I think I finally understand why some music that gets uplouded to youtube appeared to me as if the "high end" was shaven off a bit. It was probably the ISPs that were cut off. We did a test a while back and uplouded 2 of the same masters just that one was -1.0 dbTP the other -0.2dBTP and the version with -1.0 dBTP was significantly better sounding (also had more transient punch). I suspect now that since the track wasn't ISP limited, that this was probably one of the things I was hearing. Thank you so much for your videos and in also in general! Your information has helped me numerous times and I can always count on you providing clear and concise answers!

The worst part about this is that even though we know this, I've had to deliver files I know full well will suffer the consequences. - the fact that so many favourite artists/genres/etc are all peaking in this way leads sooooo many artists to insist that their tracks be mastered the same way, no matter what advice or recommendations we can make. Of course it's all relative, but I've delivered files for very very happy customers that will surely have ISPs. We do what we can.

I really enjoyed this video! Great info as always! I always try to keep my recordings as "clean" as possible so this def helps keep a few important things in mind when mastering.

The Anti Streaky

Great Video! This was really informative and interesting. And good on you for doing the random listening test!

Thank you for this. A "professional" mastering engineer said in a forum that inter-sample distortion was total bullsh*t and I was an idiot for thinking people can hear it. He actually got quite abusive and personal. But even people who think they can't hear it will admit there's a kind of fuzz if audio with inter-sample distortion goes through data compression (e.g., MP3).

Informative video! The differences are clearly audible (to my ears, anyway) and reinforces that paying attention to detail, critical listening, and vigilance are crucial for delivering outstanding masters. Thanks so much for sharing! -chaz

Love your videos thank you so much for making them!

It blows my mind that professional engineers are doing the number 1 thing i thought was mixing and mastering 101: don’t allow anything to go over 0db. Very informative video👍🏼

The video is interesting. I used to have my WIndows set to 100% for a logical reason, one that people have told me in the past: it sends all of the bits to the DAC. Thanks to this video, I might try setting Windows to 50-90%, currently trying 50% first. 50% would mean I have an output exactly at 23 bits in theory (24-bit data divided by 2 means you just drop the last bit off the end. 51-99% is still 24-bit, but the audio won't be adjusted by an integer scaling). It does seem easier on my ears if I set Windows to 50% volume, but that could just be psychological. When I had it at 100%, it sounded more brickwalled! I just tried this today after seeing this video. I also noticed that Windows had effects on, which made the brickwall sounding worse. One thing about 32-bit audio: it can go past the usual peak. However, most DACs are 24-bit or 16-bit. Want to see what I mean? Take some audio, intentionally peak it, and then save it as 32-bit floating-point uncompressed. Load it back and then normalize. You often have the original sound data. It would be an advantage to 32-bit DACs, able to handle these peaks without clipping. Windows internally processes audio at 32-bit if it needs to process it.

Great video! Kudos for doing the blind tests. Hofa blind test and Perception AB are an indispensable pair for me doing ozone development!

I always wanted to do a video about that myself! Thanks Ian for saving me time😜🥃 - Brilliantly explained, as always!

Great test, thank you so much for the video! I could clearly hear the difference, and I surely prefer non-cliped version. I think ISPs should be a thing to be aware of. I personally always use final true-peak limiting on my clients' master files. It happens to be a good practice.

It hasn't changed my mind because I always leave 1 db of headroom and have done similar tests, but the way you explain this just made is that much clearer to me! Also love that you didn't use the true peak on the SIR that s also what I do, always use a clean limiter after the SIR clipper just for safety!

Thanks for the video. How about uploading these two song clips you compared, say to Dropbox or some other site, so we can then download and hear them without KZfaq compression?

Am interesting test might be to play something back which contains a lot of intersample peaks and record the output to examine what a given DAC is doing with those peaks. Or multiple DACs to see if any handle them in a different fashion. Do any/all of them actually clip at full scale or do any have some headroom to account for this?

Very informative, thank you, Ian!

Thanks for the very informative video Ian. Much enjoyed. Two things i take away from this video 1) Why in Gods name does anyone „need“ to ride levels so near the ceiling. Just cut the music some slavk everyone! 2) Gee this song sounds so flat and stressful that i dont think i could listen to the whole length . The drums sound like its a drumcomputer playing. Everything squeezed out. Why oh why….. and i love the red hot‘s but this style of madtering is such a disaster As to your question… well it goes hand in hand. The ones that risk I.P are usually things that are mastered superhot anyway so Usually the ones causing trouble are „usual suspects“ in a way. So i dont like their sound way before i would ask myself „hmm could it have I.P issues?“ :-)

So I bounced a track to tape and then went through Mix:analog's AD converter. Nothing then should exceed 0dB in the new digital track. Is it then necessary to turn the song down an entire dB or can I get away with less reduction, say, -0.3?

Great video and audio examples! I do care about them. Most of the time, clients have preferred nonclipped versions, except for two times that I recall. One where the well-produced, intentionally clipped version actually sounded better than the nonclipped version on Soundcloud's crap 64kbps codec, even when doing test exports with increasingly lower bit-rate codecs. The other time, a client preferred the more clipped version, even though there were audible problems with it, so it is what it is.

This was a really great video with a lot of great analysis. One thing I wonder though about the encoding clipping is whether it’s actually directly correlated to the intersample peaks? In other words, isn’t it possible that you can have a ISP of say -0.01dB and still get clipping after encoding losses?

That hint hahaha thx for the video Ian!

My understanding, and what correalates to what I hear, is that an ISP that clips causes a wide-band impulse of intermodulation distortion which renders as a kind of noise. To me what is important is what I call "ISP velocity" or "density" - the number of ISP clips per unit time. The more there are more inter-mod distortion there is. My take on this is that this causes a kind of low level "mud" which can rob PLR by increasing over-all loudness. This robs dynamics and can dull transients. I worked with LVC audio to have his ISP meters show "velocity" or "density", Youlean also shows this with red dots at the top of the timeline. Another nice tool is Apple's RoundTrip but it's AU only. It will actually count the ISP clips using different codecs. In my experiments I have found, for my music, that -16 LUFS gives a good balance of loudness vs threat of ISPs. It does mean the trakcs are lower in level but it minimizes the chance of up-stream clipping because of ISP, which is important to me. But If loud is your thing, just make sure you use a good tool because its not always subjectively easy to hear it. It's one of those digital things that benefits from digital tools to measure it to eliminate it.

Something I don't quite understand is where in the D/A does it clip, is it the analog section or in the actual conversion process? I'd assume the reason some people don't hear it is because certain D/A would have higher than 0dbfs headroom but consumer don't which is why maybe engineers don't hear it when rendering. Obviously the main problem is lossy codecs because that creates digital peaks. I have heard plenty of soundcloud posts of songs which sounds clipped to hell, but then the wav beatport release sounds fine, I assume this is because of intersample peaks or worse just peaks in general.

Its worth clarifying that each sample has a maximum numerical value due to the bit depth, so a "clipped" sample peak is not playing louder to cause distortion rather its more that the peak is shaved or flattened which means the sample bit value is sent to the device multiple times which causes the distortion.

Spotify and Apple Music allow users to leave loudness normalization off, so the streaming services won’t change the loudness of the songs you play. I wonder how many people do that (I leave it off). Given that people leave loudness normalization off, it’s tough to say if the decrease in loudness needed to avoid intersample peaks is worth the reduction (i.e. loudness wars). Maybe another test you could have done is between the no intersample peaks version (which is -3dB quieter) vs. the original file that has intersample peaks (but is 3dB louder). Which one would a listener prefer? Basically, is avoiding intersample peaks worth the drop in loudness? Streaming services could force loudness normalization for every song and that would lead me to avoid intersample peaks instead of pushing my track as loud as it can reasonably go.

I would like to ask an amateur question from someone who is more interested in quality play back then production. If I have a file that has been mastered with ISPs, and assuming I can hear the degradation that it causes, would I still get that distortion if I set my dac to, say -3dbfs ? Secondly what difference would it make if that file was lossless, presumably the ISPs would not be as noticeable, but would still be audible? Please remember I am an amateur, and this question might not make sense to someone who understands this topic! Thank you Mitch

One question, since the encoded file is stored in a floating point format, does that mean that if normalization is engaged, then those peaks might be preserved? Or is normalization occurring after the transition to fixed point?

Great Video, do you have a link to the ABXer software that you showed?

Thanks for this one Ian!

For me, the kick and snare sound a little more buried in the non-ISP version, but I like the sound of both of them. I really think it's dependent on each track and its arrangement. For example, Madeon's 'Icarus' has more obvious clipping in the verses, but as soon as the drop hits, the drums, synths and FX, that gets introduced in the arrangement mostly masks the clipping.

So summing up to avoid that problem of lifeless masters with lossy streaming services codecs it`s better to leave True Peak under -1dB? Can you please explain the concept of the video briefly in more human language, as I`m even more confused than I was. Thank you

Very interesting. I frequently download popular songs from KZfaq to study them and most of the time, the m4a file that I download has lots of peaks going above 0dB so when I normalize it to 0dB, the song goes down by 1 or 2 or even more dBs in some cases. Sometimes there are thousands of waves on each channel having this issue, going above 0dB. And these are top-charted songs possibly mastered by engineers among the best on this planet so would that mean that it's the big labels who still control how commercial music has to be mastered ?

on some dacs this is really insane... bad enough intersample clipping causes effect like volume is fading after the clip and then it comes back up... audible... well... very much so when it sounds like somebody turns volume near 0 and back up real fast... :D i dont need to even abx and unsuprisingly when appling limitter the volume rollercoaster effect is removed

that was useful, thanks!

The clipped version sounded a little more rock & roll! ........... this was a pretty extreme (distorted) example, the real difficult choices come when the clipped version sounds punchier but not really more distorted, then your finger gets itchy to switch ISP protection off.

Is there any particular true peak limiter you recommend?

Crazy interesting.

Yeah, even via KZfaq codecs the difference is apparent. Thanks for the great reminder!

Yes, there is an audible difference in your example, albeit subtle. However, IMHO the crux of the discussion *should* be about why to avoid ISPs altogether - which you touch upon towards the end by pointing out that streaming services turn it down further. Unfortunately, in a world where proverbial noise pervades nearly every aspect of our lives, a healthy amount of headroom and dynamic range are both fighting - ahem - an uphill battle. I thought the loudness wars were over?

Im pretty sure the codec produces samples and the samples are not converted to something that can have inter-sample peaks until D/A. So the DA needs a little headroom to handle this but it should not be an issue in the digital domain at all.

to answer the title, yes it is a major problem

The problem with this is that your example is really extreme. +2db true peak is quite an insane amount. The average professional loud track (-5 lufs to -4lufs) with clipping only goes up to around +0.5db to +1db. And if you have to try so hard to hear the clipping here, it's definitely a non-issue on those other tracks. So while you have shown that you can hear intersample peaks, it doesn't mean that you should kill your loudness with the recommended -1db to -2db ceiling on your limiter. Overall I think it's not really a big issue.

not using true peak and oversampling in limiter is bold, i like how it saves transients and punch!

Amazing! You stop saying dB´s. Correct, we do say 2 dB or two decibels. Nice video about ISP.

I listened to on my phone on purpose and didn’t look at the screen for the listening test. I could hardly tell. Sometimes I was right but most of the time not.

The clipped ISP version sounds actually really "trashy" compared to the clean version.

If you get 7/10 twice, you still only have 7/10. 14/20 = 7/10. I would say that it can be audible, probably isn't but since technically it is there.. it has to be part of the process to avoid ISP. I've used -1.6dB and see no reason to change, it is a game of probabilities after all and going past +1.6dB means someone has royally fucked up.. you don't get those unless you almost deliberately prepare a sample of "worst case scenario".

Me recuerda a la voz de Fzst

Jfc your mic needs a deesser

Really in depth and thorough. Appreciate you and your amazing work! 🫡