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Q for Paul Frindle Dynamics Plugins
Old 12th March 2007
  #181
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Quote:
Originally Posted by lucey View Post
I agree with you, but why is it so? What is it about the way MB works?
Well they modulate the overall freq response of the programme depending on what happens in the music. I simply can't stand the sound of it - it's so unnatural. Nothing in the natural environment does this.

I have made one of these in my professional life (ages ago) - because other people like them, but it never got released as a product.


Quote:
Quality recording is a game of inches, not double blind tests on the general public.
Very true. We must always remember there is a distinction between discussions about theory - and what we must do with what we actually have now in order to make the right result artistically within the current trends.
Old 15th March 2007
  #182
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Quote:
Originally Posted by Paul Frindle View Post
I don't know why you could not hear it? I have never yet found a complex programme track that is not changed audibly by bit rate compression. Some very simple signals (like a few sine waves) are of course fairly unchanged to the ear as the coder does not need to work hard.

But I suppose one possibility might be that the track had already been compressed and decoded again before the CD was made? In this case the further coding your end would change it very little? But this would seem very unlikely?

It has to be said that the artefacts of coding at high rates can be a bit subtle (and unusual) and when first meeting them it's necessary to learn what they are - after which of course you will hear them all the time and they may begin to get on your nerves :-( ALthough for some very compressed and saturated popular music the encoding actually improves it's sound IMHO..
I've done another test for trying to perceive differences between aiff and mp3 and finally, by focusing on what there was at lower level behind a front solo I've been able to hear differences.
The track was with a rhythmic guitar behind a loud synth solo and in the mp3 was clear to hear that the guitar on the back had a uncontrolled dynamic whereas the aiff had the guitar at a natural and controlled level.
I want to thank you for this, but sometimes we have to learn what focus on, and learn to educate our ears. This your input has been very useful, thank you again...
Old 16th March 2007
  #183
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Quote:
Originally Posted by innesireinar View Post
I've done another test for trying to perceive differences between aiff and mp3 and finally, by focusing on what there was at lower level behind a front solo I've been able to hear differences.
Glad to hear your test was successful! What did you end up using as an mp3 encoder, which settings, and which decoder ?
There are so many options that just "mp3" doesn't say much
Old 16th March 2007
  #184
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I still have one question (really, just one)

Thanks Paul. I'm learning a tremendous amount here and am inspired to keep learning more. I am totally convinced of the waste of 96k and the rightness of 48k. I work in 44.1/24 and I wonder if the benefits of of working in 48k outweigh the effects of the downsample to CD.

This was "common" wisdom a few years ago, and so many of us avoided downsampling at all costs. This may be outdated information. If it is indeed better to work at 48k whose downsample algo would you trust? If there's a link on this subject I'd love to see it as well.

Thanks
Old 16th March 2007
  #185
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Quote:
Originally Posted by Kees de Visser View Post
Glad to hear your test was successful! What did you end up using as an mp3 encoder, which settings, and which decoder ?
There are so many options that just "mp3" doesn't say much
Sure, you're right, I'm referring to an mp3 converted by iTune at 192kbs
Old 24th March 2007
  #186
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Paul, if you're still here I would ask you another thing.
In this thread we told about high SR etc but I'm wondering which relationship there is between the distortion in percent and in level.
In other words, supposing we're having a signal at 100 dB SPL that has a distortion of 1%, what will be the level of the distortion? It could be possible 80 dB (1/100 of the signal power)?
If my thesis works why we need to dithering the signal, for masking what is happening many dozen of dB below, while just 20 or 30 (0,1%) dB below we have distortion? Even when talking about 24 bit we are talking about what happens below more than 90 dB below. Why?
Thank you
Old 25th March 2007
  #187
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Quote:
Originally Posted by innesireinar View Post
Paul, if you're still here I would ask you another thing.
In this thread we told about high SR etc but I'm wondering which relationship there is between the distortion in percent and in level.
In other words, supposing we're having a signal at 100 dB SPL that has a distortion of 1%, what will be the level of the distortion? It could be possible 80 dB (1/100 of the signal power)?
If my thesis works why we need to dithering the signal, for masking what is happening many dozen of dB below, while just 20 or 30 (0,1%) dB below we have distortion? Even when talking about 24 bit we are talking about what happens below more than 90 dB below. Why?
Thank you
This is a very good question and I have been asked it many times. The logic some people like to apply is that because speakers and repro systems have whole percents of distortion, why would we worry about stuff many time lower around -80 to -90dB? BTW this attitude was widespread at the time I was researching converter issues in the 1980's :-(

The reason is that the distortions from repro systems is normally low order (2nd, 3rd and sometimes 4th or 5th) and this is less annoying than very much higher orders caused by truncation distortion and the like.

Also because the speaker distortion is strictly harmonically and dynamically coupled to the programme itself (often resembling real instrument harmonics) it is far easier to accomodate within our natural perceptions - often sounding like timbre change rather than distortion per se.

Whereas the distortion from digital error is far more complex in the freq domain, largely decoupled to the programme (with unnaturally higher audible artefacts at lower levels - and stretches high in the freq domain. This makes it immediately audible and annoying even at suprisingly low amounts..

This kind of thing is one example of how we can become mislead by raw numbers produced by previous research. When new things happen we need to review our thinking and revise standards etc..
Old 25th March 2007
  #188
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Quote:
Originally Posted by Paul Frindle View Post
Also because the speaker distortion is strictly harmonically and dynamically coupled to the programme itself (often resembling real instrument harmonics) it is far easier to accomodate within our natural perceptions - often sounding like timbre change rather than distortion per se.

Whereas the distortion from digital error is far more complex in the freq domain, largely decoupled to the programme (with unnaturally higher audible artefacts at lower levels - and stretches high in the freq domain. This makes it immediately audible and annoying even at suprisingly low amounts..

This kind of thing is one example of how we can become mislead by raw numbers produced by previous research. When new things happen we need to review our thinking and revise standards etc..
Not to appear to trump Pauls expertise here ... but his answer also explains why each converter product has a unique signature, and why we've kept looking for better new ones over the last 20 years. Tiny amounts of this distortion are not tiny after all.

Or ... the world of quality is not governed by the world of quantity ... as my instructor used to say.
Old 26th March 2007
  #189
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Quote:
Originally Posted by esteso View Post
I work in 44.1/24 and I wonder if the benefits of of working in 48k outweigh the effects of the downsample to CD.

This was "common" wisdom a few years ago, and so many of us avoided downsampling at all costs. This may be outdated information. If it is indeed better to work at 48k whose downsample algo would you trust? If there's a link on this subject I'd love to see it as well.

Thanks
I too am interested in your thoughts on this dilemma, Paul. Let's say in an ITB mastering project using a short chain of high quality plug-ins with internal upsampling option.

Thanks again for the good information you're sharing here. Following your advice has made my recordings sound a whole lot better.
Old 26th March 2007
  #190
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Quote:
Originally Posted by bonne View Post
I too am interested in your thoughts on this dilemma, Paul. Let's say in an ITB mastering project using a short chain of high quality plug-ins with internal upsampling option.

Thanks again for the good information you're sharing here. Following your advice has made my recordings sound a whole lot better.
Glad to hear I have helped in some way :-)

Well I would avoid downsampling personally. It's true that they are getting better, but they can never be perfect.

Also a downsampler after your final processing treatment could cause some unwanted effects - especially if you are comp/limiting.

I can't vouch for other processing plug-ins, but if I were using the ones I designed I would know there was no significant advantage in running at higher rates anyway.

The philosophical point worth making is that your result in the format you intend is all that matters - that is the only product. If it's good at 44.1K there is no justification for running at higher rates - any more than a work of art might be considered 'better' if the artist had used different canvas.
Old 27th March 2007
  #191
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Hi Paul, Thank you for your precious explanations.
By going a bit off topic some moths ago we had a discussion about the use of very short delay, with one of the two LR channels in phase rev for getting big and fat vocal track. This tip is very cool but sometimes I get an unwanted phasing FX.
Last night I've found this old thread by GM where he told about prime numbers, pseudo modes etc. I don't understand anything...
If you want and if you have time can you explain something about that and tell me if these values GM was talking about are involved in this phasing FX that sometimes I'm getting?
http://forums.musicplayer.com/ubbthr...age/34/fpart/2
Old 27th March 2007
  #192
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Quote:
Originally Posted by innesireinar View Post
Hi Paul, Thank you for your precious explanations.
By going a bit off topic some moths ago we had a discussion about the use of very short delay, with one of the two LR channels in phase rev for getting big and fat vocal track. This tip is very cool but sometimes I get an unwanted phasing FX.
Last night I've found this old thread by GM where he told about prime numbers, pseudo modes etc. I don't understand anything...
If you want and if you have time can you explain something about that and tell me if these values GM was talking about are involved in this phasing FX that sometimes I'm getting?
http://forums.musicplayer.com/ubbthr...age/34/fpart/2

It is inevitable that adding delayed signals together (even acoustically) will cause phasing if their lengths produce nodes in the audible range. Changing polarity does not suppress it. The trick is to make them long enough to avoid a honk in the sensitive freq ranges.

In the post you refer to, GM is saying that with multiple delays he tries to pick prime numbers for the delay sizes in an attempt to avoid this. However this does not really work as what is needed are relative primes not absolute primes - and bear in mind that it's the differences between delays that cause phasing (if they were all the same there wouldn't be any etc).

What he is achieving instead is greater perceived reflection density, as the edges of the delays are less likely to paritally synchronise over time - so they remain complex :-)

We use this kind of thing in reverb when we are trying to create a characterless ambience - and in many ways much of reverb research has focussed in making just such mushy sounding reverbs with the minimum number of delays etc.. However we must remember that real spaces do not reflect this way - being related to the regular geometry of the space, much of it is specular and repetative in nature causing dramatic freq response nodes. This is actually what gives the spaces their character.

Paradoxically therefore - one of the biggest advantages of convolution reverb is that it captures these specular reflections - (basically all the stuff artificial reverb designers have been trying to supress for decades :-(..) - and this is the major reason they sound more realistic. The downside of convolution however is that it is not intrisically variable - the room size cannot be freely changed without destroying the realism etc..

Realising this, in the reverb that I designed I broke away from all this and deliberately made the synthetic reverb reflection model based on real geometrical space. In this way one can have much of the realism of convolution whilst retaining the powerful ability to vary it freely to make your own spaces and sounds - in a way that convoloution does not really allow. Often the slightest tweek can make the difference betwen something just ok or something fantastic :-)

I then did a load of set-ups based on the sounds of real spaces to illustrate this (mostly in the post section of the factory set-ups). I also included some novel set-ups for very small spaces like boxes, tubes, horn speakers and what not. Not only do these illustrate how it works, they are also useful in post pro and making instrument and amp effects etc..

This rather unconventional reverb received mixed reviews. On the one hand there were those who immediately cottoned on to what it did and appreciated it lots. On the other hand there were those who were put off by the departure from expectation for synthetic reverbs (those who were expecting the familiar mushy sound) - some of whom only realised later how close it could sound to their convolution units.

With more time and resource I could improve on this application further - there was a chunk of processing I had to leave out because at the time I could not get Sony's dev system to compile it within the time allowed. Nothing is ever perfect hey...
Old 27th March 2007
  #193
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[QUOTE=Paul Frindle;1202365]It is inevitable that adding delayed signals together (even acoustically) will cause phasing if their lengths produce nodes in the audible range. Changing polarity does not suppress it. The trick is to make them long enough to avoid a honk in the sensitive freq ranges.

Is there a formula for setting the time delay avoiding these nodes in the audible range?

In the post you refer to, GM is saying that with multiple delays he tries to pick prime numbers for the delay sizes in an attempt to avoid this. However this does not really work as what is needed are relative primes not absolute primes - and bear in mind that it's the differences between delays that cause phasing (if they were all the same there wouldn't be any etc).

What does it mean "relative primes" and "absolute primes"?

What he is achieving instead is greater perceived reflection density, as the edges of the delays are less likely to paritally synchronise over time - so they remain complex :-)

We use this kind of thing in reverb when we are trying to create a characterless ambience - and in many ways much of reverb research has focussed in making just such mushy sounding reverbs with the minimum number of delays etc.. However we must remember that real spaces do not reflect this way - being related to the regular geometry of the space, much of it is specular and repetative in nature causing dramatic freq response nodes. This is actually what gives the spaces their character.

Paradoxically therefore - one of the biggest advantages of convolution reverb is that it captures these specular reflections - (basically all the stuff artificial reverb designers have been trying to supress for decades :-(..) - and this is the major reason they sound more realistic. The downside of convolution however is that it is not intrisically variable - the room size cannot be freely changed without destroying the realism etc..

Realising this, in the reverb that I designed I broke away from all this and deliberately made the synthetic reverb reflection model based on real geometrical space. In this way one can have much of the realism of convolution whilst retaining the powerful ability to vary it freely to make your own spaces and sounds - in a way that convoloution does not really allow. Often the slightest tweek can make the difference betwen something just ok or something fantastic :-)

Are you referring to the Sony rev?
Old 30th March 2007
  #194
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[QUOTE=innesireinar;1202899]
Quote:
Originally Posted by Paul Frindle View Post
Realising this, in the reverb that I designed I broke away from all this and deliberately made the synthetic reverb reflection model based on real geometrical space. In this way one can have much of the realism of convolution whilst retaining the powerful ability to vary it freely to make your own spaces and sounds - in a way that convoloution does not really allow. Often the slightest tweek can make the difference betwen something just ok or something fantastic :-)

Are you referring to the Sony rev?
Yes - that is the only one I have ever been involved in designing. I have never done one previously so had to look at it from the ground up..
Old 2nd April 2007
  #195
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Quote:
Originally Posted by Paul Frindle View Post
Glad to hear I have helped in some way :-)

Well I would avoid downsampling personally. It's true that they are getting better, but they can never be perfect.

Also a downsampler after your final processing treatment could cause some unwanted effects - especially if you are comp/limiting.
So in the case of 48k files for a CD mastering project you would do a roundtrip to analog first, resample to 44.1 on your way back into digital and THEN do your processing.

In other words you would prefer a hit in the DA->AD roundtrip over the hit i software SRC?


The Voxengo company writes this about their r8brainPRO SRC:

Like many existing SRC programs, r8brain PRO offers you a linear-phase conversion mode. But more importantly, you also have an option of using the minimum-phase conversion mode, which finally brings SRC with true analog qualities to affordable digital audio workstations: in this mode, r8brain PRO works like an ideal digital-to-analog converter followed by an analog-to-digital converter to resample the audio. This eliminates pre-ringing associated with linear-phase designs, while introducing only a minimal amount of phase coloration.


In recent SRC shootouts the r8brainPRO is up there among the very best, along with Izotope's 64-bit algo (measurements and listening tests).

Here is a link to one such shootout: SRC Comparisons

From a technical point of view, Paul, what do you think of the "like an ideal digital-to-analog converter followed by an analog-to-digital converter to resample the audio"- claim in the quote?

Thank you

Jørn
Old 4th April 2007
  #196
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Quote:
Originally Posted by bonne View Post
So in the case of 48k files for a CD mastering project you would do a roundtrip to analog first, resample to 44.1 on your way back into digital and THEN do your processing.

In other words you would prefer a hit in the DA->AD roundtrip over the hit i software SRC?
Ok, this is a difficult situation with many subtleties.

Firstly I would not prefer an analogue round trip over a good SRC. Whilst it is the case that an SRC is notionally 'trying' to be a perfect D/A and A/D combination, real converters can introduce more errors than the digital SRC equivalent function. Obviously this all depends on the relative quality of the units in question - but given the lack of in-depth info I would opt for an SRC.

Secondly, the order of things in the example you quote has swings and roundabouts on both sides of the fence - and all this is very involved to explain without doing a massive long post. Whether it is better to put your processing pre or post the SRC depends on what exactly your priorities are and what type of programme and processing you are working with.

To give one example (probably the biggest one you would notice) is with intersample peaking. Basically this is a situation where a combination of samples that are not themselves flat out give rise to a much greater level when reconstructed in the DAC. This can cause overloads that are not visible on DAW meters because meters only show you sample values etc..

The sort of processing that causes this most often is compression and limiting - as this increases the predominance of unnaturally big value samples and therefore increases the possibility of intersample peaks.

By reconstruction we really mean the filtering that has to be done in the DAC to lose the effects of time sampling and give out a constant signal etc.. So any reconstruction of the signal (analogue or digital) can expose intersample peaks that might otherwise go unnoticed (until someone plays it all back etc). When exposed this way these peaks manifest themsleves as increased peak level.

Your SRC - whether a combination of D/A and A/D or a digital SRC application - includes a rconstruction filtering stage. So it can produce output levels that are bigger than the apprent input levels as read on the DAW meters. For some very heavily processed sounds in mixes this increase might occasionally be up to 3dB or even more in specific places.

If you are with me so far - you can now see how an SRC might change the signal, by producing higher signals, that may pass if you have left some level above average modulation - or clip if you haven't.

Now to go back to the processing order question, it is clear that putting compression before the SRC can produce a different effect from putting it after. If put before the SRC the compressor will act on sample values only - if put after it will operate on partially reconstructed signal values which could be 3dB or more larger.

This means that the same compression settings may sound qieter if put after the SRC (because it will compress earlier) than they did when put before the SRC. So if you were going for absolute max volume and to hell with what happens to people's DACs (current fashion) you are better off putting the comp/limiter before the SRC. But if you want to do the 'right' thing and avoid some of the potential intersample peaking (and put up with less overall volume) you would put it after the SRC. And since most intersample overs occur on HF, the programme may end up sounding 'softer' if comp/limited after the SRC.

I hope this makes sense - I did a much greater explaination of this stuff with examples on PSW forums in a discussion about mixing ITB and OTB. That thread is now a sticky on PSW.

But this is not even the whole story and compression is not the only thing that can change things - but I'll stop with that here for now and just answer your next query below:

Quote:

The Voxengo company writes this about their r8brainPRO SRC:

Like many existing SRC programs, r8brain PRO offers you a linear-phase conversion mode. But more importantly, you also have an option of using the minimum-phase conversion mode, which finally brings SRC with true analog qualities to affordable digital audio workstations: in this mode, r8brain PRO works like an ideal digital-to-analog converter followed by an analog-to-digital converter to resample the audio. This eliminates pre-ringing associated with linear-phase designs, while introducing only a minimal amount of phase coloration.

Ok - I am not about to comment on other people's products here in depth. But there are two things here to note.

Firstly there is nothing 'analogue' about minimum phase filtering in the slightest - the statement means nothing what so ever :-(

Secondly, minimum phase filtering (whether analogue or digital) is basically a filter that has NOT undergone phase correction. This means that the relative phases of differing freqs in the programme will be changed by the filter and the waveform shape will therefore be changed as well. It also means that the effective timing of the different freq parts of your programme will be decalibrated with respect to each other. Obviously not an optimum situation :-(

Ok now - if we go back to thinking about the previous explaination of how reconstruction filtering can increase peak levels; the other thing that can change peak levels is modifying the waveform shape (obviously).

So on top of potentially changing the sound of the programme (which they admit), using an uncorrected minimum phase filter within the SRC can do - everything we have described before - AND increase the peak level even further by changing the waveform shape - as well!!

You can draw your own conclusions from that...... For consistency and transparency you best leave it out of minimum phase mode.

And BTW, there can never be such a thing (in analogue or digital) as an 'ideal D/A and A/D' model. It is not possible.....

Why is it suddenly that this site keeps logging me out? :-(
Old 6th April 2007
  #197
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Very useful information, Paul. I can see why you would avoid SRC when possible.

iZotope's 64-bit SRC has a new take on the linear phase/minimal phase dilemma that they call "hybrid ringing control". People who have heard this and compared it to the competition tend to agree that this is (one of) the best sounding SRC algo(s) so far.

Here's a link to a brief description of their design:

http://www.izotope.com/tech/src/
Old 8th April 2007
  #198
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Hi,

This is all very interesting. I do understand the points being made about the SRC stuff but how do you explain the fact that I can clearly hear an improvement in the recorded audio quality when recording at 96k as opposed to 44.1 on my ProTools HD system? The difference to me is obvious and repeatable.
Old 8th April 2007
  #199
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Quote:
Originally Posted by Wilburguy View Post
Hi,

This is all very interesting. I do understand the points being made about the SRC stuff but how do you explain the fact that I can clearly hear an improvement in the recorded audio quality when recording at 96k as opposed to 44.1 on my ProTools HD system? The difference to me is obvious and repeatable.
i agree, but i can't help wondering if it's possible that perception is self-induced.

there's no way to hear 441.1/48kHz and 96kHz simultaneously, so it's impossible to A/B on the same system. could it also be because plugs sound better @ 96kHz? again, no way to A/B using the same system.

but fwiw i think i do hear a clear difference. been doing everything @ 96kHz since it became available, false percepetion or not.
Old 8th April 2007
  #200
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I actually did compare, recording a solo cello, same mic, same pre, split output to 2 ProTools systems similtaneously, one at 44.1 and one 96k. A room full of people all heard a difference in favor of 96k.

I was told that this is because it is easier to manufacturer a good sounding 96k converter than a good one at 44.1.

Am I dreaming?
Old 8th April 2007
  #201
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Quote:
Originally Posted by Wilburguy View Post
I actually did compare, recording a solo cello, same mic, same pre, split output to 2 ProTools systems similtaneously, one at 44.1 and one 96k. A room full of people all heard a difference in favor of 96k.

I was told that this is because it is easier to manufacturer a good sounding 96k converter than a good one at 44.1. Am I dreaming?
wow! 1st time i've heard of someone actually doing this... thanks for posting your results! thumbsup
Old 9th April 2007
  #202
Quote:
Originally Posted by Wilburguy View Post
I actually did compare, recording a solo cello, same mic, same pre, split output to 2 ProTools systems similtaneously, one at 44.1 and one 96k. A room full of people all heard a difference in favor of 96k.

I was told that this is because it is easier to manufacturer a good sounding 96k converter than a good one at 44.1.

Am I dreaming?
Was it a blind test?
Old 9th April 2007
  #203
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Quote:
Originally Posted by bonne View Post
Very useful information, Paul. I can see why you would avoid SRC when possible.

iZotope's 64-bit SRC has a new take on the linear phase/minimal phase dilemma that they call "hybrid ringing control". People who have heard this and compared it to the competition tend to agree that this is (one of) the best sounding SRC algo(s) so far.

Here's a link to a brief description of their design:

http://www.izotope.com/tech/src/
Ok personally I think the 'ringing' issue is a red herring. In all the tests I have ever done (i.e. loads) we cannot hear it at 44.1 and 48KHz. It's a popular issue at the moment, not least of all because making a non-phase corrected (or partially corrected) filter is dramatically cheaper to make! You can see the advantage to people like converter manus if ringing and pre-echo is made a big issue!

What we CAN hear however is phase distortion caused by uncorrected filters - you may like the sound of it, but it's not accurate - obviously it can't be.. The phase error issue is being played down in various publications in order to pave the way for this advantage - a liberty that many converter manus have already taken :-(

And of course we must not forget that if there IS phase inaccuracy it will partially improve at higher sample rates - as the error will reduce. But are you prepared to double your processing costs and halve the power of your system to accomodate them? You can see how promoting such ideas can benefit everyone in the supply chain - potentially at your expense..


To test accuracy you must compare the signal before and after the SRC and listen for ANY differences - not just what may sound more pleasing on some random programme.
Old 9th April 2007
  #204
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Quote:
Originally Posted by Wilburguy View Post
I actually did compare, recording a solo cello, same mic, same pre, split output to 2 ProTools systems similtaneously, one at 44.1 and one 96k. A room full of people all heard a difference in favor of 96k.

I was told that this is because it is easier to manufacturer a good sounding 96k converter than a good one at 44.1.

Am I dreaming?
But which one was correct! Watch it here :-(

Ideally what you need is a system that faithfully reproduces the sound YOU are making. Anything that sounds so different between sample rates has some issues which are potentially modifying your sound in ways you can't control. You cannot know whether these changes are compensating for other issues in your kit or if they are real :-( You cannot know that the higher sample rate is possibly even less accurate!!

Having said that - I have tested PT at different rates using the plugs we designed and as far as I can tell it works. Also I have not encountered a problem with PT converters that modifies sound differently at higher or lower sample rates specifically.

If room full of people actually heard the difference there is a problem somewhere - it should not be so.

When people report this kind of thing the most popular cause is that they are recording at too high levels and the converters are behaving differently to the reconstruction overs depending on sample rate. Try the same test by modulating both at -6dBFS max peak :-)
Old 14th April 2007
  #205
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Quote:
Originally Posted by Paul Frindle View Post
When people report this kind of thing the most popular cause is that they are recording at too high levels and the converters are behaving differently to the reconstruction overs depending on sample rate. Try the same test by modulating both at -6dBFS max peak :-)
As always, excellent post thumbsup

ruudman
Old 16th April 2007
  #206
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Paul,

I have followed this thread with great interest and while I follow your reasoning so far, most folks I have heard from that have tried both 44.1 and 96k agree that recording in ProTools at 96k sounds better than 44.1k. Are you telling me that all these folks are wrong? Are you suggesting that all folks who believe 96k is superior have something wrong with their record levels and or playback systems? I find this very hard to believe...maybe a ProTools thing?

Last edited by Wilburguy; 17th April 2007 at 10:35 PM.. Reason: to sound less argumentative and show more respect to Paul
Old 16th April 2007
  #207
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Well it does seem. that theoretically, higher sample rates should provide an advantage to the look ahead functions in dynamic processors.

I do believe that allot of the percieved sound improvements of HSR were a result of the older converters and that the newer, improved reconstruction methods on a quaity unit would not provide such a discernable difference.

heh
heh heh
heh heh heh
Old 17th April 2007
  #208
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Only the very best converters sound pretty much the same.
Old 18th April 2007
  #209
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Quote:
Originally Posted by Bob Olhsson View Post
Only the very best converters sound pretty much the same.
And altough they are decent, Pro Tools HD converters are NOT in the very best, or even best category!
Old 20th April 2007
  #210
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Quote:
Originally Posted by Wilburguy View Post
Paul,

I have followed this thread with great interest and while I follow your reasoning so far, most folks I have heard from that have tried both 44.1 and 96k agree that recording in ProTools at 96k sounds better than 44.1k. Are you telling me that all these folks are wrong? Are you suggesting that all folks who believe 96k is superior have something wrong with their record levels and or playback systems? I find this very hard to believe...maybe a ProTools thing?
On PT HD I have not been able to find any performance issues in the converters between sampling rates that would significantly affect sound quality..

What I am saying is that with todays overmodulated (and often illegally hot) programme the biggest effect you are likely to hear between sampling rates is in the way the converters respond to these common errors.

If you reduce levels to something more reasonable (i.e. without errors) the differences magically go away. Try it :-) However I realise all too well (to my personal horror) that these days you are not allowed sell anything that isn't overmodulated unless it's a film production. And it's all going to end up on 44.1KHz CD anyway, so ultimately for the final product it may be more a question of how the SRC responds to errors.
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