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Old 6th July 2018, 15:06   #61  |  Link
hello_hello
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Originally Posted by qyot27 View Post
Here's the thing, though: it only happens when forcing fpsnum= and fpsden=. It doesn't occur when leaving it as just the filename, even if you then proceed to use ChangeFPS() to do the same sort of framerate conversion.
FPSNum and FPSDen have been problematic for as long as I've been using them. I think it's due to timecode jitter and as a result, FFmpegSource sometimes drops or duplicates frames unnecessarily so it's best not to use frame rate conversion unless you need to convert the frame rate.

FFmpegSource isn't alone though. LSmash can do the same thing. Mercifully though, they don't tend to misbehave for the same sources, so usually if one doesn't work correctly, the other will.

And there's always the VFRToCFR plugin.

Last edited by hello_hello; 6th July 2018 at 15:10.
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Old 6th July 2018, 15:21   #62  |  Link
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Originally Posted by manolito View Post
This is only partly true...

The reason that I could not reproduce the issue was that AVStoDVD created a script which had "fpsnum=2997, fpsden=100" in it. With these values the output is flawless. Only after I changed the script to use "fpsnum=30000, fpsden=1001" I was able to reproduce the issues.
I don't know how timecodes work or what the source file was, but that's probably just some sort of timecode jitter that doesn't upset FFmpegSource when it's using 2997/100.

Sometimes remuxing before indexing can fix the problem, even if you have to use a different container.

I've bumped into this problem when encoding 25fps AVIs, so it's not the NTSC frame rate as such.
After trying to convince Zathor it'd be a good idea for MeGUI to use it by default, I then tried to convince him that it wasn't so clever after-all.
https://sourceforge.net/p/megui/feature-requests/591/
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Old 6th July 2018, 15:53   #63  |  Link
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It gets even weirder...

I reverted back to the older C-plugin version 1140+101 fully expecting that the fpsnum/fpsden issue would be solved (like it did in earlier tests). But not this time. The output showed the same errors as the latest C-plugin version 1315+119.

Then I went back to the ancient Standard-plugin version 2.17, and this version handled the source correctly. Of course this is not really helpful because this old version cannot decode HEVC sources.

What does this mean? Could the issue lie in the C-plugin core after all? So far the only reliable workaround when using the C-plugin seems to be to ditch fpsnum/fpsden completely and follow the ffvideosource call immediately with a ChangeFPS command, like qyot27 recommends (I still do a lot of conversions to DVD, and VFR to CFR conversion is essential for me).


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Last edited by manolito; 6th July 2018 at 15:57.
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Old 6th July 2018, 16:25   #64  |  Link
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It gets even weirder...

I reverted back to the older C-plugin version 1140+101 fully expecting that the fpsnum/fpsden issue would be solved (like it did in earlier tests). But not this time. The output showed the same errors as the latest C-plugin version 1315+119.

Then I went back to the ancient Standard-plugin version 2.17, and this version handled the source correctly. Of course this is not really helpful because this old version cannot decode HEVC sources.

What does this mean? Could the issue lie in the C-plugin core after all? So far the only reliable workaround when using the C-plugin seems to be to ditch fpsnum/fpsden completely and follow the ffvideosource call immediately with a ChangeFPS command, like qyot27 recommends (I still do a lot of conversions to DVD, and VFR to CFR conversion is essential for me).
If the source was VFR, it'd be decoded at the average frame rate, the audio would no doubt go out of sync, and ChangeFPS() won't do anything to fix that. It has to be decoded at the correct frame rate.
For minor 29.970fps to 30000/1001 corrections I'd use AssumeFPS() instead. It'll probably only change the duration by a few nanoseconds anyway.
I kind of remember TIVTC uses 23.976 instead of 24000/1001 so I generally follow it with AssumeFPS() too, or I could be remembering wrong and that's a waste of time.

It might be the source. If you can, try remuxing it while setting the frame rate to 30000/1001 to see if that makes a difference. It might be that the source wasn't exactly 30000/1001 to begin with, and when it is, the problem will go away.

All that's assuming the source really is constant frame rate. Whenever MeGUI adds frame rate conversion to a script and it's not exactly standard NTSC or PAL, I always remove it and refresh the preview to see if the frame count and total duration changes. If the frame count stays the same and the duration only changes by a few milliseconds, then it's AssumeFPS() instead.

If it is a standard NTSC or PAL frame rate I remove it anyway as I've had too many bad experiences with dropped and incorrectly duplicated frames to use the frame rate conversion unless I really need to convert VFR to CFR.
Is AVSToDVD using MediaInfo to determine the frame rate? It's just that when it's VFR MediaInfo can confabulate all sorts of "real" frame rates.

The same MKV muxed with track statistic tags:
Frame rate mode : Variable
Frame rate : 52.594 FPS

Without them:
Frame rate mode : Constant
Frame rate : 25.000 FPS

That may be just what it displays in the GUI rather than what it tells other software, although as a test I indexed both MKVs with MeGUI/FFMS2.
For the first one MeGUI added fpsnum=26297 & fpsden=500 to the script. For the second it added fpsnum=25 & fpsden=1

I think I mentioned to Zathor at one stage that MeGUI should only add standard NTSC or PAL frame rate conversions because I don't think anything else makes sense. For example, if MediaInfo says the frame rate is 28.45fps, then it's a pretty good bet the video started life as 29.970fps, only duplicate frames were removed. The main point of the VFR to CFR conversion would be to put them back. Encoding at 28.45fps would make a mess, dropping frames that don't need dropping etc. The same probably applies to VFR sources from smart phones etc. If they're an odd average frame rate, you'd want to convert them "up" to the next highest standard NTSC or PAL frame rate. And for AVSToDVD, you're limited to NTSC or PAL frame rates anyway (I assume).

Last edited by hello_hello; 6th July 2018 at 19:39.
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Old 6th July 2018, 18:00   #65  |  Link
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The source is Selur's uploaded file from this post:
https://forum.doom9.org/showthread.p...39#post1845939

It is an XviD AVI, MediaInfo says that it is 29.97 progressive. It is CFR (I believe that XviD does not support VFR).

In ffms2 the fpsnum/fpsden parameters are used to guarantee CFR output (just like "Convertfps=true" in DirectShowSource). DSS2Mod will always output CFR. I do need CFR output because the DVD specs require it.

Quote:
If the source was VFR, it'd be decoded at the average frame rate, the audio would no doubt go out of sync
I strongly question this. Decoding of VFR sources at a different average frame rate may certainly ruin the video quality, but it will NOT alter the duration. So there should be no audio sync problems whatsoever.


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Old 6th July 2018, 18:31   #66  |  Link
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That's partially why I said that it may even be down to a quirk of AVI. Since AVI has to adhere to a constant frame-rate anyway*, fpsnum/den conversions are probably not a good idea - actually, short of it being a format with no available VFW decoder, like ffvhuff or newer revisions of ffv1, I wouldn't use FFMS2 to handle AVI files at all. MKV and other containers don't have that restriction, or are more or less assumed to be natively VFR, even if the content's framerate doesn't vary.

*because the only real way to force VFR into AVI is by using dummy frames, which still means that the source is CFR - just at a higher framerate with the lower rate parts padded up to the higher rate of the rest of the content (or both/all of the parts padded out to the lowest common denominator).

So the way this might tie back to the previous fpsnum/den situation is that AVI parsing could be a lot more fragile in terms of what framerates it allows. If the fps is already stored as 30000/1001, it may be okay. But if it was created with a float 29.97 or an integer 2997/100 rate like was mentioned as working, forcing 30000/1001 could be all it takes to trip the problem. And that could be in the AviSynth plugin logic (but again, it would have to exist in both the C++ and C plugins, and since they're separate code it's either really old, or the problem is actually in libavformat and can only express itself in a constant framerate environment like AviSynth).

For the record, 3000/101 seems to work too, even though the end result of that is 29.7029703. It makes me wonder if AVI can even store framerates as integers, or whether it's always converted to float beforehand.
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Old 6th July 2018, 18:40   #67  |  Link
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Originally Posted by manolito View Post
The source is Selur's uploaded file from this post:
https://forum.doom9.org/showthread.p...39#post1845939

Quote:
Originally Posted by manolito View Post
It is an XviD AVI, MediaInfo says that it is 29.97 progressive. It is CFR (I believe that XviD does not support VFR).
Neither does the AVI container, although AVI supports some sort of "null frames" which means it can vary.

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Originally Posted by manolito View Post
In ffms2 the fpsnum/fpsden parameters are used to guarantee CFR output (just like "Convertfps=true" in DirectShowSource). DSS2Mod will always output CFR. I do need CFR output because the DVD specs require it.
Decoders always output a constant frame rate, but which frame rate....
AVISource always outputs a CFR and it duplicates frames to replace "null frames". DirectShowSource with ConvertFPS=true does the same.
AVISource is clever enough to output the correct frame rate without being told. For a 25fps AVI with a few null frames it'll output 25fps. By default, ffms2 and LSmash would treat it as VFR and output the average frame rate, which might be 24.992 without the frame rate conversion. They spread the frames out over the total duration.

Quote:
Originally Posted by manolito View Post
I strongly question this. Decoding of VFR sources at a different average frame rate may certainly ruin the video quality, but it will NOT alter the duration. So there should be no audio sync problems whatsoever.
Edit: Sorry, when I referred to the audio going out of sync I meant when the video is decoded at the average frame rate without any conversion. I did mention the audio sync when converting to the average frame rate though. I guess I wasn't thinking, but I made up for it by doing a nueron2 and editing my post to remove the evidence . And then I misunderstood you initially, so explanation deleted.... I'm over-tired.

I still think my logic is sound when it comes to converting to a constant frame rate. Converting to a standard frame rate seems logical to me. ie The appropriate peak frame rate for the source and PAL or NTSC. Unless you're happy to drop frames.

These days I've embraced VFR encoding. Not that I do a lot of it as there's not a huge need for it in PAL-Land, but I doubt I'd ever convert a hybrid NTSC DVD to a constant frame rate again. I only encode with x264 though, and obviously for DVDs you don't have the same luxury.

Last edited by hello_hello; 6th July 2018 at 19:44.
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Old 6th July 2018, 19:15   #68  |  Link
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Originally Posted by manolito View Post
The reason that I could not reproduce the issue was that AVStoDVD created a script which had "fpsnum=2997, fpsden=100" in it. With these values the output is flawless. Only after I changed the script to use "fpsnum=30000, fpsden=1001" I was able to reproduce the issues.
It seems to be as I thought it might. If you remux the AVI as an MKV the problem goes away. Well... I only checked two of the problem frames, but they were fine. Not that MKVs are immune, but in this case it seems to work.

Mind you I don't know why AVStoDVD was using "fpsnum=2997, fpsden=100" because MeGUI picked "fpsnum=30000, fpsden=1001".
Although as I type, there's a memory returning.... something about some players crashing when 30000/1001 is used....
I guess AVStoDVD is playing it safe.

http://www.avisynth.org.ru/docs/engl...ilters/fps.htm
In v2.57, the behaviour with respect to the framerate is a bit changed. The main issue is that users are allowed to specify the framerate as float, but the NTSC (FILM and Video) and PAL standards require ratios as framerate. Besides this AviSynth exports the framerate as a ratio, so when specifying a float, it will be converted to a ratio. The ratios of the standards are given by 24000/1001 for 23.976 (FILM) and 30000/1001 for 29.97 (Video). When specifying these floats, they are exported by AviSynth as ratios, but not as the standard ratios. One of the reasons for this is, that those floats are approximations (remember that 24000/1001 = 23.9760239760...), so how should AviSynth know how to choose the correct ratio? In order to overcome this issue, the user can use AssumeFPS(24000,1001) or simply AssumeFPS("ntsc_film").
Another problem is that the converted floats were (in v2.56 and older) exported with 64 bit precision, resulting in very large numerators and denominators, making some players crash. To overcome this, a smart float-ratio is added internally, and the framerates are approximated accurately by ratios of small numbers. For example, AssumeFPS(23.976) is converted to AssumeFPS(2997,125) as can be checked with Info.

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Old 6th July 2018, 21:50   #69  |  Link
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OK, I see your points, but for me repacking source files depending on their formats is not an option. I mainly use AVStoDVD for DVD conversions, and the whole point of using such a software is to have solutions which work each and every time no matter what the peculiarities of the sources are...

I think that the best workaround when using ffms2 is to ditch all fpsnum/fpsden parameters from the ffvideosource call and instead follow the ffvideosource call with a ChangeFPS command. I just posted a request for the author to change the handling of ffms2 accordingly.


Cheers
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Old 6th July 2018, 23:34   #70  |  Link
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I think that the best workaround when using ffms2 is to ditch all fpsnum/fpsden parameters from the ffvideosource call and instead follow the ffvideosource call with a ChangeFPS command. I just posted a request for the author to change the handling of ffms2 accordingly.
But if the source is VFR, ditching the conversion can potentially cause audio sync issues and ChangeFPS can't help.
That was why I requested MeGUI add it to scripts originally, as then it can convert VFR sources, even if not in an ideal way, but it turned out to be a bad idea due to the frame dropping/duplication issue.

If you use ChangeFPS instead you'd have to be converting from one constant frame rate to another, which is probably what happens most of the time anyway. Or maybe a checkbox might be useful. I assume if AVStoDVD is converting the frame rate it'd be converting to a DVD compliant one, but an ability to tell it the source is CFR or VFR would allow AVStoDVD to use fpsnum/fpsden or ChangeFPS as appropriate.

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Old 7th July 2018, 00:10   #71  |  Link
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but an ability to tell it the source is CFR or VFR would allow AVStoDVD to use fpsnum/fpsden or ChangeFPS as appropriate.
Sorry but I do not really understand...

Isn't using fpsnum/fpsden exactly the same thing as using ChangeFPS? According to the ffms2 docs using fpsnum/fpsden forces the specified output fps by dropping or repeating frames. ChangeFPS does exactly the same thing. Where is the difference?

The whole point IMO is to make sure that the output is CFR no matter if the source is VFR or CFR. And if the source already is CFR then it would not matter, using fpsnum/fpsden or ChangeFPS would not do anything in this case.


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Old 7th July 2018, 02:03   #72  |  Link
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Without fpsnum/fpsden, the frames are just output at the average frame rate. There's no frames added or deleted, they're just spread out evenly. When fpsnum/fpsden is specified the video is converted to the specified frame rate, even if it's the average frame rate, adding and/or deleting them as required to maintain audio sync.
Consider a source that's 23.976 for the first half and 29.97 for the second. You could add frames to the first half or delete frames from the second half or some combination of the two, but you can't just spread the existing frames out evenly.

It makes sense because you don't want FFMS to add or remove frames unless it's specifically told to, or at least without a method to stop it from doing so, otherwise it'd be really hard to encode VFR as VFR, where you want the existing frames with no additions or deletions.

That's what I was getting at before. If the video is VFR it has to be decoded to a specified frame rate. If fpsnum/fpsden isn't specified it's not. ChangeFPS has no idea how the frames are spread out in the source.

Of course if you're happy to assume all your sources are CFR, then dropping fpsnum/fpsden is a good idea, although it mostly works fine when it's converting real VFR to CFR, as opposed to converting 29.970 to 30000/1001, which requires an extra frame roughly every ten hours, so AssumeFPS would make more sense for that sort of thing.

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Old 7th July 2018, 14:54   #73  |  Link
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ChangeFPS has no idea how the frames are spread out in the source.
Why would it need to know this? And do you really believe that when using fpsnum/fpsden the plugin does a global analysis of the source before decoding in order to retrieve such knowlege about how frames are spread out in the source? I do not think so...

From the DirectShowSource docs:
Quote:
convertfps = false (in v2.56): When setting it to true, it turns variable framerate video (vfr) into constant framerate video (cfr) by duplicating or skipping frames.
From the ffms2 docs:
Quote:
This may naturally cause `FFVideoSource` to drop or duplicate frames to achieve the desired frame rate,
From qyot27:
Quote:
it doesn't occur when leaving it as just the filename, even if you then proceed to use ChangeFPS() to do the same sort of framerate conversion.
I still believe that avoiding fpsnum/fpsden and instead use ChangeFPS after the ffvideosource call will render the same result (or amost the same). In the end we should leave the answer to folks like qyot27 or Myrsloik who understand the inner workings of the plugin...


Generally the AVStoDVD concept for creating DVD output is to always treat the source as VFR (use DSS2Mod, use convertfps=true for DirectShowSource and use fpsnum/fpsden for ffms2). This will ensure that VFR sources are handled correctly, and for CFR sources no harm will be done. Using AssumeFPS is not an option since this will alter the video duration causing audio sync errors.


I am more interested to find out which version of ffms2 was the first to show these issues. After finding that the standard C++ version 2.17 worked fine I just tried the C-plugin version 2.17. And this one also showed no problems.

Unfortunately I was unable to find C-plugin versions between 2.17 and 2.23 (GitHub does not have them, and the MediaFire links from qyot27 were expired). So all I know is that C-plugin version 2.17 works while version 2.23 (r1140+101) does not work. The issue must have been introduced somewhere between these two versions.


Cheers
manolito

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Old 7th July 2018, 19:40   #74  |  Link
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Why would it need to know this? And do you really believe that when using fpsnum/fpsden the plugin does a global analysis of the source before decoding in order to retrieve such knowlege about how frames are spread out in the source? I do not think so...
If that's not the exact definition of indexing it's got to be a very close relative. DSS would output the specified constant frame rate, no doubt adding or dropping them as required based on each incoming frame duration. DSS would have to do it on the fly. FFMS2 runs an indexing pass first.
All you need is the frame count and duration and some easy maths to output the average frame rate, but I can't imagine how FFMS2 could convert VFR to CFR by duplicating/dropping frames if it doesn't know the duration of each frame in a VFR source, or in other words the speed at which the source frames are zipping by at any moment in time. ChangeFPS would just see the existing frames at "some" constant frame rate because that's what Avisynth needs, so it can't convert VFR that's been made constant simply by averaging out the frame durations, and if you tell FFMS2 to drop or duplicate frames to convert to a constant frame rate there's no need to ConvertFPS again.

It'd be impossible to re-encode VFR sources as VFR unless only the original frames are decoded. No more, no less.
The standard method is to extract the timecodes file, encode without any frame rate conversion, the script frame rate is irrelevant but the average frame rate seems a popular choice as it doesn't change the duration, give x264 the timecodes file so it knows where the frames really belong time-wise, or use the timecodes when muxing to distribute the frames in the same manner as the source.

Here's what the help file I have says (FFMS2 version 2.23.1)

Quote:
##### int fpsnum = -1, int fpsden = 1
Controls the framerate of the output; used for VFR to CFR conversions.
If `fpsnum` is less than or equal to zero (the default), the output will contain the same frames that the input did, and the frame rate reported to Avisynth will be set based on the input clip's average frame duration.
If `fpsnum` is greater than zero, `FFVideoSource` will force a constant frame rate, expressed as a rational number where `fpsnum` is the numerator and `fpsden` is the denominator.
This may naturally cause `FFVideoSource` to drop or duplicate frames to achieve the desired frame rate, and the output is not guaranteed to have the same number of frames that the input did.
I just checked a VFR MKV with a section at the beginning that's 25fps while the bulk of it is 59.94fps. The 25fps section has 5877 frames and ends at 3min 55 seconds. 5877 frames at a CFR of 25fps does work out to 3min 55sec. Without FFMS converting the frame rate it still had 5877 frames but in MeGUI's preview it finished at 1min 21sec and wouldn't have looked out of place as a closing for The Benny Hill Show.
With FFMS2 converting to some constant frame rate the frame count increases or decreases accordingly but the scene ends at 3min 55sec.

That's part of the issue people seem to have when ripping DVDs with MakeMKV and then indexing them with FFMS2. Unlike DGIndex, FFMS2 ignores repeat flags by default, which means for hybrid NTSC it just outputs the existing frames, and I'm pretty sure it does so at the average frame rate.
The help file says it assumes a CFR when the RFF option is enabled, although I'd imagine that's not living too dangerously given I assume the RF flags ensure the frame rate is constant anyway.

Last edited by hello_hello; 7th July 2018 at 20:27.
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Old 7th July 2018, 22:22   #75  |  Link
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If that's not the exact definition of indexing it's got to be a very close relative. DSS would output the specified constant frame rate, no doubt adding or dropping them as required based on each incoming frame duration. DSS would have to do it on the fly. FFMS2 runs an indexing pass first.
Alright, you score for ffms2, but for DSS2Mod and DSS there should be no difference when using ChangeFPS instead of "convertfps=true". In the couple of tests I did I could not see any problems with this approach.


We are dealing with two different problems here:
1. How to treat real VFR sources when CFR output is required?
2. How to deal with the regression in ffms2 where fpsnum/fpsden is broken?


For the first problem I had a dispute with Zenitram some time ago when he stopped displaying the frame rate and duration of VFR MKV files. This post by Foxyshadis
https://forum.doom9.org/showthread.p...08#post1739508
saved my life more than once. Easy to use and good quality output.

For the second problem I would really like to see current ffms2 versions fixed. It did work in version 2.17 for both the standard and the C-plugin, so it must be possible to find the error and fix it. If this is not going to happen then I will need to employ the workaround using ChangeFPS instead of fpsnum/fpsden. I am a result oriented person, I will use "whatever works", but the method has to be universal, it has to work for all kinds of sources without deeply analyzing them.


Cheers
manolito

Last edited by manolito; 7th July 2018 at 22:27.
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Old 8th July 2018, 01:59   #76  |  Link
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To be fair, the situation where I said ChangeFPS would be preferable is not VFR-to-CFR, it was AVI files like that sample specifically (I also don't really know how the fps conversions in FFMS2 perform their logic, it was a general statement about framerate).

The process of using AssumeFPS when the framerate is close enough to standard* is the most correct way, but for safety's sake I always go with sync_audio=true and then run SSRC or ResampleAudio so that the audio frequency gets fixed back to a standard 44100 or 48000 Hz, depending on what exactly it was that I was encoding for. But if you're trying to preserve sync whilst absolutely not touching the audio, the only solution is to drop or insert duplicate frames to do it.

*Even though the minor framerate differences won't result in any audible pitch shift; the shift is there, but human ears really can't tell if 44100 was slowed down to, say, 44030 before getting re-normalized to 44100; not unless you do it dozens of times in a row. Thousandths or tens of thousandths of a frame, here: I usually would find myself having to do it to correct whole 30->29.97 or 24->23.976, but we're talking about stuff like 2997/100->30000/1001; 24/30->NTSC Film/Video is a 1.001% shift, and I couldn't hear the difference - simplified 23.976/29.97->NTSC Film/Video is a difference of 0.0001%.


I kind of feel like we're getting a bit into the weeds here, though.
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Old 8th July 2018, 03:03   #77  |  Link
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When the source frame rate is already DVD complaint, AVStoDVD should be aware of it and not add any ffms2 frame rate conversion to the script. Ideally it should have no effect if it's added, but obviously that's not what happens all the time.

For VFR frame rates that aren't in the DVD compliance ballpark, and when DSS isn't an option, there's really no other way than to convert it to a CFR as it's decoded. Lsmash has the same conversion options as ffms2, so maybe if your source is VFR and there's incorrectly dropped/duplicated frames in the output, an ability to pick the indexer might be enough. I've definitely seen Lsmash go nuts in the past and duplicate/drop frames it shouldn't (ie when the specified frame rate is the same as the source frame rate) so I'm not sure it's an ffms2 specific bug, but fortunately I don't think I've had a source that neither indexers could convert successfully. Maybe it's just luck.

Still feeling obligated to argue about the ability of ChangeFPS to convert a VFR source to CFR correctly.....

When the VFR source is decoded without ffms2 converting it to a constant frame rate, FFMS2 outputs the average frame rate.
The average frame rate is pretty much the VFR equivalent of AssumeFPS, only because it's the average frame rate it can do the assuming without changing the duration. Still, it'll effectively speed up some sections and slow down others, just like AssumeFPS() changes the speed of constant frame rate video.

If you happen to get lucky and you can decode a VFR source at the average frame rate without effecting the audio sync, then of course you can convert that to a different frame rate with ConvertFPS and still not effect the audio sync any further, but it doesn't mean it converted VFR to CFR. When the frame rate varies enough or even early enough that the audio slowly loses sync when decoding at average frame rate, ConvertFPS won't fix it. It'll output the frame rate you tell it to, but the audio sync won't change.
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Old 8th July 2018, 04:56   #78  |  Link
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Also, I don't know why I didn't do this earlier, but there does appear to be something wonky with that sample file. MKVToolNix accepts it as input, but ffmpeg won't remux it into MKV, because it complains about the file's timestamps (it will allow it to mux into another AVI, but still throws a non-fatal warning about the timestamps in the file):
Code:
[avi @ 00000226993850c0] Timestamps are unset in a packet for stream 0. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly
And from running a grep on the FFMS2 source code in the avisynth, avisynth_c, and vapoursynth directories, I'm starting to suspect that this is entirely external to FFMS2, and that the issue is partially in libavformat and partially in AviSynth itself. The fps-related hits in all three of those directories were all pretty similar, and nothing seemed to stick out as being really different, which points to this disconnect between AviSynth and VapourSynth probably being more about how those programs themselves interact with FFMS2 than something entirely in FFMS2's hands. I haven't tested VapourSynth, though (mostly because I already installed Python 3.7 for all my Python things, and don't want to also have to 3.6 on here too; I'll wait until the Windows builds are built against 3.7).
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Old 10th July 2018, 19:44   #79  |  Link
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ffaudiosource problems with aac_latm audio tracks

First of all this issue applies to the standard plugin as well as to the C-plugin, I'll post it here because I usually use the C-plugin.

I've been thinking for some time that aac_latm is an evil format, but since it has become one of the two audio standards in German DVB-T2 broadcasts I have no choice but dealing with it.

I uploaded a short sample file here:
https://www51.zippyshare.com/v/KY10rZkC/file.html

This format is used by half of the public stations (the other half use E-AC3). It is not this specific file, it's all captured files with this audio format. I also doubt that my receiver has anything to do with it (a Xoro with the latest firmware). The file plays without errors in all my software players, MediaInfo has no problems with it. And using LAVFilters with DSS2Mod also handles it nicely.

Only ffaudiosource has issues, and it applies to all current versions up to the latest 2.23.1 of the standard plugin and r1315 of the C-plugin. Tested on 2 different computers under WinXP and Win7-64bit.

This is the problem:
When using ffaudiosource("filename", track=-1) I get the error "No audio track found". When I use "track=1" or "track=2" the error is "Track has no audio samples". (I did verify that the 2 audio tracks are track #1 and track #2).

First I thought that ffms2 was just unable to decode this audio format, but this is not true. When I use the shortcut "ffmpegsource2" instead of "ffaudiosource" then the audio is decoded properly. This is true for the C-plugin where the shortcut comes from "ffms2.avsi" as well as for the current standard-plugin where the shortcut is integrated into the DLL.

Can anyone explain this weird behavior? I certainly do not have a clue...


Cheers
manolito

Last edited by manolito; 12th July 2018 at 17:29. Reason: Sendspace is too slow...
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Old 11th July 2018, 02:52   #80  |  Link
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FFMS2 C-plugin r1327+120 (directs to first post in this thread)

The reason it took me so long before posting another build was that I was doing a pretty deep refactoring of the FFmpeg and mpv build guide I maintain, to switch all of the entries to out-of-tree builds (unless that was impossible; I think a grand total of 5 pieces couldn't be, out of 74-ish?). The result is that the 32-bit and 64-bit instructions in the guide can now use a single download of the source code, which saves download time/bandwidth, and it saves storage space.

Then, because I'm apparently a glutton for punishment, I'd gotten so fed up with having to build the 32-bit environment under a VM that I decided to try doing a multilib environment again (thus saving time at the beginning, because only one cross environment needs to be built, as well as being able to avoid all the nasty performance hits of a VM). This was...significantly trickier, as it exposed some pretty deep problems with how some of the pieces in FFmpeg's dependency chain have their build systems configured - and most of it caused headaches for the 32-bit parts. I had to resort to disabling assembly code on some of those 32-bit parts because their build systems won't tell nasm/yasm/gcc to build the asm for i686 in a multilib environment, so it tries building it as amd64 instead, and that causes it to fail to link at the end. Hopefully there's actually a way to force it without just having to wait on those projects to fix their autoconf setups (because it seemed to hit autoconf projects the most).

And then, because the Universe hates me, the computer I had been using as my main setup for the last couple years (a Quantum Byte mini-PC I bought in 2015) ended up getting fried due to a faulty power strip when I was just under halfway done with the move to multilib. So I had to order a new mini-PC (a Byte3), and go through setting up both the OSes *and* then get back to and finish that move to multilib in the guide. At least the new one runs 64-bit Windows 10 and I immediately migrated the Windows install on it to an SSD, so that crap shouldn't happen again (and enough free space to allow for installing MSVC, so I can get back to helping with AviSynth+ development...again, hopefully).


Quote:
Originally Posted by manolito View Post
Only ffaudiosource has issues, and it applies to all current versions up to the latest 2.23.1 of the standard plugin and r1315 of the C-plugin. Tested on 2 different computers under WinXP and Win7-64bit.

This is the problem:
When using ffaudiosource("filename", track=-1) I get the error "No audio track found". When I use "track=1" or "track=2" the error is "Track has no audio samples". (I did verify that the 2 audio tracks are track #1 and track #2).

First I thought that ffms2 was just unable to decode this audio format, but this is not true. When I use the shortcut "ffmpegsource2" instead of "ffaudiosource" then the audio is decoded properly. This is true for the C-plugin where the shortcut comes from "ffms2.avsi" as well as for the current standard-plugin where the shortcut is integrated into the DLL.
It works with the build I just posted...sort of. I always index with ffmsindex first, and using the normal ffmsindex -t -1 ARD.mts to index the video and audio fails with an error message about the audio track.

Referring to the documentation, I set the errorhandling value to 2:
Code:
ffmsindex -t -1 -s 2 ARD.mts
This allows ffmsindex to complete the indexing. -t -1 indexes all the tracks, but the error specifically occurs on track 2 - setting -t 1 (for just track 1) without -s 2, it can index successfully. The resulting FFAudioSource script with track=1 plays back.

It also - probably incorrectly - will play back the same track if you select track=2 with that first -t -1 -s 2 script.

But -t 2 without -s 2 errors out; -t 2 -s 2 does allow indexing to complete, but FFAudioSource complains about the track not containing samples.

Last edited by qyot27; 11th July 2018 at 02:56.
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