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Do I need a preamp and other stupid questions

TheSonic

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Feb 24, 2025
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I’m a recovering audio technophobe and seems like this is the place to get unstuck. If I need to post specs on anything please let me know. Please forgive the budget nature I’m just starting out!

Equipment so far (and return windows fairly open still):

WiiM Pro Plus
FIIO K11 R2R (arriving soon)
Fosi Audio Two V3 dual monoblock (db gain selectable 25/31)
Tannoy Saturn s8

Currently I have the Pro Plus connected to the amps via RCA with the DAC arriving soon. Remote “volume” is via Tidal Connect/Wiim.

1- Will attenuation still be handled via Wiim Pro Plus once I connect the DAC? Remote volume control is a must.

2- I’m considering a Wiim Ultra to get a USB out port @ 32/384 (vs 24/192 out via coax of Wiim Pro Plus) to connect to the external DAC.

3- What function would purchasing a preamp serve? I’m mostly concerned about remote volume control, but I’ve read that impedance matching is important.

Does the equipment I have mesh up even if I presumably use the streamer to change volume? (I’m fuzzy on this, is impedence effected? Gain structures too low for reasonably good sonics, etc?)
 
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Am I on the right track? I will surely upgrade as I go. The thinking behind separates is that I can replace pieces if I see a deal, get the itch or something bricks itself.
 
I'm curious what the purpose of the Fiio K11 R2R is since you've already got a perfectly good DAC in the Wiim Pro Plus.
 
Have you done a forum search on R2R?

I'm not an expert but it's generally not seen here as the best a man can get.

 
Agree that the fiio is pretty redundant here, but if you want to use it you can connect it via optical.
 
I'm curious what the purpose of the Fiio K11 R2R is since you've already got a perfectly good DAC in the Wiim Pro Plus.
There’s an annoying ringing or something of the sort. I’m running a WiiM Pro Plus direct via Blue Jeans RCA to V3 Monos. I tried adjusting the EQ in the WiiM, the various processing settings (fast/slow, NOS) and it still reminds me of a cheap brass bell glaring through the reproduction.

I’ve spent some time now with the K11 R2R via Optical out on the WiiM Pro Plus. The ringing is gone but it’s not as finessed in the highs and bloated in the lows. Overall somewhat “mixed up” sounding too.

Almost reminds me of my cheap Yamaha AVR receiver where there’s less distinction in staging, transients are lost, just more of a lo-fi sound (stage width and height also contracted). Perhaps some is to blame on the Optical out of the WiiM?

I’m fairly confident this budget R2R isn’t for me. I ordered a WiiM Ultra. I’m hoping this will improve things. Next step is better amps.
 
Oh. You can do what you want with your setup, but just fair warning this isn't that kind of audio forum.
 
Sorry to hear you’re not enjoying the Pro Plus. I had mine hooked up directly to a Buckeye amp through RCAs and it sounded fantastic. If you use AirPlay you may want to check if that feature is available on the Ultra, or other WiiM products you are interested in.
 
Hi,
You'll get a lot of similar advice here: the WiiM Pro Plus is an excellent streamer with a really good DAC. You won't get anything that sounds better (plenty will measure better, but the improvement will not be audible).

No worries if you swap the Pro Plus for a WiiM Ultra - different features which you may appreciate. No improvement in sound.

No need for an external DAC. No need for a preamp (unless you need more inputs).

Take some time to consider this. I deliberately put it quite bluntly. Think about why most people here will say something similar.

You won't change the sound by changing electronics.
Obviously, if you use the WiiMs room correction or PEQ features (and you should experiment) then that will change the sound.

Relax with your new gear, enjoy the music.
If you still want changes then start looking at your speakers and room positioning etc
 
I appreciate that this forum is based around what is scientifically measurable and was hoping there might be some technological insight given why the ringing/harsh tonality is only present when V3 monos and Pro Plus are together. For instance, the AVR Yamaha and Pro Plus, tube amp and Pro Plus, or Pro Plus-optical-K11 R2R and V3 Mono all were without this specific issue.

Of course seeking that answer would have to allow for the possibility that such a condition to arise out of the tech itself. Or perhaps just a greater understanding of combinations or wherefrom ringing in D class/digital streaming tends to originate? Unless of course this has never been anyone else’s experience in hifi and I just have a unique combo of equipment and drivers or a defective product.
 
It's a fair question, and one I don't think I can answer.

In general terms, streaming via a WiiM into class D works well. That suggests something about your specific system.

I did have one thought: what power supply do you have for the Fosi's? Is it the shared unit - which Fosi later provided a filter for.

Otherwise I'll leave it to other members more experienced in tracking down faults.

Good luck
 
If I may join in on this please.
Even though my question is very closely related to the OP‘s, I haven‘t really found an answer to what‘s plaguing me.

If I run a streamer via analoge into a power amp and control the volume exclusively in the streamer, wouldn’t that be in the digital domain?

What I‘m getting at are cut off bits. I‘m fully aware that there is a certain headroom for that. But where does this end?

In my understanding a preamp or preamp section in an integrated amp attenuate the analoge signal.

So really, I wonder if running a streamer like a Bluesound Node or a Wiim <whatever> directly into a power amp is technically a good solution.
 
So really, I wonder if running a streamer like a Bluesound Node or a Wiim <whatever> directly into a power amp is technically a good solution.
Yes, it's perfectly fine. From The Well Tempered Computer:

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I'm pretty sure the Wiim uses this method of volume control (happy to be corrected if I'm wrong though).
 
If I run a streamer via analoge into a power amp and control the volume exclusively in the streamer, wouldn’t that be in the digital domain?
Probably, but it the streamer has an analog output it might have an analog volume control.

What I‘m getting at are cut off bits. I‘m fully aware that there is a certain headroom for that. But where does this end?
Each bit represents 6dB. If you have 16-bit audio and you lower the volume by -12dB you are only using 14-bits (the two most significant bits are zeros).

If you have 24-bit audio you can reduce by -48dB and you still have 16-bits.

The same thing happens during quiet parts of the song/program but you can't hear a loss of quality because the quantization noise remains below audibility. Because of that, I say it's OK to use digital volume control for "everyday volume control" and you'll have all of the bits when you turn it up. But you may not want to use it for "gain control" or "permanent attenuation" if you're never going to turn it up and get all of the bits.

Low resolution makes the quantization noise more audible. If you want to know what low resolution audio sounds like, open a file in Audacity* and export it as an 8-WAV file. Quantization noise is like a fuzz on top of the signal. Like regular analog noise, quantization noise is more noticeable with quiet sounds but unlike analog noise it goes-away complete with dead-digital silence.

Most people who talk about resolution have no idea what it sounds like... The perception of quantization noise is not intuitive.

The real issue is if you re-amplify. And analog volume control has similar issues. If you attenuate by a lot and then re-amplify, any noise that's generated after attenuation gets amplified also and you end-up with worse signal-to-noise ratio.

Also "traditional" analog volume control potentiometers sometimes have differences in the left & right pots and the channels may go out of balance, especially at lower volumes. (There are "stepped" analog attenuators that avoid this.)

In my understanding a preamp or preamp section in an integrated amp attenuate the analoge signal.
They usually have available gain if/when you need it but in most cases we are attenuating, especially if we're not listening at maximum volume.

But noise floor is one thing. What about dynamics and the audiophile-knows-what?
Dynamics - I like the term "dynamic contrast" is the difference between the loud and quiet parts and the comes from the program. It's complicated because there are long-term dynamics like a song or symphony that starts out quiet and ends loud or micro-dynamics like a loud cymbal crash, and everything in-between. Most modern recording have dynamic compression to make "everything constantly loud" (The Loudness War). A lot of use music lovers feel that it's overdone, it makes music boring, and we just turn it down.

Basically, everything is linear and the program dynamic aren't hurt or compressed by the equipment unless you over-drive an amplifier into clipping, which is a "very bad kind" of compression. But clipping is simply heard as distortion rather than a loss of dynamics.

Note that dynamic compression is totally unrelated to file compression like MP3 which is used to make the file smaller. MP3 is lossy compression and it throws-away information (it tries to throw-away details we can't hear anyway) but it does NOT compress the dynamics.

Dynamic range relates to the equipment, storage format or transmission. It's limited by noise on the quiet-end and the maximum signal level on the loud-end. The dynamic range of the equipment far-exceeds the dynamic contrast in the program (ignoring "infinite" fade-outs, etc.) but it's still very important because we don't want to hear the noise at all!, no matter the dynamic range or signal-to-noise ratio.

With digital, it's just a range of numbers... How high you can "count to" with a given number of bits, with 0dBFS (zero decibels full scale) representing the "digital maximum". And how low you can go before you count down to zero.** A 24-bit file has bigger numbers than an 8-bit file but everything is automatically scaled to match the DAC so the 24-bit file isn't louder. And as above, when you get close to zero, the loss of resolution is heard (if heard) as quantization noise. Floating-point digital, for all practical purposes, has infinite dynamic range. IIRC, 32-bit floating-point goes from about -750dB to +750dB (with a value of 1.0 representing 0dBFS). Of course, that dynamic range only exists in the digital domain.



* If you don't already have Audacity "download without Muse Hub" so you don't accidently sign-up for anything extra. And go into the setting and turn off dither, because dither is added noise that's supposed to sound better than quantization noise.

** If you include zero, which is digital dead-silence, we could say that digital has infinite dynamic range because an amplitude of zero is minus infinity dB. But that's not helpful in the practical world. We want to know the lowest non-zero amplitude or the quantization noise.
 
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