Zp 24/96

As a (chartered) electronics engineer I agree with Keith's view! I thought about what you said, and while "utter rubbish" may be a bit heavy, nevertheless it's a fact! Why not just concentrate on enjoying the wonderful music experience that the Sonos system facilitates?

How does that work then?
See http://people.xiph.org/~xiphmont/demo/neil-young.html#toc_1ch

Distortion from the ultrasonics appearing in the audible range, as explained at the top of the section I linked.

Neither audio transducers nor power amplifiers are free of distortion, and distortion tends to increase rapidly at the lowest and highest frequencies. If the same transducer reproduces ultrasonics along with audible content, any nonlinearity will shift some of the ultrasonic content down into the audible range as an uncontrolled spray of intermodulation distortion products covering the entire audible spectrum. Nonlinearity in a power amplifier will produce the same effect. The effect is very slight, but listening tests have confirmed that both effects can be audible.

Right, so it's not inaudible then is it?

The frequency itself is inaudible, the affect on the electronics is not. In other words, you cannot hear a pure 30 kHz tone, such as a dog whistle. Blow a dog whistle, you will hear nothing. But record a dog whistle and play it back, you may then hear distortion caused by the strain on the transducers and/or amplifiers. This distortion is purely an artifact of the electronics, the recording only has the inaudible 30 kHz tone.
So that would imply that so-called "super-tweeters" that reproduce frequencies far above what we can hear could be a good thing by removing that "strain" from the tweeter's transducers right?

So that would imply that so-called "super-tweeters" that reproduce frequencies far above what we can hear could be a good thing by removing that "strain" from the tweeter's transducers right?

Firstly it's not just the speakers: it's every electrical/electronics component in the system. That means your amplifier as well.

I guess it would be possible to make an audio amplifier which handled ultrasonic noise with less intermodulation than conventional amplifiers, but it would be a lot of effort and you would end up with a very expensive piece of equipment. As far as I know no one makes such a beast.

Secondly, even if you can totally remove the audible artefacts that could be caused by these ultrasonic frequencies, the fact that the amp and speakers has to handle them in the first place puts an additional load on the system: you are burning off some of your system's power on something that, at best, you will never hear.

Ultimately what would be the point? The problem is artificial and is self-created (by using hires files). It seems a bit daft to create such a problem and then spend time and money trying to find Engineering solutions to cure it.

The best way to reduce the impact of ultrasonic noise is to filter it out. A good way to do this is to downsample to 16/44.1 or 16/48.

Cheers,

Keith

Are there other effect (maybe implication is a better word) to the sampling rate?

(Not an electrical engineer here, mostly curious)

I am thinking that higher sampling rate would give you more "interpolation" points in between the discrete therefore possibly a better resolution overall resolution. But maybe it is not a factor or it is not worth the extra points.

Thanks

Mathematically, there is no need for a "more interpolation points" in order to recreate the entire original signal. Per the Nyquist–Shannon sampling theorem, a perfect reconstruction of a signal is possible when the sampling frequency is greater than twice the maximum frequency of the signal being sampled. Therefore, for all audible frequencies (up to 22 kHz), a 44.1 kHz sampling rate is more than enough to create a perfect copy of the original audio signal.

The idea that two ultrasonic signals can collide with audible results is called "intermodulation". Even though nether a 40kHz or a 42kHz signal can be directly heard by humans, if they pass through a nonlinear system, a 2kHz tone will be clearly audible. This is a case where two "out of band" signals will interact and cause the creation of an "in band" product.

Further, since the 2kHz intermodulation product is in band, it will be subject to the usual distortions that create harmonics and further intermodulation products with the program material.

In the end, these two inaudible signals create a blizzard of in band, very audible problems. And, they are easily preventable by blocking the ultrasonics.

Actually the SACD has a CD layer that would be sourced from the same master. You can listen to either one on most universal players. (An egregious exception is the SACD of "Dark Side of the Moon" The CD layer on that was deliberately compressed to make the SACD version sound better.)


Read what I said: "You cannot compare a production SACD to a production CD." Meaning you cannot compare a SACD to it's regular CD version. I said nothing about comparing the SACD layer to the CD layer on the same SACD, for I have no data on whether they are the same mix or not. However, I do know for a fact there are SACD versions which are mastered differently than the standard CD version.

That being said, if you really want to improve sound, there are other things that will make much more dramatic improvements--improvements that you won't have to ask yourself if they are really there. First, buy the best speakers you can afford and take the time to set them up properly. Second, control vibration and EMI/RFI as best you can. Third, and IMO most important, treat your room. Strategically placed bass traps and other absorbers/diffusers made more of an improvement in my listening experience than any other change I have made over the years.

Finally, thank you Sonos for providing an incredibly good product that has given me great enjoyment over the past few years. And please give us hi rez.

The first part of the above is spot on - I would just add speaker positioning in the room to factors that have audible impact, and I guess that is what you allude to in your reference to proper set up. Once you have addressed these, hi rez won't make an ounce of audible difference. Too many people forget to address these issues that deliver 90-95% of the quality of the heard sound in these days of reliable solid state electronics upstream of the speakers.

Unless - as has been pointed out often here - the hi rez files have superior mastering. And in which case, the better sound is solely on account of this mastering.

Sonos has already indicated that the idea is not under consideration. For the short term, get used to it and down-convert (bit perfectly) your music to a format that Sonos can play. Longer-term, you can politic Sonos for a change...

I am used to it and do downconvert. I am very pleased with how things sound through the Sonos. I'm just curious.



And they would be wrong. Read the relevant information posted via links in this Thread. Science trumps the mythical "Golden Ear".

You're probably right, but I would like to know for sure. Science told us when CDs first came out that they were perfect. I listened for awhile and went back to vinyl until the CD production and CD players got their acts together. When the CDs got to the point where they sounded better in my system, I digitized my albums and went 100 percent digital. My point is that scientific orthodoxy sometimes changes over time. I wish I could be put into a blind test comparing 16/24 and 24/96 so I would know for sure if there is an audible difference.



Sure they should. Especially when the facts (not opinions) are decidedly against the "high-res" cohort.

Well, they're the guys who do this for a living (making great records), so I can't know for sure. If there is some infinitesimal improvement that one would have to strain to hear, it wouldn't be worth it. As I said before, you are probably right.



Good luck. Unfortunately, it will not be by using Sonos, at least anytime soon.

Agreed. I am sticking with Sonos because listening to music through it makes me happy.

Best of Luck

Same to you.

The argument for higher sample rates has nothing to do with trying to capture and play back higher frequency sounds. Anyone who discounts higher sampling rates because humans can't hear above 20kHz simply doesn't understand how these things work.
When you convert a signal from analogue to digital (sampling) you must first remove all content in the signal which is at half the sampling frequency or above. This is done with a low-pass filter. For CD, we want to preserve content up to 20kHz while removing 'everything' above 22kHz. This requires an extremely steep filter. Such filters are very hard to engineer, and typically will suffer from defects such as pre-ringing and poor handling of transients.
When we convert from digital to analogue (playback), artefacts are again produced which are related to the sampling frequency. Again, a very steep low-pass filter is required and again this cannot be a perfect filter. After CD was launched, many of us quickly realised that the promise of 'perfect sound forever' had not been realised. Two of the causes were: jitter (not relevant to this discussion) and the extremely steep low-pass filters required during recording and playback.
The best engineering solution was to increase the sample rate. If we sample at 96kHz, then the low-pass filters can be much less steep.
It was easy to migrate the recording studios to 96kHz, which solves the sampling problem. But the CD sample rate was by then too well established, so we still have the problem of trying to playback a signal with content up to 20kHz from a digital format whose sampling rate is only just high enough in a perfect world with perfect filters.

Let me put it another way - when the engineers were deciding on the sample rate for CD, the thinking went something like this:
Well we know we can't go below about 40kHz, because sampling theory tells us that would be flat out impossible. But the guys working on the lasers and the read mechs and the discs and the DAC chips are really struggling with bandwidth and capacity. They say we have to find the absolute lowest sampling rate that we can get to work. Lets do some tests and see how close to 40kHz we can get and still make this work. I know we'll be stuck with some issues but we have no choice.

Its no accident that audio formats which have launched since CD have supported higher sampling rates. If the argument for 96kHz was based on capturing and playing back sounds above 20kHz, as some on here have claimed, then I don't think many would be gullible enough to buy into it.

The argument for higher sample rates has nothing to do with trying to capture and play back higher frequency sounds. Anyone who discounts higher sampling rates because humans can't hear above 20kHz simply doesn't understand how these things work.
When you convert a signal from analogue to digital (sampling) you must first remove all content in the signal which is at half the sampling frequency or above. This is done with a low-pass filter. For CD, we want to preserve content up to 20kHz while removing 'everything' above 22kHz. This requires an extremely steep filter. Such filters are very hard to engineer, and typically will suffer from defects such as pre-ringing and poor handling of transients.
When we convert from digital to analogue (playback), artefacts are again produced which are related to the sampling frequency. Again, a very steep low-pass filter is required and again this cannot be a perfect filter. After CD was launched, many of us quickly realised that the promise of 'perfect sound forever' had not been realised. Two of the causes were: jitter (not relevant to this discussion) and the extremely steep low-pass filters required during recording and playback.
The best engineering solution was to increase the sample rate. If we sample at 96kHz, then the low-pass filters can be much less steep.
It was easy to migrate the recording studios to 96kHz, which solves the sampling problem. But the CD sample rate was by then too well established, so we still have the problem of trying to playback a signal with content up to 20kHz from a digital format whose sampling rate is only just high enough in a perfect world with perfect filters.



Everything you say is completely true . . . for recording and mastering!! Unfortunately, it does not apply to playback. Matter of fact, ultrasonics present in high sample rates can cause intermodulation in the transducers and amplifiers used for playback, making 24/96 recordings sound worse instead of better.

The are many valid and proper reasons for using higher sampling rates and bit-depth when recording and mastering, which you have stated eloquently. This is why it has become industry standard to oversample for recording and mixing. The snakeoil comes in when you try to apply these valid and proper reasons to the playback of those recordings, where they simply do not apply (or even cause distortion which is not present in a comparable 16/44.1 file).

This point about recording/mastering vs. playback has been stated many times in the thread, and was also emphasized in the included links. You would do well to educate yourself by reading them.

Let me put it another way - when the engineers were deciding on the sample rate for CD, the thinking went something like this:
Well we know we can't go below about 40kHz, because sampling theory tells us that would be flat out impossible. But the guys working on the lasers and the read mechs and the discs and the DAC chips are really struggling with bandwidth and capacity. They say we have to find the absolute lowest sampling rate that we can get to work. Lets do some tests and see how close to 40kHz we can get and still make this work. I know we'll be stuck with some issues but we have no choice.

Actually, the sample rate was chosen by Sony because it could be recorded on modified video equipment running at either 25 frames per second (PAL) or 30 frame/s (using an NTSC monochrome video recorder) and cover the 20 kHz bandwidth thought necessary to match professional analog recording equipment of the time.

Its no accident that audio formats which have launched since CD have supported higher sampling rates. If the argument for 96kHz was based on capturing and playing back sounds above 20kHz, as some on here have claimed, then I don't think many would be gullible enough to buy into it

Given this is an industry which sells $10,000 speaker cables, $2000 Cocobolo wood component feet, and green magic markers, you greatly underestimate the gullibility of the average audiophile.