low bit rate audio

[codename_46]

I have GKnot .27 and need to encode an AC3 file into a mono MP3 at 64kbps maybe even 48kpbs. Why such a low bit rate? I'm making a movie that can play on my PocketPC and size and bit rate are a concern since it lacks the power of my P4. At first I tried just getting besweet to encode it into 48 and 64kpbs both it would always give me errors & crash. So I just converted it to a mono .wav and then used another app to get it into a 48kbps MP3. But now when I tell GKnot just mux the video plays back at near 2x speed while the audio remains constant. :confused:
SO!
Is there a way to get besweet to just encode into 48kpbs mono?
Or what could I have bumped in GKnot to make the video run so fast?
Some help please?

[slimz]

nm everything i wrote here. im half asleep.
try avifrate maybe.
http://www.doom9.org/Soft21/Editing/avifrate110.zip

[Slogra]

When you keep the mp3 44.1khz the video should be fine... this won't work in Besweet so you'll have to make a .wav first again and then encode to mp3.

EDIT:
You can use the 56kbps voice preset with 44.1 mono with Lame, like this (i used it with a 4 hour divx once):

--preset voice --resample 44100

Or you could use this 37kbps ABR line which i picked up at http://www.hydrogenaudio.org/ :

--short --athtype 3 --ns-sfb21 -3 --lowpass 10.6 --cwlimit 10.7 --verbose --vbr-new --abr 37 --Y -q0 -b 8 -B 64 -m m --resample 44100

And if you're at http://www.hydrogenaudio.org/ you might as well ask what the best settings are for a 48/64kbps 44.1khz mp3. They know everything about mp3.

[Hiro2k]

Well I would also recomend Ogg Vorbis for low bitrate audio. It produces a smaller filesize than MP3 and also sounds allot nicer too. You can't mux ogg into AVI, so you will have to use one of the new container formats. It's not to hard to use the OGM Container, or if you want to, use the Matroska Container. This can all be done in VirtualDubMod, if your interested.

[codename_46]

Finally figured out the problem. Encoding at such a low bit rate was causing lame to change the sampling rate from 44.1kHz to 32kHz or something else. When nandub saw audio that wasn't 44.1kHz it was playing back the video at 2x speed! Once I forced lame to keep 44.1kHz and just encode down to 32kpbs nandub was able to merge the files and everything syncs perfectly.
:)

Now.. this being done. Was there some way I could have passed nandub 22.05kHz audio and set it up to where it would properly sync?? Just curious.

[Hiro2k]

Did you try VdubMod? Maybe that one will keep your audio in synch.

[bond]

using virtualdubmod really makes sense
downsampling really makes sense at such low bitrates (downsampling in lame is buggy, use ssrc for that)
and at such bitrates using ogg vorbis also makes sense

imho ;)