Thread Rating:
  • 0 Vote(s) - 0 Average
  • 1
  • 2
  • 3
  • 4
  • 5
Low sample rates and aliasing
#1
I jumped on the 96khz bandwagon a long time ago for a variety of reasons commonly cited by those in the pro audio community. One of those reasons was the absence of aliasing in the audio band when using higher sample rates. Recently the topic came up again and the scientist in me wondered how much of a problem aliasing really is at 44.1kHz, particularly when it comes to Mixbus's built in tape saturation.

Aliasing is commonly demonstrated by introducing signals of higher frequency than nyquist, either directly or by a high frequency below nyquist run into a clipper or saturator. As the frequency is increased, you can watch the alias component "walk" across to the left on a spectrum analyzer. The problem with this sort of demonstration is that the level of the original high frequency signal is much higher than what would be typically encountered in real use (anything that is not a guitar amp sim).

For example, if you look at the spectrum of any reasonably balanced mix, the signal level drops as frequency is increased. Running this mix through the tape saturation in mixbus, its the lower frequency energy acting on the saturation stage. Without the low frequency content, there is not enough energy in the high end to even get an indication on the saturation meter.

An example:

For this experiment I chose to use "Peg" by Steely Dan. Partly because of the clarity of the track and it had been a long time since I've listened to it. I loaded this track into Mixbus and routed the output to a Mixbus only. I left the tape saturation at its default setting of -10. I adjusted the track fader so that the saturation meter averaged a"straight up" indication. I then loaded a spectrum analyzer on the output of the mixbus.

I created another track and loaded a sine generator set to a high frequency (<11kHz) that was certain to generate an alias component when clipped by the saturator. This track was fed to the same mixbus as my reference music track. The level of this track was adjusted that it just poked into the spectrum of the music track while playing. The test signal was now at a real world level.

Muting the music track, there was no alias component. Theoretically its there, but its below the noise floor. increasing the tape saturation from -10 to +10 caused the alias component to appear, but down 72dB from -18dB(0VU)! -72VU can be audible in a very quiet room but given the amount of saturation required, you'll never hear the alias in the context of music.

Heres why: Say you had a mix playing and the saturation meter is straight up as in the example above. Increasing the tape saturation by 20dB would then drive the signal into oblivion, something you would only do for some sort of effect. The alias that exists at -72VU wouldn't even be discernable as the original signal would be barely recognizable anyway!

In real life, audible saturation will occur at low frequencies long before the high frequencies. Unless you like to crank up the saturation on your high-hat tracks, then you would probably hear aliasing.

Did the same test with the limited set of plugins I like to work with. Some had no problem at all, the worst had an alias down -60dB when pushed with HQ set to off.

For me, the scale has tipped back the other way in favor of working at 44.1kHz (less disk i/o and CPU).
Reply
#2
@superb:

This is a very insightful post. Thank you very much for taking time to make these tests and post your comments.

We work very hard to make Mixbus perform well in the "real world", and it's nice to know that some people "get it".

Thanks again!!!

-Ben
Reply
#3
(01-10-2018, 12:07 PM)superb Wrote: For me, the scale has tipped back the other way in favor of working at 44.1kHz (less disk i/o and CPU).

Thanks for doing that empirical work. Indeed many (most?) converters are oversampling with lower depth anyway. The bit stream is then converted into the target rate/depth. Sony's 1bit technology as the ultimate quintessence.
Here's a nice read, bit old but the first chapters teach every aspect of AD/DA conversion very well: https://books.google.com.au/books?id=yN3...&q&f=false

MMM
Reply
#4
I’m confused so is 96K good or no?


Sent from my iPhone using Tapatalk
Reply
#5
(01-10-2018, 09:15 PM)mrskytown11 Wrote: I’m confused so is 96K good or no?

96kHz is good. 44.1kHz is good enough. So no need for 96k and the extra space and I/O and DSP power 96k needs.

Spend your money rather on the analogue ends - in-chain and room treatment/monitoring.
https://sonicscoop.com/2016/02/19/the-sc...n-it-isnt/

MMM
Reply
#6
Great post - very insightful!

Regards,

Simon
Reply
#7
(01-10-2018, 05:18 PM)Ben@Harrison Wrote: @superb:

This is a very insightful post... We work very hard to make Mixbus perform well in the "real world"...

Thank you for making Mixbus so awesome! Having to worry about all of this stuff can take the fun out of things, it's good to know that nothing was overlooked. Just use it and know it will sound as good as it can.

(01-10-2018, 07:00 PM)madmaxmiller Wrote: ...Indeed many (most?) converters are oversampling with lower depth anyway. The bit stream is then converted into the target rate/depth. Sony's 1bit technology as the ultimate quintessence.
Here's a nice read, bit old but the first chapters teach every aspect of AD/DA conversion very well: https://books.google.com.au/books?id=yN3...&q&f=false

MMM

Yes, I've heard that many modern converters sample high and then downsample to the desired rate. Grabbed the book, $4.19 for the Kindle edition. I'll be sure to give it a read. Thanks for sharing!

(01-10-2018, 09:15 PM)mrskytown11 Wrote: I’m confused so is 96K good or no?

As MMM said, 96K is good, 44.1K is good enough. My intent in posting this was to show that maybe 44.1K isn't as bad as it is made out to be.

Thank you everyone for your kind words! If I posted my findings on gearslutz, I'm sure it would have turned into a major flame war. That's why I like the community here. Big Grin
Reply
#8
Hello,

Mastering and vinyl cutting engineer for many many years. I totally agree to what you can read here.

Some years ago - as part of a semi-scientific diploma work - we made the test with electronic dance music, tone generators only in the box with digital synthesizers. Production was recorded mixed and mastered in 44.1kHz - 24bit and 88.2kHz - 24bit. All identical settings on all gear - except for the sample rate. And then cut to vinyl and recorded on a studer 1/4inch tape machine.

Blind test result was: higher sample rate sounded more open, all the way through all media. Even when the higher sample rate master was resampled to 44.1kHz this difference was obvious still. No doubt. It was not possible to recreate that 'openness' in the session with lower sample rate by EQing the high end differently. From the pre-master on the difference between sample rates had a quality similar to the difference between Crome-Tape and Normal-Tape or maybe between Metal-Tape and Crome-Tape (for those who still know the difference in consumer Tapes...).

It was then that I decided: Higher rates sound better. But: I was wrong. The openness was nothing else than distortion - as mentioned in the linked article. IMD introduced into production by nonlinear processing in the digital domain. But I did not know then. I only heard some sparkle in the high frequency range that I liked. And I thought: WOW - this really MAKES a difference. I did not know how the material should sound 'originally' -

So what to do? With all this learned I was almost certain that Mixbus would have a built in Ultrasonic filter to avoid IMD distortion on higher sample rates in the dynamic section of tracks, busses and the master bus. Mhh. Testing this showed: No - there is no such filter.

I found one here - and to my opinion it works:
https://vladgsound.wordpress.com/2014/12...a-version/

For the tests I used audio material linked here in this long article:
https://people.xiph.org/~xiphmont/demo/n...ml#toc_1ch

- to download it you need to scroll down to here:
''If you're curious about the performance of your own system, the following samples contain a 30kHz and a 33kHz tone in a 24/96 WAV file, a longer version in a FLAC, some tri-tone warbles, and a normal song clip shifted up by 24kHz so that it's entirely in the ultrasonic range from 24kHz to 46kHz''

All the best wishes, Helmut.
Reply
#9
Hello Helmut and welcome to the forum!

Thank you for sharing your experience and the links. I've visted them before (love xiph.org!), but I'm glad you posted them here because they are very relavent to this discussion.

Some form of intermodulation distortion is always created when when a signal is distorted in any way, whether analog and digital. With the greater bandwidth offered at 96kHz, there are greater chances of this because ultrasonic frequencies can be recorded and IMD will show up when the signal is run through any non-linear process.

The last time I tested the plugins I use for ultrasonics (when I was still working at 96kHz), they all seemed to have filtered out anything above the audible band internally, if I remember correctly. I wasn't as detailed in documenting that experiment as I was with the above experiment. But I'm sure there are some plugins and soft-synths that really should have an ultrasonic filter placed after them when used at 96kHz.

Generally, people working at 96kHz do not want the ultrasonics touched because they believe that they add something to the sound. Having ultrasonic filters built in might make Mixbus unuseable for them, even though as your experience has shown, there would be less IMD.

96kHz allows the use of a gentler low pass filter and reduces the chances of aliasing in the audible band but with greater risk of IMD occuring. Where have all the 'benefits' of recording at 96kHZ gone?
Reply
#10
(01-15-2018, 09:46 AM)superb Wrote: Where have all the 'benefits' of recording at 96kHZ gone?

They disappeared the day converters started to oversample. No over-steep analogue anti aliasing filters, the actual filtering is in the digital domain.
One could argue that we actually convert with high sample rates by default, so 96kHz must be better and 192kHz even more. Only half true, as the oversampling will happen with reduced bit depth, ultimately at 2.8MHz/1bit, and later converted into the desired rate/depth combination. But then you have all the benefits of delta/sigma conversion, digital filtering and dithering (recording dithering not to be confused with your export dithering). No point to run it at higher rates than 44.1kHz as it doesn't win you any quality benefits.

So, the day we got modern converters, 96kHz benefits died and if a recording sounds different in different sample rates on one and the same converter - in or out - put it on ebay if you are ruthless enough or bin it if you are not.

MMM

P.S. there's one benefit of higher sample rates: at the same buffer size you have shorter latencies... but that's a timing issue when you monitor through the DAW, not a quality game.

(01-15-2018, 09:46 AM)superb Wrote: Generally, people working at 96kHz do not want the ultrasonics touched because they believe that they add something to the sound.

That must be the people who also can see infrared and ultraviolet light Big Grin
MMM
Reply


Forum Jump:


Users browsing this thread: 1 Guest(s)