herbie@watdcsu.UUCP (Herb Chong, Computing Services) (10/02/84)
Some people have been complaining about the sound of Sheffield Lab's CD releases. One of the problems with digital recording as attempted in the manner that Sheffield does is that the digital recorder does not have enough dynamic range. We have people in our physics audio lab designing and building a dynamic range compression device for digital recorders because of that. One theory has it that a digital recorder with at least 120 dB SNR is required to provide adequate dynamic range for live digital recording. If you record to capture the peaks of something like LAB-2 (I've Got the Music in Me by Thelma Houston), the lowest levels will be about 40 dB below that. This leaves you with 50 dB SNR, which is fine for analog systems, though not too listenable, but not for digital systems. This ratios is a voltage difference of only about 300 times. This means that the lowest range is quantized with only about 8 bits (2^8 = 256). What does 8 bit quantization sound like? I've heard demos at 10, 8, and 4 bits quantization. Something sounds wrong at 10 bits; at 8 bits, everything sounds harsh; at 4 bits, the sound is barely recognizable. A compressor can be used to place the signal up where more bits are being used and quantization error is not as noticeable. This is a problem with all digital recording systems handling a high dynamic range signal. More bits are required, or a signal compression system, to get around this. Of course, 120 dB SNR is hard to maintain in analog equipment, and a digital system using linear encoding would require 20 bits to get that much SNR, and finally, the number of bits generated would be 1.5 times as great. Any comments? Herb Chong, BASc Computer Consultant Department of Computing Services University of Waterloo
shauns@vice.UUCP (Shaun Simpkins) (10/10/84)
Something bothers me about this dynamic range business. Let's assume that we have a recording medium with 120dB dynamic range. Further, let's assume that MOL (maximum output level) of the recorder will produce a SPL of 130dB, the threshold of pain to the human ear. This means that the noise level of the recording medium will be at 10dB, 10dB above the threshold of hearing in an anechoic chamber and 30dB below the noise floor of a quiet listening room. We now substitute a 16-bit digital recorder for our hypothetical system, placing its MOL at the same point. Its noise floor will now be at 34dB SPL, still below the noise floor of a quiet listening room by 6dB. Put another way, the difference between ambient noise and a music signal is one bit. Will a music signal sufficient to twiddle only the LSB of the digital recorder sound bad? Maybe, but I claim that you won't notice it, since you'll be straining so hard to filter out all the ambient noise. I suspect that the digitization study cited used signals far above the noise floor of the listening chamber. What happens when the signals digitized at 4, 6, or 8 bits are played at the SPLs where they would normally occur? I suspect that distortion components would be lost in the noise floor. Even 10 bit quantization noise is lost in the environment in my example - and LSB dithering would make it even more indistinguishable from random noise. The only time such errors become perceptible is in a highly unlikely environment, the anechoic chamber. Given this, I find it hard to believe that LPs suffer less from dynamic range problems than CDs. Good vinyl, I am led to believe, has a dynamic range of about 70dB. Allowing for the ear's adaptive filtering (5dB? 10dB?) this is still a 16-20dB less than 16 bit PCM. Is the ear really that bothered by PCM's inverted distortion vs amplitude characteristics? Yes, professional recorders need to be better than the consumer environment for many reasons - Multitrack mixdowns and foresight being two. It does not mean that the consumer product must be as good. I challenge you to make a 20-bit digitizer with a 10 volt full code output. The LSB would be 10uV! Try that on comtemporary IC processes over the commercial temperature range. A final comment - I like the idea of compressing CDs, but more for stereo system protection than for ultimate dynamic range. If this is done, the compression should be multiband and the level of compression indicated by an explicit control track instead of by sensing the level of the compressed output and supposedly ``undoing'' the change, as is done now. This would reduce system distortion to levels approaching -80dB(MOL) rather than the -40 to -60dB(MOL) possible with today's systems (such as dbx), and could be easily implemented by an extra control word in the CD bit stream with appropriate sorting in the player. Of course, this would entail perhaps a 30% in playing time, about the same as 20 bit PCM but a whole lot more manufacturable. The wandering squash, -- Shaun Simpkins uucp: {ucbvax,decvax,chico,pur-ee,cbosg,ihnss}!teklabs!tekcad!vice!shauns CSnet: shauns@tek ARPAnet:shauns.tek@rand-relay
karn@mouton.UUCP (10/10/84)
On every CD I own, both those recorded from analog masters and from digital masters, the recorded-in noise level from microphone amplifiers, mixers, ventilating systems, etc, exceeds the quantizing noise level by a considerable margin. This provides a "self-dithering" mechanism that completely masks quantizing noise. The only time I've been able to hear quantizing noise on a CD is during a silent passage at the end of a recording where the recording mixer must have ramped the master gain down. The recorded-in noise breaks up and becomes irregular for an instant just before it disappears. Mind you, this is occurring at extremely low signal levels -- my volume control had to be all the way up to hear this. Phil
herbie@watdcsu.UUCP (Herb Chong, Computing Services) (10/11/84)
Which is all true, but remember that quantization noise is like IM distortion, it is audible at levels that harmonic distortion is not audible at all. I was attempting to point out that there is more to the sound of "digital" that meets the ear. For all intents and purposes, dithering the digital signal itself removes all the problems with quantization noise by randomizing the components of the noise. As for dynamic range offered by a 16-bit linear system being inadequate, it has nothing to do with the continuous output power. One can easily generate a pulse of about 100uS that can require thousands of watts to reproduce accurately, and yet not sound very loud compared to the threshold of pain. Imagine a bolt of lightning striking about 500m away. It has a peak SPL somewhere around 140+dB. You aren't immediately deafened and you don't fall to the ground in pain because the pulse didn't last long enough to transmit significant energy to your eardrum. The oft-quoted SPL's for pain is with a signal that is relatively long-lived by comparison, say one or two seconds. Bob Carver has recognized this and has made his amplifiers to have tremendous reserve power, but only for the short time it is really needed. It works because music is, by nature, transient. He has also observed that, if it were practical to do so, an amplifer capable of peak power more than 2000W/ch for about a mS is needed for accurate peak reproduction with typical speaker efficiencies. Make of it what you will. Herb... Once a hack, always a hack... UUCP: {decvax|utzoo|ihnp4|allegra|clyde}!watmath!watdcsu!herbie CSNET: herbie%watdcsu@waterloo.csnet ARPA: herbie%watdcsu%waterloo.csnet@csnet-relay.arpa BITNET: herbie at watdcs
mat@hou4b.UUCP (Mark Terribile) (10/11/84)
>A final comment - I like the idea of compressing CDs, but more for stereo >system protection than for ultimate dynamic range. If this is done, the >compression should be multiband and the level of compression indicated by an >explicit control track instead of by sensing the level of the compressed >output and supposedly ``undoing'' the change, as is done now. This would >reduce system distortion to levels approaching -80dB(MOL) rather than the >-40 to >-60dB(MOL) possible with today's systems (such as dbx), and could be >easily implemented by an extra control word in the CD bit stream with >appropriate sorting in the player. Of course, this would entail perhaps a >30% in playing time, about the same as 20 bit PCM but a whole lot more >manufacturable. I don't think it would have to cost 30% in play time; it would seem that the envelope used for compression would need a bandwidth of only 100 Hz or so and even 1f you used 16 bits for the compression amount, you would be taking less than 1/2 of 1% off the usable bandwidth. But for listenability reasons, especially in cars, you might very well want to be able to NOT decompress the signal. This would be most useful if the mastering engineers and the performers/writers had a say in what amount of compression was used. I like this idea, mainly because IF compression is needed, it ought to be in the hands of the artist. Let the artist decide how he will ``compose'' for the noisy environment. -- from Mole End Mark Terribile (scrape .. dig ) hou4b!mat ,.. .,, ,,, ..,***_*.
jj@rabbit.UUCP (10/11/84)
Uh, oh! JJ's been listening to a CD player. Last night I was listening to some Strauss and a few chamber pieces on a CD player (Magnavox, of all kinds). I was strongly impressed by the Strauss, and disgusted by the miking on the chamber pieces. The alleged "digital sound" was quite evident on the chamber works, and totaly missing on the two-miked Strauss. Needless to say, the chamber music was dreadful. The Strauss (well, the reproduction thereof, at least) was absoulutely wonderful, with a good soundstage, a VERY low background, and a dynamic range that let my speakers do a bit of what they were designed to do. This experience clearly shows that at least some of the demons of digital are recording demons that aren't even digital related. Who knows, I might even BUY one of these Magnavox players, they work nicely. <I tried the shock test, a blot test, and general abuse, and got nothing but music.> I've listened quite a bit to the Sony CDP101, and I do believe (although I did NOT have a chance to AB) that the Magnavox sounds better in some unclearly defined way that my ears perceive as transient handling. Now, I think I HAVE to buy that new cartridge! :-) Anyone have any feeling for the reliability of the Phillips/Magnavox CD players? I don't, this one was brand new. -- BE KIND TO SOFT FURRY CREATURES, THE LIFE YOU SAVE MAY BE YOUR OWN! "The car was stalled, that fateful night, ..." (allegra,ihnp4,ulysses)!rabbit!jj
emrath@uiucdcsb.UUCP (10/12/84)
Regarding Shaun's point about compression: It sounds like what you are saying is that a CD could be made where the data is 16 bit samples, but every so often a control sample is read which (digitally) controls the gain of the analog (de)amplifier used on the output of a 16-bit D/A. Assuming you could make the original recording using a 20 or 24 bit A/D (maybe someday), the samples could be converted to 16 bits by shifting small ones left and setting the control words to decrease the gain of the playback amplifier by 6dB for each bit the following samples are left shifted. Naturally, the 20 -> 16 bit conversion looks ahead to make sure that no MSBs are lost, only the LSBs of numerically large samples. The recorded signal is compressed, and proper playback entails downward expansion. It would probably sound bad without proper playback. One possible problem with this may be the inability to switch the gain of the PB amp fast enough and at the exact times, thus causing distortion and noise. (also, as you pointed out, its of dubious value) Any way, the whole idea raises the following question. Do current CDs have data format version information stored on the disk, allowing future format changes? (they must, I suspect) A drastic change, such as the one above, might mean that version n players would reject version n+1 disks, but a version n+1 player should be able to play any disk of version 1..n+1. Recalls the evolution of the cassette, with CrO2, Dolby B, Dolby C, dbx, Metal.... Perry Emrath ...pur-ee!uiucdcs!emrath DCS, U of IL emrath%uiuc@csnet-relay.arpa "Life is a bitch, and then you die"
shauns@vice.UUCP (Shaun Simpkins) (10/15/84)
If I remember my dbx specs right the vca has an attack time of less than a millisecond and a much longer decay time. They give some reasons relating to audibility of the gain riding for these times and suggest mightily that they are black magic. Who am I to argue? Anyway, this implies a control track bandwidth of at least 1KHz, perhaps 2 or 3KHz. Roughly speaking, every tenth word will be a control word if the control track is interleaved with the normal CD data stream. So, perhaps 10% play time reduction, not 30-50%. 100Hz compansion contoller BW sounds way too slow \- in particular for isolated percussive sounds, the bane of compression systems. The wandering squash, -- Shaun Simpkins uucp: {ucbvax,decvax,chico,pur-ee,cbosg,ihnss}!teklabs!tekcad!vice!shauns CSnet: shauns@tek ARPAnet:shauns.tek@rand-relay
newton2@ucbtopaz.CC.Berkeley.ARPA (10/22/84)
I reiterate that a narrow-bandwidth gain-control channel suffices. First of all, even if the notion of attack time sufficed to define the required bandwidth, there is no reason to resolve the gain-control information to sixteen bits (the implication of "every tenth word a control *word*"). That's like a floating-point representation with a sixteen-bit mantissa *and* a sixteen-bit exponent. All this to solve the alleged problem of "only" sixteen bits of resolution in the present rather impressive system. If you must have a compander "track", surely gain changes of a few multiples of 6 dB (i.e., a few bits per control word) should satisfy the yearning for more than 90-odd dB of dynamic range (personally I think the present system is all one could reasonably ask for noise/dynamic range wise, when implemented properly). Secondly, on the matter of bandwidth: Regardless of the required instantaneous rate-of-change of channel gain, the measure of control-track bit rate is the average, not the peak, rate of change to be accommodated by the compander. Since the master encoder that cuts the CD-of-the-future cum control track can easily survey the signal to be compander-encoded throughout the program's time history (or at least sufficiently "in advance" of the transient to be encoded), the gain-control bits that are interleaved with the program can appaear sufficently in advance of the gain-change they command to allow the reproducer channel gain to respond exactly on time. I'm not sure I've been clear on this: the recorder (being digital already) includes a digital program delay line, which allows the gain- control bits, calculated in real time with respect to the incoming program, to be interleaved with the delayed program as it emerges. In the reproducer, the gain control information is extracted and decoded before the program data to which it applies (and which gave rise to it) arrives. Thus the level-detectors and VCA's can compensate rather exactly for the tardiness of response inherent in the low bandwidth of the gain- control channel. This principle is the basis for a patented low-frequency pilot tone compander compatible with comventional FM stereo broadcasting, and this same notion of delaying the program to allow the slow-responding level-control detector to "catch up" is adapted by Dolby to an adaptive delta-mod digital transmission scheme, where the step-size information is coded within a bandwidth on the order of 100 Hz. Both these applications take advantage of the economies made possible by the central transmission nature of their media, which allows the expense of the delay-line and A/D's at the encoder. CD manufacture offers the same distribution configuration and thus the same economic practicality. Thus, to recapitulate, I don't think you need 30% overhead for compander information, I don't think you need 10%, and in fact I still don't see why you need an explicit control-track at all, since in an instantaneously compandered digital system tracking errors are not a problem as with analog noise-reduction systems. And I'm unconvinced that 16-bit linear PCM isn't already a miraculously generous and ambitious format to offer the mass market- after all, this is essentially (except for sample rate) the scheme that virtually everyone was/is thrilled to use commercially. Any, I remain Doug Maisel
herbie@watdcsu.UUCP (Herb Chong, Computing Services) (10/23/84)
The key point you mentioned was properly implemented. The problem with most recordings on CD (and on vinyl, for that matter) is the initial recording and the microphone setup. Admittedly, a digital recording system is pretty impressive as far as the numbers go. However, those numbers assume a lot of things, like the recording engineer is familiar with the technical side of his recorder and the consequences of what he does. The digital systems are much less forgiving of errors and techniques of analog systems don't cut it, most of the time, if you want to realize the full potential of the medium. A lot of manufacturers of CD's have admitted to compressing the initial signal so that the dynamic range is "acceptable for the consumer". Where have we heard that one before? With the compression, where do you think all the original noise goes? I would prefer to make my own decison about compression and keep the noise where it belongs. Some companies ddon't compress, but they are the ones that originally in the audiophile record business with digital recordings of their own. One final point, at $1-2k per hour just for the recording time, wouldn't you rather have a saftey blanket if you set the recording levels wrong? And wouldn't you rather have a few extra bits of dynamic range when you know the signal is going to be compressed to the final dynamic range before mastering? Herb... I'm user-friendly -- I don't byte, I nybble.... UUCP: {decvax|utzoo|ihnp4|allegra|clyde}!watmath!watdcsu!herbie CSNET: herbie%watdcsu@waterloo.csnet ARPA: herbie%watdcsu%waterloo.csnet@csnet-relay.arpa BITNET: herbie at watdcs,herbie at watdcsu
wmartin@brl-tgr.ARPA (Will Martin ) (10/23/84)
It's interesting... All these discussions of interleaving level-control info with the sound-determining data on a CD sound an awful lot like what is done in a reproducing piano. This is a player piano where the holes in the central portion of the roll determine the notes played (as in an ordinary player), plus there are holes on both roll edges that control volume and other factors determining expression. There is nothing new... Will Martin USENET: seismo!brl-bmd!wmartin or ARPA/MILNET: wmartin@almsa-1.ARPA