BLIND TEST INVITE: Do digital audio players sound different?

[Ralf11]

One of the great things about science is that it allows us to make statements about what other people are capable of hearing.

 

Also, other species.

 

Also, it allows us to extend our "sensory range" deep into the universe, where we cannot physically travel.

[Miska]

3 minutes ago, esldude said:

Okay so how does the imaging of the RME show up where we can hear it?  And I will say if someone is saying they can hear such things at -120 db they'll have to prove it. I'll say I don't believe it for all 7 billion people on the planet.

 

Some people have proven it to me in practice, that's enough for me.

 

Rather than going around asking people to prove their hearing, I rather go to the way that I don't need to think about the subject.

 

I don't understand why you are against correcting measurable and correctable problems?

 

6 minutes ago, esldude said:

Now okay the RME has some imaging at -46 db.

 

RME doesn't, Meridian Explorer 2 does. Which is mostly function of the DAC chip in question (which, by the way has 60 dB stop-band attenuation in the built-in digital filters).

 

Same DAC chip can perform much better already by running it at 352.8/384k with good external upsampling. But you probably want to ignore that too?

 

[fas42]

No-one's hearing stuff at -120dB. What may happen is that the circuitry somewhere is impacted by the presence, or absence, of particular signal, not necessarily directly correlating with the audio - which injects noise, or distorts the audoio content in an audible way ... hence, you 'hear' the difference.

 

Lots of the difficult stuff to be sorted in systems is very non-obvious, not directly connected to some clear culprit - which makes it so much harder to "debug" what's going on.

 

That said, if the circuitry is behaving itself beautifully, in a conventional, measurable sense, then the chances are that the audible results will be better, .

[Miska]

57 minutes ago, esldude said:

Here are some for the Forte, which I think you also have.  

 

OK, let's take a look at Forte. Test tones as 44.1k sampling rate 32-bit PCM.

 

First playing 1 kHz tone, wide band spectrum:

Whoops, not very clean...

 

Then a 0 - 22.05 kHz sweep:

Now we can see the noise shaping bump, two image pairs and huge HF noise bumps.

 

Let's zoom in to low frequencies:

Here we can see first image pair around 176.4k and second around 352.8k. And also remains of the modulator noise hump. This looks like Cirrus Logic DAC chip, except that for example my own implementations based on their CS4398 DAC chip are totally clean above the noise hump, and so is for example Marantz HD-DAC1 that is based on the same chip. Maybe this is some other chip model from Cirrus. Doesn't explain the HF noise humps though.

 

It also has some aliasing and funky behavior with full levels signals near Nyquist. The switching PSU makes a faint tone just above 30 kHz.

 

 

 

[Miska]

P.S. Possibly the 2.8 MHz bump is the actual converter fundamental, looks a bit different than what the old CS4328 was spitting out and that I know is running at 2.8 MHz (44.1k x64). There's also a peak at half of that 1.4 MHz. 4.2 MHz is 1.5 times 2.8.

 

[esldude]

10 hours ago, Miska said:

 

Some people have proven it to me in practice, that's enough for me.

Not for me as I've said. 

10 hours ago, Miska said:

 

Rather than going around asking people to prove their hearing, I rather go to the way that I don't need to think about the subject.

 

I don't understand why you are against correcting measurable and correctable problems?

 

I'm not against correcting measurable problems.  

10 hours ago, Miska said:

 

RME doesn't, Meridian Explorer 2 does. Which is mostly function of the DAC chip in question (which, by the way has 60 dB stop-band attenuation in the built-in digital filters).

 

Same DAC chip can perform much better already by running it at 352.8/384k with good external upsampling. But you probably want to ignore that too?

 

If upsampling fixes the issues then good, but it would be nicer not to need that fix. 

[Miska]

18 minutes ago, esldude said:

If upsampling fixes the issues then good, but it would be nicer not to need that fix. 

 

As long as you deal with PCM, at rates like 44.1k, you will need digital filters as part of the reconstruction. And you need to run those somewhere. DAC chips are very constrained in terms of DSP power, so they cut a lot of corners and leave it short. And that shows up.

 

[asdf1000]

1 hour ago, esldude said:

If upsampling fixes the issues then good, but it would be nicer not to need that fix. 

 

Record at DSD256 (as @Miska suggested) or PCM768kHz (one day...)

 

I believe Rob Watts is making an analogue-to-PCM768k converter, called 'Davina'...

 

"I have designed the first decimation filter for Davina, and this has 260 dB rejection, so this will ensure aliasing is in practice well below -300dB."

 

"It goes without saying that the Davina project (my ADC) will have none of the above shortcomings; absolutely no aliasing, and a very simple connection from Mic to the ADC to maximize transparency."

 

 

https://www.head-fi.org/threads/chord-electronics-dave.766517/page-120#post-12376871

 

https://www.head-fi.org/threads/chord-electronics-hugo-2-the-official-thread.831345/page-467#post-13653052

 

https://www.head-fi.org/threads/chord-electronics-dave.766517/page-123#post-12382623

[esldude]

10 hours ago, Miska said:

 

OK, let's take a look at Forte. Test tones as 44.1k sampling rate 32-bit PCM.

 

First playing 1 kHz tone, wide band spectrum:

Whoops, not very clean...

 

Then a 0 - 22.05 kHz sweep:

Now we can see the noise shaping bump, two image pairs and huge HF noise bumps.

 

Let's zoom in to low frequencies:

Here we can see first image pair around 176.4k and second around 352.8k. And also remains of the modulator noise hump. This looks like Cirrus Logic DAC chip, except that for example my own implementations based on their CS4398 DAC chip are totally clean above the noise hump, and so is for example Marantz HD-DAC1 that is based on the same chip. Maybe this is some other chip model from Cirrus. Doesn't explain the HF noise humps though.

 

It also has some aliasing and funky behavior with full levels signals near Nyquist. The switching PSU makes a faint tone just above 30 kHz.

 

 

 

This is unusual behavior.  It only does this at max level and between 21,750 hz and 22,050 hz with the Forte I have.  An ADC shouldn't let such signals thru, but the Forte shouldn't act that way.  Tried another DAC and it does none of this.  However, if you drop the signal to -1 dbFS that goes away. 

 

I suppose you've noticed the 31 khz tone around the PSU, also causes faint aliasing in the ADC not around Nyquist, but around the 31 khz tone.  Making me wonder if that is from the PSU.  

 

Here is a 48 khz and then 96 khz sweep recorded at 192 khz. 

 

[Miska]

1 hour ago, esldude said:

This is unusual behavior.  It only does this at max level and between 21,750 hz and 22,050 hz with the Forte I have.  An ADC shouldn't let such signals thru, but the Forte shouldn't act that way.  Tried another DAC and it does none of this.  However, if you drop the signal to -1 dbFS that goes away. 

 

Modern ADC's will let things through up to 22.05 kHz and a little bit over as aliasing. It is property of the on-chip digital filter, if you look at the level response you can see that there's a frequency reponse boost around that area. If you upsample 44.1k content to 192k that goes away. Although related to this the problem is not so much what the ADC originally has let through, but mastering in digital domain pushing levels to clipping and then that triggering such behaviors in clipped transients.

 

1 hour ago, esldude said:

I suppose you've noticed the 31 khz tone around the PSU, also causes faint aliasing in the ADC not around Nyquist, but around the 31 khz tone.  Making me wonder if that is from the PSU.

 

Yes, I mentioned that. It is from the internal switching PSU (likely some DC-DC converter), since it is independent whether the device is wall-wart powered or not. And the frequency matches such. Quite typical, I'm less worried about that than the mess it leaves at audio band.

 

1 hour ago, esldude said:

Here is a 48 khz and then 96 khz sweep recorded at 192 khz. 

 

 

Yeah, it is somewhat messy. And running it at 192k doesn't fix that either.

 

But afterall it is quite cheap device. I originally purchased it for portable room acoustic measurements because it has 48V phantom output. But they've already cut software support for that product and it works only to limited extent as UAC2 class compliant device... I replaced Forte with Prism Lyra, but I'm not entirely happy with that either...

 

[emcdade]

This is only slightly more useful than when the dullards try to evaluate how someone's system sounds through a Youtube video.  Slightly....

[psjug]

On 2/12/2019 at 5:13 AM, esldude said:

 

So what are the effects of an image at 300 khz?  The amp likely won't respond much usually, the speakers won't for sure.   Our ears don't care much about phase at higher frequencies. 

 

I read that a class D amplifier can alias these high frequency artifacts back into the audible band.  Maybe @Miska or someone else can say more about this.  If it's a problem I'm surprised it is not talked about more.

[esldude]

7 hours ago, Miska said:

 

Modern ADC's will let things through up to 22.05 kHz and a little bit over as aliasing. It is property of the on-chip digital filter, if you look at the level response you can see that there's a frequency reponse boost around that area. If you upsample 44.1k content to 192k that goes away. Although related to this the problem is not so much what the ADC originally has let through, but mastering in digital domain pushing levels to clipping and then that triggering such behaviors in clipped transients.

 

 

Yes, I mentioned that. It is from the internal switching PSU (likely some DC-DC converter), since it is independent whether the device is wall-wart powered or not. And the frequency matches such. Quite typical, I'm less worried about that than the mess it leaves at audio band.

 

 

Yeah, it is somewhat messy. And running it at 192k doesn't fix that either.

 

But afterall it is quite cheap device. I originally purchased it for portable room acoustic measurements because it has 48V phantom output. But they've already cut software support for that product and it works only to limited extent as UAC2 class compliant device... I replaced Forte with Prism Lyra, but I'm not entirely happy with that either...

 

Yes, I noticed the frequency bump in the upper few hundred hertz.  Quite a bump too like 3 or 4 db.   And that even at 192 khz the interaction with the 31 khz tone is still there. 

 

I liked it mainly because it has some nice quiet high gain microphone preamps.  Good when I occasionally use ribbon mics. 

 

I've been doing room acoustic measurements with a calibrated USB mic and REW.  All you need is the mic and a laptop.  The built in ADC seems to effectively be 16 bits worth, but for in room measures that isn't much of a handicap.  Have considered one of the Sound Devices Mixpre units when I want portable use of other microphones.  

[esldude]

23 minutes ago, psjug said:

I read that a class D amplifier can alias these high frequency artifacts back into the audible band.  Maybe @Miska or someone else can say more about this.  If it's a problem I'm surprised it is not talked about more.

I don't think the class D amp would do this.  It wouldn't be aliasing.  Most now have a switching frequency of around 400 khz.  They have filters on the output to filter most of this out prior to reaching the speaker.  They'll have good response to 40-50 khz usually.  Some are on up to 100 khz.  But as the switching noise is the last thing prior to the speaker, and the speaker isn't going to respond to it you shouldn't see anything from it.

[psjug]

23 minutes ago, esldude said:

I don't think the class D amp would do this.  It wouldn't be aliasing.  Most now have a switching frequency of around 400 khz.  They have filters on the output to filter most of this out prior to reaching the speaker.  They'll have good response to 40-50 khz usually.  Some are on up to 100 khz.  But as the switching noise is the last thing prior to the speaker, and the speaker isn't going to respond to it you shouldn't see anything from it.

FWIW here is a discussion of the issue with ncore

https://www.diyaudio.com/forums/class-d/321632-hypex-ncore-nc400-input-anti-alias-filter.html

[Miska]

It can disturb the modulation scheme in analog class-D. But nowadays it is better to use direct-digital amplifiers where the modulator is software generating 1-bit PWM (aka DSD) stream driving directly the power switching stages. Essentially the amplifier becomes "power DAC" and you don't anymore have analog stage between DAC and power amp. (for example we did that with Estelon in the Estelon Lynx speaker)

 

It was explained in more detail here:

https://www.youtube.com/watch?v=6gdbdgKWJ2s

 

I think most amplifiers (speaker and headphones) used in mobile phones and tablets these days are similar, for efficiency reasons. Also microphones used there are digital MEMS type that output 1-bit stream.

 

[fas42]

5 hours ago, emcdade said:

This is only slightly more useful than when the dullards try to evaluate how someone's system sounds through a Youtube video.  Slightly....

 

People don't appreciate how powerful lower resolution captures are for revealing obvious problems - you don't listen to YouTube to find out how good it sounds; rather, it's how bad it sounds! If there are obvious deficiencies in the SQ heard over a 'poor' communications link, how much worse will these sound in the flesh?!

 

People believe that there is some magic happening when you listen to a system in the 'right place', that this will solve its clear issues ... ummm, crap!! If the sound is audibly wrong, then it's wrong - simple as that. Listening sitting with your head between two metal clamps, in a 'perfect' room, is a fudge to get your brain to ignore the obvious ...

[bibo01]

6 hours ago, Miska said:

It can disturb the modulation scheme in analog class-D. But nowadays it is better to use direct-digital amplifiers where the modulator is software generating 1-bit PWM (aka DSD) stream driving directly the power switching stages. Essentially the amplifier becomes "power DAC" and you don't anymore have analog stage between DAC and power amp. (for example we did that with Estelon in the Estelon Lynx speaker)

 

It was explained in more detail here:

https://www.youtube.com/watch?v=6gdbdgKWJ2s

Why do you think this technology is not more widely implemented in hi-fi systems?

[esldude]

On 2/12/2019 at 4:25 PM, Miska said:

 

 

It also has some aliasing and funky behavior with full levels signals near Nyquist. The switching PSU makes a faint tone just above 30 kHz.

 

 

 

I was looking at this some more.  I'd said if you reduce levels on the sweep by 1 db it cleans up.  Actually even reducing it .1 db from full max makes all the nasty behaviour above disappear.  So it would be clean other than when max level occurs between 21,750 and 22,050 hz.  That is not going to be very common.  Yes, the Forte would  be better designed not to act that way. 

[Miska]

12 hours ago, bibo01 said:

Why do you think this technology is not more widely implemented in hi-fi systems?

 

There are some systems that work like that, Mola-Mola and NAD for example. I don't know exact reasons, maybe because it requires some development effort that spans from DSP domain to electronics, not a small, cheap or easy task. And quite different from traditional designs for fairly conservative market.

 

[Miska]

6 hours ago, esldude said:

I was looking at this some more.  I'd said if you reduce levels on the sweep by 1 db it cleans up.  Actually even reducing it .1 db from full max makes all the nasty behaviour above disappear.  So it would be clean other than when max level occurs between 21,750 and 22,050 hz.  That is not going to be very common.  Yes, the Forte would  be better designed not to act that way. 

 

It is not that device's biggest problem. I would be more worried about the amount semi-correlated noise it puts out in the MHz range. It needs better analog filter... I bet it uses one of those DAC chips that is advertised to have "full integrated low-pass filter". Old CS4328 was like that and it also puts out peak at 2.8 MHz, but less, but OTOH it wasn't designed for hires, so it cuts much more aggressively right above 20 kHz.

 

Meridian Explorer2 also cleans up little when running at lower level than 0 dBFS. And much more so when receiving content upsampled to 192k (bypassing some of the MQA filters).

 

Overall, I'd say it is bad idea to run any DAC to 0 dBFS. Even if not doing upsampling, dithered volume at -3 dBFS leaves some headroom for the DAC's DSP especially when you run in some digitally clipped content (common for modern RedBook). Some DACs like Benchmark DAC3 enforce this on you at DAC side. For this purpose, top of the line Wolfson DAC chips have configuration option for -2 dBFS digital pre-pad, but IIRC it is disabled by default. But I have some material where even -3 dBFS is not enough...

 

[audiobomber]

3 hours ago, Miska said:

Overall, I'd say it is bad idea to run any DAC to 0 dBFS. Even if not doing upsampling, dithered volume at -3 dBFS leaves some headroom for the DAC's DSP especially when you run in some digitally clipped content (common for modern RedBook).

I'm not sure I understand. I use a Korg DAC with Audiogate4 in my headphone system, upsampling to 5.6MHz. The Audiogate software has a level control slider. Should I set the slider for -3dB output?

[jabbr]

16 hours ago, bibo01 said:

Why do you think this technology is not more widely implemented in hi-fi systems?

It’s the future. 

 

At 45 MHz you can build a wireless transmitter and then the DAC/amplifier/speaker can broadcast your music across the whole town 😂

 

The only problem is that some aircraft communications systems use that frequency band, otherwise ... nothing a little strategically placed mu-metal can’t fix  

[Miska]

18 minutes ago, audiobomber said:

I'm not sure I understand. I use a Korg DAC with Audiogate4 in my headphone system, upsampling to 5.6MHz. The Audiogate software has a level control slider. Should I set the slider for -3dB output?

 

At least it doesn't harm. Many DACs give lowest THD+N at -10 dBFS. You probably still don't have analog volume control turned up to max. So it is attenuating the signal to be amplified again by the headphone amp stage. You get better quality when you don't try to push the DAC to the max just to get attenuated more at the following volume control stage... So a form of "gain matching".

 

[Miska]

6 minutes ago, jabbr said:

At 45 MHz you can build a wireless transmitter and then the DAC/amplifier/speaker can broadcast your music across the whole town 😂

 

Of course you have output filter like with analog class-D too. Higher the frequency, easier it is to remove it with simple filter without effects near audio band...