Pulling my hair out on deinterlacing from Amarec..

[ugzz]

Sorry for another new thread here on a topic i've definitely seen. I've read a ton of threads on the matter but i'm just not getting the results i'm looking for.
I'm not a total n00b, but it's been a few years since I've done anything, so most of my limited knowledge is out dated. Maybe i'm still doing things an old way even with some research too.

Here's my workflow and my problem.
I have an elgato video capture, the old one with rca/ svideo. The baked in software record was trash. So after a ton of searching here I landed on Amarectv. I got all the settings there and am using h.264. I'm at a point where using a crf of 21 I'm getting a capture that's usable and leagues better than the el gato software was getting me. However Amarectv's deinterlacing leaves a lot to be desired. So I decided to disable it and just try and go from here. My capture is interlaced, and it looks good, interlacing aside. Everything is sync'ed up no problem and it plays start to finish no problem. IF i could accept Amarectv's deinterlacing, i'd be done here, but I really don't like it. I also have Adobe Premiere, and I think it's deinterlacing feature is even worse than Amarec's. So instead of being done, i want to go one more step.

I did a ton of testing and reading up and landed on using ffmpeg in conjunction with Avisynth+ and AvsPmod. I started with QTGMC but ended up switching to yadif because I was getting similar results at twice the speed. Problem is no matter what I do, i'm getting video drift. If i compare my Amarec master file with the ffmpeg output, the audio matches exactly, but the video doesn't. It goes a few frames off after about 15 min. Lip sync is pretty noticeable by 18-20 min, and anything past that is just bad, near the hour mark is seconds off. Amarec doesn't seem to have a way of getting me a cfr that i'm seeing. I've been messing with the output by switching frame rates around but that doesn't seem to do anything. I really like using Amarec and i'm happy with the files, vfr aside.. So i'd like to keep using it if there is a way to do this. Also i really like the 60fps output when deinterlacing, so again i'd like to keep that in my endgame if possible.

So the questions is.. what's my next thing to try? Or maybe this is something obvious and i'm just missing it? Any help is appreciated.

additional info and my stupid simple scripts:
JVC-SRV101 to capture via svideo, TBC on
AvsPmod:
FFmpegSource2("amarec(20200809-0037).avi", atrack=1)
ConverttoYV12()
AssumeTFF()
yadifmod(mode=1, edeint=nnedi3(field=-2))
BilinearResize(720,540)
Crop(2,0,-4,-2)
Prefetch(14)

ffmpeg:
ffmpeg -i "deinterlaceyadif.avs" -vsync 1 -crf 21 -c:v libx264 -preset: slow -c:a copy "yadif.avi"

[davexnet]

Assuming your source is 29.97 fps,
you can add

fpsnum=30000, fpsden=1001

to the ffmpegsource2 line to give a constant bitrate

[ugzz]

Well its old home video, but yeah, if i do a raw it does show up as 29.97, but with amarec everything keeps showing variable.
..
Huh.. So running that command does ABSOLUTELY fix the sync! Fantastic!
This would be Hyper awesome.. However, Now i have a ton of stutter/ judder, whatever you want to call it. It's kind of all over the place. Some scenes have constant stutter, others may go a second or two perfect, then start. I mean it's not.. the worst thing i've ever seen, but it's certainly not great. Is there a way to compensate? Or maybe even crush it back to 29.97, which could "maybe" reduce the stuttering? I love the 60.. to the point where i was SOLD as I definitively wanted it. However if this is the trade off, i might be willing to bail on it since its stutters so much.

[TimA-C]

I may have missed it in your original post, what codec are you capturing to in AmarecTV? If you're capturing in H.264 then have you checked that you're not overloading your system during capture? Have the task manager open & visible during a test capture of a scene with lots of movement and just check that there's plenty of available overhead on both CPU usage and hdd activity. Some example clips of your captures would be helpful as well as your capture settings in AmarecTV. Also, which version of AmarecTV are you using?

[ugzz]

System overhead is no problem. It only peaks around ~30%. When i'm capturing i make sure to close down all the system "crap" and I don't use the machine. I have a 9900k.
When just running the command without the string davexnet posted, my video is butter smooth 60 and awesome, but worthless because it falls out of sync.
AmaRecTV 2.31
-
Graph1
V- 720 x 480 29.97 YUY2
A- 48000, 16bit
Graph2
4:3
Underscan
Deinterlacing: not use
graph3: default
preset: default
Recording:
Frame Rate: 30
Filter processing: filter off
Video Codec X264
Audio: uncompress
Advanced:
all at default. I see a box for "match the start timing of the audio with video", i don't exactly follow that, but it's unchecked.

H264 settings:
Slow Preset / Film tuning / Profile and Level- Auto
Zero Latency - Checked (I get an error unless i check this)
CQP Single Pass: 21
Output: VFW: H264

I can try and pull some clips later today and get them on drive. I'll need to pull a clip out from the middle of an hour long video to show the sync issues so i'll need to do a couple of exports so it'll take a few.
Appreciate the assist here.

[davexnet]

For further analysis provide a minimum 30 second clip direct from your capture,
something showing some steady movement

[TimA-C]

There is a later version of AmarecTV (v3.10?) available on the Japanese site - put your mouse cursor on the various links on the website and your browser should show you what the link is in english. DON'T download any versions that are v4 or later as you'd need to buy their bespoke codec to capture anything. (I can't access my capture PC right at this moment to check on the version no. but I'm fairly sure it's 3.10)

Do you still have these issues if you capture to a lossless codec (Lagarith, HUFFYUV, UT Video etc.) and then encode to H.264?

[ugzz]

So here are some motion clips. There's no obvious sound markers on screen so you can't really tell the audio is out. Also this is only 4 min in, and the audio isn't super terrible yet, it gets way worse a few more minutes in. Also The Amarec Recording and new yasdif with the recommended framerate setting recording are ripped exactly at 4:20 (no i didn't do that on purpose and just thought of it now), and the QTGMC one is 4:20:250. As the video was a quarter second off, even at the 4 minute marker. I would have left it right at 4:20 so you could see it was actually a different scene, but it wasn't a frame i wanted to put online.

All of these were ripped with avidemux so nothing was re-encoded so they should be direct rips, although avidemux does put them in an mkv container.

link removed, updated link below with new method

As for attempting a lossless capture, no i havn't tried that. I'm not super interested in messing with it honestly. I'm on smallish ssd drives and loving these file sizes. I guess it could be worth a shot though if i run out of options.
Cool to hear theres a newer Amarec though, I'll check that out. Although if it's deinterlacer isn't any better I assume i'll have the same workflow and have similar issues, but it's totally worth a shot. If it's built in deinterlacer is better.. maybe it could be good enough to just one shot it.

[TimA-C]

I was thinking the lossless capture more as a diagnostic than necessarily using that the whole time. Having said that, the cost of larger SSDs has come down hugely over the last couple of years and, with your setup, your captures really shouldn't take too long to process . . . I'd still try a smallish capture of a scene that you already know well, and see how it compares? Also, have you tried checking that "match the start timing of the audio with video" box?

[ugzz]

That's just in AmarecTV though right? My recordings from Amarec are all great, no audio problems at all. It's just after deinterlacing that the audio / video desync. Could doing this help? I can totally try it.

[TimA-C]

Nothing ventured etc. etc. etc. TBH I wouldn't capture to H.264 if I was doing any additional processing and then re-encoding to H.264 again . . .

[davexnet]

Capturing AVC with interlaced footage marked as progressive is not the way to go, HuffyUV, UT video even MJPG at high quality would be better

[ugzz]

I'm well aware it's not the normal way of doing things encoding it twice, but i'm liking the quality well enough, but of course I am having these issues.
I'll grab a commercial tape and run a couple captures and see what they look like. Using that tick box, and also trying a different codec. Then when i process them i'll see if i have the same problem.

Heres another thing I just got.
Trying to put them into AVIdemux to do a direct rip without re-encoding, i see this:
This video contains B-frames, but presentation time stamps PTS are either missing or monotonically increasing.

I'm not sure what this means, but could it identify some type of problem?

thanks

Oh, and for the record, these are old home movies that were taped on crappy cameras of the day, and most using longplay settings for even more reduced quality.
So the source is pretty meh, that's why i was doing everything in h264 since with a setting of 21 or 22 on CWP is giving me quality that looks acceptable enough to the original quality. Doing lossless stuff and getting 100GB files and running them side/ side with my 2GB files doesn't really look any different. Again though, if doing it fixed my problems it could still be worth it, It will just be the difference between archiving the captures or not.

[davexnet]

I'm well aware it's not the normal way of doing things encoding it twice, but i'm liking the quality well enough, but of course I am having these issues.
I'll grab a commercial tape and run a couple captures and see what they look like. Using that tick box, and also trying a different codec. Then when i process them i'll see if i have the same problem.

Heres another thing I just got.
Trying to put them into AVIdemux to do a direct rip without re-encoding, i see this:
This video contains B-frames, but presentation time stamps PTS are either missing or monotonically increasing.

I'm not sure what this means, but could it identify some type of problem?

thanks

Oh, and for the record, these are old home movies that were taped on crappy cameras of the day, and most using longplay settings for even more reduced quality.
So the source is pretty meh, that's why i was doing everything in h264 since with a setting of 21 or 22 on CWP is giving me quality that looks acceptable enough to the original quality. Doing lossless stuff and getting 100GB files and running them side/ side with my 2GB files doesn't really look any different. Again though, if doing it fixed my problems it could still be worth it, It will just be the difference between archiving the captures or not.

The point about capturing lossless is that it is saving the exact data that came out of the encoder chip on the capture device.
Then you encode it to your target format
I've seen that message from AVidemux before it always meant there was a problem with the source file,
but I don't remember the details, something about time stamps and B frames

How did you create a file that had this problem?

[ugzz]

That was the file output by using the strings and commands i put in the OP.

[davexnet]

That was the file output by using the strings and commands i put in the OP.

I see that. Something in your encode is bad. What program did use and what were the settings?
Whatever went wrong is the cause of it being jerky and giving that message in Avidemux

Here's one I did using your AmarecTV source and a script similar to yours:

LoadCPlugin("I:\program files\AVStoDVD_289_Alpha190616_NoInstall\Lib\ffms2.dll")
Load_Stdcall_plugin("J:\StaxRip_1.1.7.2\Applications\AviSynth plugins\Yadif\Yadif.dll")
import("I:\program files\AVStoDVD_288_NoInstall\Lib\ffms2.avsi")
FFmpegSource2("C:\Users\davex\Downloads\amarectv record.mkv", atrack=1)
ConverttoYV12()
AssumeTFF()
yadifmod2(mode=1, edeint=nnedi3(field=-2))
#BilinearResize(720,540)
#Crop(2,0,-4,-2)

[TimA-C]

Lossless here doesn't necessarily mean without any compression. Most of the codecs mentioned here use non-destructive compression - unlike H.264 or Mpeg2. A recent 2hr 52min capture to Lagarith (YV12) is 40.11Gb. Capturing to lossless at least gives you a chance of getting the best results from your source video . . .

[ugzz]

That was the file output by using the strings and commands i put in the OP.

I see that. Something in your encode is bad. What program did use and what were the settings?
Whatever went wrong is the cause of it being jerky and giving that message in Avidemux

Here's one I did using your AmarecTV source and a script similar to yours:

LoadCPlugin("I:\program files\AVStoDVD_289_Alpha190616_NoInstall\Lib\ffms2.dll")
Load_Stdcall_plugin("J:\StaxRip_1.1.7.2\Applications\AviSynth plugins\Yadif\Yadif.dll")
import("I:\program files\AVStoDVD_288_NoInstall\Lib\ffms2.avsi")
FFmpegSource2("C:\Users\davex\Downloads\amarectv record.mkv", atrack=1)
ConverttoYV12()
AssumeTFF()
yadifmod2(mode=1, edeint=nnedi3(field=-2))
#BilinearResize(720,540)
#Crop(2,0,-4,-2)

Ugh.. Trying to re-create this has put me into a bit of a rabbit hole.
So I can't use yadifmod2 since everything I have it 32 bit. I started upgrading things x64 but i'm still getting errors and I think the problem is AvspMod, and i don't think it's been updated in like 4 years.
If i just use yadifmod, i can basically copy / paste your stuff (minus your paths) and get the same result, and it's fine for a 30 second clip. It's only when i exceed like ~15+ minutes that I start to lose the sync. It's identical to what happens when I use QTGMC, The video seems to just keep sliding ahead while the audio is a perfect copy.

I'm going to go ahead and try a different codec out of Amarec with a new tape and see how that goes. I grabbed a movie for testing so i should get lots of action and can pull clips easily.

Is it too much to ask for help in getting something that works with x64 up and running? I'd like to try yadifmod2, but i think i need an alternate to avspmod, i like it for the preview feature, but it's not necessary. So i'm totally open to using something else as long as it has similar usability.

edit: Oh and I did upgrade to Amarectv 3.1, I'm not seeing a ton of difference but i did notice the addition of a 999 framerate mode. I ran a deinterlace test and it's just as worthless.

[poisondeathray]

I started upgrading things x64 but i'm still getting errors and I think the problem is AvspMod, and i don't think it's been updated in like 4 years.

https://github.com/gispos/AvsPmod/releases

Graph1
V- 720 x 480 29.97 YUY2
Frame Rate: 30

Is there an option to set the framerate to 29.97 ?

If you recorded YUY2 4:2:2 , ConverttoYV12() should be ConvertToYV12(interlaced=true) , otherwise you will get chroma artifacts. It looks like you used x264vfw , but what settings did you use for the pixel format ? "amarectv record.mkv" is 4:2:2 so I'm assuming that's what you used, so you'd want to use ConvertToYV12(interlaced=true)

Another weird observation. The MKV has an "original frame rate" entry of 25fps

Frame rate : 29.970 (30000/1001) FPS
Original frame rate : 25.000 FPS

What does mediainfo (view->text) say about the AVI ?

What does info() report about ffms2 as the source filter with the original AVI in the script, in terms of duration, fps ?

FFmpegSource2("amarec(20200809-0037).avi", atrack=1)
Info()

What about the actual recorded AVI ? Does that play in sync in a media player e.g. vlc, mpchc, mpv ?

Did you try another source filter? AVISource() ? Check with Info() too

AVISource("amarec(20200809-0037).avi")
Info()

You can also check for sync by playing back the script when you make changes. If yadifmod is too slow on your computer, you can temporarily replace it with yadif(1,1)

[ugzz]

[QUOTE=poisondeathray;2591957]

https://github.com/gispos/AvsPmod/releases

Wow, I didn't even know this existed. Awesome!

Is there an option to set the framerate to 29.97 ?

If you recorded YUY2 4:2:2 , ConverttoYV12() should be ConvertToYV12(interlaced=true) , otherwise you will get chroma artifacts. It looks like you used x264vfw , but what settings did you use for the pixel format ? "amarectv record.mkv" is 4:2:2 so I'm assuming that's what you used, so you'd want to use ConvertToYV12(interlaced=true)

No, Amarec only has 30, 60, 999 and when you watch the preview as it records, i can see the framerate jumping from 29-31, but the preview settings don't always match the capture. Either way, I have concerns that even though MediaInfo shows 29.97, that it's somehow still a variable..
Also, i went ahead and added the convert settings you recommended.

Instead of answering your questions about the files, i think reading all the comments here has me thinking I'm fighting too many battles at once.

I'm upgrading to the latest x64 AVIsynth, the updated x64 version of Avspmod, i'm updating to the x64 newer yadifmod2, and i'm going to bag the old Amarec and just use the newer version.
Using this combo, i'm going to recapture a video from AmarecTV 3.1 using lossless lagarith, and i'll run a long 2hr vid.
I'm going to use the Match start timing of audio and video checkbox, and pull a new interlaced source.
Then i'll run the new command using the newer versions of things, and see what I get. for comparison i'll do one with the framerate forced, and one i'll just leave at default.
I'll post the results.
Also I do appreciate your help and looking at the files, even if i am going to change up a bit

[davexnet]

That was the file output by using the strings and commands i put in the OP.

I see that. Something in your encode is bad. What program did use and what were the settings?
Whatever went wrong is the cause of it being jerky and giving that message in Avidemux

Here's one I did using your AmarecTV source and a script similar to yours:

LoadCPlugin("I:\program files\AVStoDVD_289_Alpha190616_NoInstall\Lib\ffms2.dll")
Load_Stdcall_plugin("J:\StaxRip_1.1.7.2\Applications\AviSynth plugins\Yadif\Yadif.dll")
import("I:\program files\AVStoDVD_288_NoInstall\Lib\ffms2.avsi")
FFmpegSource2("C:\Users\davex\Downloads\amarectv record.mkv", atrack=1)
ConverttoYV12()
AssumeTFF()
yadifmod2(mode=1, edeint=nnedi3(field=-2))
#BilinearResize(720,540)
#Crop(2,0,-4,-2)

Ugh.. Trying to re-create this has put me into a bit of a rabbit hole.
So I can't use yadifmod2 since everything I have it 32 bit. I started upgrading things x64 but i'm still getting errors and I think the problem is AvspMod, and i don't think it's been updated in like 4 years.
If i just use yadifmod, i can basically copy / paste your stuff (minus your paths) and get the same result, and it's fine for a 30 second clip. It's only when i exceed like ~15+ minutes that I start to lose the sync. It's identical to what happens when I use QTGMC, The video seems to just keep sliding ahead while the audio is a perfect copy.

I'm going to go ahead and try a different codec out of Amarec with a new tape and see how that goes. I grabbed a movie for testing so i should get lots of action and can pull clips easily.

Is it too much to ask for help in getting something that works with x64 up and running? I'd like to try yadifmod2, but i think i need an alternate to avspmod, i like it for the preview feature, but it's not necessary. So i'm totally open to using something else as long as it has similar usability.

edit: Oh and I did upgrade to Amarectv 3.1, I'm not seeing a ton of difference but i did notice the addition of a 999 framerate mode. I ran a deinterlace test and it's just as worthless.

I forgot to mention that version of yadifmod2 *is* 32 bit, and I encoded it with the 32-bit (as you'd expect) version of Virtualdub2

[ugzz]

That's fine, i went ahead and updated everything. I've never done any of this stuff on x64 and didn't really think that I'd have to track down x64 versions of "everything" and their dependencies. Had some hiccups but finally figured it all out.

So i'm getting basically the same thing. I added a native capture from the el gato software also for comparison. If it weren't that it spits a more washed out video with some artifacting, it's deinterlacer actually looks pretty great for a 1 pass. I did some testing with it, it's not just washed out either, there is actually less detail. I had a scene with green grass and you couldn't really even see the shadows, even if you over saturated it, but in a recording from AmarecTV the shadows were there.

https://drive.google.com/drive/folders/16YE1YZ0JkHmUsSayRozLN-J36YC2EtcA?usp=sharing

The files are:
Elgato straight cap (the program has no settings, as it's made to just be "easy")
AmarecTV 3.1 cap with no deinterlacing, using lagarith, the 30fps setting, and the Match the start timing of the audio with video box. Also i just noticed "use null frames" is checked, I didn't check that, so it must be a default on 3.1 that wasnt set on v2. Maybe that should be off, but regardless i'm getting the same results as i was with v2.
An Adobe Premiere deinterlace, i double checked that its set to top first and the settings are good, and that result is just... out of this world terrible. Also premier has a very different idea of timestamps than avidemux, because i used the same ones and hand typed them in.
Yasdif2x64 run with h.264 and no frame rate settings (audio sync just like before)
Yasdif2x64 run with h.264 and the framerate setting locked in (stuttering is back just like before)

Guess my next thing to try and to drop h.264 altogether and try running yasdif lossless also. If that fails, then i guess AmarecTV is just giving me garbage files..
I also realized I'm getting that B frame error on AVIDemux with both the output files from ffmpeg, the one with and without the frame rate settings.

If you go frame by frame in the yasdif files, you see some pretty interesting stuff. On the one with no framerate i'm seeing a pretty consistant: Frame, Dupe Frame, Frame, Dupe Frame, Dupe Frame, Frame pattern. and on the one that has the framerate set. I'm seeing dupe frames and also previous frames where the skips are. So it will go Frame, Dupe Frame, Frame, Previous Frame, Dupe Frame, Frame, and somtimes i see even 3 dupe frames in a row. Watching them the one without framerate set still looks pretty smooth, even given the odd behavior. I also went frame by frame on the amarec cap, and at least the 3 spots i checked were all good, each frame was different.

I mean if nothing else, this has been.. interesting. lol

[jagabo]

Your video is telecined film and should be inverse telecined back to 23.976 fps, not deinterlaced. In AviSynth:

LWLibavVideoSource("AmarecTV 3.1 lagarith (big).mkv") 
AssumeTFF()
TFM()
TDecimate()

You'll need LSMASH source filter and the TIVTC package, both for AviSynth.

The levels need adjustment too. And turn off all the noise reduction filters in your capture chain. All they do is eliminate detail. AviSynth has much better denoisers than any hardware.

Above script encoded with x264 CLI (video only)

[jagabo]

And just for kicks, with just yadifmod2:

LWLibavVideoSource("AmarecTV 3.1 lagarith (big).mkv") 
AssumeTFF()
ConvertToYV12(interlaced=true)
YadifMod2()

It really should be decimated to 23.976 fps too (at 29.97 fps every 5th frame is a duplicate) but I left that out for this example.

[ugzz]

Your video is telecined film and should be inverse telecined back to 23.976 fps, not deinterlaced. In AviSynth:

LWLibavVideoSource("AmarecTV 3.1 lagarith (big).mkv") 
AssumeTFF()
TFM()
TDecimate()

You'll need LSMASH source filter and the TIVTC package, both for AviSynth.

The levels need adjustment too. And turn off all the noise reduction filters in your capture chain. All they do is eliminate detail. AviSynth has much better denoisers than any hardware.

Above script encoded with x264 CLI (video only)

A few days ago i would not have been competent enough to do this.
As it stands, I got the packages you noted, and just made the AVS and a ffmpeg script to run it, and it worked first try. Nice, big thanks to everyone in the thread for just sending me stuff to try. I'm learning a bunch.
It looks good, and i'm not seeing any frame dupes.

I have the same problem when doing some home movies too, but i can't imagine they are telecined. I may have shot myself in the foot by switching the thread after starting with a home movie and going to this. Since I'm on 100% new software at this point i'll need to pull a new one and test it out. At least now i have like 3 different things to try . I know some of them are definitely 29.97, so the way you did it in ffmpeg would be better since that doesn't flip the framerate.

Also, it works great on the clip, but i wonder if i run this on the full 2hr video, if it loses audio sync like the deinterlacing scripts do.

edit: So jagabo, I'm a little confused as i pulled a clip of a home video, and i ran a deinterlace, and i ran your yadif above without deinterlace and with just ConvertToYV12(interlaced=true). I have them both full screen zoomed and I honestly can't really tell much of a difference in quality. I'm not sure why that is, but anyway the main difference i'm seeing is just that one is 29.97 and the other is 59.94. I went frame by frame zoomed in and there's a few frames where the deinterlace script look better, but theres a few frames where just the converttoyv12 interlaced=true look better. So it's basically a wash..
It would have been nice to keep the 60fps, but if having the framerates all match keeps my audio in sync i'll be happy. Time to run a long one and see how it goes.

[jagabo]

Use Yadif(mode=1) if you want 60p output.

What program did you use to encode? Maybe it saw the comb artifacts and automatically deinterlaced. Upload your encoded videos if you want further analysis.

[ugzz]

I duno wtf happened but i submitted replies to this thread.. TWICE! Neither of them are showing! Hopefully this one works!
Anyway.. So I ran the telecined script on the full 2hr 15 min video and it gets in and out of sync EXACTLY the same as the deinterlace. Like put them side/ side and the drift is idential. The clip i uploaded that has the gun fire out of sync, "yasdifx64 no framerate".. it's 100% identical. In the clip I show you can see how terrible the gun is out of sync compared to other other vids. It's weird, 10 min after the original is "mostly" in sync. it's like maybe 1/8th second off. And maybe 20 min before it's the same, maybe 1/8th off. and as you watch the 2hr 15min video it seems to go in and out of sync. Sometimes it's perfect, other times it's nearly a full second out of sync, but then just 5 or 7 min later it's not so bad again. I've never seen this behavior. I can say that at 20 min it's fine. At 35 there is some sync out, at 50 it's not terrible again, and at 1hr5 its horrible, like .5 second out, then it sort of gets OK again. At the point i included in the clip, you can see how far off the gun is. Also.... There's no way that's CZJ in that clip.. that has to be a stunt double firing that gun lol. Anyway, in the OG files the audio is perfect, but in any encode, it's awful.

I went ahead and thought maybe it was AmarecTV so I figured out direct ffmeg capture. Since the capture card requires you to select the video feed for audio capture, and ffmeg won't allow that. I tried just using my Line In on the pc for audio and the video in. I got it to work and can capture direct from ffmpeg. I had an identical problem. Deinterlacing or fixing the telecined produce identical results. The audio / video is out of sync in the exact same spots. I even tried capturing RAWVIDEO, which is more than double that of huffy.. Same results again! So i guess at this point I can confidently Say it must be my capture card. I think i have no choice but to revert to my capture card software and accept the low quality and try and fix saturation in post. OR i'll have to figure out how to tell ffmped to cut every 20 min and start a new file, then just merge them all together. That way the sync should be acceptable throughout.

I also tried deinterlacing and/or telecined fixing with huffy, same results, so it's not x264 that's the problem. It seems to be the encode or something. No matter what i use, i always end up with sync issues.

[davexnet]

Your source "AmarecTV 3.1 lagarith (big).mkv"
looks better if you inverse telecine and decimate to recover the original 24 (23.976) frame rate

vid=lwlibavvideosource("C:\Users\davex\Downloads\AmarecTV 3.1 lagarith (big).mkv")
aud=lwlibavaudiosource("C:\Users\davex\Downloads\AmarecTV 3.1 lagarith (big).mkv")
vid=vid.tfm().tdecimate()
audiodub(vid,aud)
convertToYv12().spline64resize(640,480)

[jagabo]

The audio is out of sync in your lagarith sample. So of course it's still out of sync after encoding.

[ugzz]

The audio is out of sync in your lagarith sample. So of course it's still out of sync after encoding.

Nah, the audio is good on the amarec file. even in the full 2hr15, never looses sync anywhere. I went through it pretty thoroughly. if your looking at the lips when she talks that's out on the source, sounds like a bad ADR to me.