27th July 2011
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#91 | | Lives for gear
Joined: Dec 2007
Posts: 886
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Originally Posted by espasonico I mean, if you limit a peak of 3dB, then, the intersample peak is gonna be around +3db ? | No.
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27th July 2011
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#92 | | Lives for gear
Joined: May 2007 Location: Los Angeles
Posts: 1,091
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You don't have to worry about inter-sample peaks. We are talking about a +20 work environment here (it's not a music master). If your peaks are hitting the network peak limit & your mix is at the industry average of -24 LKFS, Something is wrong with your mix. That would be one hell of a dynamic peak & should be controlled.
FYI - You can set your DDM to read "true peak".
__________________ "Any experiment of interest in life will be carried out at your own expense." |
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27th July 2011
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#93 | | 3 + infractions, forum membership suspended.
Joined: Jun 2011 Location: at home
Posts: 2,404
| Quote:
Originally Posted by Alexey Lukin You are probably right. Unfortunately, since in some special cases these analog peak levels can go very much beyond digital peak levels, these "properly designed" converters most likely do not exist.
And mastering is all about making recordings sound good on real-world playback systems, not on some ideal systems. | i say that they do -
and the problem is elsewhere
(unless you also have a poorly done d/a system)
analog "peak" always goes higher than most digital samples (would have been if that had been the original sampled analog signal)
that is just nyquist law in action
i can show examples where they can go 60dB and even higher
like i said earlier
use minimum samples eg 2x+epsilon you will have problems
but oversample 10-20x or more and the problem goes away
now if you do nonlinear processing (compress, clip, limit, yada yada) you need to low pass filter the digital before sending it to teh d/a
or else nyquist theorem will not apply
why not lower the digital level
and boost it after d/a if you want a louder signal ?
this so called problem should be fixed in the analog domain where you can control it (and see it precisely)
trying to do it in the digital domain with invalid signals is like pushing on a rope
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27th July 2011
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#94 | | Lives for gear
Joined: Dec 2007
Posts: 886
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Sorry, this doesn't make any sense to me.
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27th July 2011
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#95 | | 3 + infractions, forum membership suspended.
Joined: Jun 2011 Location: at home
Posts: 2,404
| Quote:
Originally Posted by Alexey Lukin Sorry, this doesn't make any sense to me. | i do believe that you do not understand it
it would be worthwhile for you to try to figure it out
there may be a problem
i believe some people have a problme
BUT
it is not any alleged intersample peak problem
that may be a symptom that is misinterpreted
it is not a problem
and certainly not the root cause of the problem
badly designed d/a converter system may be a problem
non linear processing that negates nyquist theorem is a problem
people who dont know what they are talkign about but just parrot every bit of wizdumb they read on the internet is a problem
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27th July 2011
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#96 | | Gear maniac
Joined: Jun 2006 Location: rio de janeiro | |
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27th July 2011
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#97 | | Lives for gear
Joined: May 2005
Posts: 928
| Quote:
Originally Posted by oldeanalogueguy i do believe that you do not understand it
it would be worthwhile for you to try to figure it out
there may be a problem
i believe some people have a problme
BUT
it is not any alleged intersample peak problem
that may be a symptom that is misinterpreted
it is not a problem
and certainly not the root cause of the problem
badly designed d/a converter system may be a problem
non linear processing that negates nyquist theorem is a problem
people who dont know what they are talkign about but just parrot every bit of wizdumb they read on the internet is a problem | Pardon me for being blunt, but where are your real-world apps that assist us in making a living and why don't you use your real name here? It's getting irritating reading a discussion that seems to be drifting into troll territory.
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27th July 2011
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#98 | | Lives for experience
Joined: Nov 2004 Location: minneapolis, mn | Quote:
Originally Posted by kk@jamsync.com Pardon me for being blunt, but where are your real-world apps that assist us in making a living and why don't you use your real name here? It's getting irritating reading a discussion that seems to be drifting into troll territory. | Drifting?
He provides no proof what so ever of his views, only convoluted hard to understand reasoning based on decades old research. Yet he attempts to take to task TC Electronic for merely making assertions. Nielson and Lund's papers are filled with examples and explanations what warrant the assertability of their positions. He provides nothing but smoke screens.
Paul Frindle, designer of the Oxford Console's DAC, acknowledges the reality of and tries to educate us about Intersample Peaks. Dan Lavry who designs his own DAC aknowledges the problem. Other designers currently working on yet to be released DAC's acknowledge the issue. Alistair, who clearly knows his digital theory, explains it to him. Alexey Lukin, who is both learned and very smart as well as a designer, provides him with clear examples and proof. And yet, Oldeanalogueguy even recommends to newbies that they read what is in his view a " good book" on Mastering, citing the Bob Katz's Mastering Audio. In Katz's book he talks about it, you can find it in the index. Even on Digido.com (Bob's Website) there is an article about it!
Drifting?
Here are a couple of links for the curious: http://www.digido.com/more-bits-please.html
Scroll down the page for Dan Lavry's response here: http://recforums.prosoundweb.com/index.php/t/4140/0/ Do inter-sample peaks only matter during D/A (Mr Frindle?)? |
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27th July 2011
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#99 | | mymixisbetterthanyours!
Joined: Oct 2006 Location: Berlin
Posts: 2,133
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anyone in mastering knows that intersample peaks exist and can actually be a problem for the DACs in consumer equipment.
If you want to get rid of them, the BW2-limiter in the S6000 is a good option. Many post-facilites will have an S6000 anyway.
For Nuendo-users there's a free plugin from SSL. On PT I don't know.
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27th July 2011
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#100 | | Lives for gear
Joined: Mar 2006 Location: the Netherlands
Posts: 5,037
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Originally Posted by oldeanalogueguy i do believe that you do not understand it | He is just being polite. This has lasted long enough so I won't be: You don't understand digital audio nor are you smart enough to realise your own ignorance. Go away, you are merely a distraction in an otherwise informative technical thread.
Alistair
__________________ Alistair Johnston - TV & Film Post, Mastering, Sound Design
-- "The first principle is that you must not fool yourself -- and you are the easiest person to fool" -- Richard P. Feynman "There's a sucker born every minute" -- P.T. Barnum |
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27th July 2011
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#101 | | Lives for gear
Joined: Dec 2005 Location: San Francisco area
Posts: 2,929
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Meanwhile--happy to report that Elephant is containing all the peaks (in 8X) quite well, by my various metering devices (incl LM100).
phil p
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27th July 2011
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#102 | | Lives for gear
Joined: Mar 2006 Location: the Netherlands
Posts: 5,037
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Originally Posted by philper Meanwhile--happy to report that Elephant is containing all the peaks (in 8X) quite well, by my various metering devices (incl LM100).
phil p | Oh excellent. Are you using it with the VST-RTAS wrapper?
Alistair
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28th July 2011
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#103 | | 3 + infractions, forum membership suspended.
Joined: Jun 2011 Location: at home
Posts: 2,404
| Quote:
Originally Posted by UnderTow He is just being polite. This has lasted long enough so I won't be: You don't understand digital audio nor are you smart enough to realise your own ignorance. Go away, you are merely a distraction in an otherwise informative technical thread.
Alistair | i understand digital audio quite well
i also know the crapp,urban myths, what high school seniors know and just plain internet wizdumb that is wrong.
if you can't see the problems with all those references people keep pointing to then you need to go back to school and really learn digital methods this time.
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28th July 2011
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#104 | | Lives for gear
Joined: Dec 2005 Location: San Francisco area
Posts: 2,929
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Originally Posted by UnderTow Oh excellent. Are you using it with the VST-RTAS wrapper?
Alistair | Nope, full-strength VST.
phil p
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28th July 2011
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#105 | | 3 + infractions, forum membership suspended.
Joined: Jun 2011 Location: at home
Posts: 2,404
| Quote:
Originally Posted by minister Drifting?
He provides no proof what so ever of his views, only convoluted hard to understand reasoning based on decades old research. Yet he attempts to take to task TC Electronic for merely making assertions. Nielson and Lund's papers are filled with examples and explanations what warrant the assertability of their positions. He provides nothing but smoke screens.
Paul Frindle, designer of the Oxford Console's DAC, acknowledges the reality of and tries to educate us about Intersample Peaks. Dan Lavry who designs his own DAC aknowledges the problem. Other designers currently working on yet to be released DAC's acknowledge the issue. Alistair, who clearly knows his digital theory, explains it to him. Alexey Lukin, who is both learned and very smart as well as a designer, provides him with clear examples and proof. And yet, Oldeanalogueguy even recommends to newbies that they read what is in his view a " good book" on Mastering, citing the Bob Katz's Mastering Audio. In Katz's book he talks about it, you can find it in the index. Even on Digido.com (Bob's Website) there is an article about it!
Drifting?
Here are a couple of links for the curious: More Bits Please!
Scroll down the page for Dan Lavry's response here: PSW Recording Forums: Dan Lavry => intersample peaks Do inter-sample peaks only matter during D/A (Mr Frindle?)? | rotflamopimp
read the reply at the pro forum
it says its the d/a
and it should be able to handle it
(if designed correctly i note editorially)
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30th July 2011
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#106 | | 3 + infractions, forum membership suspended.
Joined: Jun 2011 Location: at home
Posts: 2,404
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thanks to all for the pointers and now
having read the fine literature
my claim has been proven
by those with facts in spite of many unfounded claims
if you have intersample peaks
either your a/d d/a was badly designed
or
you have created an "illegal" signal (ie does not meet nyquist bandlimited criteria) with nonlinear processing which is a stupidity problem not an alleged intersample peak problem
and as one site notes - you are running everything way too hot
in the digital domain which causes other problems - so back off
there are better ways to make things loud
and like Bob Katz said you should make it gooder not louder or something similar to taht
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4th April 2012
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#107 | | Lives for gear
Joined: Jul 2002 Location: CT
Posts: 976
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was there ever a clear solution to this problem? -24 lkfs? I need to output 13 episodes of concert TV. Any ideas? Thanks. I need to stay within the rigid specs that have been referred to in parts of the thread. Thanks!
chap
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4th April 2012
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#108 | | Lives for gear
Joined: May 2009
Posts: 517
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G-clip.
Clip at 99%
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4th April 2012
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#109 | | Lives for gear
Joined: Dec 2007
Posts: 886
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Solution to which problem?
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4th April 2012
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#110 | | Lives for gear
Joined: May 2009
Posts: 517
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To intersample peaks.
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4th April 2012
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#111 | | Lives for gear
Joined: Mar 2006 Location: the Netherlands
Posts: 5,037
| Quote:
Originally Posted by voidar To intersample peaks. | There are many solutions. What is the context?
Alistair
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5th April 2012
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#112 | | Gear Head
Joined: Apr 2010
Posts: 71
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I've seen a few threads on other forums here at Gearslutz where oldanalogueguy has tried to peddle his unique version of digital audio theory.
Oldanalogueguy - You are completely entitled to interpret digital audio theory however you want but you are really out of your depth on this forum because you have absolutely no idea about the workflows or specifications to which we work. In TV broadcast there is a correlation between digital an analogue because this industry works on calibrated systems. And, your discussions about DACs are completely irrelevant, what have DACs (and their architecture) got to do with delivery specifications? This is a serious and very useful thread on the practical implementation of new laws and regulations governing our specific industry, so please go and troll elsewhere.
Now back to our normal programming: Has anyone tried the Compassion Compressor (DMG Audio)? It has the following relevant selectable parameters: Look ahead processing, Oversampling and Brickwall Limiting. In theory this should make it suitable for the task. Anyone tried it?
G
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5th April 2012
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#113 | | Gear Head
Joined: Jan 2008 Location: Studio City, CA
Posts: 46
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I've been using The Flux Elixir plugin on my deliverables to satisfy CALM. Works great. TruePeak. Verified with the the Nugen Meter. Elixir |
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6th April 2012
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#114 | | Lives for gear
Joined: Jun 2002 Location: LA, USA | Quote:
Originally Posted by AlexeyMohr 2) we're getting push-back from our mixers for over-limiting their mixes in layback, and I can't blame them. I wouldn't want my mix to be squashed any more than necessary either.
. |
I just started reading this thread after it surfaced.
My response to the above, is to tell the mixers to start mixing to spec, and their mixes won't be over-limited and over compressed.
It's part of the gig.
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7th April 2012
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#115 | | Lives for gear
Joined: Dec 2002 Location: U.K
Posts: 2,019
| Quote:
Originally Posted by piratepost I've been using The Flux Elixir plugin on my deliverables to satisfy CALM. Works great. TruePeak. Verified with the the Nugen Meter. Elixir | Yes - I have in intersample peak meter and correcter I use all the time when mastering. We made it some time ago but never got around to selling it, what with all the hassles of 64 bits etc..
Fact is that not only do some DACs mess up - also the SRCs contained within DAC products these days (in more expensive ones) mess up too, even before the signal gets to the DAC!.
Remember also that MP3 coding and decoding (and all other types of perceptual coding) increases peak levels too.
Now you have to be more careful than ever before :-(
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7th April 2012
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#116 | | Lives for gear
Joined: May 2009 Location: hannover, germany
Posts: 867
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+1 on the FLUX Elixir, does a great job here.
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7th April 2012
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#117 | | Gear interested
Joined: Apr 2012 Location: Paris, France
Posts: 17
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Indeed Elixir does a great job and is not so expensive. But eats a lot of CPU. It depends the computer and the DAW but hard to use in real time.
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8th April 2012
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#118 | | Lives for gear
Joined: Apr 2005 Location: amsterdam | Quote:
Originally Posted by rhythminmind You don't have to worry about inter-sample peaks. We are talking about a +20 work environment here (it's not a music master). If your peaks are hitting the network peak limit & your mix is at the industry average of -24 LKFS, Something is wrong with your mix. That would be one hell of a dynamic peak & should be controlled.
FYI - You can set your DDM to read "true peak". | I agree, I mix live television, the most uncontrolled of environments, and I don't feel the need for a true peak limiter.. I never reach close to -3dBFs, and just for safety I brickwall at -4 or so..
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8th April 2012
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#119 | | Lives for gear
Joined: Dec 2002 Location: Germany
Posts: 638
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Originally Posted by Denis Goekdag +1 on the FLUX Elixir, does a great job here. | +1 here too ... it's just that the email support of Flux is really bad. I reported a
problem with Elixir and PTHD 9.05 and got a 'reply' five weeks later to report more in detail ... yeah right ...
Nevertheless I use their Pure Analyzer System and it's very good. Although it has it quirks too but mostly I can work around them. Stopped asking support with that too. Started to see Flux as some being I have no influence on as a customer whatsoever - sad.
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8th April 2012
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#120 | | Gear Head
Joined: Apr 2010
Posts: 71
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Paul Frindle and Paradox Uncr8ted:
In TV broadcast we've never been too bothered about inter-sample peaks because the specs we worked to usually resulted in peaks no higher than -10dBFS. The new North American (ATSC A/85) and European (EBU-R128) specifications both specify new peak levels, expressed in true peak, as calculated by compliant meters.
Although of passing interest, we're not really bothered by the fine detail of what happens in the SRC or the reconstruction filters in DACs but our reputations and jobs depend on us being able to measure and confine our mixes to the loudness and true peak levels as measured by meters compliant with the appropriate specification. It maybe, after going into the fine programming detail of these meters that some bright spark notices they don't actually calculate true peak precisely. But to us, that's largely irrelevant, our job is to get through quality control, as determined by some intern running software with the same metering algorithms as we're using. Hence this thread, to discuss limiters which will brickwall limit to the true peak specifications of our meters.
BTW, if you're interested, the tech docs can be found here: EBU-R128 (website). ATSC A/85 (pdf)
Both of these are based on the ITU-R-BS1770 specification.
G
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