I have a Sonos Connect S2 linked via TOSlink to Cambridge Audio DacMagic 200M. I am subscribed to Tidal HiFi Plus..
However, the DacMagic only ever indicates 44.1 kHz , even when playing what Tidal says is a Max resolution track ( Flac or MQA). TIDAL app is set to stream Max.
What is the matter?
Best answer by GuitarSuperstar
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As has already been noted, Sonos only receives 16/44.1 from Tidal. Indeed you yourself made that observation here.
I really don’t understand this obsession with MQA. By all accounts it’s an “ex-codec” (with acknowledgements to Monty Python).
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).
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?
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?
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.
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.
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.
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.
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.
I give up.
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.
My request for a cite is for the part after "in other words". Can you cite a study which proves superposition actually "colours the sound and detail that we perceive"? Better yet, one which finds that it improves what we perceive. Because I know of several where hypersonics actually added intermodulation distortion in the playback stream, which is never beneficial to the sound.
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.
Kumar, I have seen ridiculously expensive kit displayed in hifi shops around Cape Town. My stuff are decidedly medium grade compared to those things.
But the DagMagic has brought improved detail, texture, stage acoustics, and better composure
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.
I have some such recordings and a medium grade sound system (Sonos Connect, QED Performance Graphite TOSlink, DagMagic 200M, Cambridge Audio CXA60 amp, Van den Hul Clear Water speaker cables, B&W DM1600 speakers (ages old). For CDs, I use an old Rotel RCD965BX CD player with digital out to DagMagic 200M (QED Performance digital coaxial interlink).
Other than Sonos Connect, all familiar names from a past life; what is medium grade about these bits of kit?!
And Sonos Connect belongs in this set for sound quality it delivers. Presumably, the Rotel CDP has analog line out as well, so the DAC magic interposed between that and the amp is another familiar audiophile fancy! I once had another magic box called a valve based output stage buffer which essentially was a impedance matcher, that had another separate box that was optional - a “better” power supply for the magic box…Boxes and more boxes😁
Kumar, I have seen ridiculously expensive kit displayed in hifi shops around Cape Town. My stuff are decidedly medium grade compared to those things.
Concerning the Rotel 965, I have run that one for years with analogue out. Great CDP. But the DagMagic has brought improved detail, texture, stage acoustics, and better composure when the music gets complicated and loud (when the 965’s built-in DAC would get a bit rough). It was most apparent when I briefly switched back to the Rotel analogue out after running with the DagMagic for a bit. The drop in treble clarity & detail was quite audible.
So, now the Rotel serves as a CD transport. More boxes, as you say.
I have some such recordings and a medium grade sound system (Sonos Connect, QED Performance Graphite TOSlink, DagMagic 200M, Cambridge Audio CXA60 amp, Van den Hul Clear Water speaker cables, B&W DM1600 speakers (ages old). For CDs, I use an old Rotel RCD965BX CD player with digital out to DagMagic 200M (QED Performance digital coaxial interlink).
Other than Sonos Connect, all familiar names from a past life; what is medium grade about these bits of kit?!
And Sonos Connect belongs in this set for sound quality it delivers. Presumably, the Rotel CDP has analog line out as well, so the DAC magic interposed between that and the amp is another familiar audiophile fancy! I once had another magic box called a valve based output stage buffer which essentially was a impedance matcher, that had another separate box that was optional - a “better” power supply for the magic box…Boxes and more boxes😁
agree with Buzz: Good recordings, good mixing and good mastering at 24bit/48kHz, released as 16bit/44.1kHz should sound quite brilliant on a good sound system.
Not just should sound so, but also actually do. And in my assessment of what is a good sound system, after having used many others in the days of my audiophile pursuits now abandoned, I find that Sonos can be as good a sound system as those I have used, for the sound quality it can deliver.
Why then tie oneself into knots by succumbing to all the digital snake oil being hawked by marketers today?
To the quoted words I will only add: placed appropriately in a decent domestic acoustic environment.
Even Spotify streams are, with some rare exceptions, just as good to listen to as their CD equivalents, except perhaps where high quality headphones are being used for listening.
Internally the players have had 24bit data paths since inception in 2005, however, there was not enough internal processing power or network bandwidth to deal with 24bit high sample rate content. While there are advantages when using 32 bit high sample rate during post processing in the studio, there is a complete shortage of peer reviewed studies proving that anything beyond 44.1/16 is necessary for distribution. By the way some studio people feel that 192/24 is a lame, outdated, low sample rate format.
Rather than chasing word length and bit rate, you are better off chasing quality mastering and absence of compression. I’ve had UK, Japanese, and US LP pressings, CD’s, and Mobile Fidelity CD and LP of the same master tape to compare. They sounded so different that it was hard for me to satisfy myself that they all originated from the same studio master tape. The US LP’s and CD’s were clearly the worst. Carefully comparing the Mobile Fidelity CD and LP, when using a very high quality CD player and turntable ($200 units need not apply), the CD and LP were almost indistinguishable until the LP issued a ‘tic”. After enough listening it became clear that the CD had a lower noise floor and a wider dynamic range. Casual audiophiles typically walked out mad after they misidentified CD or LP a few times, claiming that I somehow tricked them or that my equipment was not good enough.
I am shocked at the horrible mixing and mastering out there. Some songs of the 80's, 90's and early 2000's are terribly bright. Unbearably so (reason - mass music consumption on poor quality portable music players needed boosted treble content to sound OK).
But further to the point of sample rate: A good bit-stream DAC can reconstruct the sampled points with very low error, but what about in between sampled points? Those in between points are approximated by some interpolation based upon assumptions. Nyquist-Shannon theory may state you need to sample only twice the highest interesting frequency (40kHz for 20kHz) - which holds true for a pure sine wave - but in practice you need to oversample most real-world signals at least four times for reasonable reconstruction. So, 96 kHz sample rate means that you guess about half as many times as with 44.1kHz. Remember, what you have not measured (sampled), you do not know for a fact. You can only approximate it from what you do know (do have sampled). So, higher sample rates do bring more accurate reconstruction. But again, diminishing returns at rates increase.
Yet another benefit of higher sample rate is that it eases the stress on filter design. The higher the sample frequency, the slower the digital filter can be without risking aliasing folding into the audible spectrum (up to 20kHz). As you likely know, a slower filter (more gradual cut-off) has a more benign behaviour in that it has less pre and post ringing artefacts than a faster filter (faster cut-off). Reproduction tends to sound more like old analogue on LP and less bright, yet detailed - this one is subjective.
I agree with Buzz: Good recordings, good mixing and good mastering at 24bit/48kHz, released as 16bit/44.1kHz should sound quite brilliant on a good sound system. I have some such recordings and a medium grade sound system (Sonos Connect, QED Performance Graphite TOSlink, DagMagic 200M, Cambridge Audio CXA60 amp, Van den Hul Clear Water speaker cables, B&W DM1600 speakers (ages old). For CDs, I use an old Rotel RCD965BX CD player with digital out to DagMagic 200M (QED Performance digital coaxial interlink).
Cite for the bolded?
My request for a cite is for the part after "in other words". Can you cite a study which proves superposition actually "colours the sound and detail that we perceive"? Better yet, one which finds that it improves what we perceive. Because I know of several where hypersonics actually added intermodulation distortion in the playback stream, which is never beneficial to the sound.
At last - I have come across an official Sonos article about Tidal on Sonos. They support up to 16 bit/44.1kHz. No luck with better sample rates or bit depth
From their guide document: “ Stream CD quality audio (up to 16-bit, 44.1 kHz) ”
https://support.sonos.com/en-us/services/tidal?language=en_US
Which leads into the argument over how much resolution is sufficient…
Don't begin with “human ears can't hear beyond 20kHz”. That argument ignores the principle of superposition. Even though one cannot hear frequencies above 20kHz, if the signal amplitudes of such ultrasound frequencies are not at zero crossing at corresponding sample points in time of frequencies that we can hear, these ultrasound signals will either add to or subtract from the total signal amplitude - in other words, still contribute to the overall sound colour and detail that we perceive. But it is a game of diminishing returns. For 24 bit (and 16bit), beyond 44.1kHz the improvement decreases non-linearly. At 48 kHz you should still hear an improvement. At 96 kHz less so, and at 192 kHz, my guess is that one will be hard-pressed or need exquisite sound equipment to hear a clear improvement. So, perhaps, I should settle for 16bit/44.1kHz and downgrade my Tidal HiFi Plus to HiFi. Or buy a network streamer that does pass through 24bit 48kHz to my DAC.
Do you have a cite for the bold above? Preferably from an accredited scientific journal?
https://www.britannica.com/science/principle-of-superposition-wave-motion
At an AES convention Tomlinson Holman presented a paper describing his preamp design criteria. He is a limited bandwidth believer. After the presentation he was challenged by a person who claimed to have observed 400KHz on a spectrum analyzer connected to his turntable -- therefore the 20KHz limit on the Holman preamp was limiting the music. Behind me there was a phono cartridge designer and a disk cutting head designer mumbling that “we cannot play these frequencies” and “we cannot cut those frequencies”, implying that the 400KHz was a spurious response possibly the result of a cartridge resonance. Tomlinson’s argument was that limited bandwidth results in less intermodulation and much cleaner output. Mixing 32KHz and 31KHz signals can result in a 1KHz, audible intermodulation product plus harmonics. In other words mixing of inaudible ultrasonic signals can result in a blizzard of audible trash.
The wide and narrow bandwidth camps will forever be at odds with each other.
A narrow bandwidth person has no need for sample rates beyond 44.1, nor do the ears of adults.
In the studio there is plenty of room for mathematical mischief during digital processing. High sample rates and wide words minimize the risks.
Internally the players have had 24bit data paths since inception in 2005, however, there was not enough internal processing power or network bandwidth to deal with 24bit high sample rate content. While there are advantages when using 32 bit high sample rate during post processing in the studio, there is a complete shortage of peer reviewed studies proving that anything beyond 44.1/16 is necessary for distribution. By the way some studio people feel that 192/24 is a lame, outdated, low sample rate format.
Rather than chasing word length and bit rate, you are better off chasing quality mastering and absence of compression. I’ve had UK, Japanese, and US LP pressings, CD’s, and Mobile Fidelity CD and LP of the same master tape to compare. They sounded so different that it was hard for me to satisfy myself that they all originated from the same studio master tape. The US LP’s and CD’s were clearly the worst. Carefully comparing the Mobile Fidelity CD and LP, when using a very high quality CD player and turntable ($200 units need not apply), the CD and LP were almost indistinguishable until the LP issued a ‘tic”. After enough listening it became clear that the CD had a lower noise floor and a wider dynamic range. Casual audiophiles typically walked out mad after they misidentified CD or LP a few times, claiming that I somehow tricked them or that my equipment was not good enough.
I am shocked at the horrible mixing and mastering out there. Some songs of the 80's, 90's and early 2000's are terribly bright. Unbearably so (reason - mass music consumption on poor quality portable music players needed boosted treble content to sound OK).
But further to the point of sample rate: A good bit-stream DAC can reconstruct the sampled points with very low error, but what about in between sampled points? Those in between points are approximated by some interpolation based upon assumptions. Nyquist-Shannon theory may state you need to sample only twice the highest interesting frequency (40kHz for 20kHz) - which holds true for a pure sine wave - but in practice you need to oversample most real-world signals at least four times for reasonable reconstruction. So, 96 kHz sample rate means that you guess about half as many times as with 44.1kHz. Remember, what you have not measured (sampled), you do not know for a fact. You can only approximate it from what you do know (do have sampled). So, higher sample rates do bring more accurate reconstruction. But again, diminishing returns at rates increase.
Yet another benefit of higher sample rate is that it eases the stress on filter design. The higher the sample frequency, the slower the digital filter can be without risking aliasing folding into the audible spectrum (up to 20kHz). As you likely know, a slower filter (more gradual cut-off) has a more benign behaviour in that it has less pre and post ringing artefacts than a faster filter (faster cut-off). Reproduction tends to sound more like old analogue on LP and less bright, yet detailed - this one is subjective.
I agree with Buzz: Good recordings, good mixing and good mastering at 24bit/48kHz, released as 16bit/44.1kHz should sound quite brilliant on a good sound system. I have some such recordings and a medium grade sound system (Sonos Connect, QED Performance Graphite TOSlink, DagMagic 200M, Cambridge Audio CXA60 amp, Van den Hul Clear Water speaker cables, B&W DM1600 speakers (ages old). For CDs, I use an old Rotel RCD965BX CD player with digital out to DagMagic 200M (QED Performance digital coaxial interlink).
At last - I have come across an official Sonos article about Tidal on Sonos. They support up to 16 bit/44.1kHz. No luck with better sample rates or bit depth
From their guide document: “ Stream CD quality audio (up to 16-bit, 44.1 kHz) ”
https://support.sonos.com/en-us/services/tidal?language=en_US
Which leads into the argument over how much resolution is sufficient…
Don't begin with “human ears can't hear beyond 20kHz”. That argument ignores the principle of superposition. Even though one cannot hear frequencies above 20kHz, if the signal amplitudes of such ultrasound frequencies are not at zero crossing at corresponding sample points in time of frequencies that we can hear, these ultrasound signals will either add to or subtract from the total signal amplitude - in other words, still contribute to the overall sound colour and detail that we perceive. But it is a game of diminishing returns. For 24 bit (and 16bit), beyond 44.1kHz the improvement decreases non-linearly. At 48 kHz you should still hear an improvement. At 96 kHz less so, and at 192 kHz, my guess is that one will be hard-pressed or need exquisite sound equipment to hear a clear improvement. So, perhaps, I should settle for 16bit/44.1kHz and downgrade my Tidal HiFi Plus to HiFi. Or buy a network streamer that does pass through 24bit 48kHz to my DAC.
Do you have a cite for the bold above? Preferably from an accredited scientific journal?
Internally the players have had 24bit data paths since inception in 2005, however, there was not enough internal processing power or network bandwidth to deal with 24bit high sample rate content. While there are advantages when using 32 bit high sample rate during post processing in the studio, there is a complete shortage of peer reviewed studies proving that anything beyond 44.1/16 is necessary for distribution. By the way some studio people feel that 192/24 is a lame, outdated, low sample rate format.
Rather than chasing word length and bit rate, you are better off chasing quality mastering and absence of compression. I’ve had UK, Japanese, and US LP pressings, CD’s, and Mobile Fidelity CD and LP of the same master tape to compare. They sounded so different that it was hard for me to satisfy myself that they all originated from the same studio master tape. The US LP’s and CD’s were clearly the worst. Carefully comparing the Mobile Fidelity CD and LP, when using a very high quality CD player and turntable ($200 units need not apply), the CD and LP were almost indistinguishable until the LP issued a ‘tic”. After enough listening it became clear that the CD had a lower noise floor, distortion, and a wider dynamic range. Casual audiophiles typically walked out mad after they misidentified CD or LP a few times, claiming that I somehow tricked them or that my equipment was not good enough.
Seems like Sonos, which has never sold itself as a ‘audiophile’ company, may not be the right choice for you.
At last - I have come across an official Sonos article about Tidal on Sonos. They support up to 16 bit/44.1kHz. No luck with better sample rates or bit depth
From their guide document: “ Stream CD quality audio (up to 16-bit, 44.1 kHz) ”
https://support.sonos.com/en-us/services/tidal?language=en_US
Which leads into the argument over how much resolution is sufficient…
Don't begin with “human ears can't hear beyond 20kHz”. That argument ignores the principle of superposition. Even though one cannot hear frequencies above 20kHz, if the signal amplitudes of such ultrasound frequencies are not at zero crossing at corresponding sample points in time of frequencies that we can hear, these ultrasound signals will either add to or subtract from the total signal amplitude - in other words, still contribute to the overall sound colour and detail that we perceive. But it is a game of diminishing returns. For 24 bit (and 16bit), beyond 44.1kHz the improvement decreases non-linearly. At 48 kHz you should still hear an improvement. At 96 kHz less so, and at 192 kHz, my guess is that one will be hard-pressed or need exquisite sound equipment to hear a clear improvement. So, perhaps, I should settle for 16bit/44.1kHz and downgrade my Tidal HiFi Plus to HiFi. Or buy a network streamer that does pass through 24bit 48kHz to my DAC.
I don’t know precisely how the ASR 24-bit test signal was delivered but, sure, it could have been truncated. Sonos currently only offers 24-bit support from certain sources, such as local files, Amazon and Qobuz. For Tidal Sonos only claims 16-bit support, so perhaps your little blue LED was being illuminated by (horror of horrors) MQA-CD files?
My point about the Connect/ZPx0 digital out however stands. The S/PDIF has always been 24-bits wide, to allow for the volume adjustment of 16-bit samples down to -48dB without loss of information.
On the format support question, Sonos doesn’t support anything over 48kHz sampling rate. Its native sampling rate is 44.1kHz, and I have a suspicion that it may convert 48kHz to 44.1kHz before sending it out the S/PDIF. I haven’t looked in a long while. To be frank, if you’re wanting a bit-perfect streamer at higher sampling rates you’d need to consider alternatives.
I am afraid, you have it wrong, Ratty: My Connect uses Sonos OS S2, according to the Sonos controller app on my smartphone (which app is also the S2 version).
Version: 15.9 (build 75146030)
The Connect has been updated at every Sonos OS update roll-out, so it is CURRENT in as far as the latest software. In this context, I repeat my question: How relevant is the above article, dated 2019? Sonos could have updated the software not to truncate 24 bit to 16 bit.
Does anyone has authoritative, as opposed to speculative guesses, at the correct answer?
As for MQA - I am not particularly hooked on it, but rather the question is much wider than MQA: will the Sonos Connect pass through 24bit at 48 or higher sample rates of ordinary (non MQA) FLAC files to the DagMagic over digital out, or will it truncate and downsample before passing the PCM stream to the DacMagic?
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