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Michael_Grace
432
Grace Design
Sep 13, 2018
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Hi All, Following is a bit of info on the new SDAC balanced. There is some description of the new features and circuitry, design intent, and preliminary measurement data. The SDAC Balanced DAC is based on the digital to analog converter design in the SDAC. See my post on the SDAC which outlines the design philosophy and technical details and is all relevant to the SDAC Balanced: https://www.massdrop.com/buy/massdrop-x-grace-design-standard-dac/talk/1806625 (Please note my opinion on the role of the actual DAC chip in overall converter sound as I think there is much hype and judgment about which chip a product might use and how it will sound because of that choice.) Here I will discuss the addition of the SPDIF input, balanced analog outputs, and USB Audio Class 2 operation. Here I will discuss the addition of the SPDIF input, balanced analog outputs, and USB Audio Class 2 operation.
SPDIF Unlike asynchronous USB audio, a SPDIF (or toslink or AES3) signal contains an embedded clock that must be recovered by a receiver circuit. The quality of that recovered clock can vary widely depending on many factors. The design of the PLL (phase locked loop) used to synchronize the internal clocks for the d-a converter is critical to achieving high fidelity sound reproduction. The SPDIF receiver in the SDAC balanced is the AK4117 which is a solid performing hardware receiver circuit. The intrinsic jitter of the AK4117 PLL is not bad but since it is designed to acquire lock rapidly the PLL loop filter will not allow it to reject jitter in the audio band. For this reason we use a two stage clocking system with a second PLL that re-clocks the recovered clock from the AKM receiver. The second stage PLL is an extremely quiet digitally synthesized clock that has a loop bandwidth of 1Hz. This means that any incoming jitter on the SPDIF line that is above 1Hz will be attenuated dramatically to where it will not be able to produce any audible artifacts in the audio. To measure the effects of jitter on a DAC we send a relatively high frequency tone through the DAC and look for jitter modulated side bands in an FFT plot. I will upload a pictures of this that shows the output of the SDAC balanced reproducing an 11.025kHz tone with and without 1UI (177nS) of sinusoidal jitter. The SPDIF input is transformer-coupled for complete ground isolation. Note that the SPDIF interface maximum sample rate is 192kHz BALANCED LINE OUTPUTS The AK4452 DAC uses switched capacitor output filters (following the delta sigma modulator) which provide an inherently balanced signal at the output pins. From here, the balanced signal is sent to a fully differential output filter/driver amplifier. This amplifier has differential inputs and balanced outputs. Specifically this circuit is a fully differential multiple feedback 2nd order low pass filter with a Bessel response and a corner frequency of 100kHz. The fact that it is fully differential guarantees that the positive and negative phases leave the box with almost perfect amplitude matching (around +/-0.002dB). The Bessel response is chosen for the best phase response. The balanced outputs have a 300 Ohm output impedance (150 Ohms per leg) and produce exactly 2x the output voltage of the unbalance outputs at 4.34Vrms in to a 100k Ohm load. The unbalanced line outputs on the 3.5mm stereo jack operate simultaneously to the balanced outputs and put out 2.17Vrms in to a 100k Ohm load. All inputs and outputs feature robust ESD (electrostatic discharge) production. A note on balanced vs. unbalanced signals: First of all, there is no inherent difference in fidelity between a balanced and an unbalanced signal. Under ideal conditions both signals can carry identical information. It is the factors of the real world where there we can differentiate (pardon the term!) between the two. As many probably already know, balanced line signal transmission with a differential receiver can reject common mode noise. The amount of rejection is known as the common mode rejection ratio (or CMRR). An unbalanced interface is prone to noise caused by ground potential differences between a DAC and a downstream device. This can be caused, for instance, by a setup where a computer is connected to a DAC via USB, the DAC is then connected to an amplifier, and that amplifier is connected to earth ground. If the computer is also connected to earth ground then you have a ground "loop" and current will flow through the audio cable ground between the DAC and amp causing noise. A balanced interface can reject that noise but the design of the balanced driver and receiver circuitry can cause degradation to the signal in the form of distortion and amplifier noise. So you see there is no black and white way to evaluate whether balanced is better or worse in a given setup. Ideally the DAC would have a galvanically isolated (ground isolated) USB interface which would break the "loop". Galvanically isolated USB is expensive and unfortunately not in the budget for the SDAC. So in this case, if there is going to be a potential for a ground loop, a well designed balanced interface can really help. The CMRR of the receiving device will determine how well it will reject noise. USB Audio Class II For those with high resolution files the SDAC Balanced will play PCM sample rates up to 384kHz and DSD 256 This is driver-less operation on Mac OSX and current versions of Windows 10. For older versions of Windows USB I mode is required and then sample rates will be limited to 96kHz. Here are some performance plots and measurements from our pilot build of SDAC Balanced DACs. These figures are actual measurements. Guaranteed performance specifications will be more conservative.
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Frequency response @ 96kHz Fs: DC-20kHz +/-0.03dB Frequency response @ 129kHz Fs: DC-48kHz +/-0.4dB All distortion measurements are with a 22-22kHz bandwidth THD+N% @ -0.15dBFS 44.1kHz 20Hz 0.0016 100Hz 0.0011 1KHz 0.0007 10KHz 0.0008
THD+N% @ -0.15dBFS 48kHz 20Hz 0.0018 100Hz 0.0012 1KHz 0.0008 10KHz 0.0007
THD+N% @ -0.15dBFS 88.2kHz 20Hz 0.0016 100Hz 0.0011 1KHz 0.0007 10KHz 0.0006
THD+N% @ -0.15dBFS 96kHz 20Hz 0.0018 100Hz 0.0012 1KHz 0.0008
10KHz 0.0006 THD+N% @ -0.15dBFS 176.4kHz 20Hz 0.0016 100Hz 0.0011 1KHz 0.0007 10KHz 0.0006
THD+N% @ -0.15dBFS 192kHz 20Hz 0.0018 100Hz 0.0012 1KHz 0.0008 10KHz 0.0006 Dynamic Range (A-Weighted) 115.4dB Dynamic Range (Un-Weighted) 112.3dB Crosstalk @ 1kHz, -10dBFS -120dB Crosstalk @ 10kHz, -10dBFS -112dB IMD CCIF, -6.03 dBFS, 19/20kHz, 24/96k 0.00015% IMD SMPTE -2 dBFS, 24/96k 0.0012% IMD SMPTE -6 dBFS, 24/96k 0.0008% Linearity @ -90dBFS +/-0.025dB Maximum output unbalanced 2.17V Maximum output balanced 4.34V
Sep 13, 2018
pospos
70
Sep 13, 2018
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Michael_GraceHello Michael,
Thank you for these measurements!
I have a few questions regarding the spdif input if you don't mind: - My source has 110 ohms AES outputs with a 3.3V P-P voltage (or maybe a bit more). As this unit is transformer coupled I guess I can get away with a simple impedance matching voltage divider. What is your take on this? What is the maximum P-P voltage the spdif input can handle?
- Why did you choose to recover the spdif clock instead of using an ASRC to some high internal frequency like many other designs do nowadays (benchmark for example, or what ESS embeds in their DAC chips) ?
- I plan on using several DACs in a multiway active system, all fed from the same source and clock (miniDSP UDIO8, with 4 AES3 outputs). What kind of synchronization can I expect between different DACs in this situation? What will the maximum delay difference be? Is there a risk of a drift between units over time? In this situation I like the ASRC approach as it guaranties no drift over time and a synchronization that will stay within a sample of the target internal sampling frequency of the DAC. What about your approach in this regard?
Thank you :)
Sep 13, 2018
pospos
70
Sep 13, 2018
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Michael_Grace Another question :) Is the unit pop-free on turnon, turnoff, sampling frequency change over spdif, spif signal loss, corrupted spdif, etc. ?
Sep 13, 2018
Michael_Grace
432
Grace Design
Sep 13, 2018
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pospos Hi pospos, Feeding the SDAC balanced with AES at 3.5Vpp should not be a problem. The ideal method would be to build a circuit like in this article: https://www.rane.com/note149.html These are difficult to build in to normal connectors though. Another option is to use the circuit below.

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The 34 Ohm resistor at the input adds to the internal 75 Ohm resistor to make about 110 Ohms so the impedance is matched. I tested this with 120' of Belden 1800B. The scope traces below show the signal at the input to the SPDIF receiver to have excellent signal integrity. You could likely go 1000' with UTP CAT5e cable at base rate and probably about 500' at 192kHz.

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The SPDIF input is very well behaved with no pops or clicks from sample rate changes, loss of lock, invalid data etc. The SDAC balanced produces as soft click at about -40dB when powering on and off. For me this is unobjectionable Our decision to use a hardware spdif receiver rather than an ASRC is two fold. While ASRCs have gotten much better over the past few years we still believe that if you can recover the embedded clock and keep jitter to well below the dynamic range of the converter then it is always better to maintain bit accurate transmission. In the case of the SDAC we are able to fully maintain the performance of the DAC so there would be not performance increase, only possible degradation due to the conversion process. Also, while ASRC chips and DACs with SPDIF/ASRC inputs are easy to use and relieve the designer of much of the careful clock design tasks the good ones are expensive and would be outside of the budget for this DAC. In the multi-channel system that you describe the synchronous SPDIF receiver will be what you want. If all of the AES3 outputs on your miniDSP board are frame aligned then all of the DACs will also have perfect sample frame alignment. The delay between SDACs will be dominated by propagation time differences in your wiring. This will probably be in the tens of nano seconds range, so negligible. This would not be true for DACs that use ASRC since the clocks that are running the DACs are all free running and not synchronous to each other. You could potentially have up to a sample delay between DACs (22uS at 44.1kHz) Cheers, Michael
Sep 13, 2018
pospos
70
Sep 13, 2018
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Michael_GraceMichael, Thank you for that very informative response!
I like the simple AES/spdif conversion cable, I will definitely follow this path, thanks for taking the time to test this and post measurements!
Regarding the -40dB on/off click, it might potentially be a problem in my application as I am relying entirely on digital volume control, targeting around 120dB SPL/1m at 0dBFS (and 5dB(A) SPL residual noise with this particular DAC). In this scenario -40dBFS is still 80dB SPL. Not a big deal if I keep the DAC on all the time (it is less than 5W after all), as this will only happen on power failures or user errors. Still, as I will be connecting compression drivers directly I am concerned about the spectral distribution of that click. Does it have energy toward DC or is it band limited?
Sep 13, 2018
amirm
341
Sep 13, 2018
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Michael_GraceWhat excellent documentation for a DAC! Very well done. I wish all manufacturers would do the same. :)
Look forward to testing one in the future.
Amir, Founder, Audio Science Review
Sep 13, 2018
rajapruk
212
Sep 13, 2018
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Michael_GraceNice. Thanks. Are there any plans for a future Grace dac at a higher price point, with even better performance?
Sep 13, 2018
RainEr0
3
Sep 15, 2018
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Michael_GraceSo win7 can not PCM 384kHz and DSD?
Sep 15, 2018
pospos
70
Sep 21, 2018
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Michael_GraceMichael, what is the spectral distribution of that -40dB on/off click? Is there any way it could be made lower in level for applications relying on digital volume control only (eg in may case -40dBFS is 80dB SPL)
Thank you
Sep 21, 2018
rajapruk
212
Sep 21, 2018
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posposI have the same concern about the thump as pospos. I will also connect to actively driven compressiondrivers in horns.
Sep 21, 2018
Michael_Grace
432
Grace Design
Sep 21, 2018
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posposThe click is caused by the fact that the negative power supply comes up with a slight delay compared to the positive supply. There is no easy or economical way to change this in the SDAC design. However, the sound has very little LF content. It is more of a click than a pop. It should not be objectionable in a system that has appropriate gain staging. And it would definitely not be capable of any damage to your speakers!
I would recommend setting the input sensitivity of your speaker systems such that the maximum SPL you desire is achieved somewhere around 0dBFS. I am assuming you don't desire, or have the ability to create, 120dB SPL:). This would put your nominal program material at somewhere around 20dB below that and the power on click at another 20dB below that. In other words, the click will be at a level significantly lower than normal program material. I will try to post a recording of the click somewhere so you can hear what it sounds like...
Sep 21, 2018
Michael_Grace
432
Grace Design
Sep 21, 2018
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RainEr0Unfortunately not. Win 10 and OSX only for the high sample rates...
Sep 21, 2018
AndyMok
34
Sep 22, 2018
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Michael_GraceSince Galvanically isolated USB is not possible at this moment, what would be your suggestions to those who wish getting the most out-of using USB?
Sep 22, 2018
pospos
70
Sep 22, 2018
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Michael_GraceUnfortunately in my case I will need to target 120dB SPL for 0dB FS, because all the EQs are done in the digital domain. For example the constant directivity compensation (and general EQ) on my compression driver implies that the range around 3kHz is set some 14dB lower than the range past 10kHz. Here is an illustration of the EQ curve (JBL M2) : http://www.audioheritage.org/vbulletin/attachment.php?attachmentid=65945&d=1435612846 That means that my 120dB SPL target is "only" really 106dB SPL on 0dB FS peaks.
Sep 22, 2018
rajapruk
212
Sep 22, 2018
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posposIs it SPL from 1m, or in listening posposition?
Sep 22, 2018
pospos
70
Sep 22, 2018
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rajaprukOne or the other, it is the same: in the end the click will be 40dB lower than that target max SPL (with EQ margin) at a given position and a given number of loudspeakers.
Sep 22, 2018
Michael_Grace
432
Grace Design
Sep 22, 2018
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posposHere is a recording of the SDAC Balanced powering on then off. The power on pulse is at about -50dBFS while the turn off pulse is higher at -30dBFS. https://drive.google.com/open?id=1uc7VAIIvZSdPK3o5OBIuCMnNVgnqeHMp This file is 1kHz at -0.4dBFS as a level reference for the first file. https://drive.google.com/open?id=15pRvAy5FV6vDR0e7RMcHu-6FDTUjSp5D The turn off pulse could be annoying, but the power will likely be negligible as far as your diver is concerned. Of course, it is always best practice to turn power amps on last and off first when bringing a system up and down. Cheers, Michael
Sep 22, 2018
pospos
70
Sep 22, 2018
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Michael_GraceThank you Michael, very much appreciated! That level of support and transparency is pretty awesome :)
I will look that up.
Sep 22, 2018
pospos
70
Sep 22, 2018
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Michael_GraceI did an FFT of the clicks with Audacity (I don't like/trust that tool too much, but that was all I had at hand), and it looks like both on and off clicks have quite a bit of energy down low :
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Sep 22, 2018
rajapruk
212
Sep 24, 2018
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posposThis is not good for me. Too much LF. Dealbreaker for me with directly connected compression drivers :( Big thumb up for the transparency of information here from the designer. If there will be a higher priced unit from Grace later on, with no pops and state-of-the-art SNR, I would most probably buy it.
Sep 24, 2018
Michael_Grace
432
Grace Design
Sep 24, 2018
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rajaprukCan I ask what you are concerned about? At -30dB this is a power ratio of 1/1000 below full scale.
Sep 24, 2018
rajapruk
212
Sep 24, 2018
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Michael_GraceI am concerned about my JBL 476Mg and JBL 045Be compression drivers. This kind of drivers do not like LF at all. This is really really expensive stuff. Actively driven with dac and amp after digital crossover in the chain. I have protection caps on the drivers, but I do not trust those for startup or shutdown pops.
Sep 24, 2018
rajapruk
212
Sep 24, 2018
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rajaprukMy horns are about 110dB sensitivity and my current amps have 26dB gain with 150W into 16 ohms.
Sep 24, 2018
rajapruk
212
Sep 24, 2018
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rajaprukThe pop will be loud....
Sep 24, 2018
rajapruk
212
Sep 24, 2018
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rajaprukMy speakers are 4-way, so I need 4 pc of reasonable priced balanced dacs, with high SNR and no pops. Not so easy to find... This dac was close, but no cigarr.
Sep 24, 2018
Michael_Grace
432
Grace Design
Sep 24, 2018
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rajaprukHi rajapruk, I am pretty sure you do not have anything to worry about here with regard to the SCAC power off pulse causing damage. But you might want to reconsider the alignment of your system as there may be other perils lurking for your 476Mg and 045Be drivers. The SDAC Balanced output at 0dBFS is 4.2V rms on the XLR connectors. Given your amplifier gain of 26dB this will result in an amplifier output voltage of 84Vrms. The impedance of the 476Mg at 1kHz is about 16 Ohms. 84V in to a 16 ohm driver is 440W, which is 135dB SPL at a sensitivity of 110dB SPL (1W @ 1M). Your amplifier will be clipping badly at this point. Also, if your nominal listening level is in the 80dB SPL range you would be operating your digital volume controls in the neighborhood of -55dBFS and thereby throwing away 55dB of DAC dynamic range rendering your system signal to noise ratio to about 60dB (10 bits). [edit] Granted you are not listening at 1 meter from your speaker so adjust accordingly! Nevertheless, the turn on pulse would have a peak power dissipation of 440 milliwatts which will certainly not hurt your driver (especially given the duration of around 40mS), but this system has an extreme amount of gain so it would be fairly audible. Also, at this level of amplification, an accidental burst of full amplitude audio from upstream could rapidly harm your voice coil. If using the SDAC balanced I would recommend attenuating the input of your power amplifier by 20dB so the amplifier gain would be 6dB. (Or better yet, lower the gain of your amplifier so that it's noise contribution is minimized) Then you would have 8V rms to the speaker at 0dBFS which would result in a little more than 120dB SPL (4W on the voice coil). The DAC output noise scales with the amplifier attenuation so this would improve your signal to noise ratio commensurately. From here the -30dBFS the pop would be at 4mW and much less audible. What is the capacitor value of your DC blocking capacitor? It will likely attenuate the LF components of the pulse even further.
I don't know what the rated max power handling of 476Mg is but the distortion plots in the K2.S9900 White Paper (source below) suggest it can handle 7.5V rms from 200Hz to 20kHz. At 200Hz the impedance is about 10.5 Ohms so that would be 5.3W which is about 1300 times more than the peak power of the SDAC power-off pulse. As for the 045Be driver, it has an impedance of around 4.3 Ohms in its pass-band. A system aligned for 0dBFS = 120dB SPL would result in almost 15W on that driver. Now it is super unlikely that music would ever contain enough power above 15kHz to actually cause damage, but an accidental test tone at full scale would. The -30dBFS pulse would, for about 40mS, dissipate 19 milliwatts. Again, the series capacitor will reduce this depending on its value. This discussion so far has used the peak value of the pulse in the full audio bandwidth. If you want to just consider the LF content of that pulse try this: Run the pulse through a 24dB/octave low pass filter at 200Hz. The peak value of this result is around -46dBFS. This means that at below 200Hz the pulse would dissipate 1/20,000th the power of full scale. This is 200 micro watts on the 476Mg and 1 milliwatt on the 045Be. Of course it is always best practice to turn on your power amplifier last when power up your system and turn it off first when powering your system down. Furman makes some inexpensive power sequencers that can automate this.
Hopefully the upstream playback and signal processing software is robust. In my opinion that is what poses the greatest risk to your drivers. Best, Michael PS. I found the white paper at: www.keithhaddock.com/assets/_managed/products/files/K2S9900WhitePaper1-6-09.pdf The 045Be impedance came from this chart : http://www.audioheritage.org/images/projectmay/driver-data/thumbs/1500al-435be-045be-imp.jpg
Sep 24, 2018
rajapruk
212
Sep 25, 2018
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Michael_GraceThanks. More details about my system: JBL 476Mg HF-driver JBL 045Be UHF-driver JBL H4365 horn 2 x JBL 1501fe woofers per speaker Jantzen cross-cap protection caps (22uf+0,68uf on UHF. 100uf+6,8uf on HF) Autoformers on UHF for x8 impedance (to be able to drive it with the Massdrop THX headphone amp. This also gives -9dB sensitivity) OPPO BDP-103 player with VanityHD outputcard (outputs 8 channels of digital AES3) BSS BLU-160 dsp (8 channels spdif/aes3 in and out) BSS BLU BOB2 8-channel dac (faulty, noise in some channels) 4 x Crown CTs-1200 amps (on -18dB input attenuation for HF, a bit less for UHF, and the lowest gain-setting 26dB for all channels) 2 x Massdrop THX-AAA headphone amps for HF and UHF (ETA November 2018). This also has some kind of dc-blocking cap in it. HF digital correction EQ is around -10dB UHF digital correction is around 0dB LF digital correction is about -18dB (linkwitz transform for closed boxes, etc)
Sep 25, 2018
Michael_Grace
432
Grace Design
Sep 25, 2018
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rajaprukWow that is quite an impressive system. I would love to hear it! I am curious about the autoformers on the UHF. What are their specifications?
Sep 25, 2018
rajapruk
212
Sep 25, 2018
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Michael_GraceThanks. If you ever stop by Gothenburg, Sweden you are welcome! I have model 3654 from https://critesspeakers.com/autotransformers.html So the thumps are no problem then for me, is it still your opinion? Do you buy me new drivers if your dac kills them? ;)
Sep 25, 2018
rajapruk
212
Sep 26, 2018
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Michael_GraceI am now choosing between your dac and the SMSL SU-8 (I think): https://www.massdrop.com/buy/smsl-su-8-dac I have severe troubles deciding. You are part in the case of course, but what benefits does your dac have in this comparison? The specs on paper looks in favour of the SMSL.
Sep 26, 2018
Michael_Grace
432
Grace Design
Sep 26, 2018
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rajaprukHaving no experience with SMSL it would be hard for me to make a useful comparison. As for measured specifications, without detail on how measurements are made they are fairly useless. i.e. test frequency, sample rate, measurement bandwidth, filters applied, levels, ets. You need all those test conditions to be able to compare specifications. Watch out on THD+N specifications. These days it seems to be popular to make this measurement with an A weighted filter. This will make your number look really good but it completely defeats the purpose of a harmonic distortion measurement if you throw away the harmonics you are measuring with a filter! All I can tell you is that we design out products with the utmost attention to detail for sonic purity and reliability. Cheers, Michael
Sep 26, 2018
Michael_Grace
432
Grace Design
Sep 26, 2018
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rajapruk Hi rajapruk, I would love to visit Sweden. A beautiful country you live in! I can not provide guarantees for your speakers for many reasons. But you would be very hard pressed to find evidence that the SDAC could cause any sort of damage for all of the reasons I listed before. Voice coils are damaged by heat. Heat is caused by power dissipation on the coil. The pulse in the SDAC simply produces too little power to cause enough heat to damage your speakers. ..By a few orders of magnitude at least:) As a system designer you could do the calculations yourself which would probably give you more confidence. Cheers, Michael
Sep 26, 2018
pospos
70
Sep 26, 2018
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Michael_GraceHi Michael, here are measurements of the SU-8 : https://audiosciencereview.com/forum/index.php?threads/review-and-measurements-of-smsl-su-8-dac.3778/
I think what rajapruk is concerned about is not power related, but excursion related: diaphragm fatigue than can occur with metal suspension diagrams.
I think a protection cap can take care of most of it. Here are for example the same FFT of the on/off clicks as above but with a 1st order 200Hz HP filter applied:
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(nothing left to see here)
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Sep 26, 2018
rajapruk
212
Oct 12, 2018
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Michael_GraceIf possible, please post a distortion vs. level measurement. I would especially like to see how this dac performs at -15dB to -40dB input levels. My use-case is active speakers with digital crossover and digital volume-control, so I need dacs with good performance at those lower input levels.
Oct 12, 2018
Michael_Grace
432
Grace Design
Oct 12, 2018
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rajaprukHi rajapruk Here are a series of fft plots with corresponding THD+N measurements of the SDAC Balanced at various operating levels down to -45dBFS. As you can see at -35dBFS all the distortion harmonics drop below the noise floor. Of course the THD+N numbers will read higher as low signal levels because the "N" (noise) component of the measurement becomes more dominant. Cheers, Michael
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Oct 12, 2018
pospos
70
Oct 12, 2018
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Michael_GraceHello Michael,
I think rajapruck wanted to ask about IMD vs level rather than THD vs level, in reference to the "intermodulation distortion versus level" measurement Amir routinely does on the ASR forum. This test shows strange behaviors with all last generation mobile ESS dacs (ESS "hump" in the -40dB to -20dB range) which is a concern when using digital level control. Example: https://audiosciencereview.com/forum/index.php?threads/review-and-measurements-of-wesiontek-khadas-tone-board-dac.4823/
Oct 12, 2018
Michael_Grace
432
Grace Design
Oct 12, 2018
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posposHere is a plot of the SMPTE/DIN IMD distortion vs. level. Not sure if the signal parameters match the plot in the Audio Science measurement but I ran 4:1 and 1:1 for reference.
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Michael
Oct 12, 2018
rajapruk
212
Oct 12, 2018
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Michael_GraceThanks Michael. It looks very good!
Oct 12, 2018
verifonix
1181
Jul 19, 2019
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No.
Jul 19, 2019
verifonix
1181
Jul 20, 2019
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Dear lord do you see a mention of MQA anywhere at all on the entirety of the product page? Can you put two and two together, or do you need Michael's help with that...?? MQA = not supported man, unless you do software unfolding.
(Edited)
Jul 20, 2019
rslatara
287
Sep 18, 2019
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Michael_GraceUSB 2.0 works but still missing a driver, not sure if it's theyscon, xmos etc?
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Sep 18, 2019
Michael_Grace
432
Grace Design
Sep 18, 2019
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rslataraHi rslatara, The DFU endpoint does not have a driver under windows 10. The DFU endpoint is not used so you can ingnore this entry. Should a firmware update ever become necessary for the SDAC we can issue a driver for the update that will use the DFU. Cheers, Michael
Sep 18, 2019
rslatara
287
Sep 18, 2019
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Michael_GraceThanks for the update, everything works as it should with 32/384. I actually had a USB interface for a Hifiberry with a similar thing but it also works just fine. It's my first grace product and it has really improved the LCX especially SE thanks to the additional gain over RCA. For a small DAC it outputs a excellent sound even on a Jotunheim, airy and a neutral/semi-warm tone with a little sparkle in regards to imaging and the treble.
(Edited)
Sep 18, 2019
rslatara
287
Nov 8, 2020
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Michael_GraceEver consider making an artist size dac with the 4497 etc. or something like a dual burr brown pcm1795 which often get overlooked? Then throw extra connectors. If you made something like that it would be a license to print money, I'm not sure if drop are making the airist again but there is definitely a void in a matching dac for 789.
Nov 8, 2020
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