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Old 27th October 2012, 15:40   #12661  |  Link
e-t172
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Quote:
Originally Posted by 6233638 View Post
I assumed the 10dB boost was part of the decoding step - all competent receivers boost LFE by 10dB automatically when sent anything other than PCM these days for example.

Is LAV not doing that when decoding audio?
No it doesn't, and for good reason. The 10dB boost is done at the amplification stage (if not downmixing). If it was done earlier in the chain it would be implemented as a -10dB attenuation in all other channels, which reduces dynamic range, thus defeating the purpose of the +10dB boost in the first place.

That means LFE level matching is done at the last possible stage. When using a surround sound system this is normally done by the power amplifier. When downmixing this is done when the LFE channel is removed (mixed).

What nevcairiel is afraid of is that the 7dB LFE boost translates to a -7dB attenuation for other channels when downmixing with a normalized matrix. This means you're losing 7dB of dynamic range, you're raising your signal source noise floor by 7dB, and you might run into maximum gain issues. Whether or not this is an issue depends heavily on the hardware you're using.

You can also try without normalizing, in which case you get the correct LFE level and no dynamic range reduction, but you risk clipping. That said, I think that standard stereo downmix of 5.1 content is not allowed to clip per Dolby specifications.

Last edited by e-t172; 27th October 2012 at 15:43.
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Old 27th October 2012, 16:07   #12662  |  Link
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6233638
Still, check to the filter chain, after LAV Audio, were no other audio filters, which can not handle 32 float.
Examples:
Mixing AC3 5.1 in stereo (not normalized matrix):

LAVSplitter -> LAVAudio - Dump
no clipping

LAVSplitter -> LAVAudio -> ffdshow Audio Processor (enabled: Volume -> Master volume: 99%) - Dump
clipping
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Old 27th October 2012, 16:21   #12663  |  Link
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Microsoft's AC-3 has a special laptop mode that applies a 11dB gain. I don't know if that applies to just Center or all channels.
http://support.microsoft.com/kb/944436
I would be interesting to see a comparison of stereo output of MS/LAV/ffdshow.

Currently the Center level allows a maximum gain of 3dB compared to normal value. Would it make sense to allow higher values like was done for LFE?
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Old 27th October 2012, 16:30   #12664  |  Link
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Quote:
Originally Posted by Dogway View Post
So LAV Filters actually is incompatible with ffdshow, you gotta either, disable all the decoders in ffdshow, lower ffdshow merit, raise LAVFilters merit, or block in MPC-HC which is the way I chose. So if nevcairiel recommends a tutorial where the third option is promoted, what way do you think I am going to take? Unless you give a founded reason there is no incovenient to use any of the 4 options.
You can quickly disable formats in ffdshow if you do it in its installer. It has a tree structure, so 4 clicks is enough to disable everything if you click on the roots of the tree ("Decode the following formats with..." entries).
But if your prefer LAV decoders, then you don't actually need ffdshow at all. There is no point in installing it if you block it later on.
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Old 27th October 2012, 16:49   #12665  |  Link
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Originally Posted by clsid View Post
You can quickly disable formats in ffdshow if you do it in its installer. It has a tree structure, so 4 clicks is enough to disable everything if you click on the roots of the tree ("Decode the following formats with..." entries).
But if your prefer LAV decoders, then you don't actually need ffdshow at all. There is no point in installing it if you block it later on.
I intalled from the time I needed to mix down audio, play strange video/audio codecs, post-process video with avs, or as you know preview avs files, etc. I thought it was a good fallback to have there because it's very powerful with its filter system (although a bit old and messy).
But still I don't understand why you are so against raising LAVFilters merit (or lowering other decoders'). It's much easier, unless there's any drawback you haven't told me until now.
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Old 27th October 2012, 20:54   #12666  |  Link
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Quote:
Originally Posted by Dogway View Post
-Why isn't LAV Audio the default audio decoder?
Simply because FFDShow's Audio Decoder by default has a higher merit than the LAV Audio Decoder. Either uninstall FFDShow (do you really still need it?), disable specific formats in the "Codecs"-section, or lower FFDShow's merit settings, for LAV Filters to take over.
Quote:
Originally Posted by Dogway View Post
-Has LAVSplitter ever been able to connect to ffdshow?
...
By your logic and reading between lines, LAV Decoders have a merit higher than MOST decoders, but not all because "This means that if a format is enabled in ffdshow settings, ffdshow will get used". So LAV Filters actually is incompatible with ffdshow...
You do realize FFDShow lags the splitter LAV Filters does have? For FFDShow you'd always need a splitter filter to play anything. Unless FFDShow functions as a post-processor, the LAV- and FFDShow's Video Decoder will of course never connect with eachother.
I don't know about you, but I can perfectly play videos where LAV Filters does the splitting and FFDShow the decoding of the video and audio stream.
Although there's still a few exotic formats for which LAV Splitter prefers to connect to LAV Video/Audio Decoder, LAV Splitter and FFDShow can get along just fine.
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Old 27th October 2012, 23:19   #12667  |  Link
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Simply because FFDShow's Audio Decoder by default has a higher merit than the LAV Audio Decoder
That's what had to be expected, but in order to know that you need to "have messed" with (the forbidden) merit settings.

Quote:
Originally Posted by CoRoNe View Post
Either uninstall FFDShow (do you really still need it?), disable specific formats in the "Codecs"-section, or lower FFDShow's merit settings, for LAV Filters to take over.
What's so bad in FFDShow that you encourage me to uninstall it? I mean it's the same question I'm trying to get an answer for since 2 pages back. What's so bad in rasing or lowering merit settings? Can't I raise LAV Filter merits? (you didn't name it as an option)

Quote:
Originally Posted by CoRoNe View Post
but I can perfectly play videos where LAV Filters does the splitting and FFDShow the decoding
Yes sorry I had some confussion for when I couldn't split AVS with LAV but had to decode with ffdshow.
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Old 27th October 2012, 23:54   #12668  |  Link
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Don't know if any of you guys have seen or not but the latest version of AnyDVD HD seems to have implemented a cool feature -
It adds a "disc.inf" file to the root of the Blu-ray disc that contains some basic information about the disc, this could be most useful on Blu-ray's that contain fake playlist protection (Which causes LAV to pick the wrong playlist)
The correct playlist is listed in the "disc.inf" file, it looks like this -

[disc]
type=BD-ROM
totalsectors=24382848
label=WARRIOR
region=1
playlists=789, 790

Is this something that software like LAV/XBMC etc could read from to help choose the correct main movie playlist?

Thanks for the latest update Nev!
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Old 28th October 2012, 00:16   #12669  |  Link
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It's the player's responsibility to use that, not LAV. I've requested it for MC18.
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Old 28th October 2012, 00:19   #12670  |  Link
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LOL! I quit this pointless discussion. Never met anyone like you before.
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Old 28th October 2012, 01:28   #12671  |  Link
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It's the player's responsibility to use that, not LAV. I've requested it for MC18.
I'm using MPC-HC, anyway I can get that to work with the disc.inf file?
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Old 28th October 2012, 01:31   #12672  |  Link
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No. That's what I'm saying, the player itself needs to support it. Nev said he looked at it but he doesn't have a screen pass disc so didn't see the playlist info. So maybe now that you posted one he might consider looking at it. At least for exposing just the valid playlists that the player can then pick from. It would make the non-commercial players infinitely more useful. Especially now where the commercial ones all have Cinavia.
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Old 28th October 2012, 03:26   #12673  |  Link
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Quote:
Originally Posted by e-t172 View Post
No it doesn't, and for good reason. The 10dB boost is done at the amplification stage (if not downmixing). If it was done earlier in the chain it would be implemented as a -10dB attenuation in all other channels, which reduces dynamic range, thus defeating the purpose of the +10dB boost in the first place.
That makes sense, I simply assumed it was done at the decoding stage due to it being applied automatically in the receiver when it was sent anything other than PCM. Of course boosting it at the amplification stage avoids the reduction of dynamic range.

Quote:
Originally Posted by e-t172 View Post
What nevcairiel is afraid of is that the 7dB LFE boost translates to a -7dB attenuation for other channels when downmixing with a normalized matrix. This means you're losing 7dB of dynamic range, you're raising your signal source noise floor by 7dB, and you might run into maximum gain issues. Whether or not this is an issue depends heavily on the hardware you're using.
What is the reason for using 7dB and not 10dB?

Quote:
Originally Posted by e-t172 View Post
You can also try without normalizing, in which case you get the correct LFE level and no dynamic range reduction, but you risk clipping. That said, I think that standard stereo downmix of 5.1 content is not allowed to clip per Dolby specifications.
You say it is "not allowed to clip" do you mean that normalization should not be necessary to avoid clipping, or that you should be normalizing because clipping is not allowed? (of course I want to avoid clipping)

Quote:
Originally Posted by Deim0s View Post
6233638
Still, check to the filter chain, after LAV Audio, were no other audio filters, which can not handle 32 float.
Examples:
Mixing AC3 5.1 in stereo (not normalized matrix):

LAVSplitter -> LAVAudio - Dump
no clipping

LAVSplitter -> LAVAudio -> ffdshow Audio Processor (enabled: Volume -> Master volume: 99%) - Dump
clipping
Thank you. I am currently either using:
  • LAV Audio (Float) → ReClock (Float) → Windows 8 Mixer (24-bit HDMI)
  • LAV Audio (Float) → ReClock (24-bit Int Padded to 32) → WASAPI Exclusive (HDMI)
Either scenario has this "crackling" problem when LFE is above 0.04 in the stereo mix.

How are you dumping the audio for these images? (and what software is that?)
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Old 28th October 2012, 05:23   #12674  |  Link
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Quote:
LAV Audio (Float) → ReClock (Float) → Windows 8 Mixer (24-bit HDMI)
LAV Audio (Float) → ReClock (24-bit Int Padded to 32) → WASAPI Exclusive (HDMI)
I do not use ReClock, do not know how it handles 32 float.
In conjuction: LAV Audio (32 float) -> Win 7 Mixer (24 bit, 48000 Hz) -> Creative X-Fi (48000 Hz, Bit-Matched), even when playing the file with the overflow > 0dB (an example from the post above), I do not get clipping:
Image 1 Image 2
If after LAV Audio (and before mixer Win) is going audio filter not capable of processing 32 float, then there will be clipping:
Image 1 Image 2
I have got so.
Quote:
How are you dumping the audio for these images? (and what software is that?)
I use a Dump Filter from GraphStudioNext. Further open the raw-file in Adobe Audition (Import -> Raw Data).
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Old 28th October 2012, 08:07   #12675  |  Link
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If you set LAV Audio to output 32bit Integer directly will it prevent overflow/clipping in such situations? Is this overflow/clipping stuff a rounding problem?
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Old 28th October 2012, 08:09   #12676  |  Link
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Integer will always clip because you can't go over 100%. Which integer mode you use does not matter. Float on the other hand has no limitation on its value range, so you can go above 100% (which in float is 1.0), so a value of 1.1 is perfectly valid. The question just is how other components deal with this, and since DACs are typically integer, it needs to be converted at some point.
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Old 28th October 2012, 08:22   #12677  |  Link
6233638
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Quote:
Originally Posted by Deim0s View Post
I use a Dump Filter from GraphStudioNext. Further open the raw-file in Adobe Audition (Import -> Raw Data).
Thank you, I have now exported this to a raw file (32-bit Float, 48kHz) and also converted that to a 24-bit, 48kHz FLAC using Audacity for easier playback.

http://www.filedropper.com/thunder

There does not appear to be any clipping (this is at 2.24 LFE) but I am still hearing that crackling noise when including any significant amount of LFE in the downmix.

Can anyone confirm that they hear it in these files? I suppose there's always the possibility that it's my audio setup.


I'm also hearing it when I set the LAV mix to:
Center: 0.0000
Surround: 0.0000
LFE: 1.0000
Which makes me think that either something in my audio setup is causing that crackling with LFE, or it's part of the encoded audio.

Last edited by 6233638; 28th October 2012 at 08:25.
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Old 28th October 2012, 08:34   #12678  |  Link
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So normalization scales floats to (0.0, 1.0) to prevent clipping/overflow, right? Shouldn't this always be performed if integer output is explicitly requested, then?

EDIT: Yes, 6233638, i can definitely hear the crackling, but it seems to affect mainly the high frequencies here. No overflow/clipping. Don't think is has something to do with the LFE.

EDIT 2: Yes, LFE channel is still crackling. Peak meter still shows no overflow (or it is to fast for me to see). Crackling still seems to be mainly in the high frequencies. How can that be?

Last edited by bugmen0t; 28th October 2012 at 09:01. Reason: answer 6233638s questions again
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Old 28th October 2012, 08:41   #12679  |  Link
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So of course immediately after that last post, it occurred to me to disable mixing in LAV and just dump a raw 5.1 file instead.

This is the LFE channel on its own: http://www.filedropper.com/lfe

Can anyone confirm the crackling in it, or is something going wrong in my audio setup?


Quote:
Originally Posted by bugmen0t View Post
So normalization scales floats to (0.0, 1.0) to prevent clipping/overflow, right? Shouldn't this always be performed if integer output is explicitly requested, then?
I would assume that's what Normalization does.
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Old 28th October 2012, 08:49   #12680  |  Link
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LAV does not perform normalization. Don't confuse "Matrix normalization" with actual Normalization. Matrix normalization only normalizes the matrix coefficients, so that the sum of the coefficients for one channel never exceeds 1.0. If the audio is spiking over 1.0 before mixing, it will still do that after mixing.
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