Is a master clock needed?

[voodooless]

I know what it's supposed to do - the question is, "does it actually do it"?...

if it doesn’t, it really doesn’t matter how many DACs you have. It will already fail with a single one already.

curious reason? - heck yeah - curious till I'm cold and dead...

Sounds like a lot of fun

I want to assemble a small personal music listening system (something I haven't had in decades) that sounds great but doesn't puke and die after several months in service... we'll see how that goes...

Good luck. I’m pretty sure you’ll get som excellent advice here.

 

[MCH]

The only issue is if you want to measure the system because of clock drift. I bought a Behringer ADAT8200 to get a clock synchronized signal into the Digiface - you simply select your mic input as clock source.

Mixing different USB devices will always lead to clock drift, but for me it's only a problem when actually measuring the system. Vinyl playback is fine. Some measurement systems like Audiolense can compensate for clock drift, but most don't.

Sorry to the OP & friend but now that we are at it...
This is exactly where it gets confusing to me.... I use an umik to measure, as anyone else. I connect the umik to my laptop via usb and my DAC to my laptop via usb as well, i send sweeps (with timing ref) from rew to the DAC and capture the sound with the umik. Even though mic and DAC are not clock synchronized... I guess it doesn't matter... or does it matter?? and further, when does it matter to have the mic adc clock synch with the dac or dsp?? would i be doing better using an analog measuring mic and connecting it to a mic input of my dac (motu ultralite mk5) so that mic and dac work under the same clock? if yes, why?
to many questions, i know, but it is a long time i am wondering about this.....

 

[digitalfrost]

For me it mattered. I could not get good measurements with 2 different USB sources using REW unless they were synchronized to the same clock. Maybe the UMIK does some magic as a lot of people use it.

I have separate components for everything.

e: The thing is. Look at your DRC solutions. Some require precise time alignment for the measurement, i.e. you have to put the mic into the acoustic centre within a few cms. You can easily see and measure this with the time delay in REW or if you use Holmimpulse. Acourate too.

 

[antcollinet]

Resampling is only required if the sample rates are different between devices, but even that is no biggie.

But that is the point - unless syncronised, the sample rates will always be different even if only down to clock tolerance (see my explanation up there a bit...^). So either a PLL or ASRC is always needed for an SPDIF input (assuming there is no common external clock sync.

 

[antcollinet]

But they are, they are clocked by the SPDIF interface.

You joined ASR because your old? Curious reason

I'm old. And I joined ASR.

The two are not causally linked: Niether did I join ASR because I am old, nor am I old because I joined ASR.

 

[thecheapseats]

if it doesn’t, it really doesn’t matter how many DACs you have. It will already fail with a single one already.

if it's just a one dac fail - not three spread across crossovers - many people (present membership here excluded) likely wouldn't notice... did that sound jaded?... if so that was the intention...

 

[voodooless]

if it's just a one dac fail - not three spread across crossovers - many people (present membership here excluded) likely wouldn't notice... did that sound jaded?... if so that was the intention...

That’s not really what happens with SPDIF, it either works or it does not. You’ll immediately notice if something is wrong.

 

[dc655321]

I connect the umik to my laptop via usb and my DAC to my laptop via usb as well, i send sweeps (with timing ref) from rew to the DAC and capture the sound with the umik. Even though mic and DAC are not clock synchronized... I guess it doesn't matter... or does it matter??

The REW timing references in its sweeps are useful to provide an absolute timing reference embedded in the stimulus signal, for acoustic phase determination.
This can also be used to determine sampling rate shifts between stimulus and recorded signals, and correct for it in analysis.

and further, when does it matter to have the mic adc clock synch with the dac or dsp?? would i be doing better using an analog measuring mic and connecting it to a mic input of my dac (motu ultralite mk5) so that mic and dac work under the same clock? if yes, why?

Having a common time basis (clock) for input and output signals is useful for some measurement techniques (source-synchronous acquisition), or as in this thread, to enable a common "driver" from source to sink.

 

[thecheapseats]

I'm old... I think I want to try a class D amp that looks like a lego project... jigsaw puzzles are for really old people...

 

[thecheapseats]

That’s not really what happens with SPDIF, it either works or it does not. You’ll immediately notice if something is wrong.

of course you would notice - I'd notice... some don't... ran across a brand new consumer device belonging to a neighbor that wouldn't sync/drive spdif into a 75ohm coax cable longer than three feet without a 'click' every several seconds - a one foot length didn't 'click'... recently read an older comment here testifying they hooked up their mains speakers out of phase and 'discovered it' long after the fact... stuff happens... people often times 'hear' with their credit card - not their ears...

 

[Keith_W]

However, now that a computer is involved - for ripping LPs and playing them back from SSD, via Roon, I now have:

• phono stage --> A2D converter --> 4-way digital source selector ... connected with coax SPDIF cables.
• 4-way digital source selector --> SPDIF-to-USB converter (a miniDSP 'USBstreamer') --> PC (as the PC only works with USB input, not SPDIF).
• PC --> USB-to-SPDIF converter (DDHiFi TC100-COA) --> nanoDIGI --> 3x Topping E30s. So the nanoDIGI and the E30s are connected with SPDIF.

If I have understood all the posts so far correctly, this system can be divided into two parts as far as the clock signal is concerned. Before the PC, and after the PC. Please comment on these individual assumptions and let me know if I am correct or not:

- The PC takes the incoming digital stream from the Phono/ADC converter, applies convolution and DSP. Everything coming in from the ADC is treated as "data", stored in memory, and clocked at whatever clock speed the PC is running at.
- The PC then outputs via USB with a new clock signal, generated by the PC, embedded in the USB stream.
- The MiniDSP Nanodigi slaves its clock to the USB clock from the PC, then outputs to 3 Topping DAC's, which are all slaved to the MiniDSP clock.
- Therefore, because all the DAC's are slaved to the same clock, there will be no clock drift. So a master clock is not needed. The PC acts as the master clock.

Unless of course, the PC is not the greatest clock and might be producing jitter? Has anybody measured the jitter that comes out of a PC? And if so, does jitter get periodically worse if there is high concurrent CPU overhead?

 

[solderdude]

An external clock is more about the 'word clock' than the actual clock frequency.
The word-clock determines when all connected devices (that are word clock synchronized) have to start the sample.
This way the connected DACs and ADCs are synchronized and all start the sample at the exact same time.

Every DAC has their own clock which has their inaccuracies (drift, jitter) and their own internal synchronization (PLL or something similar) that can 'tune' the receiving side of the DAC device to match that of the incoming signal.
This is to prevent buffer over- and under-run so there will be no 'ticks' or missing samples if the deviation between the source and DAC device clock differ too much.

One should realize that a DAC is not a real-time device like analog.
There are always several clock pulses between the input and generated output.
So all used DAC devices should be the same type with filters set the same way as filters determine the time between the sample arriving at the input of the DAC device and the actual outputting of the corresponding output voltage dictated by that sample.

As this is the case here (all the same DAC devices with the same filter settings) and using SPDIF all three DACs will get the same word clock they will all be neatly synchronized as they are all slave to MiniDSP and thus all synchronized. The clock drift and jitter before the mini-DSP is irrelevant. This has been taken care off/changed by the mini DSP.
Jitter of the mini-DSP should be lowered/removed by the DAC device's jitter reduction (syncing between its own internal clock/PLL and what is coming from the mini DSP)

 

[voodooless]

- The PC takes the incoming digital stream from the Phono/ADC converter, applies convolution and DSP. Everything coming in from the ADC is treated as "data", stored in memory, and clocked at whatever clock speed the PC is running at.

Most probably it will use the ADC clock as a reference, but it will depend on the particular implementation.

- The PC then outputs via USB with a new clock signal, generated by the PC, embedded in the USB stream.

Nope. Most USB DACs generate their own clock. They are the master. The PC needs to resample whatever it gets to that DACs clock. If you have multiple USB DACs you’ll need either a master clock, or resample for every DAC individually. This is a totally different story from SPDIF.

- The MiniDSP Nanodigi slaves its clock to the USB clock from the PC, then outputs to 3 Topping DAC's, which are all slaved to the MiniDSP clock.

As said, this is not the case. The nanodigi is the master. If you can actually choose the sample rate on the USB interfaces there is most likely another ASRC in place to convert the USB clock domain to the DSP clock domain.

- Therefore, because all the DAC's are slaved to the same clock, there will be no clock drift. So a master clock is not needed. The PC acts as the master clock.

Yes

Unless of course, the PC is not the greatest clock and might be producing jitter? Has anybody measured the jitter that comes out of a PC? And if so, does jitter get periodically worse if there is high concurrent CPU overhead?

The clocks in the PC are more or less virtual and can be precise enough. I would not worry about that too much.

 

[MCH]

Besides the minidsp udio and the rme digiface, there are not many such devices available. Anyone knows how they work internally? What sort of ic they use? I am curious and can't find any information online...

 

[rydna]

Besides the minidsp udio and the rme digiface, there are not many such devices available.

Available to do what?

As said, this is not the case. The nanodigi is the master. If you can actually choose the sample rate on the USB interfaces there is most likely another ASRC in place to convert the USB clock domain to the DSP clock domain.

Im pretty sure Roon lets me choose the sample rate for the incoming and outgoing USB streams. They are set to 96kHz, to match:

• my A2D converter, on the input side
• and the nanoDIGI, on the output side.

Thank you for your input, V - I understand now that:
a. because all 3 (output) DACs are the same, and
b. they are connected to the nanoDIGI via coax SPDIF ... they will stay in sync with the nanoDIGI.

 

[Keith_W]

OK, I think between @solderdude and @voodooless we have our answer. Thank you gents.

 

[MCH]

Available to do what?

Multichannel USB to n stereo spdif outs. To do for instance what you do without the need of a multichannel DAC.

 

[rydna]

Multichannel USB to n stereo spdif outs. To do for instance what you do without the need of a multichannel DAC.

Sorry, I don't follow you?

If you have 'n' stereo SPDIF outs ... surely you need 'n' DACs to produce 2xn analogue outs (or an n-channel DAC)?

 

[see_no_evil]

Sorry, I don't follow you?

If you have 'n' stereo SPDIF outs ... surely you need 'n' DACs to produce 2xn analogue outs (or an n-channel DAC)?

Not really if what you want is an interface between PC (USB) and external audio gear (SPDIF).

 

[Sam Spade]

Yes, I know that a master clock makes no difference if you are using a single digital device, like a DAC. However, my question is about multiple digital devices.

I read this article (in a pro audio blog) about whether your studio needs a master clock or not. My take-away from that article was contained in this quote:



The application is for a friend of mine who has a system configured like this:

- Turntable --> Phono stage --> ADC
- ADC --> MiniDSP
- MiniDSP --> 3 Topping DAC's --> rest of the system

My understanding of using multiple digital devices is that each device might latch on to the signal at different times, and variations in clock accuracy between each device might cause clock drift over time. After some time, the difference in timing between DAC's might become audible, particularly if one DAC is driving the tweeter, and another DAC driving the midrange, etc. When I was configuring my own system, a friend of mine who is an audio engineer told me NOT to use multiple DAC's for multichannel digital output because of clock drift. Or if I wanted to, I had to slave them all to a master clock. Because of his advice, I purchased an 8 channel DAC.

However, that article I linked to mentions that a master clock is not needed in "simple" studio setups, and is only essential for complex setups involving multiple ADC's or if video is involved. Because that is a pro audio blog, I am guessing that they do not sit down for an hour listening to a single album played from start to finish, so clock drift may be less of an issue for a "simple" studio where they stop and start tracks which will give all digital equipment in the chain a chance to resynchronize.

As far as I am aware, my friend's ADC (I don't know what brand) and his Topping DAC's do not have clock outputs or inputs, so it would be impossible to slave the DAC's to the ADC, or even slave all the digital devices to an external master clock. I suppose this may not a problem if he was using a digital source, because the signal would stop and start at the beginning of each track, meaning that the DAC's would have an opportunity to resynchronize. But he is using vinyl, which means noise might be transmitted to the DAC's even between tracks, so the DAC's might not have an opportunity to resynchronize.

I did some "back of the napkin" math, and this is what I came up with. Assume we have a DAC with a deviation of 50ppm, and a "worst case" scenario where the difference between the first and second DAC is 50ppm.

- 44.1/16 * 2 channels = 44100 * 16 & 2 = 1,411,200 bits per second
- DAC clock runs at double speed = 1,411,200 * 2 = 2,822,400 cycles per second (or 2.8MHz)
- 50ppm variability at 2.8mHz = 50/1,000,000 * 2.8224 = 0.00014112 seconds (or roughly 0.1ms) every second.
- Clock drift in 60 seconds = 6ms
- Clock drift in 1 minute = 360ms
- Clock drift in 30 minutes = 21,600ms (or 21.6 seconds)

I did not study maths beyond high school so there is a very high probability that I made a mistake in my math. I would appreciate correction, because clock drift of 21.6 seconds at 30 minutes seems astoundingly high to me. I am no match for you engineering types! So please be nice to me if I got my math wrong! Also, according to my calculations, the difference at just 1 minute of playback is 360ms so it should easily be audible. I subjectively did not hear anything amiss after listening for several minutes.

Advising him to change all his equipment to allow slaving to a master clock would be a major expense for him, so I want to check with ASR whether there is any truth to the assertion that clock drift between DAC's can cause group delay, what the magnitude of the problem is, and what it can potentially add up to over time. I do not wish to give bad advice, so your input is welcome.

did you say master cLock?