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Old 20th August 2004, 18:26   #41  |  Link
skobipe
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@ursamtl

Thank you!
The guides were very helpful.
If I may, I have another minor question:
I use soft encode for encoding and in the "preprocessing" tab
I can choose "90 degree phase shift" and/or "3DB attenuation" from the "surround channel processing" menu.
I wanted to know which of the two, if any, will produce better results.

Thank You
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Old 20th August 2004, 21:37   #42  |  Link
ursamtl
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Quote:
Originally posted by skobipe
@ursamtl

Thank you!
The guides were very helpful.
If I may, I have another minor question:
I use soft encode for encoding and in the "preprocessing" tab
I can choose "90 degree phase shift" and/or "3DB attenuation" from the "surround channel processing" menu.
I wanted to know which of the two, if any, will produce better results.

Thank You
You're very welcome skobipe. The more we share our collective knowledge, the more progress we all make.

The 90° phase shift is not essential for decoding through a Dolby Digital 5.1 system, but it is if your mixed might be played back on a Dolby Pro Logic or surround system. These expect the surrounds to be phase shifted by 90°

The 3dB attenuation is usually necessary for files that will end up being played back on consumer home theater equipment.

If you want details just Google them and you'll find there's tons of stuff around on the net.

Hope this helps. Happy mixing!
Steve.
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Old 23rd August 2004, 13:05   #43  |  Link
ursamtl
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One other good reference for AC3 of course is the www.dolby.com site in their Information menu.

Steve.
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Old 1st September 2004, 21:14   #44  |  Link
ursamtl
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In another thread, I read that SurCode Dolby Digital v2 and the AC3 Encoder for Acid use newer Dolby libraries than SoftEncode. Has anybody been able to compare the results from encodes done with both libraries?

Steve.

Last edited by ursamtl; 29th October 2004 at 22:31.
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Old 11th October 2004, 08:27   #45  |  Link
keithmac
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"90 degree phase shift" and/or "3DB attenuation" from the "surround channel processing" menu.

Are these 2 options actually altering the sound itself or just metadata tags that are read by the decoder?

The 90 degree phase shift option, if it actually alters the sound data itself, and you apply it to waves derived from a dvd ac3 source (that should already have been phase shifted) surely it will be shifted by 180 degrees on the 2nd ac3?.

Same with the 3db attenuation, is it just a metadata value or is the sound itself altered while encoding?

Would be interesting to have a list of what options are just metadata for the decoder to read and what options actually modify the sound during encoding.
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Old 14th October 2004, 18:36   #46  |  Link
ursamtl
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Quote:
Originally posted by keithmac
"90 degree phase shift" and/or "3DB attenuation" from the "surround channel processing" menu.

Are these 2 options actually altering the sound itself or just metadata tags that are read by the decoder?

The 90 degree phase shift option, if it actually alters the sound data itself, and you apply it to waves derived from a dvd ac3 source (that should already have been phase shifted) surely it will be shifted by 180 degrees on the 2nd ac3?.

Same with the 3db attenuation, is it just a metadata value or is the sound itself altered while encoding?

Would be interesting to have a list of what options are just metadata for the decoder to read and what options actually modify the sound during encoding.
Hi keith,

There's a good explanation of this at YOU ARE SURROUNDED.

Regards,
Steve.
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Old 15th October 2004, 20:55   #47  |  Link
keithmac
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Thanks for the link, some good reading there!
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Old 27th December 2004, 20:39   #48  |  Link
Capturebat
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In The AC3 machine it has dialog normalization reduction, but you can't input a value. Does it automatically scan and adjust for you?
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Old 13th January 2005, 15:55   #49  |  Link
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It seems i'm in the right thread for this (I hope!)

I have a video file which has AAC 5.1 audio. I can use FAAD to convert that to either a 5.1 WAV file or a stereo WAV file. I would like to take the audio from AAC 5.1 to AC3 5.1 in order to put this video onto a dvd and still have surround. I use TMPGEnc DVD Author for mastering DVD's which I assume will remux my video with AC3 5.1, but for converting video files I use TMPGEnc Plus which downmixes everything to stereo.

What would be the SIMPLEST way (eg the LEAST amount of different software packages!) for me to convert a 5.1 WAV file into a 5.1 AC3 file?

I attempted this with BeSweet 1.4 and AC3 Machine (and BeSweet GUI 0.6) but it stated the AC3Enc.dll was missing (which I have since found) only to read that this DLL is next to useless for creating proper AC3 audio streams (or at least that's what people appear to be saying).

I'm new at this and the terminology is still way over my head, HELP!
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Old 25th January 2005, 22:04   #50  |  Link
00diabolic
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WHY? No AC3 below 224kps

Hi

I have been encoding my moviez with mp3 audio for as long as I can remember. I know all the tricks when it comes to video. Yet audio has always been mp3.

Recently I got ahold of a movie and was shocked that the audio was ac3 and file size was still normal. I figured ac3 had to be huge cuz thats whats on dvd's. So I started testing and found ac3machine, ac3enc.dll, and besweet (which I have used with gknot before) and discovered I could make an ac3 that was 128kps and was the exact same size as my audio in mp3 format. That blew my mind. 5.1 Ch 128kps audio same size as mp3 2 ch.

In the ac3machine guide it says "Set whatever bitrate you see fit. Going below 224kbit/s for a 5.1ch AC3 doesn't make much sense and going above the input bitrate doesn't make any sense either."

Then I read the above. Yet At 224 that audio was twice as big 220megs, no good. 128 same size as Mp3 @ 128 around 125megs.

So my question is this. Is ac3 5.1 128kps audio better then mp3 2ch 128kps audio that I have been using for years?

I have a 5.1 ch system and from what I can hear the 5.1 sounds good in 128kps except for a lil hiss. So what am I missing. Is it better quality then mp3 or not?

Would that ac3 encoded to 128kps be better with a different encoder?

Ac3machine uses the ac3enc.dll and I know that its not as good as other encoders. Can I use the sonic foundry soft encode dll with ac3machine, and how would I do that? or do I have to use soft encode to do it all? Soft encode is slow, is there a fast way to encode with ac3machine and not use the ac3enc.dll?

Any suggestions would be helpful. I know that both AAC and mp3 both have 5.1 surround options would using one of them result in a file around 125megs.

THANK YOU

Last edited by 00diabolic; 25th January 2005 at 22:07.
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Old 30th January 2005, 16:57   #51  |  Link
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Well, you should test that the AC3 in that file is indeed 5.1, because it may perfectly be a 2.0 file. At 128 kbps you got 64 kbps per channel, which is not that bad for AC3 compression.
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Old 23rd February 2005, 19:40   #52  |  Link
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How about the effect of sample rate on the endquality of the AC3? All DVD's I've come across are 48kHz, but SoftEncode lets me select any other resolution as well. Say my source is 44,1kHz, would it yield better results if I converted this puppy to 48 kHz before encoding?
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Old 23rd February 2005, 21:39   #53  |  Link
ursamtl
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Quote:
Originally posted by Buggle
How about the effect of sample rate on the endquality of the AC3? All DVD's I've come across are 48kHz, but SoftEncode lets me select any other resolution as well. Say my source is 44,1kHz, would it yield better results if I converted this puppy to 48 kHz before encoding?
The sample rates determine the end use of the file. 44.1kHz files are normally used for creating AC3WAV files that can be written to a regular CD as if they were tracks for an audio CD. If this CD is then played back by a DVD player hooked up to a decoding receiver via a digital coax or optical connection, it fools the receiver's decoder into thinking it's an AC3 stream coming from a DVD and thus you have a surround audio CD!

48kHz is for use as DVD video sountracks.

In general upsampling will not "improve" an audio file per se. The higher the sampling rate when recording or digitizing sound, the higher the frequency range. However, once the sound is digitized or recorded, upsampling will not add to or improve the information that's already there. Besides, if you had the same sound recorded at 44.1kHz and 48kHz with all other conditions being equal, it would be extremely difficult to hear any difference. The upper limit of normal human hearing is about 20kHz. So whether a sound source is digitized at 44.1kHz with a high-frequency limit of about 22.05kHz or 48kHz with a high-frequency limit of 24 kHz, only your dog will notice a difference!
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Old 1st March 2005, 22:01   #54  |  Link
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Guys, what is the right way to adjust volume of the AC3 file? I have several clips I'm going to use for DVD authoring and I need to set their relative volumes.

I assume I can do this by changing DialNorm parameter - without recompressing. Is it correct? Are there any tools? Does anybody know where in thje AC3 file DialNorm is located?

I did try to encode AC3 with different DialNorm and they play with different volume in WinDVD and standalone DVD player. PowerDVD and Media Player Classic play them with the same volume - seems like they normalize all AC3 clips and pay not attention to DialNorm.
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Old 29th May 2005, 12:59   #55  |  Link
livius76
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i have a movie sound track with -24.6 RMS
found in SoundForge...so i have to put in AC3Machine an Attenuate Volume by -31-(-24.6)=-6.6db ~ -7, or a Gain -7 ?
thk i try Acid Pro for a strightfourd method...


Thx a lot & keep in touch!
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Old 18th October 2005, 20:38   #56  |  Link
maa
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The first two links in the guide no longer exist - please delete this post after correction - thanks

maa
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Old 18th October 2005, 21:38   #57  |  Link
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Quote:
Originally Posted by maa
The first two links in the guide no longer exist - please delete this post after correction - thanks

maa
Thanks, edited in archive.org links.
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Old 21st October 2005, 03:04   #58  |  Link
SomeJoe
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Here are some updated links (Dolby has updated their site):

Standards and Practices for Authoring Dolby Digital and Dolby E Bitstreams

Dolby Digital Professional Encoding Guidelines

And here's a new one that has a nice explanation of every metadata parameter in an AC3 stream:

A Guide to Dolby Metadata
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Old 26th December 2005, 23:33   #59  |  Link
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Quote:
Originally Posted by SomeJoe
The second page is the Bitstream Information. Set these parameters are appropriate for your source material.
I was wondering what the "Dolby surround mode" is for? If I have a normal TV source audio, should I be setting it to NOT? Thanks for this great guide!
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Old 16th January 2006, 11:08   #60  |  Link
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Hi all...

My first post here, so please excuse me if I shouldn't be adding this as a reply rather than creating a new thread, but this seemed the most appropriate place for it to go.

I have a 5.1 AC3 file that I have separated into it's 6 mono channels via BeSweet (BeLight to be exact). I've then loaded these back into Softencode to create a new AC3 stream, but with a data rate of 384 instead of the original 448. I used softencode to get the specific information from the original AC3, which looks something like this...

File size: 195,442,353 bytes
AC-3 File type: Non-Intel byte order (0x0b)
Total frames: 109,063
Frame size: 1,792 bytes
Sample rate: 48,000 Hz
Data rate: 448 kbps
Audio coding mode: 3/2 (L, C, R, l, r) LFE
Bit stream mode: Main audio service: Complete main
Dialog normalization: -27 dB
Center mix: -3 dB
Surround mix: -3 dB
Copyright: On
Original: On
Start time: 00:00:0.00 *
End time: 00:58:10.02
Room type: Large room, X curve monitor
Mix level: 105 dB SPL


... I've used these settings to re-encode, with the obvious exception of the data rate, and then created the new file.

After finishing, I played back the new AC3 and found it was much quieter than the original. To make sure it wasn't just my ears playing tricks on me, I used BeSweet again, but this time on the new file - separating it down to new 6 mono channels. I opened them up in Sound Forge, compared them to the originals, and there was a clear drop in the peaks.

I repeated the process again, but this time turned all the filters and compression off in the preprocessing tab. Re-encoded again, and again the same results.

Thinking it might possibly be down to the change of data rate, I went through it all again, and this time kept it at 448 - the same as the source. Once again it produced the same results as before... a quieter stream.

Now I'm at a loss to think what I'm doing wrong. All I really want to do is duplicate the original AC3 file, keeping it the same volume. Other than raising the levels on the main arrange page (which I would've thought would introduce clipping), I can't think what to do.

Any thoughts or suggestion would be greatly appreciated.

Apologies again if I'm posting this in the wrong place. I did look around the forums for any answers, but couldn't find any.


Edit: Just to add, the mono waves taken from both the original and newly encoded AC3 files were created using the '32bits Mono Waves' option in BeLight... not the '16bits'... whether that would make a difference.

Last edited by desta; 16th January 2006 at 12:21.
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