing : titleTimeEnd="01:42:28:18"

now, with evodemux, number of calculated frames is 147562 which seems correct.

so after video encoding @ 23.976 fps, I end up with a file ending @ 1h42:34.xxx, which is 6 seconds longer than the timing indicated in xpl file, and also 6 seconds later than the (sd) dvd !

now if these 147562 frames were running @ 24 fps, I would have a file with the correct original timing (1h42m28s)
Am I right or Did I miss something somewhere?

[edtchie]

Yes, but unless you're starting from

xport -h movie.m2ts 1 1 2
or
xport -h movie.m2ts 1 1 3
or
xport -h movie.m2ts 1 1 4
or
xport -h movie.m2ts 1 1 5

Ron

Can you set up an option for demuxing audio track only, do not have to demux video track everytime?

[drmpeg]

sorry to interrupt the audio discussion, but I'd like to point out I'm not sure anymore Hd-dvd run @ 23.976 fps

let's take a concrete example : Casablanca
in xpl file, we have the correct timing : titleTimeEnd="01:42:28:18"

now, with evodemux, number of calculated frames is 147562 which seems correct.

so after video encoding @ 23.976 fps, I end up with a file ending @ 1h42:34.xxx, which is 6 seconds longer than the timing indicated in xpl file, and also 6 seconds later than the (sd) dvd !

now if these 147562 frames were running @ 24 fps, I would have a file with the correct original timing (1h42m28s)
Am I right or Did I miss something somewhere?
Very interesting. I would try to determine the running time of the audio to see which timeline (23.976 or 24) it matches. Since the DD+ tracks are fixed bitrate, you should be able to calculate the running time based on the size of the demuxed audio file divided by the bitrate of the DD+ track.

Ron

[drmpeg]

Can you set up an option for demuxing audio track only, do not have to demux video track everytime?
Yes. In the meantime, just select a video track that's not in the multiplex for subsequent audio tracks. Like this:

xport -h movie.m2ts 1 1 1

This will demux the video and audio track 1. Then:

xport -h movie.m2ts 1 3 2

This will demux audio track 2 and create a zero byte file for bits0001.mpv (be sure to rename the first bits0001.mpv that has the demuxed video beforehand). The demux process will be much faster since there's no video to demux.

xport -h movie.m2ts 1 3 3

xport -h movie.m2ts 1 3 4

and so on.

Ron

[MichalHabart]

Thank you. Adding ac3filter in between crashed. But adding ffdshow (tryouts) in between with output set to 16bit worked great. Now also all 6 channels have correct information in them! Yippie!!

However, there's still one problem left: Tranzcode still doesn't work. Here's what it sais:

D:\>tranzcode test.wav tmp
Input filename: [test.wav] (Multichannel Wave)
Input attribs: [Ext wav fmt, 6 ch, 16 bit, 48.0 kHz]
Output attribs: [PCM wav fmt, (6) x 1 ch, 16 bit, 48.0 kHz]
Output file(s): tmp-FL.wav
tmp-FR.wav
tmp-C.wav
tmp-LFE.wav
tmp-SL.wav
tmp-SR.wav
Processing: Completed. [in 1 sec]
Again all files are 44 Bytes long.

Edit: Asking tranzcode to resample to 44100 doesn't help, either.

That's weird because with this procedure (put ffdshow in between and set its output to integer 16) it is working for me. now i am able to create 6 mono wav files and even the multi wav is different (foobar no longer says it's wav, now it is PCM at 4800kbps for him) and it is smaller (first one was 2100MB, second one is now 1500MB). But i am still trying only dd+ from first evo file, i think that might be your problem (too big wav)

[Darth Pinous]

Thank you. Adding ac3filter in between crashed. But adding ffdshow (tryouts) in between with output set to 16bit worked great. Now also all 6 channels have correct information in them! Yippie!!

However, there's still one problem left: Tranzcode still doesn't work.
Again all files are 44 Bytes long.

Edit: Asking tranzcode to resample to 44100 doesn't help, either.

Madshi, did you try Wavewizard (http://forum.doom9.org/showthread.php?t=95265) ?

There's an option in it to ignore file header for file > 2GB / file > 4GB / every file, and an option to split a multichannel WAV in 6 mono WAV.

[Deckard2019]

Madshi, did you try Wavewizard ?
It works great. Thx. No need to convert to 16bit anymore.
All 6 wav files sounds ok. But Surcode produces weird result.
Do you know something else than Surcode to create DTS file ?

Also, in File Writer properties, there is a check box named "Truncate ?".
Do you know what is it for ?

[madshi]

Thank you guys for your suggestions. BeLight/BeSweet doesn't work, it complains about unknown input file format. WaveWizard seems to do the trick... :)

And yes, my file was > 2GB. I guess that was the problem with Tranzcode. I've only done the first EVO, not both, but still the file was > 2GB, so I guess Tranzcode won't do for some HD DVDs.

[Pelican9]

let's take a concrete example : Casablanca
in xpl file, we have the correct timing : titleTimeEnd="01:42:28:18"

now, with evodemux, number of calculated frames is 147562 which seems correct.

so after video encoding @ 23.976 fps, I end up with a file ending @ 1h42:34.xxx, which is 6 seconds longer than the timing indicated in xpl file, and also 6 seconds later than the (sd) dvd !

now if these 147562 frames were running @ 24 fps, I would have a file with the correct original timing (1h42m28s)
Am I right or Did I miss something somewhere?
Could you copy the EVOdemux's status window after read this evo (check 'Detailed info')?

[MichalHabart]

It works great. Thx. No need to convert to 16bit anymore.
All 6 wav files sounds ok. But Surcode produces weird result.
Do you know something else than Surcode to create DTS file ?

Also, in File Writer properties, there is a check box named "Truncate ?".
Do you know what is it for ?

After registering orbitlee filter, i was able to demux dd+ files but DTS tracks aren't playable anymore. I had to uninstall it to restore. It might be your problem. Also surcode has known issue, it produce longer DTS track. Try to fix it with delaycut (just open and proceed, don't change anything) and besliced 0.3 can do it also.

[Darth Pinous]

It works great. Thx. No need to convert to 16bit anymore.
All 6 wav files sounds ok. But Surcode produces weird result.
Do you know something else than Surcode to create DTS file ?

Also, in File Writer properties, there is a check box named "Truncate ?".
Do you know what is it for ?

Have you tried the 16bits/44Khz conversion in WaveWizard (the "convert samplingrate" option and the "convert sample type") ?
16 bits conversion is mandatory, but 44 KHz conversion may be necessary too.

For alternative DTS encoders, you have Nuendo DTS encoder (a plugin in Nuendo) which seems to be better than Surcode, and of course the DTS encoders from DTS Corp, the "old" DTS Pro Series Encoder and the new DTS Surround Audio Suite and DTS-HD Master Audio Suite encoders.

Take a look here (http://www.quadraphonicquad.com/forums/showthread.php?t=7487) for a review on these 5 encoders.

[chros]

now if these 147562 frames were running @ 24 fps, I would have a file with the correct original timing (1h42m28s)
Yes, that's very strange ... Some people had done various encoding with lots of sources (eg. some has AC3 audio) and they only get the untouched audio synced, if they use 23.976 fps.

So the question is what the player is doing with this xpl file during playback, and what is the tickBase , timeBase , tickBaseDivisor parameters.

A sample:
...
<TitleSet tickBase="60fps" timeBase="60fps" defaultLanguage="en">
...
<Title id="feature" displayName="Feature Presentation" titleNumber="3" titleDuration="01:42:54:16" tickBaseDivisor="4" onEnd="black">

Another sample, notice that isn't any tickBaseDivisor parameters:
...
<TitleSet tickBase="24fps" timeBase="60fps" defaultLanguage="en">
...
<Title titleNumber="2" alternativeSDDisplayMode="letterbox" titleDuration="01:58:54:22" id="MainMovie" displayName="MainMovie" onEnd="returnToMM">

[MichalHabart]

Have you tried the 16bits/44Khz conversion in WaveWizard (the "convert samplingrate" option and the "convert sample type") ?
16 bits conversion is mandatory, but 44 KHz conversion may be necessary too.

For alternative DTS encoders, you have Nuendo DTS encoder (a plugin in Nuendo) which seems to be better than Surcode, and of course the DTS encoders from DTS Corp, the "old" DTS Pro Series Encoder and the new DTS Surround Audio Suite and DTS-HD Master Audio Suite encoders.

Take a look here (http://www.quadraphonicquad.com/forums/showthread.php?t=7487) for a review on these 5 encoders.

I am little bit confused from that article. Does it mean that surcode is best or worst?

[Darth Pinous]

I am little bit confused from that article. Does it mean that surcode is best or worst?

worst.

according to this neil wilkes, quality is as follow :
DTS-HD MAS > DTS-HD SAS > DTS PSE >> Nuendo > Surcode

[MichalHabart]

worst.

according to this neil wilkes, quality is as follow :
DTS-HD MAS > DTS-HD SAS > DTS PSE >> Nuendo > Surcode

Hmm, anyone knows how to find those best encoders (i mean, if there is any version of them available on the net)

I have surcode and nuendo but they seems to be the worst :(

[Deckard2019]

To get rid off file size limit of WAV files, try :
1/ Open Graphedit, DTS/AC3/DD+ Source => Sonic Cinemaster Audio Decoder 4.2,
2/ Create AVS script with just :
DirectShowSource("audio.grf", video=false)
GetChannel(X)
where X=1 to 6 (see : GetChannel.htm in your AVISynth install dir).
Channel mapping is 1=FL, 2=FR, 3=C, 4=LFE, 5=SL, 6=SR,
3/ Use BeHappy (http://forum.doom9.org/showthread.php?t=104686) to generate each channel as WAV file directly from AVS script.

16bits/44Khz conversion can be done in AVS script (ConvertAudioTo16bit() and ResampleAudio()).
This process skips the big WAV file creation step. No more Tranzcode ;)

Still looking for a reliable DTS encoder ...

[MichalHabart]

To get rid off file size limit of WAV files, try :
1/ Open Graphedit, DTS/AC3/DD+ Source => Sonic Cinemaster Audio Decoder 4.2,
2/ Create AVS script with just :
DirectShowSource("audio.grf", video=false)
GetChannel(X)
where X=1 to 6 (see : GetChannel.htm in your AVISynth install dir).
Channel mapping is 1=FL, 2=FR, 3=C, 4=LFE, 5=SL, 6=SR,
3/ Use BeHappy (http://forum.doom9.org/showthread.php?t=104686) to generate each channel as WAV file directly from AVS script.

16bits/44Khz conversion can be done in AVS script (ConvertAudioTo16bit() and ResampleAudio()).
This process skips the big WAV file creation step. No more Tranzcode ;)

Still looking for a reliable DTS encoder ...

Then once you will find him, can you please make some sort of guide for this? So, we can try your steps and help you with testing :)

[orbitlee]

After registering orbitlee filter, i was able to demux dd+ files but DTS tracks aren't playable anymore. I had to uninstall it to restore.

Could you provide more informations? DTS(from DVD) or DTS-HD(from HD DVD/Blu-ray)? bitrate? Which dts decoder you are using?
The behavior of Sonic audio decoder is very strange, I have to feed it multiple frames(in source filter) each time to make it playback DD+ audio. This may fail other audio decoder.

[MichalHabart]

Could you provide more informations? DTS(from DVD) or DTS-HD(from HD DVD/Blu-ray)? bitrate? Which dts decoder you are using?
The behavior of Sonic audio decoder is very strange, I have to feed it multiple frames(in source filter) each time to make it playback DD+ audio. This may fail other audio decoder.

I tried DTS from DVD (terminator 3) and it had length of aprox. 4 hours and play was very fast :) I used both AC3Filter and sonic audio decoder, both had same behaviour.

[juhu]

Could you copy the EVOdemux's status window after read this evo (check 'Detailed info')?

sure. Here goes :
evodemux video stats :
---------------------

Opening file Feature_1.EVO
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
Substream id = 55
Sequence Header found
0 frames before first I-frame
Advanced Profile
Level = 3
Chroma Format = 4:2:0
Size = 1920x1080
Display Horizontal size = 1920
Display Vertical size = 1080
Aspect ratio = 1:1 (square samples)
Frame Rate = 23,976 (24000/1001)

(subs / audio info removed)

Continue on Feature_2.EVO
VOB number 5 contains 1 video , 5 audio and 4 subpicture streams.
PTM of first video frame = 105CFFB1
PTM of last video frame = 21040FDA
Duration = 1:42:34.565
VC-1 video stream 0 found!
First PTS = 105D1D05
Substream id = 55
Sequence Header found
0 frames before first I-frame
Advanced Profile
Level = 3
Chroma Format = 4:2:0
Size = 1920x1080
Display Horizontal size = 1920
Display Vertical size = 1080
Aspect ratio = 1:1 (square samples)
Frame Rate = 23,976 (24000/1001)

147590 counted frames (1:42:35.733) in video stream 0.
147562 calculated frames.

-----------------

Very interesting. I would try to determine the running time of the audio to see which timeline (23.976 or 24) it matches. Since the DD+ tracks are fixed bitrate, you should be able to calculate the running time based on the size of the demuxed audio file divided by the bitrate of the DD+ track.

Ron



audio stats :
Dolby Digital Plus audio stream 0 found!
First PTS = 00000A67
Substream id = C0
Stream 0 is Dolby Digital Plus
frame size = 256 bytes, number of blocks per frame = 6
Sampling frequency = 48 kHz
Transmission bitrate = 64 kbit/s
Channel arrangement = mono, bsid = 16
LFE channel = not present

demuxed audio filesize : 48083 ko (wow, tiny !)
...which is seen as a 1h42mn34sec file by media player classic
if someone wants to do maths with it, well please do, coz' I give up this nonsense (48083/ 8ko (64kb) = 6010.xx sec, ie 1h40m10s ?!?)


now xpl extract :
<TitleSet tickBase="60fps" timeBase="60fps" defaultLanguage="en">
<FirstPlayTitle titleDuration="00:00:22:00">
<PrimaryAudioVideoClip titleTimeBegin="00:00:00:00" titleTimeEnd="00:00:22:00"
src="file:///dvddisc/HVDVD_TS/FBILogo.MAP" dataSource="Disc"
description="FBI Warning WB Logo">
<Video track="1"/>
<Audio track="1"/>
</PrimaryAudioVideoClip>
</FirstPlayTitle>
<Title titleNumber="1" id="intro" titleDuration="00:02:00:00" description="Intro to HD DVD" tickBaseDivisor="4" onEnd="feature">
<PrimaryAudioVideoClip titleTimeBegin="00:00:00:00" titleTimeEnd="00:02:00:00"
src="file:///dvddisc/HVDVD_TS/IntroHD.MAP" dataSource="Disc"
description="Intro to HD DVD">
<Video track="1"/>
<Audio track="1"/>
</PrimaryAudioVideoClip>
</Title>
<Title titleNumber="2" id="feature" displayName="Feature Presentation" titleDuration="01:42:51:18" tickBaseDivisor="4" onEnd="black">
<PrimaryAudioVideoClip id="layer0" titleTimeBegin="00:00:00:00" titleTimeEnd="00:50:47:08" dataSource="Disc" src="file:///dvddisc/HVDVD_TS/Feature_1.MAP">
<Video mediaAttr="2" track="1" angleNumber="1"/>
<Audio mediaAttr="1" track="1" streamNumber="1"/>
<Audio mediaAttr="1" track="2" streamNumber="2"/>
<Audio mediaAttr="1" track="3" streamNumber="3"/>
<Audio mediaAttr="1" track="4" streamNumber="4"/>
<Audio mediaAttr="1" track="5" streamNumber="5"/>
<Subtitle mediaAttr="1" track="1" streamNumber="1"/>
<Subtitle mediaAttr="1" track="2" streamNumber="2"/>
<Subtitle mediaAttr="1" track="3" streamNumber="3"/>
<Subtitle mediaAttr="1" track="4" streamNumber="4"/>
</PrimaryAudioVideoClip>
<PrimaryAudioVideoClip id="layer1" titleTimeBegin="00:50:47:08" titleTimeEnd="01:42:28:18" dataSource="Disc" src="file:///dvddisc/HVDVD_TS/Feature_2.MAP" seamless="true">
<Video mediaAttr="2" track="1" angleNumber="1"/>
<Audio mediaAttr="1" track="1" streamNumber="1"/>
<Audio mediaAttr="1" track="2" streamNumber="2"/>
<Audio mediaAttr="1" track="3" streamNumber="3"/>
<Audio mediaAttr="1" track="4" streamNumber="4"/>
<Audio mediaAttr="1" track="5" streamNumber="5"/>
<Subtitle mediaAttr="1" track="1" streamNumber="1"/>
<Subtitle mediaAttr="1" track="2" streamNumber="2"/>
<Subtitle mediaAttr="1" track="3" streamNumber="3"/>
<Subtitle mediaAttr="1" track="4" streamNumber="4"/>
</PrimaryAudioVideoClip>
<PrimaryAudioVideoClip titleTimeBegin="01:42:28:18" titleTimeEnd="01:42:51:18"
src="file:///dvddisc/HVDVD_TS/Interpol.MAP" dataSource="Disc"
description="Motion Picture Association of America">
<Video track="1"/>
</PrimaryAudioVideoClip>

[Pelican9]

Opening file Feature_1.EVO
PTM of first video frame = 00000D91
PTM of last video frame = 105CFFB1
Duration = 0:50:50.297
Continue on Feature_2.EVO
PTM of first video frame = 105CFFB1
PTM of last video frame = 21040FDA
Duration = 1:42:34.565

147562 calculated frames.
Dolby Digital Plus audio stream 0 found!
frame size = 256 bytes, number of blocks per frame = 6
Sampling frequency = 48 kHz
Transmission bitrate = 64 kbit/s
Channel arrangement = mono, bsid = 16
LFE channel = not present

demuxed audio filesize : 48083 ko (wow, tiny !)
...which is seen as a 1h42mn34sec file by media player classic
<PrimaryAudioVideoClip id="layer1" titleTimeBegin="00:50:47:08" titleTimeEnd=[b]"01:42:28:18" dataSource="Disc" src="file:///dvddisc/HVDVD_TS/Feature_2.MAP" seamless="true">


I think the XPL's info is irrelevant.
The duration info comes from the DSI packet(s) of the EVOB (difference of the PTMs). I think this is the real value. The length of the demuxed audio is the same.

[JnZ]

Update: Go here:http://forum.doom9.org/showthread.php?p=966093#post966093

Hi folks,
after experimenting few hours, I found this working procedure to convert DD+ to anything else:

So here is updated Guide (old guide can't go over 4G barrier):
1) Create Graph filter: DTS/AC3/DD+ Source -> Sonic Cinemaster Audio Decoder 4.2 -> Dump (for DTS/AC3/DD+ open demuxed DD+ track, for Dump open output RAW (test.raw)). This grab RAW PCM data from Sonic decoder and put into file.
2) Pres play to render RAW PCM data. This rawdata file is depend on Sonic setting, so make sure you set 5.1 output. You've get 5.1ch 16bit (depends maybe on soundcard type), 48khz 6ch RAW data file without header. (it is critical to use Dump filter,coz it can take You over 4G barrier...which file writer evidently not). Wait until filesize of rawfile stop raising, than you can stop graphedit.
3) Use SoX to convert PCM to WAVe: sox -r48000 -t .raw -c 6 -w -s test.raw x:\test.wav (try set output to another physical disk, it speedups the process)
4) If you want 6 separate mono WAVes (for SurCode or other like-this stuff), use Wavewizard. In preferences sets Ignore size in header -> always, Stream manipulation -> Mono streams, Output format -> Wave PCM and finaly output dir. Open test.wav and press "Convert". It creates six WAVes 0-1-2-3-4-5 which match these channel order: FL-FR-C-LFE-SL-SR.
5)Encode final WAVes to anything you want.... for Besweet You need create .mux file for six mono WAVes (for AC3 encodes, etc), it contain lines like this (for AC3 encode):
x:\test_ch0.wav
x:\test_ch2.wav
x:\test_ch4.wav
x:\test_ch1.wav
x:\test_ch5.wav
x:\test_ch3.wav

Note 1: In my test, SurCode is crappy encoder, it can't handle 48khz WAVes correctly...I recomennded use other DTS encoder.

Note 2: Things You need:
- Graphfilter: http://www.doom9.org/Soft21/Filters/graphedit.rar
- SoX: http://sox.sourceforge.net/
- Dump ds filter (in graphedit package), use "register.bat".
- Wavewizard: http://www.rarewares.org/wavewiz/wavewizardv0.54b.zip

Note 3: First of all, prepare a lot of disk space: 2 hours of 6ch PCM or WAVe takes aprox 4G disk space. So You need double (for PCM and for WAVe).

EDIT: Channel order for BeSweet corrected!

[Deckard2019]

@JnZ :
Did you test this (http://forum.doom9.org/showthread.php?p=964725#post964725) ? or this (http://forum.doom9.org/showthread.php?p=964366#post964366) ?
It seems easier and lower cost disk space to me ...

[madshi]

Thanks for your efforts, JnZ, that sounds promising!

[JnZ]

@JnZ :
Did you test this (http://forum.doom9.org/showthread.php?p=964725#post964725) ? or this (http://forum.doom9.org/showthread.php?p=964366#post964366) ?
It seems easier and lower cost disk space to me ...
I test AC3 via FFDshow, and it works...but not for long DD+ (only for unilogo.ddp).
Your guide maybe works too (didn't try).

I give people another guide...so they can decide, and find best way for themselves.

Explanation: I like "old-school" "be-sure" methods... :D

[Deckard2019]

I test AC3 via FFDshow, and it works...but not for long DD+ (only for unilogo.ddp).
I've done a 2h20 movie with this method. No trouble. To be continued ;)

[JnZ]

I've done a 2h20 movie with this method. No trouble. To be continued ;)

Few minutes ago I tested AC3 produced from FFdShow and BeSweet and must say, that from FFDShow is totally crappy. LFE, SL and SR are totally different than original source. From BeSweet all OK.

So I strongly do not recomended use FFDshow output AC3!!!

Tested on UNILOGO.EVO

[madshi]

I've used EncWAVtoAC3 (which internally uses Aften) und the results look quite fine to me. In Audacity the graphs (pre and post AC3 compression) look nearly identical to me.

[Deckard2019]

So I strongly do not recomended use FFDshow output AC3!!!

So you mean, ffdshow can't encode AC3 properly ? So liba52 ?

Well, I've made several tries with different methods.
As we don't have any good DTS encoder, my goal is to produce an AC3@640kbps file directly from AVS script, without create a big WAV file. I can now do it with BeHappy, using Aften or ffmpeg.

How can I hear if the created AC3 file match perfectly the source ?
madshi, what are your Aften settings ant what is your Audacity method to compare the files ?

[orbitlee]

I think the XPL's info is irrelevant.
The duration info comes from the DSI packet(s) of the EVOB (difference of the PTMs). I think this is the real value. The length of the demuxed audio is the same.


I think the time difference is caused by drop-frame timecode vs non-drop timecode. So nothing wrong at here.

You may find this article helpful for understanding.
http://www.csif.org/html/dropframe.html

[juhu]

I think the time difference is caused by drop-frame timecode vs non-drop timecode. So nothing wrong at here.

You may find this article helpful for understanding.
http://www.csif.org/html/dropframe.html

thanks... but there's another "curiosity" :
I finally decoded DD+ (as .wav mono ) with sonic decoder 4.2->avs->behappy... And it appears sound ran slightly faster than images (@ 23.976). Choosing 24 fps for video didn't help either, so I toyed with timestrech dsp available via behappy, and finally found the perfect stretching value : 24.000 -> 23.990 to keep the A/v synched all along the film
Don't ask me why...

Now, I'm working on "the thing" ... and then again, audio decoded (with Aften this time) runs faster.
For this one, the correct audio stretching is 24.000->23.988 to have a constant sync with video (still encoded @ 23.976)
so weird...

Of course, I'm talking about progressive decay in all cases, nothing to do with the 100-150 ms a/v starting delay

[Icemaan]

Morning

Can everybody say me where i can download behappy . the webside i cant download the file is there a alternatvie download link

Thanks

Icemaan

[2439rf8hasjdert4]

Morning

Can everybody say me where i can download behappy . the webside i cant download the file is there a alternatvie download link

Thanks

Icemaan

Well, I don't think everybody can say you where download behappy, whatever that is, but I'm sure someone might tell you.

[Chumbo]

I believe there's a link a few posts up, otherwise google should prove useful.

[LRN]

Google for two keywords: "behappy" and "avisynth". It's 99.9% that you'll find BeHappy - Avisynth-based audio encoder rather than something else :)

Awesome work around here! Following guides from this thread i was able to decode and reencode Riddick HD-DVD into x264+AAC (only stereo ATM) (essentially, reencoded 16-GB VC-1 stream into 14-GB x264 stream (0.99 SSIM), so with AAC audio tracks it will fit on 15-GB HD-DVD disk). While i have some problems with fps, resync, decoding (from my usual playing toolkit [MPC/ffdshow + VLC + MPUI] no player is able to play this movie properly - either it hungs on some moment, or plays video faster than 23.976 fps, or can't open mp4 container) - it's all minor technical issues and could be solved. Currently i'm trying to smooth the whole process and to make it more open/free (i.e. without expensive software filters/decoders).
Nevertheless it works. We can decode and reencode HD-DVDs! CooL!

[madshi]

I have the same problem with using BeHappy that I had when I originally tried to create a wav directly with the Sonic Audio Decoder, only. Namely I got 2 good channels and 4 bad channels. The bad channels looked like loud noise to me with a tiny bit of audio information in the first 1-3 seconds of the track. Adding ffdshow to the graph behind the Sonic HD Audio Decoder solved the problem for me when outputting a wav file through "WAV Dest -> FileWriter". Unfortunately BeHappy doesn't work with a graph "DTS/AC3 Source -> Sonic HD Audio Decoder -> ffdShow". It complains about not being able to find a class or something like that. If I remove ffdShow from the graph, BeHappy works, but then only 2 channels are good.

Just as a warning to you guys. If it works for you that way, just BeHappy about that. But better make sure that you check whether all 6 channels are good. You can check that by converting the final AC3 file to 6 mono WAV channels by using Tranzcode and then by looking at the WAV files in Audacity.

[Deckard2019]

I tried SoundOut with this AVS :
DirectShowSource("aud.grf", video=false)
AudioDub(BlankClip(), last)
SoundOut()

Open it in VirtualDub and SoundOut pops out. Choose AC3 (Aften) and start.
But SoundOut crashes ...

[madshi]

@Pelican9,

here's a list of little bugs I found in your great EvoDemux tool:

(1) If I open the "file open" dialog to load a new EVO file - but abort the "file open" dialog, the EVO is read again. This should happen, aborting the "file open" dialog should behave as if I didn't even open the dialog in the first place.

(2) If I delete the EVO file which is currently selected by EvoDemux and then open the "file open" dialog and then abort the "file open" dialog, there's a crash and afterwards all buttons are disabled.

(3) If I rebuild the EVO with only 1 audio track, the other audio tracks are still listed in the EVO. It seems that their content is removed, but they're still listed in the header or something like that.

(4) If there's some kind of action running and I try to close the whole program, the closing stalls until I press the "Cancel" button. Trying to close the whole program should either automatically cancel the running action. Or alternatively you could open up a MessageBox asking whether the running action should be cancelled.

Keep up the great work! Thanks also to Ron/drmpeg for the original demuxing code.

P.S: Does EvoDemux support being controlled by command line parameters yet? That would help a lot in automating things.

[Pelican9]

I think the time difference is caused by drop-frame timecode vs non-drop timecode. So nothing wrong at here.

You may find this article helpful for understanding.
http://www.csif.org/html/dropframe.html

Thanks,
it was really helpful for me.

[Pelican9]

(1) If I open the "file open" dialog to load a new EVO file - but abort the "file open" dialog, the EVO is read again. This should happen, aborting the "file open" dialog should behave as if I didn't even open the dialog in the first place.

(2) If I delete the EVO file which is currently selected by EvoDemux and then open the "file open" dialog and then abort the "file open" dialog, there's a crash and afterwards all buttons are disabled.

(3) If I rebuild the EVO with only 1 audio track, the other audio tracks are still listed in the EVO. It seems that their content is removed, but they're still listed in the header or something like that.

(4) If there's some kind of action running and I try to close the whole program, the closing stalls until I press the "Cancel" button. Trying to close the whole program should either automatically cancel the running action. Or alternatively you could open up a MessageBox asking whether the running action should be cancelled.

Thanks, I'm working on it...

Edit:
1, 2, 4: Fixed.
3: Could you send me the status window of EVOdemux?
For me it's working perfectly

[zgx]

(3) If I rebuild the EVO with only 1 audio track, the other audio tracks are still listed in the EVO. It seems that their content is removed, but they're still listed in the header or something like that.I am getting other bug when it comes to audio streams. I have played around alot with my test movie Batman Begins. EVOdemux detects:

"VOB number 4 contains 2 video , 5 audio and 4 subpicture streams."

It contains three DD+ tracks and one Dolby TrueHD track. I select the main video stream, one of the DD+ tracks, the Dolby TrueHD track and a subtitle and press rebuild.

The new rebuilt file contains all the selected streams (as it should) but if I run EVOdemux on the new file it cannot find the Dolby TrueHD track but it's there as the file size is much larger then if I do not select it.

[puppydg68]

thanks to the efforts, input, arguments and collaborations of everyone on this thread, i've managed to successfully create an unadultered .wmv (playable on a 360) from a vc-1 .evo. it's only video, but baby steps always get you to where you're headed. all that remains is getting the damn audio downmixed and muxed. i'll be following closely until we get it all hammered out. thanks especially to Penguin9, drmpeg and Isochroma.
:thanks:

Could you please share this guide on getting the VC-1 video muxed unadultered into a Xbox 360 .wmv playable format?
It would be great if you could post it here http://forum.doom9.org/showthread.php?t=121168 or as an addition to this thread. Thanks in advance.

[juhu]

as noticed 2 posts above by ZGX, evodemux gives curious reports on rebuilt .evos

Once again : Casablanca.
Builiding one single evo with the 2 parts gives me this :

--------------------
Opening file Feature_1_Feature_2.rebuilt.EVO
File size: 13851 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
Substream id = 55
Sequence Header found
0 frames before first I-frame
Advanced Profile
Level = 3
Chroma Format = 4:2:0
Size = 1920x1080
Display Horizontal size = 1920
Display Vertical size = 1080
Aspect ratio = 1:1 (square samples)
Frame Rate = 23,976 (24000/1001)
(audio/sub parts removed)

147590 counted frames (1:42:35.733) in video stream 0.
73134 calculated frames.

in other words, it gives me stats from the first part, even after appending the 2nd (apart from the counted frames, but we know it's much less useful than calculated frames...)

[_xxl]

I test AC3 via FFDshow, and it works...but not for long DD+ (only for unilogo.ddp).

It works or not?
that from FFDShow is totally crappy
?
LFE, SL and SR are totally different than original source
I would like to know how much different?
1%,30%,80%,100%?
Any samples?

[Deckard2019]

Hi folks,
after experimenting few hours, I found this working procedure to convert DD+ to anything else:

So here is updated Guide (old guide can't go over 4G barrier):
1) Create Graph filter: DTS/AC3/DD+ Source -> Sonic Cinemaster Audio Decoder 4.2 -> Dump (for DTS/AC3/DD+ open demuxed DD+ track, for Dump open output RAW (test.raw)). This grab RAW PCM data from Sonic decoder and put into file.
2) Pres play to render RAW PCM data. This rawdata file is depend on Sonic setting, so make sure you set 5.1 output. You've get 5.1ch 16bit (depends maybe on soundcard type), 48khz 6ch RAW data file without header. (it is critical to use Dump filter,coz it can take You over 4G barrier...which file writer evidently not). Wait until filesize of rawfile stop raising, than you can stop graphedit.
3) Use SoX to convert PCM to WAVe: sox -r48000 -t .raw -c 6 -w -s test.raw x:\test.wav (try set output to another physical disk, it speedups the process)
4) If you want 6 separate mono WAVes (for SurCode or other like-this stuff), use Wavewizard. In preferences sets Ignore size in header -> always, Stream manipulation -> Mono streams, Output format -> Wave PCM and finaly output dir. Open test.wav and press "Convert". It creates six WAVes 0-1-2-3-4-5 which match these channel order: FL-FR-C-LFE-SL-SR.
5)Encode final WAVes to anything you want.... for Besweet You need create .mux file for six mono WAVes (for AC3 encodes, etc), it contain lines like this (for AC3 encode):
x:\test_ch0.wav
x:\test_ch1.wav
x:\test_ch2.wav
x:\test_ch3.wav
x:\test_ch4.wav
x:\test_ch5.wav

Note 1: In my test, SurCode is crappy encoder, it can't handle 48khz WAVes correctly...I recomennded use other DTS encoder.

Note 2: Things You need:
- Graphfilter: http://www.doom9.org/Soft21/Filters/graphedit.rar
- SoX: http://sox.sourceforge.net/
- Dump ds filter (in graphedit package), use "register.bat".
- Wavewizard: http://www.rarewares.org/wavewiz/wavewizardv0.54b.zip

Note 3: First of all, prepare a lot of disk space: 2 hours of 6ch PCM or WAVe takes aprox 4G disk space. So You need double (for PCM and for WAVe).

EDIT: Channel order for BeSweet corrected!

I've just tested this on the same 2h20 movie. test.wav size is 7GB, just like test.raw.
But absolutely no sound in test.wav :( And duration is 1h23 :(

It works or not?
Well it works for me. AC3 file sounds ok to me.
I would like to know how much different?
1%,30%,80%,100%?
Any samples?
+1

[madshi]

3: Could you send me the status window of EVOdemux?
Before or after the rebuild? Or maybe after reading the combined EVO?