QuadCore

John T
There is a fairly cogent argument for using higher than 44.1 for recording, but it's got nothing to do with capturing higher frequencies. It's that it's easier to build the cutoff filter when there's a bit more bandwidth to play with. Not I just say easier, which you can read as "cheaper". There are many filters that work perfectly well in 44.1.
 

 
Yes.  Cheaper converters may sound better at higher sample rates because the requirements of the anti-aliasing filters are not as critical at higher sample rates for the reason you point out.  A high quality converter may sound just as good at 44.1 because the anti-aliasing filter is doing it's job well, even when the cut off frequency is very close to the highest frequencies we want to capture.  So i opted for a good converters (Lavry blue series).

 
Don't confuse purely analog antialiasing filters with oversampled converters. Oversampled converters use a gentle analog filter combined with a very steep digital one to combine to produce the required results. Unlike analog filters, you can easily make for extremely steep digital filters within certain mathematical limits and trade offs involving processing requirements, latency, ringing, etc., along with the filter shape. In the analog world you have all that same stuff along with physics to contend with and that makes it real hard. Which is a big reason for oversampling - you can get excellent, consistent and reliable performance very cheaply. "Excellent" is of course a dangerous word in audio circles where many people are either too young, or too old and forgetful to remember the days when pretty poor performance was commonplace, and thus they have completely lost all perspective.
 
As I noted earlier, in addition to ultrasonics question, the only question with commonly used oversampling converters is whether you can hear the filter artifacts at all. If you can't hear them, all this talk of gentle filters being better is irrelevant.
 

 
I am intrigued though by the idea that higher sample rate oversampling in plugins makes for better sound across the spectrum.  Do X3 Pro Channel effects oversample?

My impression is that Sonar's bakers are more than smart enough to oversample in the cases where they should.
 
For the most part, I would think that anyone writing DSP today with any degree of competence will know where using a higher rate (or greater bit depth) is beneficial and take appropriate action. Several years ago, it's more likely that people might have left it out to save CPU, but there isn't much need for that now.

brconflict
Something else to consider with higher sampling rates, I didn't see in this thread (or overlooked): Accurate clocking. The higher the sample rates, the more accurate your clocking needs to be, since jitter and component issues may become more of a problem. This is why you see these oven-baked atomic clocks offered up from Antelope, for example.
 
If you're sampling at 192Khz with a low-quality clock, you may stand to benefit from halving the sampling rate and improving transient materials, for example. It could be a noticeable improvement.

I thought the conventional wisdom was the opposite of this. Back in the day overclocking was a strategy to mask clocking inaccuracy. As clocks became more stable and accurate, overclocking was deprecated in a lot of hi end audio playback gear, to the point where people were uninstalling the overclocking pieces/parts from their CD players and installing new improved replacement clocks as upgrades became popular. 
 
Is oversampling similar? Will oversampling make the results of clock that drifts by some small percentage seem like smaller absolute timing errors or is the effect of a clock spread evenly across the duration of the recording?
 
 

mike_mccue

brconflict
Something else to consider with higher sampling rates, I didn't see in this thread (or overlooked): Accurate clocking. The higher the sample rates, the more accurate your clocking needs to be, since jitter and component issues may become more of a problem. This is why you see these oven-baked atomic clocks offered up from Antelope, for example.
 
If you're sampling at 192Khz with a low-quality clock, you may stand to benefit from halving the sampling rate and improving transient materials, for example. It could be a noticeable improvement.

I thought the conventional wisdom was the opposite of this. Back in the day overclocking was a strategy to mask clocking inaccuracy. As clocks became more stable and accurate, overclocking was deprecated in a lot of hi end audio playback gear, to the point where people were uninstalling the overclocking pieces/parts from their CD players and installing new improved replacement clocks as upgrades became popular. 
 
Is oversampling similar? Will oversampling make the results of clock that drifts by some small percentage seem like smaller absolute timing errors or is the effect of a clock spread evenly across the duration of the recording?

 
Conventional wisdom today is that the clocking system in today's devices are better than using a master clock. That said, if you have a studio full of digital gear, you'll need a master clock to keep it all in sync anyway. It's just that the jitter response from today's stuff is pretty good, and an external clock will induce bit of jitter which is the nature of PLL circuits. In the past, the jitter from syncing to a master clock would be lesser than a device's internal clock.
 

I don't even record at 96 kHz any more !... I used to but only because my brain was playing the numbers game.. I mean 96 has to be better than 48 because its twice the size right ?
 
But seriously I decided that CD quality is good enough for me.... I'm a 56 year old bass player that records the ocassional song in a bedroom and so the technically inferior CD specification is more than adequate for my purposes

IfItMovesFunkIt
 
But seriously I decided that CD quality is good enough for me.... I'm a 56 year old bass player that records the ocassional song in a bedroom and so the technically inferior CD specification is more than adequate for my purposes

 
Ditto - I'm a 68 year old musician (bass/guitar/wind controller) and I record in an untreated room.  I tried higher kHz but I didn't notice any major difference, possibly do to my age even though I have very good hearing. Thus CD quality is fine with me.
 
Also no one has ever said this would sound better if it was recorded at a higher kHz.
 
So my question would be why record at a higher kHz if you can't tell the difference?

this thread is a great example of why i come to these forums - great discussion, lots of intelligent input, polite disagreement, always something to learn :-)

There's an expression in french that say:"there's no smoke without a fire".  So, 3 years ago I changed all my recording system.  Before, I was at 16/44.1   With my new gear and after alot of reading and questionning, I decided to go at 24/88.2  I don't know if the sample rate by itself made a difference.  I just feel that the hole thing is better now and I tried to put any small chance in my side.
I took many projet that I had recorded with my old gear and then I could hear the difference with my new stuff.  But, I still listening what I had record with my old gear in my new converters, monitors and another version of SONAR.
Very interesting conversation here and it happened before but, I can't see the end of that...

MarioD
 
So my question would be why record at a higher kHz if you can't tell the difference?

I think the bottom line for this is not the capture or final medium but the processing that is done in the middle. Some VSTs can/do take advantage of higher sampling rates which can be audible. Even with that being fact, the "how audible" continues to fuel debates. The end result really boils down to personal preference.

mettelus

MarioD
So my question would be why record at a higher kHz if you can't tell the difference?

I think the bottom line for this is not the capture or final medium but the processing that is done in the middle...

Exactly.

mike_mccue

brconflict
Something else to consider with higher sampling rates, I didn't see in this thread (or overlooked): Accurate clocking. The higher the sample rates, the more accurate your clocking needs to be, since jitter and component issues may become more of a problem. This is why you see these oven-baked atomic clocks offered up from Antelope, for example.
 
If you're sampling at 192Khz with a low-quality clock, you may stand to benefit from halving the sampling rate and improving transient materials, for example. It could be a noticeable improvement.

I thought the conventional wisdom was the opposite of this. Back in the day overclocking was a strategy to mask clocking inaccuracy. As clocks became more stable and accurate, overclocking was deprecated in a lot of hi end audio playback gear, to the point where people were uninstalling the overclocking pieces/parts from their CD players and installing new improved replacement clocks as upgrades became popular. 
 
Is oversampling similar? Will oversampling make the results of clock that drifts by some small percentage seem like smaller absolute timing errors or is the effect of a clock spread evenly across the duration of the recording?
 
 

Oversampling and over-clocking are very different, but I think I know what you're saying. We've over-sampled in the past to better remove sampling errors, caused mainly by read-errors on the source audio. For example, if you sampled 32-times a single input to create one resulting sample, there's a very high chance of accuracy vs. 2 or 3 times over-sampling.
 
However, when you're transmitting audio from one place to another, at some point the data is placed on the wire, which either is a buffered process, such as Ethernet, or USB, for example. In that process, there's two major ways this can be done. 1) Bit-by-bit, best-effort, which can lose bits of data or, 2) With error-checking, which causes jitter and delay because lost bits may be re-transmitted or corrected which adds delay. When placing bits on the wire, the higher clocking rates are more susceptible to noise and errors.
 
Jitter in itself might be the same whether the sample rate is low or high. Think of trying to read poker cards if your hands are jittery. If the cards are tiny (high sample rate), reading them correctly might be tough. If the cards are very large in size, you will certainly have an easier time reading them (low sample rate). Of course, if your eyes were trained to jitter right along with your hands, you could read the cards better. This is just a crude example, but the point is, the smaller the samples, the more difficult they are to read accurately in some scenarios.
 
Technology advances improve this ability in audio, but it's still less of a challenge for cheaper A/D equipment to sample and transmit the audio at slower speeds.