Sony Super CD: More Bits, More Bucks, Mo' Betta?

Reader dcigary pointed to this "nice writeup on the new Sony Super CD." Though the explanation of the difference between supposedly revolutionary "DSD" recording over conventional digital seems to get by with a knowing mumble, the piece does mention the price (high) and that competition from audio-only DVDs may cripple acceptance of the new format. Even if I like the idea of ultra-fidelity, my faith in the Nyquist theorum is too strong to spend a grand and a half on a CD player anytime soon ...

Comments

Please! Most people think (cough) MP3s sound fine.

[hatless]

So here's a digital format that should please nearly all the classical music afficionados out there who spend tens of thousands of dollars constructing acoustically-perfect "listening rooms". Nothing bad about that. At the very least, it finally creates a reasonably lossless way to digitize analog material for archival and preservation purposes--although any archivist will tell you that the real archives themselves for long-term preservation should be old-fashioned stamped analog discs.

These two markets--archivists and money-is-no-object audiophiles--should be covered with about 20,000 of these devices. So what about the rest of us? I have serious doubts that the difference between this and DVD-Audio can be heard on even a $3,000 home theater system.

Sony (and presumably Philips/Magnavox) intend to build support for this into all of their players starting sometime soon, maybe a year from now. The thing is, nearly all the DVD players being sold today can play the competing DVD-Audio discs. None, not even Sony's, and not any of those millions of Playstation2s shipping in the next year, can play SACDs.

Ultimately, this is about patent royalties. Sony and Philips have been collecting royalties on every CD player and CD drive sold for over a decade now, and SACD is about trying to do it again for another decade. DVD-A is the format endorsed by everyone in the industry except Sony and Philips. Is it a good professional archival format? Nah. Is it both better and more flexble than CD? Yep.

So here's the ugly truth. The MP3 revolution seems to have proven that most people have tin ears. Ask a hundred people. 98 of them will tell you that 128Kbps MP3 is "CD quality". Fact is, it's inferior to Minidisc, to FM radio and--in many respects--analog cassettes. But it doesn't have hisses and pops, and that's all most folks really notice. Heck, 320Kbps MP3 sounds crappy next to a CD, even on a $400 stereo.

If people think MP3 is "good enough"--when it can't even hold a candle to CD--why is the mass market going to embrace SACD over DVD-A? Especially when they'll have DVD-A players available from dozens of manufacturers and SACD players most likely available from... three?

CD will be superseded, not because most people want higher-resolution sound quality they can't hear on Britney Spears remixes, but (1) because DVD-A and SACD players will offer things like 6-channel sound and bundled-in DVD video clips, and (2) because the record industry will stop making CDs, just like they stopped making LPs, in order to force everyone to buy the new players and buy yet another copy of Billy Joel's Greatest Hits to go with the LP, cassette and CD they already have.

The best format won't win. The more ubiquitous one will. The question remains which coalition will blink first. Will the Sony-Philips side break down and allow their record companies to start making DVD-As once they see SACD players aren't selling well, or will companies like Matsushita start paying royalties and buying chips from Sony because the Sony/Philips DVD-A embargo has made it impossible to get record stores to carry DVD-As?

22050Hz

[jbf]

More frequency range isn't going to be recorded, played, or heard by anyone.

First of all, things above 22kHz aren't picked up by ordinary mics... Even the ultra-high-end Neumann U87Ai only claims 20-20kHz frequency response (http://www.neumann.com/mics/u87ai.htm)

Secondly, most speakers won't crank out those high frequencies without a severe falloff in response: the high-end Genelec 1038A triamped monitor gets you 33-20k Hz (-3dB). (http://www.genelec.com/products/1038a/1038a.htm)

Finally, most people can't hear above 20kHz, especially those people who are incessantly blasting their ears out with loud music.

The best reason for Super CD (or DVD or whatever) is higher bit depth, NOT higher sampling rate; going from 16/44.1 (CD quality) to 24/44.1 takes just 50% more space, for nontrivially better quality, while going from 16/44.1 to 16/88.2 brings minimal benefit at a 100% space penalty.

Re: 22050Hz

[bbrantley]

First of all, things above 22kHz aren't picked up by ordinary mics... Even the ultra-high-end Neumann U87Ai only claims 20-20kHz frequency response (http://www.neumann.com/mics/u87ai.htm)

Far from true. The mikes used in this paper, "There's life above 20 KHz!", certainly were capable of this.

Secondly, most speakers won't crank out those high frequencies without a severe falloff in response: the high-end Genelec 1038A triamped monitor gets you 33-20k Hz (-3dB). (http://www.genelec.com/products/1038a/1038a.htm)

Also not true. Unless there is a low-pass filter to prevent sending higher-frequency signal to the tweeters, most amplifiers, speaker wire, and drivers will gladly play sounds upwards of 100KHz. Whether it is necessarily FLAT is another story, as most people don't optimize (or even measure) flatness above 20k.

The best reason for Super CD (or DVD or whatever) is higher bit depth, NOT higher sampling rate; going from 16/44.1 (CD quality) to 24/44.1 takes just 50% more space, for nontrivially better quality, while going from 16/44.1 to 16/88.2 brings minimal benefit at a 100% space penalty.

This is probably true, except that "minimal" may be too harsh a term. Have YOU ever done a careful comparison between a 16/44.1 recording and a 16/88.2 recording? (I have!) On a somewhat-related note, it is remarkably interesting what effect a more accurate clock signal has on the quality of a 44.1KHz recording. The human ear can distinguish playback when the timing of these samples being played back varies by as little as 10^-10 seconds!

The reality is that the human ear's ability to differentiate is remarkably more subtle and complex than the market (and marketeers) would have you believe.

Upgrade media to new format?

[knarf]

Well, another day, another media format. Of course, the media companies will happily sell me their products. But I already have Radiohead's 'OK Computer' on CD, so I already paid the license fees. I want to 'upgrade' that CD to the format-du-jour, and am willing to pay the production costs and a little something to make it worthwile for the industry to keep on developing new products. I do NOT want to pay royalties again, since I already did. And since I have always been told that those compact discs are so expensive because of the license fees, this upgrade should be quite cheap, am I right? I mean, I only OWN the piece of plastic, which is cheap. It is the license fee which drives up the price (or so 'they' say). So, just let me upgrade my piece of plastic then...

No, unfortunately I am wrong. But I should be right...

Re:Nyquist theorem

[nathanh]

Nyquist's theorem states that the highest frequency that can be represented is one half the sampling rate. This is obvious because you must be able to detect at least a peak and a valley of the sound wave.

Entirely correct.

Nyquist's theorem does not imply, however, that the representation of the maximum [or near maximum] frequencies will be highly accurate as far as the shape of the wave form is concerned. At and around 1/2 sampling freqency, the wave forms become basically nothing but square waves [alternating between a single high, and a single low point]. In order to deal with this, some sound decoders will attempt to interpolate the waves, but they cannot reproduce the original sound accurately. This is why higher sampling frequencies ARE relevent to higher audio fidelity. Higher bit resolutions are arguable though..

You fail! This idea that the signal is not perfectly represented just because you have only two sample points is complete nonsense. Only two sample points are needed because you know the encoded signal must have been low-pass filtered at half the sampling rate before sampling (otherwise you would have introduced aliasing errors). Given this information you can entirely reproduce the original signal as it was before sampling. Nyquist's theorem states that you can exactly reproduce the signal if sampled at twice the signal's maximum frequency. I quote Oppenheim and Willsky:

The sampling theorem establishes the fact that a bandlimited signal is uniquely represented by its samples.

In layman's terms: you don't need more bits to reproduce the original signal. You just need a perfect low-pass filter on your output and infinite precision on your PCM samples. A sine wave with sampling points at the exact peaks and troughs will produce a square wave of the same frequency after sampling/modulation. This square wave will contain the frequency you want plus odd harmonics. The harmonics are naturally going to be higher frequencies and so they will be removed by an appropriately picked low-pass filter. And what's the appropriate cut-off frequency for your low-pass filter? 1/2 the sampling rate, of course. The result is the original sine wave.

Now in practise they actually do sample at higher than the low-pass cut-off frequency, but this is because of other limitations. The PCM samples are only 16-bit, not infinite precision. Also there is no such thing as an ideal low-pass filter: realistic (and affordable) filters will take several kHz to drop from 0dB to -9dB. Also you need exactly -/2 phase difference between your sampling pulse train and the source signal. There are also aliasing issues but at this point the discussion gets heavily into mathematics.

Higher resolution is what is actually needed but this is expensive to achieve. Increasing the sampling rate is far more practical (considering how fast CPUs are) and a heck of a lot cheaper. This is the real reason DVD audio samples at 96kHz. It's not because you can hear 48kHz tones but because it lets the DVD manufacturers use cheap DACs and cheap low-pass filters without sacrificing fidelity.

General Niftiness :)

[Chris Johnson]

I _hope_ this becomes common technology- there are some extraordinarily important things about it. A little background:

• 16 bits isn't enough. That's _really_ obvious at this point- no professional works in 16 bits except for the final CD output. Mix busses have to be many times that in the digital domain, but even if you mix with ideal noiseless coloration-less electronics there's a really big difference between monitoring an undigitised feed of the signal with monitoring the 16 bit output.
• 44.1K isn't enough either. This is not primarily due to people being able to hear beyond 20K (though you can sense such sounds to some extent- why do you think smashing glass or dropped plates make you jump? Viciously loud supersonic transients), it's due to the brick-wall filters required. High end amplifier designers go to great lengths to get their pass-bands up into the megahertz (and nobody claims humans hear that!) because cutting off lower causes interactions across the entire frequency band. Cutting off at 22K is just ridiculous.

Now, how does the Sony approach compare? The neat thing about the bit rate is that it's effectively infinite bit rate- it's not a finite set of voltage levels but just one bit very fast tracing a voltage level that could be anywhere. This is substantially beyond even 24 bit- a major, major advance. That's gonna be very noticable.

As for frequency, there is a surprise in store here. It may or may not be competitive with advanced PCM encoding at say 96K- but two very, very important points:

• There doesn't need to be _any_ brickwall filter on the output- provided a circuit can be made to output this stuff that doesn't merely calculate it as a super-PCM-encoding and D/A converter. If the format can feed a sort of very high frequency analog synthesiser, no filter is needed- which is critical, because...
• ...the potential slew rate of this technology is just astronomical. I hope the power supply of the players is up to it- if not there will be some very effective power supply mods waiting to be done, such as backing up the power supply with MIT Multicaps (a film cap that can produce very very high instantaneous voltage). Basically, if you fed this technology a big square wave, it might not be able to turn the corners of the wave instantly, but the vertical parts of the wave would be _vertical_- no brick-wall-filtered system can get anywhere close to this.

We're talking absurdly high transient peak voltages here: this is why high end audiophiles use absurdly heavy cables and absurdly powerful amplifiers, to let those peaks through. It doesn't hurt the speakers: this isn't RMS or even 'peak' wattage, the spikes are of such short duration that you can feed speakers many times the maximum 'peak' voltage if it's only for a microsecond, and high end systems do just that.

Where do you find such peaks? Easy- The Who ;) seriously, The Who is a _good_ example, but symphony orchestras are also good for this. The capacity for this type of extreme and essentially 'inaudible' (too brief!) transient translates to the ability to produce the _sensation_ of loudness- for instance, you could easily make many systems play 'Live At Leeds' and sound loud and bright and kind of grating and ear-splitting, but with this technology it would be less grating but more _electrifying_ and the impact would be like having the living people right there playing at you, not just a bunch of very loud sounds. Alternately, you could play big orchestra crescendos and the resulting sound would be _huge_, not just loud but as big as a live performance.

It's really not hard to make stuff sound 'loud', but making it _feel_ loud is something else. If you don't have that, the loudness ends up being just a grating, thin surface, which is actually a very good description of the sound of most pop recordings these days :) the irony is that this technology is coming around just when the recording industry's pushing sounds that are substantially worse than even CD audio can produce...

Bottom line: I want one. Specifically, I want this to _master_ to. I have quite a bit of stuff that loses about 2/3 of its potential when made into 44/16 (eight tracks of 48/20 output analog and mixed with passive resistance mixing will tend to do that- I once figured the rough equivalent resolution was about a 64 bit mix bus, possibly higher) Maybe I should try to wheedle Sony out of a recorder ;)