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Pretty sure there is no separate D90 MQA thread. You'll find comments on both in this thread, as humongous as it is. You may also find that there is a lot of negativity against the whole MQA-processing issue, but that's up for you to evaluate. I am a happy owner of the Topping D90 MQA with no regrets (except for one, next paragraph). I'm not really an MQA proponent (not really at all), but I wanted a unit capable of decoding MQA, and I think Topping has now sorted out most of the D90 MQA's issues (which were mainly due to the XMOS XU216 chip).
My only issue is that I got the D90 MQA in silver for my upstair's bedroom setup, and now I'm thinking it's better suited for my living room setup, which would've been better in black. So, for my purposes now, the D90 MQA in black for downstairs, and maybe the new Gustard X16 in silver for upstairs (a unit you might want to consider looking at as a Topping D90 MQA alternate; $300 to $350 cheaper!). Oh well, Se la Vie!
i thought the issues (loud noises during switchovers between non-MQA recordings and MQA recordings, etc) were all on the D90 non-MQA version! am i mistaken? as a result, those issues are all related to the XMOS 208 chip which is in the D90 non-MQA version (the XMOS 216 is in the D90 MQA version).
so i got confused when i read "...I think Topping has now sorted out most of the D90 MQA's issues (which were mainly due to the XMOS XU216 chip..."
the problems we are both thinking about should have been related to the D90 (non-MQA version) with the XMOS 208 chip, right?
I believe the pops, clicks, and dropouts were mostly associated with the D90 MQA (XMOS XU216) when first released; mainly in use with an Apple/MacIntosh (IOS)-based computer in playing certain sampling rates. Soon, after its discovery, Topping released new firmware which supposedly corrected it, for the most part. There have been other issues associated with pops and clicks when switching from DSD files to certain PCM rate-files; I think they have been mitigated, but not totally corrected. In my memory, they were either present or worse with the MQA than with the standard version, but I have not really encountered them on a noticeable basis with my use of the D90 MQA.
However, most forum members will probably tell you that the standard version with the non-MQA D90 is the more reliable and stable of the two. Those forum members who are keeping up with the current issues concerning these Topping models, please provide further instruction or correct me if I'm wrong. The MQA version of the D90 begins to be posted and discussed about after the 80th page of this thread.
My only issue is that I got the D90 MQA in silver for my upstair's bedroom setup, and now I'm thinking it's better suited for my living room setup, which would've been better in black. So, for my purposes now, the D90 MQA in black for downstairs, and maybe the new Gustard X16 in silver for upstairs (a unit you might want to consider looking at as a Topping D90 MQA alternate; $300 to $350 cheaper!). Oh well, Se la Vie!
Ah. I have exactly the same with my d90, but, as you said, only on PCM 768 (and then only sporadically). It does not bother me because I never use that kind of upsampling either. I thought I was the only one!
I believe the pops, clicks, and dropouts were mostly associated with the D90 MQA (XMOS XU216) when first released; mainly in use with an Apple/MacIntosh (IOS)-based computer in playing certain sampling rates. Soon, after its discovery, Topping released new firmware which supposedly corrected it, for the most part. There have been other issues associated with pops and clicks when switching from DSD files to certain PCM rate-files; I think they have been mitigated, but not totally corrected. In my memory, they were either present or worse with the MQA than with the standard version, but I have not really encountered them on a noticeable basis with my use of the D90 MQA.
However, most forum members will probably tell you that the standard version with the non-MQA D90 is the more reliable and stable of the two. Those forum members who are keeping up with the current issues concerning these Topping models, please provide further instruction or correct me if I'm wrong. The MQA version of the D90 begins to be posted and discussed about after the 80th page of this thread.
The "explosive" noise issue is with PCM768 and AFAIK has been reported on the E30 and D90 non-MQA, and it is something that sounds like an increase in volume and distortion, and it can go on forever once it happens. In a perfect world with rational people only, it would be considered a non problem since 768khz files are just a waster of space and upsampling to 768khz makes no sense BUT it is still a glitch and frankly it should not be there.
The clicks and pops, sometimes quite loud, happens with the D90 MQA when changing frequencies, i.e. as a very short event between tracks, and this is in my opinion a more real problem, but apparently firmware updates have mitigated it somewhat.
may i please ask for a brief explanation of what 768kHz files are and mean?
i have a general knowledge of audio, after having been an audiophile all my life, but am weak on the nuances of some of these digital protocols as i only have a basic knowledge of what i must know, not an encyclopedic knowledge. any advice, comments, to help me would be most appreciated. even a little knowledge to direct me in the right direction would help me along.
thank you in advance.
i'm currently using my Empyreans with my Astell and Kern Kann Alpha, which i'm enjoying a lot. still, i thought that for the convenience of being able to use my Windows 10 tower PC's large monitor to navigate with the Tidal windows app, it would be pleasant to use to make playlists, choose new albums, organize a bit, as i spend 3 hours to 8 hours each day at my computer desk, so plugging my Empyreans into a desktop set up like the D90/A90 combination would seem like a good move, in addition to my A&K Kann Alpha for when i wish to go lay down on the sofa for a while to relax.
thank you for the good folks here to help educate people like me who only know a bit...and would like to learn more and be well versed in this area.
Ok, I will try to simplify a bit, but it is not my best skill.
I was referring to the sampling frequency. In digital audio, audio signals are represented by samples, which are numerical values that represent the "height" of a point on the waveform. A CD uses 16 bit values (numbers from -32768 to 32767) and 44100 of these values per second per each channel (44.1 khz). Most "hires" recordings use 24 bits per sample and often sample at 96khz (96000 numbers per second per each channel). The process of taking the "electrical" values of the original musical signal and converting them at tiny intervals into numbers is called sampling. It is done with an ADC (analog-to-digital converter).
Now, the frequencies that can be represented by a system using N samples per second is N/2, and the human hear does not hear over 20Khz (except for some rare cases people in their youth). So values of N above 44100 makes little sense, but the matter is debated.
Some files are even sampled at 192 khz or 384 khz. Some recent DACs allow even inputs as high as 768 or 1536. Where even frequencies that not even bats can hear would be representable. But if they were not in the original recording, they cannot be created out of nothing, and thus it is a waste of energy, storage space, bandwidth.
Many audiophiles believe that upsampling (or oversampling, i.e. converting, say, a 44.1Khz file to a 352.8khz file, for instance, by inserting intermediate values between the samples of the original file) improves the sound. There is some truth to this and this is why many DACs do that work internally (it has to do with distortion introduced when converting back from digital to analog at each step, which creates additional "images" of the original signal at higher frequencies that can then interact and create distortion back into the audible range) but there is no real proof that upsampling done in software on a computer before sending the music to the DAC actually can improve the sound – at least not with the current batch of chips (I have had a DAC where I can swear I heard better when upsampling before, but I made no doubly blind test, so this is not proof).
[REMARK: In fact, sometimes the opposite is true. A lot of hires files have a LOT of noise and distortion at the higher frequencies, because they may have be upsamples badly, or it was just the raw output of an ADC that was edited leaving sampling noise in the higher frequencies (this happens often because of the way many ADCs work). This noise can actually make the internal mathematics of a DAC while oversampling FURTHER work bad. You can actually get better sound by downsampling those files first to remove the high frequency cruft, and then sending the resulting smaller and cleaner file to the DAC.]
... Many audiophiles believe that upsampling (or oversampling, i.e. converting, say, a 44.1Khz file to a 352.8khz file, for instance, by inserting intermediate values between the samples of the original file) improves the sound...
thank you @mocenigo for your thoughtful explanation. i appreciate it very much.
what you're saying is that oversampling is essentially interpolating and inserting the interpolated values in-between real world values.
i wonder why that would improve sound? the value is exactly the same as the two closest real values so no additional information is provided!
ok, back to the D90 or the D90 MQA: why do people say it happens only to 768kHz files and why do they say it is never used? is it because no real world recordings were ever recorded with 768kHz?
if that's the case, when would recordings with 768kHz even show up?
Well, the process of converting the signal back to analog using a DAC in its simplest way causes a lot of little jumps. These jumps are the differences between two adjacent samples, and are of course correlated with the original signal and you can see them as a discrete version of the derivative of the original signal. Since the original signal can be viewed as a sum of sine functions, you get another sum of sines (well, cosines). The frequencies that make up the new signal are the original frequencies but added and subtracted at all multiples of the sampling frequency.
Now, usually, a DAC has an analog filter to kill these frequencies, but some artifacts due to the interactions between these new frequencies can generate further distortion back in the audible band.
These spurious signals, being generated by signals that are correlated to the size of the “jumps”, depend on the latter as well.
Hence, upsampling and interpolation will both push these frequencies much higher AND reduce the sizes of the jumps, thereby reducing the distortion and making the analog low pass filter after the DAC chip work more effectively. So this is a good thing, what I wrote above and that you highlighted on green was a critique of doing that BEFORE a well designed and modern DAC. All of this is currently done already INSIDE the DAC chip, and therefore there is no need to do it before.
There is on top of that, an attenuation of the sound components that are close to half of the sampling frequency (the Nyquist frequency).
Now, in the past many DAC chips did not do over sampling, and many audiophiles believe that a chip should NEVER change the values fed to them.
But wait, here is where things start to get ridiculous: a lot of audiophiles look for non-over sampling chips, because they are “bit perfect” and thus better than current over sampling chips (I have explained above why this is cannot be true) AND then they feed them oversampled data. Because they feel they can tune it to their liking and they own it. So it is just a more expensive way of doing something that is done in a better way on more modern chips.
And, finally, the insanity. Feeding oversampled data to modern DACs that already do their own internal oversampling. Redundant, wasteful of resources, and potentially detrimental to the sound. But, hey, who am I to stop them?
ok, back to the D90 or the D90 MQA: why do people say it happens only to 768kHz files and why do they say it is never used?.
It is just a programming bug and it happens only at that frequency (maybe also 705.6k?). I suppose some people do over sample to 768khz on devices without such a bug.
is it because no real world recordings were ever recorded with 768kHz?
if that's the case, when would recordings with 768kHz even show up?
Trying to decide if the clumsiness of using this DAC for streaming purposes in a full AV is worth it. Currently using Audiolab 6000N Play via RCA to X4700 in preamp mode (to Halo A52+). If I understand integration for best sound quality using Tidal/Roon correctly (use Amazon HD now), It would be USB from laptop to D90 and either XLR to RCA or just RCA to RCA to the Denon. My AV equipment is on the side of my room so this isn’t ideal (obviously wireless Play-Fi app connection with 6000N is more convenient) but I do have empty PVC running from my rack to underneath the sofa so it isn’t the end of the world. Plus, if laptop is connected to network via Ethernet (as my 6000N is) then having a wired connection to the D90 too would prevent any dropouts. Anyone using the D90 in an application like I propose here? Thanks.
may i ask if there are Windows 10 settings that need to be set/adjusted when i initially connect the D90 to my Windows 10 PC tower's USB port? i keep reading that there is a steep learning curve to the D90 and that things aren't just intuitively obvious how to set it up and get it going, and this is causing me to pause and not just buy the D90/A90 set for my computer desktop.
i've read that there are bit rate settings and kHz settings on the Windows 10 pc that one needs to select to optimize the streaming out to the external D90 dac. i haven't done that before, the manual of the D90 doesn't tell you what to set it at either. If there are certain things that one needs to do to connect to a Windows 10 pc, i'd most appreciate learning about them now and i'll make a to-do list right away. thank you.
otherwise, if it were merely plug and play, i'd have gotten it immediately.
may i ask if there are Windows 10 settings that need to be set/adjusted when i initially connect the D90 to my Windows 10 PC tower's USB port? i keep reading that there is a steep learning curve to the D90 and that things aren't just intuitively obvious how to set it up and get it going, and this is causing me to pause and not just buy the D90/A90 set for my computer desktop.
i've read that there are bit rate settings and kHz settings on the Windows 10 pc that one needs to select to optimize the streaming out to the external D90 dac. i haven't done that before, the manual of the D90 doesn't tell you what to set it at either. If there are certain things that one needs to do to connect to a Windows 10 pc, i'd most appreciate learning about them now and i'll make a to-do list right away. thank you.
otherwise, if it were merely plug and play, i'd have gotten it immediately.
The D90 MQA / D90 setup to USB Windows is easy. You can rely on setting a fixed sample rate / bit depth and let the players set the source rate which is picked up by the D90 MQA during unfolding - it might say 384k PCM before hitting a MQA file / track and then show 192khz as the unfolded rate, but the USB settings are fixed - unless you want to play with them manually, but it's not needed.
Here is a "step by step" image displaying how you set up the particular device settings for the D90 MQA in the Windows Sound Control Panel:
Select the playback device in the Sound Control panel to modify it's settings. A 2nd dialog will pop-up and Stereo is already selected - you can click the Test button to verify wiring if you like, or click Next...
Then you are shown a new dialog, you only have one option to set: "Full Range Speakers", check that box and click Next:
That's all there is to the "Configure" settings, so click "Finish"
Now click "Properties" to set the Output Level, Sample rate / bit depth, and exclusive access (needed for Tidal / Amazon HD for best sound options.
A lot of people miss this important point, always set the source drive to 100% to get the best signal to noise result, less than 100% is bad - 100% drive is best. Analog or Digital always drive 100% from source to final component controling volume to AMP's. Exception is made if the driving device would overdrive the next component in the audio chain - then adjust the drive to 100% of what the component can accept.
And, the last item is the one you are overly concerned, the default bit depth and sample rate setting for the USB interface on the D90. Notice also that I have selected the Exclusive mode required by the D90 MQA to allow full control of the USB DAC by Tidal to do the final unfolding of the MQA stream:
You can pick the maximum source sample rate you will be using and limit the abilities of the D90 MQA via this control panel, but I simply set and forget it at 384khz / 24 bit, and let the source deliver to the D90 MQA exclusive access to adjust the output - you can see on the D90 MQA display what the current source is playing at through the D90 MQA - and when PCM changes to MQA detection the higher (or same) sample rate will be updated in the D90 MQA display - along with the MQA light, see example below:
Click to see full sized
That is pretty much it for D90 MQA Windows USB device setup to support Tidal MQA streaming - or for other streaming services where the D90 MQA will show PCM in the display instead of MQA.
Trying to decide if the clumsiness of using this DAC for streaming purposes in a full AV is worth it. Currently using Audiolab 6000N Play via RCA to X4700 in preamp mode (to Halo A52+). If I understand integration for best sound quality using Tidal/Roon correctly (use Amazon HD now), It would be USB from laptop to D90 and either XLR to RCA or just RCA to RCA to the Denon. My AV equipment is on the side of my room so this isn’t ideal (obviously wireless Play-Fi app connection with 6000N is more convenient) but I do have empty PVC running from my rack to underneath the sofa so it isn’t the end of the world. Plus, if laptop is connected to network via Ethernet (as my 6000N is) then having a wired connection to the D90 too would prevent any dropouts. Anyone using the D90 in an application like I propose here? Thanks.
Jenving makes Supra long run USB cables up to 15m that aren't too expensive - I haven't tried a long run, but the short run 0.7m is very nicely made. http://www.jenving.com/products/usb-121
Here in the US Jenving have a distributor that sells through eBay:
I also have had to install long runs from the AV stack to the projector / AMPS / screen and those cables were more expensive at the time for similar quality. Prices have dropped nicely over the years if you shop for competent function at reasonable prices.
Jenving's distributor on eBay was very helpful and quickly contacted Jenving to get my questions answered for some of their products, and the interaction was honest and responsive - they actually recommended another solution not available in their product lineup.
As to the rest, it's a learning experience each of us goes through, you'll have lots of fun deciding on how to go about it all - and tear it up and start again many times. I personally went from simple to complex and back to simple set up's over 50 years, there is lots of time to experiment.
A DAC isn't always the best upgrade especially if your system has improvements available to upgrade the Speakers / headphones / amp / pre-amp - perhaps even necessary in order to hear the benefits of a better / more resolving source material and the devices to play them - like a DAC / Streamer.
For me the D90 MQA + A90 for a single position headphone / IEM listening stack is a nice optimal package that I enjoy, but if I was wanting to build a distributed network of source devices and remote zones of playback, there might be the better choices.
I don't have a personal need to architect that kind of distributed system for audio with current hardware, and I haven't done much to firm up what my specific choices would be. I have read and watched a lot of information on building such a system, and there is a lot of information out there on how to spend a lot of money on it.
For me keeping it simple, running a long USB cable would be a better choice than trying to upgrade routers, switches, ethernet cards / dongles, getting upgraded low noise power supplies, and finally upgrading the in wall ethernet cables from CAT5E to CAT7/8, etc etc.
For me a fixed position at a desk, table, couch for control of media, and distributing the output to an AV stack and from that stack to a projector worked well. I also have a nice portable DAP - FiiO M15 so I can travel around the house and outside (what's that?) so I can bring it with me.
Bluetooth in range works well driving the D90 from the FiiO M15 + D90 MQA remote, it is of limited use to me right now but it could solve your PC to D90 MQA distance issue without resorting to a long USB cable. You lose some fidelity, but I think it sounds fine.
You could build gradually, getting a new higher resolution media play can reveal limitations in the equipment path you hadn't notice before, it helps to have a great headphone / IEM to sanity check what your system / speakers are doing vs direct to the D90 MQA + Pre-amp / headphone amp.
You might consider getting the A90 too and use that as the headphone amp / preamp out to your AV stack receiver / pre-amp, You could set the D90 MQA to DAC mode (fixed to maximum output) and set the A90 output via volume dial to a fixed point for driving the AV stack - and use the remote for the AV stack as volume control.
Please come back and let us know what you decided and how it worked in your system
...I'm actually in a pretty good spot right now with my B450 3700x 2070 Super USB noise situation. I have no noise with the XLR only, and by adding the RCA connections to the mix I don't have any added noise now with the Supra USB cable - unless I use GPU compute. There is probably additional noise coming along that ground path that aren't "audible" but suck amp resouces - those I would like to get rid of too.
I found another Amazon listing as a $125 bundle iFi iDefender+ CA + iPowerX 5V (save $24 with bundle) that I can use to test with / without the power with the D90 MQA (and other powered USB devices), I'll report back on how that works out. There is a cheaper $89 iFi iDefender+ CA + iPower 5V bundle (save $9 with bundle) with the previous design iPower (non-X) power supply, I went with the newer iPowerX (bundle).
I really didn't want to get a "dongle" solution, but it's a lot cheaper than the other isolator / converter alternatives and the ground breaker + external power solution should be transparent to the audio.
The single iFi iDefender+ + iPower X 5v bundle wasn't effective - until I used 2 of them on both my DAC's with USB connections to my PC.
Briefly, the first ifi Defender+ was the USB-C to PC / USB-A to DAC cable and my D90 MQA doesn't need power so I didn't connect up the iPower X external 5v power supply.
This fist configuration did nothing to reduce the GPU compute noise - a bit of a surprise. I didn't know there was a 2nd path for the noise to enter, through my M15's DK1 dock connected to the same PC via USB.
iFi sent me a second USB A - USB A version of the iDefender+ as that is what I needed and what I thought I ordered from Amazon for the D90 MQA cable to PC. Initially, I had to use a USB C to USB A adapter until the 2nd iDefender+ arrived. iFi's Amazon listing showed the A/A version in photos - in fact, my order product image showed that A/A version - but Amazon sent me the C/A version. I contacted ifi and they quickly sent the A/A unit and said they would update the Amazon listing photo's to match the actual product.
When the 2nd iDefender+ arrived I swapped the C/A iDefender+ for the A/A iDefender+, and still, no joy, the PC GPU compute noise was still there.
A few days went by, and I had a few minutes and I thought - why not hook up the 2nd Defender to my other DAC connected to my PC? (duh?)
The FiiO M15 + DK-01 stand with independent power connection - also connected to the same PC - which appeared to have no common power ground, and sure enough when I connected the 2nd iDefender+ up and plugged in the iPowerX 5v to the DK1 - the compute noise disappeared.
If I only had one of those iDefender+'s I would not have realized I needed to break the USB power/ground on both the DACs connected to my PC via USB.
Now that I know noise is also coming through that path through the power strip to the PC, I am going to swap in the another AV power conditioner and see if I can nix the noise at the source.
@hmscott Thank you for your thoughtful response. I’ve been thinking about my options and I have not come to conclusion. The Audiolab 6000N Play sounds really good using Amazon UHD files and as mentioned its just easy. Plus it found my NAS easily and plays those files flawlessly. However, the Play-Fi app is somewhat clunky and I am certain the sound quality of the D90 would be superior to the 6000N. Lastly, Tidal continues to evolve its eco system and MQA file count where Amazon seems complacent, already.
The “easy button” to moving to an MQA platform would be a Blusound Node 2i but I’d have to use its analog outputs as my Denon doesn’t unfold MQA files using its onboard DAC, and I believe my 6000N is superior when it comes to sound quality.
I don’t believe I’ll be upgrading any other gear anytime soon. The Denon and Parasound are brand new And I really like the sound. Especially at the price point.
So, it comes down to the following; extra money invested in Topping, always having to be a wired connection and using a laptop versus iPad/iPhone for listening session. The benefits? Arguably superior sound using Tidal versus Amazon, using a more widely supported service in Tidal, better sounding source in D90 versus 6000N (how much better???) and new toys.....
I don’t think I mentioned I am building a new home with a dedicated space and everything will be a wired connected via Cat7. Using the Topping via USB and having the laptop wired via Ethernet would prevent drop outs, the 6000N occasionally glitches during playback.
Does the specification of the laptop and software matter when it comes to sound quality when using the D90? Do you use Roon?
And, finally, the insanity. Feeding oversampled data to modern DACs that already do their own internal oversampling. Redundant, wasteful of resources, and potentially detrimental to the sound. But, hey, who am I to stop them?
Your attempt to explain the oversampling (interpolation) process in delta sigma DACs was quite ok, but the quoted passage doesn't make sense. If you feed an usual DAC chip with for example 44.1k sample rate signal, then it will internally oversample typically to 352.8k in the 1st stage (that's called 8x oversampling) and then in the second stage it uses a much simpler oversampling technique (often only 'Sample and Hold' one) to further significantly raise sample rate to operating frequency of delta sigma modulator, which is typically 5.6 MHz.
If you feed a DAC chip with 352.8k signal, then the 1st oversampling stage will be skipped in the DAC chip and only the second stage will apply. So there is nothing redundant. Either DAC will oversample from 44.1k to 352,8k, or you do it in a software player. One or the other. What's then the difference? We have to compare the quality of hardware oversampling vs. software upsampling. A typical personal computer is capable to perform much more precise calculations than a resource constrained few dollars costing DAC chip. The first upsampling/oversampling stage has higher influence to the resulting quality.
Hi! mind you, I was trying to convey the process in a simplified way to a posted that asked for a simple explanation. If you want to meet in a (virtual) room with a (virtual) blackboard (I hate white boards) I can be more precise ;-)
but the quoted passage doesn't make sense. If you feed an usual DAC chip with for example 44.1k sample rate signal, then it will internally oversample typically to 352.8k in the 1st stage (that's called 8x oversampling) and then in the second stage it uses a much simpler oversampling technique (often only 'Sample and Hold' one) to further significantly raise sample rate to operating frequency of delta sigma modulator, which is typically 5.6 MHz.
True, but not all DACs are sigma-delta. Some are R2R that stop at the 325.8k (or 384k, depending on which one is a multiple of the base frequency) or at higher frequencies, and do not perform sigma-delta modulation, so the argument would apply to those. But, yeah, here I simplified a bit too much in order not to overload the target reader.
If you feed a DAC chip with 352.8k signal, then the 1st oversampling stage will be skipped in the DAC chip and only the second stage will apply. So there is nothing redundant. Either DAC will oversample from 44.1k to 352,8k, or you do it in a software player. One or the other. What's then the difference? We have to compare the quality of hardware oversampling vs. software upsampling. A typical personal computer is capable to perform much more precise calculations than a resource constrained few dollars costing DAC chip. The first upsampling/oversampling stage has higher influence to the resulting quality.
What you say is true - there may be difference between the oversampling done in a computer and oversampling in a chip. However, ASICs can contain circuitry that is closely tailored to the intended purpose, and be much much more power efficient than a commodity PC. So the difference may be much smaller than you may imagine (esp. with more recent chips) and in fact also work in favour of the DAC chip.
Case: with the ES9018 (in a Gustard X20) I found that oversampling in SW yielded a better (subjective) result. With a AK4499 (Topping D90) I prefer to NOT have SW oversampling (even though, of course, with SoX I can play with a lot of parameters).
But, yeah, I confess to not have told the whole story – and in the part you criticised I omitted important information.
Thanks. True, I didn't count with R2R DACs in my post. There are differences also between software upsamplers. For example I am using HQPlayer with variety of upsampling options. In the end the choice is on listener and his preferences.
Thanks. True, I didn't count with R2R DACs in my post. There are differences also between software upsamplers. For example I am using HQPlayer with variety of upsampling options. In the end the choice is on listener and his preferences.
There are definitely differences. TBH however I haven't heard many differences during the last years between filters. And I usually catch very fine details, but being a musician (by training - degree in composition) my brain probably gets more musical details (like a contrabass playing a slightly out of tune note in a pianissimo while the rest of the orchestra blasts ffff as if that was the last thing they would do before the end of the world) than audio ones, like subtle decays or "plankton" or whatever audiophiles "hear" (I also hear no difference in detail when they claim so, and usually identify it as treble unbalance, har har har). Does anybody here hear differences between the filters of the Topping D90, for instance?
So, for instance, I would be delighted (yes, delighted) if you could me what to observe in the sound when different types of filters are applied.
For example: With HQPlayer it is typical to choose a linear phase filter for classical music and some minimum phase filter for pop recordings. The reason is that classical music content does not contain so quick changes like modern pop with artificial sounds, which may have for example very fast decay - such a thing does not exist in classical music with slower type of sounds. With classical music a listener typically wants to feel more of the natural recording space. The preringing, which is always present with linear phase filters, does not cause issues with slower nature of classical music sound and "space" information is better preserved. On the other side, for pop content, the typical case is that the "space" is artificially created in studio and is less important for the overall feel of the recording. Accurate timbre of instrument sounds, including those very fast ones, is more important and for this case minimum phase filters without preringing and with shorter decay are generally more suitable.
Bigger difference than only PCM upsampling makes when you do also software delta sigma modulation to convert PCM to DSD (on the fly) and feed your DAC with DSD signal. That has sense to do if a DAC like D90 supports the so called direct DSD path (skipping complete hardware oversampling and delta sigma modulator). That's the main reason why HQPlayer exists. Simply the quality of digital signal processing matters. HQPlayer provides variety of delta sigma modulators to choose from. So it provides an alternative way how to use a delta sigma DAC and how to listen to music.
Most people will use a free player and simpler traditional form of listening to PCM content, but there is a technical alternative for those, who want more choices and more to experiment.
