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Producing a correct 5.1 ac3 stream using free tools? (part2)
as for low volume in ac3 output from besweet, I have noticed this is a problem when encoding in 5.1 channels, but NOT with 2 channels. I can't remember how I ended up getting this fixed with 5.1 channels, but I think sometimes BeSweet likes to force some odd parameters into azid that screws up the volume... If I encounter another 5.1 source that needs converting maybe I'll figure out again how I fixed this. :P
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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 setereo 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 :arrow: 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! :wink: @ who care... i will that post stay alive now! :wink: like wroted somewhere.."use your judgement" but judge with equality and not as dictator as i was called and delete only my opinion if you want but not my informations about the audio. one more time i write here....don' t be coward and don't do exact what you accuse me...you still need to change to be a man..delete only this part cos i'm right that you read before delete!...thanks! :roll: |
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If you take anb ac3 file demuxed from a DVD and you convert it with besweet or ac3 machine in an otehr ac3 file (let say that you do not modify anything, just for doing a test) and then you listen both file on your pc the volume of the new file will be lower than the original one ! |
Back from vacations .....
Back from vacations ..... If going with an 5.1 DVD into a SAP where a "Simple Stereo Amplifier" is connected, the SAP can be settet up to output a downmixed 5.1 to Stereo Signal out of the SAP "just stereo out" to that 2 channel Stereo amplifier. And here it counts how good the quality of that downmixing of the SAP is given. If a surround-matrix 2.0ch encoded DVD is played back by a 5.1 capable DDdecoder connetcted and set to 5.1 Speakers then the surroundmatrix of that 2.0ch audio is used to split the 2.0ch to descrambled 5.1 sound components which will be delivered to the 5 internal "amplifiers" of that DDdecoder, the LFE chanel often (depends on the DDdecoder model) is provided as a low voltage output signal (unamplified) so an active Subwoofer can be feeded by that one. Quote:
Just generate a bat file from the commandline contend shown in besweet and so in that bat file you easely can set the right parameters to azid etc. You are right that theres a bug that if you go in further seetings, it can happen that the azid part of the final commandline is resettet in a wrong way. AC3machine is also a nice choice but the latest options of these used dlls are not fully supported or even up to date ;-) Maybe thats why phil gots everytime still that less volume output?? But I do think that you (phil) are aware of that besweet bug. |
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Well AFAIK (IMHO) the latest azid.dll is newer! Do check that as diff. parameter names for ds downmixing/outputting are used now. :-)
Just get my latest "Packshot with everything" package, in there you get the latest modules of all types. btw: in the case of the latest azid.exe compile you could be defenitely right ... but lets check that. (PS: You just copied my last passage of the previous post? Did your purpose was something diff.? ;-) ) And the newest beta of headac3he "should" also come with the most recent versions? |
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Note : I wonder now what we focus in the azid.dll. the problem is in ac3enc.dll, isn't it ? |
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But practize is better than theory .... and thats why I wanted to figure out all options !in commandline mode! (so nothing "could" be messed up by possible buggy GUIs) ... and also beside that, that workout via 3pass and .exe modules and finally LGPL compiled ffmpeg.exe as wav in and liba52 AC3 output. ;-) .... ähhhhm I just recognised we are again in the wrong topic for that *lol |
Is there any program that would give any detailed information about the ac3 file? It might be useful to know what level the dialog normalization and dynamic range compression are set. Also the different mix parameters are unknown with ac3enc.
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Do you mean azid ? As ac3enc just encodes what has been feeded with. Quote:
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I don't know how much detailed it information is. |
Inc,
I know those azid parameters, but doesn't the ac3 stream itself also contain the DRC and dialnorm settings (not to mention the BSI section which seems to be very important)? At least you can set those two in SoftEncode. |
I think thats the point of that "free" ac3enc that you cant tweak those settings as it "could" be that the internal routines of ac3enc do implementate these factor as default and for this using internal default values? It could be? Im not really shure.
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@ Phil ...
could you "try" this commandline via bat file for reencoding an ac3: Code:
"C:\PATH_TO_BeSweet.exe" -core( -input "X:\PATH_TO_YOUR_INPUT.ac3" -output "X:\PATH_TO_YOUR_OUTPUT.ac3" ) -azid( -g max) -ota( -hybridgain ) -ac3enc( -b 384 -6ch )In case of overflows ...: Quote:
Latest Besweet files where taken for the workouts above. |
@ Boulder ...
for parsing out AC3 related informations, do get azid.exe from here ... http://www.digital-digest.com/dvd/downloads/azid.html and use this commandline in a bat file: Code:
azid.exe -N -Q -z off "X:\PATH_TO_THE.ac3"Just interesting .... : when parsing a) an orig ac3 and b) a reencoded ac3 using directly besweetGUIs parameters with a bad volume bug ... these are the results: a) orig one: Code:
[00:00:00.000] +------------------------------+b) a reencoded one (with NOTICABLE less volume) Code:
[00:00:00.000] +------------------------------+So I dont know ... could it be that a ac3decoding unit (or also ac3filter etc.) does readout the BSI Infos from an ac3 and therefore such low "flags" do cause a lower volume playback BY THE PLAYBACK UNIT where the audio stream data itself is full ok????? Hmmmm ... |
I had similar results:
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BeSweetThe funny thing is that when I decode the original AC3 track into six mono wave files, I get downmix overflows so the audio is too high and thus the gain is actually zero! And this might be the reason why the BeSweet encoded AC3 is also more silent than the original one as the same happened when I encoded directly from AC3 to AC3. The gain for this track was about 5.5dB when I downmixed to a stereo WAV..and I feel that this 5.5dB was the amount missing from the BeSweet encoded AC3. The track was not ultra silent, but as the 5.1 tracks usually need a gain of more than 10dB, I wouldn't be surprised if they sounded very silent when re-encoded without any gain. When the original AC3 track was encoded to a 2ch AC3 track in BeSweet, there were no downmix overflows whatsoever. If anyone's up to the task, 1) decode an AC3 track to six mono channels without any gain added 2) open them in an audio editor, determine the lowest gain needed from the six tracks, 3) use that value for amplifying every track 4) feed these six tracks to BeSweet (it should work via an .lst file, right?) 5) encode to a 5.1 AC3 track 6) check the volume. I don't have the time right now..I've got to clean the whole apartment :roll: Unemployment sucks :x |
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And thats why I again added that quote related to downmix overflows. To mee it "seems" clear that a 1:1 reencode from 5.1 to 2.0 would mean overflows as you by this would just mix orig separated volumes together and that wuld be the mess. So thats why a correct downmixing incl. dynCompression solves that, thats why AC3 2.0 dwonmixed ones do result better. Now as we do a 1:1 port of 6 channels to an again 6ch Ac3 encode, we shouldnt worry about that (as explained in the quote from azid.exe) I did many tests yesterday on a very loud orig AC3 from KillBill1, the one which peaks in its center channel up to 3.5db!! in original state1 *wow* And that orig volume is very difficult to keep when using azid/ac3enc. So I eneded up to now with the line I quoted above: Code:
"C:\PATH_TO_BeSweet.exe" -core( -input "X:\PATH_TO_YOUR_INPUT.ac3" -output "X:\PATH_TO_YOUR_OUTPUT.ac3"* ) -azid( -g max) -ota( -hybridgain ) -ac3enc( -b 384 -6ch )The volume resulted up to the same level in the re-encode! (a minimal less, almost not noticable) It "could" be that the "hybridgain" does add a value or changes the value of a flag to the reencoded ac3??? I dont know. I dint recognised any distrotions in the duplicated center 3.5db parts, but I'll make a comparison in a waveform previewer this Weekend. Quote:
The point is that for instance the KillBill ac3 comes with peaks up to 3.5db! and that means no matter whats used for a 0db max changing :arrow: a pre-processing from 3.5db to 0db does end up more silent :wink: :wink: So we should try to a) keep the origs db peaks and doing a direct 1:1 volume copy to the reencode (with overflows). Or doing that reencoding using the line above with "... azid( -g max) -ota( -hybridgain ) ..." ... testing testing testing :) |
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Some reading :) : http://forum.doom9.org/showthread.ph...threadid=56020 http://www.dolby.com/pro/digaudio/pa...tandards.S.pdf http://www.dolby.com/tech/L.mn.0002.DDPEG1.pdf If I did understand it, the stream itself is OK but some metadata is added to tell the decoder how much to attenuate, how much to compress, the mix level of each channel etc., as you suspected. The Dialog Normalization works as follows (as I understood it): You want the stream to be normalized so that you use full scale at the highest peak and you have better S/N. That's what the -azid( -g max) -ota( -hybridgain ) does. But, if you do this for a couple of movies, you notice some movies will seem louder and some will seem quieter, because you normalized the loudest peak of the movies, not the average level. The idea of Dialog Normalization is to correct this so that you shouldn't need to tweak the volume knob in your amp everytime you change a disc. There's a specified target of -31dB (relative to full scale) for dialogs. This means you want the dialogs for every movie to be -31dB. If you're dealing with music, then passages that should have the same loudness as a dialog should be at -31 dB. So in the studio they pick the maximized wav and measure the loudness of a dialog. Let's say it's -27dB. When encoding to ac3 they set the dialnorm to -27dB. That's louder than the target (-27dB is 4 dB above -31dB) so the decoder will attenuate it by 4dB so that the dialogs will sound at -31dB. If you want the ac3 to sound as loud as the input wav, you have to set the dialnorm to -31 dB, so that the decoder will not need to attenuate anything. If it is like this, then one could tweak the loudness, surround levels etc. of an ac3 file by just editing the metadata, like mp3gain does for mp3. That would be an useful application :) Ac3_patch or something like that. When transcoding an ac3 you could try the following: in azid, use the command line switches to override BSI info and decode all channels to "full range" wavs. Then apply the necessary gain/attenuation to each channel so that the fixed BSI in ac3enc.dll are compensated and the end result matches the original file. This could be problably automated in a batch file or gui. |
Inc wrote:
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Code: But using Softencode the volume is the same as the original. -Anerboda |
@ GFR
Thanks for that very informative Post. :) In Azid you only can override BSI Informations when decoding to 2/x channels, means IF a downmixing process is choosen. 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. 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. |
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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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