View Full Version : BeSweet's ac3enc creates extremly low-volume ac3


Koke
13th September 2002, 17:48
Is it possible that reencoded AC3 (down to 224 kbps)
sounds too quiet. Or is it my audio setup?
I used AC3machine default settings.
Help please.

DSPguru
13th September 2002, 19:54
:logfile:

stargazer
14th September 2002, 00:28
Is it possible that reencoded AC3 (down to 224 kbps)
sounds too quiet. Or is it my audio setup?
I used AC3machine default settings.
Help please

I have the same problem (the lower the bitrate, the quieter the sound), and I've tried to solve it with extra gaining, but even 12 db gain on 448 to 320 transcoding was enough only to reach 70-80% of the original volume level.

here is the log (this is 448 to 320, but when output bitrate is lower, sound is much quieter)


BeSweet v1.4RC7 by DSPguru.
--------------------------
Using azid.dll v1.8 (b825) by Midas (midas@egon.gyaloglo.hu).
Using AC3enc.dll v0.2 by Gerard Lantau & Dg (http://ffmpeg.org).

Logging start : 09/13/02 , 12:07:40.

G:\mali_programi\BeSweetv1.4RC7\BeSweet.exe -core( -input G:\Documents and Settings\XliQ Stargazer\My Documents\avi\pottie AC3 T01 3_2ch 448Kbps DELAY 0ms.ac3 -output G:\Documents and Settings\XliQ Stargazer\My Documents\avi\pottie AC3 T01 3_2ch 256Kbps DELAY 0msnew.ac3 -logfilea BeSweet.log ) -azid( -g 12db ) -ota( -d 0 ) -ac3enc( -b 320 )

[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : G:\Documents and Settings\XliQ Stargazer\My Documents\avi\pottie AC3 T01 3_2ch 448Kbps DELAY 0ms.ac3
[00:00:00:000] | Output: G:\Documents and Settings\XliQ Stargazer\My Documents\avi\pottie AC3 T01 3_2ch 256Kbps DELAY 0msnew.ac3
[00:00:00:000] | Floating-Point Process: No
[00:00:00:000] +-------- AZID -------
[00:00:00:000] | Total Gain: 12.0dB, Compression: None
[00:00:00:000] | LFE levels: To LR -INF, To LFE 0.0dB
[00:00:00:000] | Center mix level: BSI
[00:00:00:000] | Surround mix level: BSI
[00:00:00:000] | Dialog normalization: No
[00:00:00:000] | Rear channels filtering: No
[00:00:00:000] | Source Sample-Rate: 48.0KHz
[00:00:00:000] +------- AC3ENC ------
[00:00:00:000] | Bitrate method : CBR
[00:00:00:000] | AC3 bitrate : 320
[00:00:00:000] | Channels Mode : 5.1
[00:00:00:000] | Error Protection: Yes
[00:00:00:000] +---------------------

the rest of log file looks something like this:
...
[00:00:18.282] W7: Downmix overflow (1: +1.5dB)
[00:00:18.288] W7: Downmix overflow (1: +1.3dB)
[00:00:18.293] W7: Downmix overflow (1: +1.7dB)
....

Bobby
14th September 2002, 02:56
Hi guys,
I fix this low volume problem by using ac3machine and enabling "Dynamic compression" @ normal, and also enabling "Auto find maximum gain".
I must add though that I've only tested this when transcoding a 5.1 ac3 track to a lower bitrate (192kbps) 2.0 ac3 track, But I'm guessing that it is the same principle's for transcoding 5.1 to lower bitrate 5.1
Hope that helps ya ;)

DSPguru
14th September 2002, 03:02
in fact, i've noticed that property of ac3enc too.
i believe i can change that, but it will take me some time. i'm now working on other stuff..

theReal
14th September 2002, 11:40
I also noticed this behaviour on a re-encode from 448kbit 5.1 to 320kbit 5.1 - in the end I was using azid's auto-find max. gain and normal dynamic compression, but I think it resulted in some distortions in various scenes (I don't trust the azid auto-gain anyways, it always seems to distort the sound - is that possible?)

stargazer
15th September 2002, 01:53
*Originally posted by Bobby *
Hi guys,
I fix this low volume problem by using ac3machine and enabling "Dynamic compression" @ normal, and also enabling "Auto find maximum gain".


I don't like "Dynamic compression" or any sort of compressors that reduce dynamic range, but i tried "Auto find.." and it seems that it really _only_ search maximum gain and not apply it.


I must add though that I've only tested this when transcoding a 5.1 ac3 track to a lower bitrate (192kbps) 2.0 ac3 track, But I'm guessing that it is the same principle's for transcoding 5.1 to lower bitrate 5.1
Hope that helps ya ;)

No, it's not the same principle because 2 channel 192 kbps audio is not so low bitrate (96 kbps/channel) , and in 448 5.1 you have 448/6=74.6 per channel which is lower bitrate than yours used in 2.0 ac3 track...

DSPguru
15th September 2002, 02:16
maximal gain is being applied (you can simply test it by not using '-g max'), the issue belongs to ac3enc.dll. i even sorta find it in the source-code, but i don't have the time to fix it currently.
DRC, Gain, Normalize and other methods are always relevant, but in this case there's a more basic problem.

MaTTeR
20th September 2002, 11:31
I turn every feature off and just do a straight transcode, otherwise I too have heard the distortions. Glad to hear the low volume issue was tracked down.

To avoid distortion and/or artifacts my command line usually looks like-

"C:\Program Files\BeSweetv1.4RC8\BeSweet.exe" -core( -input "d:\false AC3 T01 3_2ch 448Kbps DELAY 0ms.ac3" -output "z:\false_256kbps.ac3" -logfile "C:\BeSweet.log" ) -azid( --maximize ) -ac3enc( -b 256 ) -profile( ~~~~~ Default Profile ~~~~~ )

quantum
17th November 2002, 05:56
I've got this same low volume problem. I downsampled a 5.1 channel to 224 or 256 using default settings in AC3machine and the resulting volume is much lower, almost inaudible when played at the same volume as the original.

knight_inferno
19th November 2002, 03:52
Hello ^^
For the first time last night i used AC3Machine to create a 5.1 ac3 448, down to 256 5.1..i hadn't done anything like it before (Using AC3's and all in AVI's) after testing out the sound by putting it into the avi, indeed it was extremely quiet! I followed the doom9 guide on this, and even used the original audio just to check and that was at a decent volume, still a little quiet, but i find DVD's tend to be a little on the quiet side anyhow :rolleyes:

Anyhow, i know that you said it was a problem with azid (I think that was it) and this post was started a couple of months back, has there been some kind of "Fix" since this time that i have been unable to find? :(

Any advice would be great! If i can't do it i might as well do the usual MP3's i always do :D

MaTTeR
19th November 2002, 04:01
Originally posted by knight_inferno
has there been some kind of "Fix" since this time that i have been unable to find? :( AFAIK the problem still exists somewhat, one movie I did last week required me to crank my amp almost all the way up just to hear the dialog. However, some moves re-encode with better loudness/gain for some reason. Then you have my extreme case from last week where it seemed I had lost almost 50% of the original volume:confused:

quantum
19th November 2002, 17:07
My fix for this was to use Softencode to perform the downsampling following an older guide posted here. It may be slower, but the resulting AC3 file has the same volume.

I can't help wondering, if the volume is wrong with the ac3machine/besweet technique, are there other problems that are not so obvious?

brett
21st November 2002, 21:50
After seeing this thread, I compared some trasncoded AC3 tracks I'd done. Tracks I have transcoded from 192kbps Dolby 2.0 to 80 or 96kbps stereo are actually louder. Not many of them are very significantly louder, but they're definitely louder.

I use AC3Machine 0.41 & BeSweetv1.4 to transcode the audio for director's commentary and special features when ripping DVD -> DVD-R. I always leave the main Dolby Digital 5.1 audio track as is, so maybe the quietness people are experiencing is because of the 5.1 -> 2.0 downmix. My settings are usually:

Channels Mode: stereo
Dynamic Compression: normal
Auto Find Maximum Gain: ON
Bitrate: 80 for commentary, 96 for special features

I leave everything else at the default settings. The only problem I've noticed is that if I try to set the bitrate lower than 80, BeSweet gives me a lot of errors while transcoding.

aryan
25th November 2002, 09:37
I fix this low volume problem by using ac3machine and enabling "Dynamic compression" @ normal, and also enabling "Auto find maximum gain".

@Bobby
I must agree with you that this is the solution for solving that problem. I've made my own tests and ONLY when using DC@normal and azid autofind max gain, the results are ok. In all other cases the sound is MUCH quieter. I've used for my test BeSweet 1.5b6 and AC3 from Shrek.

@DSPGuru
I'm watching @ that thread from more than 2 months, and I still waiting for expert advice from you - which settings are really right to get maximum possible quality, and if there is a bug when it will be fixed (and where actually is the bug - in BeSweet or in ac3enc.dll).

@all
I agree with quantum that for now the best result which can be given is with Softencode. If someone wondering where to get this fine software (Sonic Foundry discontinued offering it) here is the link:
link to warez removed
But for christ sake its damn slow :(

And there is something strange I've noticed. When I reencode Shrek AC3 with Softencode, and then load reencoded or original AC3 track back, there are stream settings of the track which are same for both:
Dialog normalization: -27dB
Surround mix: -3dB
Center mix: -3dB
Ofcourse you can set these params when reencoding, and the result is just what u r setting, and I prefer to save original parameters

BUT if I load reencoded with BeSweet Shrek AC3 track I see that stream settings are changed:
Dialog normalization: -31dB
Surround mix: -6dB
Center mix: -4.5dB
That result is no matter what settings u use in BeSweet (for example I've tried to change them in azid, but the result is always the same

I think DSPGuru is the only one here which can explain that strange phenomenon:confused:

And I must say that I cant find difference using my own ear between track reencoded in Softencode, and the same track reencoded in BeSweet(using normal DC, auto find max gain and hybridgain).
Thats y i'm asking all of u do u know some program where i can load them both and compare them graphically (Sonic Foundry 5.0b for example cant open AC3 at all)

And ofcourse if U think my english is bad, dont be angry :cool: ... nevertheless I'm from Bulgaria :cool:

DSPguru
25th November 2002, 19:01
1. sorry guys, i'm too busy to currently work on this.
2. there is no bug with BeSweet.exe
3. true, ac3enc's Dialog normalization,Surround mix,Center mix values are static and never changes. (currently!)

Originally posted by DSPguru
the issue belongs to ac3enc.dll. i even sorta find it in the source-code, but i don't have the time to fix it currently.

aryan
30th November 2002, 09:42
Well, since yesterday (29.11.02) there is a new BeSweet release- 1.5b8.
And guess what ... there is new ac3enc.dll in it!:devil:
I pray that finally DSPGuru fix it:cool:
I'll give it an immediate try and inform you about results.

Also about removing link to Sonic Foundry Soft Encode as warez link, I think that this is DSPGuru last decision right and I'm not going to debate it.
But for myself I cant identify dvd.box as warez site. All the programs you can see there are freeware, and if you look at theirs first page there is poll especially asking for if you want to see shareware programs in the future.
And especially for Soft Encode, Sonic Foundry no longer support it and no longer even offer it. Here is the link you can see this:
http://www.sonicfoundry.com/Products/NewShowProduct.asp?PID=12

I'm not a licensing guru, but in my oppinion when you refuse your own children, they become noones, and thus everyone owns.
The other example is with MS Dos. They no longer support it or sell it and thus owning it is not a crime.
Am I not right?

DSPguru
30th November 2002, 13:35
the new ac3enc.dll is just a faster compile. that's all :(

jamiro
13th December 2002, 01:14
It's been 4 months and this problem is still unsolved, so there's my question: DSPguru are you able to fix this problem or not?

"(...) the issue belongs to ac3enc.dll. i even sorta find it in the source-code, but i don't have the time to fix it currently.
DRC, Gain, Normalize and other methods are always relevant, but in this case there's a more basic problem."

If it's a "basic problem" then please solve it finally because many ppl are waiting for solution.

MaTTeR
13th December 2002, 01:58
No need to be demanding, Dg's free time is obviously limited. Someone already pointed out a working solution above which is to use Sonic Foundry's Soft Encode. It work's great but only slower than BeSweet:)

theReal
13th December 2002, 02:41
And especially for Soft Encode, Sonic Foundry no longer support it and no longer even offer it. I think it is allowed by international law to spread music which has a certain age and which is not published anymore. There has been a site publishing only old, rare and unavailable reggae albums - and nobody would sue the owner because he stated clearly that if anyone wanted to make use of the songs' copyrights, he would remove the mp3s from his site immediately.
But unfortunately, something else stopped this poor guy, have a look at his page now: http://reggaescientist.org/

:( :( :(

DSPguru
13th December 2002, 07:46
@jamiro
i appreciate your concern, but you should know that lots of people are also waiting for other features/bug-fixes, which, to my understandment, are more important. (like the fact that some FRCs ends up with clicking stream).
so, i guess, i have to prioritize.

lately, i have been very busy in my personal life, and i hardly been online, not to mention having the time to code.

on a second note,
i recall lots of people trying to convince me to publish BeSweet's source with claiming that the development would be faster.
but as you can see, this is not true. until today, 4 monthes already, no-one had fixed this bug in ac3enc (being open-sourced), and no-one had written a fade-out plugin (which also has open-source samples).

jamiro
13th December 2002, 11:02
@matter @dspguru
I'm not demanding, it would be rude. I just pointed out that this problem is still unsolved and i thought that solving it won't take a long time since dspguru know's where's the bug. I'm also a programmer and i know that sometimes 90% of time takes finding the bug and only 10% or less fixing it. Please dont't take me wrong i'm not demanding, it's your spare time and it's your decision what to do with it.

Rombaldi
16th December 2002, 05:28
tried using BeSweet (and AC3 Machine) to convert the MP2 track to AC3. Conversion no problem. Remux OK. Sync, on the nose.

but it's way too low. noticably lower than it was before.

How can I boost the AC3 up and try to match the original volume.


... and please DON'T say "don't convert it to AC3". I need to for compatibility with another application. Not an option, needs to be AC3.

DJ Bobo
16th December 2002, 11:38
Set "Compression characteristic" on "None"

quantum
18th December 2002, 00:46
Is there any progress or additional information on this issue?

bira
20th January 2003, 22:21
Again,
Is there any progress or additional information on this issue?

DSPguru
21st January 2003, 07:36
i suggest you to send your comments to : ffmpeg-devel@lists.sourceforge.net

Idefixus
1st February 2003, 13:40
Hi,

i tried to reduce the bitrate to 224 kBit, but after this step, the ac3 is extrem quietly. I used the options Auto Find Maximum Gain from ac3machine, but it seems not to work.

Is it possible to normalize the ac3-stream after transcoding without transcode it again?

michaelsez
4th March 2003, 23:49
Here read this.

http://www.geocities.com/transcodeac3/#_Toc522546849

ihgl
8th March 2003, 03:19
1. Have an 23,967 AVI with Ac3 5.1 audio. Want to make a DVD PAL with AC3 Audio - authorizing program SpruceUp/Maestro
2. Rips the audio with Nandub, get a wav file
3. "Clean" the wav up with BeSlice to an AC3.
4. Uses BeSweet to convert from 23,967 to 25 fps, gives an AC3
5. The new Ac3 plays great in Winamp/Mplayer
6. Compile the m2v and Ac3 in SpruceUp/Maestro...

Problem: The amplitude gets far too low, sound hardly audible..

Have tried a lot of options in BeSweet without success. Even if I use BeSweet without framerate conversion the amplitude gets far too low. With the volume control on max you can hardly hear the sound.

My question: What does happen to the sound in BeSweet? Am I doing some fundamental mistake? The original audio file, produced with Nandub/BeSlice sounds great if you mux it with the m2v file in SpruceUp/Maestro. It is just that I have to convert the file into a 25 fps one in BeSweet - and something goes wrong there. Does anyone know what mistake I am doing here???

DJ Bobo
8th March 2003, 12:16
nothing happens, it's ac3 behaviour to lower the volume ¬.¬

You've got to decode your ac3 to 6 separate mono wav files, amplify them equally: look which file has the largest peak, and gain by that peak. For example, the largest peak is -11db (all others have for example -12, -15, and such), make a gain of 11db on all files.
Adjust the length of each wav.
Encode those 6 wavs to AC3 5.1 @ 448kbps.
The volume should be comparable to your original ac3 file.

ihgl
8th March 2003, 13:25
Originally posted by DJ Bobo
nothing happens, it's ac3 behaviour to lower the volume ¬.¬

You've got to decode your ac3 to 6 separate mono wav files, amplify them equally....

Thanks a lot. I will give it a try. Have not been dealing with AC3 audio before and knew I must have missed something.... ;-)

ihgl
8th March 2003, 15:50
Originally posted by DJ Bobo
nothing happens, it's ac3 behaviour to lower the volume ¬.¬



Well, there is one thing I do not understand though. The original AC3 that came out from Nandub/BeSlice (fps 23,967) works perfectly when muxed with the m2v in SpruceUp - no amplitude change at all. It is when I try to do the framerate conversion in BeSweet that the amplitude almost decreases to "zero". So something must happen in BeSweet. And I think I have tested all options ??

barking_mad
9th March 2003, 13:34
I appreciate everyones time is valuable here especially DSPguru as I've taken on board what's he's said about other issues which have priority over this. But has there been any progress on this matter?
I've read the posts regarding other methods, but like everyone else I suppose I'm looking for a 1 stop shop to re-encode an AC3, without the resulting audio volume being considerably lower.
My amp usually opperates at a volume seeting of about 20%. After re-encoding the AC3 it has to be pushed waaay up to 85%.
If anybody knows of a quick workaround, that doesn't involve multiple proggies I'd be extremely greatful.
Thanks for reading.:confused:

Mikel
11th March 2003, 11:33
I did some test doing AC-Trancoding as well.
Indeed the sound level decreases quite a bit.
For me, AC transcoding is useless in this way. A lot of sound quality is lost that way.
I will send e email to the above adress and i hope that others will follow my example so something might be done about it.

The more emails the higher the chance it will be fixed.

I am interested for other reasons that it will be fixed. There have been some tests on a German page comparing the audio quality of the ac3enc.dll with SoundForge SoftEncoder. The comparison show frequency responses of the signals.
However, ac3enc.dll is superior in that way.

So hopefully, someone can fix it soon.

Cheers

Mikel

gizmau
17th March 2003, 01:20
its the fault of ac3enc, not yours. take a look at this thread http://forum.doom9.org/showthread.php?s=&threadid=33535.

Kilyan
18th March 2003, 16:06
Originally posted by aryan
[B@all
I agree with quantum that for now the best result which can be given is with Softencode. If someone wondering where to get this fine software (Sonic Foundry discontinued offering it) here is the link:
[/B]

That's ok. But I do have softencode 5.1 from sonic but no way to reencode an ac3 from dvd. I mean I extract an ac3 from the dvd then load it to softencode then no buttons to reencode no menus to do that, neither can I disassemble to 6 separate mono waves, how do you reencode directly or inderectly with softencode?

Mikel
18th March 2003, 23:51
There is a function in BeSweed to Decode AC3 to 6 mono wav files.
After that you can reencode it with any AC3 encoder software.

Cheers

Mikel

Kilyan
19th March 2003, 12:49
Thanks Mikel!
Some other questions:

Some settings flags are shown by softencode for the original ac3, but
how do I know that besweet does invers this process? Can you post your batch?
I mean there is for example the 90 degree shift for the rear left and rear right(By the way that sounds good but means nothing I think, It would be stupid to delay by 90 for each frequency), what is besweet doing with that after decoding it to 6 mono waves? Does the 90 stay in the waves or this is really just a flag in the ac3 stream, and is neglected?
And please tell me also what parameters to set in softencode or scenarist ac3 encoder, because it is not described.
I mean digital deemphasis lpf/bpf/hpf and other stuff (I know what a lpf is so that's not the question == the question is that if the signal is in digital format(sampled/quanted) it had to be sampled at the input but then an lpf must be applied)

I've heard of some not free decoding tool,..name was...eer..Digigram ac3 encoder/decoder, someone knows them?

Thanks

One more:
When you decode an ac3 stream, the waves are always much below -3dB, sometimes below -20dB. Why do they do that? I accept that they should group normalize the 6 waves (One of the 6 has a 0dB or -3dB value, and the others are levelled relatively to that. And this also reduces dynamics and S/N), but they Dont!!

I think this is a trick to make you buy bigger and bigger amplifiers, it's like windows...more and more ram...cpu...

Mikel
19th March 2003, 22:50
Your question about the settings i cannot really answer. If you look further down in the forum I started another thread with more or less the same questions.

It seems that transcoding AC3 is something that is not done very often.

In fact I have only done it once so far with the movie "Green Mile" which is a 3 hour movie. Since I wanted to keep both German and English audio with all subtitles and extras, I had to find the best solution for a good 4.7GB disk. So I reencoded the audio in 320kbs which gave me a reasonable movie stream of around 2500kbs instead of 2300 kbs.

When reencoding, i set the phase of the surround to 0° to leave everything the same.

I did not do any further comparisons of the reencoded sound.
I would have liked to use Besweet for reencoding the AC3 stream, but there is a amplitude problem with the AC3end.dll.

So i using the AC3 encoder built into Scenarist, which has about the double speed of softencoder.
The parameter seem to be the same.

Cheers

Mikel

jsoto
20th March 2003, 23:46
@DSPGuru
In ffmpeg 0.4.6 (Released on 2002-12-27) changelog it can be read:

version 0.4.6pre1:
- fix quantization bug in AC3 encoder.

Only for information:
Have you compiled ac3enc.dll with this ffmpeg version in any BeSweet release?. I was trying to compile it (obviously with your sources), but I am too newbie in programming, and I am not able to success.
Could be this "quantization bug" the reason of the gain problem?

Thanks.

DSPguru
21st March 2003, 07:00
Originally posted by jsoto
Could be this "quantization bug" the reason of the gain problem?i believe not.

pago cruiser
1st May 2003, 04:05
OK, I give up. Went thru a search and found several similar threads, but nothing that my tired brain could see solves the problem.

Gnot 0.23, DivX 5.02 pro, 2 pass encoding, the usual stuff. The same old LOW LOw Low low volume MP3's in my final avi. So now I try using BeSweet like lots of folks said. Do my DVDTOAVI, do my COMP Test, and THEN I boot up a 2nd copy of Besweet. Load the AC3 file, select AC3 to AC3, set boost to 3 db, then select this new "boosted" AC3 file as my audio source when encoding/converting to MP3. No difference. Try again at 6db. Nothing. 9db. 27 db. 54 db. NADA ZIP NO DIFFERENCE.

What the heck is happening? I've now encoded this #*&@! thing 5 times, and I ain't a happy camper... :angry:

bitz4brainz
28th May 2003, 23:47
Okay. I went through all the same trials and tribulations as the other people on this thread about the way too soft AC3 encodes. Same result. I tried everything. VERY soft.

All I am trying to do is convert a PCM 48KHz soundtrack (captured analog) to 2 channel AC3 so that it will produce a STANDARD DVD (vs MP2) without chewing up a lot of space. Nothing fancy. The track already (purportedly) is Dolby Surround, and I'd like to preserve that. In a Wave editor the track looks fine and uses the full 16-bit dynamic range.

I got hold of a copy of Soft Encode and tried that. A lot better but STILL a lot softer than the WAV. Maybe by half. Why? I turned off every option I could find, especially pre-processing. All I need is an AC3 compression.

Why is this so hard???

pago cruiser
29th May 2003, 05:00
Don't know if this will help with AC3's, but I finally solved my problem using MP3 Gain. However, I had to set default volume to 100db average. The program now boosts all audio by 12 to 20 db! It does report clipping, but it is only apparant at high volumes, which I never listen to anyway. It only adds about 10 minutes post processing time, so it is acceptable to me.

mookyking
29th May 2003, 13:21
When encoding/transcoding with sonic foundry soft encode, I always set dialog normalization to -27db and turn off all input filtering and dynamic compression. The volume is about the same as the original. Dialog normalization can also be set all the way to -31db, which makes the results even louder.

Soft encode can directly transcode an ac3 stream without having to first decode to seperate .wav files; when using the File->Open dialog, select dolby digital(decode to PCM).ac3 as the file type. Takes a while to decode though! After it's done decoding, it's ready to re-ecode at a different bitrate.

FreQi
30th May 2003, 07:23
I just ran a 2 second 2Ch 48kHz WAV file through Soft Encode and the AC3 Encoder that comes with Scenarist and BeSweet to do a little testing. You can download all the files here (http://freqi.net/forums/TIABMA.zip) (985k). Here's what I did...

(a) Soft Encode -- TIABMA_a.ac3
Audio service config
Data rate: 192k
Sample rate: 49 kHz :: (auto checked)
Audio mode: 2/0 (L,R) :: LFE enabe was grey'd out
Bit stream: Main: Complete
Dialo norm: -27 dB
Audio bw: 20.30 kHz
Save frames in Intel byte order: UNchecked
Bit Sream tab
Center mix: grey'd out
Surround mix: grey'd out
Dolby surround mode: Not indicated
Copyright bit: checked
Orig bit stream: checked
Info exists: checked
Mix level: 105 dB SPL
Room type: Small
Preprocessing tab
Digital deemphasis: UNchecked
DC high pass filter: checked
bw low pass filter: checked
LFE low pass: grey'd out
90 degree phase: grey'd out
3 dB attenuation: grey'd out
Compression characteristic: Film: standard
RF overmodulation protection: checked

(b) Soft Encode -- TIABMA_b.ac3
Audio service config
(same as a)
Bit Sream tab
(same as a)
Preprocessing tab
(same as a, except...)
DC high pass filter: UNchecked
bw low pass filter: UNchecked
Compression characteristic: None
RF overmodulation protection: UNchecked

(c) Soft Encode -- TIABMA_c.ac3
(same as b, except...)
Audio service config
Dialo norm: -1 dB

(d) Soft Encode -- TIABMA_d.ac3
(same as b, except...)
Audio service config
Dialo norm: -31 dB

(e) Scenarist AC3Enc -- TIABMA_e.ac3
Audio Coding mode: 2/0: L,R
2-Ch Interleaved Input: checked
LFE Enable: UNchecked
Start time: (all zero's)
End time: (all zero's)
AC3 Settings, Edit
Data rate: 192k
Sample rate: 48
Dialog Normalization: -27dB
Bit Stream Mode: main, complete
Audio BW: 20.30
Center mix: grey'd out
Surround mix: grey'd out
Dolby surround mode: not indicated
Copyright bit: checked
Original bit stream: checked
Audio Production Info exist: checked
Mix level: -105dB
Room type: small, flat
Digital De-emphasis: UNchecked
DC highpass filter: checked
BW Lowpass filter: checked
LFE lowpass filter: checked
Surround channel 90 phase shift: grey'd out
Surround 3 dB Attenuation: UNchecked
Compression Characteristic: Film: Standard
RF Overmodulation Protection: checked

(f) Scenarist AC3Enc -- TIABMA_f.ac3
(same as e, except...)
DC highpass filter: UNchecked
BW Lowpass filter: UNchecked
LFE lowpass filter: UNchecked
Compression Characteristic: None
RF Overmodulation Protection: UNchecked

(g) BeSweetGUI -- TIABMA_g.ac3
In the pulldown menu to the left of the "Azid 1" "Azid 2" "Lame 1" "Lame 2" set of buttons, select AC3.
In the OTA section, check Auto Gain (Post Gain doesn't matter, it will go grey)
Click the "AC3 & OGG" button, check bitrate and select 192.
Click the BeSweet button and your command line will look something like "C:\....\BeSweet.exe" -core( -input "....\TIABMA.wav" -output "....\TIABMA_g.ac3" ) -ota( -g max ) -ac3enc( -b 192 )

(h) BeSweetGUI -- TIABMA_h.ac3
Same as g expect Auto Gain is unchecked.

(i) BeSweetGUI-- TIABMA_i.ac3
Same as g expect Auto Gain is unchecked, and Gain was checked and set to 50db.

Sadly, none of the resulting AC3 files have the same volume as the original WAV. Most of the AC3's actually sound a lot like each other, with the exception of (g). On that one, the logfile from BeSweet reports a Total Gain of 5.462dB was found, but the AC3 was practially mute. Unchecking Auto Gain yeilded similar results to all AC3's by Scenarist or Soft Encode, and forcing a Gain of 50 only hurt it.

So there you have it. Everything I know about encoding AC3. Here's a list of the files, their size in bytes and their checksums...

bytes file_name___ checksum
64512 TIABMA_a.ac3 4CFC89AC
64512 TIABMA_b.ac3 03EDD89F
64512 TIABMA_c.ac3 492A7865
64512 TIABMA_d.ac3 27CFC62F
63744 TIABMA_e.ac3 62DCE7B4
63744 TIABMA_f.ac3 0B8059D6
63744 TIABMA_g.ac3 BB9F6D11
63744 TIABMA_h.ac3 69B76C17
63744 TIABMA_i.ac3 8A3B531F

I find it interesting that BeSweet and Scenarist make files of the same byte count, but none of them generate the same file with what should have been the same settings.

bitz4brainz
30th May 2003, 07:59
Freaking interesting! Thanks for this!

If two supposedly professional encoders always produce lower volume and we eliminate the possibility of imbalance due to the player (since we are using the software's own player), then...

Maybe there is some technical "reason" why a proper AC3 encoder deliberately reduces the volume. Maybe it has something to do with a broadcast standard, or the way audio gear in a theatre operates. Or maybe its a silly or unthinking reason that got codified into the standard.

Like this one: Most audio CDs produced in the first decade or two of the technology do not use the full 16 bit headroom. Sometimes they use only 50% or even a third. Why? I am not a recording engineer, but I surmise that they either didn't understand the new technology (vs analog) or they didn't realize it mattered. CDs were evidently produced with the volume that happened to result from whatever setting they happen to like to use on their preamps. (?) It DOES matter because you have to boost the volume on playback, thus amplifying the noise and deficiencies of your analog playback system. It also matters, at least theoretically, because 16 bits is more accurate than 15 or 14 bits.

But audio CDs engineered in the past decade or so tend to use 90-95% or even 98% of the headroom. Even remasters of the same music on earlier CDs. Somebody woke up.

But if this is really part of the standard, then there will be no waking up. We will have to live with it.

Just conjecture and speculation, of course. I could be way off base.

I'm not doing 5.1 stuff so I'm afraid that there isn't much incentive to use AC3 on simple stereo. I think most DVD players of recent vintage, even in the US, will play back MPEG Layer II. For me, that's a known quantity.

Julio
4th June 2003, 10:12
Hi,

I'm trying to encode a WAV (16-bit stereo, 48 kHz) to a stereo AC3 file, using BeSweet 1.4.

Here's my original wave signal :

http://www.ifrance.com/InTheNuts/dvd/wav.jpg

I tried this very simple command :

besweet -core( -input country.wav -output country.ac3 ) -ac3enc( -b 384 )

... but I get this :

http://www.ifrance.com/InTheNuts/dvd/ac3.jpg

As you can see, the output level has been divided by 2 ... why ??

Is there any other way to encode a WAV to an AC3 without loss of information ?

Thanks...

Julio
4th June 2003, 16:29
Alright, my post just got moved to this thread.
I read it and it's very interesting... so it seems that there's no way to transcode WAV to AC3 without loss of level.
What a mess.
That means loss of audio information ! I'm just disgusted. Bye.

FreQi
5th June 2003, 08:13
I'm not excactly sure that there is a loss of audio information so much as the new AC3 is being "marked" to play at a lower volume. Meaning, rather than altering the waveform, maybe it's just encoded on a different dB scale. I dunno... Just a guess/hope.

Julio
6th June 2003, 09:41
Originally posted by FreQi
I'm not excactly sure that there is a loss of audio information so much as the new AC3 is being "marked" to play at a lower volume. Meaning, rather than altering the waveform, maybe it's just encoded on a different dB scale. I dunno... Just a guess/hope.

Okay, but then why do we see the diminished waveform when decompressing the AC3 into a wave editor like Goldwave, and not the "original, not-lowered" waveform ?

aklendathu
9th June 2003, 09:56
I have extracted the audio from a captured AVI to a separate WAV file and encoded it to 2.0 AC3 with BeSweet (384 kbps). However, the sound level becomes extremely low. I tried the 0db Gain as well as the Auto Gain flags and none worked well. What parameters/flags should I use to get a decent sound level on the AC3 file (at least as loud as the original WAV) ?

Thanks in advance,

Pedro

orestis
19th June 2003, 14:31
I _think_ there might be an issue with AC3Filter.

If you use a DVD-Player (e.g. WinDVD) it uses its own AC# decoder. When trying playing a transcoded AC3 it sounds just the same as the original. I haven't made any serious tests because it takes me way much time to transcode so I just put the original AC3 in and do a 2-cd rip.

Could anyone give this a look ?

raam123
12th July 2003, 09:18
I've been transcoding Mp3 to wav using BesweetGUI.

I choose the Dolby option at the right window , which allows AC3-WAV.

When finished I burn it , and play it on my DVD , though the volume is low , I have to rise it to Max to hear it well...

Any idea ?

sphinx
6th August 2003, 15:23
Hi,

I'm ripping a DVD using Robot4Rip, which uses DVD2AVI to encode the sound to AC3. Often the volume on the each segment of the dvd remain constant, but sometimes you get the odd episode where the volume is alot softer then those of the others. I've listened to the DVD and the volume is normal, but on the ac3 its alot softer.

I've tried ripping it with SmartRipper and still had the same soft volume of the ac3. I've also tried increasing the volume to that of the others using AC3Machine, but after doing that the sound is vastly distorted.

Does anyone have ideas that could help me solve this problem?

Cheers.
Sphinx

p.s. my post has seemed to have moved to this thread...
/me reads

Kermee
15th August 2003, 05:23
Hi All,

I'm trying to convert AC3 off a PAL source to NTSC AC3.

I'm using the BeSweetGUI to do this and using the preset "PAL -> NTSC (25.000 to 23.976)" to do so.

The resulting file is exactly correct in length. However the volume is at least -6dB of what the original AC3 is. How do I fix this?

Here is the command line it does:

BeSweet.exe -core( -input "C:\PAL.ac3" -output "NTSC.ac3" -logfile "C:\BeSweet.log" ) -azid( -z1 -b1 -s stereo ) -ota( -r 25000 23976 ) -ac3enc( -b 384 )

Any ideas why the resulting AC3 has much lower volume than the original? Thanks.

vmesquita
6th October 2003, 23:58
@DSPGuru

Looks like the bug has been fixed in the latest FFMpeg 0.4.8... Could you check?

http://www.dvdrhelp.com/forum/viewtopic.php?p=669661#669661
http://www.dvdrhelp.com/forum/viewtopic.php?p=672401#672401

[]'s
VMesquita

DSPguru
7th October 2003, 00:08
Originally posted by vmesquita
@DSPGuru

Looks like the bug has been fixed in the latest FFMpeg 0.4.8... Could you check?

http://www.dvdrhelp.com/forum/viewtopic.php?p=669661#669661
http://www.dvdrhelp.com/forum/viewtopic.php?p=672401#672401

[]'s
VMesquita in one word : bullshit.
in more details : this bugfix is actually something i wrote 14monthes ago to fix a flaw on 2ch ac3 encoding.
the volume bug is still there.. :(

vmesquita
7th October 2003, 01:40
@DSPGuru

Sorry, I didn't make myself clear. I was refering to the bug that prevents AC3 files produced in BeSweet to be played in Pioneer stand-alones. Has it been fixed? :confused:
I never experienced this volume bug... :)

[]'s
VMesquita

DSPguru
7th October 2003, 06:18
Originally posted by vmesquita
@DSPGuru

Sorry, I didn't make myself clear. I was refering to the bug that prevents AC3 files produced in BeSweet to be played in Pioneer stand-alones. Has it been fixed? :confused:
I never experienced this volume bug... :)

[]'s
VMesquita again, this "new" feature is there in BeSweet for almost 15monthes.

ps,
i seriously conisder removing any support for ac3enc.dll in future BeSweet versions as it seems that users ignores my warning about the amature state of this codec and use it to encode important stuff.

vmesquita
7th October 2003, 14:50
Well, I never saw your warnings... :( :( But use ac3enc mostly for tests, because of this pioneer bug. I don't have a pioneer player, so I never knew if it was fixed or not.
Unfortunatelly, there's no other free ac3 encoding utility... And SoftEncode (my favorite) has no command-line support. Do you have any software recommendation for AC3 encoding?

[]'s
VMesquita

DSPguru
8th October 2003, 00:39
Originally posted by vmesquita
Well, I never saw your warnings... :( :( http://forum.doom9.org/showthread.php?s=&threadid=15738

Unfortunatelly, there's no other free ac3 encoding utility... And SoftEncode (my favorite) has no command-line support. Do you have any software recommendation for AC3 encoding?http://forum.doom9.org/showthread.php?s=&threadid=60137&perpage=31&pagenumber=2


Cheers,
Dg.

vmesquita
8th October 2003, 03:29
This is great! I never knew softencode had command-line support! I just can't figure out how to do 2.0 throught commandline.

<edit>I just figured out, I posted here: http://forum.doom9.org/showthread.php?s=&postid=383110#post383110
</edit>

[]'s
VMesquita

J-Wo
25th October 2003, 17:37
Here is my solution to the whole low volume on AC3 encodes. After only a couple attempts, I stumbled upon the setting that PostGain to 100% solves everything. In the command line, this is -ota( -G 1 ). This setting can be found in the latest BeSweetGUI, but NOT in AC3machine. Someone previously posted that setting Autofind Maximal Gain worked for him, which is the same as PreGain Normalize to 100% in BeSweetGUI, or -ota( -g max ) in the commandline. However I found that this did NOT solve the problem. Neither did trying Hybridgain.

In my case, I was transcoding a 5.1 channel AC3 from 448 kbps to 224 kbps. The two settings I never know what to do with is Dynamic Compression and LFE to LR Channels. Can anyone tell me what I should set these two to preserve the original audio quality as much as possible? Thanks!

spacemaggot
14th November 2003, 20:59
I tried encoding a stereo .WAV to a stereo .AC3 with and without the above command-line switch, and got two identical files (same MD5 sums). No joy here.

Anyone want to sell a used copy of Soft Encoder?

-Spacemaggot.

Amnon82
7th May 2004, 11:36
Here a AC3 out of a MP3 Stereo. It keeped the loudness...

http://rj-elektronik.de/avideo/samples/Beyonce-Dangerously_in_love_by_Amnon82.ac3

Created with this

http://forum.doom9.org/showthread.php?s=&threadid=75833

adboehm
2nd July 2004, 10:23
I don't suppose there has yet been any fix to ac3enc.dll or any quick fix? (5.1-5.1 with lower bitrate)

Dirk99
5th October 2004, 17:17
Is there any other possibility to lower the bitrate in ac3-files? I´ve tested AC3Machine to downcode from 448 to 384 and you hear the different loudness very good.

The method from Amnon82 is only to convert single waves to one 5.1 ac3, isn´t it?

keithmac
5th October 2004, 18:22
You need to use a good ac3 encoder like Softencode, the ac3enc.dll is the weak link in the chain, even just a framerate change with no bitrate change dropped the volume on my ac3 by at least 50%

Dirk99
5th October 2004, 22:40
And with Softencode it is no problem? That would be great! Do you have a link where I can get the program?

daphy
6th October 2004, 06:33
your a funny fellow :D

I guess the World Wide Web would be a hot adress :rolleyes:

Dirk99
6th October 2004, 08:15
Your´re right. Even a silly question. Sorry.


I use BeSplit to join or to split AC3-files. Does this program a direct stream copy or makes it use of the bad ac3enc.dll?

Thx,
Dirk

keithmac
6th October 2004, 19:44
I use BeSweet to timestretch and add delay (if needed) and output to 6 wave files, feed these into SoftEncode and re-render the ac3 stream (this point you can set the bitrate etc)

"C:\BeSweet.exe" -core( -input "e:joinfix01.ac3" -output "e:\joinfix01.wav" -6ch -logfilea "C:\BeSweet.log" ) -ota( -r 23976 25000 -d 32 ) -profile( ~~~~~ Default Profile ~~~~~ )

Sample line, changing lenght to suit 25 fps from 24 fps ad delay and output to 6 wavs, don`t use -azid at all as i don`t need to and it actual lowers the volume slightly as well?

keithmac
6th October 2004, 19:45
the ac3.dll is only used to make the ac3 not decode it as far as I can see.

Dirk99
6th October 2004, 21:09
What do you use in the other configuration fields? In Sof Encode you can enter a "Dialog normalisation" a "Center mix level" and many more options. But I only want to change the bitrate and nothing else.

keithmac
7th October 2004, 18:07
You need to find the avaerage RMS level for the dialogue normalisation value, I use SoundForge in the Normalisation section it has an option to find the rms value, use the centre channel wav as the input.

As for the other options I`m looking into it at the moment, trying not to alter anything else really, no 90 degree shif or attenuation on the surround channels etc, if you read the help file while you`re setting the encoding options you`ll have a good idea what to change and what not to.

The dialogue normalisation is just a flag for the decode to sort the volume levels out.

ocyl
7th May 2006, 08:15
Give ffmpegGUI Beta Build 3c (with ffmpeg CVS 2006-04-28) a try, folks :cool:

http://www.free-codecs.com/download/ffmpegGUI.htm

tebasuna51
7th May 2006, 19:50
Good news!
At last a free ac3 encoder with output volume equal to input volume!

Thanks.

mutha trucka
10th May 2006, 10:28
Give ffmpegGUI Beta Build 3c (with ffmpeg CVS 2006-04-28) a try, folks :cool:

http://www.free-codecs.com/download/ffmpegGUI.htm
I tried it and compared the results with the besweet result and it has the same low volume problem, unfortunately.

tebasuna51
10th May 2006, 12:20
I tried it and compared the results with the besweet result and it has the same low volume problem, unfortunately.
Not at all, with BeSweet-ac3enc the output volume is limited to 50% of input volume.
With new ffmpeg the output volume is 100% the input volume.

Try this test:
1) Make a stereo wav sample normalized 100%
2) Encode to ac3 with BeSweet-ac3enc
3) Encode to ac3 with ffmpeg
The ffmpeg encoded is 6 dB higher (100%/50%) than BeSweet encoded.

To check multichannel wav's you need other GUI than ffmpegGUI to use ffmpeg.exe (maybe BeHappy).

mutha trucka
11th May 2006, 12:24
My bad. I didn't have the latest version. Thanks, it works great!

vlada
25th June 2006, 14:06
Since the low volume bug in FFMPEG is fixed now, can I somewhere download and updated ac3enc.dll for BeSweet?

shon3i
25th June 2006, 23:01
Since the low volume bug in FFMPEG is fixed now, can I somewhere download and updated ac3enc.dll for BeSweet?I think no, but why not just use BeHappy.

danpos
26th June 2006, 00:17
Since the low volume bug in FFMPEG is fixed now, can I somewhere download and updated ac3enc.dll for BeSweet?

If you've good skills on C++ / audio encode, so go ahead (some people tried do it but unfortunately didn't get success ... :( )

Good luck,

vlada
10th July 2006, 22:12
Unfortunately I'm not a good C/C++ programmer. I think I'll stay with BeHappy, which works fine for me.

But BeLight is a very nice and clean GUI for transcoding, so it would be nice if it worked too without any problems.

Kurtnoise
11th July 2006, 14:44
There is an alternative for BeSweet to resolve this issue : using Aften with the last bsn.dll...I didn't tried though. Maybe tonight.

vlada
20th September 2006, 19:00
Kurtnoise13> How is it looking with BeLight and Aften? Have you found any working solution? Thank you.

Btw. an updated ac3enc.dll with low volume problem fixed doesen't exist yet?

Kurtnoise
25th September 2006, 16:30
Hi,

Here is (http://kurtnoise.free.fr/index.php?dir=BeLight/&file=bsn_Aften_20060926.zip) an update for bsn.dll to use it with Aften (http://forum.doom9.org/showthread.php?t=113074). I've made some tests only with CBR/VBR Modes. So, please test it with all switches available.

Command Line example :
For Stereo Output @384kbps :
BeSweet -core( -input "C:\my_file.wav" -output "E:\my_file.ac3" -logfile "E:\logfile.txt" ) -bsn( -exe aften.exe -b 384 )
For Stereo Output with q=300 (VBR Mode) :
BeSweet -core( -input "C:\my_file.wav" -output "E:\my_file.ac3" -logfile "E:\logfile.txt" ) -bsn( -exe aften.exe -q 300 )
For Multichannel Output @448kbps :
BeSweet -core( -input "C:\my_file.wav" -output "E:\my_file.ac3" -logfile "E:\logfile.txt" ) -bsn( -exe aften.exe -b 448 -6chnew )

BeLight (http://corecodec.org/frs/?group_id=45&release_id=303#r303) has been also updated to support Aften.

LigH
16th October 2006, 14:43
I am afraid I would instead really prefer a " native" aften.dll BeSweet plugin, instead of feeding the EXE - if possible; but if you promise that there are no drawbacks, I have to believe you... ;)

Kurtnoise
16th October 2006, 15:17
Problem is : BeSweet is closed sources. So, we need to find an other way to use extra goodies...Unless to tweak ac3enc lib but BeSweet won't be able to use new things anyway.

gonwk
17th October 2006, 06:02
Hi folks,

Can someone load up the "BeHappy" GUI somewhere on Doom9 so we can download it.

I went to the Site that Hosts it and it is tough trying to navigate thru it ... I could not figure out how to download that GUI.

Thanks,
G!

LigH
17th October 2006, 08:50
@ Kurtnoise13:

Unfortunately, you seem to be right. It is possible to use plugins in BeSweet, but you seem to be only able to use parameter-less functions, which is a bad limitation.

besweet.exe -plugin( -name DLL_NAME -func EFFECT_NAME [-6ch] )

Adding parameters would be necessary for such a complex function like an AC3 encoder. Maybe something like:

besweet.exe -plugin( -name DLL_NAME -func EFFECT_NAME:par1=val1;par2=val2 [-6ch] )

Kurtnoise
17th October 2006, 09:03
Yes but the problem remains. How BeSweet can parse this "par1=val1;par2=val2" w/o tuning the code ?

Perhaps, there is an other way to do that :
besweet.exe -core(...) -dimzon( -dllname DLL_NAME ......)
You need first to build libaften then create a new lib to call aften functions...

tebasuna51
17th October 2006, 10:49
Can someone load up the "BeHappy" GUI somewhere on Doom9 so we can download it.

I went to the Site that Hosts it and it is tough trying to navigate thru it ... I could not figure out how to download that GUI.
Search in BeHappy thread.

You can use the last shon3i package (http://forum.doom9.org/showthread.php?p=829754#post829754)

gonwk
18th October 2006, 04:17
Hi TebaSuna,

I OWE YOU one ... THANKS A BUNCH!

BTW, Teba ... does BeHappy have a "BETTER" Quality than BeLight (BeSweet GUI) program.
What I have been doing ... I have bought few foreign DVDs that I think they were either copied from Satellite or they were dubbed from VHS tapes ... since they have a Hummm and Hisss ... so, I have been rebuilding them as follows but I DO NOT like my results since it seems I lose Audio Quality ... the Original's Audio tell me they are AC3 2Channel 256 ... here what I do ...

1) Use DVDDecrypter to rip in File mode (vob's).
2) Use DGIndex to demux (get Video=.m2v & Audio=.AC3)
3) Use Belight 0.22 to convert AC3 file to Wav file.
4) Use Audacity 1.3 to clean up my Wav file ... under "Effect" I use "Noise Removal" and take out noise. But I believe by doing so I also might be sacrificing my Highs and Lows!!!!
5) Export my edited Wav file. Sometimes I get daring and try the "Leveller" and even "Amplify" ... but either effect I don't care for since Results sound echoey and not so hot to my ears.
6) Use BeLight again to convert the new Wave File to AC3 ... without any Boosting or Gain. But if I have to do it because my end result is way too LOW ... then for Boosting "Ligh" works the best for me; and for Gain "HybridGain" works best for me.
7) Use Muxman to combine my earlier Video file (.m2v) with my new clean AC3. Program works great.
8) Use DVDShrink to make my ISO.
9) Use ImgBurn to burn my ISO to a DVD.

The reason I went into this detail is because I am hoping someone else might find this guide helpful ... since I had to practically read pages and pages and search so much to come with a solution that used nothing but "Freeware" to do what I wanted to do.

THANKS Again for your Help!

G!:thanks:

gonwk
18th October 2006, 04:27
Hi Kurt & Ligh,

I was so "Excited" when I saw Teba's post with URL to BeHappy downlaod site that I did not even bother to read the other posts ... Sorrry & My Apology ... since you folks took the time to respond to my Q ...

Kurt & Ligh, I am not really that Familiar with Line Codes ... matter of a fact it would be safe to say I am Big Dummy in that department ... if it does not have a GUI that walks me thru .... or a hell of a "Guide" .... then I am dead in the water.

So, if you can point me to a Guide for learning how to experiment with Command Lines Without Crashing my computer then I am game!!!

Thanks,
G!:thanks:

tebasuna51
18th October 2006, 11:00
BTW, Teba ... does BeHappy have a "BETTER" Quality than BeLight (BeSweet GUI) program.
Not at all. BeHappy is more flexible than BeLight-BeSweet because use AviSynth scripts, accept any DirectShow input, can be developed with plugins open-source and accept any encoder with stdin support.
BeSweet is closed sources and Kurtnoise can't improve it. He make a good work with the BeLight GUI and supporting the new encoders like Aften for ac3.
For your problem ac3 2.0 -> ac3 2.0 you obtain the same quality (maybe the ac3-decoder BeSweet-azid is more flexible than NicAudio-AviSynth).

Now I make some comments to audio steps in your solution:
3) Use Belight 0.22 to convert AC3 file to Wav file.
Here you have a important decision related with audio volume, apply or not the Dynamic Range Compression (DRC). The free encoders can't create DRC, by the moment, then if your audio equipment don't support a wide dynamic range (hear the whispers and the explosions not distorted) is recommended apply here the DRC 'normal' (or 'heavy' before than use Boost)
4) Use Audacity 1.3 to clean up my Wav file ... under "Effect" I use "Noise Removal" and take out noise. But I believe by doing so I also might be sacrificing my Highs and Lows!!!!
Yes, is difficult erase only the noise, but is a user choice. Remember only one thing: if the original ac3 2.0 is Dolby ProLogic encoded this kind of edition can destroy the dpl info.
5) Export my edited Wav file. Sometimes I get daring and try the "Leveller" and even "Amplify" ... but either effect I don't care for since Results sound echoey and not so hot to my ears.
If you experiment volume problems this is the moment to Amplify the maximum without clipping
6) Use BeLight again to convert the new Wave File to AC3 ... without any Boosting or Gain. But if I have to do it because my end result is way too LOW ... then for Boosting "Ligh" works the best for me; and for Gain "HybridGain" works best for me.
Here, if you have the last version of BeLight (http://forum.doom9.org/showthread.php?p=881619#post881619), you must use Aften encoder (double volume than ac3enc).
With the wav normalized/maximized in step 5 any 'PreGain' do nothing and 'PostGain' or 'HybridGain' can't be applied over an ac3 (and unnecessary with Aften with input normalized).

Boost is the last solution. Make the same than DRC but distorting the audio with undesired harmonics. Never use Boost, return to step 5 and edit the wav until you are satisfied.
If you can't hear the wav exactly the same volume than the ac3 encoded by Aften maybe you have a problem with your ac3 player

gonwk
21st October 2006, 00:07
Hi Tebasuna51,

THANKS for your detailed message and all the Great Tips that you have given me ... I saved your above Tips for my reference.

THANKS Again,
G!