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I didn't have the time this we to do some encodign test, but encoding at +3db is not really a solution. |
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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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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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 |
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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I should have said +6db in volume level :) Quote:
-kwag |
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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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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. |
of course... :oops:
sorry and excuse me Dialhot! as was posted about that i only agree and post some informatives links that show the same. Quote:
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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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Just uncheck LFE to LR Channels checkbox. http://www.digitalfaq.com/archives/error.gif |
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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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Thank you. (you can let your post Luis :lol:) |
:D
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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 ) 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 |
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 |
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). |
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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