You just did use it to play "audio" :<) The 6 us settling time corresponds to a sampling rate of ~167 kHz, not that you will ever get there or would wish to. The theoretical (real) sampling rate required for a given bandwith is Fs = 2 Bw. That requires brick wall filters and it is a lot of work to get close without significant distortion. These old DACs are all but certain to use ladder circuits [see e.g. https://en.wikipedia.org/wiki/Resistor_ladder] the settlng time will mostly come from the output buffer [see e.g. https://www.analog.com/en/technical-articles/ltc1668-dac-lt1807-opamp-achie… for bleeding edge examples]. To see something other than ringing on a scope you want at least 10 samples per cycle, e.g. for 3 kHz bandwidth (i.e. 0 - 3 kHz frequency coverage) output at 30 kHz or greater. A low pass (reconstruction in the argot) filter will round off the corners - set the corner just above the passband DMA, a local FIFO or at least double buffering are the minimum to avoid sample jitter. On basic hardware you will probably have to do what you can with a sampling clock derived from the RTC card, from 10 MHz you could get an interupt at 40 kHz or 25 kHz but maybe not 30 kHz. The interupt then controls the play out from a table or disk ;<) For testing you can do quite a lot with a single cycle sine wave table in memory. Say you are playing out at Fs = 30 kHz, and you have a 30 k sample table. By varying the step through the table from 15k to 1 you can alter the output frequency from 15 kHz to 1 Hz in 1 Hz increments; i.e. output frequency = Fs * stride / table length. From a VQ look at the AAV11 docs it uses the bottom 12 bits, doubtless <11> is ms, hopefully it will like 2's complement numbers and the analog offset voltage will be trimmed for bipolar signals. Have fun and good luck Martin -----Original Message----- From: Douglas Taylor via cctalk [mailto:cctalk@classiccmp.org] Sent: 09 July 2023 19:46 To: General Discussion: On-Topic and Off-Topic Posts &lt;cctalk(a)classiccmp.org&gt; Cc: Douglas Taylor &lt;dj.taylor4(a)comcast.net&gt; Subject: [cctalk] Talking PDP11 I have a PDP-11/53 and have just started playing with an AAV11-C D/A board.  It is a 4 channel D/A convertor with 12 bit resolution. Can it be used to play an audio bit stream? Here is simple code used to see if the thing was actually working:     .title AAV11 D/A test     ;     .asect     dbr0 = 170440     .=1000 start:     mov #7777,r0     4096 value to R0     mov #dbr0,r1     first D/A buffer out loop:    mov r0,(r1)    transfer value in r0 to D/A out     dec r0        subtract 1 from D/A value     bne loop     br start        loop back to start I was surprised to see that it took ~34 ms to run through all the numbers from 0-7777, that is about 34 Hz.  The manual says the 'settling time' is 6 microseconds.  Is this fast enough for audio? How would you convert a modern audio file into 12 bit integers? Doug

Doug You have the essence of what it can do and how it might do it, an old iron PDP-11 / LSI-11 can't organically do the signal processing necessary to go from a compact signal representation to audio. That said you could use a one bit output, which would for efficiency require a serialiser, to drive a sigma delta converter, e.g. https://digilent.com/shop/pmod-i2s2-stereo-audio-input-and-output/, to convert a word stream to sound. Generating the I2S bit stream is of course an XFU, probably doable from a DRV11 type interface, with a fair bit of logic or your favorite microcontroller as intermediary. A 7 kHz sample rate will not be easily achieved, unless you can use an external clock or plug in an "optimal" oscillator, e.g. the classic example (but not for this requirement) is one running at a multiple of baud rate frequencies, 12M288 Hz say. On these old boards, frequency = oscillator / prescaler / divisor. Generally, the divisor is the "round" number and the frequency is the fractional mess. Also, the prescalers may be powers of two, ten, or 1/2/5 : YMMV. The sort of thing you could demonstrate as audio output from your hardware is either Morse or audio composed of phonemes. The Morse would require some evolution of the test generator I described and a little tapering of the attack / decay of the di dah dit sounds to avoid harshness, or could be done from canned waveforms (returning to 0V). The phonemes might generate text to speech of 1980's (i.e. contemporary period) standard. Have fun Martin -----Original Message----- From: Douglas Taylor via cctalk [mailto:cctalk@classiccmp.org] Sent: 10 July 2023 02:20 To: Martin Bishop via cctalk &lt;cctalk(a)classiccmp.org&gt; Cc: Douglas Taylor &lt;dj.taylor4(a)comcast.net&gt; Subject: [cctalk] Re: Talking PDP11 Wow! Actual engineers responding... It looks like I could only do the most rudimentary audio. 1. Sample Rate: You got maybe 20K samples to store in lower memory.  At 7KHz sample rate that would allow 3 seconds of audio.  Voice only. 2. Samples: They must be 12 bits. Converting a modern audio clip requires, band filtering, resampling and mapping to 12 bit integers. Could be done in python, they have libraries. 3. Clocking output:  I have a KMV11, but never programmed  around it. 4. Amplify output: AAV11-C produces -10 to +10 volts, have to divide this down for input to an audio amp. In the end I will have undone all the advances made in digital audio in the last 30 to 40 years. Doug On 7/9/2023 4:09 PM, Martin Bishop via cctalk wrote: > You just did use it to play "audio" :<) > > The 6 us settling time corresponds to a sampling rate of ~167 kHz, not that you will ever get there or would wish to. > > The theoretical (real) sampling rate required for a given bandwith is > Fs = 2 Bw. That requires brick wall filters and it is a lot of work > to get close without significant distortion. These old DACs are all > but certain to use ladder circuits [see e.g. > https://en.wikipedia.org/wiki/Resistor_ladder] the settlng time will > mostly come from the output buffer [see e.g. > https://www.analog.com/en/technical-articles/ltc1668-dac-lt1807-opamp- > achieve-90ns-settling-to-16bits-83db-sfdr-small-footprint.html for > bleeding edge examples]. To see something other than ringing on a > scope you want at least 10 samples per cycle, e.g. for 3 kHz bandwidth > (i.e. 0 - 3 kHz frequency coverage) output at 30 kHz or greater. A > low pass (reconstruction in the argot) filter will round off the > corners - set the corner just above the passband > > DMA, a local FIFO or at least double buffering are the minimum to > avoid sample jitter. On basic hardware you will probably have to do > what you can with a sampling clock derived from the RTC card, from 10 > MHz you could get an interupt at 40 kHz or 25 kHz but maybe not 30 > kHz. The interupt then controls the play out from a table or disk ;<) > > For testing you can do quite a lot with a single cycle sine wave table in memory. Say you are playing out at Fs = 30 kHz, and you have a 30 k sample table. By varying the step through the table from 15k to 1 you can alter the output frequency from 15 kHz to 1 Hz in 1 Hz increments; i.e. output frequency = Fs * stride / table length. > > From a VQ look at the AAV11 docs it uses the bottom 12 bits, doubtless <11> is ms, hopefully it will like 2's complement numbers and the analog offset voltage will be trimmed for bipolar signals. > > Have fun and good luck > > Martin > > -----Original Message----- > From: Douglas Taylor via cctalk [mailto:cctalk@classiccmp.org] > Sent: 09 July 2023 19:46 > To: General Discussion: On-Topic and Off-Topic Posts > &lt;cctalk(a)classiccmp.org&gt; > Cc: Douglas Taylor &lt;dj.taylor4(a)comcast.net&gt; > Subject: [cctalk] Talking PDP11 > > I have a PDP-11/53 and have just started playing with an AAV11-C D/A board.  It is a 4 channel D/A convertor with 12 bit resolution. > > Can it be used to play an audio bit stream? > > Here is simple code used to see if the thing was actually working: > >     .title AAV11 D/A test >     ; >     .asect > >     dbr0 = 170440 > >     .=1000 > start: >     mov #7777,r0     4096 value to R0 >     mov #dbr0,r1     first D/A buffer out > > loop:    mov r0,(r1)    transfer value in r0 to D/A out >     dec r0        subtract 1 from D/A value >     bne loop > >     br start        loop back to start > > I was surprised to see that it took ~34 ms to run through all the numbers from 0-7777, that is about 34 Hz.  The manual says the 'settling time' is 6 microseconds.  Is this fast enough for audio? > > How would you convert a modern audio file into 12 bit integers? > > Doug >

On Jul 11, 2023, at 7:43 AM, steven(a)malikoff.com steven--- via cctalk &lt;cctalk(a)classiccmp.org&gt; wrote: Much the same here. I recounted this on VCFed a few months ago about building a simple 2-chip 8-bit ladder DAC with one-transistor amplifier for my Applied Technology DG680 S100 machine back in the early 80s from this absolutely excellent BYTE article on how to do polyphonic synthesis on a microcomputer (KIM-1): https://archive.org/details/byte-magazine-1977-09/page/n63/mode/2up A schoolfriend who had an Apple ][ and had not done any Z80 machine code before asked for me to hand him my Zaks book, upon which he wrote out one attempt in Z80, crossed it out and wrote a second version. Which worked perfectly. For the music piece I got it to play four-voice polyphony after painstakingly encoding Bach's Praeludium in C Major from my mothers' collection of piano music scores. A few years ago I had thoughts about porting the 6502 code to the PDP-11 and use the same sort of ladder DAC. Not sure if the slimline 11/05 would be fast enough for anything too high frequency, but if it was, the slimline 05's power supply could then temporarily come out and be perhaps be powered off some beefy batteries in that space, along with a small 1970s transistor amp and 1970s headphones topped off with a leather shoulder strap to lug it around like a giant Walkman.

And by 1979 there was the fairlight... https://en.wikipedia.org/wiki/Fairlight_CMI 73 Eugene W2HX Subscribe to my Youtube Channel: https://www.youtube.com/@w2hx/videos >Speaking of old computerized music playing technology, there are two from the PLATO system at the University of Illinois that are perhaps the earliest of all in their category and not all that well known.

And by 1979 there was the fairlight... https://en.wikipedia.org/wiki/Fairlight_CMI 73 Eugene W2HX Subscribe to my Youtube Channel: https://www.youtube.com/@w2hx/videos The first of the two is the GSW (Gooch Synthetic Woodwind), which is a four voice, 7 levels per voice, square wave synthesizer. It's fully documented in Sherwin's US patent 4,206,675. That one was attached to the auxiliary device port of a PLATO terminal and driven from the host computer, at 1200 bps. It worked quite well for playing music and was widely used for music education. It's a very simple device as you can see from the full schematics (which are, surprisingly, given in the patent). That patent was filed in 1977 but the invention is somewhat older, perhaps 1975 or 1976. The followon to that is the GCS (Gooch Cybernetic Synthesizer), unfortunately not well documented. That's a 16 voice programmable waveform (256 words by 16 bits per voice), more levels (256?), driven as a peripheral off the 8080-based "programmable PLATO terminal" from a program running in that terminal. So the musical score level definition of what to play still came from the host, still at 1200 bps, but the attack/decay etc. shaping would happen in the terminal. That one was a bit of an electrical muddle, with memory, logic, and D/A per voice followed by a 16 input combiner. Getting the analog parts to work right was a hairy task with far too many trimpots. Sherwin vowed that any followup would be digital all the way to one final D/A, which of course later became the answer in the PC sound cards, but if he did that it was after I left. The GCS was built around 1977. There were some interesting related pieces of work, such as a speed-sensing piano keyboard (so unlike an electronic organ you could have dynamics, exactly as on a piano), a music editing system with a score printing program to print on a dot matric (electrostatic) printer, and some other stuff. I'm not sure if the GCS is the earliest fully programmable waveform digital music synthesizer, but if not it is close. paul

On 07/11/2023 9:59 AM CDT Paul Koning via cctalk &lt;cctalk(a)classiccmp.org&gt; wrote:

But, judging by Wikipedia, the earlier Synclavier models were not digital sampled waveform synthesizers but rather FM synthesizers. So I still wonder if anyone did it earlier than Sherwin. paul

The Synclavier I was commercially available in 1977, based off the Dartmouth Digital Synthesizer of earlier times. The core was a New England Digital minicomputer architecture (they did sell just the minicomputer to the military, as a side). The truth is that there were quite a few digital synths in labs in 1977. -- Will On Tue, Jul 11, 2023 at 10:22 AM W2HX via cctalk &lt;cctalk(a)classiccmp.org&gt; wrote: from the PLATO system at the University of Illinois that are perhaps the earliest of all in their category and not all that well known. four voice, 7 levels per voice, square wave synthesizer. It's fully documented in Sherwin's US patent 4,206,675. That one was attached to the auxiliary device port of a PLATO terminal and driven from the host computer, at 1200 bps. It worked quite well for playing music and was widely used for music education. It's a very simple device as you can see from the full schematics (which are, surprisingly, given in the patent). That patent was filed in 1977 but the invention is somewhat older, perhaps 1975 or 1976. unfortunately not well documented. That's a 16 voice programmable waveform (256 words by 16 bits per voice), more levels (256?), driven as a peripheral off the 8080-based "programmable PLATO terminal" from a program running in that terminal. So the musical score level definition of what to play still came from the host, still at 1200 bps, but the attack/decay etc. shaping would happen in the terminal. That one was a bit of an electrical muddle, with memory, logic, and D/A per voice followed by a 16 input combiner. Getting the analog parts to work right was a hairy task with far too many trimpots. Sherwin vowed that any followup would be digital all the way to one final D/A, which of course later became the answer in the PC sound cards, but if he did that it was after I left. The GCS was built around 1977. There were some interesting related pieces of work, such as a speed-sensing piano keyboard (so unlike an electronic organ you could have dynamics, exactly as on a piano), a music editing system with a score printing program to print on a dot matric (electrostatic) printer, and some other stuff. digital music synthesizer, but if not it is close.

Much the same here. I recounted this on VCFed a few months ago about building a simple 2-chip 8-bit ladder DAC with one-transistor amplifier for my Applied Technology DG680 S100 machine back in the early 80s from this absolutely excellent BYTE article on how to do polyphonic synthesis on a microcomputer (KIM-1): https://archive.org/details/byte-magazine-1977-09/page/n63/mode/2up A schoolfriend who had an Apple ][ and had not done any Z80 machine code before asked for me to hand him my Zaks book, upon which he wrote out one attempt in Z80, crossed it out and wrote a second version. Which worked perfectly. For the music piece I got it to play four-voice polyphony after painstakingly encoding Bach's Praeludium in C Major from my mothers' collection of piano music scores. A few years ago I had thoughts about porting the 6502 code to the PDP-11 and use the same sort of ladder DAC. Not sure if the slimline 11/05 would be fast enough for anything too high frequency, but if it was, the slimline 05's power supply could then temporarily come out and be perhaps be powered off some beefy batteries in that space, along with a small 1970s transistor amp and 1970s headphones topped off with a leather shoulder strap to lug it around like a giant Walkman.

The DACs on the AAV11-C board are not marked in any revealing way.  I think they are Burr Brown DAC80, 24 pin, but I'm not sure.  I wasn't sure if they were working and was looking for a replacement. Looking at the spec sheets DAC's seem to come in Voltage or Current versions.  Life got more complicated. This started out as a simple exercise into verifying the AAV11-C operation using PDP11GUI to program up a basic program to run all the codes thru the DAC.  It worked, got a ramp out.  Now, I'm starting to look at the KWV11-C and how to use that to send values to the DAC at a controllable rate. Doug ------------------ On 7/11/2023 11:41 AM, Mike Katz via cctalk wrote:

Funny you mention that, I've got a Data Translation DT2766 and it is identical to the AAV11-C.  I mean identical!  In the day DT must have sold them based on 2 selling points: (1) Cheaper than DEC and (2) Exact drop in replacement for the DEC AAV11-C. Doug On 7/11/2023 12:33 PM, Nigel Johnson via cctalk wrote:

For info on DAC internals, have a look at https://www.analog.com/en/education/education-library/analog-digital-conver… Part 2 details basic A/D & D/A architectures Current output converters are: - less common than voltage output DACs - and, if a ladder conversion architecture is used, lurking inside voltage output DACs behind a transimpedance amplifier (I to V converter) I would be very surprised if the AAV11 does not output buffered volts Lots of other good reference material on analog.com, TI.com is also worth a look Martin PS Most contemporary audio work uses SigmaDelta converters, see e.g. Analog Devices AN-283 and MT-022 -----Original Message----- From: Douglas Taylor via cctalk [mailto:cctalk@classiccmp.org] Sent: 11 July 2023 17:29 To: Mike Katz via cctalk &lt;cctalk(a)classiccmp.org&gt; Cc: Douglas Taylor &lt;dj.taylor4(a)comcast.net&gt; Subject: [cctalk] Re: Talking PDP11 The DACs on the AAV11-C board are not marked in any revealing way. I think they are Burr Brown DAC80, 24 pin, but I'm not sure. I wasn't sure if they were working and was looking for a replacement. Looking at the spec sheets DAC's seem to come in Voltage or Current versions. Life got more complicated. This started out as a simple exercise into verifying the AAV11-C operation using PDP11GUI to program up a basic program to run all the codes thru the DAC. It worked, got a ramp out. Now, I'm starting to look at the KWV11-C and how to use that to send values to the DAC at a controllable rate. Doug ------------------ On 7/11/2023 11:41 AM, Mike Katz via cctalk wrote:

On Jul 9, 2023, at 9:19 PM, Douglas Taylor via cctalk &lt;cctalk(a)classiccmp.org&gt; wrote: Wow! Actual engineers responding... It looks like I could only do the most rudimentary audio. 1. Sample Rate: You got maybe 20K samples to store in lower memory. At 7KHz sample rate that would allow 3 seconds of audio. Voice only. 2. Samples: They must be 12 bits. Converting a modern audio clip requires, band filtering, resampling and mapping to 12 bit integers. Could be done in python, they have libraries. 3. Clocking output: I have a KMV11, but never programmed around it. 4. Amplify output: AAV11-C produces -10 to +10 volts, have to divide this down for input to an audio amp. In the end I will have undone all the advances made in digital audio in the last 30 to 40 years.

I'm reminded of a project I did in college in 1974, when I made a primitive graphics display using an X/Y oscilloscope driven by an AA-11. Since the machine was a PDP-11/20 with 8 kW of memory, I decided to use the RC-11 disk as the refresh memory, doing DMA directly from disk to the D/A data CSR. So on the scenario here: the sample rate is clearly more than adequate. 12 bits is not CD grade audio but not bad; for ears used to the distorions of compressed audio files it's probably good enough. The PDP-11 certainly won't be able to decompress modern lossy compression files. It should be fine with raw or nearly-raw files, which means you can convert externally and feed the resulting files to the PDP-11. You could convert to 16 bit raw mono with standard tools and then drop the bottom 4 bits. Band filtering? Resampling? I don't know why you would want to do that, unless there isn't a reasonable way to drive the device at the source file's data rate. For example, if you have a KW-11/P that's clearly doable. (Come to think of it, that 11/20 had a KW-11/P and I created BASIC extensions for it that would allow sampling to be driven by that clock, at a rate of your choosing.) You can't fit a whole lot of data in 64 kW of memory, but that isn't needed. That rate isn't all that high; it isn't hard to write a program that does double buffering from a disk file to memory to the D/A. That makes a really nice real time programming exercise. paul

I'm looking to see if it possible to do something similar.  Just blasting raw 12bit samples from memory out the D/A board. A fine point would be to use the KWV11-C realtime clock board for the sample rate control.  No interrupts, just polling. Harder is taking an existing MPG or WAV audio clip and converting it into raw 12bit integers. I wouldn't try this on the PDP11, I would do this in Linux, offline, using python or octave to take a wide band audio file convert it into a narrowband audio file, then resample down to the rate I would use on the PDP11 and finally convert the numbers into 12bit integers.