Topping D50
Jan 27, 2019 at 4:30 PM Post #602 of 1,054

wyup

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Hello. Amazing thread and really interesting contributions.
I am between the D50 and a Pro-Ject Pre Box S2 Digital, which has its own thread here, it is around 150€ more expensive and has a headphone amp. But I haven't heard anyone comparing both. The D50 is now 200€ on Shenzenaudio and is a good value, but seeing some people here say it is U-shaped and harsh or fatiguing makes me wary. So far everybody says the Pro-ject is not fatiguing. Yes, it's more expensive but I thought they be comparable. I plan to connect optical or coaxial from a blu-ray to powered speakers in an auditorium.

Has anyone heard both?
 
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Jan 27, 2019 at 10:44 PM Post #603 of 1,054

Fawzay

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me too, i hope we will have some good news soon

it expected to be announced on chinese new year

oh great! hopefully they release the black edition then im done after that heading to electrostats iem :)
 
Jan 28, 2019 at 9:52 AM Post #604 of 1,054

nihues

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What about D50 vs Katana Dac? I have D50 and like the sound, but still using original usb cable and a gamer htpc. I know Katana is a player/dac, but I want to remove my HTPC from the audio source (and usb cables), so it seems the best to do is to buy the allo Katana or at least USBBridge. I've settled on katana because it comes almost plug and play, can go to rpi + hat too, but katana seems better overall.

I was wondering on buying a better usb cable + usbbrige, but the costs are the same or higher, and I can sell d50, so it would cost less to upgrade to Katana instead of buying usb stuff.
After the upgrade, I'll buy a LPS to complete my audio setup.
 
Jan 30, 2019 at 8:03 AM Post #606 of 1,054

439598

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The final output OPA1612 on the D50 is a simple 2x gain stage, there are RC low pass filters on the input and the output of this op amp and im not sure what they are for exactly.
bypassing the input of this op amp completely including the LPFs and going straight to a headphone amp sounds really good, probably not as good switching to transformer or discrete I/V stage but at least a very easy way to cut the sound influence of the op amps in half.
do you need this gain though? to put in perspective Im using a unity gain headphone amp and even with the least senstive headphone, he560, ''0db'' on the D50 is still well above any safe listening level.
any ideas how this would effect SNR and THD? less gain would mean higher SNR, but you are pushing the DAC chip closer to its limit so maybe THD is worse... whatever the case it sounds fantastic. Another small benefit is less digital attenuation is needed, which with 32 bits should be perfect in theory but isnt.

An important reason not to do this:
the power on/off pops are much louder now. if you are using low-medium sensitvity headphones or your amp has a lot of gain they would be dangerously loud. The D50s useless relay might block these pops, if it does it would become an extremely useful relay... but then you have to question why they added 2 low pass filters to the signal path to dampen these very occasional pops that are already being muted by the relay.

could someone confirm if the D50 is completely silent when turning on and off?

Edit: im just learning there is 2 types of filtering that take place in a delta sigma DAC, and this is one of those filters and probably a terrible idea to remove it.
considering it actually sounded better , I wonder what the problem is with this ultrasonic noise, is it possible to cause hearing damage? I know at least it will mess with the performance of an amp.
 
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Jan 30, 2019 at 8:37 AM Post #607 of 1,054

TheSnafu

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Hello. Amazing thread and really interesting contributions.
I am between the D50 and a Pro-Ject Pre Box S2 Digital, which has its own thread here, it is around 150€ more expensive and has a headphone amp. But I haven't heard anyone comparing both. The D50 is now 200€ on Shenzenaudio and is a good value, but seeing some people here say it is U-shaped and harsh or fatiguing makes me wary. So far everybody says the Pro-ject is not fatiguing. Yes, it's more expensive but I thought they be comparable. I plan to connect optical or coaxial from a blu-ray to powered speakers in an auditorium.

Has anyone heard both?

I have heard both and, at least in my system, D50 wins hands down. S2 sound was (imo) soft and rolled of from tops and lows, not sure what measurements say but that's my memory (didn't do H2H testing).
And about that harsh or "U-shape", i think people having that opinion have other gear that gives that "U-shape" or harsh sound. D50 is very balanced dac with good highs and lows.

My D50's optical connection isn't very tight and if you pull cable it disconnects easily. RCA/USB/spdif are solid. Also, D50 is so light heavier/stiffer cables move it around...

EDIT: ASR tested both of these asked units and they measured both very nicely. At the time i tested S2 vs ME2 and had different tubes sets so take this "wins hands down" with a pinch of salt. I'm sure both are mighty fine DACs.
 
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Jan 30, 2019 at 12:01 PM Post #608 of 1,054

439598

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Or maybe the filtering is actually done at the first op amps after all!
this DAC analogue stage schematic is the closest match I can find to the D50, the component count is the same and the resistor values are similar/the same and the traces matches up 1:1 (of whats visible at least). but whats really important is hows it labeled, the first stage is being called the filter, and the second stage is an ''EQ'' ??
http://www.customanalogue.com/diy/REC/Current_Std_IV_Opamp_REC-blow.gif

it could be mislabeled, I dont know enough about electronics to be sure whats actually going on here.
it does make sense for the filtering to happen on the output of the DAC chip, as the noise would affect the performance of the first op amps, I always thought all filtering happens inside the DAC chips.

There was nothing online regarding how an unfiltered Delta Sigma dac might sound ... but for it to give the impression of being plain better, as in a drop in distortion with no real change or strangeness to the sound , doesnt seem right.

and its a differential signal, but only the non-inverting side is connected to my single ended amp, again not sure what this does but it shouldnt be good.
 
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Jan 30, 2019 at 2:07 PM Post #609 of 1,054

bogi

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I wonder what the problem is with this ultrasonic noise, is it possible to cause hearing damage? I know at least it will mess with the performance of an amp.

I think it doesn't have impact to our hearing, but could have impact on some amps, like overheating or maybe possible oscillations in some designs. But it can have also impact to audible frequency range by means of intermodulation distortion. For example two peaks in inaudible frequency range, which differ in 7kHz (for example 50 and 57 kHz) generate intermodulation distortion at frequency 7kHz and that's audible if level of that distortion is sufficient.

Look at these pictures: https://audiophilestyle.com/blogs/entry/428-ifi-idsd-micro-measurements/
Author of HQPlayer measured here iFi micro iDSD DAC frequency response up to 5MHz, when sweep 0 - 22.05 kHz was used as input signal. The 1st picture show filterless DAC output and the next 2 pictures show influence of "standard" and "minimum phase" digital filter types - they are provided as option by the iFi DAC. These filters are used yet in digital domain as part of PCM input oversampling - yet before the delta sigma modulator circuit itself. They lower the unwanted aliasing effect of oversampling. Then the next pictures show how the high frequency noise is affected when that 0 - 22.05 kHz sweep is at first upsampled by computer software to 768kHz, DSD256 or DSD512 and then sent to DAC digital input. In such a case the DAC chip has "less work" to oversample the input signal up to MHz range sample rates, which are required on delta sigma modulator input.

The first upsampling (or oversampling) stages have the most significant impact on sound quality and powerful computer can do that job in much higher quality than resource constrained DAC chip. That's the reason why upsampling players like HQPlayer exist. In a special case, when DSD signal is provided on input of for example Burr Brown TI DAC chips, even the delta sigma modulator of DAC chip is skipped and only the DSD signal conversion to differential analog signal and low pass filtering is used. That's the technical reason why DSD can sound better than PCM with delta sigma DACs - also when DSD input is result of PCM source file software conversion.

Now look at page 32 of the TI Burr Brown DSD1793 DAC chip manual: http://www.ti.com.cn/cn/lit/ds/sles075b/sles075b.pdf
You can see that the DAC chip does not provide suitable analog output signal for audio applications. It requires analog LPF stage, some example implementation appears on the picture. Each filter implementation is imperfect, because each filter has imperfect amplitude and phase response so it causes some level of distortion. Therefore less filtered output may sound to us better - if it doesn't cause audible distortion and unwanted effects downstream.

Delta sigma modulator circuits convert PCM input to 1 bit or multi bit delta sigma signal output. The delta sigma signal is of PDM type (pulse density modulation). So information is not coded by amplitude but by phase differences. DSD is 1bit two level delta sigma signal, sample rate is in MHz range. Delta sigma signal (including DSD signal) doesn't require anything more than analog low pass filtering to convert it to analog signal.

The nature of DSD signal is very simple and can be explained for example on a very quickly blinking LED lamp. Imagine a LED lamp which is able only of full light or no light and nothing between. The lamp provides 2 level high frequency pulse signal - a 'pulse' or 'no pulse' can be repeated in any amount and order. Imagine it is blinking (or not blinking) very very very quickly so our eyes are unable to distinguish individual pulses. Our eyes provide low pass filtering in such a case. The LED lamp outputs some number of pulses within some very short time interval. For example 61% of that time interval the signal level was "on" and 39% of time it was "off". It results to perceived light intensity of 61% between the minimum "off" level and maximum "on" level. This is the principle how any analog signal can be coded into 1 bit PDM signal, the only requirement is enough high pulse sample rate in relation to frequency spectrum of the analog signal.
 
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Jan 31, 2019 at 12:28 AM Post #610 of 1,054

439598

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I think it doesn't have impact to our hearing, but could have impact on some amps, like overheating or maybe possible oscillations in some designs. But it can have also impact to audible frequency range by means of intermodulation distortion. For example two peaks in inaudible frequency range, which differ in 7kHz (for example 50 and 57 kHz) generate intermodulation distortion at frequency 7kHz and that's audible if level of that distortion is sufficient.

Look at these pictures: https://audiophilestyle.com/blogs/entry/428-ifi-idsd-micro-measurements/
Author of HQPlayer measured here iFi micro iDSD DAC frequency response up to 5MHz, when sweep 0 - 22.05 kHz was used as input signal. The 1st picture show filterless DAC output and the next 2 pictures show influence of "standard" and "minimum phase" digital filter types - they are provided as option by the iFi DAC. These filters are used yet in digital domain as part of PCM input oversampling - yet before the delta sigma modulator circuit itself. They lower the unwanted aliasing effect of oversampling. Then the next pictures show how the high frequency noise is affected when that 0 - 22.05 kHz sweep is at first upsampled by computer software to 768kHz, DSD256 or DSD512 and then sent to DAC digital input. In such a case the DAC chip has "less work" to oversample the input signal up to MHz range sample rates, which are required on delta sigma modulator input.

The first upsampling (or oversampling) stages have the most significant impact on sound quality and powerful computer can do that job in much higher quality than resource constrained DAC chip. That's the reason why upsampling players like HQPlayer exist. In a special case, when DSD signal is provided on input of for example Burr Brown TI DAC chips, even the delta sigma modulator of DAC chip is skipped and only the DSD signal conversion to differential analog signal and low pass filtering is used. That's the technical reason why DSD can sound better than PCM with delta sigma DACs - also when DSD input is result of PCM source file software conversion.

Now look at page 32 of the TI Burr Brown DSD1793 DAC chip manual: http://www.ti.com.cn/cn/lit/ds/sles075b/sles075b.pdf
You can see that the DAC chip does not provide suitable analog output signal for audio applications. It requires analog LPF stage, some example implementation appears on the picture. Each filter implementation is imperfect, because each filter has imperfect amplitude and phase response so it causes some level of distortion. Therefore less filtered output may sound to us better - if it doesn't cause audible distortion and unwanted effects downstream.

Delta sigma modulator circuits convert PCM input to 1 bit or multi bit delta sigma signal output. The delta sigma signal is of PDM type (pulse density modulation). So information is not coded by amplitude but by phase differences. DSD is 1bit two level delta sigma signal, sample rate is in MHz range. Delta sigma signal (including DSD signal) doesn't require anything more than analog low pass filtering to convert it to analog signal.

The nature of DSD signal is very simple and can be explained for example on a very quickly blinking LED lamp. Imagine a LED lamp which is able only of full light or no light and nothing between. The lamp provides 2 level high frequency pulse signal - a 'pulse' or 'no pulse' can be repeated in any amount and order. Imagine it is blinking (or not blinking) very very very quickly so our eyes are unable to distinguish individual pulses. Our eyes provide low pass filtering in such a case. The LED lamp outputs some number of pulses within some very short time interval. For example 61% of that time interval the signal level was "on" and 39% of time it was "off". It results to perceived light intensity of 61% between the minimum "off" level and maximum "on" level. This is the principle how any analog signal can be coded into 1 bit PDM signal, the only requirement is enough high pulse sample rate in relation to frequency spectrum of the analog signal.
this is a really excellent explanation, especially DSD parts. very easy to understand
 
Feb 1, 2019 at 9:18 AM Post #612 of 1,054

carlosgab

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Recently I bought this USB digital cable (SK-M-b HIFI 5N OFC otg micro usb to USB Type B FOR DAC etc, SK-M-b (1.5M/4.92ft) https://www.amazon.com/dp/B01N39YYPL/ref=cm_sw_r_cp_apa_i_aofvCbD2TXX33) to connect my Onkyo DAP>D50>Amp but even this cable or my generic USB cable I still hear a noise in the background which I dont hear via COAX from my FIIO DAP. Is this the noise that is dealt with the likes of IFI Ipurifier? I would like to know because that small device now costs like USD120. Anyone having the same problem?
 
Feb 1, 2019 at 11:58 AM Post #613 of 1,054

439598

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Here is the actual configuration of the D50 op amps, these are not accurate component value, the D50 resistor values are different and the capacitor values arent known, so its not possible to work out filter characteristics, but the general working of the op amps is understood.
Untitled.png
 
Feb 1, 2019 at 1:19 PM Post #615 of 1,054

Ripper2860

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I am quite pleased with my D50 and it is now my preferred headphone rig DAC. Added a Jameco 5v linear PSU and a custom USB cable with 5v+ and GND disabled. Been running this setup for a few days and it sounds great, with no DAC sync issues. The sound profile is wonderful!!
 
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