DSPeaker Anti-Mode X4

tmallin

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Good luck! I know another person who owns the X4 and uses it with subwoofers and has had good success. I have also heard that the DAC part of the X4 is none too shabby.I still have no personal experience with either of those functions, however.

As I hope I made clear above, I do believe that the equalizer function of the X4 is head and shoulders above the previous model in terms of sonic transparency.

Please read my review of the Harbeth M40.2 on this Forum to see my comments about the relative merits of the various versions of the 40. In my opinion, the 40.2 is the best version. For one thing, it needs less midbass correction for smooth response in that area.
 

tmallin

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More on the Harbeth M40 vs. M40.2: the M40 is the least "forgiving" of much-less-than-perfect program material in the M40 series. Now, REG would probably say that is because they are the most literally accurate of the bunch. Certainly the M40 lacks the forwardness around 1 kHz that REG complained of in the M40.1. I believe the M40.2 is less forward than the M40.1, but moreso than the M40.

But, to me, the M40.1 and especially the M40, sounded too recessed in the presence range. This backed things off a bit too much, requiring close up listening to prevent the sound from getting a bit lost on the way to the listener, at least in a "narrow" room with the side walls closer than three feet to the speakers. The off-axis filled in just fine, but this made the wall reflections relatively different sounding from the on-axis output, requiring side wall damping rather than just diffusion. Now, I prefer near-field listening anyway, but this was definitely reinforced by the original M40s.

The M40.2 still has a bit of presence-range relaxation, but is quite a bit more present sounding than the original M40. It needs wall absorption much less since the off-axis sound is less different from the on-axis output.

The M40.2 is also a bit softer in the top two octaves. The M40 had a basically full return to the midrange level after the presence range dip. This caused the top two octaves to stick out just a tad, at least in comparison to the M40.1 and especially the M40.2. This is quite noticeable on so-called Golden Age classical recordings made with peaky old tube microphones. With the M40s, I had to EQ the highs to make the Mercury Living Presence recordings tolerable. With the M40.1 and especially the 40.2, while you know those recordings are too bright, the relative lack of presence range recession, together with the bit softer highs allows such recordings to be enjoyed without "needing" equalization. As I mentioned in my M40.2 review, the M40.2 is quite pleasing on a broad range of commercial recordings even without any equalization at all.

So, overall, compared to the M40, the M40.2 has deeper bottom octave extension, less midbass excess, a bit more forwardness around 1 kHz, less presence recession in the 2 - 4 kHz region, and a bit softer top two octaves (5 - 20 kHz). The newer speaker has seemingly lower distortion, more clarity, yet greater ease and authority, and will play significantly louder without distress. For more details, see my Harbeth M40.2 review on this forum.
 

samgar

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Hello Tom:

Thanks for the great write up on the X4. Based on your good recommendations I purchased one to help tame my Vienna Acoustics- The Music speakers. It works miracles. Tell Walter he owes you a commission!

Sam
 

Bobolaclune

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Part 4

Idiosyncrasies

That brings me full circle, back to the DSPeaker X4. Before describing how the unit essentially works so very well, I will deal with the, to my mind, relatively unimportant ways in which the unit could still be improved as to its basic equalization function. I acknowledge that these are minor issues in that they don't significantly impede my goals of ease of set up, sonic transparency, sonic excellence in automatically adjusting bass response, and ease of manual adjustment of response.

First, in the before-and-after frequency response graphs, the "before" response is shown as a colored line and the "after" response is shown as a black line. In the key to which is which that appears on each graph, however, the "after" line is shown as appearing in a pale pastel color, clearly incorrect. This surely could be dealt with via a firmware change. Show a colored line within the light square and a black line within the other light square, instead of a light-colored square.

Second, nowhere in the manual does it suggest that the automatic correction can operate full range. However, full range is clearly a choice. And, in fact, as suggested by Walter at Underwood Hi Fi, my dealer, most users have found the full-range correction choice to sound best. In my experience, it truly does sound best, and not by a tiny margin, either. The user's manual needs to mention and indeed highlight this.

Third, the first time measurements are made, the unit automatically populates five or six presets with its suggested response changes. These suggested presets vary in terms of the range affected and the degree of low frequency compensation added back in to subjectively compensate for the lack of overall bass weight which the removal of bass peaks can cause. But once you change those settings for each preset, there is no way to recall or reset the originally generated automatic presets. To get them back, you would have to first perform a factory reset and then run the measurement sequence again. It's best to jot down the parameters of each preset so that you can recall them manually if you wish. This is especially so since the unit's auto presets all sound very natural indeed, just with bass weights which vary a bit from each other. I would hope that this could be corrected via a firmware update.

Fourth, the "after" graphs of response do not change if you manually adjust the preset Profiles. What is shown on the "after" graph seems to remain the auto-generated "after" response as measured during set up. It would be helpful for the unit to compute and show the "after" response with manual corrections input, even if those actually are not measured responses, just calculated "after" responses. I would hope that this issue could also be dealt with via a firmware update. (In defense of the current arrangement, however, note that you can generate the actual response of any manual setting just by using the Measurement Only function of the Anti-Mode software which allows you to take measurements without altering the computed corrections.)

Fifth, the back of the remote control needs to be redesigned. I very much appreciate the high-quality heft and sturdiness of the all-metal remote, its great button feel, minimal buttons, and sure-fire operation. However, the backside of the remote should be smooth. The five round-head screws that stick up from the case are unsightly. In addition, the fact that the screws stick up can easily cause you to mistake the backside screws for the raised frontside control buttons in a darkened room, especially the three screws which are arranged in a triangle on the back side behind the frontside control buttons. Since the remote does not light up (it would be nice if it did!), you need to rely on tactile cues in a darkened room. Redesign the back so the screwheads are flat and rebated so that they are flush with the backside.

Sixth, upgrading the firmware is a bit more difficult than it should be. The only method of doing this is by downloading the firmware update file into the root directory (that is, not within a file folder) of a USB stick. The USB stick must first be formatted to the FAT or FAT32 formats, which is a bit unusual these days. Also, you'd best use a USB stick of 32 GB or smaller. Despite my using programs which were able to format a larger 64GB stick into FAT32 format, I could not get that USB stick to work with the X4. Once I loaded the firmware update file onto a FAT32 formatted 16 GB stick, the update proceeded smoothly, taking only a couple of minutes or less. Something else to note: new firmware will not load unless the file name of the new firmware is FIRMWARE.X4. If it has a date or any other characters in the file name, it probably will not load. For example, new firmware sent to me with the name FIRMWARE_22_Nov_2018.X4 would not load until I renamed it FIRMWARE.X4.

Seventh, this is the one matter that may cause some a bit of pause and this may not be correctable without a significant redesign. But I've decided to put it here rather than make a larger fuss about it since, to me, in my system, it does not significantly get in the way of using the unit to correct the frequency response as I want it to be. Probably to avoid the problems of overload encountered in the earlier DSPeaker Anti-Mode 2.0, the design seems to adjust the overall midrange and volume level of mids and highs downward as any bass compensation is applied. This makes it more difficult to directly compare various levels of bass compensation since the overall volume changes as you make changes in the bass compensation level. But as long as you boost the volume level up by the amount of bass you are adding in dB, comparisons are possible and once you have the subjective volume correct again, it is a matter of only seconds before you can determine whether the new bass level is more or less satisfying to you. Note that, for reasons I don't understand, this idiosyncrasy does not affect the bass boost or tilt (Quick Tone) controls; applying those does not affect the level of mids and highs.

That's it. Nothing earth shattering. Despite these niggles, the X4 works EXTREMELY well for its intended purposes as an equalizer.

[Continued in Part 5]

Thank you for an incredibly comprehensive review of the product which I also own but confess to only now beginning to understand it's multitudinous options and it's versatility. I have one question - you reference the fact the room correction should be done Full Range and that's how Wally at Underwood recommends it be operated. Would you mind indicating where in the menu commands you can choose that option? Thanks and kudos again. Bobolaclune
 

tmallin

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With the remote control, press the center button to enter the menu structure.

From there use the up/down buttons on the right side of the remote to scroll down one selection to Anti-Mode and select it with the center button.

Scroll down to "Corr. Range" and select that item with the center button.

You can then use the up/down buttons on the right side of the remote to select the Correction Range. The "full range" correction I refer to is named 16 kHz in the menu. Select 16 kHz with the center button of the remote. You will see an hourglass for a couple of seconds. When the hourglass disappears, again select the center button.

Then just use the center button and right-side up/down buttons to exit the menu structure.
 

Bobolaclune

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With the remote control, press the center button to enter the menu structure.

From there use the up/down buttons on the right side of the remote to scroll down one selection to Anti-Mode and select it with the center button.

Scroll down to "Corr. Range" and select that item with the center button.

You can then use the up/down buttons on the right side of the remote to select the Correction Range. The "full range" correction I refer to is named 16 kHz in the menu. Select 16 kHz with the center button of the remote. You will see an hourglass for a couple of seconds. When the hourglass disappears, again select the center button.

Then just use the center button and right-side up/down buttons to exit the menu structure.

Thanks Tom!! Appreciate the help.
 

flat4

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Apr 23, 2019
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New hear Tom. Great write up! i have had mine for a year and i am still learning all the time. Looking to try a 4 sub setup in my listening room now that they have an update to 0.4. Wonder if i could use optical out into my Dac 3 and use the analog outs of dac 3 to my mains with a hp filter and run X4 for subs only?

My thoughts are i at least wouldn't be doing double conversion from 80-100hz and up. Of course i would only be using correction for bass 80-100hz down.
 

tmallin

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I can now confirm that the headphone functions of the DSPeaker Anti-Mode X4 work very well indeed, with or without the special headspace processing engaged.

The front panel jack used to plug in the measuring microphone does double duty as a 1/4" or 6 mm headphone jack. To get this jack to output sound to the connected headphones you have to make sure through the menu system that the headphone function is in fact enabled. When you first enable this function, you may see a brief hourglass icon on the screen. When that disappears, the speaker equalization will be disabled, as will the output of the X4 to any output other than the headphone jack. The volume level is also adjusted downward automatically by the firmware so that you aren't greeted by surprise with overly loud output from the headphone jack. The Auto Toggle feature of the Headphone menu allows you to leave the headphones plugged in all the time and toggle back and forth between output to the headphones and output to your speakers.

The sound quality I hear from the X4 via headphones with no headspace processing engaged is very fine, right up there with what I'm used to from, say, the Benchmark DAC3 HGC headphone output. The Benchmark HPA4 headphone output is still slightly superior in some ways, but unless you've heard your headphones via that unit or something like the SimAudio 430HA unit, you will not know anything could be better sounding. There is plenty of drive even for the power hungry Audeze LCD-4 phones. The frequency response seems very flat and even, noise is extremely low, bass has a nice combination of extension, definition, punch, and fullness, and the mids and highs are clear, obviously very low in distortion, and the highs are nicely extended and filigreed. The background is black and the presentation has a nice sense of ease about it. The Benchmark HPA4 sounds yet a bit better in these ways, plus it has a more three dimensional and generally larger presentation. But, as I said, except by direct comparison, you'd never know that the X4 is at all lacking even when using such elite headphones.

I'm using the latest firmware for the X4, which as of this writing is the March 8, 2019, version. Improvements were made to the HeadSpace HRTF processing function back in the September 19, 2018 version of the firmware. In my opinion, the HeadSpace processing now adds a viable listening alternative to "straight" headphone playback. I would strongly recommend, however, that for the most natural presentation, the Menu's Rendering function be kept at a low setting, somewhere between 0 and 2 (the Rendering scale goes up to 10). I find that the low settings of the Rendering function provide believable improvements in the three-dimensionality and "out there"-ness of the presentation without undue addition of "phasey" "tugging at your ears" processing. The HeadSpace processing tends to move the overall presentation out from inside one's head to more in front of your eyes.

Note that I did not say that the HeadSpace HRTF processing is devoid of any "phasey" sound, just that there was no "undue addition" of phasiness. For example, true binaural recordings and a few of the coincident and quasi-coincident recording microphone techniques offer spatially enhanced playback via headphones without any "phasey" sound at all. Thus, recordings made with microphones in the ear cavities of a dummy head/torso, or even recording made with modest separation of just a pair of microphones (Blumlein, ORTF, M-S, or X-Y) can often produce more naturally spacious presentations via headphones than other types of stereo recordings. If you are sensitive to this kind of ear-tugging phasiness, you might still object to the sound of the HeadSpace HRTF processing even with Rendering at 0. In that case, just skip the processing. You still have a very fine headphone amplifier thrown in "for free" with the X4.

The Anti-Mode X4 also offers you the possibility of creating two separate equalized profiles for your headphone listening. You can do this most effectively by using the built-in parametric filters to alter particular frequency response bands as needed and then saving those adjustments as one of your headphone listening profiles. I suggest using well-recorded pink noise played back through the headphones as at least part of the process by which you choose your parametric EQ settings. Listen to the pink noise and adjust the parametrics for the smoothest sound where no frequency band "sticks out" and where you hear no distinct "tones" in the noise playback. If the meaning of "sticks out" or "tones" is unclear to you, you can get a feel for what I mean by this by just adding a 15 dB EQ peak centered somewhere between 200 Hz and 2 kHz.
 

x999666x

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@tmallin really interesting posts!! I am having same issues with the antimode dual core, it's unusable, it injects lot of noise/distortion/glare + narrower soundstage....and I've tried two different external linear powers supplies (teddy pardo and mcru) but actually they bring out the problems even more!I am talking about the unit in bypass-no calibration and none of the functions activated. It's a paradox but it sounds better analog in/out....maybe the digital in/out circuit it's not well implemented!
I have a question for you as I see you have the auralic aries : does the parametric eq activation has a bad effect on the sound quality? I am thinking of getting the aries too to go that route to tame the main 45hz room mode I have (and as a consequence some others that come from that one) using the parametric....I know it's not the right approach as It will only work in freq/db and not in time domain....but it might be just fine for me as it's the less harm! I ve demoed the x4 some months ago but didn't have the chance to try it digital in/out at the time and with the analog in/out AFTER my dac, the x4 brings its sound signature which I didn't like! (don't know how much influence it has like you ve tried in all digital)
So again....do you think I'll get a bad result using the eq from the aries?(or maybe roon parametric to the aries)? not asking about the eq result but just activating it without even mess with parameters yet....I was reading elsewhere it was worsening sound quality....but I am wondering how much comparing for example with all the equipment you' ve tried?? (I guess roon or aries eq upsamples then downsamples again)
Assuming I choose just single band eq route....what do you think about getting an aries g1or maybe an external analog parametric eq form pro audio stuff??(I guess I need parametric and not just graphic eq as I need to cut a specific freq)
Really don't know what to do anymore!!
Thank you and sorry for my english!
 

tmallin

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Digital EQ is the way to go. Analog parametric was fine in the day, but inserting a digital equalizer, either hardware or software based, is potentially cleaner sounding.

I am aware that some/many users of the Dual Core find it to sound better when used analog in/analog out. I did not find that to be the case, but I can understand that judgment if you did not find a proper after-market power supply. The extra cost power supplies provided by my dealer took enough of the "digitalis" away to make the digital in/digital out path quite superior to the veiled roundness of the analog path. I see that my X4 dealer, Underwood Hi Fi, currently offers an optional Channel Island power supply for the Dual Core for only $329. You might want to try that out if you don't want to move to the X4.

Since I find the DSPeaker Anti-Mode X4 in digital in/digital out mode to be transparent and very easy to use, I honestly have not tried the Auralic Aries G2 equalization functions which were added with a firmware update a few months back. Since I also still use an Oppo UDP-205 as a source, if I were to use the Auralic Aries G2 it would not equalize the sound from the Oppo. As I have it set up, of course, the Auralic and Oppo each feed separate digital inputs of the X4.

That set up may change down the road. I will try loading all my disc files onto a new terabyte USB stick and connect that into the USB storage input of the Auralic. (Note that Auralic recommends using an AC-powered USB hub to do this, not direct insertion of the USB stick into the Auralic's port--Auralic claims the sound is better if that connection does not draw power from the Auralic.) Since I find that my uncompressed WAV files of all my discs sound at least as good with the Oppo playing those files from its USB ports as the Oppo playing the original discs from its drawer, perhaps I really don't need the Oppo anymore. Internet streaming and local access of disc files is all I need and both those can be handled by the Auralic. I will have to test the sound quality of the files on the stick played by the Auralic to see if that path sounds as good as having the files played through the Oppo.
 

tmallin

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I'm not sure where to post this, but I'll start here. Until the last week, I had not used the USB output of my Auralic G2 streamer for many months. I had never before used the USB input of the DSPeaker Anti-Mode X4. Now I'm using a USB connection between the Auralic and the X4.

This past weekend I got around to experimenting with using the USB output of my Auralic Aries G2 streamer rather than its coaxial SPDIF digital output. The differences were of the same nature I'd heard before, but minus the large change in tonality and irritating brightness from USB connections. I'm speculating that the combination of my recently added UberBUSSes and Triode Wire Labs power cords (talked about in My Clean Power Adventures thread) must have eliminated RFI and other noise interference which were causing the tonal brightness with USB connections before. As before the USB connection sounds more three dimensional, has greater imaging stability, blacker background, and is seemingly cleaner/less distorted than the SPDIF connection. The tonality is still not exactly the same as the SPDIF connection, but certainly is no longer too bright to sound natural.

One reason I hesitated doing this experiment was that I'm not able to use USB all the way from my source to my Benchmark DAC3 HGC in this system. The path is interrupted by my digital equalizer, the DSPeaker Anti-Mode X-4. The X4 has a USB input, so I can go from the G2 to the X4 via USB, but then must use the coaxial SPDIF digital output of the X4 to feed my Benchmark DAC since the X4 has no USB output. What got me to experiment were the excellent results I'd recently achieved with a USB connection straight from my desktop computer to a new Benchmark DAC3 B(asic) in a separate headphones-only system.

I have now ordered a P.I. Audio Group/Triode Wire Labs Discrete USB cable in the recommended 1.5 meter length. Right now, in both these systems I'm using the same USB cable I found to be best of the bunch I'd tried before: https://www.amazon.com/gp/product/B00R2R77JG/ref=ppx_yo_dt_b_search_asin_title?ie=UTF8&psc=1 reviewed in TAS at https://www.theabsolutesound.com/articles/oyaide-neo-d-class-a-usb-20-cable/
 

parkerjake

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Hello Tom, I just came across your excellent posts. As it happens, I also use Aries G2 and Dspeaker X4 and Harbeth speakers, in my case 40.1's. I have combined the Harbeth's with 4 Swarm subwoofers. My DAC is the PS Audio Directstream. So, I go Aries into Directstream into Dspeaker (via analog out of DS), into main and sub amps, which means I am using the X4 as both crossover and equalizer and volume controller. It occurs to me, based upon your review, that I can modify the connection scheme by going digital from Aries into X4, running X4 as though calibrating a 2.0 system, then digital into DS and analog to main and sub amps, with low pass filter applied to subs.

Basically, what do you think of the a/d and d/a conversion of the X4 and the use of it's crossover network applied to the big Harbeths vs running the Harbeths full range with digital equalization and a separate low pass on the subs? Analog conversion would then be handled only by the DS, which also would control system volume.
 

tmallin

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I have not tried the D/A functions of the X4. I have always used it digital in/digital out. I do not want to disturb the gain structuring and overall superb sonics of my Benchmark DAC3 DAC feeding the Benchmark HPA4 for volume control, feeding the mono Benchmark AHB2 amps. There is real synergy there.

My guess would be that removing the extra A/D and D/A the X4 is currently performing in your setup would allow cleaner sonics, but that is only a guess. I recommend that you experiment and see which set up path you sonically prefer.

I also suggest that you experiment with removing the PS Audio Directstream DAC from your signal path altogether. Reviewers who have used the X4's DAC say that it is very high quality. The X4 can perform all the functions you need done (EQ, digital and analog preamp and source selection, digital subwoofer crossover, DAC, volume control, etc.) and I would guess that its digital subwoofer crossovers probably are sonically superior to using the analog low pass crossovers of the Dayton Audio SA1000 subwoofer amps which come with the Swarm. I used the Swarm once upon a time with the Stirling LS3/6 (detailed in another thread) and found the digital crossover functions of the Lyngdorf TDAI-2170 I was using at the time to sound significantly better than using the low-pass crossovers of the Dayton amps (I had two) which came with the Swarm.

I assume that you have determined that adding the Swarm to the Harbeth M40.1s adds quality and/or extension to the bass end. If you have not experimented with this, however, I think you should. As my measurements above show, my M40.2s don't need subwoofers for full bottom-octave extension even in my small room. You might also try experimenting with the the X4 by letting it EQ just the Harbeths and see how that low end sounds in your room compared to the sound of the Harbeths plus Swarm. In my room, subjectively the equalized M40.2s sound better in the low end on most material than the equalized Stirling LS3/6 plus Swarm did. The only exceptions are a few pipe organ recordings with extremely deep bass and even on such recordings, it's a close call in my small room.
 

parkerjake

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Putting aside the engineering choices made between a commercial DAC and an FPGA, as well as PCM vs DSD, the Directstream sounds better than the X4 as a DAC, and has the advantage of being continuously upgradeable. I could follow your approach and place the X4 between the Aries and the Directstream, using only the digital equalization function of the X4. I thought, by removing the low bass from the Harbeth, I would improve the dynamic capabilities of that speaker, while shifting the bass reproduction to the subs. I found, by allowing the X4 to run it's own crossover choice, that it settled on 233hz which I didn't fully understand, but which sounded awful. So, I manually set the crossover at several points and found that 75hz sounded best. What I have not tried is your suggestion of letting the X4 EQ just the Harbeths and remove the subs from the equation entirely. I was focused on whether it is preferable to run a large speaker full range with eq and add subs or use an external crossover to effectively remove low bass from the large speaker and add subs.

What I will say is that the Harbeths are special speakers in near field listening. It's weird that such large containers sound more cohesive, up to a point, the closer they are together and the closer the listener is to them. The flip side is the difficulty of arranging a living room to accommodate their special characteristics..

Anyway, you've given me another avenue to explore. I don't really care that much about very deep pipe organ bass, but I am more concerned about dynamics and how the big Harbeths fair with full EQ vs crossover and elimination of deep bass plus EQ.
 

tmallin

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I have also heard from others that the X4's automatic choice of subwoofer crossover point is often too high. I understand that this function has been improved in more recent versions of the firmware. Make sure you have the most recent firmware.

When I had my LS3/6 + Swarm, I never liked the sound of rolling off the bass of the main speakers. I always ran them full range. Eventually I settled on rolling in the subs at 45 Hz, which was the minus 3 dB frequency of the unaided LS3/6 in my room. The LS3/6, like the Harbeths, have great sounding bass on their own. You may only want to extend the bottom reach of the Harbeth bass, not replace it.
 

tmallin

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File this under "Truly Great Customer Service from DSPeaker."

A friend of mine in the USA also has a DSPeaker Anti-Mode X4. Recently, the audio output on his X4 continues to stop intermittently even after rebooting it several times. It occurs on both USB and Coax inputs using different sources which rules out the sources as culprits. Also, when the audio stops the powered subwoofer doesn't sound which rules out the main speakers' amp. My friend emailed Pasi Ojala (info@dspeaker.com) and DSPeaker is replacing the unit at no expense to my friend. Pasi said there is nothing the user can do to fix that issue. DSPeaker will issue a UPS pick-up for the defective one.

I also had occasion to contact Pasi about a puzzling performance aspect of the X4. He very promptly replied with a complete explanation and confirmed that my unit was operating as currently intended. I told him:
I still find the X4 to be the best-sounding and easiest to use equalizer I have ever used in my system. Thanks for producing such a great unit!​
[Pasi responded] Hello Tom,

Thanks!
[I continued] However, I have noted one somewhat puzzling result and thought you might be able to confirm whether my unit is operating correctly.

I feed coaxial SPDIF digital signals from my Auralic Aries G2 into the X4. I feed the coaxial digital output of the X4 into the coaxial digital input of my Benchmark DAC3 HGC DAC. On programs where the X4's screen recognizes the input as 24/176 or 24/192, the Benchmark will show the signal it is receiving from the X4 as 24/88 and 24/96 respectively. This is so even when the X4 is set to bypass the Anti-Mode DSP correction and set for MAX volume out.

If for such programs I physically bypass the X4 and feed the digital signals directly by a cable directely from the Auralic to the Benchmark, the Benchmark indicates it is receiving 24/176 or 24/192, as expected.

In other words, it appears that the X4's internal processing is limited to 24/96 and that it downsamples the signal it outputs if the input signal has a higher sampling rate than 24/96. Is this correct, or is something amiss with my unit or the set up of the unit?​
[Pasi responded] That's how it works right now.

The distortion performance of the DAC is optimal with 96kHz, and the digital output provides the corresponding signal.

There's one interoperability consideration. Toslink works with 96k - usually, but 192k can have issues depending on the devices and cable.

176/192kHz from coaxial out is on the list, but at a low priority. We're now working on some new features.

Thanks for asking.


 

parkerjake

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Hi Tom,
I'm curious about Pasi's response to your inquiry on 24/96 output of the DSPeaker. If that's the limitation of the unit, what is the advantage of the better signal capabilities of the Auralic G2 and the Benchmark DAC3? Seems like something is being lost in the middle of the signal processing. Is that loss audible when you bypass the DSPeaker entirely? If there is a loss, is it offset by the sonic benefits of equalization?
Thanks
 

tmallin

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Chicagoland
Theoretically, something might be lost, but with my Harbeth M40.2 speakers in this system and room, much more is lost by bypassing the X4's equalization than would ever be gained by bypassing the X4's 24/96 processing limitation. The sonic improvement from applying the X4's equalization vs. bypassing that equalization is easily audible with all program material in my system.

It's impossible to make a quick direct bypass comparison because even when the X4 is used digital in/digital out with the EQ bypassed, the 24/96 processing limitation is in place. You never get 24/192 or higher rates physically through the X4, in other words.

The only bypass is by physical rewiring to physically take the X4 out of the signal path, connecting, for example the USB cable straight from the Auralic to the Benchmark DAC3's USB input. That takes awhile and good aural memory is very short. By the time I get it done, I can't reliably hear any difference with 24/192 sources between that physical bypass and going through the X4 with the X4 in its 24/96 bypass mode. Thus, I conclude that, at least with the Harbeth M40.2 speakers in this system and room, it's better to have the X4 applying EQ at 24/96 than to eliminate the X4 to allow the Benchmark to reproduce 24/192 without the X4's EQ.

As I said early on in this thread, the DSPeaker Anti-Mode X4 is the most transparent equalizer I've ever used. Whatever is lost because of its current 24/96 processing limitation is very subtle indeed compared to even the relatively small frequency response smoothing the X4 does in my system. Frequency response changes, even small relatively narrow-band ones, are quite audible and important to realism by comparison to whatever sonic losses might occur when resampling 24/192 to 24/96.
 

parkerjake

Well-Known Member
Jul 14, 2019
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That makes a lot of sense. By the way, as I read through your set up, you are listening in a near field, so, presumably, the room effect is smaller than it might otherwise be, and the DSPeaker is primarily working on the Harbeth 40.2 frequency response. Have you had experience or contacts with others who have experience with the DSPeaker in a mid or far field listening application where the need for room correction would be much greater? And, in that environment, is correction above 500 hz still effective?
 

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