Sonos Connect optical out max resolution

Home audio of that type is based on the false notion that the more ridiculous the cost, the better it must sound. This taps into cognitive biases that afflict all humans. I would submit that your kit may be medium grade only in what it cost you compared to the kit you refer to, not in the sound it delivers in your listening spaces.

As to the DAC magic - you hear what you do but make sure that what you conclude is not driven by similar bias, and by sound level differences that need to be as little at 0.1dB to return similar improvements that you hear. 

The only way to be sure that the difference is real is if it survives a precision sound level matched blind listening test in a manner that is statistically reliable. 

Having installed the DAC and liking what it does, I am not suggesting you run such a test. But it might be something to bear in mind before making the next hardware uprade.

Sampling (signal processing) - Wikipedia

At each sample step all frequencies present superimpose and result in the total amplitude sampled at that sample step. The issue of precision comes with reconstruction in the DAC.

Note the remark in the above reference that the original signal can be reconstructed up to the Nyquist limit (half sample rate). Each sample is reconstructed with great accuracy by a bit stream (delta-Sigma) DAC. However, the detail between reconstructed samples are approximated by interpolation. The higher the sample rate, the fewer the in-between sample step approximations, hence more accurate reproduction.

Also, the issue of reconstruction and sample rates affects predominantly the upper audible range - above 10kHz, where much of the detail of acoustics and much of the tonal colour are represented for the human ear. Below 10kHz signal, the 44.1kHz sample rate is above 4 times over-sampled. E.g. at 1kHz, a 44.1kHz sample rate is 44.1 x over-sampled - ample, very minute interpolation errors during reconstruction. But at 20kHz, a 44.1 kHz sample rate is only just more than the Nyquist minimum of twice the recorded frequency - more scope for interpolation errors vs the case of a 1kHz signal. So, increasingly between 10kHz and 20kHz is where the issue lies with CD reproduction often sounding unnatural and bright. Here higher oversampling (48-192kHz) have the greatest benefit, if diminishing with raising sample rate.

Many players have a lot of leftover trash in their output -- due to leakage on the PC board, not because of any difficulties with the math. This trash results in intermodulation distortion. External DAC’s may minimize this trash. The external DAC could smooth out some clock jitter too.

Oversampling at playback cannot insert information that was never present, but it reduces potential mischief caused by the reconstruction filter. Too many people bolted out of the Nyquist-Shannon lecture flapping their arms, shouting “jaggies”, “jaggies”, “jaggies” and never stayed for the second part about the reconstruction filter.

I give up.  

The reconstruction filter is exactly where the interpolation happens that, in my view, is the root cause of the perceived tonal colourations of various DAC implementations. See the references below:

[https://en.wikipedia.org/wiki/Reconstruction_filter]

[https://academic-accelerator.com/encyclopedia/reconstruction-filter]

A few DAC makers and implementers get it "right” to avoid bright treble, yet have full, true detail up to 20kHz. The rest never get it spot on. But given any reconstruction filter, I maintain my point that, above 10kHz, we run into increasing challenges for interpolation, because we are going below the practical over-sampling factor of 4 as we progress towards 20 kHz, increasing the demand on the approximation made between sampling points by the interpolation algorithm - the goal being minimal error between approximation and what the true amplitude would have been at this time step, were it sampled in the first place. The simple solution is higher sample rate, at the cost of larger files and increased demand on the electronics. Higher sample rate is not only to ease the design of anti-aliasing filters.

On the other hand, my listening to various front end devices tells me that DACs are now a cheap and reliable commodity. The one in an iPhone is as good as the one in the Echo Dot that is as good as the one in the Sonos Connect/Amp which is as good as the external one I had in 2008 that was in a case worthy of a big audiophile amp of the day, with valves in the output stage that glowed!

What you say may have been true for DACs in the early days of digital audio.

The only time this may not hold good is if the listening is via high quality headphones where the room acoustics and the ambient noise floor are taken out of the frame completely, allowing for small difference to emerge to the extent that they can be perceived.

You want to up your sound quality? Get better speakers, that interact with your room well, if properly placed there. And make sure that your amp is not underpowered for what the speakers need to deliver the designer’s intent for them. Once you do this, and are using the best master for a given recording regardless, within reason, of the bitcount for it, nothing else matters, in terms of sound quality that is audible in a typical domestic environment, even if that is a quiet room.

Note that some DACs have sound shaping filters - the only way to compare these on a apples to apples with a DAC that has none is to not have the filters in use.

If you have an oscilloscope or spectrum analyzer it’s easy to see trash above 10KHz -- while playing a silent track. This is due to on board leakage. The trash causes intermodulation. Playback oversampling makes it easier to filter the trash without impacting the music.

Two questions from a layman: what you see on the scope may not be heard in a typical domestic home audio set up based on speakers, unless the trash is large enough to both be seen and heard, right? I would think it can be seen long before it can be heard.

As to the latter part of playback oversampling - what is that bit in English, with reference to heard sound quality, and is this not more relevant at the mastering stage where I keep hearing that the extra bit count/depth etc makes the mastering task easier, but is not needed once that effort is over?

Question for the OP - using amps as the analogy. Solid state amp design was a solved problem a few decades ago. But even today one can get two amps with the same published specs with one costing USD 500 and the other for USD 10000. Seeing that an amp is meant to be the classic straight wire with gain, would they deliver any better sound quality than the other? Yes, the expensive one may look better, have more features/connectivity options, and may comfortably outlast the cheaper, but in a blind test, both would sound the same as long as both are in the same working condition as when new and neither is being forced to deliver more power that it is designed to, by power hungry speakers.

The same applies to DACs today with the difference being that they do not have to deal with different speaker loads as amps have to. And yet, a few years ago there was an external DAC, with no other functionality, priced at USD 50000. There is a choice of words that can be used for those that bought it at that price.

Do you think that it did anything different for how the music sounded, if assessed objectively? If one did not know what was the price paid for it?

Hello again.

Concerning my original question under this topic, I have realised after some research that all MQA are packed either as 44.1 kHz or 48 kHz streams (depending upon the original master sample rate, which will be a multiple of either 44.1kHz or 48 kHz). The bit depth may be either 16bit or 24bit, again depending upon the particular master. So, the tracks playing via my Sonos Connect S2 that do light up the MQA indicator on my DagMagic 200M, must be 16bit/44kHz MQA streams, which are passed through bit-perfect to the DagMagic. The latter performs full, hardware-based decoding (core and render). I assume that the DagMagic actually checks for bit-perfect incoming before lighting its MQA indicator. So, at least, it is not a complete waste in terms of my Tidal HiFi subscription (setting aside the ongoing debate over the merits of MQA and hi-res FLAC). Also, the official Sonos user guide on Tidal music service states that only 16bit/44kHz streams are supported by Sonos for Tidal.

However, I reckon to have gathered some evidence that suggests Sonos Connect S2 does not output 24bit on its digital outputs - at least, not for Tidal streams. It may do so for other streams, but I am not subscribed to such and therefore cannot test it at present for any other than Tidal Hi-res streams. This finding agrees with the above-mentioned official Sonos guidance on support for Tidal music service.

Experimenting with Tidal desktop app on my Windows laptop, which streams via the custom CA DacMagic USB sound driver over a USB-B 2.0 cable to the DagMagic 200M, I can report that certain MQA tracks light the DagMagic MQA indicator when streaming via USB-B, but those same tracks do not light the DacMagic's MQA indicator when streamed from the Sonos Connect (S2 OS). This result suggests that either 24bit MQA is truncated to 16 bit over digital out on the Sonos Connect, or Tidal sends 16bit/44kHz versions to Sonos (therefore, not MQA in the case of 24bit MQA sources). Therefore, the incoming stream to the DagMagic is either not bit-perfect, or not MQA, when coming from the Sonos Connect, and consequently the MQA indicator does not light up.

Contrary, some MQA tracks do light the MQA indicator for both streaming paths - when streamed via USB-B, as well as when streamed from the Sonos Connect. I assume these cases are 16bit MQA (apparently, there are quite a number of these around).