Notice one of the waveforms NOS. |
A few weeks ago on this blog in a comment discussion, Bennet / Dtmer Hk talked about showing what it looks like when upsampling a 1kHz 16/44.1 signal in PCM side-by-side with DSD upsampling.
Sure, no problem! We can have a quick peek at the 1kHz sine tone from a couple of DACs for comparisons between direct playback with built-in filtering, upsampled in high quality with PCM SoX, and then using high quality DSD conversion with SoX-DSD.
Plus, let's also have a look at the recent Beatles: Get Back documentary series and some thoughts which I think relate not just to the "music lover" which I hope is in all of us, but also to the "hardware audiophile" side of this hobby.
First, let's show the 1kHz sine upsampled in various ways as noted above. We start by defining the parameters I'll use in SoX 14.4.2. Starting with a clean 16/44.1, -0.2dBFS (a little overhead so as not to clip), 1.00227kHz (reduce rounding off at 44.1kHz) sine wave:
Converted To PCM 24/384 with:
sox input.wav -b 24 output.wav rate -v 384000 dither -S
These parameters were as recommended by Bennet and look like excellent choices if you want to do high quality upsampling. This will create a 24/384 (we could aim for 352.8kHz as well but honestly it makes no practical difference) output file with "very good" filtering. Also with high-pass dithering.
Converted also to DSD64/128/256 - using Måns Rullgård's SoX-DSD mods:
sox-dsd input.wav output.dsf rate -v 2822400 sdm -f sdm-8
<DSD rate = 2822400 for DSD64, 5644800 for DSD128, etc.>
These are the same parameters as discussed a few weeks back when testing DSD performance with the various DACs.
Using the TEAC UD-501 and Topping D10 Balanced DACs, captured with E1DA Cosmos ADC (mono mode), here are a few side-by-side high resolution FFTs - captured using FlexASIO at 32/384 (192 bandwidth), 512k-points:
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As you can see, the TEAC graph includes the NOS mode where we can see all the ultrasonic imaging artifacts when we run the DAC filterless, top right is the standard direct playback with the typical "sharp" filter I use the vast majority of the time. The more modern Topping D10 Balanced doesn't have a NOS mode but can handle DSD256, pushing the ultrasonic noise shaping further out. As shown back in August, the D10 Balanced has by default a steep minimum phase filter for direct playback.
If we do some pixel peeping of the SoX PCM upsampled FFTs, we can make out the steeper and deeper post-Nyquist attenuation compared to the DAC's direct playback thanks to SoX's "very" high quality sharp linear filter. Of course, DACs do have noise floor limits so there's only so much attenuation possible. Even if the low-pass filter applied is capable of mathematically achieving 32-bits with better than -190dB attenuation, that's just a numbers game (that sometimes gets tossed around as if meaningful) and in real life we're not going to see that magnitude in actual playback from the analogue outputs.
Note that distortion measurements like the THD / THD+N do not and really should not change much** with various PCM and DSD processing/upsampling. Feeding a DAC with 24/384 data compared to 16/44.1 over USB could account for some of the change and likewise DSD vs. PCM will also lead to variation. Do not assume that big numbers like 384kHz upsampling or DSD256 will automatically lead to higher fidelity - as you can see, this is not necessarily the case. This is also why having test data for your specific DAC is essential.
[** Interestingly, the biggest difference I see is with the Topping D10B going from 16/44.1 to 24/384 PCM. Notice the increase in THD. I wonder if we're seeing the same phenomenon as the Topping D90SE at and above 192kHz suggesting that there are potential optimizations that can be tweaked with the ESS THD compensation parameters at higher samplerates.]
Like with the measurements, I don't hear a difference in the sound between various filters for the most part although there are exceptions. For example, the slow roll-off filter with the Ayre/PonoPlayer a few years ago is an example of one that did change the sound noticeable. Sure, we could do fancy stuff to allow even more precise filtering - Chord's WTA, HQPlayer upsample, even try out PGGB if you want. You can also play with minimum or intermediate phase settings (as discussed previously).
What's important IMO is that we keep the effect of upsampling in context and realize that this is just low-pass filtering, not more-complex DSP like FIR room correction. As per this post from Måns, you don't need "megatap/megaorder" filters to achieve more than adequate very low passband ripple and large stopband rejection. As he showed, ~400-taps is more than enough, providing 180dB stopband attenuation for any amplifier, and certainly for any human ears. Make sure to let that sink in so that we can be better informed consumers.
Again, big numbers might seem impressive at first and perhaps can spur sales based on specs, but in reality, this is just not needed and eventually people get wise to the marketing and become unhappy if pushed beyond the realm of believability. Looking at the SoX PCM upsampling debug log, it's "only" using a 573-taps FIR with 186dB stopband attenuation for the 44.1 upsampling to 32/384kHz internally. This can be increased to 645 taps with 210dB attenuation if we use -u instead of -v for "ultra" quality.
As usual, for those who advocate for really really complex upsampling algorithms and enormous tap lengths, it would be nice to see evidence that this makes a difference with actual DAC output rather than supposed "subjective" claims of improvement with no apparent basis in physical reality. Even if one insists that ears are the best tools to appreciate differences because one doesn't believe that measurements can capture supposedly audible changes, let's get some controlled listening (like blind testing) to demonstrate/confirm the claims!
Be mindful of these ideas when you're on audiophile forums and people are talking about using stuff like powerful Intel i9, Xeon, AMD Ryzen 9, Threadripper CPUs, and leveraging the power of GPUs to perform all their upsampling whether to PCM 768+kHz or DSD512+ as if this too is some kind of impressive feat. I think we've all seen examples of multi-thousand-dollar computers being used for this purpose with all kinds of claims that it improves sound (like this product). IMO, this is all highly unlikely unless one is also applying room-correction or other type of DSP that can definitively change the sound.
Audiophiles spend a lot of money (and time) on things which I think are unnecessary and suspicious, with claims of subjective improvements. Hey, if this is part of one's "pursuit of happiness", sure, go have fun! Just make sure in principle not to entertain too much snake-oil or be too giddy about fantasies, because at the very least, that would be bad for the reputation of the audiophile hobby as a reality-based technical endeavor.
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Starr & Preston playing with the Stylophone. Lennon taking a sip. |
Hi Arch, the speakers should be Altec 612 with leak TL/25 Amp below.
ReplyDeleteFound some pictures on the internet
How do they sound? I don't know
Thank you for the excellent work as usual
Thanks chancedon!
DeleteAwesome to know what kind of gear to keep an eye out for in the future to have a listen to... Maybe even opportunity to measure if able!
You could certainly argue "Let it Be" wasn't as the artist intended. McCartney didn't have approval of the mix/album and was engraged with what Spector did with some of his material - especially "The Long and Winding Road". That's the reason for "Let it Be...Naked".
ReplyDeleteI guess it depends on how you define "the artist".
Yup, good point Unknown.
DeleteThere are the stories of the music itself, and then the stories of the whole production process before the final mix and product.
I do hope that claims around "as the artist intended" blows off over time as if this has any actual meaning or even that this somehow enhances the authenticity of what we buy.
If an artist intends that there be some kind of "authenticity" process, then let's just have a description of the system, room, volume level, etc. to describe the day the artist finally listened to "Album X" and gave a thumbs up. Be done with it.
Otherwise, I honestly don't think the consumer cares to any meaningful degree as to have companies make money off this kind of claim!
I'd say there is at least some merit to HQPlayer, since some of its filters are apodising (you can usually tell those by the fact that they cut off around 21.02khz rather than the usual 21.97), and work well on badly mastered files.
ReplyDeleteSure Techie. Maybe that's beneficial.
DeleteI wonder if anyone has formally tested this? Could be a great little research project!
I wonder how many people aged 40+ would be able to discern a difference between a 21.97kHz and 21.02kHz filters?
ReplyDeleteI'm guessing 100% of mainstream audio reviewers in non-blind tests. (Happy holidays everyone)
Delete+1 to Phil.
Delete40+ is very generous. Let's talk 10+. :-)
Happy holidays to you and yours Ralph and Phil!
Did you see the interview with Taiko's CEO https://youtu.be/ggby6Vhi_9E ? (The second part of the interview is released today).
ReplyDelete"Thanks" for the link Geert.
DeletePainful series of videos. Thankfully YouTube allows 1.5x playback speed ;-).
I honestly think these guys either:
A. Hallucinates what they believe they hear. Or are delusional. Either way they're psychotic and have lost touch with reality and insight.
or...
B. Shilling. Snake-oil salesmen. Are in it to make money or achieve some ego-based attention.
I would bet it's Option B more likely than A, ya never know!
Thanks for the tests. So PCM1795's relatively short (compared to PCM1792A/PCM1794A) PCM filter does not seem to cause any penalty. Harmonic distortion and noise shaping dominate the ultrasonic frequency region with both on-chip filter and SoX resampling. On the other hand, DSD128 still has considerably more ultrasonic noise than PCM.
ReplyDeleteES9038Q2M on the Topping shows an obvious dip at around 22kHz followed by a sharp rise of noise when using DSD64. For DSD128 and 256, while the rise of noise is pushed to higher frequencies, the dip at around 22kHz is much less obvious, this phenomenon is similar to ES9038Pro: higher <20kHz noise floor when using higher DSD rates.
Yup Bennet. A little bit of this noise or that distortion...
DeleteBut nothing I'd be too worried about with either the segmented TI/BB DAC or multibit-SDM ESS. I remain unconvinced that the underlying DAC technology sounds different if the objective results look similar with low level variation...
Happy Holidays man!
I tried to watch that Beatles docu, but I guess I'm not a big enough Beatles fan so I couldn't bare longer than 40 minutes. Kinda annoying thou with the horrible picture quality, first way to much noise reduction and then some bad looking ai-upscaling on top of that. They tried to make it look modern, and I guess they succeeded since it looked like a bad smartphone camera..
ReplyDeleteGood point Tell about the picture quality with clear post-processing done. I think also pumped up the saturation for the colors.
DeleteYeah, I definitely agree with the sentiment about how much you could take. As I mentioned, my wife managed about 1 hour. We at least had some popcorn to start and enjoyed some 18-year aged, single malt whiskey at the end of the hour ;-).
A fun piece of history for the fans. And targeted to more of the "hardcore" demographic.
The original was intended for 1969 UK TV, and so was filmed in 16mm.
DeleteFor the 1969 theatrical release they turned the 16mm into 32, so it looked like S***, as it was just optically pumped up.
This new digital version looks way better than the theatrical release 50 years ago.
So it's all relative. Can't really compare video/movie tech from the 60's to today.
Could you test some Asus motherboard with headphone amp? Saw your reviews for Gigabyte (they suck from personal experience and what I saw from other people online) and MSI, but no Asus.
ReplyDelete