Recently I received this excellent question and link about ultrasonic frequencies from the Computer Audiophile site:
Hi, Archimago. Visit your blog frequently and find your posts enlightening and entertaining at times without the usual smoke and mirrors.
What caught my eye in the musings on MQA utilizing the Mytek Brooklyn DAC was the selection of a musical reference which had ultrasonic, specifically musical, content. After reading the paper by James Boyk <https://www.cco.caltech.edu/~boyk/spectra/spectra.htm>, ; I have been interested in the subject of ultrasonics and their (potential) effect on the listening experience fully recognizing that these frequencies are well above the capability of human hearing. Included would be identification of recordings that have musical ultrasonic content.
Given that formats such as PCM (24/96 or higher) and DSD (2x or higher) [Ed: remember that even DSD64 1x can go >20kHz] have the potential to capture musical content above 20kHz, I am intrigued by the possibilities. As with all HiRes formats, I understand that not only all components of the recording chain but also the reproduction chain must have frequency response greater than 20kHz to accomplish this. I do see where speakers are being offered which are spec'd to 40kHz and as high as 100kHz. There are also numerous add-on "supertweeters" being offered which have this capability as well.
IF the topic were of interest to you and worthy of your time and consideration, I for one would be most interested in your musings on the subject.
My apologies for using this Musing as a portal for my inquiry but did not know how else to contact you with the proposal.
FWIW, given the potential of the existing HiRes formats to capture the musical experience if fully realized, I too am less than interested in MQA as the latest flavor-of-the-day.
Connecticut Audio Society
Thank you Frank for the link, interesting discussion and question. I try to do what I can to collect information and synthesize posts to provide hopefully reasonable thoughts on these matters; mixed with some measurements and personal subjective impressions as appropriate.
As is essentially characteristic of all biological traits, there is a "normal" distribution of threshold for frequency detection; some of us will be able to hear to 19kHz, others have ears that poop out by 14kHz. Even if it's not "hearing" the tone, but rather "sensing" the presence of it, this concept remains valid. Since I trust none of the readers here are aliens from another planet who may have extraordinary sensory perception, I believe we can find answers based on studies that are out there exploring this physical limitation of Homo sapiens.
You've brought up one of the fundamental claims in the audiophile world when companies embarked on going beyond CD-level resolution back in the late 1990's - that ultrasonic frequency reproduction improves the audible quality of high fidelity reproduction. This is of course on top of claims that going from 16 to 24-bits made a big difference... For more on the bit-depth discussion, have a look at the blind test from 2014 (and also general concepts of expectations for high-definition audio).
I remember reading about the supposed benefits of ultrasonic frequency reproduction around 1999. "Super tweeters" with response usually at least an octave beyond 20kHz started showing up on the scene; devices like this Fostex T90A horn, or even more easily available the Radio Shack 40-1310 which I have somewhere in my electrical parts box. Offerings grew in the early 2000's with the advent of SACD and DVD-A. Since sampling rates overcame the 44.1kHz (22.05kHz Nyquist) limits of CD audio, the opportunity was there to promote add-on transducers like the Tannoy SuperTweeter, Townshend Audio Supertweeter (2004) or this Audiosmile Supertweeter by 2008. As noted by Frank, there were some multi-way speakers over the years with super tweeters incorporated or the tweeters themselves capable of extended ultrasonics (Tannoy Dimension TD12, Linn Majik 140 for example among others like B&W's with ultrasonic-capable tweeters and JBL M2 rated to 40kHz). Check out the current Madisound list of super tweeters for sale.
Despite this ~20 years history, notice that a minority of speakers these days explicitly aim for flat frequency extension significantly over 20kHz even though many, especially the metal tweeters like titanium and beryllium ones can to varying degrees. When was the last time you saw a speaker with a reasonably flat extended frequency response like -6dB at 40kHz viewed as a major selling point? Stereophile measures speakers to 30kHz but there's no suggestion that this extension is important (rather, it's a nice buffer to ensure everything looks good to 20kHz). By definition, ultrasonic refers to frequencies we humans cannot hear; at least not based on research with typical pure-tone tests.
As a starting point, a good document to analyze which originated back during the advent of "high resolution" digital audio is this Tannoy white paper "The Need for Extended High Frequency Bandwidth - or Why You Need A Supertweeter" from 1999. In there you see the link to the James Boyk measurements as well.
We can see from the white paper 3 major claims:
1. The fact that instruments have very high ultrasonic frequency harmonics - especially piccolos, oboes, triangles, cymbals. Fair enough, nobody is contesting this information from Boyk. It's obviously objectively measurable.
2. Time coherence is of importance... Yes, the concentric speaker system like in the Tannoy (or KEF, etc...) is great. Indeed, if one were to have a super tweeter incorporated into a multi-way speaker, it would be good that it remains time/phase coherent with crossovers typically up at the 5kHz-7.5kHz range. Again, I think nobody generally contests this since it applies to all speakers, not just those with super tweeters (and remember, we can use DSP processing to improve time domain performance in our sound rooms).
3. Human perception of ultrasound is possible... Ahhhhhh, here lies the contentious claim!
Is there any evidence that what are typically considered ultrasonic frequencies (>20kHz) are perceptible?
If you look around at audiophile articles, just like the Tannoy white paper, you will no doubt run across the "Oohashi Hypersonic Effect". Basically, this was a series of papers published at the turn of the Millennium by Tsutomu Oohashi and colleagues, some concepts initially discussed in an AES convention way back in 1991 before making its way to the Journal of Neurophysiology in 2000 with further additions. I'm not going to spend much time talking about this again because I have written and analysed the papers (including a more recent 2014 paper where the researchers used DSD128) back in January 2015's post "MUSINGS: What Is The Value of High Resolution Audio (HRA?)". The bottom line IMO is that this stuff is simply too speculative and clearly it's more complicated and should not be used IMO as proof that reproducing ultrasonic frequencies will always result in a beneficial effect. For example, the recent 2014 presentation actually considered safety issues and both "positive" and "negative" hypersonic effects on EEG activity were reported. My sense is that until there is actual paradigmatic understanding (not just observational reports of neurophysiological change), there is no point debating the Oohashi stuff as a pro or con. (For more on this along with links to other studies, including those that failed to replicate Oohashi, see the Wiki on "Hypersonic Effect".)
As for the other research referenced by that Tannoy whitepaper, the 1991 paper by Lenhardt is worth thinking about. Realize that this research involved placing a transducer directly against the subject's body for bone conduction (see this New York Times article related to the research). The theory is that ultrasonics are not "heard" by the usual cochlear mechanism but rather the saccule which is an inner ear organ of the vestibular system used in balance / positional equilibrium. Although I don't know how far the research has progressed, we can see based on this recent 2013 paper by Kagomiya that they used a ceramic transducer placed over the mastoid bone behind the ear and had the non-deaf subjects determine the "audibility" of "sound" transmitted through a 30kHz carrier. We're not told how much power was needed to create the ultrasonic vibrations which the subjects were able to detect (ie. what SPL in the air would be equivalent to cause these vibrations?). Suppose we accept that ultrasonic stimulation through bone conduction is beneficial for high-fidelity appreciation, how loud does a 30kHz signal from the super tweeter need to be in order to vibrate our skull so that it even has a chance to be appreciated!? Perhaps someone can let us know but I have a feeling the typical amplitude of ultrasonic material during an acoustic musical concert is far less than the amount that the researchers in this experiment utilized (obviously I'm not talking about lo-fi ear-splitting amplified rock concerts filled with distortions and all kind of wide bandwidth noise).
Finally, the Tannoy whitepaper speaks of a 10dB peak at 30kHz being audible in their internal testing. So did they publish this finding? If not, why not?! This would be fascinating, would have prompted replication research, and likely will spur on sales of their SuperTweeter. Talk is cheap, Tannoy.
Remember that recently there was also the meta-analysis by Joshua Reiss to determine perceptibility of high-resolution audio (Journal of the AES, June 2016). He cites 4 papers (out of 31 utilizing auditory experiments) which suggested the potential for hearing >20kHz. Two of these 4 are from Ashihara. Have a look at this 2006 paper where they tried to determine hearing thresholds beyond 20kHz for 15 subjects.
For convenience, here is Figure 6 from the paper where they showed the results from the 4 out of 15 subjects with best high-frequency hearing acuity:
So far, we see little evidence that ultrasonic frequencies are actually audible by assessing the claims at least put forth by the Tannoy whitepaper. Likewise, I see no evidence that typical audio playback could result in bone conduction to any significant degree as would be suggested by the Lenhardt work.
Here are a few further points to consider:
1. Again, only the young can likely hear up to 20kHz with any significance. Let's be honest guys. By the time we're 35 years old, on average, hearing acuity of an 8kHz tone is somewhere on the order of -10dB compared to 1kHz. We basically don't hear anything by 20kHz. Remember that the ladies among us are blessed with more graceful decline in acuity. Even the Ashihara paper above showed only a minority (6/15) of young folks up to 33 years old had a measurable threshold to 22kHz (Nyquist limit of CD), and nobody in their population was able to hear pure tones beyond 24kHz up to ~90dB SPL. It is unfortunate that they did not publish demographic information like the mean age or gender of those they found with best high frequency acuity.
The evidence at least based on pure-tone analysis suggests that a sample rate around 48kHz is the absolute most that any adult really can even have hope of perceiving; any frequencies that need a higher sample rate is beyond benefit barring the vague Oohashi stuff. As Figure 6 above also showed, if there were noise or other frequencies in the signal, masking happens and makes detection even worse. I think we can round up to 50kHz PCM sample rate and state with good confidence that this is all we need to capture for human consumption in the frequency domain.
2. Microphones typically are bandwidth limited to around 20kHz. You can easily see this in the vast majority of recordings (synthetic sounds of course can have all kinds of unintentional ultrasonics). An example of what one typically sees from a pop/rock studio is something like Joni Mitchell's "Both Sides Now" from the DVD-A released in 2000 (24/96):
Coincidentally, the graph also demonstrates just how noisy this recording is. Although it's presented as 24-bits on the DVD-A, there's no evidence that it needs anything more than 16-bits given the high noise floor clearly way above -96dB. Furthermore, in the production chain (I don't know if it was recorded to analogue tape, went through an analogue mixer, or had a noisy ADC), there is a rather high level 29kHz noise peak. Clearly this is not part of the music, nor would anyone claim they "should" hear this. So why bother reproducing this ultrasonic signal?
Even in bona fide high-resolution recordings, ones like this 2L DXD sample found here (I downsampled it to 96kHz to better demonstrate the roll-off in the recording at one of the more dynamic segments of the music):
Clearly this is a much better recording than "Both Sides Now" with very low noise floor so 24-bit resolution could be reasonable. But time and again, this is what you see in the spectrum of essentially all recordings. There's just nothing much up in the high frequencies beyond 20kHz in the music we buy; even those that are "high resolution" with high sample rates like 192+kHz.
I mentioned a bit more about microphones in the post last year: "MUSINGS/ANALYSIS: Is there any value in 176.4 and 192kHz Hi-Res audio files?".
Finally, just to complete this illustrations, here's a very good sounding SACD - Christina Pluhar & L'Arpeggiata's La Tarantella: Antidotum Tarantulae:
I'm showing the FFT at one of the most dynamic parts of the music - notice how the high frequency naturally drops off. As another natural sounding acoustic recording, this time to DSD, notice that there's no evidence of anything significant beyond 25kHz being recorded by the microphones. Without filtering, the DSD64 quantization noise is rather nasty as well, obviously adding nothing of value to "high fidelity" sound...
3. Adding super tweeters make the speaker more complicated and increases risk of poor integration. More expensive, potential for suboptimal cross-overs. Why bother unless there is proof of value?
4. No transducer is perfect. Speaker non-linearities result in intermodulation and subharmonic distortions that may be audible <20kHz. This is reason not to record too much ultrasonic content nor try to reproduce it. This is especially true with ultrasonically noisy content like unfiltered SACD/DSD64 as above which is part of the noise shaping used in the technology. Large amounts of noise amplified and sent to the speakers in no way provides benefits to the audiophile who desires true high fidelity! (Refer to Monty's "Intermod Tests" and have a listen to demonstrate to yourself why too much ultrasonic content might be bad in your own system.)
5. High frequencies are attenuated through air. Let's just focus on this for a bit because I think it's interesting and perhaps we don't think about it enough. Tell me, friends, how many of you enjoy listening to an orchestra or jazz band sitting from a vantage point just 4 feet in front of a cymbal, triangle, or trumpet? Nobody, I hope :-). Realize that this is the distance the measurement microphone (a 1/4" B&K 4135 condenser) was placed in the Boyk article analyzing the high-frequency content of instruments.
Assuming we could hear the ultrasonics, and our microphones are able to record >20kHz at high fidelity, and the playback chain including the speakers can reproduce >20kHz frequencies accurately, there is still significant attenuation of these high frequencies due to the air between us and the speakers. In fact, you can have fun calculating this here. As I type this on a sunny day in Vancouver, the humidity indoors is around 40%, and it's about 20°C room temperature in my sound room. Calculated attenuation per meter at 20/25/30 kHz look like this for "atmospheric absorption":
Considering that I sit 3m (about 10 feet) from my speakers, this means that at 20kHz, the sound reaching my ears attenuates another -1.7dB, at 25kHz it's -2.3dB, 30kHz -2.8dB, and -3.7dB by 40kHz. Here's a graph demonstrating the attenuation curve 10' away in a 20°C, and 40% humidity room as an average benchmark for absorption in this part of the world (typically lower temperature and humidity will increase absorption):
Comparatively, this is not a huge amount at 10-feet (3m) of course. However, if we're talking about frequencies produced by real instruments, and we're trying to replicate the sound of an actual performance at a venue, we must ask ourselves then the question "what distance should the recording sound like it's coming from?" This is important for the tonal balance of high frequencies due to the non-linear disproportionately high amount of atmospheric absorption.
As someone who enjoys going to the orchestra, when I listen to the Vancouver Symphony at the Orpheum Theatre in downtown Vancouver, the instruments are clearly much further away than the 10-feet or so between me and the speakers in the sound room. Suppose I sat front-and-center at the Orpheum, I would estimate that the guy playing the cymbal and triangle emitting all those ultrasonic frequencies at the back of the orchestra would be at least 40' away. Assuming 20°C, 40% humidity, this means that a 20kHz tone would have already attenuated by at least -6dB even if this was direct with nothing but air in the way. By 30kHz, there's >11dB of loss at this seating position; again without all the rows of musicians between me and the cymbal emanating >20kHz material. That's assuming I'm sitting close to the orchestra as an audience member - how much worse is the attenuation sitting further back with the bodies of other patrons in the rows in front of me?!
It's worth thinking about this idea of perspective as a listener (not just for the frequency response reason of course). If our home stereos are supposed to reproduce sounds in "high fidelity", based on what sounds "natural" around us, the reality is that acoustic music (IMO the type of music most appropriate for "high resolution" reproduction) heard as an audience member naturally rolls off the highs. This is potentially why research into "target curves" or "house curves" used in room correction tends to de-emphasize frequencies from 10-20kHz. An example is the "Harman Target Curve" (as described in this 2015 paper):
This data was produced empirically by allowing subjects to use tone control to find subjectively preferred tone curves starting with a flat-calibrated room/speaker system. Notice how trained listeners preferred a gradually descending frequency response of the speakers. Wouldn't you think that if ultrasonics were important, that experimental results, especially with trained listeners would at least suggest flat frequency response to 20kHz for most music?
I've often thought about the rationale for preference of rolled-off high frequencies in these target curves (the classic B&K curve has similar roll-off). I wonder whether close-mic'ed studio productions may be adding to this these days. Remember, we typically sit many meters away from the artists in a live performance. Our ears/mind naturally expect high frequency roll-off in such a venue and would also expect the same in the home sound room. When artists are recorded in a studio, close-mic techniques where the microphone is placed often a foot away from instruments could result in a rather unnatural tonal response picking up more high frequency energy than a normal listener would hear from many feet away - the kind of roll-off demonstrated by acoustic recordings like the Magnificat and La Tarantella tracks shown above.
High-fidelity speaker systems capable of flat response to 20kHz may actually sound too "harsh" or "analytical" when playing these close-mic'ed studio recordings. Who knows, maybe this is why some objectively "inferior" speakers with early high-frequency attenuation can sound natural and more "musical" in certain situations. Perhaps this is also why suboptimal digital converters like NOS DACs and those using early roll-off filters like the PonoPlayer could be preferred despite the imaging distortions above Nyquist that come along with those filter settings. The idea about close-mic'ed studio recordings and harshness is speculation on my part, so I would be interested in others' thoughts. Considering recent developments, I think it is possible that it's not "time domain" quality and short impulse response graphs that are important when looking at digital filters... Rather, it's the high frequency roll-off that can sound more "natural" (eg. PonoPlayer / Ayre filter starts rolling off by 7kHz and about -4.5dB by 20kHz for 44.1kHz material). Perhaps these digital filters are acting as mild tone control compensating in an era where our hi-fi gear no longer have those control knobs any more.
For completeness, I know that some vinyl lovers will talk about the superiority of frequency response compared to CD. While it is true that a good quality LP can contain frequencies well above 20kHz, it's not like you see much ultrasonic content on most LP's. My personal experience with vinyl rips whether it's with Denon DL-110, Shure M97xE, or Ortofon Cadenza Black cartridges have shown that LP playback typically rolls off the highs up to 25-30kHz reaching the noise floor with little content above. I have no qualms with down-sampling my vinyl rips to 48kHz these days.
The Bottom Line...However one looks at the evidence, I think it's fair to say that the likelihood of perceiving (not necessarily even hearing) a difference in sound material >20kHz is simply dubious. Then there's the question of whether this actually benefits the sound even if perceptible.
Yes, while there are the occasional research papers suggesting differences in audibility between sample rates (like this one comparing 88.2kHz vs. 44.1kHz from 2010), I have yet to see clear evidence that ultrasonic frequencies themselves have resulted in audible differences as opposed to sonic differences because of the DAC or ADC operating at different sample rates.
Among the research, I found the negative NHK study (Nishiguchi et al. 2009) particularly fascinating using the Pioneer/TAD PT-R9 super tweeter, B&W Nautilus 801 speakers, dcs digital gear, Sony and Marantz amps. In that study, they used 36 subjects ranging from teenagers to those in their 50's; a huge proportion (33/36) with audio engineering experience, 6/36 women, and the musicians who recorded the test audio also participated. There was no evidence of a difference whether the super tweeter was activated with >21kHz content. Interestingly, one 17-year old female subject actually did really well in the research trials but subsequent further trials with this individual did not pass statistical levels of significance.
Yes, I know there are some folks with strong testimonies out there. I remember interacting with a "Golden Ear" who claimed he could hear up to 30 kHz (with no evidence). Anyone can have an opinion but opinions aren't facts. Even if this fellow tried a 30kHz tone test, it's possible that he heard harmonic distortion below 20kHz but didn't know it. More often than not, proponents of super tweeters claim that they improve the "air" of the sound system, improve the spatial dimension or clarity of the treble, some even claim they improve the "transient" accuracy of bass - vague claims indeed. But time and again, the scientific literature questions the audibility and benefits wherever we look. Whether it's our human physiology (which deteriorates with age - especially frequency response), recording equipment specs (like roll-off from typical microphones used), the music signal not likely containing much >20kHz, or even the air we breathe absorbing high frequencies, all these factors collude to limit the likelihood of significant ultrasonic content in live acoustic performances, within the recorded music itself and ultimately potential audibility.
My personal experience resonates with the science. For years now I have been examining "high resolution" albums from SACD, DVD-A, Blu-Ray to digital downloads and have rarely seen what looks like actual recorded ultrasonic content especially in good acoustic recordings. I have done my own ABX listening tests with what should be very high quality recordings like those from 2L on different systems (such as discussed here). These days, in my mid-40's, I just see no point in purposely pursuing a system with ultrasonic frequency response. A system with the ability to reproduce sound reasonably flat to around 20kHz is great. Beyond that, I'll do my own tweaking of the room, and find the best sounding mastering of albums...
If we take Ashihara's research in pure-tone audibility in young people as true, we might say a sample rate of 50kHz is absolutely all we would ever need. In fact, this is why I honestly think that for the purpose of streaming "high-resolution" audio these days, what's wrong with just 24/48 FLAC? This is what I would prefer instead of a restrictive scheme like MQA.
Having said this, as a "perfectionist audiophile", I like the idea of high-resolution in the sense that deeper bit-depth (ie. 24-bits) and higher sample rate to 88.2/96kHz (because of universal compatibility at these sample rates) will allow us to capture all that decades worth of research reports have shown to be the limits of human hearing and more. As one who likes to "own" my own music collection, this would be my preference. Whatever arguments are being made about digital filter effects would be moot at the 88.2/96kHz sample rates as well. Realize that playback even of 88.2/96kHz material could result in non-linearities with poor speaker systems. So long as the high resolution recordings do not contain inordinate amounts of ultrasonic noise and respect the natural high frequency roll-off, this should not be an issue. No need in my mind for sample rate higher than 96kHz as a consumer.
Despite claims over the decades, I have yet to see super tweeters considered "must have" features of most speaker designs. Likewise, I'm not sure if there is any data to support the idea that headphones capable of frequency response significantly above 20kHz are considered to sound "better". Given the mere centimeters distance between the headphone transducer and the auditory organ, ultrasonic frequencies would be essentially free from air attenuation. Also, why is nobody (I know) asking for hybrid air/bone conduction headphones that can handle up to 20kHz or so by air conduction and offer ultrasonic stimulation through bone conduction* (as per the Lenhardt research)?!
Needless to say, I believe we can all enjoy the music fully without "super tweeters". No need for concern or regret; fearing that we're missing out.
I hope this provides some food for thought, Frank... Cheers and greetings to the Connecticut Audio Society.
*BTW, there are bone conduction headphones out there like this one, but if you look at reviews, one cannot expect sound quality to be as good as air conduction even though they do provide benefits (eg. low isolation so you still hear what's going on around, possibly easier for fit and comfort in some situations).
Over the years, I have wanted to do a blind test comparing content with >20kHz vs. filtered audio for you guys (something like the 24-bit vs. 16-bit Internet Blind Test in 2014). Unfortunately, it would be very difficult to ensure adequate blinding because anyone these days could pull up the file in an audio editor or see results on a spectral analyzer and determine which is which. I would need to ensure subjects are honest in order to have faith in the results if we did such a trial :-)...
Well, Dynamic Range Day 2017 just passed on March 31st. Always good to remember that at the end of the day, quality of the mastering (of which a big part is the retention of good dynamic range) is an essential piece of the joy of high-fidelity sound! While you're on the web site, do check out the Loudness War Research page for some great info.
This week I've been getting into the guitar blues of Hubert Sumlin (1931-2011). Perhaps best known as a member of Howlin' Wolf's band, this guy "rocks" on his solo work as well. His 1998 album I Know You (DR13) I found very enjoyable.
As usual, have a great week ahead everyone and hope you're all enjoying the music!
In response to jhwalker below in the comments about DSD and the 25kHz roll off... Not true, DSD64 can encode frequencies way beyond 25kHz. Remember those SACD ads claiming 100kHz frequency response? Those ultrasonic frequencies however then get buried in the rising noise floor - this is the price to pay for low-resolution 1-bit sampling, and why high sampling rates in the MHz range are needed. Many DAC's will implement analogue filtering so the amp/speakers don't suffer through trying to reproduce all that high frequency noise! Software like JRiver for example will by default also digitally low-pass filter DSD --> PCM playback around 24kHz.
The La Tarantella DSD track was converted with no filtering done to the conversion. Here's an example of "synthetic" music with >25kHz components in DSD64:
As you can see, that's from the Beck album. Synthetic instruments, close mic recording, studio tricks with all kinds of ultrasonic frequencies (likely noise), much of it buried below the typical DSD64 quantization noise from 35kHz onward.
Addendum 2: (April 2, 2017)
A friend just E-mailed to me his disappointment with Bob Dylan's Triplicate available as 24/192 at the usual places. Apparently there's a nasty 27-28kHz noise that runs through the songs:
I don't know if others have "heard" this. Assuming nobody has complained, I therefore assume a frequency like this above 24kHz is inaudible and undetectable.
Sadly, this is yet another example of why much of the pop/rock "high resolution" albums IMO are worthless. Why pay more for 24/192? With a noise peak like that, might as well downsample the whole thing from 192kHz to 48kHz. Plus the noise floor doesn't demand >16-bits, and the album is compressed to DR8 so by all means also dither it down to 16-bits.
Another informative post. Thanks.ReplyDelete
F.Y.I. Hubert Sumlin's Healing Feeling, featuring the great Ronnie Earl, is fabulous - one of my all time favorites. Don't know the measured DR but the sound quality (44/16 CD rip) is quite good.
Thanks for the suggestion Mark! I'll check out the album :-).Delete
Great write-up, as usual.ReplyDelete
WRT the L'Arpeggiata recording, though: isn't 25kHz pretty much a normal rolloff / filter point for DSD64, in any case? Pointing out that a DSD64 recoding rolls off at 25kHz is basically like saying a CD rolls off at 22.1kHz - yes, and?
IOW, it's entirely possible had the original recording been done at DSD128 or DSD256, etc., there might have been substantially more supersonic content, with filtering beginning at a much higher frequency.
Just a thought.
I've added a little "Addendum" on this.
Remember, DSD64 is capable of capturing higher frequencies than 25kHz quite easily. The problem is noise drowning out the signal...
Agreed, it *can* capture beyond 25kHz - but I think you and I are saying the same thing: the rising noise past 25kHz (or so) for DSD64 means that you might as well start your filtering around there, because anything above there is effectively lost in noise.Delete
Which is why everything should be recorded in DSD512 or higher ;) LOL
Sure, we can agree that DSD64 can be filtered at 25kHz. But I think the 'La Tarantella' recording highlights the fact that these natural sounding recordings generally roll-off quite quickly after 20kHz even in louder places with percussions going... Even if this were a DSD512 recording, I doubt there would be much to capture.
Yeah, if I ran a studio to record hi-res stuff, sure, DSD512 and 384kHz PCM would be great :-)... But I wouldn't argue that at home I would need DSD512 or PCM 192kHz+ wasting my hard drive space!
Some things to consider when looking at the Joshua Reiss analysis.ReplyDelete
It shows the audible threshold of artificially generated tones (not music).
It is clear that 4 individuals could well detect up to 14kHz with an SPL between -5dB and +10dB.
It probably took quite some time of having total silence around them to be able to do so in these cases.
For those that wonder how one can hear below 0dB SPL. 0dB SPL is considered the average threshold of human hearing, NOT no sound pressure at all. 0dB SPL is still 20 micro Pascal of sound pressure and is just a chosen reference point.
The research did show some people 'detecting' a presence (feels more like a pressure on the ears/head than an actual tone) of 20kHz at 80dB SPL.
Up to 24kHz needing 90dB SPL.
Quite interesting in the sense that it is objectively possible to hear up to 24kHz. 20% above the considered audible range no less.
I remember from my own experiments, in my much younger years, that I could 'detect' up to 21kHz when that tone was played very loud.
Now it starts to cut out at around 17kHz for me while my kid is covering his ears in agony.
To put the analysis in perspective (when it comes to music) one should have a look at the amplitude of high frequency contents in recordings with lots of actual harmonics.
The high frequency energy in recordings is easily 30dB to 40dB lower in amplitude relative to bass and mids.
This means that in order to 'detect' these >20kHz frequencies (@ 90dB SPL) the bass and mids in a music signal would have to be 120dB to 130dB SPL in order for that >20kHz content to become just 'barely detectable'.
The rest of the music is just happily pounding away on your eardrums at extremely loud levels when the >20kHz content would be 'just noticeable'... when coming from silence.
Detecting artificially generated tones is one thing, but we are talking about high frequency contents in MUSIC and the audibility of that high frequency content which may well yield different results.
Some research has shown that in order to actually have a beneficial effect in SQ the >20kHz content must be harmonically related to audible fundamentals and/or overtones.
This is not the case in single-tone tests which some research may already have 'shown'.
While I am convinced some people have better trained ears than others and/or have better hearing (less damage over the years or being much younger) I doubt the claimed hearing acuity of some 'audiophiles' is as good as they 'know' it is.
All I know is that all of my own experiments on this subject have shown me that I have no benefits from playing files having content well above 20kHz.
For others the possible 'ease of mind' effect when knowing they play higher res files may give them more enjoyment.
For me it just allows me to carry more music around on cheap memory cards and saves me space on hard discs.
On required bandwidth:
As Frank Zawacki and Archimago already mentioned is that to reproduce 20kHz on a proper level/quality the actual bandwidth of the transducers (microphones/speakers/headphones) and amplifiers SHOULD exceed 20kHz.
This alone is a good reason to have gear (amplifiers, transducers) that has a bandwidths of at least 30-50kHz so it can acurately reproduce 20kHz bandwidth limited files.
That fact doesn't mean you also actually need music with contents far above 20kHz, except perhaps during recording to allow for post processing.
Appreciate your analysis and personal experience as well!
Absolutely agree. Although I would advocate not to worry about crazy high samplerates and don't think there's a *need* for frequencies more than say the 24-25kHz range in the music, on the hardware side, it's good to have the leeway above 20kHz! By all means, let the speakers roll off in the ultrasonic range and these days, most amps have no problems handling 50kHz+ (but wise still I think not having a ton of ultrasonics for it to try to reproduce!).
Many interesting facts here, but it has been mentioned that higher samplerates allows simpler and more gently digital filtrering which has less impact on the impulseresponse.ReplyDelete
I have also a Teac UD-501. With my former amp and speaker, the NOS - filtermode was often to dull for my liking, but with my current setup Wharfedale Denton/ Rega Elex -R it sounds natural, but sometimes a bit laidback.
Thanks for the comment Mats.Delete
Yes, the higher sampling rate will allow simpler, and gentler filtering. This is why I think for hi-res music I download, 88/96kHz would be way more than enough for the gentle roll-off.
I discussed filtering a number of months back:
Remember that an "impulse" is not really a "sound" and the only utility is to demonstrate the filtering algorithm; long or short duration of impulse correlates with the complexity but not ultimate sound quality as if there is some 1:1 correlation. Any worries about "ringing" and the like will be up in the ultrasonic range in any event...
Remember, good music should be properly bandwidth limited so there should never really be "ringing" (pre or post) in well mastered music. Some very well regarded companies will use steep filters and sound great.
I assume the NOS roll-off synergizes with your Wharfedale/Rega. That's great and I certainly think that use of different filters likely because of the frequency effects can work better in some cases!
Firstly, Archimago, despite my inability to sign into blogspot.ca, I would like to thank you for taking the time to give considerable consideration to my inquiry re ultrasonics. Much appreciated.Delete
While you present a great case against the efficacy of trying to reproduce music content in the ultrasonic range, I will be experimenting with the Taket BatPure modules < http://taket.jp/batpure/batpure.html> to see what I can "hear", fully recognizing that my own hearing limitation of tones is 14kHz. Not a large investment for nebulous results. Just need to identify HiRes recordings with "real" ultrasonic content rather than noise artifacts. I have some unmixed/unmastered recordings made with the Sony PCMD100 which may be a good place to start.
There are a few things to consider when adding a piëzo tweeter (which the add-on is).Delete
1: The frequency response is listed as 20kHz to 150kHz but is probably already doing something at lower frequencies as there is no crossover filter needed for Piëzo tweeters.
Thus there may also be extra signal present at audible frequencies below 20kHz which you could be hearing. That doesn't mean that changes in sound are due to extension of the frequency-range but may well be due to more amplitude in the audible part of the upper treble.
2: The efficiency is not mentioned, the spec: 70dB/1m says nothing without mentioning a voltage or power combined with impedance. A meaningless specification.
The other thing they say is: The effect is appeared even in the speaker of efficiency 100dB/1m.
Strangely enough they leave out the wattage and fail to mention the impedance of the speaker used).
This means the Taket is probably louder than the tweeter in your current speaker. In that case there will certainly be an audible effect.. not due to more extension but due to a higher SPL in the treble.
3: Also keep in mind that the sound from the tweeter will interfere with the sound from the Taket. This means sharp nulls and peaks at different frequencies, also dependent on listening distance and distance from the already present tweeter. This too may be audible but doesn't mean the effect you hear is due to more extension.
I agree with Frans, I'd be careful with the interpretation of those specs. At 5,800 JPY/pair, that's <US$55 so it is indeed inexpensive to try.
If you have a measurement mic, it would be interesting measuring the sound from it to see what frequencies are actually produced especially <20kHz in the audible range. Likewise, it is important to see how well the device integrates into your speaker system...
Remember, it's just as likely that this could *harm* the quality of the sound from a good system as enhance it. Cheers!
BTW, issue with not being able to successfully REPLY appears to have been use of Firefox. Explorer works fine. :-)ReplyDelete
One a the few arguments I could conjecture up in favor of > 44 KHz is that you don’t need that steep brickwall filter right after the 20KHzReplyDelete
What are you thoughts on this?
Yeah... The concept and general negative connotations of the dreaded "brick wall" filter. Remember that a well regarded company like Chord utilizes one of the strongest "brick wall" filters I've ever seen - check out reviews of the Mojo and Hugo using their custom FPGAs. Digital upsampling precision is excellent these days.
I don't know if the brick wall was that bad back in the day (?80's, ?early 90's), but I have heard the Chord DACs at shows and they sound great even with 44kHz material where the effect would be most obvious.
I'll see about putting an article up on this in more detail in the days ahead...
If the date of this post is central to its credibility, then you have been too clever for me!ReplyDelete
IMO Oohashi's hypersonic effect needs to go in the box with cold fusion and other experiments that are "desperately in need of verification but only seeing contra-indications". When a new experimental finding contradicts a large body of prior research, the first thing needed is independent verification. When that doesn't happen, it goes in the category of outlier or anomaly, and best quietly ignored by engineers and implementers.
Don't worry about the release date :-). In the world of audiophile postings, almost every day is 4/1.
Thanks for bringing up the "cold fusion" report of the late 80's from Fleischmann and Pons. Those were my high school years - wow, how time has flown!
As much as I don't want to drag out the Oohashi stuff, considering how even these days, some people will bring in up even if tangentially as if this material seriously adds to our understanding, there's no option but to address it directly.
I find it amazing that the audiophile press all these years never question the "effect". Of course, why would they? It helps perpetuate the mystique for those who have faith...
a question totally unrelated to the (excellent) post...
It seems that you like Roon. Why ? What are the features that make it (in your opinion) excellent ?
Me (given that I'm not interested in multi-room) I see none.
I have already expressed my opinion about "favorites", talking about them as a new way of "owning" albums: non more physically on my site, but "on the cloud".
So, "owning" 1000's of albums could become not so unusual. How you handle them ?
Me, I handle and play my collection of "liquid" music (ripped CD's and not only) using JRiver Media center.
Don'you think that something similar would be beneficial for the task ("I would like to listen the Third Mahler Symphony, conducted by ... I don't remember let me see the list ...").
I'm writing something like that. Qobuz offers excellent API's. Then I shall use BASS and SQLite to handle meta-data. Worthwhile ?
I can say that I appreciated trying out Roon (remember, I don't have a subscription any more) and can see the educational benefits in the connected metadata that they provide. As I noted in the Computer Audio Part II post, if I were starting out and had money to spend, I can see the benefits of getting a Roon membership.
Most of the time, I still play music primarily through LMS. I started my collection in earnest with the Squeezebox system and still like how this is working out. These days I have multiple computer systems in the home and some have JRiver also. My workstation computer has Muso installed as well... As you can see, I'm pretty "agnostic" with what I use.
Thanks. If you appreciated the information provided by Roon, you should have a look on Qobuz. Too many times the information are the same, and with Qobuz you pay only for the music.Delete
As I said WASAPI Exclusive doesn't work well, but if you talk ASIO with JRiver Media Center then WASAPI everything work seamessly.
More: the "discovery" is a pleasant journey.
All that at the same price of Tidal, and without the "Roon tax"
I remember when I was much younger, reading about the so called benefits of an ultrasonic response, and experimenting with super tweeters / piezo tweeters etc. and I truly couldn't tell any difference, both with analogue and digital recordings.ReplyDelete
I'm beginning to wonder if in 'x' years time, we suddenly realise that 16 bits really was enough.
Published on an interesting date by the way! (01 April)... :-)
:-) Same with me Tony.Delete
If you would have asked me about >20kHz in 2000 when SACD, DVD-A, and super tweeters were first coming out, I would have been rather hyped about the excitement of all this. Alas, over time, like yourself, with experience, my own experiments and measurements, I realized that it didn't matter.
Nothing wrong with a *good* 16-bits either.
I didn't see many good April 1st pranks this year. The world is simply getting more serious it seems!
Great article. Relieved to see you are not onboard the Oohashi bus (he's kind of a *crackpot* actually, if you look up all of his published work and resume). But your link to the Harman (Dr. Floyd Toole) AES paper on target curves isn't working (for me at least) -- but this one doesReplyDelete
Thanks for the note Steven!Delete
Will update the link above...
Re DSD, nota bene that , at least in the days when SACD players were being pushed as the new thing, it was common for them (and recommended in the Scarlet Book spec) to low-pass filter the output as a final step, at either 50Hz or 100Hz, to eliminate at least some of the ultrasonic nastiness. I don't know if this is still true of hardware SACD players.ReplyDelete
> 20 kHz we really cannot hear. 88.2/96 kHz has sense mainly for 2x upsampling from 44.1/48 kHz since many common DACs oversample much worse that Sox or Ferocious Resampler https://github.com/jniemann66/ReSampler/releases/tag/v1.2.6 do. Or for recording that is then used for CD creation. > 96 kHz sample rates are total overkill and have no benefit. Recording at 24/48, 24/88.2 (if pure CD output is desired) or 24/96 (if multiple output like computer playback, DVD and CD is desired) is the gold standard and should not be pushed further. With good plain TPDF or sloped or moderately noise shaped dither, even 16 bit is very well usable for playback on common equipment.ReplyDelete
Yes, agree Honza.Delete
P.S. especially 16/44.1 - 24/88.2 or even 16/88.2 (with redithering plain TPDF or sloped TPDF) sound very well - not because they contain additional information, but because many common DACs do not filter 44.1 as well as 88.2. Even upsampling also preserve half of original "time points" (even if I know that PCm is not time-aware). Anyone can try it by using Ferocious Resampler (link above) and upsample 16/44.1 to 24/88.2 or 16/88.2 for mobile. Theoretically even 96 kHz can be used for that purpose but the results are not that elegant then. Of course the upsampled version does not contain any sound above > 20 kHz, but that does not matter.ReplyDelete
Wow! Based on the learned comments regarding the efficacy of reproducing ultrasonics, I feel like I have been living in a cave...without cold fusion. :-D Not sure why but most of the articles I came across in googling on the topic over the past year were more positive than both this Musing and the subsequent comments. Appreciate the links to additional white papers on the topic. Glad I only spent $75 on the BatPure modules! ;-) FWIW, my speakers are Martin Logan CLX Art e:stats which are spec'd to 23kHz. All of my system components have Frequency Response well into ultrasonic range making them more than capable of reproducing 20Hz-20kHz and beyond assuming there is anything of value.ReplyDelete
Really nice speakers... in a totally different league then the add-on piëzo tweeters for sure.Delete
The previous phase issue remark is not applicable here because of the large height of the membrane.
At 90dB/1W (in 8 Ohm)/1m of the speaker means the piëzo will be louder. Possibly 5 to 10dB. 10dB is about a doubling of perceived loudness.
The tweeter thus will be a LOT louder than the CLX.
I noticed at 20kHz the CLX Art is around 0.7 Ohm (not a typo). This requires something like a Krell to drive it properly. When connecting the supertweeter I would run its wires from the speaker connectors in this case and not directly from the amplifier.
This to utilize the possible attenuation from the speaker cables in this particular case.
While playing music put your ears close to the supertweeter. When you can hear sounds from it this may skew the results.
Also, when possible, put a switch between the tweeter so someone else can switch it on or off without you knowing it is on or not.
Ensure you do not KNOW if it is on or not otherwise your brain will be biased even more than your e-stats membranes.
Blind testing may be important in this case.
Thanks, Frans. I have been a "poster child" for Martin Logan going on 25 years. I owned M-L Monoliths for 20+ years before the CLXs. I also have Aerius I and Theater (center) in my video system.Delete
I am driving the CLXs with Pass Labs XA160 (160wpc) monoblocks. I have been running his amps for about 30 years going back to Threshold days. While the M-L designs can be driven by ~75wpc Pass amps, the panels DO sound much better with more power.
Installing a switch on the STs is a great idea and definitely more convenient than disconnecting them! Maybe even an LED indicator so I know whether they are On of Off.
Reading articles on recording the sounds made by bats, apparently there is a way to record them and then convert to sound to something audible. Problem is, the cost would far outweigh my ST investment. :-(
I am a bit confused about your comment re the tweeter being a lot louder than the CLX?! The supertweeters have a Sensitivity of 70dB (and assuming 1W/M) so aren't they less sensitive by 20dB? Since it was suggested that I stay away from step-up xformers, I planned to put an integrated amp with 30dB gain between the preamp and the BatPures to mitigate the difference in Sensitivity. With the integrated set to some mid level of gain, I thought the STs would "track" the gain being sent to the CLXs.
Agree with Frans... See if you can attach a switch and test out the super tweeter effect. Like I said earlier, if there's a way to measure the output, the result might or might not be surprising; plus you'll get an objective look at how much output is actually coming from it.
Regarding your comment about Google and "more positive" sentiments around the effect of reproducing ultrasonics you've found, I think this says something about the "literature" that audiophiles are immersed in.
Without a doubt, if we look at audiophile magazines and most blogs, forums, Industry web sites, it's almost always considered "positive" to have >20kHz frequencies. We can imagine the reasons. But of course very little of that material shows any objective results or provide evidence in a controlled fashion. However, the moment we switch gears and start looking at the academic literature where researchers control variables, utilize appropriate blinding, and systematically examine the facts, it's rather evident that there are reasons why we typically talk about "20-20kHz" as the audible range.
Remember, this goes all the way back to CD audio and the listening tests done even back then by folks at Sony and Philips (late 1970's) when they chose 44.1kHz as the sampling rate for their "revolutionary" digital audio format.
In any event, have fun with the experiments! And by all means open up dialogue with the other audiophiles at the Connecticut Audio Society and see what others think...
This comment has been removed by the author.Delete
"I am driving the CLXs with Pass Labs XA160 (160wpc) monoblocks."Delete
The XA160 is specified at 8 Ohm and 4 Ohm only, but the doubling of Wattage, 160W to 320W with halving of impedance, says something about decent over-dimensioning of the amp and power supply. No way to tell what the amp does with 1 Ohm impedance but will probably be better than most other amps out there.
" The supertweeters have a Sensitivity of 70dB (and assuming 1W/M) so aren't they less sensitive by 20dB?"
That part is a bit confusing as the spec '70dB/1m' means nothing. It should have stated the voltage. One can assume 70dB @ 2.83V @ 1m but may well be 70dB @ 1V @ 1m.
On their website they also say: "The effect is appeared even in the speaker of efficiency 100dB/1m."
So it seems to either play louder than the 70dB spec says or the 100dB/1W/1m is incorrect or there is a person merely claiming he heard a beneficial effect. This is why I mentioned the suspected higher sensitivity (more on this further down)
There are no plots and there is no real technical data on the website either. In any case the info is a bit sketchy. It states it can handle 150V which means (when this is RMS) you can safely connect a 25,000W amp and not blow it up. Even 150Vpp still would mean 50Vrms = 300W(8Ohm) or 600W(4 Ohm). More likely it can handle 150W / 8Ohm = 19V. Most Piëzo drivers are rated around that voltage.
All piëzo tweeters, however, have horns which are needed to increase the sensitivity in the audible range. These tweeters are known to be cheap, have a high efficiency and absolutely terrible sound quality. Think screech and sibilance galore. This is why you don't see them used in serious hifi speakers.
I dug a little deeper and found these type of transducers are sometimes used for 'pest repellant devices' putting out huge SPL (120dB) way above the audible range. Because of the high frequency we humans fortunately cannot hear anything so we can be in the same room as the ultrasonic screaming device.
This may be an indication of the audibility of US (Ultra Sound). When we are to believe Oohashi the brain can only 'do' something with US when it is harmonically related to music.
Anyway here is a plot of such a US driver: http://www.ke-sen.com/userfiles/1383036350.jpg
That one has around 73dB efficiency at 1m around 20kHz but puts out 96dB around 40kHz.
I suppose you could safely put it in parallel with your ML and experiment away.
To test for US effect: Use songs with KNOWN real US and downsample those same songs to 44.1kHz (with a known good downsampler) and compare those songs, again....BLIND, which is easy to do with blind test programs. Listen to this with the Batpure always connected.
Even then the question remains, when the tests prove audibility, is the lack of US or the sharp filter the culprit. To find that out (when blind tests are positive) you will need the switch in the tweeter path and a blind test with known good recordings only. Needs someone else operating the switch without you knowing.
Do let us know the blindtest results.
My previous ramblings are only valid if sounds come from it in the audible range at a volume about as loud as your excellent speakers.
Have fun but above all enjoy the music.
This comment has been removed by the author.Delete
One other point regarding reproduction of supersonic frequencies by loudspeaker systems: It's no secret that a driver's dispersion narrows substantially with increasing frequency. One only needs to look at the lateral response family charts on JA's Stereophile speaker measurements to see what I mean. When you get up into these ranges, most tweeters, even those designed for ultrasonic reproduction, will produce only rather narrow beams of energy. It's simply a function of the frequencies radiated vs the radiating area of the diaphragm. The upshot is that the speakers have to be carefully aimed directly at the listener, and the "sweet spot" where the ears will be in the path of the ultrasonics coming from both speakers will be quite narrow.ReplyDelete
Great point John...Delete
Yup, narrow sweet spot. Could be one heck of a difficult job to optimize the toe-in when one can't hear the frequencies!
Thanks for the continuing input, Frans, etal. Very informative.Delete
Regarding the (lack of) ultrasonic (US) dispersion, I was thinking of using something like the JBL "butt cheeks" biradial horn to improve this situation but, here again, come up short in the area of speaker design. Each obstacle is creating a learning experience...not a bad thing.
As a sanity check, have also been considering a way to measure US reproduction after being inspired by reading and , maybe even going the DIY route for a US mic, e.g.,
This capability should help in identifying songs with US content since the CLXs are spec'd out to ~23kHz.
Regardless of where this goes, I may still wind up with a couple of unique pest repellents! :-D
Excellent post! I must thank you for this informative read. I hope you will post again soon. Warehousing of merchandise....ReplyDelete
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Hello Archimago from Montreal,ReplyDelete
first off great blog (too bad I discovered it very late.. circa 2016).
And this topic is very interesting. I just want to share a few data points nothing more.
ok first off today I'm 49 so pretty sure it doesn't matter for me anymore but I do recall many years ago (20 years?) doing this experiment. I placed high frequency horn driver on top of my then two-way box speakers. I used a simple 1st order crossover (1 capacitor in series to the signal) straight from the amp terminals. Same speaker wire going from same amp terminals to the existing speaker. In other words two speaker wires now. But in the new leg for horn I put a simple on/off switch that I could throw from my sweet spot seat and turn off and on the horn. The capacitor value for this high-pass filter was chosen to allow for a -3db at 22kHz. In those years my source was a bright CD player (weren't they all) modified for NOS but I did use a mix of well mastered/recorded and mass market CDs. Most can probably argue that my mind was telling me "what to hear" as I was the one hitting the on/off of the horn but I did hear something which did sound like more "air" around the instruments for minimalistic two microphone recordings. on multi-mic rock or pop it was less obvious probably because of the recording. my rational was that frequencies below 20khz and frequencies above 22khz were combining to make harmonics both above and below 20 kHz. Anyone familiar with FFT of pure tones would know what I mean (f1+f2) and (f1-f2) and so on. end of that story.
second item I want to share is there are a few microphones that have extended range and capture above 20khz and some record labels did and do use them. one example is the Sanken CO-100K. Just search for this and add SACD to the search.
Finally, this very interesting article from 2010, which at the bottom has a list of SACDs that have musical content above 20khz. That anyone here can look for and test also my little experiment is easy and not too expensive you don't seen any fancy or expensive high frequency driver. Anything from surplus that goes above 20khz hopefully at least 28 or 30khz.
thanks for your Blog, it is a mandatory stop on my web surfing.
An interesting experiment... not blind and listening out for ultrasonic sound using a CDP which, by definition, has no signal above 21kHz to begin with.Delete
In those days it would have been better to have used vinyl instead with a cart exceeding 30kHz, a pre-amp that did not roll-off >20kHz and records with (known) recorded ultrasound.
The modifications may have given way for aliasing resulting in unwanted signals to be added in the audible range.
This may not have resulted in the sound having more 'air' though.
Curiously enough you felt you experienced more 'air'. Could be expectation bias (knowing the tweeter was on) but possibly the horn tweeter, which may have had a 3 dB or more higher efficiency, may well have given more SPL in the upper audible range than the tweeter in your speaker did.
Most likely you just may have heard a rise in the upper part of the frequency band.
Hi all and many thanks for such a good reading!Delete
In the past years, I was reading on many forums about the possibility, for some people, to actually fell some harmonics above 20 KHz, though I'm not sure how this could change how the music is perceived. Perhaps it would be great to have studio's master recording able to catch absolutely every sound, no matter of its frequency. This will most likely make everyone happy, I'm pretty sure about it. :)
Now getting back to our home audio equipment, to have an entire system able to correctly reproduce all frequencies spectrum from 1 Hz to 100 KHz would probably be a little bit dangerous. I am imagining the "perfect subwoofer" powerfully playing few Hz, then my chest and walls vibrating a lot and me thinking it's an earthquake. Also, if a bad recording will introduce ultrasonic noise then nobody will be able to predict how the DAC/amplifier/speakers combo will reproduce this. I would expect either oscillations from the amplifier, either speaker's filters will get defective, either speaker's drivers will be unable to fully reproduce correctly entire music's dynamic, especially if the ultrasonic noise is having a high amplitude (I don't even want to think about the price for speakers able to reproduce correctly up to 50 KHz or even more).
Also, AFAIK, low-pass and high-pass filters are always introduced inside amplifier to overcome woofer's very long excursion when very low-frequencies can occur (there's no reason to let 10 Hz to get to the speakers as long as driver/cabinet's resonance frequency is much higher), but also to suppress very high frequencies at amplifier's output to prevent oscillations. I was reading on audio forums about people realising that their amplifiers are oscillating a lot, but they cared not, because they really liked the output sound very much. :)
I believe Frans can explain much better why is so important for audio amplifiers to have a properly designed low-pass filter (and perhaps a high-pass filter too, though this could be done with ease by choosing the proper caps value in the input stage).
The reason why many amps roll-off in the lowest frequencies is often because of so called 'coupling caps' are in the signal path.Delete
Another reason may be (for DC coupled amps that can even amplify DC voltages) could be the presence of a DC servo. This circuit could be needed to ensure speakers do not receive DC which is potentially harmfull to woofers. That circuit has a roll-off point which dictates the bottom part of the frequency response. More often than not well below 1Hz. The point of such a design would be NOT to have coupling caps in the signal path.
Older amplifiers often had (have) subsonic filters. This was essential for playing vinyl records where unwanted (and highly amplified) subsonic frequencies could make your speakers sway back and forth without you knowing. This could lead to overheated woofers or distortion.
That's why the subsonic filter is/was there.
So IF you play vinyl a subsonic filter may be a good thing to have.
Possibly built in in the RIAA pre-amp.
For CD this is not needed nor for other digital sources.
Another reason for having a roll-off in the bass is to prevent possible DC voltages from certain sources to be amplified when the amplifier circuit itself would have been able to amplify it. In those cases there is a coupling cap present to remove possible DC output voltages which can easily and silently burn out woofers when not detected.
When no coupling caps are desired DC servo is another solution.
High frequencies are often limited solely by the design of the amplifier.
Certainly when output tranformers (think tube amps) are used.
Amps do not start to oscillate when high frequencies are fed into it.
BUT they certainly can amplify them without you knowing/realising it.
There is no reason why designs wouldn't be able to reach 1MHz. This SHOULD not be a problem for the amp nor speakers.
Still some amplifiers have low pass filters on their input. This usually is there to prevent very high frequencies to 'saturate' the amplifier circuit when feedback is used in the amp. The dreaded TIM.
When squarewaves are applied this could cause problems in the amp.
Fortunately they do NOT exist in music... That is unless you use unfiltered NOS DAC's in which case it could.
When unfiltered NOS DACs are used large amounts of hypersonic energy may be present leading to IM distortion and/or blown tweeters.
In some hires recordings high amounts of 'tape bias' signal could be present or other HF sigals one cannot hear but potentially could damage tweeters or even 'boucherot filters' in amps could go up in smoke.
Why is having a wide bandwidth still important for amps and (less so) speakers/headphones but not for digital audiofiles ?
If one wants to reproduce 20Hz to 20kHz FLAT (means well within 0.5dB) the bandwidth, which is usually given at -3dB, needs to be a few Hz to about 100kHz to reach -0.5dB between 20Hz and 20kHz.
Also to ensure correct phase response for the lowest frequencies and instruments harmonics the phase should not shift too much at these extremes.
For this too roll-off points should be chosen well above and below the audible limit.
That needs to be this way in order to accurately reproduce a file which is limited from 10Hz to 21kHz by itself.
Good discussion guys.Delete
Mytubbie - welcome! Interesting experiment with NOS and the on/off switch to the high frequency horn. I agree that there is subjectively an "air" to the effect from ultrasonic noise. Another way to "hear" this is with PCM to DSD64 conversion such as in JRiver to a DSD DAC. I can appreciate if some like this effect even though the "sound" isn't part of the original intended signal!
Thanks for the wonderful analysis. I think an important issue that is missed in the argument is that ultrasonic harmonics interact in a larger continuum of harmonics and can create new effects within the audible range. I think that the conventional model of psychoacoustic testing with sine waves keeps this aspect from the forefront of the issue.ReplyDelete
With regard to increasing sample rates, the audible difference would become more prominent if the frequency bandwidth of each component in the sound recording/reinforcement process allowed ultrasonic harmonics to pass through without being truncated. What we have now is merely increasing samples per second of the lowest resolution (frequency bandwidth) component in the recording system.
For me, a great example of what is missing in the argument is demonstrated by listening to a gong player live, then comparing it to a gong recording... striking difference!
Is the striking difference between a live gong player and reproduced recording of a gong player really due to a lack of harmonics or perhaps caused by other aspects ?Delete
Is that difference not there when recorded in DSD256 or DXD or >192/24 and only present when reproduced with say CD quality ?
Or the fact that a recording is just a simple point in space (microphone) in a certain position that catches soundwaves regardless of how they propagate and are later spewed out in a single direction with possible phase shifts as well as amounts of linear and non-linear distortions ?
The difference may very well be the way sounds propagate through a room being so much different than through speakers which causes the striking differences.
Could also be that the actual playback level is not the same as the SPL during real life performances.
What was the distance to the instrument in both cases ?
Did you hear the actual instrument from a similar distance in the recording studio and played back in the control room at the same distance ?
How can one be sure the perceived striking differences can be attributed to (inaudible ?) harmonics alone and these harmonics are the root cause of the perceived differences between recording and live ?
Thanks for the comment Dolores,Delete
Good example to consider of the gong player. I agree with Frans that there are many variables to think about which could be significant contributors to the sense of the sound being "live".
SPL, microphone placement and limitations, omnidirectional vs. forward facing box speakers, room acoustics, etc...I bet these factors play a huge, audibly different effect. Another thing to consider is getting closer to that "live" sound with multichannel vs. stereo to perhaps better simulate the original performance.
In any event, I doubt just capturing the greater harmonic bandwidth makes a huge difference. That is, I'm not sure if I captured the gong with DSD512 then sampled it down to 24/48 would result in that much difference in the "liveness" when reproduced with the same DAC, amps, speakers.
Thanks for this fascinating discussion! So many factors in play. I'm coming from a perceptual (composer/musician) and medical background. Do you mean by 'liveness', the simulation of immersive spatial qualities of sound?ReplyDelete
What I am thinking about with regard to the passing through of ultrasonic harmonic interactions, is related to spatial qualities and harmonic/phase relationships which can be lost or truncated anywhere in the (<20KHz) recording/reproduction process (chain of hardware, software encoders, compression formats, amps, speakers, etc). Optimally, these interactions can create a psychoacoustic gestalt in which the perceptual experience is greater than the sum of isolated parts (harmonics, phase relationships).
I also wonder about the limitations in accuracy of current digital recording formats, to encode and render the inharmonics (non integer multiple harmonics), and aperiodic waveforms of the live gong sound.
The digital format preserves phase relations pretty well. This is demonstrated by recording a squarewave. When higher frequencies are cut-off the reproduction of that squarewave will look like the plots you see on this (and other websites). The Gibbs phenomenon is what you should investigate to see what phase correct bandwidth limited signals should (and do) look like.Delete
When the phase would be altered the squarewave would look much different.
The biggest elephants in the room is not the digital format nor actual ADC and DAC at all. I am quite convinced that, when well designed and suited equipment is used, this isn't a problem. Also the recording formats are not the biggest issues here.
The biggest elephant in the room is the speaker. It's frequency response and its phase response errors are magnitudes bigger than that of any proper designed ADC and DAC.
At 30kHz the wavelength is just 1cm.
This means a 5mm difference in the path between your ears and the midrange driver and tweeter of a speaker will rotate the phase by 180 degrees already. Gone is your well preserved phase response as well as the frequency response.
Other big elephants are the listener and its position as well as the room sounds are reproduced in.
Invisible elephants also exist. In the studio or room where something is recorded in. Sound engineers may have altered many aspects of the microphone signal so it sounds 'good' to the general public or the producers or those with financial interests in the end product.
Not to mention the quality of the used microphones and equipment and knowledge of how to place microphone(s). Not all studios are top notch quality all the way. There are quite a few excellent studios and sound engineers around.
The chance they have produced a popular music album is slim though.
Those are far greater issues than any digital linear-phase wideband recording system.
That system and a suitable microphone doesn't care about the relation between harmonics (integer or non-integer) and periodicity of the waveforms arriving at the microphone. It just converts soundpressure differences into voltages and so does a proper recording chain. It doesn't need to be harmonic at all to be recorded properly.
Should I ever have to research this I would start in the biggest zoo's where the biggest elephants are.
The last thing I would research is in the digital recording playback chain... why ?
Because this is already very well documented.
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I did a simple 44.1 vs 96 blindtest earlier and i preferred the 44.1 KS/s version 30/43 times, in all other blind test the result was the opposite but then i never got good statistics.ReplyDelete
My system is really cheap(dynavoice dm5+t-amp) so i cannot rule out IMD products. I only noticed a difference in very specific types of music, very strange.
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