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WiiM Amp Ultra Streaming Amplifier

Rate this streaming amplifier:

  • 1. Poor (headless panther)

    Votes: 3 1.0%
  • 2. Not terrible (postman panther)

    Votes: 8 2.6%
  • 3. Fine (happy panther)

    Votes: 66 21.6%
  • 4. Great (golfing panther)

    Votes: 228 74.8%

  • Total voters
    305
Also synchronising devices is completely different from keeping the overall latency low. Try playing an electrical instrument through your superior Sonos gear.and report back.
That's exactly why I don't like consumer DSP solutions. Too many variables and before you know it, you'll end up with 50-100ms latency which is fine for just listening, but renders the whole thing unusable for any musical application. Imagine pressing a key on a synth and the actual sound is 100ms late. Hilariously bad and worse than the PC-based software instrument situation of 1999. :D

By stark contrast, professional DSP-based systems like Neumann active speakers manage a system latency of 2-3ms. Completely different ballpark.
 
…Using this method, the Room EQ filters are always enabled and actually adress room modes and you can change the target frequency response with EQ presets. Having used other Room Correction systems before, you really get the best spacial results when your speakers are time-aligned and phase corrected. But at least cutting out those boomy, bloaty standing waves makes everything sound clear and fast.
Hence why I keep hammering the point about DIRAC Live being in an entire different league compared to RoomFit.
 
That's exactly why I don't like consumer DSP solutions. Too many variables and before you know it, you'll end up with 50-100ms latency which is fine for just listening, ...
Well, that's exactly what consumer electronics are designed for, no? ;)

Hence why I keep hammering the point about DIRAC Live being in an entire different league compared to RoomFit.
Unfortunately, hammering and making sense are often not affiliated. Time alignment of individual drivers can be helpful, especially at higher frequencies. Time alignment and phase correction of a complete loudspeaker in a frequency range, where is mostly behaves as a minimum phase system ist vastly overestimated.

And no, by that I don't mean that RoomFit was better or the best or even perfect.
 
My question remains: why doesn’t someone invent a solution. A variable delay can’t add much cost, because Sonos has it in products that have a line in.
Latency reporting between devices was already standardised in eARC, but some manufacturers implement it better than others. Presumably Sonos is one of the good ones.

Similarly audio processing pipelines like PipeWire in linux allow each step in the pipeline to report latency. That makes it possible for AV applications to compensate for the difference in audio and video latency to keep things in sync. Some applications support this, but others don't. I think there's some latency reporting in recent bluetooth audio standards too.

Unfortunately the legacy formats like toslink don't have any way to report latency so the best you can do is a manual adjustment to compensate.
 
Well, that's exactly what consumer electronics are designed for, no? ;)


Unfortunately, hammering and making sense are often not affiliated. Time alignment of individual drivers can be helpful, especially at higher frequencies. Time alignment and phase correction of a complete loudspeaker in a frequency range, where is mostly behaves as a minimum phase system ist vastly overestimated.

And no, by that I don't mean that RoomFit was better or the best or even perfect.
Generally speaking, my advice is to keep room correction to no more than twice your room’s (estimated) Schroeder frequency, regardless of algorithm used anyway. And even in that correction range, DIRAC trounces RoomFit. Of course it could be worse…the room correction in Eversolo streamers is a baby rattle compared to RoomFit.
 
Hi Marti,

Did you ever write up your room correctiion process using REW as I also have the Wiim Amp Ultra (and a minidsp umik-1) and would like to follow your procedure. If you have not I would really appreciate just a few bullet points of the overall process. Thanks in advance.

Hey I'm really sorry because I both took a nasty fall on my shoulder and I'm feverish at the date time. All of my remaining attention went to my kids for the last two weeks. But I do feel like everyone should at least get the chance to experiment with Room EQ and build on my findings.

The biggest "problem" I found with Wiim's Room EQ is that it just does an Auto EQ from the measured response to a target. The correction is being done by 10 PEQ filters, no time alignment or attempts at reducing standing waves. BUT as soon as you made an auto-correction profile, you can edit all of the 10 bands, allowing you to virtually build your own Room Correction EQ.

So step 1 would be to make a couple of auto correction's and save them. At this point, decide if you want to make corrections for each channel separately or for both channels at the same time. Because my focus was finding room modes, I choose both channels at the same time, but I want to experiment with individual channels in the future.

Step 2, connect a mic to a laptop and connect the laptop to the Wiim streamer. Install REW on the laptop, import the measurement file for your microphone and set the volume to your preferred reference level. If you only use music, output a -12dB signal and increase the volume to 85dB, if you also watch movies, output a -20dB signal and increase the volume to 85dB.

Step 3 is to do measurements in REW. Position the mic in several different locations of the room for each measurement. Average the measurements.

Step 4, fill in your room dimensions in REW.

Step 5, open the EQ window and in the right sidebar, scroll all the way down to room modes. Pay attention to the ones with a long time or a loudness over 85dB. Display the theoretical room modes based on your dimensions and you should be able to find matches. First order room modes and the ones below 150Hz are the most important to correct.

Step 6 create modal EQ filters using the data in the detected room modes. Using the waterfall graph, lower the peak so it loses energy just as fast as other frequencies and correct the frequency if you feel that it's necessary. Once ready, the modal filters can be changed to peak filters and anything with a Q over 24 can be set to 24.

Step 7, after you have spend your 10 EQ bands on the worst modal frequencies, create a measurement of the predicted frequency response. In the main window, set the smoothing to psychoacoustic.

Step 8, go back to EQ but now create an in-room target. (I made straight downward slope that matches the predicted measurement the best) Make sure the low frequency roll-off matches your measurement, detect reference level based on measurement and do an auto EQ.

Step 9, if you're satisfied with the results, overwrite a Room EQ filter in the Wiim app with the filter settings from step 6. Make a PEQ preset in WiiM with the filter settings from step 8.

Bonus step: make a new target. For quite listening, I made a loudness target. First I generated a flat target with a bass shelf and a treble shelf (using the settings that Yamaha uses on their Auto EQ), and then I applied that to the same slope. You only have to make a new PEQ preset for a new target.

Using this method, the Room EQ filters are always enabled and actually adress room modes and you can change the target frequency response with EQ presets. Having used other Room Correction systems before, you really get the best spacial results when your speakers are time-aligned and phase corrected. But at least cutting out those boomy, bloaty standing waves makes everything sound clear and fast.
Hi Marti,

Sorry to hear about your fall, take your time recovering.

Many thanks for the room correction outline, I didn't appreciate it was so involved and will put some time aside to get my head around it.
 
And even in that correction range, DIRAC trounces RoomFit.
I'd like to take your word for it ... but I wont. This exaggerated language alone keeps me from believing that you've been following all the development steps of RoomFit in detail. Sorry.
 
I'd like to take your word for it ... but I wont. This exaggerated language alone keeps me from believing that you've been following all the development steps of RoomFit in detail. Sorry.
I own a WiiM Ultra. I keep trying with it. It’s still not up to par. They still have basic things missing (albeit in the pipe, including properly implemented subwoofer time-sync, not the current broken algorithm relying on the mic inside the streamer itself). These kinds of things are so basic as to be laughable.
 
I own a WiiM Ultra. I keep trying with it. It’s still not up to par. They still have basic things missing (albeit in the pipe, including properly implemented subwoofer time-sync, not the current broken algorithm relying on the mic inside the streamer itself). These kinds of things are so basic as to be laughable.
Dirac Live room correction sets the subwoofer time sync automatically? That's news to me, indeed. After reading the miniDSP tutorials I assumed this was a manual process.

The price of Dirac Live bass control aline (which requires an existing room correction license) is half that of a WiiM Amp Ultra.

I don't know why WiiM still didn't manage to use the phone's mic for synchronizing mains and sub, but in reality the automatic result isn't that far off in many typical setups. Plus it can be corrected manually. If you had complained about the precision being limited to 1 ms steps, I could have followed you to some degree.
 
How do you know beforehand and for each combination what needs to be delayed, sound or video?

If you think that a variable delay for 4k video doesn't cost much, go ahead, bring it to market. ;)

Also synchronising devices is completely different from keeping the overall latency low. Try playing an electrical instrument through your superior Sonos gear.and report back.
I wasn’t arguing that it is a simple problem. But I think it is not an expensive hardware problem.

People here spend multi-thousand dollars. I think most source material now is digital, and I think a variable input delay could add maybe $50 per component. A click track and rotary delay dial could tune each source. A stand alone delay component would be significantly more expensive.

Without this capability, every component that does DSP or video processing makes a mess of things.
 
Dirac Live room correction sets the subwoofer time sync automatically? That's news to me, indeed. After reading the miniDSP tutorials I assumed this was a manual process.

The price of Dirac Live bass control aline (which requires an existing room correction license) is half that of a WiiM Amp Ultra.

I don't know why WiiM still didn't manage to use the phone's mic for synchronizing mains and sub, but in reality the automatic result isn't that far off in many typical setups. Plus it can be corrected manually. If you had complained about the precision being limited to 1 ms steps, I could have followed you to some degree.
It does not, on that you were correct. You are also correct that whole millisecond increments aren’t acceptable even for manual calibration via REW.

However, it does not lead you into thinking that it will do it automatically, unlike the WiiM, which has the feature, but is completely broken, resulting in even worse results than if it didn’t have the feature at all. This is my fundamental issue with WiiM.

Depending on the device you are using for DIRAC, you need to utilize whatever time-synchronization capability that specific device offers. In the case of the miniDSP, it has the necessary precision, but you need to take manual measurements and apply them.

Full disclosure: there is no specific control over subwoofer to speaker timing on the M1. I utilize physical time alignment (subwoofer’s and speakers’ acoustic centers are precisely the same distance from the main listening position). Were it not for the fact that I am not willing to mess with my perfected placement, I would otherwise be happy to move the sub to a different spot and rerun DIRAC to see if/how it breaks the sound.
 
I picked up a WiiM Ultra and found it to be a excellent sounding streamer. I do run it and my Technics TT (which uses a Muscial Surroundings NovaII pre all of this feeding a Schitt Freya S. It's a large tall room with glass smooth walls (time and phase correction needed here) so for room correction I feed it through a MiniFlex dsp with Dirac. It all dances well together and I could not be happier with the sound on a 2.3 system
 
I’ve never noticed a delay (with the Peo Plus, not ultra - but I can’t see why it would be different) with the TV connected via optical.

For multi-room, WiiM has an inbuilt feature to sync across devices, though I haven’t needed to use it.
 
Can anyone please nominate any differences between the Wiim Amp Ultra and the Wiim Ultra, besides the obvious (that one includes a 100W amplifier and neither have airplay).

I have a Buckeye 502MP amp sitting idle and thought I would combine it with a Wiim Ultra. Thanks in advance.
 
Can anyone please nominate any differences between the Wiim Amp Ultra and the Wiim Ultra, besides the obvious (that one includes a 100W amplifier and neither have airplay).

I have a Buckeye 502MP amp sitting idle and thought I would combine it with a Wiim Ultra. Thanks in advance.
Ultra is a great preamp/DAC/player with somewhat questionable features that do not really add value (headphone amp), lack of LDAC, MM
Amp Ultra has amp but none of those things.
Performance wise the same, feature wise Amp Ultra is more well rounded
 
However, it does not lead you into thinking that it will do it automatically, unlike the WiiM, which has the feature, but is completely broken, resulting in even worse results than if it didn’t have the feature at all. This is my fundamental issue with WiiM.
I cannot say I fully agree with this and I'm not sure how you ended up with this judgement.

My assumption is that all this started with a misunderstanding in development pretty early. I don't have to tell you, but there are clearly to very different reasons for delay between the subwoofer and the main speakers. One is delay due to placement (different runtime of the soundwaves), one is delay due to internal processing of each device, in particular if a DSP is used. WiiM's approach (which I will agree on is far from ideal) should still be able to pick up the delay caused by internal processing about accurately. It cannot and will not take into account the delay due to placement, unless you move your WiiM into the MLP.

Yes, this isn't good and it's not entirely clear why they cannot simply switch to using the smartphone mic instead. OTOH, picking up timing differences in the lowest register is not exactly trivial, because of the lack of sharp transients emitted by the sub. Many AVRs are doing a really bead job automatically aligning the sub with the other speakers. Even upmarket high-end systems like Lyngdorf's RoomPerfect don't even attempt to automate this task. RoomPerfect asks the user to manually measure the distances and to enter the DSP delay number provided by the sub manufacturer instead.

In my experience the auto sync value is a reasonable starting point and manually adjusting this value be 1 or 2 ms either direction is usually all it needs. This is best done looking at the summed frequency response, in particular if you have just a smartphone for measuring.

Admittedly, this leaves customers with the main annoyance: making it look like it would work magically when it really doesn't go the final step. Yep, not good. Telling people how to get around this limitation is one of the most common tasks over on the WiiM forum.
 
… I'm not sure how you ended up with this judgement…
Here’s a hypothetical scenario:
A listener has their WiiM set up at the media console against the back wall, which is over two and a half meters away from the main listening position. Directly underneath the WiiM, is their Rel subwoofer, which is non-DSP, so effectively zero DSP-induced delay. The speakers are placed equidistant from the main listening position as the subwoofer, so they are approximately 3/4 meters away from the back wall, and some meter and a half away from the subwoofer and the WiiM, since they are placed dead center between the speakers. The listener runs the auto-sync, which the WiiM ends up applying a substantial delay to the subwoofer so that it hears the sub the same time as the speakers one and a half meters away, except they should have little to no delay at all because the speakers and sub are actually set up equidistant from the main listening position in a nearly equilateral arc. The listener would have been better off not using this broken feature at all.
 
From the last roadmap they posted:
  • Discrete Dual Subwoofer Control: Independent calibration and management for two subwoofers to smooth out room modes.
Not sure how this works with one sub out…


It’s annoying how every thread about a WiiM product devolves into complaints and speculations over the lack of Airplay. It doesn’t have Airplay, get over it.

There should be more complaints about the high latency rather than a missing feature that can be remedied with $30 raspberry pi…
Yes. I agree.
 
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