Welcome to Doom9's Forum, THE in-place to be for everyone interested in DVD conversion.

Before you start posting please read the forum rules. By posting to this forum you agree to abide by the rules.

 

Go Back   Doom9's Forum > Capturing and Editing Video > New and alternative a/v containers

Reply
 
Thread Tools Search this Thread Display Modes
Old 23rd March 2015, 15:24   #18821  |  Link
leeperry
Kid for Today
 
Join Date: Aug 2004
Posts: 3,463
Quote:
Originally Posted by Liandri View Post
I have a 5.1 headphones and some videos come with 5.1 channels, but in most of them I got a bad impression due to voices being too silent in 5.1 mode.
Sounds like what you want is increase the coeff of the center channel, ffdshow audio can easily post-process LAV but it might just be that the mixing matrix of your 5.1'ish phone is wonky to begin with.

And going back to lipsync, a 1ms difference is a lot with headphones IME......I mean ±1 ms and it's easy to see that sync is broken, I'd love a decimal figure tbh.
leeperry is offline   Reply With Quote
Old 23rd March 2015, 16:00   #18822  |  Link
huhn
Registered User
 
Join Date: Oct 2012
Posts: 5,833
Quote:
Originally Posted by leeperry View Post
Sounds like what you want is increase the coeff of the center channel, ffdshow audio can easily post-process LAV but it might just be that the mixing matrix of your 5.1'ish phone is wonky to begin with.

And going back to lipsync, a 1ms difference is a lot with headphones IME......I mean ±1 ms and it's easy to see that sync is broken, I'd love a decimal figure tbh.
how will you notice 1 ms of a difference there is usually only a frame every ~42 ms. even reclock with default settings has a max latency of 100 ms (20% of audio buffer which is 500 ms at default). i mean how do you even know it is 1 ms in the first place.

i wouldn't be shocked that it is totally impossible with current hardware to get a perfect sync better than +/-1 ms.
i mean soundcard/DAC delay, audio renderer delay, leo bodner tester "- Provides accurate measurement within 1 millisecond accuracy." and even that is questionable. and all the clocks in a PC that fight each other...
huhn is offline   Reply With Quote
Old 23rd March 2015, 16:11   #18823  |  Link
leeperry
Kid for Today
 
Join Date: Aug 2004
Posts: 3,463
Quote:
Originally Posted by huhn View Post
i wouldn't be shocked that it is totally impossible with current hardware to get a perfect sync better than +/-1 ms.
I run Invariant TSC in W7SP1 with a 0.5ms granularity together with an async USB DAC(running off two low-jitter clocks for 44.1/48kHz multiples), using a 1m long DVI cable to my TV and a 1.8m long cable to my headphones. I also use a hub that reclocks USB data for better timings and Slysoft Reclock in 88.2kHz Exclusive mode WASAPI prevails(it runs its audio thread in realtime priority BTW).

Fact is that ±1ms delay in LAV kills lipsync in mVR, I'm quite happy with my current rig but I wouldn't mind a decimal figure to play around with that's all

Of course 5.1 loudspeakers in a room would be a completely different story and only a proper AVR calibration could compute the perfect delay required for each speaker to reach the hot spot simultaneously.

Last edited by leeperry; 23rd March 2015 at 16:25.
leeperry is offline   Reply With Quote
Old 23rd March 2015, 16:30   #18824  |  Link
nevcairiel
Registered Developer
 
Join Date: Mar 2010
Location: Hamburg/Germany
Posts: 9,794
A 0ms perfect sync is practically impossible with DirectShow, just due to the way it syncs playback and how video presentation works. Anyone that claims 1ms makes a difference is delusional.
__________________
LAV Filters - open source ffmpeg based media splitter and decoders
nevcairiel is offline   Reply With Quote
Old 23rd March 2015, 16:39   #18825  |  Link
leeperry
Kid for Today
 
Join Date: Aug 2004
Posts: 3,463
Right, 42 ms @24Hz ought to be enough for anybody LOL gotta love humorous forums ^^
leeperry is offline   Reply With Quote
Old 23rd March 2015, 16:56   #18826  |  Link
nevcairiel
Registered Developer
 
Join Date: Mar 2010
Location: Hamburg/Germany
Posts: 9,794
Its not about whats enough or whatnot, its about technical fact, and the fact is that DirectShow is not capable to give you 100% perfect 0ms sync. In fact, I would be surprised if any PC player can really do that, simply because of how VSYNC works.
Knowing that your sync is never going to be precise to the millisecond, its also clear that a 1ms difference is not going to change anything, since its within the margin of error we're going to have anyway.
__________________
LAV Filters - open source ffmpeg based media splitter and decoders

Last edited by nevcairiel; 23rd March 2015 at 17:01.
nevcairiel is offline   Reply With Quote
Old 23rd March 2015, 17:14   #18827  |  Link
Sulik
Registered User
 
Join Date: Jan 2002
Location: San Jose, CA
Posts: 215
Here is a 1ms sync adjustment you can do yourself: just move one step back or one step forward (sound waves travel at about one foot (~36cm) per second)
Sulik is offline   Reply With Quote
Old 23rd March 2015, 18:15   #18828  |  Link
James Freeman
Registered User
 
Join Date: Sep 2013
Posts: 919
Quote:
Originally Posted by Sulik View Post
..sound waves travel at about one foot (~36cm) per second)
man thats sllloooowwww....
__________________
System: i7 3770K, GTX660, Win7 64bit, Panasonic ST60, Dell U2410.
James Freeman is offline   Reply With Quote
Old 23rd March 2015, 18:51   #18829  |  Link
Liandri
Registered User
 
Join Date: Aug 2012
Posts: 14
I'd like to correct myself about 2.1 bass feature. Choosing 2.1 in KMP Audio is not the only thing that is required to make it work. It has a checkbox for "LFE Redirection", which makes it work for stereo source. Same with PotPlayer.
Liandri is offline   Reply With Quote
Old 23rd March 2015, 18:52   #18830  |  Link
jmonier
Registered User
 
Join Date: Oct 2008
Posts: 183
Quote:
Originally Posted by Sulik View Post
Here is a 1ms sync adjustment you can do yourself: just move one step back or one step forward (sound waves travel at about one foot (~36cm) per second)
I think you mean about 1 ft/ms.

Last edited by jmonier; 23rd March 2015 at 18:55.
jmonier is offline   Reply With Quote
Old 23rd March 2015, 18:54   #18831  |  Link
nevcairiel
Registered Developer
 
Join Date: Mar 2010
Location: Hamburg/Germany
Posts: 9,794
He probably just meant to say milliseconds, since thats what we talked about.
__________________
LAV Filters - open source ffmpeg based media splitter and decoders
nevcairiel is offline   Reply With Quote
Old 23rd March 2015, 19:00   #18832  |  Link
huhn
Registered User
 
Join Date: Oct 2012
Posts: 5,833
Quote:
Originally Posted by Liandri View Post
I'd like to correct myself about 2.1 bass feature. Choosing 2.1 in KMP Audio is not the only thing that is required to make it work. It has a checkbox for "LFE Redirection", which makes it work for stereo source. Same with PotPlayer.
this may work in your sound driver too. you may have a LFE redirection or you just have to set your speaker to small/not full range or something like that. my creative/ESI soundcards have options like that.
huhn is offline   Reply With Quote
Old 23rd March 2015, 20:09   #18833  |  Link
e-t172
Registered User
 
Join Date: Jan 2008
Posts: 568
Quote:
Originally Posted by leeperry View Post
Fact is that ±1ms delay in LAV kills lipsync in mVR.
Quote:
Originally Posted by leeperry View Post
Right, 42 ms @24Hz ought to be enough for anybody LOL gotta love humorous forums ^^
Your claim is absolutely ridiculous (not quite as ridiculous as your previous claims about high sample rate audio, but close). 1 ms is equivalent to sound traveling 35 cm. Are you seriously claiming that you can perceive the difference in lipsync just by moving 35 centimeters away from the speakers? For reference, my arms are longer than that. That must cause you real trouble in the real world, because that means you would notice lipsync issues just by looking at people talking to you from the other end of a room…

Also, the international standard for sync requirements in professional video production, ITU-R BT.1359, states that detection thresholds are as high as 45 ms, while the recommendation allows for delays as high as 90 ms (end to end). This means that your personal sync requirements of <1 ms are two orders of magnitude more stringent than the equipment the material is produced with in the first place.

Last edited by e-t172; 23rd March 2015 at 20:32.
e-t172 is offline   Reply With Quote
Old 23rd March 2015, 20:52   #18834  |  Link
infiniter
Registered User
 
Join Date: Dec 2003
Posts: 36
Hello! I searched this thread, but couldn't find the proper answer, so I ask here.

I got LAV Splitter installer 0.64 downloaded and only installed the splitter (don't need the rest). I got Windows 7 Ultimate x64. Neither the x86, nor the x64 version of lavsplitter.ax would install with regsvr32 lavsplitter.ax. RegSvr32 always said "cannot load module".
I used of course admin shell and there is no former version of LAV splitter on my system. Is this because the installer put all the files in the x86 path of C:\program files? Any idea? Thanks in advance.

Update: I found a ZIP (https://1f0.de/downloads/) of the splitter, unpacked to C:\program files\lavcodecs, ran the install_splitter.bat and it worked. Though the BAT actually does the same as I tried before. It can only mean, that when using the installer of LAVSplitter and only selecting "Splitter (x86)" and "Splitter (x64)" it does not install all files necessary. Doh!

Last edited by infiniter; 23rd March 2015 at 21:16. Reason: Found solution.
infiniter is offline   Reply With Quote
Old 23rd March 2015, 21:39   #18835  |  Link
Liandri
Registered User
 
Join Date: Aug 2012
Posts: 14
Quote:
Originally Posted by huhn View Post
this may work in your sound driver too. you may have a LFE redirection or you just have to set your speaker to small/not full range or something like that. my creative/ESI soundcards have options like that.
The device is from C-Media, and as far as I can see it doesn't have such options.
Liandri is offline   Reply With Quote
Old 24th March 2015, 02:00   #18836  |  Link
Arm3nian
Registered User
 
Join Date: Jul 2014
Location: Las Vegas
Posts: 177
Quote:
Originally Posted by huhn View Post
a LFE crossover is defiantly not a bad idea. i recommend to add this with the panned addition of up mixing. https://code.google.com/p/lavfilters...ary%20Modified

and there are a lot of headphones with multiply driver and even a driver for LFE. here an example picture: http://www.tacticalsites.com/~squint...enearphone.jpg 4 is the LFE.
these system have a huge problem with bass reproduction when a source is played that's isn't using a LFE channel so a LFE crossover for stereo is a must have for these.

a lot of smaller 5.1 or 7.1 system work the same way and get into the same problem without a LFE channel when they are used analog.
If a system has problems with bass when a LFE channel is not present then that system is a piece of crap...

Almost all content, besides the audio on bluray do not have an LFE channel. So when listening to music or watching videos on youtube there would no bass... but of course there is. The bass is either reproduced by the left and right speakers or fed into to the subwoofer at the crossover frequency. The purpose of an LFE channel is to isolate specific sound effects from the other sounds, so the subwoofer reproduces them instead of the other speakers (since it does a better job). LFE can contain frequencies up to 120hz. An LFE channel can allow you have a higher crossover frequency if your main speakers cannot reproduce low frequency sounds. Headphones do not need this feature as they can reproduce the audible range of tones, but some speakers might. Quality systems focus on a specific frequency response for each speaker.

Basically, any quality headphone has 2 drivers, you should know this. What hifi gear have you seen with 8 drivers that costs $80. That makes no sense when a proper headphone with 2 drivers costs more. Any decent headphone can reproduce bass without a dedicated subwoofer driver/cone. Notice how standalone subwoofers have huge cones? You need them to produce the low frequencies at audible volumes. A half inch subwoofer inside a headphone isn't going to provide much benefit compared to the main drivers.

The surround sound aspect does not make much sense either. Notice how a proper surround system has the surround sound speakers on the opposite sides of the other speakers? The auditory system is good in positioning, but it doesn't make up for the fact that the surround speakers are right next to the other ones in a headphone. Any proper headphone will have a good enough sound stage and imaging capabilities to mimic what multiple drivers do in such a confined space.

tldr: Why should software support badly designed hardware that sells purely on marketing?

Quote:
Originally Posted by Liandri View Post
I have Somic E-95 v2010. Marketing or not, it has more than 2 speakers. I can easily feel 5.1 sound if the source has it. And it has a subwoofer, I've checked it and it works - disabled all channels except LFE and I could hear what I usually hear from my sub from Defender Blaze 60 (2.1) system, plus vibration. So my headphones have .1 channel, and if they do not, I still have a working 2.1 system right here. Extra channels are not lost, otherwise I'd not hear something in 5.1 content, but I can hear everything in 2.0, 2.1 and 5.1 modes. I just don't feel like using 5.1. (and please don't suggest that it's broken)

I can clearly hear the difference. I've checked with the same content. Even for stereo-only content, when KMP Audio is set to 2.1 - LFE channel works (as for how it works, I'm not sure - it probably gets added by mixing both channels) and I can feel bass with vibration, with 2.0 - not. With LAV Audio I can only feel that in 5.1 and above.

As I already mentioned, moving LFE slider did not do anything when Stereo mode was active. If you are suggesting to use 5.1 mode, I can only repeat that I prefer using 2.0/2.1 over 5.1 right now.

LFE in both my 2.1 system and 5.1 headphones can work with both Stereo and 5.1 content. Even if you say there is no 2.1 content, I don't see why there shouldn't be such an option.


That's too bad. Thanks for the honest answer.
I was not suggesting to use 5.1 but I can see why you are not using it. Your headset has a hardware mixer attached. If you have left the settings on your computer as default, and the voices are too quiet, then the mixer is configured wrong... Blurays are professionally mastered (at least I hope most are), so it should sound correct with no downmixing. I would suggest lowering the volume of all the other channels and leaving the center higher, but from what I can tell the mixer level controls are also the attenuator, meaning you can't amplify after mixing, another limitation of your setup.

Once again, if you are certain the clip you are testing has significant LFE content, and the slider does nothing, then your left and right drivers can't reproduce the sounds. If the "subwoofer" in your headset is not doing anything at all when you have stereo selected in LAV then your headset doesn't even have a passive filter to send to low frequency content to the sub. So if 5.1 doesn't work correctly, and the sub doesn't work unless there is a dedicated feed, then what good is your headset?

Which leads to 2.1. Seems your options for decent bass are to use a directshow filter or to get another pair of headphones
And to be clear, I'm going completely off what you're posting. I don't know how your headset performs, but it seems to me it's causing more trouble than it's worth. You're losing the benefit of the extra speakers for movies, and the bass is weak in content without a dedicated LFE stream. The configuration in your player is basically doing what ffdshow/ac3filter would do, which is either downmixing to 2.1 or upmixing 2.0 to 2.1 for the headset to use the subwoofer/manage the bass properly.

Last edited by Arm3nian; 24th March 2015 at 02:10.
Arm3nian is offline   Reply With Quote
Old 24th March 2015, 07:40   #18837  |  Link
kalston
Registered User
 
Join Date: May 2011
Posts: 164
Quote:
Originally Posted by e-t172 View Post
Your claim is absolutely ridiculous (not quite as ridiculous as your previous claims about high sample rate audio, but close). 1 ms is equivalent to sound traveling 35 cm. Are you seriously claiming that you can perceive the difference in lipsync just by moving 35 centimeters away from the speakers? For reference, my arms are longer than that. That must cause you real trouble in the real world, because that means you would notice lipsync issues just by looking at people talking to you from the other end of a room…

Also, the international standard for sync requirements in professional video production, ITU-R BT.1359, states that detection thresholds are as high as 45 ms, while the recommendation allows for delays as high as 90 ms (end to end). This means that your personal sync requirements of <1 ms are two orders of magnitude more stringent than the equipment the material is produced with in the first place.
Haha, lipsync issues in real life, I hadn't thought of that but that's a fun way of illustrating how ridiculous his claim was.
kalston is offline   Reply With Quote
Old 24th March 2015, 08:21   #18838  |  Link
huhn
Registered User
 
Join Date: Oct 2012
Posts: 5,833
Quote:
Originally Posted by Arm3nian View Post
If a system has problems with bass when a LFE channel is not present then that system is a piece of crap...

Almost all content, besides the audio on bluray do not have an LFE channel. So when listening to music or watching videos on youtube there would no bass... but of course there is. The bass is either reproduced by the left and right speakers or fed into to the subwoofer at the crossover frequency. The purpose of an LFE channel is to isolate specific sound effects from the other sounds, so the subwoofer reproduces them instead of the other speakers (since it does a better job). LFE can contain frequencies up to 120hz. An LFE channel can allow you have a higher crossover frequency if your main speakers cannot reproduce low frequency sounds. Headphones do not need this feature as they can reproduce the audible range of tones, but some speakers might. Quality systems focus on a specific frequency response for each speaker.

Basically, any quality headphone has 2 drivers, you should know this. What hifi gear have you seen with 8 drivers that costs $80. That makes no sense when a proper headphone with 2 drivers costs more. Any decent headphone can reproduce bass without a dedicated subwoofer driver/cone. Notice how standalone subwoofers have huge cones? You need them to produce the low frequencies at audible volumes. A half inch subwoofer inside a headphone isn't going to provide much benefit compared to the main drivers.

The surround sound aspect does not make much sense either. Notice how a proper surround system has the surround sound speakers on the opposite sides of the other speakers? The auditory system is good in positioning, but it doesn't make up for the fact that the surround speakers are right next to the other ones in a headphone. Any proper headphone will have a good enough sound stage and imaging capabilities to mimic what multiple drivers do in such a confined space.

tldr: Why should software support badly designed hardware that sells purely on marketing?
i don't use hardware like this and i have personally no need for such a feature with my beyerdynamics T5p...

that doesn't change the fact that people buy logitech speaker on mass. and a system without digital processing can't do the cross over by it self so a cross over is needed.

if you run an sub with an analog connection without digital processing a crossover is needed it's that simple. and this can happen with a more expensive system too.

luckly some soundcards can do this be them self.

nevcairiel clearly said this is not going to happen so no need to discuss about this here.
huhn is offline   Reply With Quote
Old 24th March 2015, 09:40   #18839  |  Link
nevcairiel
Registered Developer
 
Join Date: Mar 2010
Location: Hamburg/Germany
Posts: 9,794
The next nightly of LAV will feature the libdcadec DTS decoder.
It supports full bitexact DTS-HD decoding of up to 96/24, as well as much broader support for DTS extensions than the ffmpeg decoder.

Most prominently (things the ffmpeg decoder didn't do):
- DTS 96/24 (96 kHz support for lossy DTS streams, and DTS-HD HRA)
- Bitexact DTS-HD MA up to 7.1, 96 kHz, 24-bit

The only missing features I'm aware of right now is support for 192kHz DTS-HD MA, which I hope will be added soon, and DTS Express, which may take a bit longer (but is also much less important).
__________________
LAV Filters - open source ffmpeg based media splitter and decoders

Last edited by nevcairiel; 24th March 2015 at 10:11.
nevcairiel is offline   Reply With Quote
Old 24th March 2015, 10:50   #18840  |  Link
kalston
Registered User
 
Join Date: May 2011
Posts: 164
Awesome! Can't wait to try it.

Last edited by kalston; 24th March 2015 at 10:56.
kalston is offline   Reply With Quote
Reply

Tags
decoders, directshow, filters, splitter

Thread Tools Search this Thread
Search this Thread:

Advanced Search
Display Modes

Posting Rules
You may not post new threads
You may not post replies
You may not post attachments
You may not edit your posts

BB code is On
Smilies are On
[IMG] code is On
HTML code is Off

Forum Jump


All times are GMT +1. The time now is 15:04.


Powered by vBulletin® Version 3.8.11
Copyright ©2000 - 2019, vBulletin Solutions Inc.