Saturday 21 August 2021

SUMMER MUSINGS: Room EQ Preferences - Vintage Spendor SA1-like curve, Hang Loose Convolver (HLC), and the increasing relative importance of software.

Getting ready for an evening of digital filter evaluation!

I mentioned a few month back in my Spendor SA1 (1976) review/measurements that I really enjoyed the sound of these classic speakers!

While it would not be possible to exactly replicate their sound - these are complex devices with all kinds of unique properties including distortion idiosyncrasies and unique radiation patterns - to some extent, we can try to replicate the frequency response via DSP.

These days, we have the technology available to modify the sound we hear in the soundroom without great difficulty; the trick is to make sure the sound has improved, at least subjectively, when we add things like EQ or applied other signal processing.

Although some (typically more traditional) audiophiles will speak of maintaining "purity" of sound with minimalistic set-ups, if one is a contemporary technologically astute music lover and audiophile, especially one already using complex streamers, computer gear, networked libraries, I don't believe there should be any fear in going all the way using high-quality DSP to shape and adjust what we hear. In fact, from the frequency response perspective, we can "emulate" almost whatever we might want depending on the flexibility of the software used.

For fun, recently, using the Spendor SA1 (1976) measurements, what I did was extract the "predicted in-room" frequency response for those speakers and basically traced out what an idealized curve might look like as shown in the graphic below:

Using my previously captured room log-sweep (I just used the one I measured here since nothing much has changed in my room) with Acourate, I can then redesign the target room frequency curve to something that looks like the Spendor's:

Unlike the actual Spendors which are not full-range speakers, with my Paradigm Signature S8 + subs combo, I can keep a flat frequency response to 20Hz. I used Acourate's "PeakNDip" function in combination with the usual "Basic" adjustments. Added a "HF Roll-Off" up top.

From there, I created a simple FIR DSP filter to conform to that general frequency response (as per usual procedure). And here's the resultant filter simulating the pattern we see from the Spendor compared to my native un-DSP'ed measured response at the listening position:

For completeness, here's the time-domain step response post-DSP as well:

Note that this is a more idealized step response than what we get through the passive filter network from the Spendor thanks to Acourate's excessphase correction. When experimenting with Acourate (and related software), depending on your room and speakers, you'll notice that there's only so far you can go when tweaking the filter parameters so as not to have too much pre-ringing or group delay artifacts. No worries if there are slight anomalies, just make sure to listen and you're satisfied with the sound! ;-)

Even if one's speakers measured anechoically flat typically at 1-2m, the in-room "steady state" frequency response will naturally "tilt" downward in the treble region on account of frequency-related directivity at the listening position. In the graph above, we can see this natural tendency with the Paradigm Signature S8's actual "In-Room" curves.

As such, I usually will create filter variants with different amounts of "tilt" in the high frequencies. Through a high-fidelity system, compressed modern pop and rock recordings/productions are often harsh in the high frequencies, so for those, I will often listen with something like a -6dB tilt as opposed to classical or jazz where I might only apply -3dB.

For this "Spendor SA1-like" filter, I've created -3dB and -5dB variants to tame the treble as you can see here:

Based on subjective listening, I prefer the -3dB variant most of the time. For the harshest overly-bright pop or sibilant content (like say Cowboy Junkies' The Trinity Session, "Misguided Angel"), I will happily dial in the -5dB variant.

As discussed here and elsewhere, empirical research looking at listener preference "target curves" in home theaters and small rooms typically show a bass bump which I also find satisfying so long as it doesn't go overboard (see this Figure 14 from Toole, 2015):

For the average listener, there's a preference of about +5dB from around 30Hz to 70Hz. Interesting but probably not surprisingly, the "untrained" listener likes a much higher bass boost even above +10dB (and more treble as well to create that "U" shape)! Since we're all sophisticated music lovers and audiophiles here, let's not just aim for "average" ;-). I'll add +3dB bass boost to the "-3dB tilt" version for added "euphonia" - here's the frequency response design in "Target Designer" and simulated sweeps using the filters:

Now, typically I would then pop these DSP settings into Roon presets:

And away I go to listen to the sound of the "new" settings with the Spendor SA1-like frequency response characteristics.

While simply using Roon or the JRiver built-in DSP will work, insightful audiophiles know that human sensory memory where we hold on to much of the details/nuances (echoic memory for audio) is actually very short; usually a "buffer" on the order of <5 seconds. I think many audiophiles, especially the "Golden Ears" find this fact hard to believe (while at the same time arguing against controlled listening tests). While I don't doubt that major differences can be remembered and declared to be "obvious", this time duration is usually too short to allow "definitive" judgements to be made about more subtle variations. Unless one has fast fingers, going into menus and changing listening settings can take longer than this, plus it distracts one's attention which is also important for focused evaluation (listening requires both sensory acuity and cognitive attention).

Enter Mitch Barnett's Hang Loose Convolver (HLConvolver/HLC) (US$129 for license), available for Windows and MacOS (works on the new ARM CPU models). I was able to "beta test" and provide feedback to Mitch on the software over the last year - thanks for the opportunity, Mitch!

Have a look at the software "operations guide" to get an idea of how it works. Installation is clear and takes a few minutes to get up and running. There are different ways of operating the program including a standalone app used in combination with VB-Audio Virtual Cable allowing processing of loopback audio from programs like Roon, HQPlayer, or even system wide. For audio software that is compatible with Steinberg's VST3, there's a plug-in that can be used in JRiver 27.0.52+ (prior to this version, only VST2 was supported), Reaper DAW, and Audirvana can handle VST3 in Windows (use HLC's AU plugin for Mac). I see Chris Connaker's article on HLC also discussed applying convolution to DLNA streams; alas I have not used DLNA in years here at home, but this is definitely significant.

For my listening, I used the current JRiver 28 with the Surface Pro laptop, accessing test music either on the internal SSD or through the WiFi network. Notice that I just have the USB 2.0 cable connected to the RME ADI-2 Pro FS R BE DAC (16' Amazon Basics cable, no worries about sound quality - it's digital bits), and the DAC's XLR output to my Emotiva XSP-1 preamp:

And here's what the JRiver 28 screen looks like with the VST3 plugin activated, loaded with some DSP filters to try out as shown on the right window:

HLC loaded filters can be any samplerate up to 384kHz and 32-bits. ZIP files with multiple samplerate filters supported. Efficient processing with <10% CPU utilization even with the Intel i5-4300U-based Surface Pro from 2013.

I've got a couple of albums showing in the screenshot. Ēriks Ešenvalds' There Will Come Soft Rains (2020, DR13), a wonderful collection of choral works beautifully conducted by Richard Nance and performed by the Pacific Lutheran University Choir of the West (Tacoma, Washington). This album has a complex overlay of voices, wide and deep soundstage, expansive reverb. And Olivia Rodrigo's recent album SOUR (2021, DR7) which gives us a taste of highly produced modern pop.

The beauty of HLConvolver is the combination of instantaneous filter switching and the flexible volume control facilities including an "autogain" function. You can use the included pink noise tone (file in the HLC folder) to fine calibrate the filter amplitudes with the "bypass" level. Then with a single click, you can switch between each of six "filterbanks" or bypass the DSP instantaneously. The "Master Trim" control allows you to add some attenuation if you see the output level meters clip. There's simply no easier or better way to accurately compare the differences between DSP room correction filters than this! The instantaneous switching not only allows you to appreciate the frequency-domain changes as I did with the Spendor SA1 curve (and variants), but also the effect of time-domain correction. There's also a convenient Load / Save feature to quickly recall all the settings and filterbanks.

For anyone interested in DSP room correction, this is the "next level" in filter evaluation.

How does it sound with those Spendor-like filters? Obviously it's impossible to fully describe in words. The differences are not hard to hear compared to a typical "flattish" room target curve.

The dip going from 1-3kHz adds a "polite" factor to the sound that's less fatiguing. Some describe it as adding a subjective impression of depth as well, perhaps related to Linkwitz's comments here (see "Psycho-acoustic 3kHz dip") discussing the difference in diffuse field frequency response of microphones vs. human ears. In combination, I like the "punchy" sound in the "Spendor SA1 -3dB Tilt & +3dB Harman Bass Bump" version for rock music with that extra low-bass kick.

Here's what the impulse response looks like in order to "correct" for my left speaker + room (focusing on -50ms to +200ms of the 65k-taps FIR filter) with both frequency and time-domain correction. Notice the complexity. Whatever you think about MQA these days, their simple filters would obviously perform little if any actual correction for "time smearing" in playback systems! To this day, we still have little idea of what they're trying to "de-blur".

Time-domain changes are a bit harder to put the finger on but can be readily appreciated using HLC's quick A/B switching. I created versions of the Spendor SA1 filter with and without Acourate doing excessphase correction. In general, I hear a better "focus" with phase correction on. Voices like the choral parts and solo for example in "Amazing Grace" on There Will Come Soft Rains "snapped" into place better in the soundstage. Even the reverb of the "space" around the voices sounds better controlled and not as broadly diffused, as if localized to a smaller "volume" in the soundscape (obviously better appreciated with actual listening than words!).

Even with adult-contemporary pop as I revisited Peter Cetera's World Falling Down (1992) the other night, it wasn't hard to notice that his voice had extra depth and became more front-focused with time-domain correction when flipping back and forth instantaneously. This is a good example of the kind of change that I would not be able to tease out listening to an unfamiliar sound system or room such as at an audio show or the local hi-fi dealership.

While I believe correction of "time smearing" of the speakers and room is worthwhile and would be my preference, I can appreciate this opinion is also a subjective preference. Although less focused, the spacious subjective "bloom" in my system without DSP can be appealing as well without the time-domain correction. Perhaps we can draw an analogy here, sort of like how vinyl and tube amps have more distortion which to some can sound "fuller" and judged to be "better". Presumably this is why stripping away the time correction in PGGB can sound fine for some users even though I believe this lowers the ultimate rendering quality.

Even though I would not characterize my Spendor SA1-like filters as capturing all that is special about those speakers, with DSP, I believe I'm capturing a significant chunk of the mid/upper midrange tonality especially when listening to vocal music and some of the "presence" those little speakers can convey. Just another playback option to use as the "spirit moves" on those enjoyable evenings when not trying to be analytical about sound!

As I discussed in the past, feel free to design whatever subjective preference or even flight-of-fancy EQ curve you might want to use. Whether it's a flat response, B&K-like, Harman-like, throw in a "BBC Dip" (the Spendor curve would be a variant of this), accentuate the bass, emulate a favourite speaker, etc., by all means audiophiles, tweak to your ears/mind/heart's content and explore for yourself what sounds "good" to you.

Don't forget though that while we can create and explore countless filter design variations, there is still something to be said about "accuracy". This is especially important if one is an artist or in music production since you're creating "the art" itself and presumably you would want this to be experienced as intended with a neutral reference playback system which is really the most logical standard to aim for rather than some arbitrary "colored" sound. Sure, one could be selective and target the music to sound awesome on Apple AirPods but how's that going to come across with a good hi-fi system?!

Compared to the electromechanical transducers (speakers, headphones), the electronics side of audiophile hardware has over time evolved to become extremely "accurate" these days. For example, DACs even at very low price points (like the Topping D10 Balanced, or D10s) are capable of fantastic resolution which will likely never be needed by most playback systems (to a large part dependent on your room's noise floor)! If we combine that resolution potential with the computing resources we have available, we can transform the sound of our audio systems in ways that home hobbyists could not achieve with such ease (and low cost) until recent years. 

[For a fascinating look at an early digital correction device, have a look at the SigTech TF 1120 "Time Field Acoustic Correction System" review recently posted on Stereophile. The cost was about US$11000 in 1996 or ~$19,000 inflation-adjusted today, if you want the "full meal deal" including hardware, software, a computer and omni mic to do what we're talking about here but with lower resolution and less flexibility.]

I would argue that future developments that significantly elevate the audiophile's experience of sound quality and overall "quality of life" has already started shifting from hardware devices (such as streamer, DAC, even amps) towards developments in software (such as playback apps, DSP room correction abilities, virtualization of multichannel streams, metadata availability, UI).

As a music lover, I think the rise of computer audio and streaming allowing us to access the music through all kinds of apps and user interfaces has been the most important change in the last decade plus. Roon for example with its extensive metadata embedding I think has added more to my enjoyment of music than DAC, amplifier or speaker swaps in the last few years. Even though I enjoy testing out technical performance, on the evenings listening to music, it is the interaction with the software and how it allows me to tweak the sound, and learn about the artists and albums, or browse the lyrics easily that has been the most rewarding "upgrade". In terms of sound quality, software has allowed audiophiles to explore with infinitesimal precision things like playback filtering (eg. HQPlayer), providing opportunities to transcend whatever limits of the filter implemented in the hardware. Of all these techniques, DSP room correction is currently the most powerful, allowing the audiophile great latitude; not just targeting for "accuracy", but also achieving one's subjective preferences as discussed above.

Over the years, I have talked about nonsense hardware tweaks. Stuff like the ridiculous Synergistic Research HFT stickiesweird "Vibratrons", or even just expensive cables in general. Then there are folks who think otherwise bitperfect server computers can have a major effect, or fear "noise" from switching power supplies and suggest various typically expensive options with no evidence of anticipated effect. Are we sure these things truly change the sound, or allow the user to have any degree of control?

Audiophiles, let's make sure to synchronize with what we can accomplish in the 21st Century and start implementing fine-tuning that can be measured, and objectively demonstrated to unquestionably affect the sound. Ultimately, to combine objective techniques of verification and fine-tuning informed by insightful subjective individual preferences is powerful and highly rewarding; arguably I believe the most fulfilling forward direction that makes sense for the "high-fidelity" hobbyist.

For those of you who might not be experienced with DSP and all this stuff above, check out Mitch Barnett's Accurate Sound website and his services. I'm sure he'll be able to provide guidance and ideas depending on your needs.


Pssst... With Mitch's permission to post, here's a sneak peek at what he's working on ahead as he expands HLConvolver:
I have started developing an easy-to-use, headphone calibration application that uses high resolution FIR correction filters. The application is called Kick Back DSP (KB-DSP), and Hang Loose Convolver will host the FIR filters as part of the same application/plugin. To summarize, calibrated binaural microphones would be worn in the ear. Headphones worn over top. Test tones will run in both ears, and within 90 seconds, the headphones will be accurately and precisely matched to the wearer’s ears for perfect sound reproduction. Such an application could become the ultimate high resolution personal EQ system for headphones in that the correction filters are orders of magnitude greater in frequency correction resolution than a handful of PEQ filters. In addition, existing PEQ filters are based on measurements using a “dummy head” with faux ears. The measurement difference between faux ears and your ears is considerable, which renders the dummy head PEQ ineffective. Those who try out the application need to get ready for the most accurate sound reproduction they have ever heard.
Nice! Looking forward to what he can achieve in the headphone DSP correction space (I assume this procedure will be best for larger circumaural headphones). Just like our rooms will treat the sound differently, so too are the nuances of our head/pinnae.


Patricia Barber's Clique! (2021, DR14) is out now for those who like her style of vocal jazz. The audiophile press highlights music like this because it's well recorded so hardware audiophiles will like this, but to be honest, the audience among music lovers I suspect is limited. I see there's also the SACD listed to be released on September 3rd which might be a better deal for those looking for something to spin since it includes the 5.1 multichannel mix.

The CD sounds great, very clean recording with excellent tonality. Wide dynamic range (DR14 average), fantastic resolution, love the smooth decays on the cymbals. Also love the soundstage and guitar work on "I Could Have Danced All Night". Barber's voice perfectly captured without excess sibilance. Will need to listen to it a few times to get a better sense of the album as a whole.

The other CD I got this week was The Killers' Pressure Machine (2021, DR6). This one will definitely split fans of the band! Still "alt. rock" genre, but the songs are generally a bit slower, more meditative, more melancholic (although "Quiet Town" and "In The Car Outside" are quite sprightly). The standard version has "snippets" of commentary at the start of a number of the songs from rural American folks. There's an Abridged version without the comments (less distracting for streaming and playlists). At a time of diminished attention spans, I think using this system with interspersed comments between songs is an interesting way to have listeners engage with the album as a whole instead of just picking individual songs since the spoken comments do add to the mood and will "prime" the listener to the music coming up.

We've got stories of small-town tragedies like a railroad-related death, injured horse in stampede, themes around evangelical Christianity, opioid addiction, pyromania, hunting, physical abuse, working-class life, etc. The Killers do Americana, channeling their inner Springsteen.

As I said above, I think fans will have mixed reactions to this one but it's good to see the band expand their repertoire with some quieter, stripped down tunes (like "Runaway Horses"). And there's a sweetness and empathy across the album away from the glitz of Las Vegas. My favourite track is "In Another Life". Sound quality's decent, I think better than most of their other albums; admittedly, I have not heard their previous album Imploding The Mirage from 2020 yet.

Stay safe everyone! Hope you're all having a good end of August. I'll be out-of-town visiting family next week, so off on vacation again for a little bit ;-).


  1. Mitch's project seems super cool! Personally I have little experience of what accurate sound reproduction actually sounds like. Sure I listen to live music and and try to replicate those sounds and emotional connections in my home audio. I measure my main system and apply room EQ to a preference curve to both my main system and headphones but am I really getting accurate? I'm quite happy with the results. Being able to hear an accurate reference is very interesting.

    1. Hey Doug,
      Yeah, I think the projects are very interesting and this is the kind of direction I think audiophiles need to consider beyond the typical "Hardware Acquisition Syndrome" thinking that $$$ correlate with sound quality or apparently so easily accept the stories told by countless Industry salesmen and spokesmen.

      My belief is that it's important to know what "accurate" sounds like as best we can achieve with our gear, rooms, etc. From that reference point of the "high-fidelity standard sound", we can play around with subjective preferences.

      I would like to think that if all audiophiles appreciated "accurate", "neutral", "undistorted" sound as defined by objective means, then it gives us all a much better perspective from which to understand each other even with "subjective" descriptions! I would even consider this a type of pre-requisite before we jump in or strongly make gross generalizations...

  2. Yet another interesting experiment!

    I always appreciate your tests Archimago because they come with an undercurrent of "audio philosophy" - the deeper meaning - which appeals to this audiophile/philosophy geek (with an science/empirical bent).

    I remember the hoopla with the SigTech devices showed up. How does a device that old compare to the DSP features/processing power you'd have with Roon or JRiver of today?

    Also interesting are your comments on playing with phase and settings for the speaker DSP, and noting the apparent effects of "time smearing."

    I know that the audible relevance of time/phase coherence has been controversial, some arguing it's essentially a non-factor (Toole as I recall) others, including speaker DIYers saying they definitely hear a subtle but appreciable difference.

    My anecdotal experience certainly won't decide the question, but along those lines: I always appreciated the sound of Thiel speakers which of course were known for maintaining time/phase coherence with elaborate 1st order crossovers. In the stores, they always had a particular sense of focus, precision and density to their imaging I found fascinating and which sounded very "right" somehow.

    Eventually I ended up with Thiel CS6s, 3.7s and 2.7s in my home over the years, along with tons of other speakers. And a consistant characteristic in comparison with other designs has always been that "regular" speakers seemed to image just great on their own, but when the Thiel speakers were switched in, it was like looking through those opticial lenses where after a few clicks the image just seemed more focused and distinct. The non phase/time coherent speakers tended so sound a bit more swimmy, vague and "see-through" in their imaging.

    Whether it's due precisely to the time/phase coherence or other aspects of the Thiel design (e.g. coaxial mid/tweets) I can't say, but sure has been a very strong, persistent impression.

    I know audiophiles can get a bad rap for always talking about "system synergy" because lots of b.s. occurs under that banner, but in my case that's how it worked out. While I loved the density and focus to the Thiel sound, I always found them a bit "dry" (e.g. string sections less lush) and reductive (e.g. a saxophone would have laser image focus and vivid transient presence, but it sounded "more mouthpiece, not so much resonating body/bell of the instrument." So just a bit thinner. I found pairing my tube amps with Thiel to be nirvana - that focus and density and palpability I loved seemed to remain, but the sound filled out a bit more and I'm less conscious of a sense of thinness.

    I also use Joseph Audio Perspectives which are rightly known for incredible soundstaging and vivid imaging. And tonally I like them a bit better than the Thiels. But even with the tube amps kept as a base amplification, the time/phase coherence of the Thiels seems to result in that tighter image focus when I put them back in the system.

    (That said, another approach to "system synergy" is, quite reasonably seeking "accuracy" in which you ensure each component is capable of measurably accurate performance - from DAC up to Speakers - to the degree possible. Then you aren't worrying about compensating for problems in one bit of gear by trying to find complementary "colorations" in another)

    1. Hey Vaal,
      Appreciate the thoughts man!

      What is life without a bit of philosophy, consideration of ideals, exploration of motivations of self, and ultimately envision our own goals?

      Like any hobby which can be dominated by consumerism, I remember thinking about this stuff long ago, reading the audiophile magazines, even obsessed about tweaks and cables and such. Eventually, I decided that gear acquisition was clearly not the point for myself.

      I like the word "synergy" as well (although 'Synergistic Research' is kinda triggering in a very bad way ;-). Beyond hardware synergy on the level of the electronics and sound, I think it's a synergy between hardware and "the self". Recognition of our own psychology needs to enter the discussion in a desperate way. The synergy of objective performance and subjective preferences which is a big part of the topic here is in the big picture what this is all about.

      To deny one or other part of this "complex" is to risk becoming reductionistic. By nature, audio companies (and often those who mindlessly "review" for self-gain rather than truthfully critique) have something to sell you will come at it with this simplistic approach - "Come audiophile, buy our next generation DAC - it'll blow you away because of !!!" says nothing about truly how you perceive or what preferences you have. They assume that just because their hardware is different, that you might respond that way... Nonsense of course!

      We each need to be philosophically smart about what we truly want. As I've said before, I believe audio technology is very much mature these days. As such, even at reasonable price points, we find all kinds of very competent products. If we understand our personal philosophy and can make it rationally known to others before getting into heated "holy wars", I suspect we'll not only get along better, but also propel ourselves toward better understanding of self and others.

      The truth IMO is that once we hit reasonable price points (let's say something like $5000 for a digital system, we can argue on the specifics if need to), there's no need to expect a huge amount of "magic" beyond this.

      And if a company does achieve a new level of "magic", it would be nice to see the evidence before telling us the price tag. ;-)

    2. As for Thiel speakers, I have only heard a pair in the past at a dealership (or was it audio show?); can't remember now. Unfortunately without intimate experience, it would be hard for me to say...

      I think in terms of time-domain performance, a lot of this will also depend on the current level of performance from one's system. Using HLC, volume controlled, for sure, the difference can be audible with A/B switching. Without DSP, my system sounds good already and I can enjoy my music, but as a perfectionist "hi-fi" audiophile, I'll for sure take that extra bit of temporal fidelity!

      Through the lens of "objective-subjective synergy" as an audiophile, I think we can clearly appreciate the reasons why not everyone will like the same things. To a large degree, our psychological makeup and philosophies are different! Again, if we can appreciate this in ourselves and express these preferences, that will go a long way to understanding each other.

      We can imagine that a person auditioning a hi-fi DAC might instead prefer some kind of lower resolution tube mod digital player. Or prefer the "bloom"/"live" sound of a SET tube amp instead of the "clinical"/"precise" sound of a higher resolution Benchmark AHB2 (these adjectives like "clinical" of course can be chosen to enhance or devalue the product).

  3. Hi Arch, thanks for taking HLC for a spin! Being able to switch instantly between level matched filters (or no filter) allows one to zero on which filter sounds best, over speakers or headphones. Switching between filters with and without time domain correction is a real ear opener as you have heard for yourself. For anyone using DSP room correction or headphone correction, it is a great way to train one’s ears to really start discerning audible differences in a meaningful way.

    You raise a great point about the increasing relative importance of software to audiophiles. I participated in the DSP revolution in the pro audio industry a number of years ago. I spent 10 years as a recording/mixing engineer working on physically large 24 track mixing consoles connected to a 700 lb 24 track 2” tape machine with racks of h/w effects directly behind the mix position. The effect racks were comprised of analog plate reverbs, compressors, limiters, phase/flange, etc., all analog electronics. A studio like this outfitted with the top analog gear in that era cost several hundreds of thousands of dollars, often going over a million.

    Then digital delays and digital reverbs started appearing in the effects rack. Then came along digital 24 track tape recorders. Then digital music samplers. Finally the Digital Audio Workstation (DAW) once computers became powerful enough.

    Flash forward today and one can outfit the same state of the art studio, using the “same gear” to get the same sound for under $10K, including computer. I paid $5K for a Lexicon 224 digital reverb during that era. Today, the 224 is modelled as a DSP software plugin for $139. That giant 700 lb Studer A-800 24 track machine has had its electronics modelled using DSP, right down to the infamous tape head bass hump. Cost: $329. Almost all professional DAW’s are under $200. Digital revolution.

    Some of these DSP concepts have made it into the audiophile world. I wonder what the DSP audiophile will be listening to in 10 years.

    Thanks again Arch for taking HLC for a spin and your input during development.

    Keep up the great writings!


    1. Mitch,

      I do Sound Editing/Design for films and TV, having started in the industry back during the analog tape/film days, and doing the slow transition to digital too.

      Frankly, I harbor no romanticism about the days of film and mag/tape.
      Dragging 4 reels through a synchronizer (and inevitably having something jump out of the gangs, to screw things up), cutting and storing the tiniest bits of mag all over the place, scribbling with wax pencils, all the transferring...what a drag. I pinch myself when I think of how much easier, efficient and ergonomic things are now with digital. (Though, yeah, brings it's own problem. Trouble-shooting digital problems can be it's own form of hell).

    2. Vaal,

      I hear you. I don’t harbor any romanticism either. My point was about the pro audio industry transformation from analog to digital and the dramatic cost (and size/weight) reduction going from h/w to DSP software. I wasn’t fond of cutting 2” master tape either :-) I really don’t miss the maintenance and calibration of all of the gear. Calibrating the 24 track tape machine with 24 channels of record and 24 channels of playback took all day, once a month. But I don’t think it compares with all of that mag tape editing!

    3. Thanks Mitch and Vaal,
      Great discussion and nice example of how software has changed the direction in the production world profoundly. Will be interesting to see how in the years ahead software changes will also convert the experience of audio both in stereo and *multichannel* on the consumer end.

      Mitch, I was thinking in our recent E-mails about the time-domain optimization experience. So often we hear audiophile reviewers treat time accuracy as "transient response" or "precision" of percussion attacks. Certainly what I had suspected and which HLC has made clear as I played with the various filters is that the experience is not so much a conscious one like being able to "time" that percussion hit or the plucks on a harpsicord or the rapid strums of a pipa.

      That sense of space, depth, focus is certainly a more difficult one to describe but once you hear it, it's hard not to want to make sure the system still has it!

  4. Hi, great sw, great test. Additional: For around 60 Euro you can use "aroio" image with raspberry pi + dac. There you can switch between several (with accourate made) filters quick an easy (headless control, maybe logitech media server on 2nd Pi or on NAS necessary). BR Jürgen

  5. Sorry I forgot: "aroio" image for free from abacus electronics. BR Jürgen

    1. Thanks for the suggestion AudioWolf,
      I see that AroioOS includes BruteFIR in the distribution... Interesting!

      It doesn't look like there's much (if any) information in English out there on the software. Will keep an eye on this in the future.

    2. Yes, German. Maybe try "Google Translator" (via Google account), like I do reading your very good articles 😉 BR Jürgen

    3. Note that Brutefir allows you to switch filters in real time using the command line interface and telnet. That's how I evaluate filters. I've written a Web GUI that triggers the CLI commands with the push of a button.

      And regarding bass boost preference; I have a separate filter with additional bass boost that I use late in the evening for low level listening. It mimics a loudness control. I think it's often overlooked how your average listening level relates to your preffered level of bass.

  6. Hi Archimago, thanks for a very informative post and the description of HLConvolver! I was recently setting up my desktop system and I agree that the correction done by Acourate (or a similar tool) helps to improve the accuracy and naturalness of imaging. You can find my full story here:

    I've done my setup in stages: first making sure that the reflections are minimal, that the speaker setup is symmetric, then done some PEQ via the built-in soundcard DSP. Acourate was left as the last step for two reasons:
    1) the default length of FIR filters adds too much latency, I've ended up using 'trimmed down' 8192-tap versions, which have less effect at low frequencies, and 2) I never consider PC- or Mac-based setups as "solid" as real hardware because they always tend to self-update and then require a reboot exactly when you don't want to do that, and a major OS upgrade (especially on Mac) can easily break down your DSP tools due to some incompatible changes that the tool vendor hasn't yet worked around.

    The individual headphone calibration topic is actually very interesting! I had some success with Dr. Griesinger's DGSonicFocus app, and by using it + Redline monitor + reverb I could simulate a stereo speaker setup and pass the stereo imaging tests from "Chesky Records Jazz Sampler & Audiophile Test Compact Disc"--on headphones! But in the end it's still hard to convince your mind that the sound is coming from far away from you--it still remains very close to your head (but not "inside" anymore). Also, headphones obviously lack that physical sensation of low frequencies with your whole body that you have with speakers, which again renders the headphone output less real. Plus, the headphones need to be very comfortable and lightweight.

    1. Great post Mikhail. Nice work on a very challenging room situation! Appreciate the ideas and your hybrid usage of both the PEQ and FIR filters; along with the 8192-tap trimming.

      Good point about the non-"solidity" of computer-based systems. A pain every 2 weeks or so when Windows 10 rolls out another update and basically a mandatory update.

      Also painful that about every month or so, I'll do a search and yet another update for the Windows Server machine as well. Enough stories of system vulnerabilities that I basically feel I "have to" update just in case.

      I looked at the DGSonicFocus app but have not given it a try. Glad that you are able to simulate a stereo speaker set-up with that (and other tweaking like Redline crossfeed). I'll need to spend some more time on this with headphones in the days ahead. Mitch's ideas around the KB-DSP certainly sounds very promising. I've had some binaural in-ear microphones for more than a year already and need to play with those to see if I can make some recordings with my own head/pinna. ;-)

    2. There seems to be a difference between where the microphone probe is located in the ear. A lot of researchers (and Dr. Griesinger is one among them) argue that for such tuning of headphones the probe must be located in a close vicinity to the eardrum, like Etymotic ER-7C (, and leaving the ear canal open. Whereas others (such as K. Iida, the author of the book on HRTF: are fine with placing mics at the entrance to the blocked ear canal.

      As for DGSonicFocus—you need to spend some time getting used to compare loudness of banded pink noise. It took me a couple of weeks to start using it efficiently. With probes, tuning could be done easier. That's actually what I'm going to try soon.

      Keep up great posts, Archimago!

  7. I've been using a free VST plugin called MeldaProduction MConvolution EZ for years:

    It also allows to use different filters and quickly switch from one to another. I use it for digital room correction (I adjust the frequency AND the phase).

    1. Thanks fgk, good to know! I currently use HAFConvolver—also free, but I’m not sure if you can switch between filters quickly.

    2. Hey fgk,
      Thanks for link. I'll have a look when I get back home after the time away. Seems like a straight forward plugin (without the volume adjustment fine tuning and level meter that HLC has). Wondering which audio playback software you're using this with? I'm thinking this might be useful in foobar as a VST2 plugin.

      Mikhail, is that a Home Audio Fidelity plugin?
      And is that the "Room Shaper"? Looks like they're also using the Virtual Audio Cable technique for software like Roon with a standalone app.

      Given the "high end" nature of Roon (features and price), it would be nice I think if it could support VST plugins and open up the world of possibilities... But I see the official word seems to be "not in the near future":

    3. Hi Archimago,

      As for MConvolutionEZ plugin, I use it in Foobar in the following way:

      1) I installed George Yohng Foobar2000 VST wrapper as a Foobar component:
      You have to add it to the list of active DSPs in Foobar > Preferences > Playback > DSP manager > Active DSPs.

      2) Install Console ART-Teknika VST (v.1.6.2)
      It's a visual, very convenient VST chainer, it allows to use other VST plugins simultaneously.

      3) You have to configure the George Yohng's VST wrapper so that it would use ART-Teknika's ConsoleVST when it starts with Foobar.

      4) You have to configure ART-Teknika's Console so that it would know when you install your VST plugins.

      5) Make any VST chains you want, using your VST plugins, in ART-Teknika. Just connect VST plugins with virtual wires.

      This is the chain I use when I listen to speakers:

      Some plugins, even though they are in the chain, are in Bypass Mode (StereoTool for declipping and increasing the dynamic range, EQ). I activate them on as needed basis.

    4. Archimago, I only use their free convolver plugin. It's not actually easy to locate info about it on the site, but here is a Google Drive folder that contains both plugins and the manual:

      I host the plugin in Ardour DAW, and route all audio to it via my MOTU soundcard which has an internal mixer.


  8. Archimago

    I may have misread/misunderstood your reference to playing with 'phase' in your article. My mind went to time/phase coherence, but were you instead talking more about maintaining "absolute phase?" As in switching between inverted polarity vs getting back to absolute phase?

    I ask because I know my CJ preamp inverts polarity and they suggest compensating for this by, for instance, switching the + and - of the speaker leads going in to the speakers, to maintain absolute phase.

    The reason I've never bothered doing this is that many have said it's unlikely to make any consistent difference, since many if not most recordings are produced without regard to absolute polarity. So it's still just a crap shoot even if your own system maintains phase.

    Was this the type of phenomenon you were switching between, where you noticed a difference? Wondering if I should bother switching my speaker leads.

    1. Archimago has mentioned creating two sets of filters for Splendor: one with excess phase correction, and one without. The term "excess phase" refers to the difference between the measured phase behavior of the system, and the behavior of a minimum phase system that would yield the same frequency response. There are good explanations on the RoomEQ help site: The main issue which non-minimum phase behavior creates is time incoherence between groups of frequencies. So correcting the excess phase (what Acourate does) helps to make the time response of a speaker to better conform to "ideal" minimum phase behavior where all frequencies are delayed at the same amount.

  9. Hi Arichimago. This relates to your excellent earlier posts on Topping DACs. I've ordered a D10s that's expected tomorrow. From my earlier subjectivist days, I've got a couple of IFI iPower supplies lying around. I'm sure you know how they work. There's an iFI dongle you plug into computer USB port, then you plug iPower into that, and finally plug USB cable running to the DAC into the dongle. This substitutes the iPower for the computer power via USB, while the audio signal passes through as usual to the DAC.
    It's supposed to provide cleaner power by taking computer noise out of the equation. Short question - do you think there's any value in that, or have I wasted my money yet again?

  10. IMO, both the pure and the digital approaches are valid and especially so when used together. If you put "garbage in" (impure audio) no amount of digital correction can restore what was corrupted to begin with. That said, the purer the audio both before and after digital correction the greater the fidelity.

    Hook up an outstanding system to crappy speakers and you'll get crappy sound. Offer an outstanding system low file rate MP3 files and you get the same crappy fidelity. In either case all the nuance will be lost. Common sense recommends blending the strengths of each approach (objective & subjective) to create something greater than the sum of its parts.