Quantcast Producing a Correct 5.1 AC3 Stream Using Free Tools? (Part2) - Page 2 - digitalFAQ.com Forums [Archives]
  #21  
11-08-2004, 04:30 AM
Dialhot Dialhot is offline
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Quote:
Originally Posted by incredible
So, to keep the orig volume we should also try to rise the gain, even over the 0db mark, as 3db seems still non distorting.
Not really a solution for me ears. Don't forget that gain is also added on the receiver to adapt to the speaker configuration. In my case, center channel is already to +3db. As you say, there is no distort but... +3db +3db = +6db. And THEN the distortion can be heard on certain scenes.

I didn't have the time this we to do some encodign test, but encoding at +3db is not really a solution.
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  #22  
11-08-2004, 05:09 AM
incredible incredible is offline
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Well actually I recognised that some DVDs got in their orig state peaks up to 4db. So these ones you also "would" overgaining if you see it like that

But your amplifier does not gain the volume when increasing the gain
Theres a constant amplifiying from in to out and you do only do set the input level from your sound input according to the channels.

Thats the same if youre using a PA Mosfet-amplifier as there also you only adjust the input-levels of what will be amplified.

So a problem would be: If your source coding ends up in a distortion, then shurely hearing loud a distotioned source it will sound even more horrible.
(Remember these Gabba-Techno-200bpm Hardcore songs? There they did apply to that RolandTR909 Kickdrum a distortion effect so it sounds more brutal, but WONT harm to your speakers.

BUT!: Even if you got a mega-clean excellent mastered Sound input like from an orig CD and you force the increasing of the "input adjust" (= Volume on your Amp.) too much, then the internal technical compounts cant actually handle that input and "clipping" will be the result.
That kind of clipping "could" exist even before your ears can recognise it. So In case of clipping the sound weave change from DC to AC (I hope I dont mix it up now). The technical compounts by that of your speakers will do heat up and finally your speakers are dead if you force the "clipping" to much and too long.

So thats why its even better for speakers IF your amp gots more power than the speaker itself, as by that the amp gots more reserves to void "technical" clipping
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  #23  
11-08-2004, 05:50 AM
GFR GFR is offline
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Quote:
Originally Posted by incredible
@ GFR

I noticed the following:

When decoding an AC3, some DVDs got AC3s which do exceed a lot the 0db mark! Like mentioned above in case of KillBill where the center peaks up to 3.5db! So no matter if -g max in azid used or a normalize to 100% (=0db) in an external Audio Utility, the volume by this will be decreased anyway.
I think this +3.5dB comes when you feed a maximized wav to the encoder, and the encoder will then quantize in the frequency domain. When you decode it back to time domain overshoots can occur. The same thing happens with MP3 (see the MP3Gain software). Since the metadata in the BSI is telling the decoder to attenuate a few dB, the decoder can be implemented in such a way that it won't clip.

Also, if 0dB is a full scale sinusoid, then +3.5dB doesn't mean it goes over full scale (it can't, of course - digtal clips hard, it's not a soft saturation like a tape recorder). It's just that the waveform is more "busy" than a sine wave. For example, a full scale square wave will be over 0dB if a full scale sine is 0dB.
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  #24  
11-08-2004, 06:12 AM
Dialhot Dialhot is offline
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Quote:
Originally Posted by incredible
too much, then the internal technical compounts cant actually handle that input and "clipping" will be the result.
For me clipping equals distortion. Else, what is the distortion definition according to you ?

Quote:
That kind of clipping "could" exist even before your ears can recognise it.
Yes it is. And if you encode at +3db you already have clipping that you do not notice and that is this "clipped" curve that you give to your amp, that will "clip" it more and then the distortion will be heard. That's why I told that "+3db +3db = +6db" even if this has no real "mathematical" meanings.

Quote:
So In case of clipping the sound weave change from DC to AC (I hope I dont mix it up now). The technical compounts by that of your speakers will do heat up and finally your speakers are dead if you force the "clipping" to much and too long.
Don't worry, all my Home Theater installation is far well protected against this
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  #25  
11-08-2004, 11:36 AM
kwag kwag is offline
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Just a little note on decibels.
Remember that a change of +3DB, is twice the volume level.
And the same for negative values. A change of -3DB is exactly half the volume level.


Correction:
A change of +6DB, is twice the volume level

-kwag
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  #26  
11-08-2004, 12:35 PM
GFR GFR is offline
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No,

+3dB is twice the POWER level.

Twice the volume (loudness) level is a couple more dBs (something between +6dB to +10dB).
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  #27  
11-08-2004, 07:11 PM
kwag kwag is offline
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Quote:
Originally Posted by GFR
No,

+3dB is twice the POWER level.
Yes, indeed
I should have said +6db in volume level
Quote:

Twice the volume (loudness) level is a couple more dBs (something between +6dB to +10dB).
It's exactly 6db change for volume level, and 3db change for power level

-kwag
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  #28  
11-14-2004, 02:43 AM
jeo jeo is offline
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Quote:
Originally Posted by jorel
AC3-5.1 give good volume ONLY in dolby surround receiver cos that receivers have 5 or 6(depend of the model) separeted amplifiers for EACH channel.
then using the same dolby surround as stereo and as 5.1 decoder, ALWAYS the 5.1 have better volume cos in stereo only 2 of the reminders is working.
i really will that everybody understand what i wrote thousands times:
in stereo,we have 2 channels working with the whole signal but in
dolby surround we have MORE 4 speakers working
using the same source ac3-5.1 and the SAME volume.

my hobby is audio for 44 years and my work is eletronic for 32 years.
i know what i posted and IF someone do tests following exactly what i wrote,will see ..or better, listen the strong differences in quality and in the volume.
if nobody understood what i wrote....the only way is....a single calculator show that 1 source for 2 targets give less power that for 6 targets...
is so hard to understand?...i can' t trust!

example:
2 * 10 Watts always give less potence that 6 * 10 Watts
using the same source and the same volume for 2 or 6 channels...
if nobody understood....excuse me..i'm really sorry!

ac3 5-1 when you encode to mp2 or stereo loose lots of details....do tests putting and removing the surround.....in ac3 they increase...in stereo they sometimes loose no matter if you're using besweet,headac3.....etc.
only tests can show and proove for you, not my poor english!
agree!

http://www.audioholics.com/techtips/...iverpower2.php
http://www.audioholics.com/techtips/...iverpower3.php
http://www.audioholics.com/techtips/...eiverpower.php
http://www.audioholics.com/techtips/...nelsdriven.php

read too: "No Subwoofer Output in 2-Channel Mode"
http://www.audioholics.com/techtips/...tin2channe.php
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  #29  
11-14-2004, 04:57 PM
Dialhot Dialhot is offline
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Quote:
Originally Posted by jeo
agree!
May I ask you to read completly a thread before to start to answer to it ?
We (I) already said to Jorel that, even if true, this has nothing to do with the current problem.

We are dealing here with volume loss during 5.1 -> 5.1 process with ac3enc based encoder.
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  #30  
11-14-2004, 05:24 PM
jeo jeo is offline
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of course...
sorry and excuse me Dialhot!
as was posted about that i only agree and post some informatives links that show the same.

Quote:
Originally Posted by Dialhot
We are dealing here with volume loss during 5.1 -> 5.1 process with ac3enc based encoder.
right, i saw that too cos i read all and this thread is very impressive.
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  #31  
12-30-2004, 11:26 AM
Dialhot Dialhot is offline
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I found the solution !

It was the only thing I never tried : the Postgain
Hybrigain screw too much the dynamic so just forget it.

Do not use AC3Machine, use BesweetGui. In AC3 panel select "do 5.1 AC3". This select automatically the parameters " -azid( -g 0.95 -L -3db )". I would like to not have the "-L" but it's not very important. The gain to 95% is good as it prevents to have distort on peaks.
Go then on main tab and select "Postgain 100%" (that's add "-ota(-G 1)".

I did only one try, so let me know if you suffer some problems I didn't heard.
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  #32  
12-30-2004, 11:49 AM
Prodater64 Prodater64 is offline
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Quote:
Originally Posted by Dialhot
I would like to not have the "-L" but it's not very important.
You can get rid of that parameter in AZID1 options.
Just uncheck LFE to LR Channels checkbox.

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  #33  
12-30-2004, 11:57 AM
Dialhot Dialhot is offline
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Quote:
Originally Posted by Prodater64
You can get rid of that parameter in AZID1 options.
Just uncheck LFE to LR Channels checkbox.
Pro, try.. and then you'll remove your post
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  #34  
12-30-2004, 12:02 PM
Boulder Boulder is offline
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If you check the surround mix mode, then the LFE option won't appear when you leave the Azid settings. It shouldn't matter what the surround settings are as they are meant for downmixing to two channels. I suppose that's just a bug and not intentional.

EDIT: Hmm, it only reappears if no options are enabled in Azid. Phil, what happens on your system?
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  #35  
12-30-2004, 12:08 PM
Dialhot Dialhot is offline
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Quote:
Originally Posted by Boulder
EDIT: Hmm, it only reappears if no options are enabled in Azid. Phil, what happens on your system?
The same . You're right : if something else is selected (as a downmixing mode), then the "-L" does not reappears once I leave the azid1 tab

Thank you.

(you can let your post Luis )
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  #36  
12-30-2004, 12:21 PM
Prodater64 Prodater64 is offline
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  #37  
06-05-2005, 04:43 PM
rds_correia rds_correia is offline
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Hi Phil and guys,
This is what I've used.
I take it that this is what you were talking about, right Phil?
Code:
BeSweet.exe -core( -input "E:\MATRIX_I\AC3\matrix_i AC3 T01 3_2ch 384Kbps DELAY -23ms.ac3" -output "E:\MATRIX_I\AC3\matrix_i AC3 T01 3_2ch 384Kbps DELAY -23ms-new.ac3" -logfile "E:\MATRIX_I\AC3\matrix_i AC3 T01 3_2ch 384Kbps DELAY -23ms.log" ) -azid( -g 0.95 ) -ota( -d -23 -G 1 ) -ac3enc( -b 224 -6ch )
Do notice that I didn't include the -L -3db on the azid parameter like you pointed out.
I don't even know what those do but since I'm only searching to kill the low volume and I have a feeling that those won't do much about my problem...
I'll post the results in a minute after besweet finishes.
Cheers
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  #38  
06-05-2005, 05:35 PM
rds_correia rds_correia is offline
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Nope, no luck here.
Still much lower than the original though somewhat better than my previous trials.
I don't have Cooledit or softencode.
Can't buy them and I can't understand how DSPguru doesn't try to fix this damn thing...
I know it's not directly BeSweet related.
But it gets in the way of BeSweet when it tries to transcode 5.1 material.
If he understands the problem and he knows his coding then he could have done something already.
Or at least put the pressure on the FFmpeg guys...
Any more ideas?

@Zyphon,
How about your packshot transcodings?
Do they get exactly the same volume as the original ac3-5.1?

Cheers
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  #39  
06-05-2005, 06:21 PM
Dialhot Dialhot is offline
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Rui,
You missed something into your line. I told to use the postgain. YOu should have a "-ota(G 1)" in your command line. Without this you can't have a correct result.

Note : -L sends the .1 to the front channel and on the subwoofer. You don't have normal 5.1 (the bass are louder than original).
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  #40  
06-05-2005, 06:35 PM
rds_correia rds_correia is offline
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Hi Phil ,
But I do.
Check my post again.
There is -ota( -d -23 -G 1 ).
Isn't that what you advised?
It just has a -d -23 because of the ac3 track delay.
Cheers
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