03-08-2006, 08:05 PM
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Iv always had this problem when converting a uncoppressed wav file (off a movie) to 2 channel ac3 for dvd. Thats why iv always left it and encoded to mp2 instead, but i though id take time out and investigate why this is.
whenever iv tried to encode to ac3, using besweet, or tmpg3, or recently the audio transcoder in Sonic scenraist... the sound has been way to quiet, too low for normal use, i transfer to dvd and i can almost hear it ino my tv with the sound maxed out, this has been the problem on all 3 encoders. if say i use bitrate of 320 kbps ac3 sound is dead quiet same setting fopr mp2 and the levels hit the roof .
Am i doing something wrong, lets take besweet. How can i increase the volume on ac3 encodes so it is at a normal level? Iv tried using pregain option in besweet it didnt help much? also tried boost, the sound ended up sounding like its off the radio
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Someday, 12:01 PM
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03-09-2006, 03:10 AM
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If the three encoders have the problem, then it is not in what you do, but in what youuse to read it. Don't you think ?
WIth my experience, there is no problem with besweet and 2 channels AC3 (the sound is just the same as in mp2). The know problems are for 5.1 AC3.
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03-09-2006, 09:01 AM
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thats what i thought but i cannot work out what i am doing wrong. say when i encode using besweet, i test it playing it back on media player classic, it works but is too quiet. I have found that ac3 encoded video files on my pc are normally quiet so i turn rthe volume up, but the ones i encode myself are too quiet. i then dump the ac3 encoded sound along with video in tmpg or dvd lab pro and author the dvd. Burn it and test on my dvd player/s and the sound is still too quiet. What do you think is worng, could it be the ac3 encoder engine
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03-09-2006, 09:14 AM
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ac3enc.dll used by besweet has a problem with 5.1 encoding.
Other tools don't use the same engine that's why I assume that your problem is elsewhere.
It's possible that you do the same mistake on your PC and on your SAP  .
Do you have a 5.1 system ? I assume that normal DVD played on your SAP have a correct sound ? And if you play them on your PC ?
What you can do to be sure is to take an ac3 that you encoded and simply convert it into wav with besweet, using absolutly no filters at all. Play this wav : if the sound is too low, then the endoced ac3 had a problem. If not, then this is the codec you used to PLAY the ac3 that has a problem.
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03-19-2006, 05:19 PM
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thanks dialhot, i burned some tests. the ac3 sound i encoded with tmp3, besweet, came out too quiet. even at max volume the sound was too low. but the sound i encoded with as3 encoder for sonic senarist came out perfectly, i cant for the life of me work out what the problem is with the other two ( i tried everything, from changing codecs, to encoders, re installs etc) but i will stick with sonic.
Thanks again
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03-21-2006, 12:55 AM
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Some people have had some success with ac3enc-produced 5.1 AC3s by using a +6dB gain when encoding to AC3. But if you have problems with 2-channel tracks, your problem probably lies elsewhere. Can you upload a short sample clip of the source and resulting AC3 file somewhere? You can use DelayCut to cut the files.
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03-21-2006, 05:12 AM
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Quote:
Originally Posted by Boulder
Some people have had some success with ac3enc-produced 5.1 AC3s by using a +6dB gain when encoding to AC3. But if you have problems with 2-channel tracks, your problem probably lies elsewhere. Can you upload a short sample clip of the source and resulting AC3 file somewhere? You can use DelayCut to cut the files.
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Yupp, I did that once too and had a look a the resulting AC3 using Audition.
And there ... at the peaking areas cippings where the result.
If you do normalize the source PCM stream to 100% (what besweets Normalize 100% does) and apply a +6db later a clipping at the final ac3 is inevitable.
Maybe some users just did apply a +6db ONLY to their source ac3 before reencoding to ac3. Then for shure it gets almost the same volume like the original DVD but .... you dont know if clippings do occur AND .... at 50% of the DVDs I know/have the original AC3 already is too silent. So such originals are not to be seen as loudness reference.
Reference is a 16bit 48.000Hz Pink-Noise PCM Wav file at maximum gain if testing the ac3enc.c core in FFmpeg/Mencoder/Ac3Enc.dll.
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03-21-2006, 07:50 AM
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i will try to upload some of the sourse and destination clips later on.
im actually going from pcm (wav) to ac3 ( 2 channel), my sources are wav.
i will try to normalise the pcm, but the 6db gain ? is that in the overall output configuration or should i do that through pregain on besweet?
Should i do that in two stages, first normalise the audio and add the gain, encode to a new wav pcm,
then stage two encode to ac3 with no adjustments
that does sound promsing what you wrote, but unfortunatly some of those options are not available in tmpg 3, so we will give that a miss now altogether. i will continue trying on besweet.
sonic ac3 encoder was successful when i had burned to dvd
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03-21-2006, 07:57 AM
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Quote:
Originally Posted by Boulder
Some people have had some success with ac3enc-produced 5.1 AC3s by using a +6dB gain when encoding to AC3. But if you have problems with 2-channel tracks, your problem probably lies elsewhere. Can you upload a short sample clip of the source and resulting AC3 file somewhere? You can use DelayCut to cut the files.
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so u mean successes going from 5 channel ac3 to 5 channel ac3, or 2 channel to 5 channel? Now that would be amazing if it could be dont without a lot of work
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03-21-2006, 08:03 AM
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Quote:
Originally Posted by khusru
so u mean successes going from 5 channel ac3 to 5 channel ac3, or 2 channel to 5 channel? Now that would be amazing if it could be dont without a lot of work
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Read also what Incredible told : these "success" reported by Boulder aren't really convenient as distort is more than probable if you use a such +6db gain.
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03-21-2006, 09:22 AM
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@khusru
NEVER ... do apply a manual db increase AFTER a normalization has been done!
When normalizing a source its already at its maximum gain (if the normalizing is setted to max gain) THEN its up on the encoder how he treats the volume, ... furthermore how it'll get affected in its volume.
The goal is NOT only that your resulting AC3 "sounds" loud, but it should sound loud enough and beside this it should NOT be distorted.
If your SAP accepts the results of ac3enc (ffmpeg/mencoder/besweet) and if its loud enough then just go for it. In my cases I just have to increase the knob of my (already loud enough) Amplifier to compensate these missing db's. As said, ... I've to do that on 50% of my original DVDs when playing back them.
Alternative is using Softencode (no longer available -> get it at ebay) ... or your way paying thousands of bucks by purchasing the whole Scenarist package
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03-30-2006, 10:13 AM
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Quote:
NEVER ... do apply a manual db increase AFTER a normalization has been done!
When normalizing a source its already at its maximum gain (if the normalizing is setted to max gain) THEN its up on the encoder how he treats the volume, ... furthermore how it'll get affected in its volume.
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im sorry mate, im not too sure how u mean i should do this? okay here is the scenario, i have a 10 second wav pcm clip that i want to convert to ac3. i have normalization selected at 100%, and i have added a 6db pregain, is that fine? or should i leave the pregain out because the track is going to be normalized hence its at its maximum gain without distort ?
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03-30-2006, 11:09 AM
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Quote:
Originally Posted by khusru
or should i leave the pregain out because the track is going to be normalized hence its at its maximum gain without distort ?
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Yes you should. But you can use postgain as your problem is after the normalized wav has been encoded to ac3. But there is few chance that you don't have any distort in the end if you use a "+6db" value.
I personnaly always use a "postgain to 100%" that is a post encoding normalize.
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03-30-2006, 04:20 PM
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I just ran a test in besweet, i encoded a pcm wav clip , ( already normalised), into ac3 using postgain, at 384kbps. The sound is quiet not as quiet as before, but when turning the sound up there is a lot of hiss in the background.
i did another test on the same clip, but used pregain 6db, this clip was the louder of the two,
both clips sound a lot better on my pc, than my old tests did using besweet. I have run out of dvdrs at the mo so i cant check it on my dvd player and tv.
but will get back to u guys when i do
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03-30-2006, 04:42 PM
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Quote:
Originally Posted by khusru
i did another test on the same clip, but used pregain 6db, this clip was the louder of the two,
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How did you normalize the wav ?
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04-04-2006, 08:51 PM
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i just checked azid 1/ normailze 100%
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04-05-2006, 04:29 AM
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Quote:
Originally Posted by khusru
i just checked azid 1/ normailze 100%
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I never used that, I prefer to use the besweet OTA options (pregain or postgain to 100%). Try with this.
Or use normalize.exe on your supposed normalized pcm to see if it was really normalized because what your describe sound weird to me.
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04-15-2006, 09:29 AM
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To me it seems pretty much the same output level  .
How can one create an ac3enc.dll for besweet?
Any instructions out there?
Thanks for sharing these great news with us Andrej.
Oh, and BTW, what about AC3 to AC3 transcoding?
It was also affected by the same bug.
Has it been solved either?
Cheers
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Rui
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04-16-2006, 07:11 AM
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Yesterday I tried to compile a new vers. of ac3enc.dll.
DspGuru released the wrappercode for accessing the ac3end.c source.
But that wrappercode seems to rely on an old ffmpeg release structure where some headers now are obsolete, so no success till now
Anyhow, that should be DspGurus Job as he never thought about releasing the sources of the besweet project itself
Now ac3->ac3: Thats also possible now via ffmpeg.exe.
BTW. you should have a look at Belight from Dimzon @ Doom9.
That is to me now the way of handling/transforming/encoding audiostreams.
It uses avisynth to frameserve the audiodata via NicAudio.dll or via Directshowsource.dll. As Avisynth is able to serve multiple audiochanells and as avisynths internal audio processors (channelmapping, timestretching, resampling(SSRC), ... etc etc etc) are quite good - well at least as good as besweets - you should test it.
It internally uses Faac, ffmpeg, Nero etc etc for audioencoding.
The new version of Purebasic now allows serving data directly to an exe via stdout/stdin. Means it could be possible to build a "wrapper" exe to access ffmpeg.exe (all in one file). That wrapper could also include a 2-pass maximizer etc. ---- or directliy accessing avcodec.dll --- Thinking about it, when times permits it
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