Hi-Resolution Audio and Sonos



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I couldn’t see a need for multiple Boost devices since you can hook your Ethernet to any primary (not sub/surround/move) SonosNet v2 (not early ZP units) Sonos device and get essentially the same thing.

I wired every Sonos that was easy to reach with a cable and I can happily stream FLAC to Groups. I do pick a wired Sonos as the first device (group coordinator) in the group though.

Oh dear. It sounds like a brief education in Fourier analysis is also required.
Oh dear. It sounds like a brief education in Fourier analysis is also required.

Well, to the defense of the OP, Fourier analysis is usually not covered in a first year college course. 😉

In the real world how many of our multi speaker Sonos systems would have a hope in hell of keeping up the bitrate over wireless of HiRes.

I have always wondered if Sonos was not capable of buffering a greater amount of information to help with streaming in Flacs. 

 

 

Now that I extensively use Firestick + Netflix HD, I believe that the second para quoted is the solution to the first. If HD video streams can be viewed in HD quality across the home as long as the broadband pipe is big enough, I can’t see why Sonos HD audio should be affected IF the buffer is large enough. I suspect Firestick uses a large buffer and because both audio/video are still in perfect sync, the size of the buffer isn't an issue for that sync.

I needed a ethernet wire to get Sonos to play in sync with TV because that sync means that Sonos can’t use a buffer. But where it can, I see no reason why HD audio streaming should be an issue when HD video clearly isn't.

@ninjabob : what happens if you use the option Sonos now gives you to set the buffer at the extreme end - 2 seconds?

Pushing voluminous quantities of air costs energy which in turn costs money. Physics.

 

 

While I am off the Sonos bandwagon where future buying is concerned, I am still a fan of the 1 pair + Sub combination for audio where listening distances are such that do not need a larger speaker than the 1. 

I agree with the physics comment - and physics will always prevail except for headphone listening - but the Sonos Sub meets this requirement smartly because the air movement requirement is only for low frequencies, and the Sub does that as well as any comparable floor standing speaker. And by separating this from the midrange sources - the 1 units - the result is better than many floor standing speakers that suffer from bass bloat affected midrange quality.

All other things like budgets and room acoustics being the same, I would always go for a satellite speaker+ Sub set up for home audio over a pair of all in one floor standers. Also because the high prices are often dictated in the latter case by the needs of very expensive cabinetry and engineering needed to keep the midrange sound quality not affected by the need to deliver accurate bass - a floor stander that sounds as accurate/natural as a sat+Sub set up will have to be at much higher price points.

This, without the additional benefits that are available by DSP that tunes the heard sound for room responses, that needs active cross over tech of the kind that is usually missing in passive floor standing speakers.

While therefore I still believe that Hi Res audio is a red herring, I do not think that this is the case just because Sonos at its best does not have the ability to bring out the better resultant sound quality that is claimed for it. 

And if floor standers that meet the test of accurate midrange while delivering bass energy are a preferred solution, Sonos, via the amp/port still have a place in the system based on merits if preferred as a solution. Which still won't show any audible benefit via Hi Res audio.

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@ninjabob : what happens if you use the option Sonos now gives you to set the buffer at the extreme end - 2 seconds?

I would love to know how to increase the buffer?

@ninjabob sorry my mistake, from being an extensive user of line in jacks, for which Sonos now allows one to set the delay/buffer from as small as 75 milliseconds all the way to 2 seconds. Where line in is being used for audio only streams, the 2 second setting is recommended for stable wireless play. The use of the 75 millisecond setting is recommended where TV lip sync is needed, but this usually also needs ethernet wired Sonos for the audio play to be stable.

It may well be that when Sonos brings HD audio capability to its platform, it will tweak the buffer size, or allow for the Line In option to be made available for all kinds of use.

And even if you insert blank plugs into your jacks, you will see the selection I refer to and I wonder if it then applies to streams originating from the unit for non line in sources. Probably not, but easy to test.

Buffering of files for playback varies with the available memory and the online service in question. It’s typically more than 2 seconds, sometimes tens of seconds. You can test it by cutting the connection to the source (NAS, internet) and seeing how long playback lasts.

@ratty : now that makes it more mystifying - if there is this buffer, why does Sonos have issues with wireless streaming of FLAC files? When Amazon devices easily do HD Video+Audio?

@ratty : now that makes it more mystifying - if there is this buffer, why does Sonos have issues with wireless streaming of FLAC files? When Amazon devices easily do HD Video+Audio?

Personally I don’t have an issue with FLAC and wireless. SonosNet is not over-endowed with bandwidth however, and too many 192/24 streams could (unnecessarily) gum it up. It would be sustained throughput, not buffering, which would be the issue.

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I understand that this was banned by the original post, but I can’t resist.
Hires audio is pointless. Red book CD is as good as it gets and Sonos supports it. So who cares? Why bother investing time, effort, energy, disc space etc to something for absolutely no benefit whatsoever?

I don’t think it’s useful to troll audio-files.  (pun intended).  

Nobody thinks Sonos will ever support more than 16 bit 48kHz.  Yet here you are, trolling.

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Martin is a producer.  His experience with digital sampling theory and the math behind it is probably slim to none.  His position in the realm of “experts” can be likened to the difference between an expert Photoshop user, and the guy who programs the Photoshop application.  You do not need to be an expert on low level image processing to be the former, you most certainly have to be for the latter. 

The same can be said for a guy like Martin vs a guy like Monty at Xiph.  Some of the most absurd, incorrect “facts” I’ve ever heard about digital audio have come from producers and artists.  Just ask Neil Young, he lost his shirt touting the advantages of Hi-res.  Great artist, one of my favorites, but the extent of his actual knowledge about digital sampling couldn’t fill a thimble.

One thing to consider is that the paper spec for Redbook is tight and complete, but the actual clean implementation of that spec was impossible at first (hardware and real world limitations).  Audible digital artifacting and distortion of the brick wall filter was obvious in the 1980’s with a digital workflow.

Sampling theory is very simple to understand, as is Nyquist.  The hard part is implementing and engineering a recording and playbvack system to meet the spec.  It’s been my opinion that the two camps are arguing past each other - one ensconced in the defense of Nyquist and the spec and or best understanding of sound perception by human ears.  The other side is on the engineering and use section of the thing - and the initial shortfalls of digital was never related to the spec, but the impossibility of meeting the spec with the technology of the time.

If the sample rate were increased to 60-80kHz, then a clean anti aliasing filter could have been implemented without a lot of digital processing (which took the better part of the generation to perfect!).  It’s not because of Nyquist and sampling theory.  It’s because you eventually have to build what you are describing.

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Martin is a producer.  His experience with digital sampling theory and the math behind it is probably slim to none.  His position in the realm of “experts” can be likened to the difference between an expert Photoshop user, and the guy who programs the Photoshop application.  You do not need to be an expert on low level image processing to be the former, you most certainly have to be for the latter. 

The same can be said for a guy like Martin vs a guy like Monty at Xiph.  Some of the most absurd, incorrect “facts” I’ve ever heard about digital audio have come from producers and artists.  Just ask Neil Young, he lost his shirt touting the advantages of Hi-res.  Great artist, one of my favorites, but the extent of his actual knowledge about digital sampling couldn’t fill a thimble.


If the sample rate were increased to 60-80kHz, then a clean anti aliasing filter could have been implemented without a lot of digital processing (which took the better part of the generation to perfect!).  It’s not because of Nyquist and sampling theory.  It’s because you eventually have to build what you are describing.

I think it is also worth noting that the AES spec for recording audio is currently 24/96, so they have sufficient bit depth and sampling rate to make the recording to be limited only by the microphones, and have enough margin to your output file (usually 24/44.1 if it is going to be ripped to a compressed format) or decimated to 16/44.1.  

This places the main burden on playback equipment.  It would be trivial today to hve 24/96 playback so you can just take the master file and not anything that’s been decimated to CD quality at this point.  Whether or not you feel there is a big difference or not.  

And these days most of the differences in the sound of recordings has more to do with the mastering direction given by the producer or artist than anything particularly technical.

@buzz : my thinking is based merely on how uncompressed line in works wirelessly in a stable way more often with a 2 second delay instead of one of just 75 milliseconds. Is there something in this that will allow HD audio streams to also work much better with the 2 second delay? Or are these two unrelated and therefore different aspects?

And here I am only referring to the distribution within the home - never mind if the source is NAS or a stable broadband feed.

Compressed or uncompressed relates to the amount of data that needs to be sent over the network. Obviously, less data will result in fewer music interruptions if there is a communication struggle. Lossless compression would be mandatory for HD streams, but this does not result in minimal network traffic.

Severe local issues can limit effective communication with the NAS and communication between the coordinator and members of a Group, stereo pair, or surrounds and SUB.

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They asked for results of a scientific study. Here you go:

http://www.aes.org/e-lib/browse.cfm?elib=18296

 

 It would be trivial today to hve 24/96 playback so you can just take the master file and not anything that’s been decimated to CD quality at this point.  Whether or not you feel there is a big difference or not.  

And these days most of the differences in the sound of recordings has more to do with the mastering direction given by the producer or artist than anything particularly technical.

What would not be trivial is this playback not working on older players that are an inevitable part of multi room set ups - which are Sonos target markets. There is also the question of the higher density streams playing in a stable way wirelessly in groups even where all players are able to decode the bitstreams.

Given the above, that there isn't an audible difference by going down that road other than for the reasons in the last sentence quoted, means that unless the proposed Sonos HD players can downsample to older player compatible formats on the fly, allowing 24/96 playback isn't a trivial matter even if present Sonos products already have the hardware to do this.

 Is there something in this that will allow HD audio streams to also work much better with the 2 second delay? Or are these two unrelated and therefore different aspects?

 

The “delay” is actually a buffer -- similar to Internet radio. Line-In is real time data. Delay does not imply compression.

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Thanks Ratty,

Just read the article. Excellent explanation. Totally agree with the conclusion regarding better 'Masters'.  Best sound I ever heard was from a remastered Mobile Fidelity LP back in the 70's. And to think I sold it for a fiver when CD's came out and I replaced it!

I admit to lazy thinking.

The buffer will only serve for the source unit to get incoming streams without stuttering via the buffer. But since grouped play by definition has to be in perfect sync, no buffer can exist to remove stuttering in downstream units, because it will affect the sync. If this arises due to unavoidable circumstances, the only solutions are ethernet wiring or compressed streams if these serve to overcome the WiFi issues and allow stable music play.

And since it is silly to use compressed HD files, the solution for grouped play of these may have to be wiring alone if there are WiFi issues.

Although this compression may not be so silly - if a better master has been used, even compressed HD music will sound better than CD quality streams from not as good masters.

A question now: What is then the point of the options that Sonos now offers for audio delay for Line In? Assuming that this is just a buffer/delay and no compression is involved. The source feed is via a wire, so why the need for any delay more that the 70 millisecond one that is needed for the ADC/DAC and other Sonos processes?

I also realise that the comparison with Firesticks has a fallacy - they must use a hefty buffer to allow HD play before it degenerates to SD quality if the WiFi issues persist. But if they had to do grouped play in sync, with other Firesticks, I am sure the downstream Firesticks will not be able to cope via WiFi.

Lossless compression will not compromise the quality of an HD stream.

 

Yes, but I am not sure that this will allow enough bandwidth saving to overcome stuttering in HD group play where no buffer can be used.

And for Line In, the compressed mode that is used for stable play in compromised WiFi environments is lossy compression. Although even this is not something that is audibly a compromise to most unbiased ears.

Lossless compression will not compromise the quality of an HD stream.

 

Yes, but I am not sure that this will allow enough bandwidth saving to overcome stuttering in HD group play where no buffer can be used.

Buffering is always used. It’s impossible to deliver a real-time stream over an asynchronous network without a playout buffer. 

 

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With respect to Groups, players might (depending on which nodes are wired and wireless) be facing the polluted wireless environment multiple times, once to fetch the source, once for each member of a Group, plus once for a stereo paired unit in the Group. In a SonosNet wireless mesh, the data must deal the wireless environment as communication is passed through multiple nodes. While a mesh can be self healing, there is actually more data “in the air” than in a scheme built around a central server and access point. In a polluted environment, total data is also elevated as corrupt packets are re-transmitted. At some level of pollution and traffic, any scheme will saturate and break down.

I sort of understand this, the bandwidth is shared between all devices, and can go up exponentially if there are re-transmissions. I’m trying to figure out the ‘best practice’ for minimising the wasted bandwidth in the SonosNet mesh.

I have learnt that starting a group from a wired device gives me best results, is there any advantage from starting it from a wired stereo pair? I recall its the left speaker that is best to be wired if possible?

Thanks!

 

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i can hear differences in sound quality in all the above scenarios.
I don't doubt that you can hear these differences. But the question to be answered is how many of these will survive in a controlled single variable level matched blind test.

Until then, all I can say is that what your subjective experience is, is just that and therefore of not much use to me.

Particularly since I have been there, done that and I have come to realise what matters to my perceptions of sound quality. And what does not.


This I agree - that neither you nor me can generalize based on either of our experiences

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