i don't know why someone would mix a session recorded at 44.1 /48 @ 96 kHz. what's the point?

What I took from it is that @96k any plug-ins used create cleaner harmonics, @44.1k plug-ins such as compressors and equalisers create aliasing foldback below 44.1k that add clutter to the audio spectrum. The more plug-ins used add to the harmonics which add to the aliasing foldback.

That makes sense. So if we run our stuff 24/96 it moves much of the aliasing (NSF or other) to beyond the point of human hearing. I wonder if any of the 44.1 problems would go away if it was 24/48? That's what i usually run in...I can do 96 or up to 192.

I'd be curious what would happen if you ran it at 192?

Also, are many people using 32 bit float?

I don't quite get the phase inversion part of it. I understand phase, I understand how passives affect phase, how crossover slopes affect phase, how eq's can affect phase. I've worked in environments where in order to get things to sound right, you need to flip the phase of a driver. I perhaps don't understand it as much as most of you, but I have a pretty good handle on it. I had a fully adjustable phase switch for a long time, so I got to play around with phase a lot.

What I don't get is inverting the phase of the track at 44.1 and 96? Or can anyone explain how that phase inversion track is working? I understand what he is proving with the track...I don't understand what he did to make that track?

Here's a better way to ask it, here is a quote from the paragraph above that track...

"Below is a stream of the phase inverted difference between the 96kHz session bounce and the 44.1kHz session bounce."

Can someone explain what he is talking about?

If you have 2 signals and they are exactly the same, when you invert the phase of one it cancels out so in effect you should have silence.

If the signals are not exactly the same and you invert the phase on one, you hear the difference between the two.

What has been done is one is at 44.1 and the other at 96k so what you are hearing is the difference between the two signals when one is inverted, or what is left behind after what is the same with both signals has been cancelled out.

What you are effectively hearing is the difference between the two files.

Excerpt from the last paragraph...

"In the above phase inversion test you can clearly hear the aliasing on the open hi-hat. Second, there is a distinct amount of high frequency detail that is prevalent in the 96kHz bounce that is not captured in the same way in the 44.1kHz bounce. This high frequency detail can be heard in what remains of the vocal in the phase inversion test above. Third, higher sample rates allow you to control transient detail with more precision and less distortion than at lower sample rates. This is why we see oversampling features built into many popular digital mastering limiters...."

Sean G, post: 441212, member: 49362 wrote: "In the above phase inversion test you can clearly hear the aliasing on the open hi-hat. Second, there is a distinct amount of high frequency detail that is prevalent in the 96kHz bounce that is not captured in the same way in the 44.1kHz bounce. This high frequency detail can be heard in what remains of the vocal in the phase inversion test above. Third, higher sample rates allow you to control transient detail with more precision and less distortion than at lower sample rates. This is why we see oversampling features built into many popular digital mastering limiters...."

I get that part...your first post pretty much answered my question.

Im gonna have to mess with it myself to answer my other questions. Thank you Sean G.

So do most people here use a 96khz sample rate?

I do track, mix and master in 96khz, sounds better to me ;)

Everything sounds different than everything else.. regardless of what it is.. No 2 Mics. Pres. Convertors,.. Speakers.. Rooms. Performances.. Are exactly the same on a measurement scale.. If it sounds good.. it is.. if it doesn't it probably has alot more to do with than 96k vs 44.1

When I render using real time vs regular in Cubase using the same sample rate they don't null.. to me the real time sounds better..

Also.. all this time I thought the industry was trying to add analog distortion/saturation and character to mixes.. I wonder what the same tone generator tests with analog gear would be like.
Would they have the same less harmonic distortion as 96k or more? less than 44.1 or more?.

I wonder.. If there is a difference.. would the plug in manufacturers try and get the exact modeling to 44.1 ..48..96k.. or 192 because if they are different.. then bench marking would be different as well.. and the correct exact model would have to be at one of those sample rates..

I just read that flow chart.. that's a bizzarre way of doing things.. record in Ableton live then upsample to Protools at 96k .. Do a mix at 96k.. Then copy the 96k mix and then downsample it to 32 bit 44.1 ... render both and the compare them...

Why not do a mix at 96 or 44.1 and then copy the mixer settings to the other using the original source files for both..

Chris Perra, post: 441230, member: 48232 wrote: I just read that flow chart.. that's a bizzarre way of doing things.. record in Ableton live then upsample to Protools at 96k .. Do a mix at 96k.. Then copy the 96k mix and then downsample it to 32 bit 44.1 ... render both and the compare them...

Why not do a mix at 96 or 44.1 and then copy the mixer settings to the other using the original source files for both..

I thought the process in the flow chart looked at little weird too

pcrecord, post: 441223, member: 46460 wrote: I do track, mix and master in 96khz, sounds better to me ;)

Merci. I've always done 24/48 bc of computer restrictions, but I don't have those now. 96 here I come.

Chris Perra, post: 441229, member: 48232 wrote: I wonder.. If there is a difference.. would the plug in manufacturers try and get the exact modeling to 44.1 ..48..96k.. or 192 because if they are different.. then bench marking would be different as well.. and the correct exact model would have to be at one of those sample rates..

That is actually an interesting thought. I wonder as well now...

Chris Perra, post: 441230, member: 48232 wrote: I just read that flow chart.. that's a bizzarre way of doing things.. record in Ableton live then upsample to Protools at 96k .. Do a mix at 96k.. Then copy the 96k mix and then downsample it to 32 bit 44.1 ... render both and the compare them...

Why not do a mix at 96 or 44.1 and then copy the mixer settings to the other using the original source files for both..

I did not understand that part at all either. I understand what he is saying...I don't understand why anyone would do what he shows.

Please excuse this if it's a stupid question...I never really thought about it. But according to that chart, he is changing the bit depth, and sample rate, often up! 24 to 32, 44.1 to 96.

Once the track is recorded, can you actually change the track? Not on paper, but, actually alter the track? Once it's been recorded in 16/44.1, you can change it to 24/96 to the and it will sound as if you recorded it in 24/96?

I thought past the recording stage, you could change the bit depth and sample rate down...but not up. Can someone tell me which is correct?

I just took some drums at 44.1 then up sampled them to 96k.. Then rendered both at 44.1 and 96k with no faders or plug ins to 16 bit 44.1. They dont phase cancel.. It seems it has nothing to do with plug ins.

I wonder what others encounter

Brother Junk, post: 441238, member: 49944 wrote: I thought past the recording stage, you could change the bit depth and sample rate down...but not up. Can someone tell me which is correct?

You can upsample and increase the word length, but you don't gain anything directly.

In Sony Vegas you don't necessarily have to upsample your files, you can just change the project properties to whatever rate you like and it will resample on the fly. I suppose there's a CPU penalty for that, but it does it.

Hey, wait, I though resampling was supposed to be a bad thing. Now all of a sudden it's a good thing?

By up sampling I loaded the 44.1 project into a new 96k project.. Faders and all

The Ryan Schwabe article that Sean linked in the first post of this thread is interesting, but the author falls down on much of what he attempts to find fault with.

First example: take the diagram that shows the spectrum of a waveform whose fundamental is not stated, but presumably at about 7KHz. Quote: "At 44.1kHz sampling rate the 3rd harmonic is below the anti-aliasing cutoff filter, but above the Nyquist-Shannon frequency". If you look at what he has drawn as an anti-aliaising filter, it's only about 5dB down at the Nyquist frequency, so it's simply the wrong frequency of filter to be using. Anti-aliasing filters designed for the CD audio frequency of 44.1KHz are 3dB down at 20KHz and somewhere around 70dB down at the Nyquist. His incorrect example leads to false conclusions.

Second example: the assumption that filters that perform anti-aliaising in the analogue domain ahead of sampling work at the top end of the audio band. In the last 15 - 20 years, the only types of ADC and DAC used for sampling and reconstruction of audio sample well outside the audio frequency band, typically at many MHz. At this rate, it's possible to reduce the wordlength from 16 down to 4 or even 1 bit, and then recreate the longer wordlengths during the decimation (down-sampling) process in the digital domain. The point is that, when sampling at these MHz rates, the analogue anti-aliaising filters can be very simple designs that roll-off gently in the tens or even hundreds of KHz, causing minimal phase irregularities in the audio band. The anti-aliaising filters for containing the information in the band 0 - 20KHz are implemented digitally as part of the decimation process.

Ryan Schwabe is right to raise many of the issues that he does in his article, but he goes about justifying his conclusions in the wrong way.

The further subject of whether DAW plug-ins can generate aliaising products is down to the coders of the individual plug-ins and the structure of the DAWs they are running in. Certainly, a plug-in that intentionally creates distortion (generating harmonics not present in the original waveform, or increasing the amplitude of those that are) lays itself open to problems attributable to the coder's lack of full understanding of the physics involved. A properly designed and implemented algorithm will suppress harmonic components that would otherwise fall above the Nyquist frequency. Note that algorithms of this type do not automatically know what clock rate they are running at; they can only deal in terms of fractions of the sampling frequency, whatever that happens to be at the time of use. If, for example, a DAW allows a plug-in to run at 96KHz in a 44.1KHz session, then it's up to the DAW and not the plug-in to perform SRC at the input and output of the plug-in, together with any anti-aliaising band-limiting associated with conversion to a lower rate. It's a tricky subject, and I've had suspicions for a long time that many DAWs do not implement this correctly, let alone any problems due to the use of digital SRC. I've yet to be persuaded that the benefits of running plug-ins at sampling rates different to that of the session outweigh the problems that the necessary implied SRC can cause.

Boswell, post: 441244, member: 29034 wrote: If, for example, a DAW allows a plug-in to run at 96KHz in a 44.1KHz session, then it's up to the DAW and not the plug-in to perform SRC at the input and output of the plug-in, together with any anti-aliaising band-limiting associated with conversion to a lower rate. It's a tricky subject, and I've had suspicions for a long time that many DAWs do not implement this correctly, let alone any problems due to the use of digital SRC. I've yet to be persuaded that the benefits of running plug-ins at sampling rates different to that of the session outweigh the problems that the necessary implied SRC can cause.

Thanks for sharing your understanding of this Bos, the issue of SRC and the downside of this is something I came away scratching my head over.

With plug-ins that oversample to perform their given function and the DAW performing the SRC at the input and output stage of the plug-in, say in the case where the session is at 44.1kHz, if the DAW is not implementing this correctly is it safe to say that some aliasing foldback maybe added to the audio spectrum?

Sean G, post: 441245, member: 49362 wrote: With plug-ins that oversample to perform their given function and the DAW performing the SRC at the input and output stage of the plug-in, say in the case where the session is at 44.1kHz, if the DAW is not implementing this correctly is it safe to say that some aliasing foldback maybe added to the audio spectrum?

Yes, except in the case where a plug-in is specifically written to run at a sampling higher rate than the DAW session. In that case, the responsibility is with the plug-in to perform the required SRC and anti-aliaising at its output. These types should be safe to use with any DAW and not cause foldback, but they may separately still suffer from digital SRC effects (my pet gripe).

Interesting thread, thanks Sean.

44.1/24, 48k/24 sounds very good to me. I do hear an extra sweet smoothness at higher SR but the extra sweetness has never been a game changer; to always be set on example: 96k.
My thinking... when using top quality converters, a win win for both sound quality and DAW performance can be accomplished from lower SR . Budget converters don't seem to have the same results.

Samplitude, PCIe interfacing and a well built PC mutitracks very smooth @44.1 for me, which is why I choose this platform as my "go to" DAW as well as why good converters are worth the investment.

My modo is; everytime you process audio, you degrade it. Some degradation are a good thing, like when going for analog external gear, but ITB it's bad. I don't thrust computers to do the job perfectly each time. (Yes I'm an IT ;))
If you record at 44, mix at 44. I could even say master at 44 as well because most high end plugins have oversampling anyway so the excuse that they will sound better drops right there... I haven't always thought about it this way, but this is my actual opinion ;)

pcrecord, post: 441256, member: 46460 wrote: everytime you process audio, you degrade it.

+1(y)

pcrecord, post: 441256, member: 46460 wrote: My modo is; everytime you process audio, you degrade it. Some degradation are a good thing, like when going for analog external gear, but ITB it's bad. I don't thrust computers to do the job perfectly each time. (Yes I'm an IT ;))
If you record at 44, mix at 44. I could even say master at 44 as well because most high end plugins have oversampling anyway so the excuse that they will sound better drops right there... I haven't always thought about it this way, but this is my actual opinion ;)

I believe exactly what Marco just said. Which is why I never SRC (bounce down) ITB unless it is a last resort. I record and capture at my destination SR. Which is why I choose two DAW's to get that done without bouncing. There is never a need to bounce down. One DAW is your archive and the other is the commercial product.

My conclusion to everything Pro Audio is: Use two DAW's to work with and you will never look back.

Chris, I am aware that using the 2 DAW approach avoids SRC issues, when it comes to mixing at a higher sample rate then capturing at say 44.1 is there a beneficial audible difference from your experience?

Sean G, post: 441261, member: 49362 wrote: is there a beneficial audible difference from your experience?

The difference is sonically tighter, smoother. Once you have 2 DAW's all connected like I experience, the ability to flow between tracking, mixing and mastering is a game changer.

I do not need analog mixing or mastering gear like I need 2 DAWs and a 3 way monitoring system.

Late to the party here. I agree that minimizing sample rate conversions is good practice. I think in general (opinion based) that upsaming type things work better in visual things like pictures and video than they do audio. To me up sampling is an SRC and isn't necessary.

I auditoned 44.1 vs 192 on my old m audio interface and couldn't hear a differnce at home thru a set of mackies, doing crappy original demo stuff. So I think Chris was correct about the converter quality.

Also I've read things where even solid comverters sounded better at 96 than their max 192. So it's likely just opinion and validation for the users own reasoning (cpu performance, lack of quality room/monitoring?) I think that there's probably some truth to it. The gear was probably designed with a particular use in mind. Lavry comverters don't do 192 and there's a long explanation as to why. I've heard good claims against some of Lavrys reasoning as well.

Recording at higher sample rates does offer lower latencies, which is a plus in many cases.

My thought is record/mix at the highest sample rate possible. For archival reasons, and for any sonic benefits (assuming there are some), and to account for the redicously large number of listening formats. Otherwise record at the destination sample rate. Either way I don't SRC until the last possible stage of the production.

It took a surprising amount of effort to put together a rig that was 192 thru and thru. I'm dying to see how the newer computers handle it. From interfaces to channel counts to adat to plugins it was tiresome to get things that's met the needs at 192.

I have eyes toward the future and 96 has been standard for at least 5-7 years.

That said all of my work at the studios the last 6 or so years was 44.1/24 and all was just fine. No grainy sounds or overbright or absurd 3-5k ailments.

I've said it a million times but eventually you'll get to the point where you can record a song in one huge sample i.e. No peices/linear and many of the ailments will go away. Just a theory.

kmetal, post: 441268, member: 37533 wrote: Lavry comverters don't do 192 and there's a long explanation as to why. I've heard good claims against some of Lavrys reasoning as well.

Dan Lavry in one of his white papers on sample rates believes that the optimal sample rate is 88.2 kHz

Sean G, post: 441269, member: 49362 wrote: Dan Lavry in one of his white papers on sample rates believes that the optimal sample rate is 88.2 kHz

Was that in the consideration of end going to 44 for CD ?
I Wonder when he wrote that.. technology runs fast these days and boundaries get broken everyday ;)

pcrecord, post: 441277, member: 46460 wrote: Was that in the consideration of end going to 44 for CD ?
I Wonder when he wrote that.. technology runs fast these days and boundaries get broken everyday ;)

It was referenced here a few times I think in the past. I may have even linked it myself.

I know he has written one or two....this is the one I was referring to http://www.lavryengineering.com/pdfs/lavry-white-paper-the_optimal_sample_rate_for_quality_audio.pdf

I think it was written in 2012...?

pcrecord, post: 441277, member: 46460 wrote: Was that in the consideration of end going to 44 for CD ?

I was thinking that too.

Also things get even more complicated when vsti are involved. I know BFD for instance, all the samples were recorded at 44.1, and the software/player (or daw??) oversamples the sounds of your using higher session sample rates.

So when I print them I'll actually be printed over sampled versions. I veiw this as a degradation (at least technically) although I'm not sure if it is in reality, or would be perceptible for better or worse.

What's also very interesting (to me at least) is Sequoiacan using source tracks of different formats like MP3, and different sample rates, all in the same session! Some video editors allow different formats in the same session too.

Also wild is samplitude and sequoia support 384k sample rates! I don't know of any interfaces that go that high, although antelope has the 2ch mastering converter that does go that high (6k price tag....)

I wonder if we will eventually move into a true single standard sample rate at least for rec/mix/mastering. Particularly since there's so many final delivery formats, I think it could eventually happen.

This is nothing new even tape had different sizes and speeds.

kmetal, post: 441293, member: 37533 wrote: Also things get even more complicated when vsti are involved. I know BFD for instance, all the samples were recorded at 44.1, and the software/player (or daw??) oversamples the sounds of your using higher session sample rates.

That's a good point.
This makes me think... Is upsampling less damageable than downsampling ? I'm tempted to say yes. Adding zeros is easier than removing some no ?

kmetal, post: 441293, member: 37533 wrote: Also things get even more complicated when vsti are involved. I know BFD for instance, all the samples were recorded at 44.1, and the software/player (or daw??) oversamples the sounds of your using higher session sample rates.

So when I print them I'll actually be printed over sampled versions. I veiw this as a degradation (at least technically) although I'm not sure if it is in reality, or would be perceptible for better or worse.

What DAW are you planning on using Kyle?...was it Samplitude??

Maybe Chris audiokid would have a better knowledge of how Samp handles the SRC from a DAW perspective when it comes to the SRC.

audiokid, post: 441262, member: 1 wrote: The difference is sonically tighter, smoother. Once you have 2 DAW's all connected like I experience, the ability to flow between tracking, mixing and mastering is a game changer.

This is why I have been putting my toe in the water when it comes to the 2 DAW setup...for me the only thing letting me down is not having high-end converters and a high-end monitoring system to take full advantage of what I am doing.

Chris you have been using the Dangerous Monitor correct?

How would the Cranesong Avocet compare?...have you tried it??

I see many top notch studios are using the Avocet and its something on the wish list for me. Can you achieve the same 3 way monitoring with it as you have with the Dangerous Monitor???

Chris Perra, post: 441240, member: 48232 wrote: I just took some drums at 44.1 then up sampled them to 96k.. Then rendered both at 44.1 and 96k with no faders or plug ins to 16 bit 44.1. They dont phase cancel.. It seems it has nothing to do with plug ins.

I wonder what others encounter

Did you flip the phase of one or both of the tracks 180*? Sorry if this is a dumb question, but you didn't list it in the steps. W/o that part of it, it wouldn't cancel, it would sum, and you wouldn't be able to hear any differences. As far as I understand it, in theory, you would have to bounce one track out of phase to hear the difference. I'm not sure which track. I would try it both ways. I would think that you can't do them both out of phase, because then, they are back in relative phase. They would both just be 180* out.

However, in this case he is adding a component. The two tracks have a different sampling rate, so I'm not sure. But even with the sampling rate change, I would think only one track would be out of phase, bc you don't get cancellation if they are both out of phase. Someone please correct me if I'm wrong.

Is it possible for you to upload ur tracks?

I will try it myself later, just so I can observe the phenomenon that diagram is describing. I have never changed the sampling rate during a project. Once I've opened the project, almost always in 24/48, I leave it. So I can't comment at all on that part of it. I'm not doubting the OP. I'm not talking about that part of it at all. So forget the sampling rate part of it...

But I've never flipped the phase of an entire track and compared it to another one, no matter what the sampling rate. I look forward to messing with it. In theory, with the way monitors work, around the crossover point, you will not get as much cancellation. While the cancellation will not be absolute, it should do what the guy says, if aliasing appears.

I'm kind of ticked off at myself that I didn't think of this.

bouldersound, post: 441242, member: 38959 wrote: Hey, wait, I though resampling was supposed to be a bad thing. Now all of a sudden it's a good thing?

I think it was just to prove the point that a 96khz sample rate is better than 44.1. At least, that is what I thought, he was showing that 96 is better. Which leads me to the question, is 192 even better?

Boswell, post: 441244, member: 29034 wrote: First example: take the diagram that shows the spectrum of a waveform whose fundamental is not stated, but presumably at about 7KHz. Quote: "At 44.1kHz sampling rate the 3rd harmonic is below the anti-aliasing cutoff filter, but above the Nyquist-Shannon frequency". If you look at what he has drawn as an anti-aliaising filter, it's only about 5dB down at the Nyquist frequency, so it's simply the wrong frequency of filter to be using. Anti-aliasing filters designed for the CD audio frequency of 44.1KHz are 3dB down at 20KHz and somewhere around 70dB down at the Nyquist. His incorrect example leads to false conclusions.

I see what you mean about the diagram. I don't know why he is not telling us what his f0 is. I'm assuming the white vertical line is 20khz as he says the aliasing is folded back into the audible spectrum. Also he doesn't say what the NSF is, but the sample rate is 44.1, so it seems like that line should be 20khz. But there is not scale to the rest of it either. He lists it as "ideal filter" but not what it is. However, he also says (paraphrase) that the diagram is simplified for understanding. But I agree, I find that diagram to be more confusing than helpful.

Also, that's all those aliasing filters remove? A couple db? The natural slope of the harmonic isn't much higher than the orange line he has drawn. I mean it's better than nothing. With that diagram, I think he is just trying to show how foldback works, and literally nothing more.

Boswell, post: 441244, member: 29034 wrote: Second example: the assumption that filters that perform anti-aliaising in the analogue domain ahead of sampling work at the top end of the audio band. In the last 15 - 20 years, the only types of ADC and DAC used for sampling and reconstruction of audio sample well outside the audio frequency band, typically at many MHz. At this rate, it's possible to reduce the wordlength from 16 down to 4 or even 1 bit, and then recreate the longer wordlengths during the decimation (down-sampling) process in the digital domain. The point is that, when sampling at these MHz rates, the analogue anti-aliaising filters can be very simple designs that roll-off gently in the tens or even hundreds of KHz, causing minimal phase irregularities in the audio band. The anti-aliaising filters for containing the information in the band 0 - 20KHz are implemented digitally as part of the decimation process.

You explained that extremely well. I've bookmarked this page.

I still have the question of recording, and I'm asking you bc you seem to understand this well. If a girl I work with sets her session to 16/44.1, and records her verses; she then bounces at 16/44.1 and sends me the tracks in wav format. Is that track not written at a bit depth of 16 and 44,100 samples per second once it has passed through the ADC? If I take it, and set it up in my session at 24/96, it will function, but it's not all of a sudden as if it was recorded at 24/96 right? Wrong?

Now suppose she sets her session to 24/96 and records her verses. She bounces a wav at 16/44.1 and sends it to me. Can I actually use that as a genuine 24/96 file? Meaning is the info there, in the bounce, because the session was set to 24/96? Or once you set those parameters in the bounce, you have a 16/44.1 bounce (this is what makes sense to me).

Or even if in my own session, I'm at 16/44.1 and I record something. I then change the session to 24/96....would that recording quality then change, but only because it has the analogue info to look at? Essentially rewriting the file in a true 24/96? Or would it stay 16/44.1 because that's what it was when it was recorded?

As you can see, I know very little about daws

Boswell, post: 441244, member: 29034 wrote: let alone any problems due to the use of digital SRC.

As if I haven't bothered you enough, can you expand on that? You can pm me if you feel like I'm hijacking the thread.

@Sean G Great thread.

***Edit, some of you may find this page interesting. People have differing opinions on many of the subcategories of this, but if you find this stuff interesting, this is thought provoking

http://www.wescottdesign.com/articles/Sampling/sampling.pdf

Brother Junk, post: 441317, member: 49944 wrote: Did you flip the phase of one or both of the tracks 180*? Sorry if this is a dumb question, but you didn't list it in the steps. W/o that part of it, it wouldn't cancel, it would sum, and you wouldn't be able to hear any differences. As far as I understand it, in theory, you would have to bounce one track out of phase to hear the difference. I'm not sure which track. I would try it both ways. I would think that you can't do them both out of phase, because then, they are back in relative phase. They would both just be 180* out.

However, in this case he is adding a component. The two tracks have a different sampling rate, so I'm not sure. But even with the sampling rate change, I would think only one track would be out of phase, bc you don't get cancellation if they are both out of phase. Someone please correct me if I'm wrong.

Is it possible for you to upload ur tracks?

I will try it myself later, just so I can observe the phenomenon that diagram is describing. I have never changed the sampling rate during a project. Once I've opened the project, almost always in 24/48, I leave it. So I can't comment at all on that part of it. I'm not doubting the OP. I'm not talking about that part of it at all. So forget the sampling rate part of it...

But I've never flipped the phase of an entire track and compared it to another one, no matter what the sampling rate. I look forward to messing with it. In theory, with the way monitors work, around the crossover point, you will not get as much cancellation. While the cancellation will not be absolute, it should do what the guy says, if aliasing appears.

I'm kind of ticked off at myself that I didn't think of this.

I think it was just to prove the point that a 96khz sample rate is better than 44.1. At least, that is what I thought, he was showing that 96 is better. Which leads me to the question, is 192 even better?

This probably arises through a misunderstanding of what happens when upsampling. Going from 44.1 to 96 KHz is not a simple process, but involves digital padding and then filtering. The 96KHz result does not contain the same samples as the 44.1KHz waveform, so will not null when subsequently down-converted to 44.1 KHz. What it does contain is an approximation to the same original waveform, but it certainly does not have any more information in it than the original. This means that if the original was 44.1/24, then there will be no information in the frequency range 20-40KHz in the 96/24 version of the track. If the original was 44.1/16, then although the 96/24 (upsampled) version will typically have non-zero values in the bottom 8 bits of each sample, the track will not have any more than 16 bits of information in each sample. This is not immediately easy to comprehend, but Shannon information theory says this is so. You can expand this reasoning to 192KHz.

If the original analogue waveform was sampled at a lower rate than the DAW processing rate, the lower rate imposes a limit on the amount of information captured, and no amount of processing can restore what was not captured in the first place. You can, of course, add information during processing (such as deliberate distortion), and this will occupy the full bandwidth available within the DAW rate.

It's interesting that in the transfer of old cylinder and 78rpm recordings to CD, it's preferable to leave some of the system hiss in the result, as that fools the ear into thinking there was more high-frequency content in the original recording than there actually was. Restricting the output bandwidth to the known bandwidth of the recording makes the transfered result sound very dull, even though it's a more accurate capture than the version that has more hiss.

Brother Junk, post: 441317, member: 49944 wrote: I still have the question of recording, and I'm asking you bc you seem to understand this well. If a girl I work with sets her session to 16/44.1, and records her verses; she then bounces at 16/44.1 and sends me the tracks in wav format. Is that track not written at a bit depth of 16 and 44,100 samples per second once it has passed through the ADC? If I take it, and set it up in my session at 24/96, it will function, but it's not all of a sudden as if it was recorded at 24/96 right? Wrong?

Right.

Brother Junk, post: 441317, member: 49944 wrote: Now suppose she sets her session to 24/96 and records her verses. She bounces a wav at 16/44.1 and sends it to me. Can I actually use that as a genuine 24/96 file? Meaning is the info there, in the bounce, because the session was set to 24/96? Or once you set those parameters in the bounce, you have a 16/44.1 bounce (this is what makes sense to me).

There's no more information than in the 44.1/16 version of the track.

Brother Junk, post: 441317, member: 49944 wrote: Or even if in my own session, I'm at 16/44.1 and I record something. I then change the session to 24/96....would that recording quality then change, but only because it has the analogue info to look at? Essentially rewriting the file in a true 24/96? Or would it stay 16/44.1 because that's what it was when it was recorded?

It would change to a 96/24 file because that's what you set it to do. There would be no more information than in the original recording. There would indeed slightly less information, as any sampling rate change process is lossy.

Sean G, post: 441315, member: 49362 wrote:

Maybe Chris audiokid would have a better knowledge of how Samp handles the SRC from a DAW perspective when it comes to the SRC.

I can't currently comment on how Samplitude deals with SRC because I have been using two DAW's and an analog pass between uncoupled DAW's to change SR for years now. I can "up or down" sample with what I hear as more pleasing, open, tighter result.
The down side to this method: SRC is not as fast because everything is in "real time". The pro's (beside sound quality) I mix into the capture for a variety of reasons anyway.

That being said, there is an interesting thread somewhere on RO about how Tim Dolbear dithers in Samplitude.

I took a 44.1 project.. Then loaded that into cubase all individual l tracks and faders into a new project set to 96. It did the conversion to 96..
I then rendered both the 96k project and the 44.1 project.. Both with no plug ins down to 16 bit 44.1

Then I put one of them 180 degrees out of phase using the phase switch button on the stereo fader. They didn't phase cancel.. So in this case plug ins had nothing to do with the differences in the 2.. I'm going to check though I may have had a limiter and dither on the e main buss. Perhaps that might have done something if so..

Sean G, post: 441316, member: 49362 wrote: Chris you have been using the Dangerous Monitor correct?

How would the Cranesong Avocet compare?...have you tried it??

Edited:
I've not tried the Avocet but (if I had a choice ) I would also not be interested in it.

For my particular monitoring workflow.

I like the Dangerous Monitor ST. The Dangerous "Monitor" (without the ST) is the predecessor to the "ST" version. I like the ST because it doesn't have a converter inside it (forcing me to think boxed in... and to be held hostage to the eventually dated conversion over time.
I want an independent controller with DA options in various sections of my workflow for a variety of reasons, if that makes better sense? Converters are constantly evolving /improving. The Dangerous Monitor ST needs no improvement.

imho: The ST is the most amazing monitor controller made . As of Sept 2016... it still does everything right.

Boswell, post: 441322, member: 29034 wrote: It's interesting that in the transfer of old cylinder and 78rpm recordings to CD, it's preferable to leave some of the system hiss in the result, as that fools the ear into thinking there was more high-frequency content in the original recording than there actually was. Restricting the output bandwidth to the known bandwidth of the recording makes the transfered result sound very dull, even though it's a more accurate capture than the version that has more hiss.

Great stuff Bos.(y)

Some of this is why I do a two DAW "uncoupled" pass. I do not believe analog gear (even a straight wire for that matter) is improving the audio, but in "just the right amount", for those harsh mixes, analog recaptured on a second box adds (or fitters) enough of the edge, creates a a faux finish that simply sounds better in comparison to a cleaner digital mix.
NOTE: If a mix was stellar to begin with, that last thing I would do is smear it up with anything analog.

That being said: The reasons I might include a straight wire or added character gear pass could be exclusively to "help"crappy sound of bad conversion. The analog pass between two DAW's (example: 96k to 44.1 or simply both at 44.1 for that matter) can improve the stereo perception. Even some dirty hiss recaptured on a second DAW will give us the perception that the mix has more high end (broader bandwidth) (more FM like) after we re-process a mix through a two DAW system and some analog EQing.
The addition of analog gear in a pass is more about smoothing out the sharp edges, fitting the changes (altering or downsampling audio back together (left, right and centre) in a way that doesn't seem to sound as good just digitally.
The addition of analog (OTB mixing) is never is about making the audio clearer and more pristine.To me its more about changing it while trying not to sound like I blurred it up. Blurring it up happens right away so you really need to be listening. Its super easy to fool yourself into thinking a change of smear is the improvement while missing you just crossed the transients over and actually blurred up a clients mix. Thus, those who see the benefit of transformeless summing and mastering tools.

Two uncoupled DAW's with some character analog gear gets us closer to the world of Sound Design. Two DAW's makes it easier to study cause and effect.

Remember the days before we had a DAW, when we used samplers ? Well, now we have DAW's and the ability to do even more.

Boswell, post: 441322, member: 29034 wrote: It's interesting that in the transfer of old cylinder and 78rpm recordings to CD, it's preferable to leave some of the system hiss in the result, as that fools the ear into thinking there was more high-frequency content in the original recording than there actually was. Restricting the output bandwidth to the known bandwidth of the recording makes the transfered result sound very dull, even though it's a more accurate capture than the version that has more hiss.

Great stuff Bos.(y)

Some of this gives reason why I do a two DAW "uncoupled" pass. I do not hear analog gear (even a straight wire for that matter) as improving the audio, but "just the right amount", especially for harsh mixes recaptured on a second box adds (smears or filters) enough of the edge, creates a faux finish that simply sounds more natural, pleasing in comparison to a cleaner digital mix that is too truthful per-say. It can also help broaden up the stereo spread.

That being said: The reasons I might include a straight wire or added character gear pass could be exclusively to "help" distract the brittle edge of poor conversion. The analog pass between two DAW's (example: 96k to 44.1 or simply both at 44.1 for that matter) can improve the stereo perception too. As Bos touched on, even some dirty hiss recaptured on a second DAW will give a perception that the mix has more high end (broader bandwidth) (more FM like) after we re-process a mix through a two DAW system and some analog EQing.
The addition of analog gear in a pass is more about smoothing out the sharp edges, fitting the changes (altering or down-sampling audio back together (left, right and centre) in a way that doesn't seem to sound as good just digitally.
The addition of analog (OTB mixing) is never about making the audio clearer and more pristine.To me its more about changing it while trying not to sound like I blurred it up. Blurring happens right away so you really need to be listening. Thus, the benefit of transformerless summing and mastering tools .
Its super easy to fool yourself into thinking a change of smear is the improvement while missing you just crossed the transients over and actually blurred up a clients mix.

Remember the days before we had a DAW, when we used samplers? Well, now we have DAW's and the ability to do even more.

• Two uncoupled DAW's makes it easier to study cause and effect.
• Two uncoupled DAW's with some character analog gear gets us closer to the world of Sound Design.

Boswell, post: 441322, member: 29034 wrote: This probably arises through a misunderstanding of what happens when upsampling. Going from 44.1 to 96 KHz is not a simple process, but involves digital padding and then filtering. The 96KHz result does not contain the same samples as the 44.1KHz waveform, so will not null when subsequently down-converted to 44.1 KHz. What it does contain is an approximation to the same original waveform, but it certainly does not have any more information in it than the original. This means that if the original was 44.1/24, then there will be no information in the frequency range 20-40KHz in the 96/24 version of the track. If the original was 44.1/16, then although the 96/24 (upsampled) version will typically have non-zero values in the bottom 8 bits of each sample, the track will not have any more than 16 bits of information in each sample. This is not immediately easy to comprehend, but Shannon information theory says this is so. You can expand this reasoning to 192KHz.

If the original analogue waveform was sampled at a lower rate than the DAW processing rate, the lower rate imposes a limit on the amount of information captured, and no amount of processing can restore what was not captured in the first place. You can, of course, add information during processing (such as deliberate distortion), and this will occupy the full bandwidth available within the DAW rate.

It's interesting that in the transfer of old cylinder and 78rpm recordings to CD, it's preferable to leave some of the system hiss in the result, as that fools the ear into thinking there was more high-frequency content in the original recording than there actually was. Restricting the output bandwidth to the known bandwidth of the recording makes the transfered result sound very dull, even though it's a more accurate capture than the version that has more hiss.

Right.

There's no more information than in the 44.1/16 version of the track.

It would change to a 96/24 file because that's what you set it to do. There would be no more information than in the original recording. There would indeed slightly less information, as any sampling rate change process is lossy.

Thank you!