Just an additional piece of info I’ve noticed, the songs on Spotify that seem to sound bad (off and on constantly while playing) are the latest remastered stuff and seems to be the same songs. My normal playlists play fine but I tried the latest remastered 2023 u2 (yes I know 😂) and that’s all over the place, I went back to one I knew worked and was fine and then went back to the u2 and exactly the same again, all over the place. 
I’m thinking my kit is probably dated too much now it could be time to move up to era’s? 

I’d take a look at the data rates your selections are using, if the new stuff is pushing more data you may simply have wireless issues that can usually be resolved with out buying new gear that would suffer the same problems.

https://support.sonos.com/en-us/article/improve-your-sonos-products-wifi-connection

I’ve only just realised I meant to post this on my thread but just realised I’ve added it to yours instead FFS 🤦🏼

Sorry 

This is profoundly incorrect. Sampling rate has nothing to do the frequency of the audio data that it contains. it's the frequency at which the data is captured. there is plenty of audiophile snake oil out there, this isn't one of em.

Honestly, you couldn’t be more wrong. 

https://www.asel.udel.edu/speech/tutorials/instrument/sam_rat.html

See the bolded?

Sampling Rate

Sampling rate determines the sound frequency range (corresponding to pitch) which can be represented in the digital waveform. The range of frequencies represented in a waveform is often called its bandwidth. Waveforms sampled at a high sampling rate can represent a broad range of frequencies and hence have broad bandwidth. In fact, the maximum bandwidth of a sampled waveform is determined exactly by its sampling rate; the maximum frequency representable in a sampled waveform is termed its Nyquist frequency, and is equal to one half the sampling rate. Thus, for example, a waveform sampled at 16,000 Hz can represent all frequencies up to its Nyquist frequency of 8,000 Hz.

Also (more technical)

https://en.wikipedia.org/wiki/Nyquist%E2%80%93Shannon_sampling_theorem

The Nyquist–Shannon sampling theorem is an essential principle for digital signal processing linking the frequency range of a signal and the sample rate required to avoid a type of distortion called aliasing. The theorem states that the sample rate must be at least twice the bandwidth of the signal to avoid aliasing distortion. In practice, it is used to select band-limiting filters to keep aliasing distortion below an acceptable amount when an analog signal is sampled or when sample rates are changed within a digital signal processing function.

. . .

The sampling theorem introduces the concept of a sample rate that is sufficient for perfect fidelity for the class of functions that are band-limited to a given bandwidth, such that no actual information is lost in the sampling process. It expresses the sufficient sample rate in terms of the bandwidth for the class of functions. The theorem also leads to a formula for perfectly reconstructing the original continuous-time function from the samples.

All those jaggy stairstep diagrams the audiophiles have been selling you are complete lies.  There are no estimations in sound sampling, there’s no “filling in” of the parts in between.  Lossless sampling is truly lossless, up to ½ the sampling rate.

just because sampling rate determines bandwidth doesn't mean it IS bandwidth. there's a lot you can do with bandwidth. 

Yes, you can store sounds that only your dog can hear and charge gullible humans a premium for them.  Because I can't think of anything else I can do with sounds above 20 kHz.

Oh wait, yes I can.  You can use them to cause audible intermodulation distortion in the playback hardware, making the audio sound worse!  Silly me.

And thanks for the acknowledgement that my post was not "profoundly incorrect".  I appreciate it.

During the lecture on sampling theory the audiophiles ran out in mid lecture flailing their hands, shouting about “jaggies”. Had they stayed for the full lecture they would understand why the jaggies are not present in the final reconstruction. This is somewhat like complaining how bad an empty construction site looks before it is painted and furnished.

Processing, at any bandwidth, can be constructive. While uncompressed music might sound better to some listeners in a quiet room, compressing music that will be played through earbuds on a busy street results in music that is much more comfortable to listen to.

By the way, there are two types of “compression”. One type is an attempt to reduce the amount of data that must be transferred from here to there. This type of “data compression” is simply some data processing tricks. Once lossless data compression is undone at the receiving end there is no way to determine that the data had been compressed along the way. Nothing is added and nothing has been removed. Some audio data compression methods, such as MP3, will permanently alter the music. Audibility of this compression depends on how aggressive the compression was. Early in this digital age when music was sent over voice grade lines and a 20MB hard drive was HUGE, music compression had to be very aggressive and there were clearly audible consequences. At this point we don’t need to worry much about disk space or the connection bandwidth.

The other type of compression can be done in the analog or digital world. This should really be described as “dynamic range compression”. “Dynamic range” is the difference between the quietest and loudest notes in a session. In noisy listening environments one must increase the Volume in order to prevent the quiet parts from being lost in the environment’s noise. Unfortunately, loud portions may then be above the threshold of pain and you’ll be racing for the Volume control. Dynamic range compression is an automatic Volume control adjustment. Tastefully used, dynamic range compression can improve listening enjoyment in difficult environments. Music producers have proven that highly compressed music sells better than uncompressed music. This is why it is difficult to purchase uncompressed releases. You need to actively seek uncompressed releases. If you are listening in a good (quiet) environment, premium releases (often touted as “Hi-Res”) sound better if they have not been compressed.

In my opinion, music should be distributed without dynamic range compression. When needed, compression can be easily applied by the playback system. Historically, compression has increased the equipment costs, been misunderstood by the public, and has always been dumped on by the audiophile community. At his point playback compression hardware is virtually costless. Unfortunately, there will need to be an added button or two and the user will need to spend a couple minutes learning to use the feature. This might result in a unit being branded as “complicated” by some users. And, the audiophiles will continue to dump on the feature. A great feature would be to use a phone/pad/computer microphone to measure the ambient noise and adjust the playback compression automatically.

I loved my dBx audio compressor, it had a lot of uses but I sure didn’t want original audio that someone else had messed with.

Some purchased stuff that had been over-compressed at the studio did sound better slightly uncompressed but artifacts were always an audible issue. Being able to compress at home was nice, either to bring the quiet passages up above ambient noise or to dampen the loud sounds at night, I usually set the range gate to only act above normal volume levels though.

The place it really worked well was in making cassette tapes for the car. Setting a pretty large compression ratio with no range gate brought the lower level sounds up over the car noise while not blowing your ears out on the loud stuff. In a quiet setting it was obvious the music had suffered badly but in the car it worked well.