Old 12-09-2010, 05:53 PM   #1
djz
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Default 192khz? Does anybody actually track at this rate?

Does anybody here track at 192khz? I know this has been beat to death all over the internet, but really, unless I was planning on recording some slomo audio, trying to detect ghosts in my studio (I'm sure they're there!), I can't really see any legitimate reason for tracking at this rate... for processing/vstis that might have aliasing issues, I can see the benefit of 96khz...but 192???

As an aside, I had been doing a lot of tracking/mixing at 44.1khz simply because my computer couldn't really handle it. Then I started tracking and mixing at 48khz and I noticed the audio quality became worse... I believe this to be because my crappy interface might clock poorer at 48 and thus has a higher rate of jitter...
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Old 12-09-2010, 06:42 PM   #2
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for example using hydrophones or recording bats / insects
Not to mention you'll need a mic that can actually capture frequencies that high to begin with.

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Old 12-09-2010, 07:16 PM   #3
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IIRC, the supposed primary benefit of higher sampling rates is not the capture of higher frequency content, but better rendering of content at audible frequencies.
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Old 12-09-2010, 07:25 PM   #4
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Originally Posted by Analogy View Post
IIRC, the supposed primary benefit of higher sampling rates is not the capture of higher frequency content, but better rendering of content at audible frequencies.
I agree but the term "tracked at 192" was included.. peaked my interest.

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Old 12-09-2010, 08:15 PM   #5
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I did a 2 track classical guitar recording for a client a while back at 192 khz. He asked for it. His reason was for 'archival' purposes. Hedging a bet on the future mass playback capabilities. That actually makes some sense.

It sounded great at 192 khz but by the time it was knocked down to 16 bit 44.1 khz it did lose a bit of 'shine' and 'body. But it still sounded great.
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Old 12-09-2010, 08:26 PM   #6
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Originally Posted by karbomusic View Post
I agree but the term "tracked at 192" was included.. peaked my interest.
It makes sense to track at the same sample rate that you're mixing at to avoid a conversion step. If you're mixing at 192 you're eating a lot of CPU just on mixing and processing, no need to add conversion overhead to that as well.
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Old 12-10-2010, 10:21 AM   #7
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maybe / maybe not - the hydrophones I use go to 100k+ which I think is pretty normal. The speed of sound underwater is a bit over 4 times faster than in air. This affects stereo imaging and the frequency content (depending on what you are recording). It can be useful to slow the waveform down.
What hydrophones do you use, and what sort of sounds do they capture? I have long wanted to do underwater recording experiments, and never even thought of looking for something called a hydrophone...
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Old 12-10-2010, 12:25 PM   #8
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Originally Posted by djz View Post
As an aside, I had been doing a lot of tracking/mixing at 44.1khz simply because my computer couldn't really handle it. Then I started tracking and mixing at 48khz and I noticed the audio quality became worse... I believe this to be because my crappy interface might clock poorer at 48 and thus has a higher rate of jitter...
One thing to be aware of is that you want to track in an even multiple of your final medium (ie tracking for a cd shoud be done at 44k or 88k, tracking for dvd audio should be done at 48k, 96k, or 192k) The reason for that is that on render, it eliminates the averaging errors inherant in trying to distribute 48,000 samples across 44,000 samples. And yes, the difference between cheap A/D / D/A converters is the way they handle high sample rates. A cheap A/D does almost as well as a high end one at 44k, but is noticably worse at 96k.
As for tracking at 192k, it would be mostly for laboratory purposes, or very critical recording situations where you don't want to use pass filters to prevent aliasing around the nyquist frequecy (half the sample rate), so you just make sure that it is well outside of the audible range by ramping up the sample rate.
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Old 12-10-2010, 12:56 PM   #9
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Dan Lavry (who makes some of the best DACs going) argues that converters do not perform as well @ 192k than at 96 so even for archival purposes you are probably better off @ 96 with current SOA converters.
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Old 12-10-2010, 01:15 PM   #10
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This is in-line with what I know about the subject too and goes hand in hand with the fact that higher sample rate don't need as steep filters and although we can't hear the frequencies we can supposedly here timing differences (between the left and right speaker) that equate to wavelengths way above what we can hear.

BUT, it is a total mistake to think that higher frequencies make the frequencies we can hear more accurate BUT, if you have a good/correctly designed filter on the output your get just sine waves and it doesn't get more pure than "harmonic motion"

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One thing to be aware of is that you want to track in an even multiple of your final medium (ie tracking for a cd shoud be done at 44k or 88k, tracking for dvd audio should be done at 48k, 96k, or 192k) The reason for that is that on render, it eliminates the averaging errors inherant in trying to distribute 48,000 samples across 44,000 samples. And yes, the difference between cheap A/D / D/A converters is the way they handle high sample rates. A cheap A/D does almost as well as a high end one at 44k, but is noticably worse at 96k.
As for tracking at 192k, it would be mostly for laboratory purposes, or very critical recording situations where you don't want to use pass filters to prevent aliasing around the nyquist frequecy (half the sample rate), so you just make sure that it is well outside of the audible range by ramping up the sample rate.
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Old 01-25-2011, 12:09 PM   #11
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Default An Essay I have been working on for a while now ...

As I see it , there are three slightly overlapping frequency ranges of interest in what we call "music" ...

The first is what I would call "Temporal" - this is the frequency at which individual musical events take place. Intuitively this is perceived as tempo/beats/rhythm. Temporal events occur in a range of frequencies with a low end of essentially "0" - Cage's 4:22 for example - to perhaps 10-20 Hz. Above a certain (fairly low) frequency the individual events are perceived as cyclic (the "second" range - see below) and are no longer really temporal in the sense I've identified here.

The second range is what most folks would call "Cyclic" - with fundamental frequencies in the range of 16Hz to 4200Hz. Virtually all musical instruments used in modern compositions produce sounds within this range. This is not to say this is the only range of sounds they produce - it is just to say that the fundamental frequency is within this range.

The final range is what I would call "Timbreal" (OK - I made up that word). This is the range of frequencies that comprise the overtones and initial metallic percussion sub harmonics. It spans maybe 2000Hz on the low end to 20KHz and possibly beyond. It is important to note here that every particular instrument generates it's own somewhat unique upper harmonic signature.

For the "Best Sampling Rate" question I think that both the Cyclic and the Tambreal ranges are involved.

The important distinction here is that nearly everyone can hear the Cyclic range "critically". By this I mean that most people can hear fairly subtle anomalies in this region. As a result, having a detailed representation of this frequency range is very important to just about everyone.

The Tambreal range is far more challenging to hear with the same "critical" ear. There are people in the world who can hear in this range with great precision. I'm just not one of them

IMO, for the vast majority of listeners the Cyclic range (16Hz-4200Hz) is way more important than the Tambreal range.

And it turns out, for example, that even at 44K1 Samples/Second all frequencies in the Cyclic range are captured with less than 0.5 dB error (off of peak) and that means what I claim we perceive as "music" is captured quite well at the 44K1 sample rate. Frequencies in the Tambreal range are not recorded with the same accuracy but the errors in this range are not nearly as important - as long as the fundamental frequencies are captured properly the music sounds pretty darn good.

So why would anybody want to sample at a higher rate? The answer I give these days is two-fold ...

The first consideration is "Transients, Cymbals and Synths" because these are the three "musical" things that have significant and potentially important energy in the Tambreal region. A fast-rising transient could have energy exceeding 20K. A spectrograph of a cymbal reveals lots of energy above 10K. The complex waveforms produced by modern synths often contain energy above 8K - particularly during filter "sweeps".

The second consideration is Transition Band Width - explaining this in detail is difficult and kind'a puts people to sleep. Briefly, the Transition Band Width is the difference between the highest frequency you want to hear (the "pass band") and the Nyquist-Shannon limit of the Sample Rate / 2. Check this out :

Code:
S/S    Transition Band          Cents   dB/Oct  Fltr
44k1   22050 / 20000 = 1.1025   168.9   568.27    48 
48k    24000 / 20000 = 1.2000   315.6   304.14    26
60K    30000 / 20000 = 1.5000   702.0   136.76    12
64k    32000 / 20000 = 1.6000   813.7   117.98    10
88k2   44100 / 20000 = 2.2050  1368.9    70.13     6
96k    48000 / 20000 = 2.4000  1515.6    63.34     6
S/S - Samples per Second
Transition Band - region between Nyquist-Shannon limit (SR/2) and highest frequency in the pass band
Cents - Transition Band width in 100ths of a semitone
dB/Oct - for aliasing to be -80dB
Fltr - Number of 12dB stages for -80dB anti-aliasing

Note that going from 44K1 to 48K almost doubles the size of the Transition Band Width. Going from 44K1 to 96K is nearly a 9X improvement. These improvements translate into less complex anti-aliasing filters as shown above. I should add that modern DACs use internal oversampling to reduce the complexity of the anti-aliasing filters and this seems to help a lot; at least on paper

peace y'all
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Old 01-25-2011, 12:34 PM   #12
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Originally Posted by semiquaver View Post
Dan Lavry (who makes some of the best DACs going) argues that converters do not perform as well @ 192k than at 96 so even for archival purposes you are probably better off @ 96 with current SOA converters.
Yes. He's the man to listen to and I believe it's fair to let him have the final say. This post explains all and there's a more in depth white paper on his site.

http://www.gearslutz.com/board/1234224-post72.html

To cut a long story short while using higher sample rates lets you use a mellower anti-aliasing filter, if you go too high the converter has less time to digitise accurately. He believes 88.2 - 96khz to be the best compromise of speed/accuracy.
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Old 01-25-2011, 12:45 PM   #13
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192KHz??? That is SOOO old hat!!!! 384KHz/48 bit fixed is where it's at now


On a more serious note, after being a long time proponent of recording at 96KHz, I am totally rethinking my stance on that. The gain I get in productivity due to the significantly lighter CPU load at 44.1KHz is a greater benefit to me than any plugin accuracy increase mixing at 96KHz.

The newest album I have just started recording will all be done at 44.1KHz. I had forgotten just how much more I can do with the extra CPU overhead available and I'm really loving rediscovering that!

Besides, a good listen to a really well recorded CD is a healthy reminder of just how good even 16 bit/44.1KHz actually can be.
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Old 01-25-2011, 01:16 PM   #14
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There is someone here in Germany (to some he is something like a "restoring pope") who restores old recordings afaik often from original master tapes of recordings for records to be released again an CD by some record companies. He actually uses 384 kHz at 64 bits. On a radio show he was asked what tips he could give to the average user, listening to his mp3s e.g. in a car. His answer, somewhat offended, went like "I never ever deal with such things!!!!" (Well, the exclamation marks are my interpretation).



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Old 01-25-2011, 03:54 PM   #15
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Originally Posted by Mr. Data View Post
There is someone here in Germany (to some he is something like a "restoring pope") who restores old recordings afaik often from original master tapes of recordings for records to be released again an CD by some record companies. He actually uses 384 kHz at 64 bits. On a radio show he was asked what tips he could give to the average user, listening to his mp3s e.g. in a car. His answer, somewhat offended, went like "I never ever deal with such things!!!!" (Well, the exclamation marks are my interpretation).



-Data
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Old 03-04-2011, 12:19 PM   #16
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Originally Posted by djz View Post
.......I can see the benefit of 96khz...but 192???
I found this after I had agonized about, and then actually bought a Delta 192 specifically because it had such a high sampling rate, and then I read just the introduction to this:

"Nyquist pointed out that the sampling rate needs only to exceed twice the signal bandwidth. What is the audio bandwidth? Research shows that musical instruments may produce energy above 20 KHz, but there is little sound energy at above 40KHz.

Most microphones do not pick up sound at much over 20KHz. Human hearing rarely exceeds 20KHz, and certainly does not reach 40KHz. The above suggests that 88.2 or 96KHz would be overkill.

In fact all the objections regarding audio sampling at 44.1KHz, (including the arguments relating to pre ringing of an FIR filter) are long gone by increasing sampling to about 60KHz.

Sampling at 192KHz produces larger files requiring more storage space and slowing down the transmission. Sampling at 192KHz produces a huge burden on the computational processing speed requirements. There is also a tradeoff between speed and accuracy. Conversion at 100MHz yield around 8 bits, conversion at 1MHz may yield near 16 bits and as we approach
50-60Hz we get near 24 bits.

Speed related inaccuracies are due to real circuit considerations,
such as charging capacitors, amplifier settling and more. Slowing down improves accuracy."

The link to the pdf:

http://www.lavryengineering.com/docu...ing_Theory.pdf

Courtesy of Dan Lavry

Sampling Theory For Digital Audio
By Dan Lavry, Lavry Engineering, Inc.

I now do everything at 48KHz; including brushing my teeth.
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