Which is higher quality--mp2, mp3, aac

[Erik Norgaard]

Originally Posted by Daniel Ross

No. Don't save to a lower format.

The 48 is a container, and can hold up to 24khz in it.

I'd need to brush up on my physics to give a better explanation. Can't remember right now.

Simple: a 24kHz sound has 24.000 oscilations per second, if you look at the sinus curve, there are 24.000 ups and 24.000 downs. To represent all of them you need 48.000 numbers minimum per second, which can exactly indicate the max amplitude of the wave in each up or down.

Of course, this will be very squared, lower frequencies are better represented. But this is OK since 24kHz is way beyond recognizable by the human ear. The higher the sampling frequency, the less "squared" the curve is in the audible spectrum.

The sample frequency just means how many chunks you split a second into. So, for 24kHz, you need 48k chunks per second, or 48kHz sampling frequency.

Cheers, Erik

[Jim Andrada]

I always also thought that frequencies above 20kHz were imperceptible, but lately there seems to be some thought that the higher frquencies are in fact perceived although not "heard" in the normal sense of the word.

Isn't DVD sample rate 96kHz? If there's no effect perceptible to humans, why would they use the higher sampling frequency?

Not being snippy, just curious.

[Erik Norgaard]

Originally Posted by Jim Andrada

I always also thought that frequencies above 20kHz were imperceptible, but lately there seems to be some thought that the higher frquencies are in fact perceived although not "heard" in the normal sense of the word.

Isn't DVD sample rate 96kHz? If there's no effect perceptible to humans, why would they use the higher sampling frequency?

Not being snippy, just curious.

It is possible that audio-dvd (not to confuse with video dvd) have a higher sampling rate, and I think that the production stage is done at a higher sampling frequency.

But think of this: At 48kHz, one oscillation at 24kHz is represented by two numbers giving a very square wave. At 12kHz you get 4 numbers, but still a very square wave. At 6kHz you have 8. It starts to get detail but now you get into the sensible area.

If you start at 96kHz you get a much smoother wave form at 6kHz, you will have 16 numbers for one oscillation. So, the higher sampling frequency doesn't add more in terms of audible frequencies, but does give a truer representation of the sound.

I guess that is why audiophiles claim to be able to hear the difference between analog and digital recordings.

Cheers, Erik

[Jim Andrada]

I think you're on to something there because my classical pianist wife always complains about the "coldness" of digital recordings. I notice it too. Could well be due to the "squaring of the higher frquencies.

So we got a CD player with a vacuum tube final stage - sounds better to both of us.

[Daniel Ross]

Thanks, Erik. Now it's less vague. That's generally what I thought, but it was very unclear.

[Dylan Pank]

Forget FFmpegX - it's great for making mpeg4 files and not bad for progressive scan DVDs or VCDs, but it's not a serious audio tool (and was never intended to be).

You want Audacity: http://audacity.sourceforge.net/. It's free, despite being a brilliant audio editor! (the 1.2.6 version is stable and a little easier to use and has the features you need.)

It can import .ogg files and output them as 48khz .aiff files which is what FCP wants. There are loads of settings that allow you to optimise the resampling from 24khz or 44.1Khz up to 48Khz, but if you leave it on the defaults it will do a pretty good job. There are extensive help files, online documentation and a forum, but basically it will quite easily do what you want.

[Elmer Lang]

Another interesting thread and thanks to all for the tips and explanations!