ESS Sabre Reference DAC (8-channel)

Re: digital volume control?

schro20 said:
I always thought that digital volume control was "bad" and all volume control should be done in analog. Reason: you only have so many bits, so if you digitally scale values from (let's say) 0-255 to go only from 0-15 you have only 16 different levels left when you started with 256. If you scale in analog you still have 256 levels between 0 and 15. The latter presumably having more fidelity. Is that argument missing the point?

(I hope not to unleash a big argument here... Just point me someplace I can educate myself.)

Or is this something that's special about the ESS chip that it can do this the "right" way and the above argument is still correct in a more generic context?

peter

Hi Peter,

Here is what I think, though I am sure some will disagree. :)

If you have a DAC with very good DNR its very difficult if not impossible to do better than use its built in attenuation.

If you attenuate on the digital side then you put the onus on the DAC to have great DNR. The Sabre has this in spades, so its a perfect candidate.

Show me an analog attenuation circuit that can match to 0.005dB or so across channels and still make a -132 DNR (that's about -250nV integrated across 20kHz when compared to 2Vrms, or roughly 1.78nV/root Hz). It's just not likely to happen. :)

That said I use analog volume control too, but when I do I use something like my Joshua Tree stepped attenuator which has very good channel matching, but not anything close to .005db :).

There is nothing "wrong" with either approach. Its mostly going to come down to user preference. It's just different degrees of goodness. :)

My personal experience is that using the ES9008 volume control yields exceptionally good results. Better than any analog attenuation I have heard yet. Keep in mind, to get the best possible DNR (by reducing the noise floor) a balanced output is ideal when using the DAC volume control as there is nothing down stream to attenuate noise. The good news, is there is none to be heard at all in my setup. The color of the music seems to emerge from a perfectly clean and clear canvas. That's what I love about this DAC.

Cheers!
Russ
 
Sabre Chip Output Summing in Stereo Mode

If I remember correctly, when in stereo mode, the sabre chip internally parallels 4 DACs for each channel. I have searched, but found no info on the following, probably trivial questions.

(a) Does this mean that the signals at all 4 outputs are equivalent and it doesn't matter which one I use?

(b) How is the internal paralleling done?

(c) Could summing the 4 outputs somehow be done outside the chip? If so, could this potentially improve sound quality? Maybe by summing the 4 currents in the I/V converter, or, in the most extreme case by using a total of 4 I/V-converters per channel?
 
Originally posted by schro20
I always thought that digital volume control was "bad" and all volume control should be done in analog. Reason: you only have so many bits...

Digital volume control *can* be bad if badly implemented. Say without proper bit-depth for the processing or no/bad dither techniques. It got quite a bad rap from some poor implementations in the PC software world among others.

here's a good start:
http://extra.benchmarkmedia.com/wiki/index.php/Digital_volume_control
(you might also find good info in app notes and such from MFG's or finding some material on DSP techniques. I've been meaning to study up on this myself, but time is limited ;-)

It would seem from all accounts here that the Sabre does a fine job of it.

-Chris
 
eclectic2k said:


Digital volume control *can* be bad if badly implemented. Say without proper bit-depth for the processing or no/bad dither techniques. It got quite a bad rap from some poor implementations in the PC software world among others.

here's a good start:
http://extra.benchmarkmedia.com/wiki/index.php/Digital_volume_control

The volume control in the Sabre DAC has been designed so that it won't have the distortions that are discussed in the link.

I haven't designed volume control into my Sabre DAC because I'm sending full volume analog signal to a preamp, so the rest of this post is just conjecture. :D

A volume control, simply stated, will just shift bits down in magnitude. If you have a 16 bit signal, you can turn it down and the 16 bits will shift down until the top 8 bits are all zero, the 16 bits of data now reside at bits 9 to 24 and you will not lose any resolution if you have a DAC with a noise floor at -144dB.

Then if you turn it down some more you will lose bits of resolution. If you have a 24 bit signal, you will lose bits of resolution with any turn down of the volume setting.

So, if you have a setup with your amp where you have to turn down the volume way down all the time, it will not be good. You will always lose bits of resolution. A lot of amps are this way with way too much gain.

If you can adjust the gain of the amp so that the loudest you ever listen will correspond to the DAC volume control full up, then that would be ideal. Now a typical listening session would have the DAC volume control only reduce the signal by 6dB to 12dB.

If I were to implement a volume control, I would try it by not allowing a reduction of more than 18dB. That would correspond to a shift down of 3 bits on the signal. That would mean that 24bit data would lose 3 bits of signal, and 16 bit data would be shifted down to bits 4 thru 19. Therefore, If you have a -120db noise floor, you will still have 17 bits of signal above the noise.

This would be implemented by setting Sabre registers 0 to 7 and restricting them by only allowing a range from 0x00 down to 0x18 or 0x24.

There's a good experiment for someone to try! :D
 
Re: Sabre Chip Output Summing in Stereo Mode

Javin5 said:

(c) Could summing the 4 outputs somehow be done outside the chip? If so, could this potentially improve sound quality? Maybe by summing the 4 currents in the I/V converter, or, in the most extreme case by using a total of 4 I/V-converters per channel?

Answer C is what we are doing. Look at the ESS web site for the schematic for the Sabre8 stereo demo board and a white paper describing the internal architecture of the part.

http://www.esstech.com/techsupp/drivers.shtm
 
Hi.

I am running a digital volume control (64bit float) on my PC even on 16bit material . I am allowing not more than 12db attentuation.
After all, this is sounding ways better then any preamp (passive / active) I tried before.
DigitalVolumeControl with direct DAC-Amp coupling get's you lot of problems solved.
No impedance mismatches and variations, coupling caps, cables, asf.

The key parameters to consider when talking about digital volume control are IMO more then mentioned before. You need to dimension you entire chain to be able to have lowest impact on the digital side. You need to look at the voltage of your DAC output stage, the amp gain, the speaker sensivity and your typical listening volume to get digital volume control going at lowest impact.
Keep also in mind that you easily can have 12db difference on the
music material depending on the recording level, which is increasing your "working" range to the disadvantage of digital volume control.

The best way would be IMO some kind of passive shunt-to-ground volume control on the DAC output (see also how the Lessloss guys are doing it). Unfortunately my current DAC is not really supporting this.

I'll soon face the same challenge with the Buffalo dac.

I am wondering if a passive I/V conversion on the DAC out
would be recommended? Did anybody try this?
 
I have just received my Buffalo and rigged up a passive output using 10uf Evox Rifa film caps. It is not plugged into my main system yet (tomorrow) but through an old mixing desk into headphones sounds full of potential.

If I can figure out how to modify my RBroer solid state I/V stage I shall try that next. (OK, already tried marrying the two togther, with poor results due to differences in the output voltage/current of the Buffalo and Monica chips)

All good learning, I am sure.

Mark
 
Disabled Account
Joined 2008
ESS Sabre GUI incomplete

Hi Dusfor99 (or whoever can help me),

i need you help for a problem with ESS Sabre GUI.

I'm owner of an ESS Sabre EVB. I have a problem with some
registers setting through the most recent release 2.0.2.1. (but the problem there is also with the previous version) of Sabre8 GUI (2.0.2.1.): I was not able to find in any section of the GUI the window called "Input Settings (Quantizer Setting)" where to editing the number of bits each quantizer will use. While setting in stereo mode the DAC (2 Channel) i cannot set the value of the quantizer. Through the "List all register" button I see the Register 15 setting to 00000000 (6 bits quantizer), but i cannot find the "Quantizer Settings (Input)" Window where to modify the value to 7, 8 or 9 Bits. I attach the screen capture taken from the Sabre8 Gui Help file show the windows that I cannot find.
Oherwise could you sugest another way to modify those values?
Also I don't find where to midify the "differential" from "True" to "Pseudo". I use the EVB with I2S input.

Sabre firmware is updated to the latest version available in
http://www.esstech.com/techsupp/drivers.shtm#sabre.

Thank you.
 

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Spartacus said:
With music at full volume via either a media player app or the windows mixer, I can hear a low level distortion that sounds like clipping.
When I turn the volume down on the PC (and up on my pre-amp to compensate) it gets worse.


I really don't know much about digital processing..... but I do know that Windows can do some weird things when changing volume in the digital domain. I would suggest looking into something like ASIO to get a bit perfect output, or try testing from a non-PC source.

I don't know why it would show up on the Buffalo, but not Opus though :confused:
 
Just got myself a Denon DVD-1920 off of ebay :) Would it be beneficial to use a TP SPDIF receiver inside it to transfer PCM audio to the outside? I remember this being done on the Oppo. Also, I might be wrong on this, I remember Russ or Brian talking about a board to be designed for use with the Denon to get DSD to the DAC. Is this something that is being done, or was I dreaming :)
 
Russ White said:


I am planning on producing an SDIF3 DSD output board at some point yes. :)

I would not bother with using the SPIDF board as a transmitter for the DVD-1920. It would just be redundant. The reason it makes sense for the Oppo is that it converts the DSD to PCM. The Denon does not.

Cheers!
Russ

My thinking was to use a SPDIF module for PCM and a separate connector for DSD. That was the need for some kind of switch inside the Denon would be avoided. But I would still need a switch in the DAC for this, but only to activate the DSD signal, the PCM would be regular SPDIF. Another way of doing it might be to run both PCM and DSD inside the same cable, say a cat5 network cable. But I would still have to switch it in the DAC. Did you get anywhere on the automatic switching using the SACD LED?

Looking forward to the SDIF3 module :)
 
Re: Dustin on the radio through Sabre, of course

wildmonkeysects said:


Ross, could you elaborate on the reference system(s) that you heard? Amps, speakers, power conditioners, room treatment, the works. We wanna know what toys the boys at ESS get to play with...and to help frame a reference for us to compare our experiences with.

Straight ESS demo board, or something snazzier? Do you know what filter settings, modulator bits settings, etc?

WMS


Sorry to take so long to reply, vacation schedules and it took a little while to get back in touch with Bob at ESS.

The ESS demo room is 8'x12'x24' with 8" thick walls with internal sound dampening. It's HVAC is isolated from the building. A concrete slab physically separates the building slab from the demo room.

Equipment list:
CD/SACD Player: Denon DVD-1940CI (modified with DSD/Serial out jumper cable)
DAC board: ESS 8-channel Sabre Reference board (with metal film caps and Crystek crystal)
DAC: ES9008 Sabre Reference DAC
Op Amps: AD797 (3 per channel)
Receiver: Denon AVR3808CI
Front speakers: B&W 801 Series 2 with stands
Center speaker: B&W HTM1
Rear speakers: B&W 802
Subwoofer: None
Speaker wire: Std Monster cable
AV connectors: Std RCA cables

The demo board was the same one that ESS has made available on their web site. So there was nothing special that they were doing that wasn't disclosed by them. It's also the same board that was used in the comparisons done the an outside company. I was impressed how open and fair they were regarding the comparison testing. They needed a true comparison of the Saber chip for themselves, and they were happy to hand out copies of the test results.

In the list above, you will notice no subwoofer. Didn't need it, the 801's sounded awesome.

The interesting thing about listening to various CDs, not every CD sounded incredible. A lot of the sound depends on the quality of the recording, of course. For example, my standard CD copy of Dark Side of the Moon was lacking in sound quality...but then again it was recorded in '72. But every piece of music we listened to was noticably improved by the Sabre chip. Well recorded pieces had an openness and depth that the sound straight from the Denon (and into the B channel of the amp) didn't have.

RossG
 
Re: Re: Dustin on the radio through Sabre, of course

krgaunt said:

The interesting thing about listening to various CDs, not every CD sounded incredible. A lot of the sound depends on the quality of the recording, of course. For example, my standard CD copy of Dark Side of the Moon was lacking in sound quality...but then again it was recorded in '72. But every piece of music we listened to was noticably improved by the Sabre chip. Well recorded pieces had an openness and depth that the sound straight from the Denon (and into the B channel of the amp) didn't have.

RossG


I will whole-heartedly agree with that statement. New material that has been mastered well and the newer remasters of old tapes can sound fabulous. Older CDs that were not mastered with care... well, I don't listen to them very much any more because the flaws in them are much more apparent.

Have you guys noticed something. I don't know if it's just me, but... when I listen to a good recording on my system with the Sabre8 DAC, then go away to do something, the good recording will stick in my brain and play over and over in my mind. The badly recorded material I just forget about and it doesn't stick in my brain.

Am I going crazy or what? The good recordings just stick in my head because they sound so real. :D

Ross L.
 
OK - more progress!

The buffalo is assembled and everything powered up exactly as expected. +/- 15V for the IVY seems very stable and the 6V/6.5V for the DAC seems very stable too. I've just connected it to my PC (USB to SPDIF converter) but what I'm hearing is heavily distorted with crackles all the way through the music. It also seems to have a hard time locking on to the SPDIF signal. Crackling is worst on bass notes.

I'm off to try it with a non-PC source next although I've never had problems before.

Will report shortly with photos and more details.


Fran
 
Hi,

@Russ / anyone...

I've just ordered a Sabre ES9008 chip on it's own and will be starting a board design very soon. I'm looking at using it for both stereo and surround material as part of my first amp build...

I just wondered if anyone had given thought as to how to multiplex the DAC outputs so you can combine the eight outputs into two channels for stereo, then back to eight channels for surround playback? This would give the best sound for stereo music while keeping the costs down for multichannel (should hopefully also sound awesome from what I've read on this DAC).

What about an (8:1) x 2 mux IC from Analog? Would this give resonable crosstalk figures and matching?....

http://www.analog.com/en/switchesmultiplexers/multiplexers-muxes/products/index.html

I've thought about using relays, but that would require eight DPDT relays (since the outputs are differential) and would be a tad cumbersome - is there a simpler way of doing it (without using manual switches)? Or, you could use a set of relay contacts 'across' each pair of outputs to be combined so they will tie the outputs together when activated?

Do the digital inputs on the Sabre also need to be combined when in stereo (quad-differential) mode?

Thanks,
OzOnE.
 
What about separate I/V converters for all current outputs, and then do the summing to stereo with summing amplifiers after the I/V stage? In other words, you do not sum the currents at the DAC outputs, but rather after separate I/V stages. Correctly configured, one can then have both stereo and surround from the same chip, just a matter of switching the digital inputs between 2 channel and surround.

Rolv-Karsten
 
Hi,

Aaah, I see. So I could just use buffers after the outputs which would permanently give a combined output (alongside the separate outputs), then just switch to that for stereo?

But, how much would that degrade the output compared to joining the DAC outputs directly (from the point of view of the added SNR that the combined outputs would achieve)?

Would using buffer amps even work properly as far as the DAC is concerened - ie. are there strict phase / delay / freq requirements for combining the DAC outputs to achieve the added SNR?

OzOnE.

P.S. btw, does anyone have a working link to the ES9008 datasheet as I seem to have 'misplaced' the copy I had?