Convert at Low Bitrates at 48Khz in Stereo

[ashwin.terminator]

I've got the latest K-Lite Pack and all that, but in my virtualdubmod and virtualdub, I get the stereo option at 48Khz and 44.1Khz only upto 128kbps, If i gotta encode at 64 kbps or 96kbps or 112 kbps, it has to be in one channel. How do I get to convert audio at 48Khz or 44.1Khz at the above bitrates in stereo using VirtualDub or VirtualDubMod/

[ashwin.terminator]

Also, any idea why my VirtualDub doesn't encode at EXACTLY the same bitrate I enter?? Also how do i use the In-built bitrate calculator in Vdub and Vdubmod to get files of exactly 700MB?? Using the same thing, my files are generally ablve 700MB...

[jagabo]

Many MP3 encoders will not allow very low bitrate with 48 or 44.1 KHz samples because they know it will sound bad. Using a lower sample rate, although it eliminates the high frequencies, will sound better because there's less data to encode.

Many encoders often have problems meeting the requested bitrate -- especially at low bitrates. Try a different codec.

[tomlee59]

Also, any idea why my VirtualDub doesn't encode at EXACTLY the same bitrate I enter?? Also how do i use the In-built bitrate calculator in Vdub and Vdubmod to get files of exactly 700MB?? Using the same thing, my files are generally ablve 700MB...

I'm glad jagabo was able to figure out what you were asking -- you meant bitrates *down to* 128kb/s, not "up to." And indeed the reason for the lower bound is exactly as he says: Low bitrates will sound bad, so the author of the software wanted to prevent you from doing something that will produce results you'll complain about. If you are really determined to go below the recommended bitrates, you can use a different tool for the audio conversion. Audacity (free), for example, will allow you to go to absurdly low bitrates, so you can hear for yourself what happens.

As for the bitrate/filesize issue, there are two factors to consider. One is that the target bitrate you select is just that: a target. The encoder will adjust things dynamically in an effort to meet the target, but it is not an exact science. Expect some variation. If needed, adjust the target downward if the output is consistently too high.

The other factor is the difference between binary and decimal reporting of filesize. A binary kilobit is 1024 bits, instead of the decimal 1000 bits. Similarly, a binary megabit is 1024*1024 bits, not 1000000 bits. Thus, there can be several percent seeming discrepancies depending on what numerical base is being used.

[ashwin.terminator]

But in that case, ain't I supposed to make an audio file for every individual file I wanna encode at 32 kbps at 44.1Khz?? Is there absolutely no way for me to do that at VDub or DubMod??

[tomlee59]

First, I suggest that you try a simple experiment: Encode a short audio sample at 32kb/s and see what happens. A stereo track will be unbearably, intolerably bad if the sampling rate is 44.1 or 48kHz. Even mono will sound terrible at that bitrate. Well-designed tools employ "forcing functions" to prevent users from doing things that will be a waste of time. If you are really determined to do this, you will likely have to use a different tool. Perhaps there is a way to make Vdub do this, but I've never attempted it.

What is your actual objective? Are you trying to encode an audio track for a standard format, like VCD or DVD? If not, then is there some other reason that you need to adhere to a 44.1/48kHz sampling rate? If you are free to alter the sampling frequency downward, then you can get much better-sounding results at the bitrates you've been talking about. The result will have less treble, but what does get encoded will suffer much less from the "hollow" and "ringy" artifacts that accompany excessively low bitrate MPG encodes.

Perhaps if you could say more about your goals, someone here could offer better advice.

[chh]

If - for whatever reason - you need to modify the audio sample rate from 48KHz to 44.1KHz or anything else, and/or you need to reduce the number of channels from 2 to 1, or whatever audio settings which can not be handled within Virtualdub, recode audio separately with SUPER from Erightsoft:
https://www.videohelp.com/tools/SUPER

Multiplex (=merge) the video- and audiofiles with Virtualdubmod.

[tomlee59]

Of course there are many tools that can do this. Audacity, IMO, is a superior tool, since it is dedicated to audio processing alone.

[JimmyS]

You can make any MP3 encoding setting changes directly from within VDub. Make sure you have the LAME MP3 ACM codec installed. If you don't have it get it from the tools section. Use the direct link. The ACM codec is a seperate zip file that is in the download, so you'll have to unzip twice to get it. Right-click the LameACM.inf file and select install. Open Vdub, go to Audio - Compression, select Lame MP3. Check the "Show all formats" checkbox to the right. You should see settings from 48 KHz, 320 kbs to 8 KHz, 16kbs stereo and mono. If you are changing the sampling rate and/or stereo to mono etc., you have to open the Audio - Conversion dialog and make your changes there, or VDub will give an error message when you convert.

[ashwin.terminator]

No, I still don't get it! I still don't get the option of Stereo at 48 or 44.1Khz at 64 kbps! Any other codec?

[jagabo]

Lame will not let you go below 128 kbps with 44.1 or 48 KHz sampling. I don't know of any ACM MP3 encoder that will. Windows includes an MP2 encoder that allows 64, 96, and 112 (as well as higher) kbps.

[ashwin.terminator]

hmm, but audacity takes too much parsing the video file(I'm on a 256MB RAM, 1.7Ghz P4 system)

[poisondeathray]

avidemux will allow you to use stereo mp3 below 128kbps with 44.1 or 48Khz

But it will sound better using aac than mp3

[ashwin.terminator]

btw, can any of you mention the difference between K-Lite Codec Pack and K-Lite Mega Codec Pack?

[jagabo]

btw, can any of you mention the difference between K-Lite Codec Pack and K-Lite Mega Codec Pack?

One screws up your system, the other screws it up even worse.

[ashwin.terminator]

lol!!

[zoobie]

nobody in the know uses them
...and you need a new thread

[ashwin.terminator]

fine enuf of off topic discussions... any more ideas??

[tomlee59]

hmm, but audacity takes too much parsing the video file(I'm on a 256MB RAM, 1.7Ghz P4 system)

Since you still haven't stated what your actual goal is, your comment makes little sense. No matter what you do, something will have to "parse" your media file to convert the audio. Audacity is not the fastest too in the world, but it's not terribly slow, either, as it uses the same Lamelib module that just about everyone else uses.

The most efficient workflow is to demux first (that cuts out the overhead of parsing) and then convert the audio separately.

But again, without your telling us what your actual overall goal is, we can only provide limited guidance. And you haven't mentioned whether you've actually tried any of these experiments to see if your goals are even sensible.

There are several levels to your problem, you see, ranging from the strategic to the tactical.

[JimmyS]

Lame will not let you go below 128 kbps with 44.1 or 48 KHz sampling. I don't know of any ACM MP3 encoder that will. Windows includes an MP2 encoder that allows 64, 96, and 112 (as well as higher) kbps.

Just to make sure I wasn't having a deja vu all over again, I took a small clip of MPEG2-128k MP2 video and ran it through VDub to XVid and 32KHz-64Kbs_mono MP3. After the transcode, MediaInfo says that its audio is what I set it for. Do I have special powers and abilities far beyond that of mortal men? (wait a minute, thats Superman )

[Gavino]

Lame will not let you go below 128 kbps with 44.1 or 48 KHz sampling. ...

I took a small clip of MPEG2-128k MP2 video and ran it through VDub to XVid and 32KHz-64Kbs_mono MP3. After the transcode, MediaInfo says that its audio is what I set it for. Do I have special powers and abilities far beyond that of mortal men? (wait a minute, thats Superman )

No superpowers, I'm afraid - the issue is about 44.1 or 48KHz, and stereo too.

[JimmyS]

Lame will not let you go below 128 kbps with 44.1 or 48 KHz sampling. ...

I took a small clip of MPEG2-128k MP2 video and ran it through VDub to XVid and 32KHz-64Kbs_mono MP3. After the transcode, MediaInfo says that its audio is what I set it for. Do I have special powers and abilities far beyond that of mortal men? (wait a minute, thats Superman )

No super powers, I'm afraid - the issue is about 44.1 or 48KHz, and stereo too.

I miss read jagabo's reply. Went back in to VDub. Re-looked at Lames compression settings (as per the reply). 44.1K stereo will go down 118Kbs ABR stereo though. (B.F.D.)

Your right, no super powers, more like UnderDog.

Edit: I also have another MP3 codec installed. I think it's an older version of the Fraunhofer codec. It will allow 48 and 44.1 stereo down to 96 Kbs. I don't remember where I got it though.

[jagabo]

You're right, Lame ACM does have 124 and 118 kbps ABR stereo with 44.1 KHz samples. The command line version of Lame goes all the way down to 32 kbps with 44.1 and 48 KHz samples.

[JimmyS]

You're right, Lame ACM does have 124 and 118 kbps ABR stereo with 44.1 KHz samples. The command line version of Lame goes all the way down to 32 kbps with 44.1 and 48 KHz samples.

Cool, I'll have to remember that for future reference . I just never use command line apps. any more cause I'm such a shi**y typist, and point and shoot has made me so very lazy.