newton2@ucbtopaz.CC.Berkeley.ARPA (10/12/84)
Newsgroups: net.audio Subject: Re: Sheffield CDs and why they sound bad References: <460@watdcsu.UUCP>, <46@vice.UUCP> I agree with just about everything in Shaun's posting on dynamic range, but I'm a little uncertain about what one gains from using an explicit control word (the equivalent of a pilot-tone compander). The distortions of dbx-style companders are due to two sources: 1. The inherent distortion of the (analog) gain-control elements (the 20-cent VCA's) 2. The envelope-following distortion caused by the control voltage modulating the channel gain; this distortion is cancelled if the decoder tracks the encoder *exactly*. Aside from shoddy component tolerances, mistracking is inevitable in an analog system because the level-measuring circuits don't measure the same signal at the same time- it's been analoged somewhat out of shape via a nonlinear, non-flat and time-dispersive recording medium (ignore the matching of the supposedly "rms" level detectors). In a digital system, a feedback-compressor/feedforward-expander topology would be inherently *perfect* in the level-measuring/matching department, since both ends of the system would "measure" (Really just transcribe) the same digitized level data. So distortions due to non-complementarity would arise only from the step-size errors of the A/D/A converters. Again, I don't see what putting the control information on a separate track, as it were, gains you, since everything on that track is implicit in the digital data together with knowledge of the companding algorithm. You've got to compute the data for the control track from the data, so why not do it in the reproducer? If you want non-standard or variable algorithms, you could load them at the beginning of the selection (or periodically) but I don't think you need anything like 30% control-track overhead. The multiband observation is certainly a valid one, however, if you don't want to reintroduce all the breath, swishing and sucking artifacts of the pre-Dolby (and post-dbx :-)) era.
newton2@ucbtopaz.CC.Berkeley.ARPA (10/12/84)
I can't type and read screen prompts at the same time (maybe I should run for Pres.) My signature on the captioned article shouldn't have been "~r digicomp", but rather Doug Maisel @ Univ. of Calif., Berkeley CA USA
herbie@watdcsu.UUCP (Herb Chong, Computing Services) (10/15/84)
For those that are interested, dbx makes a delta-modulation PCM
encoder/decoder that is capable of more dynamic range than a 16-bit
linear encoding can. I believe that it is a 12-bit system, but I may
be wrong. It is not terribly popular because it is incompatible with
everything else on the market. I do not know how many bits they use
in the A/D and D/A converters, but they store fewer bits than used
by the converters. For those that don't know, a delta-modulation
system records the difference signal between successive samples and
saves them onto the storage medium. The dbx unit attaches to a VCR.
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 watdcs5121cdd@houxm.UUCP (C.DORY) (10/16/84)
It seems as though we need a little clarification on what the dbx 700 is all about. True, it employs a form of delta modulation to code waveforms, however it is not a "12-bit system" as described previously. (I think what the previous contributor was thinking of was Differential PCM.) The sampling rate of the unit is about 700K Hz. (By the way, this is about the same bit rate as current PCM devices, however their sample rate is 1/16 that.) Each sample (bit), the unit decides whether the waveform went "up" or "down" in absolute voltage and writes the corresponding "1" or "0". Now, it's not quite as simple as that in real life -- delta mod has a few inherent limitations. The dynamic range is limited as well as the ability to track transients adequately. Therefore, dbx uses spectral companding accross different frequency bands to boost the dynamic range. (No folks, this is not a dbx 150 in front of their A/D, this companding works quite well as it is tied in with the A/D through a feedback loop.) As well, to enable the unit to track transients, elegant prediction circuitry has been added to the delta-mod coders. Hence, dbx coins the acronym CPDM (Companding Predictive Delta Modulation) for their dbx 700 Digital Audio Processor. For further reading, I recommend an article appearing in the current "AES JOURNAL" written by Bob Adams, the dbx 700's designer. I know that some of you are just dying to hear the specs, so here they are (to the best of my recollection): dynamic range -- >110dB (mid band), frequency response -- 10 Hz to 20 KHz +-0.5dB, etc. As well, since this unit has a sampling rate at 700 KHz (or thereabouts) you don't need those nasty elliptical anti-aliasing filters ala PCM units (thereby greatly reducing delay distortion - pinched highs, etc.) Don't get me wrong, the dbx 700 also has anti-aliasing filters, however these are Bessel (linear phase, no overshoot and ring) and the knee (-3dB point) is at 38 KHz!! What does it sound like? Glad you asked -- better than the Sony PCM F1. I've made several recordings in parallel (i.e., from the same pair of Schoeps mics and custom built mic preamps) with both devices using my wife (a concert violinist) as a single-blind lintener. In all cases, she was able to distinguish the two units and preferred the sound of the dbx. (Yes, I was careful to match the outputs of both units -- this was done using a 1 KHz tone. Matching was good to <.1dB. However, as pointed out by L. Miller, this might not be good enough.) Subjectively, the dbx 700 reproduces massed string tones and massed voices much better. As well, percussive instruments (i.e., triangles, etc.) have a much cleaner sounding "ting" rather than sounding like "splang" as on the Sony PCM F1. This is due, I believe, to the lack of significant delay distortion in the anti-aliasing filters. Albeit we were monitoring with very good equipment: Stax Lambda Pro earspeakers -- with excellent loudspeakers (by Nestorovic) the differences were still audible however not as prevalent. The unit is built as a professional tool rather than a consumer device like th Sony PCM-F1, et. al. In use, it has many advantages over the Sony. Inputs and outputs are adjustable and are at line (+4dBm) level. Also, the inputs and outputs are balanced. The Sony F1 has three error states: error correction, error concealment, and mute (while it collects its senses). This is all well and good for non-critical use, however when mastering a record it leaves a lot to be desired. Error correction is just that and it seems to work fairly well. Error concealment can be clearly audible and manifests itself as strange perturbations of the waveform. Muting is totally unacceptable, for 1.5 seconds the unit does a "stable clear" while it gathers its marbles to go on. The dbx 700 seems to be much more tolerent to errors, the most you will get from an uncorrectable error is a barely-audible "tick". The ballistics of the meters on the dbx are much easier to follow than the meters on the F1. In addition, th dbx when overloaded behaves a little more gracefully than the F1. As for the dbx 700's popularity, there are a few hundred in use currently. The uses are widely varied from on-location recording (like myself) to use as a mixdown/mastering deck in studios as well as remote links for WBGH (WGBH?) FM in Boston. The major complaint seems to be that there is currently no way to transfer digitally from the dbx 700 to the Sony PCM 1610 for CDs. I don't see this as much of a problem -- most CDs have been through a half a dozen A/D / D/A trips anyway. At least the dbx 700 sounds better (and doesn't introduce the gross group-delay distortions as the PCM units do). Why doesn't some DSP jock rig up a way to perform this digital transfer -- they'd probably make a mint as well as a lot of friends. Craig Dory AT&T Bell Laboratories