I notice that the Freq Analysts show frequencies up to 48 kHz when my tracks recorded at 96 kHz.
 
Should I low pass filter the final mix before exporting to 44.1. I think I don't have an EQ that allows me to filter out above 20 kHz although I'm not positive about that.
 
When I export to 44.1 from a 96 kHz project, is there a risk of artifacts from reduction in available frequencies?
 
I feel like it is a silly question, but mixing at the double and quad rates is still sort of new to me. I was mixing on an old laptop before Fall 2013, so I always recorded at 44.1 or 48 due to processing constraints. Now the ceiling has been raised sufficiently that I can record at 96 and mix a many track project bumping against ceilings for Hard Drive read/write or processor consumption.
 
I'm guessing that there is no extra step required, but I thought I'd ask. I know that a lot of convertors apply low pass filters above the audible range when converting.

Not sure if I understood this correctly... you have an audio artifact at 48KHz?? How are you viewing this (with what)? That is way out of audible range.

Yes, I can see the sound up to 48 kHz and yes way outside of the audible range.
 
I think the maximum frequency that can be reproduced is about 1/2 the sample rate minus a little because at the top of the range there needs to be a low-pass filter. This is why they use 44.1 rather than 40 kHz. The extra 2K of frequency inclusion is for the filter. That's why 48 is a little preferred because it puts the filter a littler farther outside of the audible range. The filter usually causes some minor artifacts that hopefully never fall within the audible range. Having more frequency outside of the audible range to work with makes it safer.
 
The bottom line is I'm not sure how it works when you go from 88.2 to 44.1 or 96 to 44.1. Is a filter applied? Should I apply the filter manually? Is there no need for a filter? Does the filter only matter at the time of digitizing?
 

 
http://stabilitynetwork.blob.core.windows.net/g-tunes/Screenshot_96.png
 
From Digital Audio Explained for the Audio Engineer by Nika Aldrich (Sweetwater Press)
P124

Aldrich
Aliasing
If the material to be sampled contains frequency content above the Nyquist Frequency then a type of distortion to the signal called aliasing occurs. If frequencies above the Nyquist Frequency are present then the samples will not be taken twice per highest frequency, but will be 'assumed' to have been taken twice per highest frequency and the signal will be reconstructed differently than it originally existed.
 
Since there is only one possible way to reconstruct a "legal" signal through the sampling points, and since we assume that the waveform is indeed a legal waveform and will be reconstructed as a legal waveform, any frequencies above the Nyquist Frequency will be re-created as frequencies below the Nyquist Frequency. The frequencies created are actually very predictable and mathematically determinable. The Nyquist Frequency acts as a sort of mirror in that any frequency content above it gets mirrored around it and creates frequency content below it.

Basically, I'm worried about aliasing resulting from exporting the tracks at 44.1 without proper preparation.
 
I'm pretty sure there can be aliasing if I try to listen to a 96 kHz recording on a stereo that can't handle it.
 

Aldrich p346
By recording higher frequencies than the ear can hear we create the possibility that the analog equipment in the signal path will create unnatural and inaccurate results. This very effect is often blamed for tests in which square waves at 15kHz are said to sound different than sine waves at 15kHz. We already know that the human ear cannot hear the difference between a square wave and a sine wave, each with a fundamental of 15kHz, as the first overtone of the square wave is at 45kHz, well above the hearing range. If a difference can be heard it can often be identified to be the creation of harmonic material within the hearing range because of nonlinearity and distortion in the playback equipment upon attempting to recreate high frequency wave forms. (This, and the fact that a square wave and a sine wave of equal amplitude have different amplitudes of the 15kHz fundamental, resulting in the square wave version sounding louder than the sine wave version by a few decibels.) Since any analog component is non-linear, recording material that should have no effect on audibility only provides the possibility that distortion may be added within the audible range, thereby affecting the accuracy of the playback of the recorded material.

 
Aldrich is making the case that recording at double and quad sample rates is sometimes detrimental for reasons beyond disk space and processor consumption.
 
With respect to Aldrich, I have to feel that if I record at 96 and export for playback at 44.1, as long as the export is done in such a way as to not introduce Aliasing, then the problem should not be evidenced. In other words, the general listeners to the music should not have any problems with Aliasing if you export to 44.1 before distributing.
 
It must be that there is no need to filter out high frequency data when exporting 96 sample rate Waves to 44.1, but I wanted to ask the question.
 

Sorry for the delay, I saw that post on my cell and wasn't sure if I was reading it correctly. I have no answer to this, and have never seen an audio spectrum analyzer go that high before. The best way to find out is to actually op test it and listen to the results in a good environment.
 
My gut reaction is to low pass that scenario (only because I can see it), but I am also very curious if anything audible comes out in the mix-down from not low-passing it.

Whenever you reduce sampling rate you should be dithering

Sampling audio at 96khz is not the same as producing an audio wave of a 96khz frequency.

I don't know about dithering for sample rate reduction. I believe there is no artifacts due to truncation when changing the sample rate. I could be wrong though. 

So I emailed Nika Aldrich who wrote the book I quoted, and he answered with this...
 

Nika
The proper way of doing sample rate conversion involves filtering the material below the new Nyquist frequency prior to the sample rate conversion.  This would eliminate any aliasing due to the conversion. Assuming you used a good sample rate converter, there should be no aliasing in the result.

 
So, I guess this is really a question for the bakers... Is material above the Nyquist frequency being removed prior to the sample rate conversion? Is Nika Aldrich correct that this material needs to be filtered before doing the conversion? And if so, I don't have any filters that filter above 20kHz so how would I go about it?
 
Changing the sample rate from 96 to 44.1 changes the Nyquist frequency from 48 to 22.5. Does this sound right to you guys? Am I correctly understanding this?
 
I think this is just handled for us by Cakewalk, but I'm checking to be sure.

mudgel
Whenever you reduce sampling rate you should be dithering

I agree, Mudgel. Nika Aldrich who wrote the book I quoted also agrees with you. I'm also aware that a lot of people in forum do not agree with dithering for Sample Rate conversion but do for bit depth conversions. I don't know the science behind the argument against using it. Nika Aldrich explains his reasons FOR using dither on sample rate conversions in his book Digital Audio Explained for the Audio Engineer.
 

mudgel
Sampling audio at 96khz is not the same as producing an audio wave of a 96khz frequency.

 
I agree with this too. I believe the Nyquist frequency is 1/2 the sample rate so for a sample rate of 96, the maximum frequency is 48 and in practice a bit lower due to the requirements to filter below the Nyquist Frequency. This filtering that I'm referring to should happen within your interface before the original digitization.