Should phantom power be supplied by a linear supply?

[My3gger]

Extra tap with shield/insulation is very cheap here, at least for most stuff we do here. I rather have extra tap than lytics for linear supply, although modern ones would probably last a very long time at low ripple current and proper voltage (i buy ones with 1/3 more V).

 

[rjb5191]

Played with those a little bit last night. First thing that threw me off was that the original files are 32bit float format dumped from a DAW. The older software (Audio DiffMaker) didn't recognize that at all, DeltaWave did, but didn't play nicely with my audio interface (seemed to want to send 32 bit audio to the interface, even though it should be advertising it is a 24 bit device).
Eventually got them both working well enough, and DeltaWave seems to do a better job time aligning. Even after time and amplitude aligning, the best null achieved was about -55-ish dB.

Listening to the amplified difference there did not seem to be noticeable distortion, frequency response changes, etc., so my preliminary guess was that the playback was started at slightly different points in the original source, that some amount of compression was still being applied, and that the interaction between attack and release time of the compressor meant that there were very slight differences in gain at each point in time in the resulting files.
Of course if rjb5191 says that the compressor was off, and the unit was only being used as a buffer amp, then that hypothesis would be right out the window.

I did listen to the files, of course, but I could not hear the differences rjb5191 described.
If anyone wants to do some blind checks to see if you can identify the files without knowing what you are listening to this ABX playback and testing tool is handy:
Lacinato ABX software
tutorial on using Lacinato ABX

Thank you for doing this. It's enlightening. The compressor was on.

 

[abbey road d enfer]

The compressor was on.

AAARGH! All this discussion for figuring out operator's mistake...

 

[user 37518]

AAARGH! All this discussion fo figuring out operator's mistake...

It happens, has happened to me more than once, at least good ideas were exchanged.

 

[rjb5191]

AAARGH! All this discussion for figuring out operator's mistake...

I didn't really think about the timing issue with the compressor on. I thought it would be better to have it on as any differences with the power supplies could be more noticeable with the compression on. That was my reasoning anyways.

 

[JohnRoberts]

I didn't really think about the timing issue with the compressor on. I thought it would be better to have it on as any differences with the power supplies could be more noticeable with the compression on. That was my reasoning anyways.

too many variables...

JR

 

[ccaudle]

I didn't really think about the timing issue with the compressor on.

That (indeterminate timing of gain changes) only affects a nulling test.

any differences with the power supplies could be more noticeable with the compression on

In general that is valid reasoning, it is possible that power supply noise or ripple could affect the behavior of the compression in some way, and that is how the device will be used, so better to test it how it will be used.

Did you look at the ABX software I linked? That lets you load the two files you provided, and switch back and forth between the two while listening. If you are convinced you can hear a difference between the two, then you click the test button, and the program will play a file for you, and you pick whether you think it is the first or second file. You have a 50% chance of getting it right even if you just guess or flip a coin, so you do that multiple times, maybe a dozen or 15, and then the program will calculate how often you identified correctly.

There are a lot of ways that two audio files can sound different, even if the difference isn't really relevant. Slight level differences are the most tricky to catch, because even a 0.1dB level difference can be heard as a difference in a good listening environment.
A timing difference can also be heard if you are switching back and forth between two. I think the two files you provided are not quite exactly aligned, but that isn't so likely to matter if you are listening to the unlabeled sample all the way through and trying to identify. If you were listening to the unlabeled sample, and switching between that and the two reference files you might notice a slight glitch as you switched to the file which wasn't perfectly lined up, and catch that as a difference.

 

[abbey road d enfer]

I thought it would be better to have it on as any differences with the power supplies could be more noticeable with the compression on.

With the compressor on, there is a difference, whatever the power supply. You can't make an analog compressor (as well as any analog circuit) behave exactly the same every time.

 

[Somkereki]

First thing I did when I got my MOTU M4 was take it apart and look at how it was designed. It uses three switching power supplies to step USB 5V to + and - audio rails and a 48V phantom rail. It's a studio grade interface from a prestigious manufacturer. I don't perceive any audible noise when phantom is turned on, even at high gain.

It's important to note that even if switching noise makes it into the 48V rail, as long as the 6.8k feed resistors (as well as other circuit impedances) are well-matched, that noise will be common-mode and should be cancelled out in the preamp.

Mismatches in other circuit impedances will manifest in more ways than just phantom power noise.

Phantom-powered devices generally have fairly low output impedances in the audible range (<100 Ohm), and should be pretty capable of shunting away any audible noise from the phantom supply. The 6.8k feed resistor and the device's output impedance make a voltage divider that should attenuate any noise voltage quite a bit.

If you're really worried you can follow up a switching regulator with a linear post-regulator with good PSRR, and/or a capacitance multiplier.

Hi ! My experience is that devices with SMPS have a much more unpleasant sound! For example, the jitter of the clock signal deteriorates measurably, interference occurs during the z A-D conversion, etc. I never use an SMPS power supply in studio technology! My instruments: Neutrik A2-D, Tektronix TDS224 oscilloscope, but if I'm looking for noise I use the "good old" Tektronix 465 - five. (analog) By the way, the Neutrik A2-D also has an analog power supply!!! But my DUNE HD media player, which I use for listening to music at home, also sounds better with an analog power supply than with the supplied SMPS! I am an electrical engineer and was a sound engineer when I was younger.

 

[abbey road d enfer]

Hi ! My experience is that devices with SMPS have a much more unpleasant sound! For example, the jitter of the clock signal deteriorates measurably, interference occurs during the z A-D conversion, etc.

These smps and the equipment they were in must have been very poorly designed. Could you measure how much jitter increased? It's been shown times and times again that it takes enormous jitter to actually impair audio quality.

I never use an SMPS power supply in studio technology!

It must take a lot of thinking avoiding smps in a studio.
First, no computer. That seriously restricts the possibilities in terms of publishing mixes. Only tapes and vinyls.
Let me see, in my cabin I have one in the computer, two in the screens, three in the monitors (2.1), one in the mixer, two in the hardware reverbs, one in the CD player, even some of the lightbulbs have smps.
I wonder how I can produce anything with this doomed environment...

 

[ccaudle]

It's been shown times and times again that it takes enormous jitter to actually impair audio quality.

Since jitter results in a modulation effect, the audibility depends not only on amplitude, but spectrum of the jitter, as well as the transfer function of clock input to audio output of the converter (some converter topologies are more sensitive to jitter than others, with switched capacitor type circuits being noted as less sensitive to jitter).

In general I agree with your sentiment, but I learned long ago that you should not overestimate the engineering competence of audio companies, especially if they are marketed as "audiophile" boutique brands. At least the consumer companies, the pro focused brands that got lumped in with audiophile labels like Weiss and Prism seemed to have pretty good measurements.

 

[abbey road d enfer]

Since jitter results in a modulation effect, the audibility depends not only on amplitude, but spectrum of the jitter, as well as the transfer function of clock input to audio output of the converter (some converter topologies are more sensitive to jitter than others,

I have long lost interest for the sterile discussions around jitter audibility. I just remember that, in order to produce significant effects, enormous amounts of jitter had to be accounted for, much more than anything that could happen in a half-decently designed digital chain.

with switched capacitor type circuits being noted as less sensitive to jitter).

I have no idea why it would be so...

In general I agree with your sentiment, but I learned long ago that you should not overestimate the engineering competence of audio companies, especially if they are marketed as "audiophile" boutique brands.

I certainly do not underestimate their marketing ingenuity, which results in shedding $5k for a rubidium clock that's used to drive a PLL.

 

[Ilya]

I certainly do not underestimate their marketing ingenuity, which results in shedding $5k for a rubidium clock that's used to drive a PLL.

That's because of the word "rubidium". This single word adds approximately 4k to the price)

 

[ccaudle]

I have no idea why it would be so... [switched capacitor converters less sensitive to jitter]

Most switched capacitor filters are designed with a little margin, so that all of the charge transfers from one capacitor stage to the next in less than the specified clock cycle time. If the clock cycle is marginally shorter or longer than nominal it does not really affect the charge transferred, which is what eventually determines the output voltage.

Wow, this is way off topic. I can barely remember how we got from discussing phantom power supply to obscure details of DAC design.

 

[john12ax7]

Since jitter results in a modulation effect, the audibility depends not only on amplitude, but spectrum of the jitter, as well as the transfer function of clock input to audio output of the converter (some converter topologies are more sensitive to jitter than others, with switched capacitor type circuits being noted as less sensitive to jitter).

I think the idea of jitter spectrum causing audible changes has some merit. If you listen to different clock and converter combos you will hear a difference.

 

[abbey road d enfer]

I think the idea of jitter spectrum causing audible changes has some merit. If you listen to different clock and converter combos you will hear a difference.

I certainly don't doubt that different converters may sound different. But all the attempts at making me believe that clock X resulted in better sound that clock Y have failed to convince me, as long as the system was properly locked.

 

[Ilya]

There’s a widespread myth (local?), that Yamaha CL (and older M7CL) consoles sound better, when externally clocked. I’ve never conducted any tests of this in person though.

 

[abbey road d enfer]

There’s a widespread myth (local?), that Yamaha CL (and older M7CL) consoles sound better, when externally clocked. I’ve never conducted any tests of this in person though.

I heard that a lot, not only for Yamahas. Unless the internal clock is extremely bad, the most significant difference is that using an external clock results in increased jitter.
Someone suggested that the effect of such jitter would be kind of similar to dithering, which some may find pleasant.

 

[john12ax7]

I certainly don't doubt that different converters may sound different. But all the attempts at making me believe that clock X resulted in better sound that clock Y have failed to convince me, as long as the system was properly locked.

I mean in the context of the same converter. Some converters are more susceptible than others to charges in sound when using different clocks. We can debate whether there is an improvement, but there is often an audible difference. This shouldn't happen if it was simply a matter of only very large amounts of magnitude of jitter making a difference.

 

[abbey road d enfer]

Some converters are more susceptible than others to charges in sound when using different clocks. We can debate whether there is an improvement, but there is often an audible difference.

I've been invited at several of these clock shootouts and none of them was conducted with the necessary level of discipline. Too much cognitive bias.
I'm not saying it's the case everywhere, but this is my experience.
I'm not terribly interested in this kind of experiment, since, even if actual differences appear, there will be a debate as to which is best.
As we say here, let's not sodomize diptera.