Are pre-amps necessary

[watchnerd]

I used to work with a Revox PR99 Mk II... in 1985?

Yeah, mine are from that era, one from 1984, one from 1989.

Both bought within the last year, in my case.

 

[restorer-john]

Bottom line: for normal use, a good, fixed analog output range DAC, with digital volume control, will probably outperform a normal, not State of the Art, analog preamp.

Right, OK...

Now consider every D/A converter Amir tests, he loads its line outputs with a 200K ohm analyzer input. How is that remotely representative of typical power amplifier input impedances? Hint: it's not. Most power amplifier input impedances range between 10K and 50K, and with amplifiers incorporating input level pots, the impedance will be likely at the low end of those numbers.

D/A converters (line output) will have output impedances ranging from a few ohms (the good ones maybe) to many hundreds (or thousands) of ohms for the average ones. All of that combines to make the otherwise flat FR and low THD numbers less likely (if at all) to occur in a real system. Take the industry darling NC-500 (or equiv) with its unbuffered 1.8K ohm per input and tell me a typical D/A converter is going to perform well into such an impedance. And yet people are plugging sources into these modules without using an outboard buffer in some cases.

Good preamplifiers have output impedances below most of the headphone amplifiers tested on ASR. They can swing large voltages at very low impedances and happily drive long cables and multiple amplifiers with little or no modification of the ruler flat FR and low THD.

People obsess over headphone output impedances but rarely even consider the line output impedance of their D/A converter before eschewing a proper preamplifier and plugging in their power amps.

 

[Rja4000]

Hi
Thanks for your input.

I'm an engineer, but not an EE, and even if I'm a long term user of HiFi and pro audio gear, I'm learning here, so I, for sure, am happy to get more knowledgeable people's input.

Your comment raises several questions, for me.

First, do you think it's correct to guess that most DACs meant for pro audio are likely not to have that kind of issue ?
This is the case for the RME ADI-2 Pro, in my experience.
(And, according to the spec, its Output impedance is 200 Ohm for balanced, 100 Ohm for unbalanced outputs, while its Input impedance is 18 kOhm for balanced and 9 kOhm unbalanced - which may mean it's mostly comparable to a typical amp for testing purpose).

Second, I don't know about the load @amirm is using to perform his measurements.
If it's not representative of a typical power amplifier, then that probably means that, for the DACs with volume control at least, we miss an important measurement here, which is their response under a load similar to those amplifiers. Or a measurement of their line-level output impedance.
And, for power amps, at least listing their input impedance.
Is that correct?

What does @amirm think about it ?

 

[rodtor]

Quite a few commentators on line refer to a minimum 1 to 10 ratio for matching the output impedence of a dac to the input impedence of an amp. The point, then, would be for the latter to exceed the former by at least a magnitude of ten. @solderdude referred to this 'rule of thumb' in another ASR thread last April. If this is right, the RME, with output impedence of 200 in balanced, would presumably match very well with an amp based on the Hypex 502MP, for example, whose input impedence is 47 k. But, like @Rja4000, I would definitely appreciate hearing from others about this important question,.

 

[solderdude]

Actually I keep a factor 100 as a 'rule' which isn't a rule in the case of DAC's at all.
A minimum of 10 would be more correct.
Inputs of amps are resistive in most cases, at least within and well outside the audible range.
There is no impedance matching here. Just output and load resistance.

Only in very rare cases (think passive pre-amps with transformers and tube amp outputs with a very high output resistance) some thought is needed.

 

[watchnerd]

Right, OK...

Now consider every D/A converter Amir tests, he loads its line outputs with a 200K ohm analyzer input. How is that remotely representative of typical power amplifier input impedances? Hint: it's not. Most power amplifier input impedances range between 10K and 50K, and with amplifiers incorporating input level pots, the impedance will be likely at the low end of those numbers.

D/A converters (line output) will have output impedances ranging from a few ohms (the good ones maybe) to many hundreds (or thousands) of ohms for the average ones. All of that combines to make the otherwise flat FR and low THD numbers less likely (if at all) to occur in a real system. Take the industry darling NC-500 (or equiv) with its unbuffered 1.8K ohm per input and tell me a typical D/A converter is going to perform well into such an impedance. And yet people are plugging sources into these modules without using an outboard buffer in some cases.

Good preamplifiers have output impedances below most of the headphone amplifiers tested on ASR. They can swing large voltages at very low impedances and happily drive long cables and multiple amplifiers with little or no modification of the ruler flat FR and low THD.

People obsess over headphone output impedances but rarely even consider the line output impedance of their D/A converter before eschewing a proper preamplifier and plugging in their power amps.

My question is:

If you're going to use a DAC with volume control, why not just go fully active speakers?

Why use a power amp at all?

There are many, many pro style interfaces/ADDA that are designed to hook up to monitor speakers, over balanced connections, with all the input / output impedances sorted out.

 

[Rja4000]

If you're going to use a DAC with volume control, why not just go fully active speakers?

Actually, @restorer-john says, if I understood him correctly, that @amirm 's DAC measurements are not valid for that use case, since the AP has much higher input/load impedance than a typical amp or active speaker.

I don't know if @amirm uses some kind of load adaptation for his measurements.

My Genelec 1032A, as an example of pro active monitors, are given for an input impedance of 10kOhm.

 

[Blumlein 88]

Actually, @restorer-john says, if I understood him correctly, that @amirm 's DAC measurements are not valid for that use case, since the AP has much higher input/load impedance than a typical amp or active speaker.

I don't know if @amirm uses some kind of load adaptation for his measurements.

My Genelec 1032A, as an example of pro active monitors, are given for an input impedance of 10kOhm.

On the other hand, if a DAC does poorly with the easy load Amir uses, using a more difficult one isn't going to improve performance. I thought Amir used 100 kohm anyway.

The old 10:1 rule being bandied about means an input of 10 kohm needs an output impedance of 1 kohm or less. Not much of a problem with all other than pathological DAC designs.

 

[restorer-john]

I don't know if @amirm uses some kind of load adaptation for his measurements.

He may, but I've seen no mention of it and all the dashboard indications are 200K ohms. It's possible the FR plot is at another load value. I'll let @amirm comment.

But I am a little surprised the RME-ADI pro is 100ohms for unbalanced. Probably a series resistor to protect the output stage in case of a short.

Here's one of my vintage preamplifier specs:

Unbalanced output impedance is 1 ohm.

Input impedance is 20K ohms to the preamp itself.

 

[Rja4000]

I am a little surprised the RME-ADI pro is 100ohms for unbalanced. Probably a series resistor to protect the output stage in case of a short

The RME outputs seem, indeed, to be well protected against short.
Even the headphone outs are.

 

[Rja4000]

all the dashboard indications are 200K ohms

100kOhm for unbalanced and 200kOhm for balanced, for what I checked.
But that's an indication of the internal load.
I don't think the AP measures any external load adaptation, like they describe here:
https://www.ap.com/technical-library/apx585586-input-impedance/

The APx555 may have difference internally-switchable input impedance, but I couldn't find any evidence of that.

 

[restorer-john]

100kOhm for unbalanced and 200kOhm for balanced, for what I checked.
But that's an indication of the internal load.
I don't think the AP measures any external load adaptation, like they describe here:
https://www.ap.com/technical-library/apx585586-input-impedance/

The APx555 may have difference internally-switchable input impedance, but I couldn't find any evidence of that.

@amirm may use the BNC T-connector method to add parallel loads but realistically, the best test would be the same output impedance test he does for HP amps. Or just do it loaded vs unloaded with a nominal parallel resistance and calculate the output Z. My guess is most D/A converters will be around 100-500 ohms, but we don't know that.

CD players with just 200 ohms output impedance (100R+100R) still specified load impedances greater than 10K and the variable (pot) outputs needed load impedances greater than 50K ohms to prevent FR anomalies. A 50:1 ratio.

Passive volume controls after medium output impedance digital sources will affect FR.

 

[Theo]

Coming back to the OP, let me rephrase the question: in the real world, do our ears detect a change (noise or distortion) in the signal when using a digital volume control?

The SNR indeed decreases (or THD rises) when decreasing the volume. I concur with this 100%, it seems indeniably correct. But is it audible?
My personal experience is that the noise floor remains the same (denominator of SNR): the signal is reduced (numerator of SNR), which seems to me the purpose of volume control! So unless the noise produced by the DAC is audible when no music is played, digital control seems harmless to me. Of course, the volume of the signal output by the DAC with no input can be increased to a point where the noise can be heard (independently from the digital volume control). If you try this experiment, I would recommend that you don't start your CD player afterwards without checking the DAC volume setting first!

Ah! Some may say: the THD rises with decreasing volume control and degrades the signal, and that may be audible! That's probably true if the CD (or whatever digital source you use) is played with the DAC volume setting at -60dB. But to be able to listen to this, the DAC output would need to be amplified significantly. But, what is the point apart from demonstrating that digital is bad?

Conclusion: digital volume is ok if properly implemented. However, a preamp (with an analog volume button) between the DAC and the amp may be a good idea as a safety precaution in case the digital volume control is set inadvertently to 0dBb before you start playing the music...

 

[Willem]

In continue to think that with a good DAC with properly implemented digital volume control a pre amplifier is a waste of money, and potentially degrades the sound quality, or at least measured performance. With good modern gear and digital sources I never hear any noise anymore, but that may be partially due to my preferece for speakers that happen to be quite inefficient, such as my Quad 2805 electrostats or my Harbeth P3ESRs (both about 83 dB).

 

[yodog]

Digital attenuation is less about loss and more about noise floor. A 16 bit stream can represent a dynamic range of 96 db, hence it has a noise floor of -96 db. If you would keep the stream at 16 bit and lower the amplitude of the signal, with each 1db of lowering you will get 1db closer to the noise floor, until at -96db your signal will be lost in the noise floor. Of course, you will hear a rise in noise way before that.

But digital volume control doesn't usually work like that. If you take a 16 bit file and stream it over a 24 bit stream, which has 120 db SNR, you can potentially lower the amplitude of the 16 bit audio by 24 db before any loss of SNR will take place. And it gets better as you improve your bit depth. At 32 bit you have potentially 192 db of SNR, so you can almost lower the 16 bit audio by its full 96 db SNR and just barely scratch the noise floor of the 32 bit stream.

You need also to consider two other factors. First is the practical SNR of your entire system, and of your own hearing. CDs were made with 16 bit because that’s way more than you realistically need to play any sort of music in any setting other than maybe an anechoic chamber. Since your base noise floor in whatever room you're are listening in is probably higher than 30 db, you don't really need even 96 db of SNR. And most systems will have less SNR, where amplifiers and mostly speakers are the limiting factor.

Dacs are usually the least limiting piece in the chain, and well-engineered dacs can easily have SNR of 100 db and more. And that’s the second factor to consider – you can only realistically use the bit depth whose SNR is lower than the SNR of your dac (called linearity sometimes). Say, if your dac has 110 SNR, you can't even use the full potential of 24 bit audio, let alone 32 bit. But that is not really a problem.

Think of it like this – say you have a base noise of 30 db SPL in your listening room (which is really low). If you set up your system to play 16 bit audio such that the quietest possible bit is at 30 db SPL, that would make the loudest possible bit (0dbFS) at 126 db SPL! Which is higher than the threshold of pain for humans, and is as loud as a jackhammer close by. You'll probably never listen to music at that volume, and I doubt most speakers can play at that volume without distorting terribly.

That means that most dacs has more than enough SNR to digitally control the volume without any perceptual noise floor. If a dac has that option built in, make sure it's converting the stream to 24 bits at least. 32 bits is the professional industry standard for audio editing, so anything beyond that is completely overkill. Nothing in the world has more than 192 db SNR anyway.

You have also clever solutions like the one in the RME ADI-2-DAC that uses digital volume control. In the default mode it has more than enough SNR to control the volume seamlessly, but if you are still concerned, you can optimize its noise floor by setting the base reference level to your liking using actual analog circuitry, to get the best SNR possible. It even has a mode in which it blends together digital and analog control to automatically achieve the lowest possible noise floor. But this is a massive overkill in most cases.

So to conclude, fear not from digital volume control. If the dac is decent and you resample your stream to 24 bit at least, it will be completely transparent.

mind blown. Thank you for this, I thoroughly enjoyed reading your post.

 

[RichB]

mind blown. Thank you for this, I thoroughly enjoyed reading your post.

Here is a presentation form ESS about their 32-bit volume control:

@restorer-john made a good point though, measurements from the AP 200 K Ohms. Stereophile measures into 100 K Ohms.
Many amps are 10 K Ohms (and vary with frequency).

The Oppo BDP-105 was not buffered so there were hardware failures when connected to some amps.
I believe the 205 is buffered and it sounds great in my game room connected to an AHB2.

- Rich

 

[thejck]

getting back to this after a short hiatus.

I got my infinite baffle subwoofer built for my HT setup and am now trying to integrate it with my 2 channel setup. What are my options to add this into a seperate stereo setup along with a nice quality DAC and power amps?

If I got something like the RME dac I could use the xlr outs to my speaker power amp. and the rca out to the sub amp? but what options to add a crossover for my sub? i was looking at the minidsp crossover and dsp solutions but would i have to add it in front of the dac or after? in which case how do i control the volume.
if its in front of the dac i can use the time delay to keep the sub phase equal to the speakers. based on location I need the phase of the sub to be slightly ahead of the speakers (sub is a short distance further than speakers). and i can use the crossover but I wont be able to use the volume control of the rme unless the crossover is after the dac.

I am not sure what my options are here unless I get a preamp or dac with a seperate sub output? any suggestions?

hope this makes sense.

in short.
built sub for HT setup ( works fine with HT receiver)
need to integrate it with stereo setup and either stereo preamp or dac.
trying to determine what dac to buy as well as how to do a crossover and phase adjust for sub.

what kinds of pieces of equipment would I need to add. I like the idea of less is more.

 

[LTig]

I got my infinite baffle subwoofer built for my HT setup and am now trying to integrate it with my 2 channel setup. What are my options to add this into a seperate stereo setup along with a nice quality DAC and power amps?

If I got something like the RME dac I could use the xlr outs to my speaker power amp. and the rca out to the sub amp? but what options to add a crossover for my sub? i was looking at the minidsp crossover and dsp solutions but would i have to add it in front of the dac or after? in which case how do i control the volume.

You could get an analog crossover like the Behringer CX3400 Super X Pro V2 or Behringer CX2310 Super X Pro V2. You feed its input with the RME or a preamp and the outputs go to the poweramps of satellites and subs.

if its in front of the dac i can use the time delay to keep the sub phase equal to the speakers. based on location I need the phase of the sub to be slightly ahead of the speakers (sub is a short distance further than speakers). and i can use the crossover but I wont be able to use the volume control of the rme unless the crossover is after the dac.

I am not sure what my options are here unless I get a preamp or dac with a seperate sub output? any suggestions?

Alternatives are digital crossovers / speaker management systems like the Behringer DCX2496 or DBX DriveRack PA2 or similar. Most of those have delays and additional graphic and/or parametric EQs which could be used for room eq as well. Some also have a digital input which could be fed by the RME so you prevent one additional DA/AD conversion. Of course the quality of the DACs in those crossovers are not as good as the RME but probably good enough.

 

[thejck]

ok maybe i dont understand phase and time delay.
If my L+R speaker is perfectly even from my main listening position and my subs are about 3-5 feet further away from my listening position do I need to be concerned with time delay? I am planning on setting a crossover for my subs to be below 40-50hz.
Do i need to care to delay the fronts to line up with the subs or at that sub 50hz frequency you cant even tell the difference.

is this time delay I am referring too any different from phase?

Do all I need to set this up is just a crossover to only pass the low frequencies to the sub and some soft or dsp or eq for the whole frequency range to kill any peaks?

 

[Killingbeans]

Take the industry darling NC-500 (or equiv) with its unbuffered 1.8K ohm per input and tell me a typical D/A converter is going to perform well into such an impedance. And yet people are plugging sources into these modules without using an outboard buffer in some cases.

I wouldn't base my decisions on the moronic actions of other people. NC500MP is 47K ohm, and NC400 is 104K ohm