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@ jagabo. thanks a lot for your mask but it seems that the bobed variant isn't the right solution for this case as you are left with basically all the crap frames doubled and they can't be fixed well. But for other cases which aren't a mix of 3 different formats I will give this another go.
@ hello_hello: that is some pretty impressive work you did there but working like this analytic section based "every section gets its own special treatment" kind of work would probably kill me in the long run/would take me forever to get that right for a whole movie like this so I would prefer it to be a bit more simple but never the less thanks for your hard work as it shows basically a lot of clever thinking how to deal with special cases like this in general.


that is pretty impressive I didn't notice any blending in the film sections so your dupe interpolation seems amazing.
Would you mind sharing the script you used for this conversion? So basically the 29,97 variant of your script in post #12
and no worries if its VapourSynth I can manage to re-write those to an AVS+ variant :-)

thanks again for all the help everybody, you guys are out of this world impressive.

You will laugh but that is the idea which came to me while asleep. I just played around with it and it looks pretty good regarding smoothness. However of course the stair pattern which we so greatly eliminated becomes more obvious again as we don't remaster any frames.
Doing some SeparateFields() and throw away the odd keep the even or throw away the even and keep the odd experiements only lead to worse results.
So if you have a clever idea how to fix the stair pattern (obvious in sample1: car and sample 2 rolling shutter) that would be awesome

it has to be progressive. yea my first idea was to interpolate the 23,976 back to 29,97 to match to the deinterlaced 29,97 parts but that of course only leads to stuttery playback as it can only duplicate frames in order to achieve the 29,97. So I guess my only chance is to either live with all the dropped frames in the ntsc video parts which of course are mostly noticeable in fast movements (the frame drops are the same with and without (pp=0) ) or to interpolate the ntsc video parts to 23,976 with the most clever interpolation around what ever that is as I never did that before and have a hard time understanding select every frame patterns :-/and after that cut them into the film

here is the sample of the telecine/interlace mix: https://workupload.com/file/gxYsu4xAczq

thanks hellohello and alex_ander for the input on how to deal with such cases.
The mkv samples I created with mkvtooolnix from the original dvd_vob source because I wouldnt know how else to make a lossless sample from a vob format DVD
Your suggestion that the broken parts of the samples jagabo mentioned are a result of a wrong placed cut by toolnix is probably right.
I'm working with the original VOBs here of course and I index them with MPEG2Source just as you describe.

I havent posted any of the actual telecine/video mix scenes yet but will do so in a couple of hours. thanks for your help and later

the project just turned into a nightmare. I just realized that it is actually a composite cut out of telecined film and fully interlaced video.
Before making hellohello's suggestion with the Trim section based filter work a reality I kindly ask if there is a way to detect the scene parts which are fully interlaced? So as an alternative to the "detect combing when everything is red" a detect the fully interlaced sections?

alright, great thanks a lot

thanks you guys for working on this so much it means a lot.

@Selur that worked awesome, very impressive. I used AssumeFieldBased() to replace the Vaporsynth SetFrameProps command as SeparateFields seemed wrong to me.
For santiag the options "nns=2, qual=2, pscrn=2" don't seem to be available in AVS or maybe they are there under different names?
Is there a santiag vaporsynth documentation somewhere so I could compare their setting to the ones from the avs wiki? Couldnt find anything and am curious if that would improve the results even more
thanks so much :-)

I didn't apply any resize to that video but didn't set the display aspect ratio to 4:3 when I encoded it as it was just a test so it automatically changed to 3:2 as it always does when doing that with 720,480 which only looks right with 4:3 dar.
I made another export with hellohellos suggested resize CropResize(0,0, 18,0,-12,0, ResizeWO=true, CropDAR=4.0/3.0, InDAR=15.0/11.0) but the problems remained. If you check out sample1 between 00:00:02 - 00:00:07 you'll notice the bug and it is expected that this will happen every time when that weird unstable stair pattern occours of this "film which was edited on video converted back to film + pulldown for DVD" mess...

Here are all 3 samples extracted from the original DVD with mkv toolnix which should be a lossless sample extract of the original DVD source to work with: https://workupload.com/archive/pjqNBpCjP9
thanks

Thanks a lot for your further help.
I usually use the lossless mode for the standard cases QTGMC(Preset="fast", NoiseProcess=0, SMode=0, TR2=0, lossless=2, sourcematch=3)
this way you don't lose any sharpness. The standard settings for QTGMC often appear to blurr/unsharp. Not for all sources of course but I noticed it
on some DVD's which had good sharpness before. However this mode doesn't do any good in this special case it was probably only meant for the fully interlaced sources wheras for the progressive repair jobs the standard settings are better suited.

Regarding your samples:
the qtgmc method appears to be a bit sharper than the tfm only method but both methods have their issues on specific scenes.
For example if I use TFM(metric=1, CThresh=-1, pp=5) or TFM(metric=1, CThresh=12, pp=5) I'm getting weird things such as in frame frame 112 (looks even more weird when played in real time with mpc)
https://workupload.com/file/F3kbhy4W9hJ
and if I turn it off with 255 (no pixel will be detected as combed) I'm getting the weird grey pixels again as seen in 00:00:18 - 00:00:20 when applying TFM() TDECIMATE() without special settings on the sample of the main post

thanks a lot you guys for the info on how to configure the combing threshold and how to disable the post-field-matching order. I never thought about such settings before. The only thing I ever did if something wasn't detected right was a AssumeTFF() but other than that the standard TFM setting always did the job great.
Both of your suggested methods worked equally fine. However there are still plenty of interlaced leftover frames in the source. Here is a sample of a scene where with TFM(pp=0) already applied where you see it if you look at the knife and his mouth https://workupload.com/file/nqg7s8Gua4N
I tried to fix it with qtgmc progressive repair QTGMC(Preset="Fast", NoiseProcess=0, SMode=0, TR2=0,InputType=1)
but it didn't help much so I guess those are baked in?

thanks a lot for your further help

hello,
I usually always inverse-telecine 29,97 NTSC DVD's which are a mix of progressive and interlaced frames back to 23,976 by using the classic
TFM().TDECIMATE() from avs.

However in this special case of a TV show which was probably edited on Video but was converted back to film along the chain or whatever happened here on this particular DVD this method introduces strange errors probably due to some field problems but as this is not fully interlaced it doesnt do much to just decide between even or odd, in fact in this case both even and odd look much more terrible than an inverse telecine. Here is a sample: https://workupload.com/file/UCXaT7kWvDj
maybe some of the experts have a tip on how to deal with such cases.
thanks in advance for looking into it
regards
-skh

you are right that the file size encoded in a lossy aac or ac3 will not much differ between 16,24 and 32 bit but in an uncompressed format such as .wav it will. It is also very performance intense working with 24 or 32bit wav files in adobe audition takes ages to load on my PC. 16bit is much faster and doesn't take as much space also. I am just used to using 16bit my entire life for everything and most people even on pro releases do that too. You could always argue that as long as every step before the final encode was lossless with an as high bitdepth as possible it is ok to use 16 as a final encode but the reality is that the majority of the sources are already lossy so most who don't know better would just convert ac3 16bit to wav integer 16bit and then to ac3 16bit or aac 16bit again
That would generate the loss pandy warned about. However working like that for many years I never actually could tell a difference hearing wise such as more compression artefacts or that quantization noise. I have some descent monitors and a very good stereo setup and am pretty picky when it comes to sound quality so whatever I mess up can't be noticed by my hearing and when comparing before and after in spectral view also doesn't seem to differ much from each other.

ok so you would say that it is ok for to use 16bit as an output bitdepth and just not as a processing format. That is ok I have no problem if ffmpeg processes everything automatically higher than 16bit to guarantee a minimum loss in quality. so how would it be then if I process the aac in a sample_fmt="s32" wav and after that use 16 bit ac3 for the final conversion stage. Would that be good?

re-encoding is inevitable.
In a perfect world everything you're working with is in a lossless codec and after you're done you make an encode for the final stage. But that is not the reality. The majority of stuff I have to deal with is already encoded so when I change things I have no other choice than to re-encode it.

I always work with 16 bit PCM. And in the end convert that to AC3 192 (if its an old mono source) or 384, 448 kb/s if its a clean stereo source. Shure I made some comparisions with 24bit but I really hear no difference so never understood why I should bloat everything up to 24.
I don't see how a mono TV source originally recorded on VHS over 40 years ago encoded wrongly in some weird aac 96 kb/s 24bit format will sound any worse when I re-convert it to 16 bit pcm and after that to 192 kb/s ac3 - 16bit. There really can't be any audible difference for the human ear.

I'm not doing any high end produced music related masters here just some random old TV crap which does'nt have the best quality to begin with.

I also do a lot of DVD or Blurays for myself which have to be compatible to my stand alone Blu-ray players or if single file media for their internal usb ports and meant for streaming media AAC codecs or any sort of 24bit just doesn't make any sense to use there space and compatibility wise. Even for official stuff: Did you ever see a DVD with a 24bit AC3 audio track? I even find Blu-rays using 24bit lpcm for mono sources from an already degenerated hiss and crackle 35mm print from the 70's a total joke and a waste of disc space which could be better used for good video bitrate. Those audio tracks would sound the same in ac3 192 so whats the point other than marketing some voodoo?

I don't see a point in converting the audio to 32bit. Even for wav pcm I find that a maximum of 16bit is enough. If you do a wav to ac3 conversion for an audio file in ffmpeg it always is 16bit.
So if a direct aac to ac3 is only possible in 32bit due to the unset bitdepth of aac I wonder if there is a way to make a batch with an additional conversion step so that it just copys the video without re-encode. reencodes the aac audio to 16bit wav and after that re-encodes it to ac3 and mux it into a mkv container.

hey there,
I sometimes do audio conversions from AAC to AC3 when changing a video container.


-i %INPUTFILE% -c:v copy -acodec ac3 -b:a 384k -f matroska %1%.mkv


I just found out that no matter what you do it will always encode the audio with a bitdepth of 32 which doesnt make much sense.
I tried a lot of commands and most wouldn't work at all. For example as listed in this thread

thanks I think I got it now -)

is there a way to automate the output filename in a regular ".bat" ?

I sometimes do things like adding audio tracks from another video, keep the audio of the original and change the container such as:



"%FFMPEG%" -i "C:\Test\Video1.mp4" -i "C:\Test\Video2.mp4" -map 0:v:0 -map 1:a:1 -map 0:a:1 -c:v copy -c:a copy -f matroska "C:\Test\Video-NEW.mkv"


which adds an audio track from a different video to the first video as the first audio track, keeps the original audio as a second track and changes the container to mkv.

hello,
when encoding in 2-pass with ffmpeg it always generates a mbtree and a log file.

I am curious if it is possible to skip the 1st pass and instead accessing the infos from mbtree / log which basically contains all the info which is needed for the second pass?

For example you do a 2-pass and have to cancel it during the 2nd pass or your computer shuts down during 2nd pass. The mbtree/log is already written so it would be kinda sensless to let it write the same thing all over again and instead you just let it read the files and do a 1-pass. Or after a while of encoding you get a feeling which algorithms fit which kind of video best. For example "lots of action", "medium action" "almost no action at all" and "no grain", "medium grain" and "a lot of grain" , cartoon/live action whatsoever. For each variant you would have a mbtree and would just use that and do 1pass instead of crf.

I googled a lot but couldnt find anything regarding that toppic. Maybe some of the ffmpeg specialists here would know?
thanks for reading
all the best

you are a master of your craft. Thanks a lot. That totally makes sense to me :-)

ah ok thanks a lot that makes a lot of sense. I usually work with clips and have to get used to the "last" term. It is a bit confusing as I am never shure what status so to speak I declare as last. For a single video where I just apply the overlay on the entire thing it is totally logical of course. As I just work in a certain pattern there.
But for section based filtering I always work with clips which is easier for me to remember which section is filtered in which way.
Here is an example for your awesome idea. I'm curious what you think of it :-)


Source = LWLibavVideoSource("").AddBorders(0,0,4,0)
clip1=Trim(Source,0,12566).RemoveDirtMC(15,false)
clip2a=Trim(Source,12567,15398)
clip2b=Trim(Source,12567,15398).RemoveDirtMC(15, false)
clip3=Trim(Source,15399,72235).RemoveDirtMC(15,false)

thanks for the great explanation. The mask works fantastic. One last question to make shure I did everything right.

Is here in this command "last" the RemoveDirtMC image and (src) the original image just like it is in Overlay ?

thanks so much

thanks a lot man that is nice to know. So the removedirt bugs didn't occure because of the flickering they were somethings else. I tend to always crop everything away I don't need so from now on I will just leave it in standard resolution and crop after remove dirt.
The weirdness shown on the 2nd picture in my main post remain but I will definetley play around with the diffmask. thanks a lot for your help

ok I tested it and it seems that I'm still having the issues.

Frame 273 still shows weirdness.




I used

RestoreMotionBlocks(clensed,input,alternative=alt,pthreshold=30,cthreshold=8, gmthreshold=40,dist=3,dmode=2,debug=false,noise=limit,noisy=4, grey=_grey)

thanks for that suggestion. I'm not really shure if I understand it but I will definetley give it a go.
The video I uploaded is without RemoveDirtMC. Only the photos I posed are.

hello,
I'm having problems dirt filtering specific scenes. It seems that RemoveDirtMC is somewhat confused by faster movements in the style of flickering but not flickering as a deffective video more like the wind is blowing or the light is flashing.
I also tried Despot and spotless but they worked far worse than the various variants of Removedirt.
Maybe some of the experts here who are very good in understanding special configurations of functions could help. I would be very grateful.

here is a sample: https://www.transfernow.net/dl/20231014rCKBNzmE


and here are 2 screenshots of the issues which are caused