Choosing a DAC for DSD playback?

[Music1969]

Talking about PCM upsampling I don' t think any kind of upsampling can be usefull.

What DAC are you using?

If it uses an ESS/AKM/TI (examples) DAC chip, there’s a good chance it resamples incoming PCM to a higher sample rate... I assume they do this because it is useful... and it’s not to add more information...

There are some that believe doing some DSP before the DAC can be useful...

 

[diegooo1972]

A scarlett 2i4 that use akm 4382et.
From what I learnt here almost every chip nowadays do oversampling to help reconstruction filter, avoid phase problems and rising signal to noise ratio.
I see what you mean. Maybe i can't hear differences because my scarlett already do oversampling.
That's true but Nyquist confirm that I can't hear that differences anyway.

 

[Music1969]

A scarlett 2i4 that use akm 4382et.
From what I learnt here almost every chip nowadays do oversampling to help reconstruction filter, avoid phase problems and rising signal to noise ratio.
I see what you mean. Maybe i can't hear differences because my scarlett already do oversampling.
That's true but Nyquist confirm that I can't hear that differences anyway.

Noted. I don’t believe anyone here suggested upsampling before the DAC creates new musical information. I don’t believe anyone here (this thread) is against Nyquist...

 

[diegooo1972]

It's only that I read that people do oversampling while from what I understand it's useless.
Just wondering why and see If I understood correctly. You can never know if you miss something while you learn.

 

[Music1969]

It's only that I read that people do oversampling while from what I understand it's useless.

I won't comment on audibility, but here are some objective measurements, up-sampling before an RME ADI-2...

https://audiophilestyle.com/forums/...-rme-adi-2-dac/?do=findComment&comment=926151

 

[Music1969]

Somebody will then say these images are well above 20kHz...

https://audiophilestyle.com/forums/...s-use-nos-dacs/?do=findComment&comment=800099

"Because those high frequency high level directly correlated components easily generate intermodulation products in the audio band, down to 0 Hz..."

 

[Music1969]

Even with PCM up-sampling, Archimago's measurements of his RME ADI-2 Pro... 16/44.1 vs 24/384 shows improvements...

So I'm, not sure about 'useless'. Whether you/can hear differences, who knows. There's objective data though...

 

[Music1969]

@MC_RME have you found similar objective results as above, feeding ADI-2 DSD256 (Direct DSD Mode) and PCM384kHz (and even PCM768kHz).

Or you don't get these measured improvements yourself?

 

[diegooo1972]

Which software was used to upsampling ? Maybe the algoritm there is a little invasive ?
considering also that from 352khz to 384 there' s no linearity in conversion like from 44.1 to 48khz.
Honestly seems to me something in the conversion more then the happyness of RME with upsampled audio.
Something like the linear phase filter or impulse filters in the conversion. Jez I don't like conversions. They lead me to even more confusion
But Honestly I don't know.
I''d like to know what's going on there.

 

[Music1969]

Which software was used to upsampling ? Maybe the algoritm there is a little invasive ?

When you say invasive, you mean worse measurements with higher sample rates?

 

[diegooo1972]

No I mean waveform corrections with filters. I'm not sure all filters are transparent.
Just wondering.

 

[Eirikur]

It's only that I read that people do oversampling while from what I understand it's useless.

The only reason I can think of is reducing jitter on a synchronous physical interface like SPDIF, i.e. sending a 44.1/16bit stereo signal only needs 1.4mbits/sec, while 352.8/32bits stereo needs 16 times that (22.6mbits/sec).
Higher bandwidth allows for a more accurate clock recovery, and I think it is necessary to increase the bit depth when you oversample to avoid quantization noise. Simple example: 2*upsample a straight line through -1,0,1 and you would need -1/2 and +1/2 as values, i.e add another bit.

Interestingly, even very old DAC chips like the BurrBrown PCM58P internally use a similar mechanism: they sample an input stream of 44100Hz/16 at 176.4kHz and operate on 18 bits internally.

All of this was my motivation to buy a USB DAC, this completely removes the transport jitter problem and all upsampling is done inside the (very capable) DAC chip - simply feed it the original input data unchanged -- and then we're back at DSD: when you have DSD source material you want to feed it directly to the DAC and be confident that the DAC processes it in the proper domain.

 

[diegooo1972]

I agree with the things you say.
I prefer USB DAC too to avoid intermediate unnecessary components. A DAC that I trust. I trust less in software implementation of algorithm upsampling. I prefer to keep things as simple as possible. On scientific bases of course.
About DSD I still need time to read everything I can here on ASR. I just have rudimental knowledge. Can't consider anything atm.
I started from PCM. I'm quite new here.

 

[zym1010]

Newbie here. I'd like to know the difference in sound quality between the "DSD direct" path (no digital volume control; supported by some AKM-based DACs, such as RME ADI-2 DAC) and the normal path (with volume control); but there seems to be nobody talking about this (neither here nor other forums).

 

[Music1969]

t's only that I read that people do oversampling while from what I understand it's useless.

In addition to measurements above, some more measurements, up-sampling PCM44.1 to DSD256, with another DAC.

https://audiophilestyle.com/forums/...elta-sigma-dac/?do=findComment&comment=622406

 

[diegooo1972]

That' s interesting ofc. But i'd like to know if this is a better behaviour of the DAC using converted pcm to dsd or it's the conversion itself that correct the waveform in some way.
What I really care are the implications of PCM to DSD conversion.
Bah I don't know. I need to get into dsd a lot more. But it's quite interesting so i'm gonna have fun.

 

[Music1969]

What I really care are the implications of PCM to DSD conversion.

You can see the objective measurements above though right? One implication is improved measurements, as shown above?

Do you mean sound quality implications?

 

[diegooo1972]

No that seems to me a good things in any case at first sight.
Just curious to know what's going on there.
How can you say that this is a better behaviour of this IFI DAC with DSD or it's just the conversion that removed or corrected things in the PCM making DSD better ?
I don't like to change things but this is just my attitude.
I'm not even sure there's going to be a difference in hearing so quality sound should be safe after all.
As I said i'm not that into DSD.
But now i'm curious.
Tonight i'll try to study that.

 

[diegooo1972]

In this link he states that PCM and DSD come from same original data but where or how he get DSD from PCM ?
I think it was with a conversion right ?
Every PCM to DSD software converter can have a lot of filters to apply.
Those can change and correct things and I don't like that. If you abuse filters is going to be a DSP after all.
Don't you think ?
I'm just talking about fidelity of the sound and probably we can't really hear that anyway.

 

[Music1969]

How can you say that this is a better behaviour of this IFI DAC with DSD or it's just the conversion that removed or corrected things in the PCM making DSD better ?

In the case of the iFi DAC, when you feed the Burr Brown DSD1793 DAC chips DSD, there is no DSP done in the chip...

Same with the RME DAC above, when "DSD Direct" mode is enabled...

Some believe one of the benefits of up-sampling to DSD before these DACs come from taking all (or most) DSP out of the DAC chip.... let the DAC just do D-to-A.... whether you agree or not, the measurements above are the measurements.