Saturday 6 January 2018

Audiophile Myth #260: The Detestable Digital Filter Ringing and Real Music...

For the 2016 edition, go here.
In the audiophile world, there are many targets of angst and anxiety that "practitioners" of this hobby are well aware of.

For years we've been worried about the "dreaded jitter". However, we know that these days, with asynchronous interfaces like USB and ethernet, there's nothing to be concerned of. Sure, we can see jitter anomalies with old S/PDIF, but I doubt anyone should purposely not use the interface for fear of audible issues assuming otherwise decent gear. For years and still to this day, various "practitioners" of audiophilia hang on to beliefs around cables of all sorts; assuming normal hook-ups with decent quality cabling (and even with poor quality cables), we are typically hard pressed to find evidence of audible differences.

Then we have beliefs that bit-perfect streamers sound different, really? We also hear of esoterica like folks who think lossless file formats sound different, seriously? How about the folks who think that computer OS's make a difference, or software players sound markedly different (assuming it's all bitperfect of course and sent to the same DAC)...

Feel free to browse this blog for discussions and thoughts around countless other audiophile items of faith. Today, let's address the audiophile "myth" that has gained prominence among those trying to sell things and those that advertise said "things" over the last few years.

As per the title, today, let's explore this "myth" of the detested ringing with digital filtering and audio playback since I've been posting a bit of a series on this topic over the last little while.

I. A Little History... How in the world did we get here?

Anti-imaging filters in digital-to-analogue conversion have been with us since the start of digital audio of course. They're needed to reconstruct the continuous nature of the audio information encoded in our digital files and prevent unintentional frequencies to seep through. Let's stroll through the archives of Stereophile over the years to see how a focus on filtering has crept into the audiophile mindset as a topic of discussion and debate. As I've said in the past, despite my criticisms, at least Stereophile is the most technical of the mainstream audio magazines easily found in North America and good stuff can be gleaned especially in the measurements section of their reviews.

Audiophiles had jumped on the idea that CD's specs were not up to perceived expectations shortly after the introduction of the format (the ideas were reviewed in this 1986 article). These days, all kinds of people claim they "knew from the start" that CD fared poorly against vinyl LPs in terms of sound quality. Sure... We're in an era where there's a retro-coolness to owning turntables and LPs so unless it's documented somewhere years ago, it doesn't count :-). All I remember about the early 80's digital audio was marveling at the beauty of the discs themselves thinking how "cool" lasers are. A few years later, reading the excellent review of the first Sony CDP-101 in an old copy of Stereophile I found, I remember how I "needed" one of these by the time I reached my teens!

As I re-read the articles now and the followups including J. Gordon Holt's later comments (like this one), it's hard not to have respect for the man who stands his ground when he saw real progress and could clearly describe the hows and whys the changes represented a step forward in fidelity. Clearly the CD format was a step forward compared to the obvious deficits of vinyl. From the start, questions about dynamic range, the importance of dithering, and uncertainties about antialiasing/anti-imaging filtering had already been posed. Remember though that those were the very early days of digital, when most consumer CD players and later DACs were not truly capable of 16-bit resolution, before digital upsampling and oversampling, and before the use of delta-sigma modulation or complex hybrids of multibit and bitstream as we see these days. Even then, CD's sounded good!

Some of the earliest devices that focused our thoughts on digital filtering were devices from Theta and Wadia. As far as I can tell, Theta devices performed standard linear phase digital filtering with DSP chips of the time (starting with the DS Pre in 1988, with measurements of the DS Pro Prime by 1991). Wadia went a different direction with their filtering algorithm - like this review from 1990 of the Wadia Digimaster X-32 "digital processor". Check out the Measurements section and marvel at the use of atypical (at that time) slow roll-off filtering with clearly measurable roll-off before 10kHz and strong imaging artifacts above Nyquist. Interestingly, the square wave result looks like they used an intermediate phase filter. All this being done by 36MIPS of processing power (not bad back in 1990)! (For comparison, remember that the humble Raspberry Pi 3 today is capable of about 2400 Dhrystone MIPS, and 190 Linpack double precision MFLOPS - all for <$40US - isn't Moore's Law great?)

Over the years, Wadia continued to release new versions of the Digimaster - such as this 4th version of the Digimaster 2000 Mk.2 by 1996 and the 27ix by 1999. As you can see, they continued with the tell-tale slow roll-off filter but the square wave in the latter review suggests that they went back to a linear phase symmetrical filter. By 1999, technological advancements allowed the Wadia device to be capable of 96kHz input and true >18-bit resolution.

One does wonder whether the filter designs back then were truly what the engineers intended or simply the result of compromises made due to computational limitations. Just the same, even if known to be suboptimal, there's a market to be made by being "different" even if these filters were actually not any better in terms of fidelity.

Another interesting device worth mentioning was the Meitner IDAT (Intelligent Digital Audio Translator) DAC from 1993 (square waves also shown here). This "intelligent" device used four Motorola DSP56001 processors (90MIPS, baby!) and would flip between FIR and IIR filtering. Hmmm, I don't see a point to doing this these days, but I wonder how the detector would select which filter and whether distortions would be introduced by the switching; presumably it could switch mid-playback and perhaps multiple times in a song...

The Japanese "major" consumer electronics companies got into marketing of filter tweaking only to a small extent from what I could tell. For example Pioneer had their "Legato Linear" / "Legato Link Conversion" spline-based interpolation in the early 1990's (eg. Pioneer Elite PD-65 in 1992, mentioned as a feature with Pioneer PD-S 507 in 2000). I have not come across measurements of this filter.

In the late 90's and early 2000's, there were claims put out by companies like Audio Note and Sakura Systems that digital reconstruction filters were unnecessary. "1x oversampling" was the rallying cry for Audio Note and Sakura hailed the low complexity of Wadia's 13-tap filter and Luxman's DA-07 "Fluency" filter (released in 1988, go here to download a paper on the interpolation method, measurements for the 2011 Luxman DU-50 here) claiming that Non-OverSampling (NOS) was the way to go. Yes, NOS DACs do sound different, but if aliased, squared off "stepped" digital sound, with some high frequency roll-off is what you want in 44.1kHz playback, then those devices are for you. These days, we still see NOS devices like the MHDT DACs, Computer Audio Design devices, and the Pro-Ject DAC Box typically still using the old Philips TDA154x chips hailing from the early 1990's (you can of course get cheap versions on eBay like this one I measured awhile back).

By 2005, we have Ayre and their C-5xe Universal Disc Player providing intriguingly various digital filter settings labelled "Measure" and "Listen". Later they released their white paper with the introduction of the MP upgrade of devices by 2009 (also the same year as the QB-9 USB DAC). The "Listen" filter is the familiar slow roll-off, anti-imaging reconstruction filter used in the PonoPlayer among other Ayre-designed gear.

By 2006, I remember reading this Stereophile article and becoming curious about this whole digital filter business. But although Ayre had done something similar earlier as noted above, it wasn't until 2009 with the Meridian 808.2 and their intriguing use of the term "apodizing" (which really was just "minimum phase" when it came to that filter) that I sense the whole idea of using digital filtering as a differentiating factor marketed to consumers at large started to take hold of the audiophile psyche. I remember asking about this by 2012 in the Slim Devices forum as I read through the material and started to wonder just what the big deal was about! For awhile I played with custom upsampling using Viruskiller's settings on my Squeezebox Transporter as documented on that Slim Devices thread.

Fast forward to this decade, and we see an evolutionary progression where impulse response measurements are typically included in the suite of objective assessments (including here on this blog). Many DACs will allow the user to switch settings (including high end consumer players like the excellent Oppo BDP-205 UHD-BluRay player). There's also a move to standardize the filter characteristic (à la MQA for compatible devices).

And that friends, brings us to today... In 2018. After all these years, do we have a clear concept of what we want in a digital filter? Do we understand whether there's actually anything to worry about? Specifically, are there "ringing" or time-domain issues we need to worry about with an orthodox linear sinc filter - supposedly the "ideal" time domain filter for low-pass, antialiasing/anti-imaging?

I think if you've read through my last 2 posts (here, here, plus the one in 2016, plus the blind test in 2015), you will have seen that on the whole, I believe having digital filtering is important (no NOS for me) but let's not "make a mountain out of a mole hill"! In fact, as you saw last week, my preference is clearly more along the lines of high quality linear phase filters being more frequency and phase/time coherent than minimum phase filters as employed by Meridian, Apple, MQA, and Ayre.

II. Demonstrations - with actual, real, music! :-)

Today, let's get into the real world! I'll provide impulse response graphs for correlation only, no synthetic square waves like last week... Let's look at actual music and the magnitude of supposed issues like the much detested "ringing".

If we are to pixel-peep, let's zoom into the leading edge of perhaps the most dynamic sound recorded on digital - a cannon explosion on the CD layer of Telarc's 2001 1812 Overture SACD. Let's look at the waveforms after running through SoX upsampling using different anti-imaging filter settings:

Notice the impulse response graphs to the right. To the left, you see the original 16/44 data up top - I've zoomed in enough to reveal the actual sample points. And then below are the 3 upsampled/filtered/antialiased waveforms with a basic cubic interpolation underlaid to provide a time-domain comparison after upsampling to 352.8kHz. What do we see? The filter with the least temporal change is the linear phase filter. Even with the impulsive leading edge of a cannon blast!

Using the same filter steepness, the minimum phase variant with its non-linear frequency-phase/time relationship creates a slightly different upsampled waveform. And you can see at the bottom that I've included my intermediate phase suggestion from last week which in this situation with an excellent recording, only shows a very slight temporal change from the "ideal".

How about another example? This time of a snare drum. Snare drums are a great test of temporal accuracy. Here's a good sounding free snare sample. I normalized the peak to 100%, zooming in on the highest peak in the "attack" (this whole waveform shown below is only 7ms). SoX -b 95 used again to upsample to 352.8kHz. As above, I've underlaid the cubic upsampling waveform for temporal comparison:

Feel free to try this with other "impulsive" sounds such as cymbal clashes (yes, I've looked at this as well, but not shown). Looking at the samples above, do you see any extra ringing anywhere whether it be "acausal" pre-ringing or "likely masked" post-ringing even with a steep filter (95% passband) used?


The key here is to remember that within a properly bandwidth limited signal where all the frequencies are below Nyquist, a linear phase FIR filter actually does not create ringing regardless of the impulse response appearance. As I have said in the previous weeks, any decent recording will follow this rule. And if it does, then the ideal filter to use is clearly a linear phase, sharp filter that can reconstruct all the frequencies in the audio data with essentially ideal temporal resolution.

And that folks is the "myth" we need to say goodbye to in 2018! Linear phase, steep "brick wall" type antialiasing/anti-imaging digital filters performed with high precision, and with no intersample overloading, do not "ring" with good recordings that only contain "legal" frequencies below Nyquist. Sure, some people might prefer minimum phase slow roll-off filters because they sound different (as per Ayre's "Listen" filter), but technically if you care about time domain performance and frequency domain accuracy of 44.1kHz playback, you would go with a high precision, reasonably sharp, anti-imaging linear phase reconstruction filter (as per Rob Watts' video linked 2 weeks back).

But there is a slight issue we have to be aware of! What if the music we listen to is poorly engineered and not properly bandwidth limited? Sadly, these days with loud, dynamically compressed, "limited", and all-out clipped music, there are quite a number of "indecent" recordings out there.

They aren't hard to find. Just look for tracks in your music collection with low DR (typically DR5 or less), then have a look for squarish waves. Here are a few examples of pixel-peeping into rock/pop songs and finding "problematic" waveforms in the FLAC CD rips:

It literally took just a few minutes to identify those songs above. As we know, Red Hot Chili Pepper's Californication album is a classic tragedy of the loudness wars. Indy/garage rock albums like the "lo-fi" sound of The Black Keys' El Camino have a reason why they sounds "lo-fi". Modern pop music is literally filled with squared off waveforms - Bieber's Believe, Pink's The Truth About Love, recently Train's A Girl A Bottle A Boat (Train, what are you guys doing when Bulletproof Picasso was much better mastered and sounded so much less congested?!), and U2's Songs of Experience where apparently the old guys are trying to sound young and fresh with the "modern sound". And that's even without me digging into the hard rock, metal, grunge, or modern remasters collection :-(.

"So what?" You say... Well, let me show you what happens when we put the segment like Pink's song through a steep digital filter making sure we avoid intersample overload (as above, I include the Cubic Interpolation comparison):

Voilà, your "ringing" as the digital filter negotiates the "illegal" parts of the waveforms. For context, that whole waveform above is only 4ms of the Pink song!

Underwhelmed by the severity of ringing? :-)

You see, in real music, even with egregious examples of degenerate waveforms like these, the intensity of the ringing is never as impressive looking as what is shown in the impulse responses when looking at DAC measurements.

As you can see circled in pink for the linear phase filter (with a 95% passband), the "acausal" pre-ringing is actually very tiny. In fact, this P!nk sample is one of the worst examples of pre-ringing; most of the time it's not even as noticeable as above! With the minimum phase filter we see that the post-ringing is more intense and notice what happened to the double "flat tops" due to the post-ringing.

I also present the same waveform using the "-p 45" intermediate phase suggestion which again I think is a great compromise that clearly reduces the pre-ringing and still maintains the "flat tops" quite well due to much less intense post-ringing; at the expense of some very mild phase shift primarily with higher frequencies of course.

In some situations such as the Californication sample (zoomed to about 5ms so we can see the ringing better!):

Notice how in this sample even though poorly bandwidth limited, the linear phase filtering changed the waveform only very slightly and there was actually no meaningful ringing added. In fact, as you can see, by going minimum phase, even if it's primarily post-ringing, the waveform was altered much more significantly.

This is an example of a situation where using minimum phase filtering gained nothing in the process and in fact worsened fidelity. There was no pre-ringing to suppress and all it did was cause temporal delay and "blurring" of the frequency time relationships. Again, it's hard no to be underwhelmed by how little there is to be concerned when it comes to "ringing"!

III. So... In Summary...

While it's not fully a "myth" that ringing cannot happen with playback, here are the key points:

1. Generally there is nothing to worry about with ringing. They only happen with poorly mastered music - typically those with strong dynamic compression, low DR, and a propensity to clip.

Strictly speaking, these types of recordings are not worthy of a "high fidelity" audio system any way! Surely as audiophiles, I hope we're not spending thousands of dollars just for listening to The Black Keys, Justin Bieber, P!nk, Beyoncé, Miley Cyrus, Imagine Dragons, Bruno Mars, The Weeknd, Taylor Swift, Lady GaGa, etc. without throwing in a bit of acoustic, jazz, and classical into our musical repertoire. The truth is that one would never be able to appreciate the nuances and dynamics that high quality equipment can convey when fed essentially compromised audio material that doesn't even need 16-bits resolution, much less 24-bits. This is not a judgement against the value of the artists or genres (hey, I love my rock and pop!), but rather an observation that high fidelity hardware can deliver way more resolution potential than what this kind of "software" demands.

Ringing is seen only with albums that contain poorly bandwidth limited waveforms. In general, classical albums and acoustic recordings will not have this problem. No surprise then that even with very steep filters, our blind test back in 2015 did not show a clear listener preference between linear and minimum phase filtering using good recordings. As shown above, even when it happens, they're not of strong amplitude and typically inaudible as discussed before.

2. If you typically listen to good quality recordings - you know, most classical, vocal and jazz recordings, stuff from DCC, Telarc, Channel Classics, MoFi, Analogue Productions, AIX, Wilson, Chesky, classic RCA Living Stereo, etc... Stick with a high quality digital filter that is: linear phase, relatively steep that doesn't roll-off in the audible spectrum, and provides overhead protection against clipping/overload. Something like Rob Watts/Chord's accurate high-tap sinc filter or using piCorePlayer with the "Chord-like" SoX settings from a couple weeks back will be best for these recordings from the perspective of highest fidelity. As I said in that previous post, I don't think a very steep filter is necessary, something like 95% bandpass with SoX is more than good unless you are sure your hearing acuity extends beyond 21kHz :-).

In this light, I actually cannot help but think that when companies introduced certain "new" filtering ideas, these steps actually worsened fidelity. For example, though intriguing at first, these days I consider the Meridian "apodizing" steep minimum phase filter from 2009 a step backward in terms of fidelity. Sure, it created a stir in the audiophile community, maybe sold a few Meridian units, and put the idea of switchable filters on the feature list of many DACs; but it causes more time-domain issues than it actually solved. All because it dazzled even those in the audiophile press with the lack of pre-ringing impulse response? Here's a quote from that review:
"The brevity of these preresponses that occurs with high sample rates has led some commentators to decide that this is why high sample rates produce better sound quality, and not the octave extension in bandwidth itself. On the face of it, however, this improvement seems paradoxical: the preresponse comprises ringing at half the sample rate (called the Nyquist Frequency), which, even with CD, is above the limit of human hearing. But the preresponse does appear to have an audible effect, perhaps because the human ear/brain system acts as a wavefront detector rather than as a spectrum analyzer."
The most fascinating part is the piece that I italicized. Notice the ambivalence of the sentence before where he expressed factual recognition that the ringing is at Nyquist. But then right afterwards, takes it as some kind of subjective "truth" that the effect "does appear to have an audible effect" with no evidence for such a thing whatsoever! Who said they could hear this and with what music? Probably not Mr. Atkinson himself because earlier in that quoted paragraph, he referenced the 2006 article where his own listening test did NOT clearly show evidence of an audible "preresponse"! Such is sadly the state of the audiophile mainstream media... Full of contradictions and an unwillingness to be critical of claims no matter how unsupported. Of course Meridian wants us to believe pre-ringing is audible to sell their goods; but do we as audiophiles need to believe this, and does the media take any responsibility for educating, clarifying, and investigating before promoting such ideas?

BTW, I think Apple's switch to minimum phase playback in iPhones and iPads (compared to the classic iPod) also doesn't help fidelity - there's no point getting picky playing typical AAC iTunes/Apple Music lossy audio in any event.

3. If you listen to lots of 44.1kHz modern pop/rock/electronica/loudly remastered CD's and likely will run into anomalous waveforms, go have fun with my "Goldilocks" intermediate phase setting from last week. IMO, it's fine for all that I listen to and incorporates small tweaks that may not be "ideal" but I do not believe any human would be able to notice. The intermediate phase setting will significantly reduce further any risk of audible pre-ringing (assuming we're still worried after all I've said and shown here!), phase anomaly will be minimal compared to a steep minimum phase filter, imaging suppression is excellent, and there won't be any issue with intersample overload in actual music with 4dB overhead as suggested.

4. If you listen to 88.2/96+kHz high resolution, don't worry about the filters for the obvious reason that any acausal ringing from a linear phase filter would be way beyond human perception. Of course, if we're buying 88/96+kHz music, it better be well engineered! Even then, I'd just stick with linear phase, and ensure the passband isn't insanely low to cause roll-off below 20kHz - something is very wrong if one ever sees this. The most important aspect of a good digital filter for high samplerate music is to make sure it doesn't overload when the DAC upsamples!

5. In an ideal world, all recordings will be properly bandwidth limited so nobody could point to ringing during playback! If that happened, then follow point 2 above.

Folks, it comes back to the quality of the recording, mixing, mastering. Fooling around with filtering in the hardware is at best just an honest attempt to plaster over poor recordings (and at worst a marketing attempt to sell something "different" without admission that it likely compromises fidelity). Support what you can to encourage better quality engineering and studio techniques. An end to the Loudness War would of course go a long way, support Dynamic Range Day, don't buy crappy recordings, etc...

Have fun listening everyone and let's stop getting excited/worried/panicked about "the detestable ringing" in 2018. With that, I think I'll take a break from talking about filters for a bit. I've probably covered the topic enough for awhile.

Time to just enjoy the new albums I got over the holiday season... And time to get back to the real day job... :-)

[BTW, if you're wondering why this is Myth #260, this is my 260th blog post! I must have missed a few other ones here and there along the way :-). Almost a full 5 years into the blog with regular updates, my how time flies...]


The 34th annual Chaos Communications Congress wrapped up a couple weeks back in Leipzig. This is the largest "hacker" convention in Europe, drawing something like 15,000 attendees!

For audiophiles, here's an interesting presentation on MQA and DRM by Drs. Christoph Engemann and Anton Schesinger. The slides are also available as PDF. No question MQA implements a form of "conditional access" even if it doesn't prevent the ability to copy files at this point.

I'll leave you to review the information provided by these gentlemen.

Until next time, wishing you all happy listening!


  1. Thanks for great post! Good arguments!

    I think that minimum phase fans can now move to phase 45 setting, and linear phase users can stay where they are :)

    And it also supports steep filtering (e.g. 95 percent cutoff or more) as best option.

  2. Thanks for this. Your Pink and RHCP samples are pretty convincing evidence that -- even with heavily clipped (and hence non-band-limited) samples -- linear-phase still gives the best time-domain behaviour.

    For extra fun, you could run those samples through the "MQA-like" filters you discussed in previous posts. But I am not sure that the sets

    * People who still need convince
    * People who are amenable to this sort of evidence

    have a non-empty intersection.

  3. Wow, great reading (again), thanks! Very nice to see these wavefronts and to be able to pixelpeep them. One question about "Goldilocks intermediate phase setting": why allow a little bit of aliassing? What is wrong with having the stopband at 100%? Maybe because then the filter gets steeper, so the filter has to move more to the minimum phase side?

    Still enjoying 16/44 music via Goldilocks :-)

    1. With today's knowledge and computing power, I would avoid aliasing at all. There is reasonable evidence that proper steep filtering can be done without problems and aliasing brings new unwanted post-mastering frequencies into playback chain. The main reason for allowing aliasing was to make filter less steep and decrease overall ringing somehow. Also steep filters were computation intensive which has been problem for hardware and software 20 years ago.

      From recent posts it is seen a) that ringing at cutoff 95 or even 99 does no harm to properly recorded tracks b) even if you allow some alias ringing is still present to some degree when filtering 44.1 with content preserved up to 20 kHz which is desirable.

      And if somebody still wants/needs to deal (shift from preringing to postringing) with ringing (which I think is not necessary in majority of tracks/playback chain), he can choose phase 45 goldilocks setting and do not introduce aliasing.

  4. Great post, thanks! What actually has shocked me in the italicized fragment is the attempt to justify the audibility of pre-ringing by rejecting the theory that human hearing acts as a spectrum analyzer. Nice! After all, we live in post-truth era, so emotions count more than facts.

  5. Absolutely excellent. Many thanks for a brilliant article.

  6. Forgive me for diverging a bit from the topic, but I had never read the Holt piece, nor the Fremeresque diatribe that inspired it, and it took me back...

    I will never forget buying my first CD player (a JVC) in 1988, listening to my first disc (Crowded House, IIRC), and nearly fainting at the amazing, beautiful, accurate sound. After having to live for over 30 years with LPs, this was nirvana! Let me add that I never thought CDs sounded harsh until I started reading Stereophile (and their unindicted co-conspirators) in the late '90s. Those fools ruined music for me for a time.

    In this spirit, you might want to check out the letter about digital filtering in Issue 16 of the late, lamented The Audio Critic, where in Don Moses of Wadia defends his company's theories against criticism from Dr. David Rich and Dr. Stanley Lipshitz:

    Apologies if you've read it already. Me, I'm stickin' to 16/44.

    1. :-( Sorry to hear about reading audiophilia stuff ruining the hobby for you at the time!

      Thanks for the link to the Audio Critic article (starting page 6). Classic from 1991. Amazing that after all these years, we're still talking about the "ringing" and reminding folks that this ringing isn't an issue when dealing with bandwidth limited signals!

    2. Thanks for the thoughtful response, and keep up the excellent work!

  7. By the way, after some experiments now I use SoX's 95.4 cutoff with no alias since new year and it sounds really great after longer listening. For offline Ferocious Resampler, I use 95.2 cutoff, with violet noise TPDF dither to 16 or 24 bit format. I use 16/88.2 bit upsampling format for mobile playback for majority of files to save space. On desktop I use 24 bit or online upsampling.

  8. A clipped recording sounds bad regardless of filtering. Here is an example of unintentional clipping in one of my ripped audiophile CD. I attached a flac at the end of the post so that everyone can listen to it.

    Also, for believers of "declippers", don't use your eyes or any automated algorithms to judge the declipped waveform, use your ears to listen!,113420.msg933979.html#msg933979

    1. Yes clipped recordings sound bad. Audacity's Clipfix can help in cases of light clipping, e.g. with settings 98 per cent threshold (or 95 - depends on track) and -2 dB reduced amplitude.

  9. Finally someone that uses real-life dynamic audio samples.

    What would be interesting is 'nulls' as a waveform that appears the same at a first glance on a 'scope' picture may still be different at dynamic differences of say 20dB.

    Indeed the ringing (pre-and post) do not appear in well recorded music.
    Ringing is only shown in special test signals (squarewaves, step and needle pulses and clipping of course) and is performed to give some insight in the filter's behaviour.

    How come various filters still sound different when those differences are not caused by the dreaded (pre-)ringing ?
    Phase variations between lower and higher frequencies in the audible band can be audible.
    Roll-off IN the audible band (all slow filters) is quite audible.
    No reconstruction filtering in NOS filterless DACs is very audible.

    Something to keep in mind when listening for 'attack' in instruments.
    'Speed' and 'attack' in music is found in the 4-6kHz range, not in the 20kHz range and is not 'blurred' by any type of filter-ringing.

    Something else to consider. I have measured step, needle and squarewave responses of many headphones. Including the very expensive ones.
    Without exception the 'post ringing' of all drivers, most certainly the more expensive headphones, is MUCH more severe than any ringing caused by reconstruction/anti-alias filters and is mostly in the audible range.
    The filter ringing of DAC's is outside of the audible range when it occurs.
    And yes... tweeters also show ringing.

  10. Wow, wow

    Hi Archimago

    This is all, I can say. Really great work in this area.


    1. Thanks Juergen!

      Much appreciate the feedback and assistance over the years :-).

  11. Happy New Year Arch! Wow, fantastic article on digital filter ringing using real music! What a treat :) I hope this myth is finally busted for good. Now, if we can only get the music industry to produce better recordings, mixes and masters: Dynamic Range: No Quiet = No Loud

    Congrats on your 260th blog post over 5 years! Really quality writing Joe, I am in awe.

    Wishing you all the best!


    1. Hey there Mitch!
      You know, honestly I'm a bit shocked looking back that I've written this much.

      The blog started basically because I could not believe what some "audiophiles" were saying about high bitrate MP3; claims that they could "obviously" hear the difference between lossless and well encoded MP3. This started the initial blind test and from then on, it just became a "fun" thing to do :-).

      Keep up the great work with your articles on Computer Audiophile!

  12. Great article. You're leading us right back to where we should be: it's the content. And once you break away from the bit perfect goal and head down the path of what sounds best to you then you can see what a marvel, and a true audiophile bargain, DSP really is.

  13. Thanks for the comments guys! Wish I had more time to respond to each, but alas, this week's looking very busy with other deadlines. Hope you're all having a great 2018 so far...

    I enjoy following various tech hobbies over the years - computers, mobiles, photography, home theater, etc. But when we look at the media and websites for those pursuits, it's amazing to me how audiophilia in comparison is riddled with much misunderstanding and questionable claims passed along as "facts" hobbyists hold on to. This "ringing" business is I think an example of one where even the core belief needs to be challenged after all these years.

    Let's make sure to do what we can to dispel all this nonsense with fears around "ringing" we've been fed by certain manufacturers and accepted without question by the audiophile press all these years. Bit by bit, with education and facts (rather than just subjective claims), audiophiles interested in the technology whether the hardware or encoding formats can hopefully be seen as actually informed hobbyists rather than viewed as the crazy cousins among the technologically-based hobbies!

    1. Many thanks again for great work.
      By the way new version of Ferocious Resampler is out

      and I use these parameters for offline 44.1-88.2 upsampling

      -r 88200 -b 24 -–dither --ns 2 --gain 0.98 --mt --doubleprecision --nometadata --lpf-cutoff 95.2 (for desktop - 24 bit)
      -r 88200 -b 16 --dither --ns 2 --gain 0.98 --mt --doubleprecision --nometadata --lpf-cutoff 95.2 (for mobile - 16 bit)

      Fans of steeper cutoff can try 97.8 or 99 cutoff but I am perfectly happy with 95.2 now.

      I was able to adjust these settings mainly based on what I have read here and that is great.

  14. 30 years have passed and I still have no idea what's "digital glare".
    None of hardware synths from 80s I know of, have that preset.. :)

    Could someone elaborate?

    1. Perhaps this is the wrong place to ask :-)
      but acc. to ArnyK:

      Digital glare is a marketing phrase that has been used to show down the encroachment of digital technology on traditional strongholds of analog technology. Widely used to dupe innocent young people into buying expensive turntables.

      The reality is that there is a sonic effect that does to the ears what bright lights do to the eyes. It's been around forever, but back in the days of just analog, we called it "glare" because we didn't have digital to kick around.

      The sonic effect of glare is like bright gone really bad. Bright relates to accentuated response in the upper midrange and treble. Glare may or may not be more restricted to midrange.

      Glare is most frequently due to bad choices related to speakers, rooms, and media mastering. It can often be tamed with a good equalizer.

  15. Would you care to have a look at this?

    Marantz unique filtering technology
    MMDF: Marantz Musical Digital Filtering)
    This unit (ND8006) is equipped with oversampling and digital filter functions using an original algorithm created by Marantz. The unit supports PCM signal.
    Toggle between the two types to suit your preferences.


    1. Thanks for the heads up Nuwanda,
      I don't know anyone here with that device at this time but will keep an eye out.

      Looks like it uses an ESS9016 DAC chip; presumably programmed with Marantz's own filter parameters to select between 2 settings. Hope they selected to implement good filtering techniques :-). The product sheet also says nothing about this MMDF and I don't see any measurements out there.

      Until there's evidence of something interesting, I'll assume it's just the usual variation... Maybe a minimum phase slow roll-off or something like that for one of the choices.

    2. BTW: Here's Stereophile's measurements of the NA-11S1 from 2013:

      Looks like on that they used a slow-roll linear phase and a steeper, probably "apodizing" intermediate phase setting. Unfortunately no steep linear phase "ideal" filter IMO.

    3. Yes, ND8006 uses an ESS9016 DAC chip. It is the newest addition to the line up. Not a bad polivalent machine that I want to keep an eye on it. I always thought Marantz makes decent products and I like seeing them moving away from the Cirrus Logic DACs.

      The one you mentioned NA-11S1is from the reference series and I guess based on a Spartan-6 FPGA like PS Audio Direcstream. That allows the manufacturer to practically create a custom software based DAC.

      Thanks for all your insights.
      Appreciate a lot your work. Makes me keep learning.

    4. Still I would upsample to 88.2/96 kHz, DAC of this level would handle the rest very well.

  16. One of your best blogs. Fitting for your 260th entry shining the light of truth and reality on the audiophile world.


  17. Great job as usual archimago. I always wondered why, with all of the debate over filtering through the years and especially since Stuart and Meridian made a big deal of it, nobody ever tried to prove a point using actual music signals. Until now.

    So now people who do think they hear subtle differences between filter types are choosing between two artifacts whose audibility is often questioned: ultrasonic ringing or phase shift in the audio band.

    Now, maybe someone will be inspired to do similar waveform comparisons on Hi-res vs. MQA versions, like the 2L recordings for example. Probably not you, as I know you're tired of that hot mess :)

    1. There really are subtle differences between various filter types. They can even be demonstrated in blind listening tests.
      Those differences, however, are NOT caused by the 'ringing' but are caused by audible frequency roll-off within the audible band and or phase shifts in the audible band even though the latter is a matter of training what to listen for.
      This roll-off in the frequency range can be clearly seen on frequency plots.
      People like to blame 'pre-ringing' for this just like jitter is often mentioned as an excuse.

      Then there is the ultrasonic crap some slow filters pass through which could (and maybe even do) produce audible spurious signals in loudspeakers/headphones as they are fed signals they may not be able to reproduce that well and 'distort' in the audible band.
      That last one will be a harder to 'prove' objectively as it depends on transducers.

  18. I must say I'm a bit disappointed. You show the signal after filters calculated on the computer, everything in the digital domain, and nothing that would equal the real world behaviour. I think I suggested that before, and here would have been the place to show it: the intersample-peak-samples of real music measured at the output of a DAC, with its own filters and no upsampling nonsense (sorry) in-between. Missed opportunity with interesting results. AFAIK DACs usually have about 2 dB headroom, so they would handle these examples perfectly without drastic distortion. Your Teac might not be eligible for that test as it uses upsampling, which can not be bypassed/defeated completely. Even if not upsamling it might stay in the audio path as filter and add its own ringing.

    1. Hey Techland,
      This is basically a demonstration of digital upsampling filters and a way to achieve excellent settings without worrying about ringing and intersample overload. We can talk about the DAC output another time when actually measuring devices...

      What do you mean "no upsampling nonsense"? There's essentially always upsampling these days... Whether in the computer or in the DAC unless you're using a NOS which IMO is not the way I like to listen to my music. One can easily defeat the upsampling on the TEAC UD-501 and it's automatically turned off when sent 352/384kHz data.

    2. I had measured the 503 (which I believe has the same upsamling code and basic path in its FPGA), and it was clearly visible that its own filter was always present additionally to the DAC's filter when measuring at the analog output. Even with upsampling off.

      There is a clear difference between the neccessary oversampling in the AD and DA process, and the superfluous additional upsampling from 44.1 kHz to 192 or so. No one was ever able to show real-world advantages to me. The output signal never got better. The often quoted 'aliasing filters shifted outside the hearing range' is double nonsense, as it never was in (at least my) hearing range, but also because any upsampler includes the exact same filter at 22.05 kHz again. Didn't you notice that all your upsampled signals clearly show 22 kHz ringing - not 179 or 192 kHz? So what is all that good for then? I can save all my time by just selecting the right filter on my DAC and enjoy my music...

    3. Indeed when upsampling a 44kHz file the 22kHz ringing does not move upwards.
      It can't because the intention of that filter is to eliminate aliases above 20kHz. The ringing also is not a problem as it is inaudible.
      It is above the audible range IF these signals are present anyway.
      Clipping signals may cause (again inaudible) ringing but the fact that a signal clips is more sound deteriorating than inaudible ringing.
      It is time people stopped worrying about ringing.

      There is an advantage of upsampling but this is after the DAC in the analog filter that has to remove the 'stairsteps' and HF garbage that are moved upwards in frequency (with 4x upsampling to 176.4kHz)
      This filter can now have a much simpler and less steep construction and won't ring. Much less components (and higher tolerance in values is allowed) in the construction of the analog post filter is needed reducing costs considerably.

    4. Hi Solderdude, I am not worried abpout ringing. I just mentioned it as obvious proof that upsampling doesn't improve anything because upsampling includes the same filter as the DAC itself, at 22 kHz. The only logical advantage of upsampling would have been if that filter can move upwards, a) far away from the audio hearing range, and b) improving the impulse response (even if we can't hear it). But both does NOT happen!

      > This filter can now have a much simpler and less steep construction and won't ring.

      I don't understand that logic. There is no such 'analog' filter, it is inside the DAC already (these days) and doesn't ring any other than what one sees from the anti-aliasing filter at 22 kHz. So where exactly is the advantage? I still don't see it. Also current chips like the AK4490 are extremely (!) clean outside the audio band. There is no advantage to adding analog filters behind the DAC.

    5. Just take a look at the schematics of a complete DAC or the datasheet of DAC chips.
      DS always upsamples. The frequency where it upsamples to is ao high that a simple 1st or 2nd order is already enough to filter out HF noise.
      For Multibit DACs that upsample a lower frequency filter is needed.
      Usually something like 4th order.

      you won't see any of this filters actions in the signal.
      It simplifies the analog postfilter, fewer and cheaper parts can be used. That's the bonus/gain from upsampling... a simpler post filter.

    6. Nope. You both are moving sideways, missing the point now. I wrote:
      > There is a clear difference between the neccessary oversampling in the AD and DA process, and the superfluous additional upsampling from 44.1 kHz to 192 or so.

      Solderdude then replied:
      > There is an advantage of upsampling but this is after the DAC in the analog filter that has to remove the 'stairsteps' and HF garbage that are moved upwards in frequency (with 4x upsampling to 176.4kHz)

      This has nothing to do with what SD then explains after I disagree:
      > Just take a look at the schematics of a complete DAC or the datasheet of DAC chips. DS always upsamples. The frequency where it upsamples to is ao high that a simple 1st or 2nd order is already enough to filter out HF noise.

      Correct. And this is what I already said in the first place. And it is NOT about 'There is an advantage of upsampling...with 4x upsampling to 176.4kHz.'

      That analog filter is not part of the discussion as it (as SD correctly writes) has no influence on ringing etc (simple low pass 12 dB/oct at about 100 kHz).

      So back to square one: where is the advantage of upsampling from 44.1 to 192 kHz?

    7. Well upsampling 44.1 to 192 is a bit daft... to 176.4 seems more logical.
      When using a NOS filter-less DAC or a NOS DAC that can do multiple sample speeds then it has advantages as the output signal may well be 'better'.
      When you do not find this to be the case ... well than don't.
      It seems rather pointless to upsample to a higher sample rate and then feed it to a DS DAC as that one will upsample even further.

      In the AD process it makes a lot of sense to sample at at least 176.4 or 192 kHz also due to the input anti-alias filter being less steep and gives more options after the initial recording.

    8. Hey Techland,

      Hmmmm, I'm not sure who's upsampling to 192kHz... In this post I was showing upsampling to 352.8kHz (which turns off the internal interpolator on my TEAC DAC) to demonstrate the intermediate phase setting suppression of ringing while maintaining good temporal accuracy. No different than say using HQPlayer to do high quality upsampling.

      Maybe you can show me a link of what kind of measurements you were thinking about.

  19. This might be of interest:

  20. I follow and completely agree with all your conclusions.

    There is only one remaining possibility for an audible effect. The pre-ringing causes issues for playback devices such as IMD distortion in pre or power amps and output stages.

    The worst case IMD generally requires two HF signals spaced 1 KHz apart (per stereophile 19 and 20KHz) in order to create and IMD tone at 1KHz (where hearing is most sensitive)

    So perhaps the area to investigate might be how ringing at a pure tone of 22.05 KHz from a brick wall filter could Intermodulate with something else and cause audibly objectionable somewhere close to 1KHz? The clock signal runs at a multiple of 44.1KHz (upsampling sticks to integer multiples) so perhaps we have a parasitic analog signal from that or close to 22.05KHz bleeding into the audio?

    This is the adaptive filtering approach that EMMlabs patented (MDAT is the trade mark) :-

    Anyway just some thoughts - perhaps the intelligent folks here can think of other ways any of this pre-ringing might be audible. As it stands I can’t help but agree with every conclusion from Archimago. Clearly Archimago fully understands the mathematics of time series analysis - something I find absent in most discussions on audio.

    1. To add to my comment. The A to D process must create ringing on all HF transients at a specific frequency related to the anti-alias filter used - call this Signal A. The D to A process creates ringing on all transients at the brick wall filter used to remove ghosts - call this Signal B.

      Could IMD distortion between A and B be the audible problem?

    2. Thanks Jeremy,
      Well, Jeremy, I certainly cannot say I fully understand the mathematics of DSP and filtering because like any field, folks spend years learning the ins and outs! What I am most interested as a hobbyist are the implications and practical value. I'll leave the intricacies to guys who actually program this stuff as part of the domain of their day job - folks like Miska and Mansr for example.

      That's an interesting patent there. The block diagram Figure 5 shows us that a comparator and timer mechanism that switches between the 2 filters. Presumably the switch happens quite a number of samples ahead.

      The question of course is "how much difference does this make?" in actual music fed into this system based on the settings they determined? For example, how often is this transient detection system triggered? And where is this triggered (ie. let's see a waveform example)? What kind of music are we seeing this in?

      As for IM, this might be a factor if there is indeed ringing to be created going through the system at different frequencies. I can certainly create ringing by sending an unfiltered square wave into an ADC; but that's not a natural waveform!

      In the real world, what kind of instruments actually create such a massive transient that causes pre-ringing? I'd like to see the data and examples in real life. For companies like MQA that insist on "end to end" control, they need to demonstrate "first things first" before selling a claim...

      Give us an example of a sample of real music through an actual ADC where ringing or time domain issues should be corrected. Then show us how a typical DAC might not reproduce the audio correctly. Then demonstrate the superiority of MQA decoding/filtering. They have not bothered to do such a thing...

    3. Agreed. The HF transients shouldn’t be there. The anti-alias filter should have removed them before digitization. That said it is possible to create transients at half the sample rate by changes in bit level or bit toggling from one sample to the next - what I am postulating probably means that there are errors in the digital file. Another possibility is DAC differential linearity errors - these are DAC non-linearity errors between bit levels.

      Read this for some thoughts of how upsampling noise may help randomize noise

      Anyway what I am postulating is that the ringing might be causing trouble somehow (even though it shouldn’t and even though the ringing itself isn’t audible)...

  21. Why is there so much interest in HQ Player and upsampling and various filters?

    No decent filter should affect the meat and potatoes of the music. So why are so many folks hearing stuff?

    Are most DACs so badly built and designed that their performance is sample rate specific?

    What gives? Surely Archimago has burnt a few brain cells trying to figure this out? It makes no sense mathematically.

  22. Problem solved. I also ran some tests to confirm. See this article

    What folks are hearing with MQA and other maximum, minimum or anything other than a linear phase filter is Phase Distortion!

    I can hear it too. Because of the way the ear functions you need to play the music sufficiently loud to hear it. With Roon I can quite easily hear the difference between Linear Phase and Minimum Phase filters that they offer. The linear phase is the only filter that sounds correct. I can hear why folks might like the distortion of a minimum phase filter - being distortion it is perceptively louder or more dissonant. It certainly makes the transients or loudest bits stand out a bit more.

    The track to try this test is Nils Lofgren Live Keith Don’t Go. Simple guitar and vocals. Probably the track doesn’t have much phase errors to begin with. Complex multitrack multi microphone rock or pop is probably not the kind of track to test out linear vs minimum phase filter audibility.

    1. Enstablish if a dac is better than another can be an adventure.
      I'm not sure that the digital measurement method is capable of understanding what is wrong, what the human brain processes when it hears a heavily manipulated sound.we are still not able to explain through measurements why cables sound different, and many technicians do not accept because they do not have a measure that tells them this.the signal is born at 16 bit, why should you improve the original signal already registered?
      there is no logic that justifies this.
      but mathematics does not understand the question.
      I would like to divide all the big questions of audio reproduction between 2 categories.
      if we want accurate reproduction, faithful reproduction, or as you want to call it, you must extract and reproduce the information already recorded with the fewest possible errors.this applies to all the components of the audio chain, including the loudspeakers and to know that the system is l.t.i. up to the speaker clamps, from the speaker on, the system is not l.t.i. This means that you cannot use dsp equalizer or other to change acoustics.
      if instead you think that you can modify everything you want, without knowing how the sound is born and all the rules of science that govern the reproduction of sounds,
      you are not on the road to accurate reproduction, you are in creative reproduction.creative playback knows nothing about the original sound already recorded already present on the playback media.
      creative audio through mathematics changes the sound at will.
      the measures were born to try to understand why our brain feels a difference.
      now the measurements are used inappropriately to subvert what our brain feels by processing a sound.
      and return to the concept of Sakura
      our brain prefers a sound with fewer errors than a sound more extended in frequency but containing more's like the speakers, there are those who prefer the consistency of a single-track and those who are more focused on the frequency extension of a 4-way speaker.
      there are those who prefer to hear the 3D to have an illusion of sound like live, some don't listen to 3D.
      I don't think mathematics can answer this.
      all these new dac are just technological expression, if it sounds better you don't know ..
      it will sound different, but different is often only the need to change an unsatisfactory will sound different, but different is often only the need to change an unsatisfactory sound.

  23. hello, what's the best way to do pre ringing ? i like how it make snare transient and hihats smooth.

  24. I'm wondering if you've been able to do testing using digital filters of different lengths (e.g. using HQPlayer, PGGB of Chord's products) - theoretically, looking at the impulse response, the longer the filter, the longer the ringing that it produces. It would be interesting to see if the length of the ringing - which you have shown to sometimes manifest itself in the poor recordings - is actually affected by the length of the filter. If that were to indeed be the case, then that would reinforce the case for shorter filters (or "less long", when compared to e.g. 1M+ taps) being better suited to poorly-mastered recordings.