Log in

View Full Version : Audio, delay and conversion


Shandra
2nd August 2004, 19:40
Ok, as I just started to experiment/learn on other containers then avi (including the research here and over at the HA forum) I encountered the audio/video sync. issue again.

When using Matroska with mp3/ogg I can stick to my used to GK, and for aac I just use the mkvtools...

But now I also wanted to try out mp4 as a container (and yes, I have read the faq (http://forum.doom9.org/showthread.php?s=&threadid=62723) and the guide about mp4 streams at doom9) and as I so far hasn't found another statement as that you have to take care for the av sync. before muxing the two streams into the container (and that there is nothing like a delay switch for the utils so far (?)).

So I have to incorporate the delay within the encoded audio file itself, and as I haven't done this since early divx times where I used HeadAC3 (if I remember it right) for that purpose (and nearly forgoten all about it), and the guides at doom9 name aacmachine/beesweet for that purpose and so far I only used BeSweet through GKnot (where the delay is set via VDubMod and not through BeSweet) I am not experienced with BeSweet itself (and somehow likes the idea of encoding to aac with Foobar) I now am only wanting to convert the original ac3 to a lossless format modified with the delay information
(ok, I could still use HAC3, but lets assume that at this stage I consider to keep the 5.1 information asfaik thats not possible with it).

So I now want to know/verify a few things about the BeeSweet GUI (0.7b4) (or some other alternatives) -

The delay is incorporated (the file cut/padded right) if I just set under the BeSweet Tab within the OTA section the delay option with the desired value and thats all to it (the append silence option is confusing me?) ?
And is using 5.1 DD-Wave as Output ok in this case (don't know about wav, always thought that there are lossless and lossy types of wavs) ?
And should I still use under the azid 1 Tab the LFE to LR Channels option with -3db, or should I have all options unmarked (except maybe for the dynamic compression) ?

As I am more confused or distracted by all the threads I found with some searches (or may have not found by it) & I at least want the subtle feeling of "knowing what I do" is the above correct, or is there another "fool-proof" way ;) Thanks!

Edit: Damned, I just should had run HeadAC3 before posting - ok, all I have to do is using it (6 channels, set delay) and having a large amount of Diskspace free - Sorry, but I am still curious wether I have understood the BeSweetGUI properly or not ;)

killingspree
2nd August 2004, 21:35
sorry, but alternative containers are not really within the scope of the newbie forum! for your convenience, I'll move you to the 'New A/V containers forum'! - there'll you'll be able to get all the help you need!

cheers
steVe

bond
2nd August 2004, 21:42
you can use an a/v delay when muxing with 3ivx as described here (http://www.3ivx.com/support/windows/encoding/ge_shift.html), but basically its indeed better to change the delay already in besweet

if i understood it right, your questions are mainly related to how to handle the delay settings with multichannel tracks in besweet, right? so i pass it on to the audio forum :D

Shandra
16th August 2004, 00:16
Ok, after the forum was down... and I am still not able to paraphrase my question corectly.... but have learned much through reseacrh in the meantime (and you all know - research just add up to irritation because you just encounter more contratictions)....

The Guides I found here are for anything but not wav....

So to obtain a delay modified and downmixed to 2ch wav from a 5.1 dts/ac3 source.... to use it as a base for any codec I like (wich codec with what settings may maintain the dolby information - well, thats the faq here... but nowhere is mentioned if it is transparent in plain stereo PCM wav)...

In regards to BeSweet... Should I use LFE to LR in such a case, should I enable dialog normalization in on BeSweets Azid Tab2 or not (the Doom Guides never mention the 2nd settings page, brother john (http://home.arcor.de/brotherjohn/) (german) advices to set it....

Either I really am in need of a whatever source (multichannel/dolby 2ch/mono/etc) to wav guide - or some transfered explanation of those settings as I am unable to do it myself (its just that the FAQs and guides are mostly with a single purpose in mind or those who are talking about the topic in question (dolby, etc.) are either make use of another program or are just too far beyond my grasp)....
[Edit: its an or... as you may just look at the audio guides at doom9 - with the exception of AAC they make use of BeSweets Gain, with AAC its Azids gain used...it make sense in that application context - but is irritating if I focus on wav as an inbetween... (more irritating is that new BSGUI uses the new azid tags dpl and dplii, but AACMachine, etc. are still using the old cli parameters - and no guide/faq I found is telling me if the newest azid is backward compatible to them or ignoring them... sigh]



I do not share, but I go to other users to verify (as I think that if other OS/Systems are able to properly playback what I am doing - the chance of it beeing a good choice is better (especially as I am afraid of not beeing able to playback my encodes in the same way as it is now with the next system I buy - and on my 4 systems my latest MP4 container test with AAC sound was good [edit: ok on 2, as neither my Cel500, nor my plain P200 (nonMMX) were able to offer the power needed - but the sound alone was good on all of them] on all my PCs, but the sound failed under *nix systems (was 5.1 AC3 downmixed to LFE to LR -3db, dpl with normal dynamic compression (& no dialognormalization on azids 2nd tab, downsample to 44k), Postgain by BeSweet & delay incorporated by BeSweet... on my systems the AAC from that wav & derived from that AC3 sounded ok, on a *nix system with MPlayer that MP4 (muxed with mp4box) was strange - the xvid stream was ok but the sound had really strange effects wich I did not experienced on my systems [all win, 95B,98SE,2kPro,XPHome] (voices only on right channel, echo just on left&really strange up/down (volume) bubbling sound in general)... &that is something I do not want to experience If I may switch my own system - that previously good encodes turn out to be totally BS.... We tried it again with mkv as a container and it was the same - on my systems (soundcard[ISA SB64AWE,SB PCI128,SBLive (1st generation),Teratec XFire1024]) ok, on his a nogo (AC97 onboard Sound)....