Soundbites: News in brief from the Wi-Fi audiophile files Listen up! Here are some soundbites - so to speak - from Wi-FIispeaker/player suppliers. Sonos is intending to provide access to Deezer Elite streaming lossless audio for its US customers. It was announced in a Sonos blog and beta testing started on September 15 with a 30-day free trial. Deezer, it says, is only available on …
“Music quality has gone backwards from vinyl to CDs to MP3s and Deezer Elite reverses that trend.”
That isn't backwards, that is a bell-curve...
I love my LP's, but suggesting that the format allows music of higher quality than red book cd? a very backwards statement to make indeed.
"We understand 16-bit FLAC is equivalent to CD-quality and much better than the lossy MP3 files we’re all used to. You can get better-than-CD quality with 24-bit audio and DVD-Audio uses this with a 96kHz or 192kHz sample rate, both giving many more samples per second than 16-bit FLAC at 44.1kHz."
Almost right and then just wrong - 16bit/44.1Khz FLAC file isnt' the equivalent to "CD-Quality", it is them _same_ as "CD Quality".
Better than CD quality?
24bit audio - yes, this can increase the resolution of a recording and in theory allow for greater musical fidelity. Difficult to tell the difference in practice, but yes, at least its there (as long as its all the way from the studio master at least).
96Khz or 192khz? - 44.1 Khz is a higher 'resolution' already than my (and your) ears can hear. At 96khz and with the right equipment you are either tormening your dog or delighting him with bits of your music that you will never hear.
Silly Sono :)
"96Khz or 192khz? - 44.1 Khz is a higher 'resolution' already than my (and your) ears can hear. At 96khz and with the right equipment you are either tormening your dog or delighting him with bits of your music that you will never hear."
Higher sample rates improve the phase accuracy of high frequencies, not an issue for mono but in stereo it helps you pick up the direction a sound is coming from.
Higher sample rates also make the anti-aliasing filters much easier to design, and reduces their effects on the audio you can hear.
Higher sample rates are useful, you better define of the sound at e.g. 16kHz:
With 44kHz sampling you get 3-4 samples per lambda, so you can tell it's there, but nothing else. With 96Khz you're sampling it nearly 8 times per lambda, with 192Khx that's 16 points.
Draw a sine curve with only 3-4 points, now do the same with 16.
Now assume you don't hit the perfect sync between them, you might have two points at "-50db up" and 2 at "-50dB down", then repeat. Is that a sawtooth, a sine wave, a square wave? They all sound different. Take 16 points and you have defined the shape of the wave, better allowing you te recreate it.
By the time it comes out of most home speaker / headphone combinations then I'll agree. But when I'm recording I'll go higher rate, then drop it down as the very final step.
I think you'll find square waves and saw-tooth waves are synthetic special cases which are made up of lots of high harmonic frequencies. You really have to switch from thinking of the waveform as a series of points, to a representation of the true waveform with higher freqencies filtered out (see https://www.xiph.org/video/vid2.shtml)
Google "Nyquist" sometime. Essentially the jist is you can *flawlessly* reproduce any frequency, by sampling at double the rate.
So if you want a flawless 22khz signal (which you cannot hear), you sample at 44khz. Hence why most music is sampled at 44.khz.
It has absolutely zip to do with the smoothness of the output waveform. You will get a flawless sine with 2 points per cycle.
You've been reading the Wikipedia article on the Nyquist Frequency, and particularly the section on Aliasing sinusoidal waveforms.
This is a very special case, and does not mean that you can reconstruct any waveform from a sample of 1/2 of the frequency of it's highest component. It's really pointing out the minimum sampling rate that allows you to differentiate between one sine wave and another with an integer multiple of the it's frequency. The important thing is that you have to know is that it is a sine wave before you start.
There are many special cases, and the one that I like to think of is a sine wave at 1/4 of the sampling frequency, which at 44.1 KHz sampling, would make the frequency of the sinewave 11.25KHz, well within the hearing range of most people. This would mean that if sampled at exactly 90 degree intervals, you would get something between a perfect sawtooth and a square wave. Of course, if you know it is a sine wave, you can reconstruct it, but on a CD player it would be stupid to assume that everything you play will be a sine wave, so it tends to use some mathematical spline to smooth the waveform, and this is what will be fed to the analogue part of the system. Different implementations of CD use different smoothing functions, but none of them can perfectly reconstruct the original signal in every case.
As has been pointed out, this is a pathological case, but it illustrates that digital sampling can never be anyway close to perfect unless the sampling rate is many times the maximum frequency, certainly more than twice, whereas a mechanical system could be perfect within a range of frequencies, even though it is unlikely to be so because of material physics.
To say that the limit of human hearing is 20khz (or whatever) is to say that the highest frequency a human can hear is 20khz. But there's only one "sound" that's completely defined by it's frequency and that's a sine wave. If you could tell the difference between a 20k sine wave and a 20k square wave (or anything else) it would be because you were hearing the higher frequency components which you can't do - your ears don't have hardware to respond to those frequencies.
Stop listening to shit music then.
It's not about shit music, it's about shit post production. Today's sound engineers seem to have only one aim - make it LOUDER. This results in digital 'remasters' that sound like they were recorded on a 70s cassette recorder with ALC, every breathy cymbal becomes an explosion, all the instruments disappear into noisy audio mud.
Digital recording promised us a dynamic range way above 100dB, but the music industry decided to just use those two at the top.
"Pleasantly surprised...by the absence of audiophile twattery on this thread."
Yeah, that's because on this forum the twattery is all about the Nyquist theorem, sampling rates, network packets and Wi-Fi interferometry (oh, I made that last one up).
The principle is the same and each group feels intrinsically and infinitely superior to the other - would be interesting to put both in one room and watch the proceedings on CCTV :-)
We don't have TV/Monitors which include infra red and and ultra violet light as humans cannot see in that range. Same with Sound. We hear in a finite range which is fully catered by the quality of CD. Anything more is just noise for dogs, bats, elephants and marine life. Sampling at hight bit rates is good for production but adds nothing to consumer playback.
"We understand 16-bit FLAC is equivalent to CD-quality and much better than the lossy MP3 files we’re all used to. You can get better-than-CD quality with 24-bit audio and DVD-Audio uses this with a 96kHz or 192kHz sample rate, both giving many more samples per second than 16-bit FLAC at 44.1kHz."
Of course you can also get 24-bit FLAC at 96kHz or 192kHz too. Sonos just doesn't support it (or didn't at least, haven't checked recently)
As has already been said here you almost certainly won't hear the difference between 16-bit 44.1 and higher with the important proviso that both are made from the same master. I've bought a few 24-bit flac recordings simply because they are mastered better than the CD version, Muse's 2nd Law for example which seems not to have everything turned up to 11 unlike the cd.
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