Welcome to Doom9's Forum, THE in-place to be for everyone interested in DVD conversion. Before you start posting please read the forum rules. By posting to this forum you agree to abide by the rules. |
|
|
Thread Tools | Search this Thread | Display Modes |
20th August 2004, 18:26 | #41 | Link |
Registered User
Join Date: Aug 2003
Posts: 7
|
@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 |
20th August 2004, 21:37 | #42 | Link | |
Registered User
Join Date: May 2004
Location: Montreal
Posts: 729
|
Quote:
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. |
|
23rd August 2004, 13:05 | #43 | Link |
Registered User
Join Date: May 2004
Location: Montreal
Posts: 729
|
One other good reference for AC3 of course is the www.dolby.com site in their Information menu.
Steve. |
1st September 2004, 21:14 | #44 | Link |
Registered User
Join Date: May 2004
Location: Montreal
Posts: 729
|
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. |
11th October 2004, 08:27 | #45 | Link |
Registered User
Join Date: Mar 2003
Posts: 99
|
"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. |
14th October 2004, 18:36 | #46 | Link | |
Registered User
Join Date: May 2004
Location: Montreal
Posts: 729
|
Quote:
There's a good explanation of this at YOU ARE SURROUNDED. Regards, Steve. |
|
13th January 2005, 15:55 | #49 | Link |
Registered User
Join Date: Jan 2005
Location: Great White North
Posts: 326
|
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! |
25th January 2005, 22:04 | #50 | Link |
Registered User
Join Date: Jan 2005
Posts: 7
|
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. |
23rd February 2005, 19:40 | #52 | Link |
Guest
Posts: n/a
|
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?
|
23rd February 2005, 21:39 | #53 | Link | |
Registered User
Join Date: May 2004
Location: Montreal
Posts: 729
|
Quote:
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! |
|
1st March 2005, 22:01 | #54 | Link |
Registered User
Join Date: Nov 2004
Posts: 19
|
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. |
29th May 2005, 12:59 | #55 | Link |
Registered User
Join Date: Feb 2005
Posts: 5
|
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! |
18th October 2005, 21:38 | #57 | Link | |
Registered User
Join Date: Jun 2003
Location: Great Lakes, USA
Posts: 1,433
|
Quote:
__________________
KpeX Audio FAQs: General | BeSweet | SVCD/MP2 | MP3 | Vorbis | AC3 | DTS | AAC Linux Audio/Video FAQ |
|
21st October 2005, 03:04 | #58 | Link |
Registered User
Join Date: Jan 2003
Posts: 315
|
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
__________________
- SomeJoe |
26th December 2005, 23:33 | #59 | Link | |
Registered User
Join Date: Dec 2002
Posts: 218
|
Quote:
|
|
16th January 2006, 11:08 | #60 | Link |
Registered User
Join Date: Jan 2006
Posts: 220
|
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. |
Thread Tools | Search this Thread |
Display Modes | |
|
|