poser@csli.Stanford.EDU (Bill Poser) (12/15/90)
Can anyone tell me what sampling rates are supported by the digital-to-analog converters on the NeXT machine? Bill
eps@toaster.SFSU.EDU (Eric P. Scott) (12/15/90)
In article <16894@csli.Stanford.EDU> poser@csli.Stanford.EDU (Bill Poser) writes: >Can anyone tell me what sampling rates are supported by the digital-to-analog >converters on the NeXT machine? 8012.8210513 Hz (CODEC, used by Lip Service and voice alerts) 22050 Hz (used primarily for beep sounds and Macintoy compatibility) 44100 Hz (CD quality) -=EPS=-
garton@cunixa.cc.columbia.edu (Bradford Garton) (12/15/90)
In article <16894@csli.Stanford.EDU> poser@csli.Stanford.EDU (Bill Poser) writes: >Can anyone tell me what sampling rates are supported by the digital-to-analog >converters on the NeXT machine? > I had heard that 2.0 allows arbitrary sampling rates -- anyone know if this is true? Brad Garton Music Dept. brad@woof.columbia.edu
jsd@arcadien.rice.edu (Shawn Joel Dube) (12/16/90)
In article <16894@csli.Stanford.EDU>, poser@csli.Stanford.EDU (Bill Poser) writes: |> Can anyone tell me what sampling rates are supported by the digital-to-analog |> converters on the NeXT machine? |> If I read the man pages right, you have a choice of 11kHz or 22kHz with either 8bit logrithmic or 16 bit quality. The 22kHz w/ 16 bits is the same as CD sound. I'm sure it will support about any other speed except the packaged software won't support it. Supposedly (and I heard this from a NeXT rep), the DAC is fast enough to emulate 9600 buad modems and even a fax machine. -- rrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrr r ___ _ "...but then there was the r r /__ | \ possibility that they were r r ___/hawn |__\ube LaRouche democrats which, of r r jsd@owlnet.rice.edu course, were better off dead." r rrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrrr
edwardm@hpcuhd.HP.COM (Edward McClanahan) (12/18/90)
poser@csli.Stanford.EDU asks: > Can anyone tell me what sampling rates are supported by the > digital-to-analog converters on the NeXT machine? jsd@arcadien.rice.edu answers: > If I read the man pages right, you have a choice of 11kHz or 22kHz > with either 8bit logrithmic or 16 bit quality. I believe the two rates supported are 22KHz and 44KHz stereo 16-bit/channel. The 8-bit logrithmic rate is the sampling rate of the A->D converter attached to the microphone input. Several manufacturers (such as Ariel) offer 16-bit/ch stereo A->D converters which connect to the DSP Port. > The 22kHz w/ 16 bits is the same as CD sound. Uh... The 44KHz rate is that used in Audio CDs. This rate was chosen so that frequencies up to 20KHz (actually 22KHz) could be represented digitally and be reconstructed accurately in an D->A converter later. Nyquist is credited with the theory that twice the sampling rate is required to represent an analog signal of a particular frequency (20KHz is higher than most people can here). As the 44KHz standard was being proposed, some manufacturers wanted a 50KHz signal (giving more breathing room at the high frequency end), but the technology of the day suggested using the slower rate. If I'm not mistaken, a 48KHz rate was also proposed because it matched a "standard" used already in the recording process. Skeptics doubt the accuracy of converting 48KHz to 44KHz sampled data. Also, I believe current digital recording equipment samples at twice and/or four times the 44KHz rate. The conversion from 88KHz or 176KHz to 44KHz is obviously mathematically simpler than the 48KHz to 44KHz conversion and has quieted most of the skeptics (such as Chip Davis of Mannheim Steamroller fame). > I'm sure it will support about any other speed except the packaged > software won't support it. Well, you could probably pump data to the DAC connected to the DSP chip at an arbitrary rate, but recall that an extremely steep low-pass filter is required before and/or after the D->A step. This filter is difficult to design because it must try to avoid phase (and other) distortion near the Nyquist frequency (half the sampling frequency). The 22KHz (Nyquist) frequency may be well beyond your hearing, but filters arn't perfect. The low-pass filters in CD players (and the NeXT) tend to invert or shift the phase of frequencies near their "cutoff" point. Also, those frequencies may be boosted or attenuated as well. Suffice it to say that in order to minimize these effects, the low-pass filters are designed for set "cutoff" points. The one in the NeXT apparently supports two points corresponding to the 22KHz and 44KHz sampling rates. Incidently, this filter is far more important during the sampling phase (A->D). That is why Ariel's (and others') stuff costs so much. The D->A function is much cheaper. I'd even go so far as to say the reason many of my CDs sound worse than their corresponding albums is due to the A->D conversion process used at the recording studios. On several Leo Kottke CDs, an audible ring at around 10KHz cannot be avoided by playing them on any CD player I have tried (some costing over $3000). Alas, even albums arn't immune to this problem if the studio used digital mastering (more and more common these days). > Supposedly (and I heard this from a > NeXT rep), the DAC is fast enough to emulate 9600 buad modems and > even a fax machine. The hottest question in comp.sys.next last year... =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-= Edward McClanahan Hewlett Packard Company -or- edwardm@cup.hp.com Mail Stop 42UN 11000 Wolfe Road Phone: (480)447-5651 Cupertino, CA 95014 Fax: (408)447-5039
cpenrose@sdcc13.ucsd.edu (Christopher Penrose) (12/19/90)
In article <108170005@hpcuhd.HP.COM> edwardm@hpcuhd.HP.COM (Edward McClanahan) writes: >Well, you could probably pump data to the DAC connected to the DSP chip >at an arbitrary rate, but recall that an extremely steep low-pass filter >is required before and/or after the D->A step. This filter is difficult >to design because it must try to avoid phase (and other) distortion near >the Nyquist frequency (half the sampling frequency). The 22KHz (Nyquist) >frequency may be well beyond your hearing, but filters arn't perfect. The >low-pass filters in CD players (and the NeXT) tend to invert or shift the >phase of frequencies near their "cutoff" point. Also, those frequencies >may be boosted or attenuated as well. Suffice it to say that in order to >minimize these effects, the low-pass filters are designed for set "cutoff" >points. The one in the NeXT apparently supports two points corresponding >to the 22KHz and 44KHz sampling rates. It would still be extremely useful to have the ability to convert signals at arbitrary sampling rates. First, we don't know what the exact cutoff points are on these filters. Anyone have the frequency response of the NeXT DAC filters? I have used MTU digisound-16 dacs that had a ceiling sampling rate of 48KHz. The filters, however, rolled off (-60db) at about 18.9KHz. As disk and cpu resources are limited, I chose lower sampling rates for my computer music pieces. My piece "Lesion", was converted at 30KHz using the 18.9KHz filter. There was a small amount of high frequency aliasing around 12KHz-15KHz, but I liked it in context. No one else has noticed this distortion. A later piece: "CircusCircus" was converted at 36KHz. I perceived no aliasing distortion and the frequency response spanned the bounds of the filter. >On several Leo Kottke CDs, an audible ring at around 10KHz cannot be >avoided by playing them on any CD player I have tried (some costing over >$3000). Alas, even albums arn't immune to this problem if the studio >used digital mastering (more and more common these days). I don't want to get into the classic analog/digital debate, but each audio medium clearly influences its output sound. You may notice digital phase cancellation, reinforcement, and aliasing, but I can't help hearing poor channel seperation, wow & flutter, high frequency distortion, and noise associated with turntables. The digital medium provides me, as a composer, with a lot of control, and I'd like to be able exploit the "limitations" of the medium if I choose. Arbitrary sample rate conversion would be cool. If 2.0 doesn't support it, maybe I'll hack it myself. Again, anyone have the frequency response handy for the dac filter(s)? As it stands the converters are half a sample out of phase for stereo conversion. But this seems to be nitpicking. The conversion interface is the best that I have ever used, and its sitting in my living room. Christopher Penrose jesus!penrose
cbenda@unccvax.uncc.edu (carl m benda) (12/19/90)
In article <1990Dec15.144028.17387@cunixf.cc.columbia.edu>, garton@cunixa.cc.columbia.edu (Bradford Garton) writes: > In article <16894@csli.Stanford.EDU> poser@csli.Stanford.EDU (Bill Poser) writes: > >Can anyone tell me what sampling rates are supported by the digital-to-analog > >converters on the NeXT machine? > > > > I had heard that 2.0 allows arbitrary sampling rates -- anyone know if > this is true? > > Brad Garton > Music Dept. > brad@woof.columbia.edu ACK.... correct me if I am wrong, but I always thought that "SAMPLING RATE" applied to ANALOG --> DIGITAL conversion and NOT the other way around. I believe that the NeXT machine reproduces "CD" quality sound, thus 44.1Khz is the frequency at which the machine takes a 16 bit quantity and USES the Moto DIGITAL SIGNAL PROCESSOR to convert said 16 bit quantity into an ANALOG voltage level to be sent to your audio equipemnet. /Carl
poser@csli.Stanford.EDU (Bill Poser) (12/19/90)
In article <3020@unccvax.uncc.edu> cbenda@unccvax.uncc.edu (carl m benda) writes: > >ACK.... correct me if I am wrong, but I always thought that "SAMPLING RATE" >applied to ANALOG --> DIGITAL conversion and NOT the other way around. Nope. The sampling rate refers to the relationship between the discrete-time signal and the continuous time signal that it is considered to represent, not necessarily to the actual sampling process. So it is quite legitimate to refer to the sampling rate of a DAC, and this usage is quite common.
gessel@masada.cs.swarthmore.edu (Daniel Mark Gessel) (12/19/90)
>is the frequency at which the machine takes a 16 bit quantity and USES the >Moto DIGITAL SIGNAL PROCESSOR to convert said 16 bit quantity into an ANALOG >voltage level to be sent to your audio equipemnet. /Carl The A to D converter is located in the monitor (on the old cubes), I'm pretty sure. I don't think it's the DSP that does it (I don't even know if it can), since I remember a mention of a 16bit ADC in the literature about the machine (some advertising). The DSP is used to generate synthesized sounds, (and generate images of the mandelbrot set, of course). Dan -- Daniel Mark Gessel Independent Software Consultant Internet: gessel@cs.swarthmore.edu and Developer I do not represent Swarthmore College (thank God).