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Topping D70s MQA Review (DAC)

nighteagle

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You don't have to use any attenuation. It's just there to get the 4 V output level for reviews.

I don't know much about this but isn't XLR supposed to use 4V? What will the difference be between 4v and 5v? Like a loudness function or?
 

Atanasi

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I don't know much about this but isn't XLR supposed to use 4V? What will the difference be between 4v and 5v? Like a loudness function or?
4 V XLR is common for consumer devices, but 5 V should work, especially with A90 you can adjust as you want. 5 V is that 2 dB louder.
 

nighteagle

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4 V XLR is common for consumer devices, but 5 V should work, especially with A90 you can adjust as you want. 5 V is that 2 dB louder.

What would give the best sound quality with the A90 using XLR. Using the D70s in pure dac mode or at -2dB or -1.5dB? Considering its 5V in pure dac mode.
 
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Veri

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What would give the best sound quality with the A90 using XLR. Using the D70s in pure dac mode or at -2dB or -1.5dB? Considering its 5V in pure dac mode.
I would just use -2dB, you will get exactly as Amir measured that way...
 

Mikechw

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What would give the best sound quality with the A90 using XLR. Using the D70s in pure dac mode or at -2dB or -1.5dB? Considering its 5V in pure dac mode.
I would just use 5V. A90's input sensitivity is 8.8Vrms for mid gain. 2.9Vrms for high gain..
That means if u use high gain, just dont turn the volume too high, otherwise signal will be clipped.
 
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JohnYang1997

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I would just use 5V. A90's input sensitivity is 8.8Vrms for mid gain. 2.9Vrms for low gain..
That means if u use low gain, just dont turn the volume too high, otherwise signal will be clipped.
What? 2.9V is for high gain.
 

wyup

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I was talking about the ES9068AS which is what is used in the Gustard x16 which has excellent S/PDIF jitter.
Thanks for the clarification. I would think using the incoming spdif clock would be better as it's the one with the correct timing for the samples? So does the AK4118 "sync" the onboard clock to the spdif clock, or the onboard clock operates independent of the spdif clock and the input is resampled to match the onboard clock instead of spdif clock? That's what I'm confused on. In the case of the Gustard x16, would it of been using the onboard clock or the spdif clock, if the spdif receiver is built into the DAC chip and has it's internal PLL? I'd assume it'd be using the spdif clock..

Going through the thread I find this jitter and USB/SPDIF, AKM/ESS discussion very interesting regarding performance of chips and choice of inputs. I am not an expert but I'd like to understand it better:

AP generates jitter patterns into the dac for the measurements to test its correction mechanism. USB usually performs the best, being generally better than SPDIF on ESS and AKM, but AKM SPDIF is significantly worse here and on M400/D90 (Toslink always worst) and Amir says it is so because AKM dac chips don't do resampling on synchronous inputs (coax and toslink). So only the ESS chip does jitter correction to SPDIF? What role does the AK4118 spdif receiver have, just locking to the signal clock? Isn't there any jitter correction for spdif?

I'm interested in Toslink performance since I believe many of us will be using dacs to connecto to our TVs to stream multimedia (Netflix, youtube, movies, music) to home stereo hifi setups, so spdif and jitter management is important without a better source, and I prefer stereo dac performance to AV receivers through hdmi. AKM chips don't seem to do well on toslink, except maybe RME ADI-2. For ESS, there is SMSL SU-9 but I'd like Topping and SMSL to catch up with their AKM flagships.
 

Rottmannash

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4 V XLR is common for consumer devices, but 5 V should work, especially with A90 you can adjust as you want. 5 V is that 2 dB louder.
Going through the thread I find this jitter and USB/SPDIF, AKM/ESS discussion very interesting regarding performance of chips and choice of inputs. I am not an expert but I'd like to understand it better:

AP generates jitter patterns into the dac for the measurements to test its correction mechanism. USB usually performs the best, being generally better than SPDIF on ESS and AKM, but AKM SPDIF is significantly worse here and on M400/D90 (Toslink always worst) and Amir says it is so because AKM dac chips don't do resampling on synchronous inputs (coax and toslink). So only the ESS chip does jitter correction to SPDIF? What role does the AK4118 spdif receiver have, just locking to the signal clock? Isn't there any jitter correction for spdif?

I'm interested in Toslink performance since I believe many of us will be using dacs to connecto to our TVs to stream multimedia (Netflix, youtube, movies, music) to home stereo hifi setups, so spdif and jitter management is important without a better source, and I prefer stereo dac performance to AV receivers through hdmi. AKM chips don't seem to do well on toslink, except maybe RME ADI-2. For ESS, there is SMSL SU-9 but I'd like Topping and SMSL to catch up with their AKM flagships.
I listen to my LG C9 via optical thru the Topping E30 and sounds flawless. Always.
 

samshaver

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Will there be a D70s whose price doesn't include the MQA tax?

I can’t definitely say, but the User’s Manual would seem to suggest so...
EDCD72E0-043A-40AF-AA01-9CEB428B6ED0.jpeg

However, with the fire at AKM I would wonder if they would release the “Standard Version” or not.
 
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SHENZHENAUDIO

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I can’t definitely say, but the User’s Manual would seem to suggest so...
View attachment 102367
However, with the fire at AKM I would wonder if they would release the “Standard Version” or not.
Yep, the standard version surely will be released, while the AKM chip is really rare, time is the question.
 

nighteagle

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Got the d70s DAC yesterday. Paired nicely with the A90. Its a bit bigger than the A90 but works fine when I put the amp on top of the DAC. Using balanced connection. Fixed the issue I had with noisy USB with my previous D50s paired with the A90. Set it to -1.5dB on the volume (Saw wolf got better measurements using -1.5 compared to amirm using -2.0).
 

SHENZHENAUDIO

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Got the d70s DAC yesterday. Paired nicely with the A90. Its a bit bigger than the A90 but works fine when I put the amp on top of the DAC. Using balanced connection. Fixed the issue I had with noisy USB with my previous D50s paired with the A90. Set it to -1.5dB on the volume (Saw wolf got better measurements using -1.5 compared to amirm using -2.0).
Plz picture. :D
 

wyup

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The jitter magic in the ESS DACs is in the manner in which they do clock detection and how that detection insulates the phase locked loop from jitter in the incoming data stream. They perform a high speed sampling of incoming data and determine moving averages of the inter-edge period. They do this as part of the data reading process from the Manchester encoded data. This gives them a very stable and controlled metric of the actual data rate which is used to control a DPLL.
This is a neat trick, and very different to using an ASRC for jitter isolation.

If i understand it well from this and previous posts, the ASRC is done at the input as Veri says:
I believe ASRC (Asynchronous Sample Rate Conversion) happens at the input, so it manipulates the data before feeding it to the DAC, whereas the DAC oversampling filter happens in the DAC stage, so after the input. The standard OS filtering won't help external jitter.
So can I assume that a first jitter correction is done at the USB input by the xmos receiver resampling the incoming signal (because USB chip is known to be asynchronous these days) to clock it with the dac master oscillators. And then for SPDIF input, ESS does in its turn a high speed resampling to the signal at the chip to get a good controlled metric.

Everyone knows that ESS has built in high performance ASRC. The PLL you are mentioning is extra clock circuit for convenience. You can bypass it and still run 44.1k and 48k with single oscillator.

What is the role of the extra PLL clock circuit, a way to further reduce jitter and align the clock as close to the input signal? I've read this article in wikipedia related to this, but it's a bit hard to understand:
https://en.wikipedia.org/wiki/Phase-locked_loop#Jitter_and_noise_reduction
 

Francis Vaughan

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Very briefly, as there is a lot going on.
For jitter the entire problem is to create a stable local sample clock in the face of incoming data that both defines the sample rate and has timing variations. Furthermore, we are only concerned with jitter that has energy in a range of frequencies that can affect the audio.
Ideally the DAC would provide its own sample clock and the incoming data stream would be slaved to that. In general this is just not viable. But for specialised cases, such as a USB DAC that is using the asynchronous transfer protocol this is possible. But the moment you get any more complex that a single device, you need to synchronise the local sample clocks.
A phase locked loop is simply a way of managing a local clock in some manner so that it tracks, on average, an external clock. By comparing the phase of the external and local clocks one can derive an error correction signal and use that to nudge the local clock to keep it in sync. But that nudging itself adds jitter to the local clock. The question becomes how to do the correction in such a manner that the management of the local clock is done so that jitter in that clock outside of frequencies that matter for audio. This turns out to be harder than one might hope. The obvious thing is to perform the changes to the local clock gently, taking an average of the error signal and applying that. This is another way of saying that a low pass filter is applied to the error signal. The trouble with that is that eventually the error is applied so slowly that the PLL is unable to find synchronisation with the input in the first place, or drops out of sync if the input varies too much. The overlap between audible jitter energy, a slow enough filter to remove such energy, and yet fast enough to allow the system to be able to keep sync is very delicate.
Lots of tricks and adaptive controls are possible here. Only using occasional samples from the incoming clock is another way of trying to remove unwanted energy from the jitter spectrum of in the incoming clock. The ESS mechanism is another way of finding a benign way of estimating the incoming clock and creating a local error estimate. Again one that can be applied to the local clock in a smooth manner that does not put energy into the audio in bad ways.

The other way of managing jitter is to resample the incoming data and feed that to a DAC that is running with a local sample clock. But what does this mean? In the abstract, you want a way of arbitrarily resampling the data, taking any possible incoming sample rate and converting it to a different sample rate. That is near black magic. In the extreme, you feed it to DAC, feed that to an ADC, and you can resample anything. But it misses the point. If you have two sample rates that are simple ratios of one another sample rate conversion is easy. For instance converting 48kHz to 32kHz was done to provide digital feeds to FM radio stations (that are bandwidth limited to 15kHz anyway.) This was one of the reasons for 48kHz being adopted as a pro standard. Arbitrary sample rate conversion came a lot later. Conversion from 441.kHz to 48kHz was for a while simply not possible. Next, doing so where the two sample rates could vary relative to one another (the asynchronous part) came later still.

Ideally, with an ASRC there is an incoming stream with its wobbly data clock, and a different local clock. Bingo, you can have your local clock, and don't need to slave from the external clock. But that ignores a lot of grief inside the ASRC. ASRCs don't work well if the two sample clocks are close to one another in frequency. Which is why you often see internal clocks of non standard sample rates used, and samples converted both up and down in frequency to meet the one internal rate. They are not magic, and still need to be fed reasonably stable clocks.

In general a ASRC is not the same as the oversampling mechanism in the DAC. The nature of things is such that you could conceivably optimise the operation of a system so that there was some commonality, but oversampling has a very different reason for existing and is independent of an ASRC. You could trivially use an ASRC with a non-oversampling DAC. Delta sigma DACs oversample intrinsically as part of their operation in a manner quite independent of any external sample rate conversion.

So can I assume that a first jitter correction is done at the USB input by the xmos receiver resampling the incoming signal (because USB chip is known to be asynchronous these days) to clock it with the dac master oscillators. And then for SPDIF input, ESS does in its turn a high speed resampling to the signal at the chip to get a good controlled metric.

This is confusing uses of the the word "asynchronous" in lots of different contexts. Asysnchonous simply means without synchronisation. Simialrly "sampling" is being confused. All references to resampling in the this context refers to changing the sample rate of the audio. Sampling however can mean sampling anything, and here, usually the clocks. Nothing to do with the audio. Asynchronous USB simply means that the USB receiver is not synchronised to the USB source. It says nothing about the payload. A USB receiver will not resample the audio.
The ESS jitter control does not resample either. It samples the clock to get a metric of the clock rate, but that refers to the data clock, and not the audio samples. There may be a ASRC present, but that is quite separate. Oversampling is separate again.
 

direstraitsfan98

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does anybody have an issue with their d70 where occasionally, while opening and closing programs (happens when I close mpc media player usually after fininshing a video or movie) something will mess up, windows 10 wont recognize the dac anymore and you'll need to turn the dac off and on to reset?
 

SHENZHENAUDIO

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does anybody have an issue with their d70 where occasionally, while opening and closing programs (happens when I close mpc media player usually after fininshing a video or movie) something will mess up, windows 10 wont recognize the dac anymore and you'll need to turn the dac off and on to reset?
D70 or D70s?
 
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