Hi, and welcome to the community.
To answer your questions:
@Ermos Thanks for your reply. I’m new to all of this! For 1, just to confirm: if voltages are given per leg, does that mean the dCS quoted 2v balanced setting for example is equal to 4v total? In other words 2v carried on the positive leg and 2v carried on the negative leg of the balanced cable for a total of 4v? most standard outputs i’ve seen quoted are 2v single ended and 4v balanced.
Actually I found the answer I was looking for in the Stereophile review. I should have looked their first!
"With the Bartók’s volume control set to its maximum, a 1kHz digital signal at 0dBFS resulted in a balanced output level of 6.03V into 100k ohms with the output level set to “6V,” 2.04V with it set to “2V,” and 603mV with it set to “0.6V.” The maximum levels from the unbalanced outputs were all very slightly lower and the maximum level from the headphone output was 6.82V. "
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The reason for this is the way that balanced differential circuits work. The signal is split into two polarities which are inverted in respect of each other; + and -. When they are combined at the receiving end to reduce noise that has been picked up the two outputs are summed. Most manufacturers provide the single ended output by using only one half of the existing balanced circuit. Hence the single ended output is exactly half the voltage of the balanced one. 2V is commonly found as it was (is) the specified output voltage for CD players that meet the redbook standard ( ISO 13490).
dCS do not do this and actually build separate output circuits of different design for single ended and balanced. Hence they can make it so that both provide the same output voltage ( or as close as is possible).
Note that Stereophile measure the voltages using a signal of 0dBfs. Even if you set Bartok output as 0dBfs the music signal of well mastered recordings will be a few dB below this to provide some headroom so as not to drive everything into distortion. That means that although you can play with the numbers so that output voltage = input sensitivity this will not be consistent in real life. Another reason to follow Ermos’ advice to set the line output voltage so that the volume control is in a comfortable position . Further as Bartok’s volume control is digital and as digital volume controls work by reducing bit depth (i.e. resolution), when using it directly into a power amp you do not want to have to have the average position of the control set too low in order to produce a comfortable listening level . So in most home set ups where direct connection is used this may mean that it is better to set the line voltage level lower than at the maximum available.
No, 2V out is 2V out, not 4V total. The + and - are both 2V, in opposite phase.
@PAR and @Ermos thanks very much for clearly explaining how the voltage works.
I will start by running the Bartok into my pre amp (Bryston BP17(3)). According to Bryston the unit was designed around the standard 2V CD input you mentioned above.
As some point I’ll experiment going direct into the Amp. 2V will likely be the correct setting but experiment as you suggest to get the volume range on the Barktok between -30 and -10 as the FAQ suggests. My Bartok is on the way but it will be a few weeks yet…
Thanks again for your help!
I found in the data sheet of Bartok that the recommended input impedance is 10k-100kΩ
1 x pair balanced outputs on 2 x XLR connectors. Output levels: 0.2V, 0.6V, 2V, 6V rms for a full-scale input, set in the menu. Output impedance: 3Ω Maximum load: 600Ω (10k-100kΩ is recommended)
Next, I look at the input impedance of my ATC SCA2 preamp and see that for a balanced input it is only 2k7 per leg.
At the RCA input, everything is pretty standard.
What does the recommended 10k-100kΩ mean and how can this affect the sound in the case of working with a preamp where the input impedance is 2k7?
The term “maximum load” here is a bit confusing. The load is actually higher with a lower input impedance value. I’m not sure what 2k7 means, but assuming it’s something like 2000 ohms you should be fine, since it’s a lower load than the 600 ohm maximum. It’s not quite in the recommended range but I expect it won’t be a problem.
No problems. Your Bartok’s balanced output is absolutely fine with your ATC SCA2 preamp.
This is all about power transfer and the usual rule of thumb is that the receiving equipment’s input impedance should be at least 10 times the sending equipment’s output impedance for the most efficient power transfer . As 3 ohms output is 900 times less than the input impedance ( 2,700 ohms) it is completely suitable.
The dCS spec. basically just quotes the most commonly found minimum or maximum input impedance numbers for pre or power amps. In other words it will work with just about anything except for unusual outliers.
Remember that I also own an SCA2 !
Its just a bit of engineer’s jargon and uses K instead of thousand or 000. so 2K7 is 2 thousand 700. 8,600 would be 8K6 etc.
After almost a year with my Bartok and now Rossini Clock, I finally got around to comparing the Bartok run direct as a digital pre into my 200W Bryston 3B(3) vs 2v line into my Bryston BP17(3) preamp.
Round 1. Bartok 2v out into my Bryston pre amp.
Round 2. Bartok 2V direct into my Bryston Amp
Round 3. .2v and .6v Direct into my Bryston amp
For normal to loud listening levels running direct sounds best. The sound is absolutely beguiling. For casual low level listening, adding the pre amp wins.
The other revelation for me is how much less power I must be using. My amp is rated: 200w at 2.8v and 100w at 2V input on the 23db setting. So running at 0dB at .2v or .6v I must be consuming a hand full of watts at most! Speakers are Harbeth SHL5+. I listen in midfield.
can I jump in here as I’ve learned a few things in this thread…
I’ve been running the bartok at 6V into my AMR-AM77’s as monoblocks and out to Dynaudio Confidence C2 Platinum’s, where the bartok volume dial listening is generally around the -40db going to the louder end at -35db… is this incorrect…?
should I be trying to reduce to a 0.6v output which allows me to up the output volume on the bartok to around -20db / -10db… or even use v0.2 and use bartok volume between -15db up to 0db -(which is loud but comfortable) - (but run out of headroom)
I hadn’t thought of trying this before
dean
It’s easy just to try it yourself and listen, personally using my Rossini as pre-amp did not work in my system. To keep it in the -30/-10dB range I had to lower the output to 2V or below.
Digital volume controls work by discarding bits and bit depth is one of the things that defines resolution.
So is is important that the bits lost in practice have an inaudible. result. In legacy dCS products that means attenuation with direct connection should not exceed ( from memory) around -23dBfs and with current models -30dBfs. However I would try to find a voltage level output that provides the lowest attenuation degree required to provide a comfortable average listening level.
Those lower voltage levels for output that you cite may be correct for you.
As @imprezap2 implies, using no pre-amp is not the best solution for everyone so if you have a chance it could be worth experimenting.
Hi Guys,
Can I just refer you all to this posting - it’s actually further up in this thread - but basically when you’re working internally at 32 bits of precision with 16 (or even 24) bit source data then volume control doesn’t have to involve throwing away bits if you do it properly…
"Now, the points that have been made about digital volume controls throwing away bits and decreasing dynamic resolution – these are not correct. The idea that a digital volume control truncates the signal and throws away bits to achieve a volume reduction would only be true if the volume control was to take place before the filtering stage in the DAC, which is not the case – at least with a dCS DAC. Considering how a digital filter operates, there are hundreds, thousands or potentially even millions of multiples/accumulates throughout the signal path. The volume control is effectively just the last one in this chain.
The architecture of a dCS DAC also means that the interface between the output of the filtering stage and the input of the DAC is carried out inside the main FPGA, so we don’t have to deal with fixed width interfaces which are present when for example sending digital audio signals between ICs. We have much more control over the whole process.
As an example, consider a 16 bit sample from a CD, along with a 16 bit volume control. All possible permutations of volume and sample are guaranteed to fit within 32 bits. Even if the signal was upped to 24 bit, there is still a digital noise floor of -192dB, with zero distortion or quantisation artefacts, assuming it is dithered properly. This is way below any real-world analogue noise floors.
The point to consider related to volume control however is that the DAC itself has an analogue noise floor, which is fixed. Reducing the signal by 30dB for example means the ratio of signal to the analogue noise floor of the DAC is reduced. This does not mean the DAC is simply throwing away bits to attenuate the volume. Another point worth considering is that with some DAC types (notably ladder DACs) the distortion products of the D/A conversion structure are essentially fixed. They become proportionally bigger as the signal gets smaller (i.e. when the volume is turned down), so poor low-level linearity is an issue. Not the case for the Ring DAC, as the distortion goes down faster as the signal gets smaller."
Phil
Thanks for the clarification Phil.
I’m now running my Bartok direct into my Bryston 3B3 direct at either .2V or .6V. At .2v I can run at 0dB for most chamber music and .6v 0–6dB for most orchestral programme.
It is a fairly easy to Switch voltage in Mosaic (I like to use the app in slide over mode over Roon - I prefer this than allowing Roon to control the volume)
Benefits for me at least are a subjective sense of greater clarity and depth and improved micro-dynamics. PPP passages are clearer because the noise floor is lower. I can easily hear this as the idle white noise from my speakers has been reduced to almost nothing.
If I’ve understand the APEX messaging correctly, there is a reduction in the analogue noise floor of the APEX boards. So does this mean the volume control of APEX’d Vivaldi and Rossini units has been improved as a result of the lower analogue noise floor?
I think the non APEX manual says the digital volume works well between -30 to -10dB. What’s the impact of APEX on this range? Really just curious
In my system when using the Rossini as a pre-amp there is definitly a dynamic decrease (at 2V and below) and it is not a comfortable way to change the volume (with DSD and classical music I needed to keep changing the output voltage)
With a “modest” integrated amplifier I don’t have this dynamic decrease when changing the volume.
I think the Rossini can work well as a Pre-amp, but it will depend on the gain of the power amp and the sensitivity of the speakers in the system.
Adding to the conversation… I just purchased a Benchmark AHB2 and I’m testing it directly driven by my Bartok. I listen mid to near field, so 1/3 W to 10W is typical for me. I can run the Bartok at 6V between -18dB to -6dB depending on programme material with the AHB2 on the lowest gain setting.
Assuming the Bartok DAC delivers a SN of 117.5 dB; at -16dB I’m only a couple of dB below the 105dB spurious response figure quoted by dCS (assume this is analog noise floor?). Long way of saying the noise floor of the AHB2 is around that of the Bartok during normal playback
What does this mean for subjective sonics?
Few other subjective findings on the AHB2