Comments on: Avoiding inter sample peaks

By: Keith

Keith — Fri, 23 Sep 2011 23:23:11 +0000

This is groundbreaking. I recorded an album on great gear, had it mastered. The whole process wasn’t ideal, but I was never happy with the final product. I always felt that there were parts in the songs that sounded “pop and click”-y….

It had to be these intersample peaks. Your example is just like my album. I continue to learn!

By: david

david — Wed, 07 Sep 2011 21:41:46 +0000

Interesting read. I see where mike S is coming from and if you do compress everything and make your breakdowns compressed and normalised as well as the main part of the track it can sound too loud all of the time. And even though yes this kind of production will give rise to the probable chance of more inter sample peaks i still dont see why the producer should have to worry about them because its the designers of the DAC that should sort it out. Even if a track not involved in the ‘loudness war’ i.e. a classical recording was normalised that could still contain these clippings too, granted not as much of course but normalising a whole track to 0.db is a must at the end of a track at least just so when listening to lots of songs u dont have to fiddle with the output level every 2 minutes. Yes each song is percieved louder but at least if its normalised we have a bit more constance overall. Plus if the peaks on any given track where that noticeable to the audience listener am sure they would turn their system down anyway.

By: coriolis

coriolis — Tue, 16 Aug 2011 18:12:06 +0000

Very informative – and a bit confusing after reading some of the comments.
I don’t know how modern DAC’s work in practice, but I have certainly heard
recordings with way too many overs to be listenable.

An acoustical engineer-friend of mine had a rather sophisticated hifi preamp
once, which blinked an led every time clipping occurred on the digital input.
Listening to the debut album by The Darkness, it was on a lot of the time.
Great songs, but horrible sounding recording because of the clipping…

By: mike spitzer

mike spitzer — Sun, 10 Oct 2010 00:17:28 +0000

It has been said before by guys like Bob Katz and Mr. Ludwig but the best way to avoid these problems is ……… AVOID THE LOUDNESS WARS !!!

All these CDs these days that are squashed to a constant square wave maxxed out to a constant 0.00 sound LOUD, but they sound BAD.

Look at your mastered WAV files …….. if they look like a big rectangular box and the meters don’t move more than 0.5 db during the whole song, then you are overcompressing and limiting the life out of your music.

Let the music keep some dynamics and BREATHE some up and down …..
Real music needs to have some dynamics of AT LEAST 3-6 db.

Yes, that means you may have to turn the volume knob on your stereo up a notch to be as loud as some commercial releases — but at least it will sound good and not sound like a clicky, poppy. wall of constant noise

By: da professor

da professor — Mon, 20 Sep 2010 16:42:43 +0000

Mr. Sixpack is correct.

We are talking two different domains.
If you sampled at greater than twice the highest frequency
then you *could* recreate the original signal exactly.

And that is what is happening!!
You get the ANALOG peaks higher than the digital because they were higher originally.

There is no guarantee that your samples will all be at the highest analog peaks. In general most samples will be well below the peak, with an occasional sample perhaps being close to the highest analog value. The higher the sampling rate the more samples that will be near the peaks though. And the less this impact (alleged) “problem” will have.

Now , to be fair, in practice you need at least 10x as many samples as nyquist said was mathematically sufficient. This is a practical engineering limitation.

As is the analog domain after D/A. A good design will allow for those peaks because they are a normal phenomenon. If there is not enough headroom then back off the gain on the digital signal so you do get a clean analog output. Then turn your amp knob to the right and make it as loud as you want.

By: joe sixpak

joe sixpak — Fri, 17 Sep 2010 19:10:54 +0000

intersample peaks are a meaningless red herring !!!

if your analog amplifier has headroom then you get a great result no matter what the blather on the various bbs etc. try to tell you. and most of those idiots never had a real sampled digital system course in a real grad school.

if your analog out cant handle the signal then even peak digital may cause problems — it really depends on how hot you normalised the digital signal.

you can always eliminate the alleged problem by lowering the digital volume level and turning the output amp knob to the right. no distortion , still loud enough to make your ears bleed.

this should all have been handled by teh people designing the a/d/a chain. so you dont need to worry.

now if you have two different a/d and d/a boxes then they may not match. so just turn the digital level down and be safe.

By: 10 Myths About Normalization | homerecordingartist.info

10 Myths About Normalization | homerecordingartist.info — Tue, 25 May 2010 01:37:29 +0000

[...] Normalizing to near 0dbfs can introduce inter sample peaks. [...]

By: Adam!

Adam! — Tue, 18 Dec 2007 05:43:44 +0000

Heres a (late) question: if the target format is MP3, does explicitly avoiding inter-sample overs help? The reason I ask is because it was my impression that the mp3 decoding process changes the waveform, introducing new inter-sample overs. Infact the mp3 example you posted, once decoded to wav, actually has regular overs, which have been induced by the decoding process.

Surely keeping the track out of the red is good advice regardless, but I wonder if having zero inter-sample overs in your mixed down WAV means that an mp3 encoding of that WAV will also be free of them?

By: des

des — Thu, 15 Nov 2007 14:23:10 +0000

> If it goes above red then I reduce the main volume. A sound practice :-) > I am having a hard time believing they could be as much as 1.0 dB, though. The Neilsen/Lund paper that lays out the theory for this shows a contrived example where the inter sample peaks are 6dB (!!!) higher than the sampled level. They go on to say that many mastering engineers agree 3dB is a safe, conservative allowance. (And 1dB, like I said above, is the general consensus on the web, and among my engineer friends.) Here's the paper, if you want more details: http://www.tcelectronic.com/media/Level_paper_AES109.pdf

By: Michael

Michael — Wed, 14 Nov 2007 20:56:47 +0000

Thanks for this, so often I hear digital anomalies. but there are so few explanations as to where they are coming from. I am having a hard time believing they could be as much as 1.0 dB, though. That is a very large interpolation for any DAC to make.