Reply by Paavo Jumppanen●October 21, 20042004-10-21
Stephan M. Bernsee <spam@dspdimension.com> wrote in message news:<2tp1o6F238b6qU1@uni-berlin.de>...
> On 2004-10-20 23:52:57 +0200, PaavoJumppanen@iname.com (Paavo Jumppanen) said:
>
> > I'd agree that there is a lot more to a good mix than just spectral
> > balance but the area that is most commonly stuffed up in music
> > recordings is the spectral balance, and if you think about it there
> > are very good reasons for this.
>
> I don't agree. I think the use of too may effects like reverb, cheap
> equipment (or using too many of the free effects plug ins on the
> software side) and poor placement of instruments in a mix is about as
> popular as spectral imbalance (to borrow the subject line of another
> topic). To quote a former employee of a German audio software company
> recently sold to a video company "the fact that you can purchase a
> hammer in a store doesn't mean that you can successfully build your own
> house". IOW: even if you have access to the same tools as the "pros" it
> doesn't mean your mix will sound that way. That's the primary reason
> why bad mixes exist in the first place, and it accounts for 90% of the
> income of mastering facilities (I know because I used to work for one).
Yes, and a large reason for that is cos there are many people doing
recording who haven't got a good understanding of what they are doing
and why. A big reason for that is not being able to correlate the wat
things sound and what you need to do to improve it. Our customers tell
us that our product helps them in this regard by better training there
ears, since they learn to correlate certain overall tonalities with
the spectral shapes they see in the analysis.
...snip...
>
> MBC, if used correctly, does balance the overall spectral shape of a
> recording and makes it sound more pleasant. However, it can easily
> squeeze the life out of a recording, too. If I understand correctly,
> your device balances the overall spectrum without that disadvantage
> because it's an EQ and not a dynamics compressor. The snake oil smell
> probably comes from people who fell for the marketing of matched
> filtering devices and software that claims to accurately model
> microphones, or to transfer the sound of one mix to another. This is a
> dangerous claim because in most cases the trick won't work right and
> people are disappointed.
>
> For you, that's just tough luck I think...
You make it sound much worse than it is (ie. the snake oil tag). That
tag has long gone along with any carping that we had, and we have
never taken the line that you should use this software to try and copy
a mix from another track since it is the wrong thing to do. It ignores
the intentions of the producer in coming up with a particular mix in
the first place. That is why our software is "manual adjustment
only". There are no magic one button fixes or presets to go with (much
to the dissappointed insistense of a couple of users). I think a some
of our supposed claims come from reading between the lines things that
were never said in the first place, though, like I made out earlier,
it isn't an issue anymore. It only was for the first few months until
we had a host of positive magazine reviews. Contrary to normal
practice we openly offer a 30 day money back guarantee on our software
and we have a refund rate of 2% or less so I guess most people who buy
it are more than satisfied with how it can help them.
Cheers,
Paavo.
Reply by Stephan M. Bernsee●October 21, 20042004-10-21
On 2004-10-20 23:52:57 +0200, PaavoJumppanen@iname.com (Paavo Jumppanen) said:
> I'd agree that there is a lot more to a good mix than just spectral
> balance but the area that is most commonly stuffed up in music
> recordings is the spectral balance, and if you think about it there
> are very good reasons for this.
I don't agree. I think the use of too may effects like reverb, cheap
equipment (or using too many of the free effects plug ins on the
software side) and poor placement of instruments in a mix is about as
popular as spectral imbalance (to borrow the subject line of another
topic). To quote a former employee of a German audio software company
recently sold to a video company "the fact that you can purchase a
hammer in a store doesn't mean that you can successfully build your own
house". IOW: even if you have access to the same tools as the "pros" it
doesn't mean your mix will sound that way. That's the primary reason
why bad mixes exist in the first place, and it accounts for 90% of the
income of mastering facilities (I know because I used to work for one).
> Another issue is the prevalence of multiband compression in popular
> recordings these days. I've got so many that simply sound bad because
> of compression artifacts. It would appear that much of the problem
> comes from a particular portion of the spectrum dominating and hence
> preferentially being compressed more by multiband compressors,
> resulting in strong changes in instrument colour with time. If the
> balance is more uniform to start off with these side effects don't
> seem to be as pronounced. That being as it may, I wouldn't be
> dissappointed if multiband compression vanished tommorrow but it seems
> to be entrenched in the industry today.
Yes, that's indeed the case. Since the appearance of a very popular MBC
device made by a danish company and its success in the 1990s things
have become a lot worse than before, when only a few engineers had such
a device at their disposal. But we've seen this type of misuse before:
in the 1980s, people came to the mastering facility with tapes or DATs
that were almost unusable due to excessive use of exciters and stereo
width enhancers. In the 1990s the multiband compression era began, and
now it appears we're still right in the middle of the pitch
quantization decade. Of course you could point your finger at me saying
that because I'm working at a company involved in developing these
devices I am partly responsible for this, too (and I wouldn't
disagree), but it's always a problem with people using and mis-using
things. But if it's any comfort, I can hardly turn on the radio these
days without hurting my ears from all the artifacts and effects - if
you're into developing this stuff you tend to develop a
hypersensitivity to them. :-)
MBC, if used correctly, does balance the overall spectral shape of a
recording and makes it sound more pleasant. However, it can easily
squeeze the life out of a recording, too. If I understand correctly,
your device balances the overall spectrum without that disadvantage
because it's an EQ and not a dynamics compressor. The snake oil smell
probably comes from people who fell for the marketing of matched
filtering devices and software that claims to accurately model
microphones, or to transfer the sound of one mix to another. This is a
dangerous claim because in most cases the trick won't work right and
people are disappointed.
For you, that's just tough luck I think...
--
Stephan M. Bernsee
http://www.dspdimension.com
Reply by Paavo Jumppanen●October 20, 20042004-10-20
Stephan M. Bernsee <spam@dspdimension.com> wrote in message news:<2tmhj5F216dceU1@uni-berlin.de>...
> On 2004-10-16 06:51:05 +0200, PaavoJumppanen@iname.com (Paavo Jumppanen) said:
>
> >
> > Thanks Rick,
> >
> > We had a bit of an infamous start about a year ago being branded snake
> > oil merchants.
>
> I must admit I too was about to dig out my popular Furutech CD
> Demagnetizer link again when I first saw the web site... but not!
> Apparently this is an EQ of some sort. Even though the MP3s don't
> really help much (for some tracks, the "before" version sounded better
> than the processed version to me [too much midrange], but I guess
> that's just the normal exaggeration on the demo files to demonstrate
> the effect) I bet it can be used to help correcting problems.
>
> There's a lot more to a good mix than just the proper overall spectral
> balance (stereo placement of instruments/stereo width, proper
> application of reverb, good adjustment of the dynamics etc.) so I
> wouldn't quite agree with all of the claims made on your web site, but
> it's a start and probably a nice tool for people who just want to make
> some quick and easy to set up adjustments.
Marketing is marketing. How often do you read a brochure that doesn't
add some spin.
I'd agree that there is a lot more to a good mix than just spectral
balance but the area that is most commonly stuffed up in music
recordings is the spectral balance, and if you think about it there
are very good reasons for this. Apart from the usual limitations of
human hearing (TTS, masking, loudness effects, fatigue etc) every
recording studio control room will have unique acoustic
characteristics and somewhat unique loudspeaker responses which can
end up colouring the mix for one reason or another leading to a
recording that isn't very transportable. Really good studios will have
well defined acoustics (and perhaps applied EQ) that avoid this issue
but how many really good studios are there in the overall picture.
It's probably only a small fraction and that is certainly reflected in
the quality of my CD collection. In any case, we're not proposing
anything radical hear. EQ in mastering is standard practice and most
commercial CD's these days go through mastering houses. Our software
just makes the process of mastering EQ more objective, more efficient
(time spent wise) and less error prone.
Another issue is the prevalence of multiband compression in popular
recordings these days. I've got so many that simply sound bad because
of compression artifacts. It would appear that much of the problem
comes from a particular portion of the spectrum dominating and hence
preferentially being compressed more by multiband compressors,
resulting in strong changes in instrument colour with time. If the
balance is more uniform to start off with these side effects don't
seem to be as pronounced. That being as it may, I wouldn't be
dissappointed if multiband compression vanished tommorrow but it seems
to be entrenched in the industry today.
Regards,
Paavo.
Reply by Jon Harris●October 20, 20042004-10-20
"Andrew Reilly" <andrew-newspost@areilly.bpc-users.org> wrote in message
news:pan.2004.10.20.04.48.13.320576@areilly.bpc-users.org...
> On Tue, 19 Oct 2004 20:54:00 -0700, Bob Cain wrote:
>
> >> I'm wondering whether it would be useful to have a phase controller in
> >> addition to the standard parametric eq controls. The phase controller
> >> would go from minimum over linear to maximum phase response.
> >
> > I think that is a _very_ good idea. The problem I see is
> > that while those three points are well defined, what you
> > might do between them is enormously flexible. Any ideas
> > that you would be comfortable discussing publicly?
>
> Many years ago ('95-ish, from memory) David McGrath and I presented a
> paper at an AES conference on "optimal phase" FIR EQ synthesis. I think.
> It's certainly been a feature of the EQ software for Huron systems for
> about that long. The user can choose minimum or linear phase, or dial-in
> a desired group delay. I don't think that we ever bothered to allow
> dialing-in group delays that would result in maximum-phase filters though.
> I'll see if I can dig up a paper reference for you. The papers used to
> be on-line on the web site, but I think that they've fallen of in one
> of the re-designs. The implementation is a simple generalization of the
> Hilbert transform process that one usually uses to get a minimum-phase
> impulse response.
> http://www.lake.com.au/driver.asp?page=main/products/huron/modular+software
>
> You might also be interested in Lake's "Contour" product line, which
> includes some fairly fancy arbitrary-curve FIR EQ features, and a neat GUI
> to drive it. http://www.proaudio.lake.com/ [I do work for Lake, but was
> not invloved directly with these particular products, so I'm afraid that I
> can't tell much more about them.]
What I am curious about is how they do the so-called "Mesa" filters, which have
"the ability to separate the sides of a parametric section, change center
frequencies and adjust slopes independently". That has to be more than a simple
biquad per band. Or maybe it is FIR?
http://www.lake.com.au/proaudio/Lake_Mesa.htm
Reply by ho●October 20, 20042004-10-20
Jerry Avins wrote:
> Roman Rumian wrote:
>
>>Hello,
>>
>>maybe my question is stupid, but does such a device/algorithm exist in
>>practice ?
>
> I know what a parametric amplifier is, but not a parametric equalizer.
> There is such a thing, but I'm not familiar with its specs. Matlab code:
> http://www.mathworks.com/matlabcentral/fileexchange/loadFile.do?objectId=1963
>
> How would one change the equalization parameters in an FIR?
A time domain method is to window the the impulse response of an IIR
parametric equalizer. With enough taps, the FIR can approximate many
but not all frequency responses of the IIR. In particular, the FIR
will need to be quite long for low filter frequencies or high Q's, and
combining those two would probably increase the FIR length too much.
Reply by Stephan M. Bernsee●October 20, 20042004-10-20
On 2004-10-16 06:51:05 +0200, PaavoJumppanen@iname.com (Paavo Jumppanen) said:
>
> Thanks Rick,
>
> We had a bit of an infamous start about a year ago being branded snake
> oil merchants.
I must admit I too was about to dig out my popular Furutech CD
Demagnetizer link again when I first saw the web site... but not!
Apparently this is an EQ of some sort. Even though the MP3s don't
really help much (for some tracks, the "before" version sounded better
than the processed version to me [too much midrange], but I guess
that's just the normal exaggeration on the demo files to demonstrate
the effect) I bet it can be used to help correcting problems.
There's a lot more to a good mix than just the proper overall spectral
balance (stereo placement of instruments/stereo width, proper
application of reverb, good adjustment of the dynamics etc.) so I
wouldn't quite agree with all of the claims made on your web site, but
it's a start and probably a nice tool for people who just want to make
some quick and easy to set up adjustments.
--
Stephan M. Bernsee
http://www.dspdimension.com
Reply by Stephan M. Bernsee●October 20, 20042004-10-20
On 2004-10-18 10:48:39 +0200, an2or@mailcircuit.com (Andor) said:
> Roman Rumian wrote:
> ...
>> Many thanks for your explanation - I read this brochure carefully.
>> Have you heard about the similar device, or the EQI-LP is the only one ?
>
> It seems every harddisk editor and plug-in manufacturer is nowadays
> offering a linear-phase eq. Google for linear phase eq. I don't know
> of any other hardware eq though.
>
> Regards,
> Andor
Well, yes, and to what end. If you ask me, it's because a linear phase
EQ is easy to design (from a non-DSP developer's perspective who thinks
poles and zeros are nationalities) - "FFT and multiply" is fairly easy
to understand. Not forgetting that today processor power is no longer
an issue, so FFT away and be done with it. Linear phase EQs also have
the desirable property of sounding good wrt. marketing (not necessarily
wrt. the sonic results). The "linear" buzzword is always good for
sales! :-)
If you really compare a good non-linear phase EQ (all analog EQs I know
are of this type) to any of the recent linear phase EQ you'll soon
realize that linearity doesn't make it sound good. In fact, most linear
phase EQs sound horrible. I for one would rather spend my money on
stuff that sounds good, be it linear, non-linear or else.
In a way, this reminds me of the discussions I've seen in the days when
the CD was something new: is "DDD" better than "AAD"? Discuss! :-)
And in the end, the final result is presented to us through a highly
non-linear device anyway: our very ears!
--
Stephan M. Bernsee
http://www.dspdimension.com
Reply by Andrew Reilly●October 20, 20042004-10-20
On Tue, 19 Oct 2004 20:54:00 -0700, Bob Cain wrote:
>> I'm wondering whether it would be useful to have a phase controller in
>> addition to the standard parametric eq controls. The phase controller
>> would go from minimum over linear to maximum phase response.
>
> I think that is a _very_ good idea. The problem I see is
> that while those three points are well defined, what you
> might do between them is enormously flexible. Any ideas
> that you would be comfortable discussing publicly?
Many years ago ('95-ish, from memory) David McGrath and I presented a
paper at an AES conference on "optimal phase" FIR EQ synthesis. I think.
It's certainly been a feature of the EQ software for Huron systems for
about that long. The user can choose minimum or linear phase, or dial-in
a desired group delay. I don't think that we ever bothered to allow
dialing-in group delays that would result in maximum-phase filters though.
I'll see if I can dig up a paper reference for you. The papers used to
be on-line on the web site, but I think that they've fallen of in one
of the re-designs. The implementation is a simple generalization of the
Hilbert transform process that one usually uses to get a minimum-phase
impulse response.
http://www.lake.com.au/driver.asp?page=main/products/huron/modular+software
You might also be interested in Lake's "Contour" product line, which
includes some fairly fancy arbitrary-curve FIR EQ features, and a neat GUI
to drive it. http://www.proaudio.lake.com/ [I do work for Lake, but was
not invloved directly with these particular products, so I'm afraid that I
can't tell much more about them.]
--
Andrew (also A.Reilly at lake.com)
Reply by Bob Cain●October 20, 20042004-10-20
Andor wrote:
> Perhaps you are hearing some
> short-comings of the implementation.
That's always possible, of course, but calculations for
deriving the filter are all long floating point and the
resulting impulse responses, at short floating point
precision, pass the tests I can think of to test for linear
phase and minimum phase using Matlab as the analytic tool.
> I would strongly suggest you try some other product (ours? :-). At
> least, I would consider to try a parametric linear-phase eq. In my
> opinion, constant Q graphic EQ is ergonomically unsuited for audio
> work (a debatable matter of taste, I agree).
I agree with you.
My goal was a curve based equalizer using various selectable
control point algorithms for specifying an arbitrary
magnitude function. This is quite different than what I
understand a parametric eq to be.
The ability to switch between linear and minimum phase
realizations was as much for my own edification as for the
end product. As I said, and in considerable disagreement
with Gerzon as to the audibility of very minor phase
changes, I find to difference to be subtle at best and at
the edge of my ability to discriminate. Yet, for me, there
is a clear subjective preference for the minimum phase that
I find very difficult to characterize. It's about as
elusive as the idea of listening fatigue which I know is a
weak fallback. I must plead guilty also to not putting my
discrimination yet to the ABX test.
>
> Bob, I know your reservation towards linear-phase eq with regards to
> pre-ringing.
Yes, and Gerzon (from the excellent discussion you link to
below) said it well before I began to even think about it,
"Everything that is known about the way the ears perceive
transients suggests that, all other things being equal, a
pre-response (ie before the main impulse) in a filter will
have more audible effect than a similar mirror-image
post-response after the main impulse. This is not just
consistent with Lagadec�s findings on his digital filter,
but is also consistent with the Haas Effect, whereby
transient sounds tend to be preferentially localised by the
transient arriving at the ear first, with later transients
(up to about 40ms later, when separate echoes are heard)
playing a reduced role. This is also consistent with the
physiological effect of forward inhibition or temporal
masking, whereby the perception of stimuli tends to suppress
or reduce the sensitivity to the perception of stimuli
following immediately afterwards."
> From
> what I gather, our linear-phase eq customers are correcting tracks in
> a way which is impossible for them with minimum-phase (the EQ1-LP is
> switchable between linear- and minimum-phase).
Well, there's no arguing with subjective impression, I have
just found my own to be different.
>
> Some time ago, there was a discussion here whether in fact the optimal
> phase response was somewhere in between linear and minium (or perhaps
> even maximum?). I base this assumption on an article by Michael
> Gerzon:
> http://www.audiosignal.co.uk/Why%20do%20equalisers%20sound%20different.html
>
> What do you think of this?
I think he covers all the various considerations in the
fashion that can be expected of Gerzon. Few can do it
better, AFAIC. I am surprised at his emphasis on the
audibility of minute effects. My ears aren't that good.
This does correlate well, though, with the interminable
discussions I've seen among A/D designers and high end
production people as to the effects of the front end filter.
They claim audibility of ridiculously small effects near
Nyquist (but what may be telling is that none seem willing
to put their discrimination ability to the ABX test.)
I must disagree with the following observation of Gerzon's
based on a good deal of work with measurement based speaker
and microphone transformations:
"One area of pessimism concerns the viability of using
equalisers to compensate for defects in other equipment
(microphones, loudspeakers and even multiple stages of bass
roll-off in audio electronics). The problem here is that
even very tiny residual errors in the frequency and phase
responses may turn out to be almost as audible (or in some
cases even more so) than the original unequalised errors.
Equalisation may improve the tonal accuracy in such cases
but it can (and often does) increase the audible colouration."
It is always dangerous to disagree with Micheal Gerzon,
though, and I well know that. :-)
I wasn't aware of that archive of his work and want to thank
you for the link. I'm working currently on a method for
calculating encoders empirically, based on measurement, for
the tetrahedral array he invented and there is much there
that I want to read.
> I'm wondering whether it would be useful to have a phase controller in
> addition to the standard parametric eq controls. The phase controller
> would go from minimum over linear to maximum phase response.
I think that is a _very_ good idea. The problem I see is
that while those three points are well defined, what you
might do between them is enormously flexible. Any ideas
that you would be comfortable discussing publicly?
Thanks,
Bob
--
"Things should be described as simply as possible, but no
simpler."
A. Einstein
Reply by Jerry Avins●October 19, 20042004-10-19
Richard Owlett wrote:
> Andor wrote:
>
>> [ *SNIP* ]
>
>> In my opinion, constant Q graphic EQ is ergonomically unsuited
>> for audio work (a debatable matter of taste, I agree).
>
>>
>
> /preamble
> Admitting that someone(s) will be able to say "I/We told you so" ;)
> I'll dive into my question.
>
> My interest in DSP derives from my interest in improving robustness of
> restricted forms of speech recognition when subject to restricted forms
> of acoustic interference. I know just enough of both fields to get me in
> trouble.
> /end preamble
>
> I wish to manipulate an audio stream (speech), sampled at 44 kHz.
> Real time is *NOT* required - source is a CD.
>
> Is there a filter (FIR/IIR/other) such that given ONE input producing
> SIX outputs
>
> 1. All outputs have the same time delay (measured in sample clocks)
> 1.a. Specifically number of calculations for each output NOT relevant
> 2. Q >=10 and easily adjustable
> 3. center frequencies will vary over ~< 20:1 range
> 4. available as Scilab (or equiv ) code
>
> Bonus Question
>
> What should I have been asking for?
>
> BTW I've discovered a similarity between "teaching DSP" and "instructing
> a child in theology" -- answer only question asked ;}
If you need a filter with *the* delay, then you need a linear-phase
filter, which pretty much (but not absolutely) specifies a symmetric
FIR. Getting all the filters with the same delay is then easy; just make
them the same length. The other details I leave to you.
Jerry
--
Engineering is the art of making what you want from things you can get.
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