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audio sampling rate question

Started by Phil April 27, 2008
with the newer flash recorders I now have higher resolution and sample 
rate options then my older 44.1/16 PCM

I realize I can record at lower safer levels with the 24bit resolution 
but I was wondering if sampling at 96 or 88 and resampling to my final 
level of 44.1 after post processing has any advantage.


-- 
Phil


Interesting question. We are investigating the similar topic, for
industrial data acquisition purpose. In our case, the advantage of
retaining higher sampling rate and dynamic range is obvious. It leaves
more room for post-processing. Is your application pure audio?

James
www.go-ci.com
Phil wrote:
> I realize I can record at lower safer levels with the 24bit resolution > but I was wondering if sampling at 96 or 88 and resampling to my final > level of 44.1 after post processing has any advantage.
I would wonder if there are consumer devices out where 48/16 is the limiting factor. Usually the SNR is limited more or less below 90dB by anything else in the signal path. And most devices use oversampling anyway. Normally the dynamics and bandwidth of 48/16 is sufficient for any home audio application. Of course, if we are talking about professional recording devices at least the 24 bit are useful because it makes the recording level much more safe. And for measurement purposes 96kHz is useful too. Whether 88 or 96kHz is useful for your post processing is another question. In theory it is not, unless you do a frequency transformation. Marcel
Phil wrote:

> with the newer flash recorders I now have higher resolution and sample > rate options then my older 44.1/16 PCM
> I realize I can record at lower safer levels with the 24bit resolution
I agree. I now have a 24 bit WAV file and a C program I wrote to reduce it to 16 bit with the appropriate shift.
> but I was wondering if sampling at 96 or 88 and resampling to my final > level of 44.1 after post processing has any advantage.
Not that I know of. Unless you have a better digital filter than the device has to reduce to 44.1 the advantage should be pretty small. -- glen
> I agree. �I now have a 24 bit WAV file and a C program I wrote to reduce > it to 16 bit with the appropriate shift. >
On average how much compression ratio have you achieved? James www.go-ci.com
Marcel M�ller wrote:
> Phil wrote: >> I realize I can record at lower safer levels with the 24bit resolution >> but I was wondering if sampling at 96 or 88 and resampling to my final >> level of 44.1 after post processing has any advantage. > > I would wonder if there are consumer devices out where 48/16 is the > limiting factor. Usually the SNR is limited more or less below 90dB by > anything else in the signal path. And most devices use oversampling > anyway. Normally the dynamics and bandwidth of 48/16 is sufficient for > any home audio application. > Of course, if we are talking about professional recording devices at > least the 24 bit are useful because it makes the recording level much > more safe. And for measurement purposes 96kHz is useful too. > Whether 88 or 96kHz is useful for your post processing is another > question. In theory it is not, unless you do a frequency transformation. > > > Marcel
Marcel basically covered it. In general, it will depend on what processing you do. Because sampling with more quantization levels and at a higher rate will give you a better SNR. And any processing, processes the signal and the noise. So if your processing has some feedback that creates unhappy situations with processing noise. But I am guessing in you situation any such processing is probably not going to be encountered or significant. Most ADCs internally oversample and then downsample to get better SNR. Interestingly many also use less quantization levels and much higher sampling rate, which sort of cancel each other out. J. Elms
DigitalSignal wrote:
>>I agree. I now have a 24 bit WAV file and a C program I wrote to reduce >>it to 16 bit with the appropriate shift.
> On average how much compression ratio have you achieved?
The advantage is that you don't have to guess as accurately what the record level will be. I don't know about the OP, I was recently using a Roland R1 recorder. It has the choice of 16 or 24 bit WAV, and some choices of MP3. With 24 bits, I then wrote a C program to find the minimum and maximum sample values, along with a some other statistical values. I then did the conversion to 16 bits by rounding and shifting the appropriate number of bits. -- glen
On Tue, 29 Apr 2008 07:25:29 -0800, glen herrmannsfeldt
<gah@ugcs.caltech.edu> wrote:

>DigitalSignal wrote: >>>I agree. I now have a 24 bit WAV file and a C program I wrote to reduce >>>it to 16 bit with the appropriate shift. > >> On average how much compression ratio have you achieved? > >The advantage is that you don't have to guess as accurately >what the record level will be. > >I don't know about the OP, I was recently using a Roland R1 >recorder. It has the choice of 16 or 24 bit WAV, and some >choices of MP3. With 24 bits, I then wrote a C program to >find the minimum and maximum sample values, along with a some >other statistical values. I then did the conversion to 16 bits >by rounding and shifting the appropriate number of bits.
This doesn't sound right. If you want the best quality possible in the final 16-bit signal, look into dithering and noise-shaping rather than doing rounding, and it's best to do all your processing in floating point (that way you can change the level to any arbitrary value, rather than only in the 3dB steps that bit shifting gives). OTOH, the difference can be subtle, and many people can't tell even under the best listening conditions.
> >-- glen
Ben Bradley wrote:
(snip, I wrote)

>>I don't know about the OP, I was recently using a Roland R1 >>recorder. It has the choice of 16 or 24 bit WAV, and some >>choices of MP3. With 24 bits, I then wrote a C program to >>find the minimum and maximum sample values, along with a some >>other statistical values. I then did the conversion to 16 bits >>by rounding and shifting the appropriate number of bits.
> This doesn't sound right. If you want the best quality possible in > the final 16-bit signal, look into dithering and noise-shaping rather > than doing rounding, and it's best to do all your processing in > floating point (that way you can change the level to any arbitrary > value, rather than only in the 3dB steps that bit shifting gives). > OTOH, the difference can be subtle, and many people can't tell even > under the best listening conditions.
I thought about it, but so far I haven't tried. By finding the peak and appropriate shifting, the quietest parts aren't so far down. It is a recording with a live audience, and the background isn't all that quiet, anyway. It would be nice, though. Do others do it? -- glen
On Tue, 06 May 2008 20:51:45 -0800, glen herrmannsfeldt
<gah@ugcs.caltech.edu> wrote:

>Ben Bradley wrote: >(snip, I wrote) > >>>I don't know about the OP, I was recently using a Roland R1 >>>recorder. It has the choice of 16 or 24 bit WAV, and some >>>choices of MP3. With 24 bits, I then wrote a C program to >>>find the minimum and maximum sample values, along with a some >>>other statistical values. I then did the conversion to 16 bits >>>by rounding and shifting the appropriate number of bits. > >> This doesn't sound right. If you want the best quality possible in >> the final 16-bit signal, look into dithering and noise-shaping rather >> than doing rounding, and it's best to do all your processing in >> floating point (that way you can change the level to any arbitrary >> value, rather than only in the 3dB steps that bit shifting gives). >> OTOH, the difference can be subtle, and many people can't tell even >> under the best listening conditions. > >I thought about it, but so far I haven't tried. By finding the >peak and appropriate shifting, the quietest parts aren't so far down. > >It is a recording with a live audience, and the background isn't all >that quiet, anyway. It would be nice, though. Do others do it?
Yes, I'm pretty sure most all audio editing software (everything from Audacity to Pro Tools) have been doing it as I described for many years now. It might be easier to write some script file for an audio editing program to do what you want. I don't know what programs have what features nowadays, but the old Cool Edit 2000 has/had scripting so you could do several automated things with audio files.
> >-- glen