best way to convert 24b multi channel into 16 bit?

The audio was recorded 48,000 24 bit but i need 16 bit. The files have 3 or 4 channels embedded. Is there a cheap or free way to convert those from 24 bit to 16 bit?
 
Why do you need to convert the files? A more comprehensive view of the workflow may suggest the most practical answer. Presumably you have 24 bit files to start with. But does the next step (not disclosed) accept only 16-bit files? Or can it perform the conversion upon import, or as a explicit operation? We typically want to preserve original data (including audio files) in the original format to avoid premature or arbitrary loss of quality, etc.

Have you investigated the "usual suspects" of free audio apps? I would be that most (all?) of them will do this conversion. http://www.hongkiat.com/blog/25-free-digital-audio-editors/

Note also that simply chopping off the low 8 bits is not considered to be an optimal method. Better methods use dithering to produce less artifical results.

If the question is specific to "multi-channel", then we may need to get down to specifics examples as there are no practical generic answers.
 
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i think you extract the tracks to mono files in SD's WaveAgent. Don't know about the bit depth conversion ability of it. For editing poly files, I use Sound Forge Pro with the iZotope suite for any re-sample or bit depth conversion. (Which is rare)
 
I was going to suggest Sound Forge. It is a great program, but I don't know if they offer a free trial version. Richard is right, use dither rather than straight conversion.

GJ
 
Keep them 24-bit throughout the editing process, converting to 16 when you render the final mixed stereo soundtrack with the video.
 
Any question about software has to start with what platform you are on.

In this case the question is probably not a real on in that I doubt you need to convert to 16 bit. You probably do need to separate the individual tracks but? Most programs you would be working with will do both seamlessly on import. Though as others have said I would keep the audio 24 bit till the final mixdown.
 
To expand on the questions.
Basically i have a bunch of interviews recorded with Canon 7D with audio recorded externally at 48,000 24bit. When ingesting the video and audio into FCP 7 and trying to sync we notices audio drift (when trying to do the same with Sony's Vegas everything was fine - which leads me to belief that the way FCP converts 24 into 16 bit is funky). When i converted 24 into 16 bit (where 3 tracks got merged into 2) and outputted the file and tried to sync up the audio in FCP this time audio lined up perfectly. So again my assumption is that "my" FCP simply doesn't like 24 bit for whatever reason so i need it converted. Hope it helps.
 
Final Cut does support 24bit. You have to go into your sequence settings and change it from 16 to 24. See if that does the trick although I haven't heard of this before. There are other issues with audio and Final Cut but your issue is alarming. Changing bit depths doesn't change the speed of the audio, the sample rate does.
 
There is really no way that changing the bit depth will have any effect on the audio-video sync. There are many possibilities in that work-flow and it is some other factor (or combination of factors) that is affecting the sync phenomenon. Bit-depth is the wrong path. It is only a coincidence.
 
You are right in assuming that FCP is funky with sound but the bit depth has nothing to do with it. I suspect that you are coming across the FCP frame rate bug. Is your footage by any chance in a full frame speed (30fps, 25fps, or True 24fps)? If so and you have worked at NTSC speeds before (29.97, 23.98, 59.94) then this is very likely the problem.

Here is my last blog post about this with a fix. If you want to go way down the rabbit hole with this you can look at a very long thread on it at the DUC.

The gist is that FCP can get confused and will then convert your sound with out telling you and do it wrong.
 
As Noiz suggested it's the FCP's way of dealing with frame rates.
After converting the audio to right frame rate through Wave Agent everything seems fine.
 
The irony of course is that there is no "frame rate" for audio. The conversion just puts the FR stamp in the metadata that FCP wants to see.

Things like this are why everyone who loves Apple also hates them, at least some of the time.
 
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