LAV Filters - DirectShow Media Splitter and Decoders

[nevcairiel]

Originally Posted by mogli

Why would anyone do that? One would eventually end up with the same volume as matrix normalization for most movies

Thats far from the truth. Especially if you don't include LFE in the downmix, as eg. Dolby recommends, most 5.1 streams can be downmixed to stereo without a change in volume, or only a very small one.
Mastering of commercial content rarely maxes out single channel levels.

Originally Posted by mogli

No benefit at all but very annoying.

You are free to not use it. But given the above reasoning, its a perfectly reasonable experience.

[mogli]

Originally Posted by nevcairiel

Mastering of commercial content rarely maxes out single channel levels.

We're talking about downmixing so channels will add up and e.g. even overload at half the full volume.

[Asmodian]

Originally Posted by mogli

We're talking about downmixing so channels will add up and e.g. even overload at half the full volume.

I often use it for headphones, I very rarely notice the volume decreasing due to a loud noise.

Have you tried it? It isn't a bad experience. You are free to use the normalized matrix option instead too, that is why it exists.

[nevcairiel]

Originally Posted by mogli

We're talking about downmixing so channels will add up and e.g. even overload at half the full volume.

Good that I explained the line above what you quoted why that isn't a problem in typical content.

Yes, it can theoretically happen that the volume lowers to the same level, however in the real-world with typical consumer-grade content this is not the case.
There are various factors that allow this. As said above already, channels are never maxed out, so there is headroom to do mixing in. Additionally, surround channel are often quieter for a more ambient experience then fronts, and due to the way PCM works, only if the audio is coherent would peaks actually add up perfectly. Additionally, not including LFE enables it to be much more reliable. And of course this only applies to 5.1, if you downmix 7.1 without dropping the rear channel, the chances of running into a volume reduction are much higher.

This is factually a technique used in many downmixing applications to various degrees. Sometimes in a hybrid fashion, where the mixing matrix is reduced so incoherent audio signals will never clip under any circumstances (disregarding existing headroom etc), but a coherent one still would. Preserving volume when downmixing for headphones is important for many users.

It does sound to me like you haven't even tried it, nor fully understood how and why it works.

At the end of the day, I'm not even sure what you are arguing for. This mode has been popular over the various years it has existed. But its also not the only mode you can use. You can just let it overflow (by checking no options), or you can use a fully normalized matrix where clipping is impossible, and handle the reduced volume in other ways. There is also no secret how it works, the potential for sudden volume changtes is even mentioned in the tooltip of the option. If you don't like one particular mode, you can just use another, there is nothing to gain from complaining that it exists and how bad it might be in your eyes.

[manolito]

What I do not understand is why the two options "normalize matrix" and "clipping protection" can be ticked at the same time. Since both options should prevent clipping when only one of them is ticked (as Asmodian explained) then in which situation could it be useful to activate both options?

[nevcairiel]

Originally Posted by manolito

What I do not understand is why the two options "normalize matrix" and "clipping protection" can be ticked at the same time. Since both options should prevent clipping when only one of them is ticked (as Asmodian explained) then in which situation could it be useful to activate both options?

Its not really useful, unless the signal would overflow on its own, which is only really possible with float codecs, so not really a common thing.
I suppose I could've locked the checkbox, but eh, doesn't really change anything.

[Sunspark]

Originally Posted by nevcairiel

Its not really useful, unless the signal would overflow on its own, which is only really possible with float codecs, so not really a common thing.
I suppose I could've locked the checkbox, but eh, doesn't really change anything.

A streaming radio station I like, the input states vorbis codec, 32bit Float and the output states PCM codec, 32bit Float. Checked another one, the input codec was aac, but it was also 32bit Float.

Maybe float is more common with streaming radio?

[nevcairiel]

Originally Posted by Sunspark

A streaming radio station I like, the input states vorbis codec, 32bit Float and the output states PCM codec, 32bit Float. Checked another one, the input codec was aac, but it was also 32bit Float.

Maybe float is more common with streaming radio?

Those are just lossy codecs that decode to float, but they wouldn't really overflow.

[mogli]

@nevcairiel

I just hope for an improved clipping protection.
If it's overflowing often it can't be helped and the overall volume needs to be decreased permanently for that movie. However, as you said yourself, that's rarely the case. Maybe only a few seconds, miliseconds or even samples overflow in a whole movie. Using a limiter in such a situation would allow to keep the overall volume steady and loud. So essentially I hope for short time (limiting) and long time overshoots (normalization, leveling) to be handled separately.
At 2 minutes in decreasing the volume by 2dB because 2 samples overflowed for the rest of a 2 hour movie doesn't sound reasonable to me.

[Sunspark]

Personally I don't use either normalize or clipping. I don't think it's needed because the player is set for 85% volume (though I leave it at 100% for web stuff), the PC is set for 73% volume, and the powered speakers have their own volume control as well. If you were running at 100% in everything, then yes, but in my example I am not, so I believe there is headroom.

[clsid]

@nev
Can you make a MinGW GCC 10.3 (or 11.1) build?

[Astray]

Hey there, got another audio/subtitle language order question. I watch a lot of anime and made a pretty crazy subtitle preference instructor

eng:*|f, eng:eng@Signs, eng:off, jpn:eng@Full, jpn:eng@Subtitles, jpn:eng, *:eng|d, *:*|df

The issue I'm having is with a lack of flexibility in the audio language currently. Generally I want to have jpn as the default language, but on rare occasions there are some anime that have better, or at least comparable, English voice acting and the .mkv files have it as the default audio language. This gets completely overwritten by the audio language preferences though. I don't even think forced audio languages would be respected in the files, in which case I can just remux the files, but I haven't tested.

Is there a way to do something like

*|df, jpn, eng

for the audio language preference and make it work?

[nevcairiel]

The advanced selector syntax is only available for subtitles, for audio only a list of languages is available currently.

[Astray]

Originally Posted by nevcairiel

The advanced selector syntax is only available for subtitles, for audio only a list of languages is available currently.

Where would be a good place to submit feedback and/or feature request for this? I would imagine it's not too difficult to add the feature in since you could probably copy and paste most of the code used for the subtitle selection.

[sofakng]

Is it possible to use DXVA2/D3D11 copy-back for 8K videos? (ie. 7260x3630)

I'm using an i7-8700k (PCIe 3.0 x16) an RTX 3080 but it's extremely slow.

Native-mode works fine but I need copy-back to use AviSynth and some filters on the video.

If it's not possible, what is the bottleneck?

[el Filou]

It's hard to pinpoint a single component that makes copyback slow or OK. In theory it should be fine with all systems but some even with IMC already had issues with it in 4K.
We had a thread dedicated to that here: https://forum.doom9.org/showthread.php?t=176642

[LigH]

I would say it is a quite safe bet that elaborate filtering of 8K UHD video in AviSynth+ won't work in real time. Certainly slower than untouched copy-back rendering.

[clsid]

I am now using a tool to convert GCC debug info into PDB, so that the crash reporter of MPC-HC is able to create proper stack traces for crashes that happen inside the FFmpeg code. It seems to be working well for traces, although it lacks values of variables.

One of the crashes it has logged so far is this one in the Matroska parser:
https://imgur.com/a/EvddqYY
readbytes is called with a nullpointer for the buffer. The code lacks a check for memalloc failure.

The crash dump indicates that user had 400+ MB of free memory. But that might have been heavily fragmented.

[GMJCZP]

I'm having a problem playing this .ape file, it sounds noisy, both with WMP and MPC-HC, when opening it in Adobe audition it sounds fine.

SampleAPE

Edit: I am using BeHappy 0.3.0.19814 and Monkey's Audio Console Front End 3.99 as encoder. I am using LavFilters 0.75.

[filler56789]

Originally Posted by GMJCZP

I'm having a problem playing this .ape file, it sounds noisy, both with WMP and MPC-HC, when opening it in Adobe audition it sounds fine.

SampleAPE

Confirmed, noisy with LAV Filters, plays OK with MPV.

Monkey's Audio Console Front End 3.99

FWIW: latest Monkey's Audio is version 6.29 :-|