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:[email protected]]
Sent: 09 July 2023 19:46
To: General Discussion: On-Topic and Off-Topic Posts <[email protected]>
Cc: Douglas Taylor <[email protected]>
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
