Re: [linux-audio-dev] Using Jack for Control Volt continued
On Sunday 09 November 2003 03:11, Paul Winkler wrote: Most audio circuits include coupling capacitors that block DC voltages. Transformers work too. I would be amazed if any soundcard will pass a DC voltage. I know of at least one manufacturer (Digital Audio Labs, http://www.digitalaudiolabs.com/) that makes cards capable of this, although even there you need to set an internal jumper to enable it (default=DC blocked). I'm told that this is because their products are frequently used in general-purpose instrumentation setups, as well as strictly audio applications. Cheers! |-| | Frederick F. Gleason, Jr. | Director of Broadcast Software Development | | | Salem Radio Labs| |-| | Some people say a front-engine car handles best. Some people say a| | rear-engine car handles best. I say a rented car handles best.| | --P.J. O'Rourke| |-|
Re: [linux-audio-dev] Using Jack for Control Volt continued
Thanks for the all the interesting responses. I did some thinking and a few experiments, and I think the best way to do it would be to use an amplitude modulated sine ( actually triangle ) wave at one have the nyquist, precisely 1/4 of sample rate. Here's why I think this: At half nyquist, we have exactly four samples always per full wave. This means any sample is either a peak ( our desired data ), or a zero. It also means sine wave and triangles look exactly identical, and geometric production of the wave as a triangle will work perfectly, while production of the wave as a sine will get forced into a triangle shape. At the nyquist, the wave is either all peaks or all zeros, so that's no good because most digitial synthesis wave generators will start at 0 phase, meaning we get all zeros and no data! We could force everyone to use nyquist with phase 90%, but I think that makes things needlessly difficult to generate and represent. At frequencies other than half nyquist, we get long range distortions if a sine wave is used, ( try it out, I never realized just *how* lame 48khz is until trying this! ) but at half nyquist sines and tris are the exact same so it should be easy for any app to generate these perfectly. The same holds true for quarter nyquist, eight, etc, which could be optional for lower res signals, but I can't think of any advantage to using lower than 1/4 sr. Pipe up if you can please. I think the recomendation should be: Use a triangle if you are not absolutely positive the frequency calculation is perfect. If you know it is precisely 1/4 sr, use a sine or a tri. At half nyquist, it should also thus take only 2 samples to get the data, and it seems to me ( though I'm no DSP programming expert by *any* stretch ) that this should make demodulating very simple. The needed control data can be reached by a two sample sample-and-hold, or a conditional evaluation that just checks whether the data is not 0, or an averaging equation on any power of 2 samples range ( ie the average amp of four samples multiplied by two gives us the control data. ) Our peak control rate is thus 1/4 sample rate, which is high enough for nice sounding amplitude envelopes, and if less is needed the same frequency wave can be either modulated or demodulated at a lower resolution. As long as the window is a multiple of 2, we should get good accuracy. Further, this would I think allow easy hardware integration as standard sound cards can be used, again demodulation should be straightforward for outboard gear ( easy microcontroller programming again with the 2 sample sample and hold for example ) and we don't have to worry about telling people to be careful with potentially dangerous DC signal. And . . . we already have a plethora of gear and software that at the very least allows automation of volume. This way that facility can be bastardized to become whatever we want. This also looks nice in standard wave views, a nice graphic view of the data is clear as a block of solid colour in most viewers. I am interested to hear any and all feedback of course. Thanks, Iain Duncan
Re: [linux-audio-dev] Using Jack for Control Volt continued
Iain Duncan [EMAIL PROTECTED] writes: Thanks for the all the interesting responses. I did some thinking and a few experiments, and I think the best way to do it would be to use an amplitude modulated sine ( actually triangle ) wave at one have the nyquist, precisely 1/4 of sample rate. Why modulate at all? You can represent the voltages directly or proportionally. Note that making everything a function of the Nyquist frequency causes a shift when running at different sample rates. -- joq
Re: [linux-audio-dev] Using Jack for Control Volt continued
On Sat, Nov 08, 2003 at 07:50:22PM -0800, [EMAIL PROTECTED] wrote: Sorry, forgot to ask a couple of things. Will Jack or other similar APIs allow passing of DC signal? jack doesn't care what kind of signals pass through it. If so, do we have to worry about speakers blowing up or do soundcards protect against DC? Most audio circuits include coupling capacitors that block DC voltages. Transformers work too. I would be amazed if any soundcard will pass a DC voltage. If not, can anyone tell me what the most CPU efficient way of modulating/demodulating a control signal might be? It occured to me that one could just use a very high frequency signal remember that nyquist theorem limits the frequency to half the sampling rate. -- Paul Winkler http://www.slinkp.com Look! Up in the sky! It's ANCIENT COWARD! (random hero from isometric.spaceninja.com)
Re: [linux-audio-dev] Using Jack for Control Volt continued
On Sunday 09 November 2003 09.11, Paul Winkler wrote: [...] remember that nyquist theorem limits the frequency to half the sampling rate. OTOH, if there's only one control per channel, one could just modulate with a sine at Nyqvist, and demodulate using fabs(sample). For two signals at half the sample rate, use the sign to determine the signal subchannel when demodulating. However, as long as you stay digital, there should be no need for modulation of control signals. AFAIK, JACK doesn't process the audio data in any way in between clients. You most probably *do* need to modulate if you need to go analog, unless you use modified sound cards or industrial AD/DA boards for the I/O. You'll have to use lower frequency carriers for analog I/O, as anything at Nyqvist will probably (should!) be totally eliminated by the converters. //David Olofson - Programmer, Composer, Open Source Advocate .- Audiality ---. | Free/Open Source audio engine for games and multimedia. | | MIDI, modular synthesis, real time effects, scripting,... | `--- http://audiality.org -' --- http://olofson.net --- http://www.reologica.se ---
