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Does Phase Distortion/Shift Matter in Audio? (no*)

You are assuming that the impulse is band limited therefore a band limited signal can create it.
I'm not quite sure what you mean here. You can get a digital unit impulse from a band limited continuous time impulse. A normalized sinc function is band limited to half the sample rate and if you sample it you get a digital unit impulse. Why? Because the samples all fall exactly on the zero crossings except for one single sample at x=0. The caveat is that the normalized sinc does contain frequencies up to and including Fs/2 (but not above Fs/2), so it doesn't strictly obey the band limit imposed by the sampling theorem.

"Approximately?"
Yes, because I was allowing for noise, jitter, etc. in a real ADC as well as a non-ideal input signal. If you ignore that and put an impulse through the described finite filter, the result is exact.
 
Mod note: moved this from another thread as it could use some expert review…

Unfortunately, I couldn't find any information from WIIM itself about the “Volume Limit” setting. The manual simply states that this setting exists. If anyone here can provide more first-hand information from WIIM, that would be greatly appreciated.

Regarding my Pro Plus, to avoid any misunderstanding, the device does not appear to be faulty. I will know for sure in two or three weeks because I am meeting up with friends in the country, two of whom have the same Pro Plus. We will compare it directly with my device in every respect.

We can also rule out analog clipping in the various preamps.

I have now also connected my EAR 864, which can also handle high input voltages, and the picture is the same. Furthermore, analog clipping sounds completely different to my ears.

Everything that follows refers to the use of the analog RCA outputs with the internal DAC.

What I'm experiencing is that the Pro Plus simply doesn't sound good with the parameters set to normal – EQ off, room correction off, analog output 2V (I also tried it with 1V output and it sounds equally unsatisfactory), pre gain neutral, volume limit 100%, volume in the range between 60% and 90%.

When I heard the Pro Plus for the first time, I thought it had a very nice, pleasing and appealing sound. But it was somehow very strangely compressed, with a certain sharpness and a loss of spatiality. My ears tired quickly and I was on the verge of returning the Pro Plus for another brand.

On the last evening before returning it, I played around with the volume limiter. I hadn't read anything about this setting and its effect yet. When I reduced the voltage range by 5% = -3 dB, all these faulty sounds disappeared. Incidentally, not many people seem to understand how this volume limiter works. At least I didn't find any forum members who obviously had any knowledge of it (at the time). I will continue to post under the link to the other forum.

If I understand correctly, the volume limiter reduces the digital signal after the PEQs etc. but before the analog filters. So exactly at the point in the circuit where @totti1965 talks about “inter sampling over.” And I think he's hit the nail on the head. It's great that I can set the Pro Plus to make me very happy. But it would be disastrous if 99% of all users never thought of using the volume limit sensibly, especially since Wiim doesn't say anything about it. Users can understand that you set a limit if, for example, you exceed the headroom with a 9 dB bass boost in the EQ. But hardly anyone will ever know that the volume limiter is indispensable for a really good sound, even when the EQ is turned off. Once again, without volume limitation, the Pro Plus doesn't sound damaged, but only with volume limitation does it sound exceptionally good.

It concerns the problem described here:


And here dBFS is described very well; without this, it's impossible to understand what it's all about.


Here is finally the link to the other forum that I promised above. There is a poster there who explains the issue really well. His name is “onlyoneme.”


It's only six pages... but they're really worth reading. All the nonsense there talking about bit perfect isn't important to me, though; it just shows that some people don't understand what this thread is about. Incidentally, the video explains this problem very well too.

 
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Unfortunately, I couldn't find any information from WIIM itself about the “Volume Limit” setting. The manual simply states that this setting exists. If anyone here can provide more first-hand information from WIIM, that would be greatly appreciated.

Regarding my Pro Plus, to avoid any misunderstanding, the device does not appear to be faulty. I will know for sure in two or three weeks because I am meeting up with friends in the country, two of whom have the same Pro Plus. We will compare it directly with my device in every respect.

We can also rule out analog clipping in the various preamps.

I have now also connected my EAR 864, which can also handle high input voltages, and the picture is the same. Furthermore, analog clipping sounds completely different to my ears.

Everything that follows refers to the use of the analog RCA outputs with the internal DAC.

What I'm experiencing is that the Pro Plus simply doesn't sound good with the parameters set to normal – EQ off, room correction off, analog output 2V (I also tried it with 1V output and it sounds equally unsatisfactory), pre gain neutral, volume limit 100%, volume in the range between 60% and 90%.

When I heard the Pro Plus for the first time, I thought it had a very nice, pleasing and appealing sound. But it was somehow very strangely compressed, with a certain sharpness and a loss of spatiality. My ears tired quickly and I was on the verge of returning the Pro Plus for another brand.

On the last evening before returning it, I played around with the volume limiter. I hadn't read anything about this setting and its effect yet. When I reduced the voltage range by 5% = -3 dB, all these faulty sounds disappeared. Incidentally, not many people seem to understand how this volume limiter works. At least I didn't find any forum members who obviously had any knowledge of it (at the time). I will continue to post under the link to the other forum.

If I understand correctly, the volume limiter reduces the digital signal after the PEQs etc. but before the analog filters. So exactly at the point in the circuit where @totti1965 talks about “inter sampling over.” And I think he's hit the nail on the head. It's great that I can set the Pro Plus to make me very happy. But it would be disastrous if 99% of all users never thought of using the volume limit sensibly, especially since Wiim doesn't say anything about it. Users can understand that you set a limit if, for example, you exceed the headroom with a 9 dB bass boost in the EQ. But hardly anyone will ever know that the volume limiter is indispensable for a really good sound, even when the EQ is turned off. Once again, without volume limitation, the Pro Plus doesn't sound damaged, but only with volume limitation does it sound exceptionally good.

It concerns the problem described here:


And here dBFS is described very well; without this, it's impossible to understand what it's all about.


Here is finally the link to the other forum that I promised above. There is a poster there who explains the issue really well. His name is “onlyoneme.”


It's only six pages... but they're really worth reading. All the nonsense there talking about bit perfect isn't important to me, though; it just shows that some people don't understand what this thread is about. Incidentally, the video explains this problem very well too.

I tried to digest all that information but most of them is impossible to verify to me. Having various DACs, I mean.

Pro Plus has not a state of the art DAC but should be enough except very quiet passages in which if you’re in total silence and have very good ears you can listen some noise.

That you describe is more like a bad unit or just some personal psychoacoustic phenmenon, ruled out clipping.

Something like horns hurting your ears lies more on the clipping or saturation side, a medium DAC is perfectly flat and wont distinguish a horn from an electric guitar, only bits and voltages
:)

You can also consider what I did, demand for an exchange if you can, and verify if second unit sound ok.

The only case in which I ever sound a poor quality sound that can attribute to a line signal was some dongles, always confimed by Amir measurements of audible HD at high volumes.

For the current dynamic range of music, Pro Plus is fine, and has the advantage of very confortable software.

You have better scores on the Ultra model, but I have doubts 1% higher petcentile of audiophiles can tell any difference
 
But it would be disastrous if 99% of all users never thought of using the volume limit sensibly, especially since Wiim doesn't say anything about it.
This doesn't happens, first all DACs are tested at their 0dBFS and they behave perfectly linear, if your statements were true instruments will detect them.

I have my WiiM on the 80% max, but that allows me to fine tune the volume.

Consider that this can be the key to understand what you're experiencing: if you usually remain between 25- 45 % of volume (usually my case) chances are you cannot find the good loudness to your taste: 30 can be to quiet and 32 can be bothering.

Allowing your range to operate between 20- 70 allows us to better "feel" the volume.

Also take into account that recordings are normalized in such a way that even played at 0dBFS (which is consider the only way to play bit perfect) they don't reach full scale.

0dBFS will be the highest value attained by the album, and usually editors let a margin.

EDIT: oh, I turned up and down the max volume setting, I can listen some slight bass reduction as a subtle change on the FB slope.

That can means (or not) that the reducer acts in the voltage of the output, which changes slightly the FB curve but in an acceptable way (smoothly).

You can have the same impression if you go from 2V to 1V output. This makes more sense because the voltage reduction leave intact the bits dedicated to digital volume avoiding eventual digital degradation when playing too low in the digital domain.
 
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Mod note: moved this from another thread as it could use some expert review…

Unfortunately, I couldn't find any information from WIIM itself about the “Volume Limit” setting. The manual simply states that this setting exists. If anyone here can provide more first-hand information from WIIM, that would be greatly appreciated.

Regarding my Pro Plus, to avoid any misunderstanding, the device does not appear to be faulty. I will know for sure in two or three weeks because I am meeting up with friends in the country, two of whom have the same Pro Plus. We will compare it directly with my device in every respect.

We can also rule out analog clipping in the various preamps.

I have now also connected my EAR 864, which can also handle high input voltages, and the picture is the same. Furthermore, analog clipping sounds completely different to my ears.

Everything that follows refers to the use of the analog RCA outputs with the internal DAC.

What I'm experiencing is that the Pro Plus simply doesn't sound good with the parameters set to normal – EQ off, room correction off, analog output 2V (I also tried it with 1V output and it sounds equally unsatisfactory), pre gain neutral, volume limit 100%, volume in the range between 60% and 90%.

When I heard the Pro Plus for the first time, I thought it had a very nice, pleasing and appealing sound. But it was somehow very strangely compressed, with a certain sharpness and a loss of spatiality. My ears tired quickly and I was on the verge of returning the Pro Plus for another brand.

On the last evening before returning it, I played around with the volume limiter. I hadn't read anything about this setting and its effect yet. When I reduced the voltage range by 5% = -3 dB, all these faulty sounds disappeared. Incidentally, not many people seem to understand how this volume limiter works. At least I didn't find any forum members who obviously had any knowledge of it (at the time). I will continue to post under the link to the other forum.

If I understand correctly, the volume limiter reduces the digital signal after the PEQs etc. but before the analog filters. So exactly at the point in the circuit where @totti1965 talks about “inter sampling over.” And I think he's hit the nail on the head. It's great that I can set the Pro Plus to make me very happy. But it would be disastrous if 99% of all users never thought of using the volume limit sensibly, especially since Wiim doesn't say anything about it. Users can understand that you set a limit if, for example, you exceed the headroom with a 9 dB bass boost in the EQ. But hardly anyone will ever know that the volume limiter is indispensable for a really good sound, even when the EQ is turned off. Once again, without volume limitation, the Pro Plus doesn't sound damaged, but only with volume limitation does it sound exceptionally good.

It concerns the problem described here:


And here dBFS is described very well; without this, it's impossible to understand what it's all about.


Here is finally the link to the other forum that I promised above. There is a poster there who explains the issue really well. His name is “onlyoneme.”


It's only six pages... but they're really worth reading. All the nonsense there talking about bit perfect isn't important to me, though; it just shows that some people don't understand what this thread is about. Incidentally, the video explains this problem very well too.


Hallo Heinz,

It was exactly this Hans Beekhuizen Video I wanted to show you, after I have begun to read your post!

Hans is not so well regarded in the community here, but in this case, he may be right.

@Miguelon: the DAC in the WiiM is State of the Art. You can buy a US$ 60.000 Streamer and it will not sound better.
It is that simple: The WiiM Pro Plus is not „good for the money“ that would be something typical YouTube Bullshitter would tell you - it is perfect on an absolute scale!
I have had many discussions with friends about that topic!
WiiM in 2025 is now something as Sonos was 10 Years ago.
Perfect Hardware, monthly updated by a team of 70 engineers, who know exactly what they do.
Am I saying this, because I am an „objectivist“ or can’t afford a better streamer???

No, I own at least four Streamer, also a pimped Linn Majik and a heavily upgraded Sonos, each in the EUR 2.000,— range.
The „Subjectivist“ in me compared the WiiM Pro Plus (-114 dB) with an even better measured SMSL D6s (SINAD -120 dB) driven digitally by the WiiM.
The WiiM with his own AKM D/A Converter sounds better in my ears.
So in 2025 I use the WiiM and not the expensive stuff.
The WiiM simply works: fast, risposive, perfect usability with roon, and also I can see my beloved youtubevideos from @amirm or @crinacle over my Loudspeakers, without any dropouts or problems. Nothing, Linn or Sonos would be able to do (at least not that flawless!).
 
It's kind of funny how important phase is to speakers in general, other than when it comes down to the sound from a single speaker.

In large-scale live sound systems, phase control is important to all the major's line-array deployments; especially with beam steering systems like in Martin's MLA and EAW's Anya. Many/most manufactures work to keep the phase traces of their various sized models to be as similar as possible despite the varying number of ways employed in each type box, as it allows the main speakers and all the different size fill speakers, to blend smoothly together. Time and phase alignment of subs to mains is a huge goal at all levels of live production, even from smaller PA operators.

In home, we know the phase response of L&R needs to be the same for optimal stereo. It's easy to wonk up stereo by overlaying all-pass filters on just one side.
Muti-sub works with phase. So do many room correction products.

So looking broadly at audio, it seems to me that phase is surely important whenever we are concerned with multiple sources, be they two or more speakers, or simply the drivers in a single speaker.

The most ironic thing about phase to me is on a finished single speaker.
I know how important it is in the speaker design to get the phase and time relationships correct between drivers.
Phase trace overlay between drivers' summation regions is necessary for producing the best spinorama results possible (focusing only on frequency magnitude response.)
Time alignment of drivers' individual measured impulse responses, is also necessary for optimal results.

But once those phase and time alignments are in place to get best spins possible, phase is deemed no longer an issue, as it is truly hard to hear whatever amount of phase rotation (or lack of rotation) went into the speaker's alignment.

I think debate about linear phase vs IIR low order vs IIR high order, etc...is a bit of a shame because it masks how fundamental to audio, phase and time are.
I look forward to the day when hard science exists to show how many degrees of phase rotation per bandwidth span per bandwidth location, it takes to be audible or not.
 
Unfortunately, I couldn't find any information from WIIM itself about the “Volume Limit” setting. The manual simply states that this setting exists. If anyone here can provide more first-hand information from WIIM, that would be greatly appreciated.
You open the WiiM app, select your device and open the settings dialog. Activate "fixed volume" and set the max. volume to 90-95%. Even if the WiiM app now shows always 100% volume, it is limited to the max setting. This is an error within the app.

I am using Tidal and a lot of streams do indeed clip at 100% volume. Therefore I am using Tidal with 95% volume setting only.

Stupid loudness war is - and remains - a mess...
 
It's kind of funny how important phase is to speakers in general, other than when it comes down to the sound from a single speaker.
That is not the topic at all. See below.
In large-scale live sound systems, phase control is important to all the major's line-array deployments; especially with beam steering systems like in Martin's MLA and EAW's Anya. Many/most manufactures work to keep the phase traces of their various sized models to be as similar as possible despite the varying number of ways employed in each type box, as it allows the main speakers and all the different size fill speakers, to blend smoothly together. Time and phase alignment of subs to mains is a huge goal at all levels of live production, even from smaller PA operators.
Phase is of critical importance in design of speakers of all kinds. Not because phase by itself is important but for other roles it plays. In the case of speaker arrays, beam forming relies on that to accomplish its said goal (narrowing and controlling beam width). You don't even have to go to such an exotic example. Even the design of a 2-way speaker involves understanding of phase as it is the complex sum of the two drivers which determines the frequency response.

What we are discussing is not any of that. In the above examples, if you don't get the phase right, the other parameter we care about, beam control and frequency response become distorted.

The discussion here is even if those things are done right, whether you should care about phase correctness *in your room* as you listen to speaker. Answer to that is that it is not.
In home, we know the phase response of L&R needs to be the same for optimal stereo.
We don't "know" that. I am confident that the phase you are hearing from two speakers across full frequency range absolutely does not match as it arrives at your hear. You are never seating locked in a vise, nor is there any precision in how speakers are designed and placed to accomplish that.
Muti-sub works with phase. So do many room correction products.
Again, that is to accomplish another goal: flatness of frequency response. No one is disputing the role of phase in that regard. Get the frequency response right and then see if you can argue whether preference is correlated with phase.
 
Mod note: moved this from another thread as it could use some expert review…

Unfortunately, I couldn't find any information from WIIM itself about the “Volume Limit” setting. The manual simply states that this setting exists. If anyone here can provide more first-hand information from WIIM, that would be greatly appreciated.

Regarding my Pro Plus, to avoid any misunderstanding, the device does not appear to be faulty. I will know for sure in two or three weeks because I am meeting up with friends in the country, two of whom have the same Pro Plus. We will compare it directly with my device in every respect.

We can also rule out analog clipping in the various preamps.

I have now also connected my EAR 864, which can also handle high input voltages, and the picture is the same. Furthermore, analog clipping sounds completely different to my ears.

Everything that follows refers to the use of the analog RCA outputs with the internal DAC.

What I'm experiencing is that the Pro Plus simply doesn't sound good with the parameters set to normal – EQ off, room correction off, analog output 2V (I also tried it with 1V output and it sounds equally unsatisfactory), pre gain neutral, volume limit 100%, volume in the range between 60% and 90%.

When I heard the Pro Plus for the first time, I thought it had a very nice, pleasing and appealing sound. But it was somehow very strangely compressed, with a certain sharpness and a loss of spatiality. My ears tired quickly and I was on the verge of returning the Pro Plus for another brand.

On the last evening before returning it, I played around with the volume limiter. I hadn't read anything about this setting and its effect yet. When I reduced the voltage range by 5% = -3 dB, all these faulty sounds disappeared. Incidentally, not many people seem to understand how this volume limiter works. At least I didn't find any forum members who obviously had any knowledge of it (at the time). I will continue to post under the link to the other forum.

If I understand correctly, the volume limiter reduces the digital signal after the PEQs etc. but before the analog filters. So exactly at the point in the circuit where @totti1965 talks about “inter sampling over.” And I think he's hit the nail on the head. It's great that I can set the Pro Plus to make me very happy. But it would be disastrous if 99% of all users never thought of using the volume limit sensibly, especially since Wiim doesn't say anything about it. Users can understand that you set a limit if, for example, you exceed the headroom with a 9 dB bass boost in the EQ. But hardly anyone will ever know that the volume limiter is indispensable for a really good sound, even when the EQ is turned off. Once again, without volume limitation, the Pro Plus doesn't sound damaged, but only with volume limitation does it sound exceptionally good.
Easy test of your theory: replace your Wiim with some other DAC and see if all the problems return when playing, say, a CD. If your theory is right, they should return, because all normal DACs don't have the 'volume limiter solution' available to them.

I suspect that the problems won't return, and that your theory is wrong.

By far the best way to check if your Wiim is faulty is with a signal analyser, comparing the output and distortion when varying the Volume Limiter settings. Doing it by ear, "what the audio sounds like" compared to other Wiim Pro Plus owners' units when they visit you with their Wiims, is fraught.

cheers
 
That is not the topic at all. See below.
Phase is of critical importance in design of speakers of all kinds. Not because phase by itself is important but for other roles it plays. In the case of speaker arrays, beam forming relies on that to accomplish its said goal (narrowing and controlling beam width). You don't even have to go to such an exotic example. Even the design of a 2-way speaker involves understanding of phase as it is the complex sum of the two drivers which determines the frequency response.

What we are discussing is not any of that. In the above examples, if you don't get the phase right, the other parameter we care about, beam control and frequency response become distorted.

Agreed. I was simply saying how the stark contrast of the importance of phase depending on what we are talking about, strikes me as funny.

The discussion here is even if those things are done right, whether you should care about phase correctness *in your room* as you listen to speaker. Answer to that is that it is not.
Disagree. Proper tests have yet to be done imo. We need tests like jj was talking about doing.
Besides, what is my preferred ratio of direct to reflected, in my room?

We don't "know" that. I am confident that the phase you are hearing from two speakers across full frequency range absolutely does not match as it arrives at your hear. You are never seating locked in a vise, nor is there any precision in how speakers are designed and placed to accomplish that.
I think as a principle that we do know that phase of speakers needs to match for optimal stereo.
I'm also equally confident that the direct sound I get from two speakers does have phase matching across the full frequency range, at least from the mid range up.
For direct sound, changing listening positions is about time alignment differences between the two isn't it, and not phase alignment.
If listening position has unequal time arrivals, the phase of each speaker will still be the same for each speaker, but the time alignment will be off. Time and phase, different animals.

Again, that is to accomplish another goal: flatness of frequency response. No one is disputing the role of phase in that regard. Get the frequency response right and then see if you can argue whether preference is correlated with phase.
Agree, First goal is surely to get the frequency magnitude response right. Then and only then can we assess the audibility of phase.

Personally disagree on the preference part though....I don't think preferences belong in audio science.
I see preferences to be inside a behavioral science; and expect preferences to continually change, as real audio science and products, continue to evolve.
 
Disagree. Proper tests have yet to be done imo.
You can't disagree using data you don't have. For 50 years, Dr. Toole and crew did not consider something useful to go after. The few others who researched it said it is only audible with specialized signals and even there, its value for preference can't be established.

Given the above, the best position you can take is neutral and not advocate for it especially since such phase "accuracy" (whatever that means) likely won't come for free.

Personally disagree on the preference part though....I don't think preferences belong in audio science.
??? That is how we determined what frequency response we should aim for in speakers. There are many compromises in speaker design and we need to be clear on what brings the most listener satisfaction, and what not.
 
You can't disagree using data you don't have. For 50 years, Dr. Toole and crew did not consider something useful to go after. The few others who researched it said it is only audible with specialized signals and even there, its value for preference can't be established.

Given the above, the best position you can take is neutral and not advocate for it...
Indeed, it's the best, most rational and science-based position to take. The whole "remember Einstein" response, in resisting today's best science-based working conclusion, is an attitude that reeks of science denialism.

There are a number of members here who use best available science to educate on audio matters when they agree with it, which I applaud, then backflip and enter blatant science denialism and all its tactical trickery whenever they find the current consensus hypothesis to be inconvenient. THAT I don't applaud.

cheers
 
...When I reduced the voltage range by 5% = -3 dB, ...
Voltage reduction by 5% is -0.45 dB.
I'm guessing it's a proprietary scale called "Volume Limit" and it's not the voltage that is reduced by 5%, but the total volume range, and the company makes its own decision on how to scale it. I assumed that Heinz measured it.
 
I look forward to the day when hard science exists to show how many degrees of phase rotation per bandwidth span per bandwidth location, it takes to be audible or not.
There is no interest from the industry (and thus AES etc) to show audibility of phase distortion.

But we are in the lucky position to create dependable evidence ourselves with quite little effort, relatively speaking:
  • Take an existing decent setup (good speakers, good room with proper balance of absorption and diffusion, of course acoustical symmetry)
  • Apply professional DRC (like Acourate etc) to it at the LP, to create flat (to target) mag response and min-phase'ish phase response
  • Make long-term AB comparisons (and then ABX tests once the learning curve has settled), with introduced global phase distortion like what regular speaker XO would introduce (same on both channels, of course), listening for a number of effects like timbre change in the bass, speed of bass transients, soundstage aspects like perceived size and sharpness of phantom sources), etc etc
  • Draw your conclusions
 
Voltage reduction by 5% is -0.45 dB.
Now my post was moved to this section which is more a general threat about phase shift, etc
Nevertheless, I want to reply on how the volume limit works and this threat linked below explains it very well by @onlyoneme
- each 5% less means 3dB less.

 
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I was not thinking in more expensive DACs but in the top SINAD classification that Amir shows in his list.

Many of the top tier are around 300$.
The Wiim is (audibly) equally top tier - at least in the Pro+ and up variants.
 
The Wiim is (audibly) equally top tier - at least in the Pro+ and up variants.
Nothing to complain at audibly level by my needs and capacities, but numbers are what they are.

123 SINAD of Topping D90 III is better than 113 WiiM Pro Plus, what is audibly or not I don't know.

I found diverse answers in ASR, some of them arguing imperceptible over 120 SINAD and other enough over 110.
 
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