Audio sync issues

[OutsidaII]

I created the d2v file straight from the vobs and the ac3 file and the video are exactly the same length, but for some reason when i run the ac3 through BeSweet to make the mp2 1 extra second is being added on and causing major sync issues to this video. I've been working on this problem for awhile so any input would be greatly appreciated.

[DSPguru]

:logfile:

[OutsidaII]

I didn't have a log originally, and here's the one after i redid it, this time there's not extra second but it's still .08 longer than the video.

[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : C:\esca\esca AC3 T02 3_2ch 448Kbps DELAY -188ms.ac3
[00:00:00:000] | Output: C:\esca\esca188.mp2
[00:00:00:000] | Floating-Point Process: Yes
[00:00:00:-661] | Overall Track Gain: 9.916dB
[00:00:00:-661] +-------- AZID -------
[00:00:00:-661] | Input Channels Mode: 3/2, Bitrate: 448kbps
[00:00:00:-661] | Output Stereo mode: Dolby surround compatible
[00:00:00:-661] | Total Gain: 0.000dB, Compression: None
[00:00:00:-661] | LFE levels: To LR -3.0dB, To LFE 0.0dB
[00:00:00:-661] | Center mix level: BSI
[00:00:00:-661] | Surround mix level: BSI
[00:00:00:-661] | Dialog normalization: No
[00:00:00:-661] | Rear channels filtering: No
[00:00:00:-661] +------ Shibatch -----
[00:00:00:-661] | Source Sample-Rate: 48.0KHz
[00:00:00:-661] | Dest. Sample-Rate: 44.1KHz
[00:00:00:-661] | Attenuation : 0.0db
[00:00:00:-661] +------- MP2ENC ------
[00:00:00:-661] | Bitrate method : CBR
[00:00:00:-661] | MP2 bitrate : 224
[00:00:00:-661] | Channels Mode : Stereo
[00:00:00:-661] | Error Protection: Yes
[00:00:00:-661] +---------------------
[03:04:03.904] E7: CRC error
[01:32:01:291] Conversion Completed !
[01:32:01:291] Actual Avg. Bitrate : 224kbps
[00:20:50:000] <-- Transcoding Duration
Logging ends : 01/15/03 , 17:03:51.

[DSPguru]

How To Submit A Bug Report (http://forum.doom9.org/showthread.php?s=&threadid=15738)

[OutsidaII]

I believe this meets the specifications. Hopefully you can help me with the problem i'm having. BTW this is a new log because something happened to the other one.

BeSweet v1.4 by DSPguru.
--------------------------
Using azid.dll v1.8 (b825) by Midas (midas@egon.gyaloglo.hu).
Using Shibatch.dll v0.2 by Naoki Shibata & DSPguru (shibatch.sourceforge.net).
Using MP2enc.dll v1.129 (23/5/2002), Engine 1.129 <http://www.cdex.n3.net>.

Logging start : 01/16/03 , 02:38:22.

C:\Documents and Settings\KII\Desktop\SVCDrippack\04-BeSweetGUIv0.6b61\BeSweet.exe -core( -input C:\esca\esca AC3 T02 3_2ch 448Kbps DELAY -188ms.ac3 -output C:\esca\esca-188.mp2 -logfilea C:\Documents and Settings\KII\Desktop\SVCDrippack\04-BeSweetGUIv0.6b61\BeSweet.log ) -azid( -L -3db --maximize ) -ota( -d -662 -g max ) -ssrc( --rate 44100 ) -2lame( -m s -b 224 -e ) -profile( myprofile )

[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : C:\esca\esca AC3 T02 3_2ch 448Kbps DELAY -188ms.ac3
[00:00:00:000] | Output: C:\esca\esca-188.mp2
[00:00:00:000] | Floating-Point Process: Yes
[00:00:00:-661] | Overall Track Gain: 9.916dB
[00:00:00:-661] +-------- AZID -------
[00:00:00:-661] | Input Channels Mode: 3/2, Bitrate: 448kbps
[00:00:00:-661] | Output Stereo mode: Dolby surround compatible
[00:00:00:-661] | Total Gain: 0.000dB, Compression: None
[00:00:00:-661] | LFE levels: To LR -3.0dB, To LFE 0.0dB
[00:00:00:-661] | Center mix level: BSI
[00:00:00:-661] | Surround mix level: BSI
[00:00:00:-661] | Dialog normalization: No
[00:00:00:-661] | Rear channels filtering: No
[00:00:00:-661] +------ Shibatch -----
[00:00:00:-661] | Source Sample-Rate: 48.0KHz
[00:00:00:-661] | Dest. Sample-Rate: 44.1KHz
[00:00:00:-661] | Attenuation : 0.0db
[00:00:00:-661] +------- MP2ENC ------
[00:00:00:-661] | Bitrate method : CBR
[00:00:00:-661] | MP2 bitrate : 224
[00:00:00:-661] | Channels Mode : Stereo
[00:00:00:-661] | Error Protection: Yes
[00:00:00:-661] +---------------------
[03:04:03.904] E7: CRC error
[01:32:01:291] Conversion Completed !
[01:32:01:291] Actual Avg. Bitrate : 224kbps
[00:21:29:000] <-- Transcoding Duration
Logging ends : 01/16/03 , 02:59:51.

[htc10825]

If the ac3 file has the crc error This meens mostly that there are read errors on dvd during rip or your comuter system has storage errors(chipset issue or overclocking). This will cause the incorrect decoding in time base of the ac3 file and thus a/v asyncron(after the error points).

[OutsidaII]

okay so here's the thing. this is the fourth time now that i've ripped this movie and had this problem. I use dvd2avi with nothing else running and i get my d2v and the ac3. I honestly don't know what else different to do to keep from getting this crc error. Any words of wisdom would be greatly useful.

[htc10825]

1. Check it with VDub(video=.d2v+avisynth, audio=the decoded wave file) to find out the asyncron point.
2. Set delay in audio/interleaving to get the best syncron(after the asyncron point).
3. Load the wave file into a wave editor(WaveLab/CoolEdit Pro) and move the audio data(after the asyncron point) a littel(using the value of VDub) to match the video, save it to a new file, encode to mp2.

[OutsidaII]

how do i find the asynchron point in virtualdub? i've been looking through other posts and stuff and i can't figure out how.

[htc10825]

1. read all the guides of VDub
2. read my prev. post carfully
3. report what you have done with all the source and VDub.

[OutsidaII]

I read all of the stuff about VDub on here and I tried the manual synching that it recommends. Since the sound is off from the very beginning I did the source displacement like it recommended and tried it and it worked on the sample i made. This doesn't help me though because i didn't find a specific asynchron point that i can use to fix in cool edit pro. My question is since virtual dub will cut off any audio in excess of the movie length, is it working because it's getting rid of the extra time at the end of the audio track causing it to synch correctly.

What was done with the source:
1. dvd2avi to get d2v and ac3
2. created a wav file with BeSweet with delay -188ms set by dvd2avi
3. Loaded the an avs with the d2v and the wav file.
4. Set the audio to interleave every 500 ms and delayed the audio track by 188ms

[htc10825]

first thing to do with VDub is changing the value(can be negative) of audio delay to get 1st A/V syncron point. Starting at 0.0 ms

[OutsidaII]

I realize that just telling me what to do exactly would accomplish nothing because then i would not have learned anything, but these vague instructions, while appreciated, just bewilder me even more. So are you saying don't run it through besweet with a set delay and let virtual dub do it? And i can't find in any guide anywhere on the web how to calculate an asynchron using virtual dub? Maybe i'm just slow but I've never had this much trouble with any movie ever.

[htc10825]

In my last post I wanted only say so:
In your (2nd)last post, the steps 1-3 are very well. In step 4., set the delay first at 0ms, the click play and watch if it syncron.
If not then estimate the real delay velue and its polarity. And then change it in the VDub.

Sometimes there is a intro in the 1st VOB which has no sound and the DVD2AVI cannt recognize it. So goto play the dvd and see the duration of this intro(press the "next"-key). Then set it in VDub, assume 36s, the delay at 36000ms.

Do not be impatient. Just report your progress and I'll help you throu the whole procedure.

[OutsidaII]

Alright so in VDub i set the delay to 0ms and made a sample clip and in the clip everything matched up. I'm not sure what you meant by watching if it synchron, but there was one part in the sample where the video kind of stuttered for a second. I'm not really sure what to make of this. I hope that i'm on the right track.

[htc10825]

Sorry to say that.
Let us do it from the beginning:
1. get a .d2v and .ac3 with Dvd2avi
2. create a wav file with BeSweet with delay -188ms set by dvd2avi
3. create an avs file containing the .d2v and nothing else
(except the loadplugin mpegdec.dll, of cause!)
4. load the an avs in VDub Set the audio delay first at 0ms. The values of preload and interleave are not important at all because we donnt use it.
5. click "play input" in VDub and watch if the A/V syncron, follow the advice of my last post, post your progress
It's wrong to create a clicp. Just "preview" the input stream.

[OutsidaII]

One quick question, all this time we've been talking about watching for the synchron, i just want to make sure i'm looking for the right thing, so if you could just clarify for sure what i should see if i find a one.

[htc10825]

if in step 5 the a/v syncron at the begining, but lost synchron after 40m35s120ms, and the "40m35s120ms" is the change point. This is your 1st asynchr point. Go to next step:

6. Estimate the time diff. of asynchr and set the value of audio delay in VDub. Repeat it till the a/v synchr after the 1st asynchr point. Write down the asyn-point and the value of delay.

7. unload the wave file in VDub, load the wave file in WaveLab(or your favourite weve editor) and add or delete a few time at the asyn-point with the value which you've got in step 6. Save wave.

8. reload the wave in Vdub and go on checking, if there is a 2nd asynchron point, determine it and repeat the step 6,7,8.

[OutsidaII]

When you do play input video you get no sound from virtual dub, at least I don't. Is there a setting that needs to be on to enable sound output during the preview video?

[htc10825]

you must load the wave file into VDub.
I read all of the stuff about VDub on here... It's not true...
You have to read more...

[OutsidaII]

I know that I haven't known some stuff but honestly do you think I would have post that I couldn't get sound if I hadn't looked around first to see what was going on. I loaded the avs file and then the wav file. I even checked to make sure that there was sound on the wav file, but still when i go to preview input or preview output video i get no sound. It honestly baffles me why it doesn't work.

[htc10825]

the wave file must be a "real wave file"(uncompressed), i.e. for 1 min movie the file size must be 10MB(44.1kHz) or 11MB(48kHz). A movie with 180min and 48kHz the audio file size must be near 2GB.
Check it plz.