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PICList Thread
'SINEWAVE OUT OF PWM'
1995\09\21@065452 by *DO_LATER

I am currently doing a project and would like to know if it is possible to generate a sinusoidal analogue waveform using the PWM on the PIC16C74. I would also like to know if it is possible to generate a quadrature pair of sinusoids.

'ITU programmers? In business?'
1996\05\02@182907 by Eric T. Brewer

Hi folks, I have been trying to call ITU all day today and their phone has been busy. There Fax line is busy as well. Their web page is till working... Don't know what's up? Anyone else know? Based upon a couple of old mails to the list, I am intereseted in buying one for a client. Are they still good? Has anyone had problems? thanks, eric

1996\05\03@090605 by Odriozola Belden

I've been using an ITU programmer and had no trouble. The only thing that bothers me is that I learned to use the ASPIC that came with it and nobody else seems to use it, or am I wrong ? Almost all the source code examples you find in the net comes in another asembler. Anyway its an easy change from one to the other. Rogerio At 03:20 PM 2/05/96 -0700, you wrote: {Quote hidden}>Hi folks, > >I have been trying to call ITU all day today and their phone has been >busy. There Fax line is busy as well. Their web page is till working... >Don't know what's up? Anyone else know? > >Based upon a couple of old mails to the list, I am intereseted in buying >one for a client. Are they still good? Has anyone had problems? > >thanks, >eric > >

1996\05\03@092848 by Norm Cramer

I also have the ITU programmer and it works great. I hve only used it to program the 16C84 at this point. I have the ASPIC assembler but use MPASM since it is available free from Microchip. Norm At 08:04 AM 5/3/96 -0500, you wrote: >I've been using an ITU programmer and had no trouble. The only thing that >bothers me is that I learned to use the ASPIC that came with it and nobody >else seems to use it, or am I wrong ? Almost all the source code examples >you find in the net comes in another asembler. Anyway its an easy change >from one to the other. > >Rogerio >

1996\05\03@102015 by chris

Eric, We are very much still in business! If you are still having problems contacting us today, please send e-mail directly to me. Thanks for your interest in our products! ---------------------------------------------------------------------- Chris B. Sakkas (spam_OUTchrisTakeThisOuTitutech.com) http://www.itutech.com ITU Technologies (.....infoKILLspam@spam@itutech.com) ftp://itutech.com *** Complete PIC programming packages starting at only $39! *** *** VISA and MasterCard accepted (513) 574-7523 ***

1996\05\03@145441 by David E. Queen

At 08:26 AM 5/3/96 -0500, you wrote: >I also have the ITU programmer and it works great. I hve only used it to >program the 16C84 at this point. > >I have the ASPIC assembler but use MPASM since it is available free from >Microchip. > >Norm > >At 08:04 AM 5/3/96 -0500, you wrote: >>I've been using an ITU programmer and had no trouble. The only thing that >>bothers me is that I learned to use the ASPIC that came with it and nobody >>else seems to use it, or am I wrong ? Almost all the source code examples >>you find in the net comes in another asembler. Anyway its an easy change >>from one to the other. >> >>Rogerio >> > So, is it generally accepted that a newbee like me would be better off learing with MPASM? Does MPSAM work well with the ITU unit and does anyone use the pair?

1996\05\03@155628 by Martin J. Maney

On Fri, 3 May 1996, David E. Queen wrote: > So, is it generally accepted that a newbee like me would be better off learing > with MPASM? Does MPSAM work well with the ITU unit and does anyone use the pair? I dunno, sounds like a religious issue. :-) Yes and yes.

1996\05\03@170712 by Norm Cramer

At 02:56 PM 5/3/96 -0500, you wrote: >On Fri, 3 May 1996, David E. Queen wrote: > >> So, is it generally accepted that a newbee like me would be better off learing >> with MPASM? Does MPSAM work well with the ITU unit and does anyone use the > pair? > >I dunno, sounds like a religious issue. :-) Yes and yes. > > Well it is sort of a religious issue but there are good reasons to use MPASM. Mainly since most people use it, code examples from the list can be used directly. Also assembler help is easyer to find. It runs as a Windows application not as a dos shell. I am sure there are more but these are enough for me. In fact the first one is enough for me. Norm

1996\05\03@204702 by Steve Budde

>> So, is it generally accepted that a newbee like me would be better off >learing >> with MPASM? Does MPSAM work well with the ITU unit and does anyone use the > pair? > >I dunno, sounds like a religious issue. :-) Yes and yes. > > I use both the MPASM compiler and the ITU PIC-1 programmer. I am extremely happy with both of them (but I wish their new programmer, that handles all of the PICs was out at the time I bought mine) One thing, the MPASM is great because it is free. And, as several others have mentioned, most of the code examples both on this list and in all of the app notes are aimed toward this compiler. This makes it much easier to start learning how to program these little goldmines. I initially bought my programmer for using on my senior project this semester. We used a 16C65, and had no trouble with using the programmer (even through a dos window within Windows 95). The kit was easy to build, and had the whold thing working in about an hour. So from my limited experience anyway, I would recommend using MPASM, and using an ITU programmer. I have used the PROMATE programmer, and was happy with it, but for a poor student like me, I could not afford one. I have also used a Tribal Microsystems programmer, which programmed just about everything under the sun......but about twice the price of PROMATE. anyway, my $.02 Steve -- Steve Budde e-mail: sbuddeKILLspamsiu.edu WWW: http://www.silug.org/~budde http://www.prairienet.org/~sbudde

1996\05\04@014133 by David E. Queen

At 04:04 PM 5/3/96 -0500, you wrote: >At 02:56 PM 5/3/96 -0500, you wrote: >>On Fri, 3 May 1996, David E. Queen wrote: >> >>> So, is it generally accepted that a newbee like me would be better off >learing >>> with MPASM? Does MPSAM work well with the ITU unit and does anyone use the >> pair? >> >>I dunno, sounds like a religious issue. :-) Yes and yes. >> >> > >Well it is sort of a religious issue but there are good reasons to use >MPASM. Mainly since most people use it, code examples from the list can be >used directly. Also assembler help is easyer to find. It runs as a Windows >application not as a dos shell. I am sure there are more but these are >enough for me. In fact the first one is enough for me. > >Norm > I guess there are no Windows 95 issues or you would have said something.

1996\05\06@090059 by Norm Cramer

At 12:46 AM 5/4/96 -0500, you wrote: >> >I guess there are no Windows 95 issues or you would have said something. > I have been running the programmer in a DOS window under Windows 95 without any problems.

'62.5 Hz sine wave'
1996\10\29@101626 by Shawn Ellis

Hey all, this will be the first real problem I have presented this list and I hope SOMEONE can come through for me. Buttering up aside, I have a 12-bit DAC (AD737) interfaced to a PIC16C74. I'm useing the PCW "C" compiler from CCS. I am trying to build a ultra-stable AC synchronious motor driver. So I have a sine wave represented as a 160-point data set that I generated with a basic program and imported into my code as an array. This data-set represents one half of one period of the sine wave. The PIC sends each value in the set out to the DAC and at the end of the set subtracts the numbers from half scale to produce the negative part of the period, then the whole thing loops. The PIC is looking at a 20Khz, 90% duty cycle ttl square wave that comes from the PWM module. This square wave also updates the DAC. The result should be a 62.5 Hz, +5VDC to -5VDC sine wave coming out of the DAC. The PIC loads the new DAC value only after seeing the DAC get updated by the 20 Khz pulse and then waits for the next DAC update before loading the next number. The DAC update is then fed into an APEX high voltage amp to produce a large sine wave. The problem is that the resulting sine wave is really 62.52 Hz (5 us too fast). I have used several different sources for the 20Khz update pulse with no change in performance so I don't think that could be the problem. The device this control is intended for needs stability better than +/- 1us. (Thats less than +/- 1us per revolution at 62.5 Hz) Right now we are running it with a 62.5 hz ttl square wave, being fed into an inverting amplifier, then a transformer to produce a +/- 60 V square wave. Anybody have any suggestions on: 1. What could this problem be? 2. Is there a better/easier way to achieve an ultra-stable sine wave? 3. Is there a ultra-stable motor-driver I can buy ready-made? 4. Is there another solution besides a synchronious motor that could provide a more stable rotation at these speeds? I hope there's someone out there smarter than me! Shawn

1996\10\29@111509 by Martin J. Maney

On Wed, 30 Oct 1996, Shawn Ellis wrote: > The problem is that the resulting sine wave is really 62.52 Hz (5 us too > fast). I have used several different sources for the 20Khz update pulse > with no change in performance so I don't think that could be the problem. I can't see offhand how the system described could be causing this error, especially if you've used an external clock source. I wonder if you have a problem caused by the measurement technique? If you're measuring the period of a single cycle, you may be seeing the effect of a bit of hysteresis in the input conditioning. Well, it's a thought.

1996\10\29@114157 by Shawn Ellis

>I can't see offhand how the system described could be causing this error, >especially if you've used an external clock source. I wonder if you have >a problem caused by the measurement technique? If you're measuring the >period of a single cycle, you may be seeing the effect of a bit of >hysteresis in the input conditioning. Well, it's a thought. > Well, I've used the frequency counter on a fluke oscilliscope but the ultimate sign of error is the time between activations of an optical sensor I have looking at a hole in a wheel attached to the motor is 5 microseconds too fast. Also bear in mind that the 62.5 Hz square wave I described before runs this same setup at the correct frequency, but with more acceleration/deceleration variences than I'd like.

1996\10\29@162722 by Scott Dattalo

Shawn, You might want to check out some simple sine wave theory: http://www.interstice.com/~sdattalo/technical/theory/sinewave.html And an implementation with a PIC: http://www.interstice.com/~sdattalo/technical/software/pic/picsine.html These may give you a couple of more ideas. I realize that "generating" your sine wave is not necessarily the problem. From your description, it sounds as though if there is a timing problem. Specifically, you're generating 320 samples for a 62.5Hz sine wave, which translates to 1/62.5/320 = 50us per sample. This means you need to update the DAC exactly once every 50us. And I assume that you're synchronized to the 20kHz PWM output to achieve this update rate. Is it possible that your crystal is too fast? (62.52-62.5)/62.5 ==> 0.032% error is not much error. This problem is solvable by adding more precision to your frequency variable. Currently, I assume every 50us you automatically increment some kind of "frequency" variable to access the next location in the sine wave table: f = f + 1; /* Increment at each sample */ if f>=160 then ... rollover/reverse However, instead of incrementing by exactly 1.000, you could do something like so: f = f + 1 + eps; /* Increase f at each sample */ where |eps| << 1. (way less than, not shift left by) In your case, if the problem really is a fast crystal, then eps will be 0.00032. So that after 160 samples, spaced 99.968us apart (fast crystal) f will rollover at: 160 (1 + eps) * 99.968us = 62.5 times per second The down side is that now you must (perhaps must is too strong) interpolate between entries in the sine table array. There are scaling issues that greatly simplify the problem of dealing with floating point numbers. Furthermore, the scaling can be designed such that you can take advantage of roll-over arithmetic. I'll elaborate if this is not too clear. I've used a variation of this technique for phase-locking to a 60Hz frequency before. There I used 20bits of resolution in the phase increment. Scott

1996\10\29@164342 by Beacham, Bill HSD

From: owner-piclist To: Multiple recipients of list PICLIST Subject: 62.5 Hz sine wave Date: Wednesday, October 30, 1996 10:09AM Hey all, this will be the first real problem I have presented this list and I hope SOMEONE can come through for me. Buttering up aside, I have a 12-bit DAC (AD737) interfaced to a PIC16C74. I'm useing the PCW "C" compiler from CCS. I am trying to build a ultra-stable AC synchronious motor driver. So I have a sine wave represented as a 160-point data set that I generated with a basic program and imported into my code as an array. This data-set represents one half of one period of the sine wave. The PIC sends each value in the set out to the DAC and at the end of the set subtracts the numbers from half scale to produce the negative part of the period, then the whole thing loops. The PIC is looking at a 20Khz, 90% duty cycle ttl square wave that comes from the PWM module. This square wave also updates the DAC. The result should be a 62.5 Hz, +5VDC to -5VDC sine wave coming out of the DAC. The PIC loads the new DAC value only after seeing the DAC get updated by the 20 Khz pulse and then waits for the next DAC update before loading the next number. The DAC update is then fed into an APEX high voltage amp to produce a large sine wave. The problem is that the resulting sine wave is really 62.52 Hz (5 us too fast). I have used several different sources for the 20Khz update pulse with no change in performance so I don't think that could be the problem. The device this control is intended for needs stability better than +/- 1us. (Thats less than +/- 1us per revolution at 62.5 Hz) Right now we are running it with a 62.5 hz ttl square wave, being fed into an inverting amplifier, then a transformer to produce a +/- 60 V square wave. Anybody have any suggestions on: 1. What could this problem be? 2. Is there a better/easier way to achieve an ultra-stable sine wave? 3. Is there a ultra-stable motor-driver I can buy ready-made? 4. Is there another solution besides a synchronious motor that could provide a more stable rotation at these speeds? I hope there's someone out there smarter than me! Shawn ............................................................................ ......... The 62.5 hz wave is a 16 ms signal. With 160 points, this corresponds to 100us resolution. If you are interrupt driven this should not be a problem. I would wonder, however, that since you are only 5 us off, what about the time effect of the code that is executed after the interrupt is received. The delay in code, if equal for all points, would shift the waveform equally for the first and last point, resulting in no error. If you have a decision process in the code that is different netween the first and last point, then one or the other could be shifted. I'm new to the PIC arena so I don't know what timing is consumed for each instruction but 5 us doesnt sound like much - Perhaps only a few instructions could account for the error. Good luck

1996\10\29@191620 by Steve Hardy

> From: Shawn Ellis <.....sellisKILLspam.....MERC.RX.UGA.EDU> > >I can't see offhand how the system described could be causing this error, > >especially if you've used an external clock source. I wonder if you have > >a problem caused by the measurement technique? If you're measuring the > >period of a single cycle, you may be seeing the effect of a bit of > >hysteresis in the input conditioning. Well, it's a thought. > > > Well, I've used the frequency counter on a fluke oscilliscope but the > ultimate sign of error is the time between activations of an optical sensor > I have looking at a hole in a wheel attached to the motor is 5 microseconds > too fast. > Also bear in mind that the 62.5 Hz square wave I described before runs this > same setup at the correct frequency, but with more acceleration/deceleration > variences than I'd like. > I'm in a very analogue mood today, so I would suggest feeding your existing square wave into an analogue band pass centred on that freq. I don't think the motor would notice a little 3rd harmonic distortion. If the motor still runs fast you definitely have a problem in your measuring equipment. Certain analogue filters will also give sine _and_ cosine output, which could be put to good use generating a nice rotating magnetic field: two coils angled at 90 deg to each other. Stick a permanent magnet inside this and voila - a synchronous motor. Regards, SJH Canberra, Australia

1996\10\29@220202 by Gerhard Fiedler

At 10:09 30/10/96 -0500, Shawn Ellis wrote: >The problem is that the resulting sine wave is really 62.52 Hz (5 us too >fast). I have used several different sources for the 20Khz update pulse >with no change in performance so I don't think that could be the problem. >... >1. What could this problem be? >2. Is there a better/easier way to achieve an ultra-stable sine wave? You might try to check your sine wave output synchronized to your 20kHz input (a counter does this very well), this might tell you whether the precision of the input frequency is your problem. And you can reduce the number of your points to the half -- did you ever notice that the second half of your half sine wave is exactly the first half in reversed order? With half the points you go forth and back, then negative forth and back...

1996\10\30@010421 by dfr

Shawn Ellis wrote: > The problem is that the resulting sine wave is really 62.52 Hz (5 us too > fast). I have used several different sources for the 20Khz update pulse > with no change in performance so I don't think that could be the problem. Just some thoughts on this one... The error is not really 5us. Note that 50(62.52-62.5)=1, which implies that an additional cycle is added every fifty seconds (3125 cycles). Also note that 3125*320=1,000,000. Hmmmm. The sine table is being advanced an extra position once every 1,000,000 sync cycles. Are you also running an ISR at the same time, and possibly over-correcting the table pointer? If the 62.52Hz figure is accurate then the above numbers look significant to me. Revealing their meaning is another matter. Hope this serves as food for thought. Regards, Dana Frank Raymond

1996\10\30@090406 by Shawn Ellis

> >Certain analogue filters will also give sine _and_ cosine output, which >could be put to good use generating a nice rotating magnetic field: >two coils angled at 90 deg to each other. Stick a permanent magnet >inside this and voila - a synchronous motor. > Interesting suggestion, let me clarify by saying the motor I'm using is a Globe model with 3 coils at 120 Deg. apart.

1996\10\30@090615 by Shawn Ellis

> >And you can reduce the number of your points to the half -- did you ever >notice that the second half of your half sine wave is exactly the first half >in reversed order? With half the points you go forth and back, then negative >forth and back... > I tried that at first, but the PIC had to make too many decision that way and I had trouble doing everything before the next update pulse.(50us) And I'm already running at 20 Mhz! The main time-eater is the fact that I half to keep track of and load three different DAC's for three different phases of the motor. (0, 120, 240)

1996\10\30@091853 by Shawn Ellis

> > The 62.5 hz wave is a 16 ms signal. With 160 points, this corresponds to >100us resolution. If you are interrupt driven this should not be a problem. >I would wonder, however, that since you are only 5 us off, what about the >time effect of the code that is executed after the interrupt is received. >The delay in code, if equal for all points, would shift the waveform equally >for the first and last point, resulting in no error. If you have a decision >process in the code that is different netween the first and last point, then >one or the other could be shifted. I'm new to the PIC arena so I don't know >what timing is consumed for each instruction but 5 us doesnt sound like much > - Perhaps only a few instructions could account for the error. > >Good luck > I wish that were the problem, but the DAC's all update on an independent 20 Khz square wave no matter what the cod is doing. I just have to get the value intot the DAC's regesters before the update pulse... And I can't see that taking to long, since I've timed it by fliping a bit before and after the loads and watched this on the 'scope with the 20 Khz and it seemed to be happening well within the 20 Khz

1996\10\30@092311 by Shawn Ellis

At 02:23 PM 10/29/96 -0800, you wrote: {Quote hidden}>Shawn, > >You might want to check out some simple sine wave theory: > >http://www.interstice.com/~sdattalo/technical/theory/sinewave.html > >And an implementation with a PIC: > >http://www.interstice.com/~sdattalo/technical/software/pic/picsine.html > >These may give you a couple of more ideas. > > >I realize that "generating" your sine wave is not necessarily >the problem. From your description, it sounds as though if there >is a timing problem. Specifically, you're generating 320 samples >for a 62.5Hz sine wave, which translates to 1/62.5/320 = 50us >per sample. This means you need to update the DAC exactly once >every 50us. And I assume that you're synchronized to the 20kHz >PWM output to achieve this update rate. Is it possible that your >crystal is too fast? (62.52-62.5)/62.5 ==> 0.032% error is not >much error. > > >This problem is solvable by adding more precision to your >frequency variable. Currently, I assume every 50us you automatically >increment some kind of "frequency" variable to access the next >location in the sine wave table: > >f = f + 1; /* Increment at each sample */ >if f>=160 then ... rollover/reverse > >However, instead of incrementing by exactly 1.000, you could do >something like so: > >f = f + 1 + eps; /* Increase f at each sample */ > >where |eps| << 1. (way less than, not shift left by) > >In your case, if the problem really is a fast crystal, then eps >will be 0.00032. So that after 160 samples, spaced 99.968us >apart (fast crystal) f will rollover at: > > 160 (1 + eps) * 99.968us = 62.5 times per second > >The down side is that now you must (perhaps must is too strong) >interpolate between entries in the sine table array. > >There are scaling issues that greatly simplify the problem of >dealing with floating point numbers. Furthermore, the scaling >can be designed such that you can take advantage of roll-over >arithmetic. I'll elaborate if this is not too clear. > > >I've used a variation of this technique for phase-locking to a >60Hz frequency before. There I used 20bits of resolution in the >phase increment. > >Scott > Good suggestion! But I've never done that sort of math before... also things don't really point to the 20 Khz since I already tried an independent update pulse coming from a counter in a computer sitting on my desk and the results were EXACTLY the same. I also tried three different 20 Mhz crystals in the PIC prduceing the 20 Khz, with no change... So I can only assume this frequency is correct.

1996\10\30@214233 by Steve Hardy

> From: Shawn Ellis <EraseMEsellisspam_OUTTakeThisOuTMERC.RX.UGA.EDU> > > >[SJH wrote:] > >Certain analogue filters will also give sine _and_ cosine output, which > >could be put to good use generating a nice rotating magnetic field: > >two coils angled at 90 deg to each other. Stick a permanent magnet > >inside this and voila - a synchronous motor. > > > Interesting suggestion, let me clarify by saying the motor I'm using is a > Globe model with 3 coils at 120 Deg. apart. > Motor theory is one of the many things I'm ignorant of. Do these 3-coil motors have a permanent magnet rotor? 3-phase AC motors have an inductively coupled 'squirrel cage' rotor which would slip relative to the rotating field if non-zero torque was applied. I presume this is the same for those small single phase 'synchronous' motors used to drive tape recorder capstans etc. I presume for a permanent magnet rotor, it will stay in lockstep with the rotating field, with a varying lag depending on the required torque. Anyway, now I see why analogue filtering of a single square wave wasn't satisfactory for you, since you need three phases. Since your clock is 20KHz, it needs to be divided by 320 to get the 62.5Hz fundamental. Rather than using the PIC to generate 3 sine waves, how about using it to divide the 20KHz by 320, producing 3 square wave outputs 120 degrees apart. The three square waves are fed into separate filters tuned to the same frequency (62.5Hz). Unfortunately, 320 is not divisible by 3. You can get around this in several ways: . Use the PIC to interpolate. . Use a PLL to multiply the 20KHz up to 60KHz, and use this as the clock. . Just don't worry: use phase spacing of 107, 107 and 106 cycles. In the latter case, the worst case angular error of the field vector will be of the order of 12 minutes of arc. This would probably be better than the accuracy at which the field coils were placed. Of course you would have to ensure there is no torsional resonance at harmonics of the rotational speed in any case. Another technique may be to use phase shift networks to produce successive phase lags of 120 degrees. This may have the advantage of being able to tweak the outputs for maximum smoothness of the rotation. If you used a 320-count shaft encoder (Litton, HP etc.) you could compare the 20KHz output from the encoder with the clock signal - use an XOR gate which would show up any speed difference as a varying PWM, or the PIC could count the encoder pulses to derive an error signal. The possibilities are endless. Good luck. Regards, SJH Canberra, Australia

1996\10\30@234943 by dfr

Shawn Ellis wrote: > Well, you'll be interested to know that I tried putting a check in my code > and at every 3125 th data point, I held all the indexes back one point. > This worked! But is very bad for stability since the wave jumps by one data > point every 50 secs. Now if only I could figure out why that happens or how > to spred that one point across 3125 points... Its one of those "gotta be there" kind of problems. In previous messages you mentioned that you are running your PIC at 20Mhz. As well, you stated that you are measureing the 62.52Hz at the spindle. This is what I would do: 1) Verify that my frequency counter is not the source of there error. 2) Measure the frequency of the sine waves themselves as the mechanical elements of the system may be converting instantaneous error (glitches) into a smooth frequency error. 3) Use a logic analyser (with a large memory) to log all values written into the 3 DACs over a long period of time. Analyze the data for discontinuities or other instantaneous errors. Veryify that the output frequency (not the spindle RPS) is in fact in error. 4) Verify that worst case timing meets minimum spec at all times. 5) Ensure that the 20Khz sync signal does not contain any noise or sloppy transition delays (slew rate, etc). 6) Examine the CCS code for errors or funnies. I've had to do this for other PIC C compilers, though I have no experience with recent CCS versions. 7) Ensure that the CCS produces code that DISABLES interrupts, not just IGNORES them. What about peripherals unused and THOUGHT to me inactive? 8) Post the code for public comment. (there may be non-disclosure issues here). I can't think of anything else. Happy hunting shawn. -- Regards, Dana Frank Raymond

1996\10\31@041724 by fastfwd

Shawn Ellis <PICLISTspam_OUTMITVMA.MIT.EDU> wrote: > things don't really point to the 20 Khz since I already tried an > independent update pulse coming from a counter in a computer > sitting on my desk and the results were EXACTLY the same. I also > tried three different 20 Mhz crystals in the PIC prduceing the 20 > Khz, with no change... So I can only assume this frequency is > correct. Shawn: Is there a chance that you're not really sending exactly the number of DAC values that you think you are? Maybe you're "doubling-up" on one of your lookup-table values? You can check this by replacing all the sine values in your lookup table with alternating full-scale (FF or whatever) and zero-scale (00) numbers, then look at the resulting output square wave and see if any double-wide pulses or gaps are generated. -Andy Andrew Warren - @spam@fastfwdKILLspamix.netcom.com Fast Forward Engineering, Vista, California http://www.geocities.com/SiliconValley/2499

1996\10\31@042354 by Bjarne Nebelong

Shawn, I think that I want to bother you with some vector theory that I am using on a three phase generator that I made for a costumer (in a 87C749, sorry). You got three coils and the current in these can only go in two directions: Coils I ____ ---|____|--- --> <-- ____ ---|____|--- --> <-- ____ ---|____|--- --> <-- This gives only six possibilities. i.e. with 3-bits you can control a rotation. uvw A A: 100 1=Current + to - _____ B: 110 0=Current - to + F / \ B C: 010 / \ D: 011 Count A-F-A-F... gives clockwise dir. \ / E: 001 Count F-A-F-A... gives the other dir. E \_____/ C F: 101 D With the above table you got a rotation in only six values, six corner vector. The problem with this is that the corners is rather long from a perfect circle, it gives overmagnetisation there (harmonic distortion), this is solved by adding corners: A _____ B \__A \ \B \ This gives 18 corners, that are suitable for most situations. I can provide you with that table if you want. (48 steps) If you want to control the output voltage, with your sinewave you change the amplitude, you have to switch to a non "currentfloating" mode, 000 or 111. Intelligent stopping means that you select the one that only change one bit to do the stop. i.e.: 100 -> 000 110 -> 111 010 -> 000 011 -> 111 001 -> 000 101 -> 111 This can be done with a trianglewave, a comparator and the reference. I use the PWMoutput on the MCU that got a built in comparator. Input the comparator output in a MUX 74HC157: u ----> A1 Y1 ----> U v ----> B1 Y2 ----> V w ----> C1 Y3 ----> W stopbit -+--> A2 +--> B2 +--> C2 STOP ----> Select A/B stopbit 1 if 111 0 if 000 STOP 1 = stop (from comparator or PWM output) 0 = RUN I am doing this outside the MCU becouse the timing is essential here, no time for IRQ. To control the output stage you need the complement: uvwuvw UVW +++--- A: 100 100011 B: 110 110001 C: 010 010101 D: 011 011100 E: 001 001110 F: 101 101010 + _________________________________ | | | b |c b |c b |c u+ ---|< v+ ---|< w+ ---|< |e |e |e | | | +-- L1 +-- L2 +-- L3 | | | b |c b |c b |c u- ---|< v- ---|< w- ---|< |e |e |e |_______________|_______________| - You will also nead some kind of deadtime generator, but I suppose that you got that already. Hope you got some help out of this, feel free to contact me if you nead more. Best regards and good luck. Bjarne Nebelong _____________________________________________________________ Bjarne Nebelong Internet : KILLspamb.nebelongKILLspamhelsingborg.se nebelong systems Phone & Fax : +46 42 12 82 21 Wrangelsgatan 25 Mobile : +46 70 579 05 50 25440 Helsingborg, SWEDEN ______--_--__-__--_____-__-__-_-_-__-__-_-_-__--_-___-_______

'Running a Consulting Business'
1996\12\12@053341 by fastfwd

Dudes: I really don't want to spend too much time discussing this topic on the list, but the following may be helpful to anyone contemplating going into my line of work. There's a guy here on the list (he just lurks, though... I think he's posted exactly ONE message) who taught me what I think are the three most important things you need to know about running an engineering-consultation business. Aside from his sure-fire way of knowing exactly WHEN and BY HOW MUCH to adjust your rates for inflation, which I won't describe until I get his permission (and maybe not even then, since it's sorta embarrassing to admit that I set my rates that way), he told me: 1. When an engineer who's only done garage-type development tells you how long it'll take him to perform some task, multiply it by two and bump the unit of time to the next-larger convenient measure. Example: I ask an eager but inexperienced engineer how long it'll take him to change a resistor value on his schematic. He says thirty seconds. I multiply thirty by two and get sixty, then bump the units from seconds to minutes... I come up with sixty minutes, which is about what it'll REALLY take him by the time he's made the change to his schematic, printed it, checked it, noticed that he forgot to update the revision level, bumped the rev, printed it again, checked it again, looked up the new part number and made the change to the BOM, and informed the PCB designer and anyone else who needs to know. This works on big projects, too... Witness the recent back-and-forth list discussions regarding the time that it takes to do an "average" PIC project. One guy says 2 weeks; the next says 4 months. 2. Take only: a) interesting jobs, from b) nice people who c) have money. Once you're established and have a reputation, you'll be offered more work than you can handle... There's no reason, at that point, to settle for anyone who doesn't meet all three of the above-mentioned criteria. When you're first starting out, of course, you may have to take some uninteresting jobs... But don't waste your time working for anyone who fails the "nice people" or "has money" tests. -Andy === Andrew Warren - RemoveMEfastfwdTakeThisOuTix.netcom.com === === Fast Forward Engineering - Vista, California === === === === Custodian of the PICLIST Fund -- For more info, see: === === http://www.geocities.com/SiliconValley/2499/fund.html ===

'Measuring frequency of sine wave'
1997\01\07@082724 by Stephen Brown

Hi, I'm interested in measuring the frequency of a pretty clean sine wave (output from a switched cap (not to narrow bandwidth) bandpass filter) using a 16c84. This is for a musical instrument tuning application. Presumably I would need a 0-5v square wave coming into the pic for a frequency counter routine. Any ideas on the best way to convert a sine wave to a square wave for this purpose (e.g. comparator, overdriven amplifier, PLL)? I played around with a LMC568 PLL circuit to lock onto the sine wave signal and output a pulse train, it works but the problem is that the lock in frequency range for this device is +-60% of the center frequency -- too narrow for this application. Ideally, I need to be able to handle a bit more than an octave above and below 440 Hz (i.e. about 200Hz-1000Hz). The CD4046 ?? PLL chip can be configured for a much wider range, but it is my understanding that it needs a square wave input itself (am I wrong?). The nice thing about these PLL things is that they can create a logic signal indicating when they are locked onto a frequency signal -- i.e. they can tell you when a valid input is there. Anyway, any hints would be appreciated. Steve Brown spamBeGonesbrownspamBeGonened.ara.com TakeThisOuTbrownsEraseMEspam_OUTsover.net

1997\01\07@085640 by BLEGER

Steve, How about this for simplicity: Rectify the sine wave with a diode, then use a zener diode and resistor to limit the voltage to logic levels (approx 0 to 4 volts) then use a schmidt trigger to clean up the edges of the waveform. This assumes the sine wave is high enough. If not it can be amplified with a trasistor or op amp. In__|\|___ |/| / \ R |--------| /__________|Schmidt |_______ Out | |Trigger | __/ Zener |Logic IC| //\ |--------| | | ///Ground ______________________________ Reply Separator _________________________________ Subject: Measuring frequency of sine wave Author: pic microcontroller discussion list <RemoveMEPICLISTTakeThisOuTMITVMA.MIT.EDU> at smtp Date: 1/7/97 8:29 AM Hi, I'm interested in measuring the frequency of a pretty clean sine wave (output from a switched cap (not to narrow bandwidth) bandpass filter) using a 16c84. This is for a musical instrument tuning application. Presumably I would need a 0-5v square wave coming into the pic for a frequency counter routine. Any ideas on the best way to convert a sine wave to a square wave for this purpose (e.g. comparator, overdriven amplifier, PLL)? I played around with a LMC568 PLL circuit to lock onto the sine wave signal and output a pulse train, it works but the problem is that the lock in frequency range for this device is +-60% of the center frequency -- too narrow for this application. Ideally, I need to be able to handle a bit more than an octave above and below 440 Hz (i.e. about 200Hz-1000Hz). The CD4046 ?? PLL chip can be configured for a much wider range, but it is my understanding that it needs a square wave input itself (am I wrong?). The nice thing about these PLL things is that they can create a logic signal indicating when they are locked onto a frequency signal -- i.e. they can tell you when a valid input is there. Anyway, any hints would be appreciated. Steve Brown sbrownEraseME.....ned.ara.com EraseMEbrownssover.net

1997\01\07@094353 by efoc

Stephen Brown wrote: {Quote hidden}> > Hi, > > I'm interested in measuring the frequency of a pretty clean sine wave > (output from a switched cap (not to narrow bandwidth) bandpass filter) > using a 16c84. This is for a musical instrument tuning > application. Presumably I would need a 0-5v square wave coming into the > pic for a frequency counter routine. Any ideas on the best way to > convert a sine wave to a square wave for this purpose > (e.g. comparator, overdriven amplifier, PLL)? I played around with a > LMC568 PLL circuit to lock onto the sine wave signal and output a > pulse train, it works but the problem is that the lock in frequency > range for this device is +-60% of the center frequency -- too narrow > for this application. Ideally, I need to be able to handle a bit more > than an octave above and below 440 Hz (i.e. about 200Hz-1000Hz). The > CD4046 ?? PLL chip can be configured for a much wider range, but it > is my understanding that it needs a square wave input itself (am I > wrong?). The nice thing about these PLL things is that they can create > a logic signal indicating when they are locked onto a frequency signal > -- i.e. they can tell you when a valid input is there. > > Anyway, any hints would be appreciated. > > Steve Brown > > RemoveMEsbrownEraseMEEraseMEned.ara.com > RemoveMEbrownsspam_OUTKILLspamsover.net Why not simply drive it into a schmitt trigger inverter Peter ....... -- ================================== = New Ideas come from those who = = didn't know it wasn't possible = ==================================

1997\01\07@120416 by Stuart Allman

try measuring the zero crossings. It seems to work for other projects I have seen. Stuart Allman RemoveMEstudioTakeThisOuTspamhalcyon.com

1997\01\07@171208 by peter

Stephen Brown wrote: > > Hi, > > I'm interested in measuring the frequency of a pretty clean sine wave > (output from a switched cap (not to narrow bandwidth) bandpass filter) > using a 16c84. This is for a musical instrument tuning > application. Presumably I would need a 0-5v square wave coming into the > pic for a frequency counter routine. Any ideas on the best way to > convert a sine wave to a square wave for this purpose > (e.g. comparator, overdriven amplifier, PLL)? just use an opamp overdriven, this will give you a nice square wave to feed the pic, you'll probably find you don't need to use the filter Alternately you could try an opamp conected to a diode conected to a zener conected to a schmitt conected to an inverting schmitt conected to the washing machine -- Peter Cousens email: EraseMEpeterspamspamBeGonecousens.her.forthnet.gr snailmail: Peter Cousens, karteros, Heraklion, Crete, 75100, Greece, phone: + 3081 380534, +3081 324450 voice/fax After Bill Gates announced to the world that he was Microsoft, his wife was asked to comment. She said that as his wife, she had been the first to notice this problem

1997\01\07@191658 by n Medical Technologies

Stephen Brown wrote: {Quote hidden}> > Hi, > > I'm interested in measuring the frequency of a pretty clean sine wave > (output from a switched cap (not to narrow bandwidth) bandpass filter) > using a 16c84. This is for a musical instrument tuning > application. Presumably I would need a 0-5v square wave coming into the > pic for a frequency counter routine. Any ideas on the best way to > convert a sine wave to a square wave for this purpose > (e.g. comparator, overdriven amplifier, PLL)? I played around with a > LMC568 PLL circuit to lock onto the sine wave signal and output a > pulse train, it works but the problem is that the lock in frequency > range for this device is +-60% of the center frequency -- too narrow > for this application. Ideally, I need to be able to handle a bit more > than an octave above and below 440 Hz (i.e. about 200Hz-1000Hz). The > CD4046 ?? PLL chip can be configured for a much wider range, but it > is my understanding that it needs a square wave input itself (am I > wrong?). The nice thing about these PLL things is that they can create > a logic signal indicating when they are locked onto a frequency signal > -- i.e. they can tell you when a valid input is there. > > Anyway, any hints would be appreciated. > > Steve Brown > > RemoveMEsbrownKILLspamned.ara.com > brownsSTOPspamspam_OUTsover.net To convert a Sine wave to a square wave, you could use either a single-supply comparator or op-amp. A bit of hysteresis would be good. Actually, depending on the amplitude of the sine wave, the best and simplest bet might simply use a CMOS Schmitt Trigger. I haven't used a PLL in a bunch of years, so I've forgotten how to do this. Sorry. Al Cohen Boston Medical Technologies

1997\01\08@022456 by Oyvind Kaurstad

Stephen Brown wrote: {Quote hidden}>Hi, >I'm interested in measuring the frequency of a pretty clean sine wave >(output from a switched cap (not to narrow bandwidth) bandpass filter) >using a 16c84. This is for a musical instrument tuning >application. Presumably I would need a 0-5v square wave coming into the >pic for a frequency counter routine. Any ideas on the best way to >convert a sine wave to a square wave for this purpose >(e.g. comparator, overdriven amplifier, PLL)? I played around with a >LMC568 PLL circuit to lock onto the sine wave signal and output a >pulse train, it works but the problem is that the lock in frequency >range for this device is +-60% of the center frequency -- too narrow >for this application. Ideally, I need to be able to handle a bit more >than an octave above and below 440 Hz (i.e. about 200Hz-1000Hz). The >CD4046 ?? PLL chip can be configured for a much wider range, but it >is my understanding that it needs a square wave input itself (am I >wrong?). The nice thing about these PLL things is that they can create >a logic signal indicating when they are locked onto a frequency signal >-- i.e. they can tell you when a valid input is there. >Anyway, any hints would be appreciated. >Steve Brown You could do this: Take two resistors (100k for instance) and use them for a voltage divider from your 5v and to ground. This will give you half the supply voltage. Feed this into the comparator. (Either input) Make a similar voltage divider for the other comparator input and couple your sine wave to this input using a proper capacitor (0.1 uF will do). This will DC shift (and high-pass filter) your incoming signal to half the supply voltage. Cut-off frequency of the filter will be at approx 30 Hz. The comparator will give you a nice square wave. You will probably not need any hysteresis since you have a clean input signal. -Oyvind spamBeGoneoyvindSTOPspamEraseMEnofac.abb.no

'PIC as controller for sinewave inverter - magic si'
1997\02\03@071245 by Mike

Hi gals and guys, I'm looking (again) at feasability of using HC11 architecture as controller for sine-wave inverters 48V DC in for 240V AC out 50Hz and 5000VA. Now after being exposed to PICs - I'm considering using a PIC with built in A to D... Although I have most experience with 8051 architecture, it seems that the HC11 with OTP would be a more viable solution in terms of usability of programming model and level of integration - thats an option but, can anyone care to suggest a suitable PIC for such a task ? There has been some discussion in various hobby magazines about 'magic sinewaves' in 384 bit sequences - anybody know about these or the background maths etc - I was going to look at Taylor's series etc to produce the HC11 table constants but there might be a more efficient way ? I'm interested in any +ve/-ve comments/observations on this... Rgds Mike KILLspamerazmusspamBeGonewantree.com.au There is no a'priori reason that the ultimate truth will be interesting or even useful, those moments of frustration during philosophical debate would be replaced by the sheer terror which accompanies true knowledge.

1997\02\03@191707 by Dave Mullenix

Mike wrote: >There has been some discussion in various hobby magazines about 'magic >sinewaves' in 384 bit sequences - anybody know about these or the >background maths etc - I was going to look at Taylor's series etc to >produce the HC11 table constants but there might be a more efficient way ? > >I'm interested in any +ve/-ve comments/observations on this... Don Lancaster has written about "magic sinewaves". Don has a homepage at http://www.tinaja.com/ Enter the Reference Library, then the Magic Sinewaves Shelf and find this blurb: Don Lancaster's Magic Sinewave Library "Magic sinewaves" are repeating long sequences of ones and zeros. Digital sinewaves with precisely controlled amplitudes and amazingly low distortions. Compared to traditional PWM, magic sinewaves can offer far higher efficiencies and lower distortions. With circuitry that is elegantly simple and microcontroller friendly. For use in such applications as induction motor speed controls, electric autos, solar panels, power factor correction, inverters, home energy efficiency improvers, and specialized utility services. There are more magic sinewaves than there are particles in the universe. Only recently have specialized analysis tools become available. Don Lancaster, Synergetics, and several Synergetics Partners are in the forefront of this billion dollar opportunity. Extensive seminars, consulting, sourcecode, tutorials, and a new Partners program are now being offered. These files are available for downloading: Magic Sinewave Development Proposal. Formal proposal outlining seminars, partners, and services. A wide range of opportuinities and possibilities are listed, along with full references. Blatant Opportunist #35 MAGSINE.PDF (*) Magic sinewaves are a brand new method of driving ac induction motors, solar panels, inverters, or electric cars. Offer ing outstanding simplicity and efficiency. Introductory Don Lancaster tutorial. Blatant Opportunist #37 EMERGOP4.PDF (*) Fourth annual release of emerging technical opportunities. Push me pull you. The TeraHertz mystery band. PIC microcontollers. Magic Sinewaves. Solitons. DNA computing. Binary chain codes. Bionomial Goes Binary ONESWORD.HTML * A PostScript utility to generate all possible binary words of length "n" having "k" ones in them. Use for magic sinewaves and exploration of the binomial theorm. Requires two way recordable com. An analysis of Magic Sinewave Higher Harmonics. Just where do those square corners come from on a low harmonic waveform? Jim Fitzsimons analyzes the first 999 (!) harmonics of five typical 384 bit non-delta magic sinewaves. Demo sampler of 288-Bit Magic Sinewaves. A detailed analysis pf a few of the 288-bit "delta Friendly" magic sinewaves. There are 49 amplitudes in the complete 288 sequence, having distortions as low as 0.01 percent. Tech Musings #101 MUSE101.PDF ( June 1996 ) Using Delta-Wye transforms, Professional audio resources, Acrobat amber beta software, Building your own Tesla coil, Three phase magic sinewaves, more. Tech Musings #93 MUSE93.PDF ( October 1995 )* More on magic sinewaves, stepper motor driver chips, yet another patent horror story, mid range Mench computer, a collection of PCMCIA plug-in card resources. Tech Musings #90 MUSE90.PDF ( July 1995)* Fourier Series analysis, alternatives to plated through holes, classic computer resources, a $200 GPS receiver, Newtek's new video toaster for Windows. Hardware Hacker #87 HACK87.PDF (April 1995 )* AC motor drive fundamentals. New Basic Stamps from Parallax. PIC Microcomputer resources. Additional details on forced zero magic sinewaves. Dealing with "too good to be true" results. Hardware Hacker #86 HACK86.PDF (March 1995 )* The DNA computer language. Some wavelet book resources. Magic digital sinewave codes. Engineering economics review. Hot new fringe FM RBDS tuner.

1997\02\03@194411 by Steve Hardy

{Quote hidden}> From: Mike <EraseMEerazmusEraseMEWANTREE.COM.AU> > Hi gals and guys, > > I'm looking (again) at feasability of using HC11 architecture as controller > for sine-wave inverters 48V DC in for 240V AC out 50Hz and 5000VA. > > Now after being exposed to PICs - I'm considering using a PIC with built > in A to D... > > Although I have most experience with 8051 architecture, it seems that the > HC11 with OTP would be a more viable solution in terms of usability > of programming model and level of integration - thats an option but, can > anyone care to suggest a suitable PIC for such a task ? > > There has been some discussion in various hobby magazines about 'magic > sinewaves' in 384 bit sequences - anybody know about these or the > background maths etc - I was going to look at Taylor's series etc to > produce the HC11 table constants but there might be a more efficient way ? > > I'm interested in any +ve/-ve comments/observations on this... So-called 'magic sinewaves' seem to be a pet subject of a bloke called Don Lancaster. The actual mathematics to generate the sequences is not in the public domain; however there are a few published sequences of length 384 and above - look at his web page. I'm not sure about this, but I think it is possible to patent algorithms in the US. However, you can't in Australia so if you can discover his algorithm for yourself there would be nothing to stop you commercially develping products in Oz. Basically, there are an infinite number of PWM sequences which will generate 'sine waves' with a greater or lesser harmonic content. The aim of 'magic sinewaves' is to minimise (or even zero) the first n harmonics, such that the first non-zero harmonics will be at a very high frequency and thus easy to filter using an LC low-pass power filter. The complete cycle (20ms) is divided into fixed time slots (20ms/384). Each slot is then assigned an output voltage of +V, 0 or -V. Naturally, because of the 4-fold symmetry of a sine wave only one quadrant of the sine wave needs to have table entries. The power of the fundamental (which is what you want) is proportional to the total number of +V slots (= the number of -V). The mathematical technique is based on Fourier analysis (not Taylor series, AFAIK). If you need to regulate the output, you can presumably narrow the utilisation of the +/-V slots. Thus you would aim for 50% duty cycle (of the slot) for no-load output, allowing this to rise to 100% for full load. As it so happens, I am currently designing a 6KW inverter, except that I am using rectified 3-phase as the power source (giving about 580V avg). The DC (with 4.2% ripple) is switched by a full bridge of MOSFETs using opto-isolated control. Control will be provided by a 16C74 - primarily because I have a few of these floating around. The '74 will be used to monitor voltage and current (current being crudely measured from the voltage drop across the MOSFETs Rds(on)). It will also determine frequency of operation and PWM sequencing. Might also add temperature sense. One of the most difficult things about controlling 600 odd volts at 50KHz is the tremendous dV/dt (those FETs switch fast!) which will couple through the tiniest capacitance to potentially disrupt ones delicate control circuitry. I am having to build up my own opto-isolators and isolated power supplies. Total capacitance has to be limited to 0.2pF, otherwise currents of over 1mA will be induced (6000V/us). I would be interested in your overall converter topology. Please correspond privately since this is wildly off-topic. Regards, SJH Canberra, Australia

1997\02\04@111945 by Brian Boles

Before you go for that HC11, check out the 17C756 that will be sampling soon. It will "out motor" a HC11! Rgds, Brian. ______________________________ Reply Separator _________________________________ Subject: PIC as controller for sinewave inverter - magic sinewaves Author: Mike <@spam@erazmus@spam@spam_OUTWANTREE.COM.AU> at Internet_Exchange Date: 2/3/97 8:10 PM Hi gals and guys, I'm looking (again) at feasability of using HC11 architecture as controller for sine-wave inverters 48V DC in for 240V AC out 50Hz and 5000VA. Now after being exposed to PICs - I'm considering using a PIC with built in A to D... Although I have most experience with 8051 architecture, it seems that the HC11 with OTP would be a more viable solution in terms of usability of programming model and level of integration - thats an option but, can anyone care to suggest a suitable PIC for such a task ? There has been some discussion in various hobby magazines about 'magic sinewaves' in 384 bit sequences - anybody know about these or the background maths etc - I was going to look at Taylor's series etc to produce the HC11 table constants but there might be a more efficient way ? I'm interested in any +ve/-ve comments/observations on this... Rgds Mike spamBeGoneerazmusKILLspamwantree.com.au There is no a'priori reason that the ultimate truth will be interesting or even useful, those moments of frustration during philosophical debate would be replaced by the sheer terror which accompanies true knowledge.

1997\02\04@113613 by Mike

At 09:04 AM 4/02/97 -0700, you wrote: > Before you go for that HC11, check out the 17C756 that will be > sampling soon. It will "out motor" a HC11! > > Rgds, Brian. Sounds OK, I checked it out on microchip's site, nice chip. Might have a bit more horsepower than I need, can I get it for A$9 each in tube lots and the assembler for free ? Rgds Mike

1997\02\05@205411 by sdattalo

Mike wrote: > > There has been some discussion in various hobby magazines about 'magic > sinewaves' in 384 bit sequences - anybody know about these or the > background maths etc - I was going to look at Taylor's series etc to > produce the HC11 table constants but there might be a more efficient way ? > I don't believe in magic. I know for a fact that this routine efficiently generates sine waves with 8-bit resolution: http://www.interstice.com/~sdattalo/technical/software/pic/picsine.html And so does Eric Smith's: http://www.brouhaha.com/~eric/pic/sine.html And if you want to examine a few more ways to possible generate sine waves then check out: http://www.interstice.com/~sdattalo/technical/theory/sinewave.html Scott

1997\02\09@131012 by Sten Dahlgren

For all of you interested in PIC's and power inverter, check out there web site for the RD001 paper, "Uninterupptible Power Supply Reference Design". --- Sten Dahlgren CelsiusTech Systems ! "I'd rather have 39 Hp under my arm S-175 88 Jaerfaella Sweden ! than one under my backside" .....sedaspam_OUTcelsiustech.se +46-8 58084430 ! join your nearest karting club now !!

1997\02\09@151012 by peter

Sten Dahlgren wrote: > > For all of you interested in PIC's and power inverter, check out there web > site > for the RD001 paper, "Uninterupptible Power Supply Reference Design". > --- > Sten Dahlgren CelsiusTech Systems ! "I'd rather have 39 Hp under my arm > S-175 88 Jaerfaella Sweden ! than one under my backside" > TakeThisOuTseda.....TakeThisOuTcelsiustech.se +46-8 58084430 ! join your nearest karting club now !! Which web site are you refering to ?? -- Peter Cousens email: TakeThisOuTpeterKILLspamspamcousens.her.forthnet.gr snailmail: Peter Cousens, karteros, Heraklion, Crete, 75100, Greece, phone: + 3081 380534, +3081 324450 voice/fax After Bill Gates announced to the world that he was Microsoft, his wife was asked to comment. She said that as his wife, she had been the first to notice this problem

1997\02\10@000947 by Mike

At 07:09 PM 9/02/97 +0100, you wrote: >For all of you interested in PIC's and power inverter, check out there web >site >for the RD001 paper, "Uninterupptible Power Supply Reference Design". >--- >Sten Dahlgren CelsiusTech Systems ! "I'd rather have 39 Hp under my arm >S-175 88 Jaerfaella Sweden ! than one under my backside" >.....sedaRemoveMEcelsiustech.se +46-8 58084430 ! join your nearest karting club now !! Which web site are you referring to - here ? Rgds Mike

1997\02\11@121200 by Zack Cilliers

Hi there! Can some one send me some more information as how the sinewave can improve on a squarewave. thanks ---------- {Quote hidden}> From: Scott Dattalo <RemoveMEsdattalospamBeGoneunix.sri.com> > To: Multiple recipients of list PICLIST <spamBeGonePICLIST@spam@spam_OUTMITVMA.MIT.EDU> > Subject: Re: PIC as controller for sinewave inverter - magic sinewaves > Date: Thursday, February 06, 1997 3:51 AM > > Mike wrote: > > > > There has been some discussion in various hobby magazines about 'magic > > sinewaves' in 384 bit sequences - anybody know about these or the > > background maths etc - I was going to look at Taylor's series etc to > > produce the HC11 table constants but there might be a more efficient way ? {Quote hidden}> > > > I don't believe in magic. I know for a fact that this routine > efficiently generates sine waves with 8-bit resolution: > > http://www.interstice.com/~sdattalo/technical/software/pic/picsine.html > > And so does Eric Smith's: > > http://www.brouhaha.com/~eric/pic/sine.html > > And if you want to examine a few more ways to possible > generate sine waves then check out: > > http://www.interstice.com/~sdattalo/technical/theory/sinewave.html > > > > Scott

1997\02\12@180550 by sdattalo

Zack Cilliers wrote: > > Hi there! > > Can some one send me some more information as how the sinewave can improve > on a squarewave. > > thanks Zack, Are you asking how the sinewave can be generated from the square wave? Well, if you aren't then I'm answering the wrong question... A couple of weeks ago John Payson and I were discussing some details on how it is possible to suppress the 2nd and 3rd harmonics in a sampled square wave. This led me on an analytical rabbit chase that turns out to be the inverse of the so called "magic sine wave". Skipping over the details, let me just pull a formula out of my magic hat. Suppose you have a very narrow duty cycle pulse train: ^ | tau | <----> | | +----+ +----+ | | | | | -+-----+ +------------------------+ +---------> t phi <-----> <-------------- T ------------> In other words, if you have a pulse stream that has a frequency of 1/T, pulse widthes of tau, and an initial phase of phi, then one (of an infinite many) series expansion is: f(t) = d + inf ---- \ sin(n*pi*d) / n*2*pi tau \ 2* / ----------- * cos| ------ *( t - --- - phi) | ---- n*pi \ T 2 / n=1 where, d = tau/T is the duty cycle pi = 3.141592653... t = time For 50% duty cycles, you may recall that there are no even harmonics present in the fourier series. So as one simple check, you can substitute d=0.5 and see that the sin(n*pi/2) kills the even harmonics in this parameterized expansion. The goal of the magic sinewaves is to suppress many if not all of the harmonics beyond the fundamental. This can be accomplished by chopping the period into many fine pieces; in other words, let tau<<T. Mike made a reference to there being 384 divisions. Now if you strategically place these narrow pulses among the 384 possible slots, it's possible to suppress higher harmonics. The technique of finding the optimum placement is far from trivial. However, here are few helpful observations. The fundamental frequency is 1/T. If the number of subdivisions is N (i.e. tau = T/N) then 1) Harmonics that evenly divide N/2 can be totally suppressed. 2) If the n'th harmonic is suppressed then the N-n'th is also suppressed. 3) To suppress the 2nd through the n'th harmonic at least 2^(n-1) pulses are needed. As an example, if you wanted to suppress the 2'nd, 3'rd and 4'th harmonics then the number of needed subdivisions is N = 2*3*4 = 24 Out of the 24 possible pulses, 2^(4-1) = 8 pulses are needed. (And if I had my notes, I would tell you where these optimum pulse positions are located). In addition to suppressing these harmonics, the 20'th, 21'st and 22'nd are also suppressed. The fifth harmonic is not completely suppressed. There is also a 1/n factor that will attenuate those harmonics that are not suppressed. However for small n, the sin(n*pi*d)/n*pi is approximately n*pi*d/(n*pi) = d. So the 1/n factor only becomes important for larger values of n. There are many, many more details that I'm ignoring... Scott

1997\02\14@131742 by sdattalo

This is my second time to send this to the list. The first time apparently didn't make it. (At least it was not in this month's log.) Zack Cilliers wrote: > > Hi there! > > Can some one send me some more information as how the sinewave can improve > on a squarewave. I believe the context of your question is WRT power inverters? In which case the sine wave drive is lethal to the switching devices. The so called "magic sine waves" is a glitzy synonym for pulse width modulation. (O.k. there's a little more to it than PWM, but the underlying theory is identical.) And PWM is the proper technique for driving many these devices. A couple of weeks ago John Payson and I were discussing some details on how it is possible to suppress the 2nd and 3rd harmonics in a sampled square wave. This led me on an analytical rabbit chase that turns out to be the inverse of the so called "magic sine wave". Skipping over the details, let me just pull a formula out of my magic hat. Suppose you have a very narrow duty cycle pulse train: ^ | tau | <----> | | +----+ +----+ | | | | | -+-----+ +------------------------+ +---------> t phi <-----> <-------------- T ------------> In other words, if you have a pulse stream that has a frequency of 1/T, pulse widthes of tau, and an initial phase of phi, then one (of an infinite many) series expansion is: f(t) = d + inf ---- \ sin(n*pi*d) / n*2*pi tau \ 2* / ----------- * cos| ------ *( t - --- - phi) | ---- n*pi \ T 2 / n=1 where, d = tau/T is the duty cycle pi = 3.141592653... t = time For 50% duty cycles, you may recall that there are no even harmonics present in the fourier series. So as one simple check, you can substitute d=0.5 and see that the sin(n*pi/2) kills the even harmonics in this parameterized expansion. The goal of the magic sinewaves is to suppress many of the harmonics beyond the fundamental. This can be accomplished by chopping the period into many fine pieces; in other words, let tau<<T. Mike made a reference to there being 384 divisions. Now if you strategically place these narrow pulses among the 384 possible slots, it's possible to suppress higher harmonics. The technique of finding the optimum placement is far from trivial. However, here are few helpful observations. The fundamental frequency is 1/T. If the number of subdivisions is N (i.e. tau = T/N) then 1) Harmonics that evenly divide N/2 can be totally suppressed. 2) If the n'th harmonic is suppressed then the N-n'th is also suppressed. 3) To suppress the 2nd through the n'th harmonic at least 2^(n-1) pulses are needed. 4) It is possible to suppress all of the even harmonics. This is accomplished by an pulse train that is followed by it inverse. For example, if you had a traing of pulses like 011000 the even harmonics are suppressed by the doubled chain: 011000100111. To illustrate, suppose you wanted to suppress the 2'nd and the 3'rd harmonics. The number of needed subdivisions is N/2 = 2*3 = 6 or N = 12 Out of the 12 possible pulses, 2^(3-1) = 4 pulses are needed. And their locations are: 101101000000 (actually any cyclic rotation of this stream works). In addition to suppressing the 2'nd and 3'rd harmonics, the 10'th and 11'th are also suppressed. The fourth harmonic is not completely suppressed. If you wanted to suppress the 4'th or all of the even harmonics for that matter, then the following pulse stream would do the trick: original <----------> 101101000000010010111111 <----------> inverted from the original There is also a 1/n factor that will attenuate those harmonics that are not suppressed. However for small n, the sin(n*pi*d)/n*pi is approximately n*pi*d/(n*pi) = d. So the 1/n factor only becomes important for larger values of n. There are many, many more details that I'm ignoring... Scott

1997\02\15@160058 by Sten Dahlgren

---------- > From: Peter Cousens <TakeThisOuTpeterspamcousens.her.forthnet.gr> > To: PICLISTEraseMEMITVMA.MIT.EDU > Subject: Re: PIC as controller for sinewave inverter - magic sinewaves > Date: den 10 februari 1997 07:09 > > Sten Dahlgren wrote: > > > > For all of you interested in PIC's and power inverter, check out there web > > site > > for the RD001 paper, "Uninterupptible Power Supply Reference Design". > > --- > > Sten Dahlgren CelsiusTech Systems ! "I'd rather have 39 Hp under my arm > > S-175 88 Jaerfaella Sweden ! than one under my backside" > > RemoveMEsedaEraseMEspam_OUTcelsiustech.se +46-8 58084430 ! join your nearest karting club now !! > > Which web site are you refering to ?? Sorry, I was a little bit faster than i should be. It is http://www.microchip.com Note also that it is already in it's b version as i discovered thursday.

1997\02\15@191005 by Dmitry Kiryashov

Scott Dattalo wrote: [some tech details skipped] {Quote hidden}> > The goal of the magic sinewaves is to suppress many > of the harmonics beyond the fundamental. This can be > accomplished by chopping the period into many fine pieces; > in other words, let tau<<T. Mike made a reference to there > being 384 divisions. Now if you strategically place these > narrow pulses among the 384 possible slots, it's possible > to suppress higher harmonics. > > The technique of finding the optimum placement is far > from trivial. However, here are few helpful observations. > > The fundamental frequency is 1/T. > > If the number of subdivisions is N (i.e. tau = T/N) then > 1) Harmonics that evenly divide N/2 can be totally suppressed. > 2) If the n'th harmonic is suppressed then the N-n'th is > also suppressed. > 3) To suppress the 2nd through the n'th harmonic at least > 2^(n-1) pulses are needed. > 4) It is possible to suppress all of the even harmonics. > This is accomplished by an pulse train that is followed > by it inverse. For example, if you had a traing of pulses > like 011000 the even harmonics are suppressed by the > doubled chain: 011000100111. > > To illustrate, suppose you wanted to suppress the 2'nd and > the 3'rd harmonics. The number of needed subdivisions is > N/2 = 2*3 = 6 > or N = 12 > Out of the 12 possible pulses, 2^(3-1) = 4 pulses are needed. > And their locations are: > > 101101000000 > (actually any cyclic rotation of this stream works). > > In addition to suppressing the 2'nd and 3'rd harmonics, the > 10'th and 11'th are also suppressed. The fourth harmonic is > not completely suppressed. If you wanted to suppress the > 4'th or all of the even harmonics for that matter, then > the following pulse stream would do the trick: > > original > <----------> > 101101000000010010111111 > <----------> > inverted from the original > > There is also a 1/n factor that will attenuate those harmonics > that are not suppressed. However for small n, the sin(n*pi*d)/n*pi > is approximately n*pi*d/(n*pi) = d. So the 1/n factor only > becomes important for larger values of n. > > There are many, many more details that I'm ignoring... > Scott While i've read this i've thinking - "May be exist reverse technique ?" For example we have an input digitized signal ( only logical "1" and "0" sequence). And we want to detect some fundamental frequencies in this stream while harmonics should not have influence on result of detection. Are there some way to apply that magic sinewaves technique reversly i.e. not for synthesis but for detection ? I think there are ability in use simultaneosly magic waves and correlation techniques to achieve cutting harmonics in each correlation sum (in DTMF detection for example). I hope that i have not speak some nonsense ;) And second question - what happen in case when we take into account N = odd value, i.e. not 384 but for example 385 or 383 pieces ? WBR Dmitry.

1997\02\17@145839 by sdattalo

Dmitry Kiryashov wrote: > > > While i've read this i've thinking - "May be exist reverse technique ?" > For example we have an input digitized signal ( only logical "1" and "0" > sequence). > And we want to detect some fundamental frequencies in this stream while > harmonics > should not have influence on result of detection. Are there some way to > apply > that magic sinewaves technique reversly i.e. not for synthesis but for > detection ? Yes. But unfortunately there are some practical problems... > And second question - what happen in case when we take into account N = > odd value, > i.e. not 384 but for example 385 or 383 pieces ? Well for starters, it's not theoretically possible to suppress all of the even harmonics if N is an odd value (recall N is the number of subdivisions of the fundamental frequency; ie. if the fundamental frequency is f=1/T, then the time resolution for pulse transitions are limited to the time quanta T/N.) Pushing theory aside however, it is possible to come fairly close to suppressing the even harmonics. And with a little magic, it's possible to suppress a large subset of the lower harmonics. Think about how you would generate a pulse stream that could be low-pass filtered to generate a sine wave. Intuitively, you might expect to have some sort of frequency modulated square wave. When the the frequency is "low" the filter has a large output and when the frequency is "high" the filter has a small output. Your intuition may further lead you to this equation: g(t) = (1 + sign(A1*cos(2*pi*f1*t) + A2*cos(2*pi*f2*t)) )/2 This dual tone signal (which for the context of our current discussion has nothing to do with DTMF) consist of two harmonically related sinusoids: f2 = m*f1 f1 is the frequency of the fundamental waveform that we wish to synthesize (e.g. DTMF generation) or detect (DTMF decoding). f2 is a much high frequency that can be easily filtered with our low pass filter. The sign() function returns the sign of its arguement. Loosely speaking, it's the mathematical function of a comparator. The rest of the equation for g(t) shifts the DC offset so we have a stream of 1's and 0's. There's a simple way to see what this function is doing. Imagine that the lower frequency sinusoid's amplitude, A1, is twice the higher one. When the low frequency cosine is passing through zero, there are many zero crossings in the higher frequency cosine. For every zero crossing, there is a corresponding pulse transition. Now as the lower frequency cosine approaches it's peak, the higher frequency cosine at some point will no longer be able to cause the sum (of the sinuoids) to change signs. Here's another way to see what's happening. If the amplitudes A1 and A2 are equal then the sum can be expressed as a product: sum = A1*cos(2*pi*f1*t) + A2*cos(2*pi*f2*t) = A1*(cos(2*pi*f1*t) + cos(2*pi*f2*t)) = 2 * A1 * cos(pi*(f1+f2)*t) * cos(pi*(f1-f2)*t)) Which is seen to be one cosine wave being modulated by another. Now to address Dmitry's question. So far we've made no assumption about the sampling rate. But it's sort of obvious that the faster we can generate the pulse stream the easier it will be to filter the high frequency components. If it's not obvious, then please take my word for it. I haven't investigated the case for there being an odd number of samples per T1 (=1/f1). I can only speculate that it might induce a subharmonic oscillation. There are two reasons. First, the phases of the cosine waves could be such that every other cycle differs by one extra pulse. The frequency of this extra pulse is 1/2 the frequency of the fundamental (f1). Thus the introduction of the subharmonic. Secondly, if you take the odd number of samples and multiply it by two you get an even number of samples that is twice as long. Using the same theory as above, it's then possible to generate a sine wave of half the frequency of f1. And as we all know from O.J., if it's remotely possible then we can assume it as true. One more point. Think about what happens when the relative amplitudes of the cosines are varied. For example looking at just the extremes: if A1>>A2 then g(t) is a low frequency square wave and the low pass filter's output will have a large output. if A1<<A2 then g(t) is a high frequency square wave and the low pass filter's output will have a small output. The actual output will be somewhere between these two. One more point. Amplitude control of the synthesized sine wave can be traded off for harmonic supression. Scott

'Magic Sine waves'
1997\03\16@021208 by James and Iliana

Me again. I have a lead on a possible commercial application of Magic Sine waves a s I understand them. Does anyone know where I can get information on them ?? I realize that information of this type is not free. Correct me if I'm barking up the wrong tree. A company needs to vary the speed of some AC motors. Toshiba says that thier controller will not work in this situation. Long cable run and high voltage. They said the " Harmonics " would not allow thier controller to work. If I understand Magic Sine Waves, using them can eliminate most of the harmonics thus allowing the control of the motors. The company is paying DEARLY for some controllers now.. Any info is greatly appreciated. ---------------------- James and Iliana Austin Texas Automotive Technician Electronics Hobbyist Software Engineer ----------------------

1997\03\16@051156 by Mike

> I have a lead on a possible commercial application of Magic Sine waves as >I understand them. Does anyone know where I can get information on them ?? >I realize that information of this type is not free. > Correct me if I'm barking up the wrong tree. A company needs to vary the >speed of some AC motors. Toshiba says that thier controller will not work >in this situation. Long cable run and high voltage. They said the " >Harmonics " would not allow thier controller to work. If I understand Magic >Sine Waves, using them can eliminate most of the harmonics thus allowing >the control of the motors. The company is paying DEARLY for some >controllers now.. > Any info is greatly appreciated. Have a look at the addenda, by the way I would be interested in how you progress, I was looking at magic sinewaves recently and also about 2 years ago. But in both cases other jobs came up :( I think its a great idea but, needs a lot of work, in basic theory and practice in implementation etc. Rgds Mike --------------------------- Addenda ---------------------------------- Don Lancaster has written about "magic sinewaves". Don has a homepage at http://www.tinaja.com/ Enter the Reference Library, then the Magic Sinewaves Shelf and find this blurb: Don Lancaster's Magic Sinewave Library "Magic sinewaves" are repeating long sequences of ones and zeros. Digital sinewaves with precisely controlled amplitudes and amazingly low distortions. Compared to traditional PWM, magic sinewaves can offer far higher efficiencies and lower distortions. With circuitry that is elegantly simple and microcontroller friendly. For use in such applications as induction motor speed controls, electric autos, solar panels, power factor correction, inverters, home energy efficiency improvers, and specialized utility services. There are more magic sinewaves than there are particles in the universe. Only recently have specialized analysis tools become available. Don Lancaster, Synergetics, and several Synergetics Partners are in the forefront of this billion dollar opportunity. Extensive seminars, consulting, sourcecode, tutorials, and a new Partners program are now being offered. These files are available for downloading: Magic Sinewave Development Proposal. Formal proposal outlining seminars, partners, and services. A wide range of opportuinities and possibilities are listed, along with full references. Blatant Opportunist #35 MAGSINE.PDF (*) Magic sinewaves are a brand new method of driving ac induction motors, solar panels, inverters, or electric cars. Offer ing outstanding simplicity and efficiency. Introductory Don Lancaster tutorial. Blatant Opportunist #37 EMERGOP4.PDF (*) Fourth annual release of emerging technical opportunities. Push me pull you. The TeraHertz mystery band. PIC microcontollers. Magic Sinewaves. Solitons. DNA computing. Binary chain codes. Bionomial Goes Binary ONESWORD.HTML * A PostScript utility to generate all possible binary words of length "n" having "k" ones in them. Use for magic sinewaves and exploration of the binomial theorm. Requires two way recordable com. An analysis of Magic Sinewave Higher Harmonics. Just where do those square corners come from on a low harmonic waveform? Jim Fitzsimons analyzes the first 999 (!) harmonics of five typical 384 bit non-delta magic sinewaves. Demo sampler of 288-Bit Magic Sinewaves. A detailed analysis pf a few of the 288-bit "delta Friendly" magic sinewaves. There are 49 amplitudes in the complete 288 sequence, having distortions as low as 0.01 percent. Tech Musings #101 MUSE101.PDF ( June 1996 ) Using Delta-Wye transforms, Professional audio resources, Acrobat amber beta software, Building your own Tesla coil, Three phase magic sinewaves, more. Tech Musings #93 MUSE93.PDF ( October 1995 )* More on magic sinewaves, stepper motor driver chips, yet another patent horror story, mid range Mench computer, a collection of PCMCIA plug-in card resources. Tech Musings #90 MUSE90.PDF ( July 1995)* Fourier Series analysis, alternatives to plated through holes, classic computer resources, a $200 GPS receiver, Newtek's new video toaster for Windows. Hardware Hacker #87 HACK87.PDF (April 1995 )* AC motor drive fundamentals. New Basic Stamps from Parallax. PIC Microcomputer resources. Additional details on forced zero magic sinewaves. Dealing with "too good to be true" results. Hardware Hacker #86 HACK86.PDF (March 1995 )* The DNA computer language. Some wavelet book resources. Magic digital sinewave codes. Engineering economics review. Hot new fringe FM RBDS tuner.

'Another Magic Sine Wave question'
1997\03\17@011513 by James and Iliana

Hello All. Is there anyone that has used Magic Sine Waves in an application ?? I've looked at all the data from Don's web site and it sounds like, as he would call it, " Pseudo Science ". ( maybe because I don't know enough to tell what is smoke or not ). Before I make a fool out of myself ( and spend money that I don't have ) could someone just confirm that they are not smoke and mirrors ?? Thanks a bunch for your time. ---------------------- James and Iliana Austin Texas Automotive Technician Electronics Hobbyist Software Engineer ----------------------

'Magic Sine waves'
1997\03\17@011514 by James and Iliana

At 06:01 PM 3/16/97 +0800, you wrote: > >Have a look at the addenda, by the way I would be interested in how you >progress, >I was looking at magic sinewaves recently and also about 2 years ago. But in >both >cases other jobs came up :( > >I think its a great idea but, needs a lot of work, in basic theory and >practice in >implementation etc. > >Rgds > >Mike > Hi Mike, I read all the data on the Magic Sine Waves that I could find on his site.. still confused. I understand the idea. Make your own sine waves based upon digital signals that are not regular so you eliminate harmonics. Is there some sort of formula available ?? Is that something that I would have to pay for ?? I'll send Don some email and see just where he stands and what is available. Thanks for the reply to my post. ---------------------- James and Iliana Austin Texas Automotive Technician Electronics Hobbyist Software Engineer ----------------------

1997\03\17@034240 by Mike

At 12:10 AM 17/03/97 -0600, you wrote: {Quote hidden}>At 06:01 PM 3/16/97 +0800, you wrote: > >> >>Have a look at the addenda, by the way I would be interested in how you >>progress, >>I was looking at magic sinewaves recently and also about 2 years ago. But in >>both >>cases other jobs came up :( >> >>I think its a great idea but, needs a lot of work, in basic theory and >>practice in >>implementation etc. >> >>Rgds >> >>Mike >> > >Hi Mike, > I read all the data on the Magic Sine Waves that I could find on his >site.. still confused. I understand the idea. Make your own sine waves >based upon digital signals that are not regular so you eliminate harmonics. >Is there some sort of formula available ?? Yes - his formula is written in postscript language. Is that something that I would >have to pay for ?? I'll send Don some email and see just where he stands >and what is available. Good luck - I tried about 3 times over last 6 months... Rgds Mike Some say there is no magic but, all things begin with thought then it becomes academic, then some poor slob works out a practical way to implement all that theory, this is called Engineering - for most people another form of magic. Massen

'Another Magic Sine Wave question'
1997\03\17@053444 by Mike

At 12:00 AM 17/03/97 -0600, you wrote: >Hello All. > Is there anyone that has used Magic Sine Waves in an application ?? I think that was Don's intention - to raise the capital to produce fully functioning units as part of a business plan - perhaps with the eventual aim of floating the company. I've >looked at all the data from Don's web site and it sounds like, as he would >call it, " Pseudo Science ". ( maybe because I don't know enough to tell >what is smoke or not ). It seems very reasonable and real. I see that minor variations in a regular PWM approach can give improved results:- the 'magic sinewaves' is just a logical extension of this - well logical in the sense that it is more focused on producing the bit stream directly from some formula instead of via some iterative method - which is likely to be very time consuming indeed ! > Before I make a fool out of myself ( and spend money that I don't have ) >could someone just confirm that they are not smoke and mirrors ?? I think is worth while pursuing - but then you would be a fool if you spent money you don't have, then that would be a debt you'd be lumbered with ;) Anyway I think its a worthwhile gamble at least for basic research purposes. > Thanks a bunch for your time. And you get the electrons free :) Rgds Mike Some say there is no magic but, all things begin with thought then it becomes academic, then some poor slob works out a practical way to implement all that theory, this is called Engineering - for most people another form of magic. Massen

1997\03\17@112509 by sdattalo

James and Iliana wrote: > > Before I make a fool out of myself ( and spend money that I don't have ) > could someone just confirm that they are not smoke and mirrors ?? The "they" being Magic Sinewaves... The only way you'll make a fool out of yourself is by claiming that this stuff is Magic. It's not. I've studied various pulse streams: 12,15,24,36 and was able to demonstrate the harmonic suppression claimed by Lancaster. I was unable to come up with a generic formula. However, I have made several observations and have written a program in MATLAB to calculate the harmonic content of a given pulse stream. I can re-post the observations and send a copy of the MATLAB program to those interested. Scott

'Magic Sine waves'
1997\03\17@113520 by 71177.202

James and Iliana wrote: > > Me again. > I have a lead on a possible commercial application of Magic Sine waves as > I understand them. Does anyone know where I can get information on them ?? Try Don Lancaster at http://www.tinaja.com.

1997\03\22@221442 by Michael S. Hagberg

i recall someone asking about magic sine waves, and ran accross an article in June 1995, Circuit Cellar. check it out. michael

'Programmable Sinewave Generator'
1997\04\07@192538 by kotay

I remember some discussion on this list a while back about sine wave generators, so when I saw this info in Electronic Component News I thought some list members would be interested: ML2037 DC-500kHz 65536 steps Frequency programmed via 3-wire SPI interface Output gain digitally programmed via two external pins Requires no external components Manufactured by Micro Linear Corp. 408-433-5200 http://www.microlinear.com Keith

'Magic Sinewaves'
1997\07\21@143505 by Miller, Steve

Howdy, Has anyone used Don Lancaster's magic sinewaves with a PIC? These binary strings are supposed to generate sinewaves with very little harmonics. How does this technique compare to using an eight bit port and a resistor ladder arrangement? (Obvious advantage of the magic sinewave is that it only requires one or two output pins.) ( Link to magic sinewave is http://www.tinaja.com/ ) Another question : How do I evaluate the harmonic content of a particular scheme? I have the student edition of MATLAB and MATHCAD. (These are not the current versions, as my student days was a while ago). Does anyone have a sample file on how to get MATLAB to calculate this? Thanks. ------ Steve

1997\07\21@150410 by sdattalo

Miller, Steve wrote: > Another question : How do I evaluate the harmonic content of a particular > scheme? I have the student edition of MATLAB and MATHCAD. (These are not the > current versions, as my student days was a while ago). Does anyone have a > sample file on how to get MATLAB to calculate this? Steve, Here is the MATLAB program: % sqseries.m % % The purpose of this program is to draw the frequency % components of a pulse train % T. Scott Dattalo, 14FEB97 clear; clg; T = 1; N = 1000; %Number of samples t = [1:N]/N * T; %Time vector % 0 4 8 c 0 4 8 c 0 4 8 c bitstring = '00000001001101111110110010000000'; [m,P] = size(bitstring); % Number of pulses in the period of T bits = zeros(1:2*P); % double the length for m=1:P % First half is non-inverted bits(m) = abs(bitstring(m)) - abs('0'); end for m=P+1:2*P % The second half is inverted bits(m) = 1-(abs(bitstring(m-P)) - abs('0')); end P = 2*P; phase = [0:(P-1)] .* bits * T/P; harmonics = 50; mag = zeros(1:harmonics); ph = zeros(1:harmonics); tau = T/P; d = tau/T; sqwave = zeros(1:N); for n=1:harmonics w = n*2*pi/T; for m=1:P if bits(m) ~= 0 sqwave = sqwave + 2*sin(n*pi*d)/(n*pi)*cos(w*(t-tau/2-phase(m))); mag(n) = mag(n) + 2*sin(n*pi*d)/(n*pi)*cos(w*phase(m)); ph(n) = ph(n) + cos(w*phase(m)); end end end ti = sprintf('SQUARE WAVE'); title(ti) subplot(211) plot(t,sqwave) subplot(212) plot([1:(harmonics)], mag,'*') This program will calculate the first 50 harmonics of a repeating pulse stream. It does this by adding the harmonics of the individual pulses that build the pulse stream. If you have an arbitrary pulse stream: ^ v(t) | tau | <----> | A| +----+ +----+ | | | | | -+-----+ +------------------------+ +---------> t phi <-----> <-------------- T ------------> The harmonics of it can be found from: infinity ---- sin(n*pi*d) / 2*pi / tau \\ v(t) = d + \ ------------- * cos|n* ---- * | t - --- - phi || /___ n*pi \ T \ 2 // n=1 If you want to really know the theory then check out the gory details at: http://www.interstice.com/~sdattalo/technical/theory/sqwave.html Scott -- "Life in two dimensions is a mass production scheme" Neil Peart

'Dual Sine Waves'
1997\10\31@042844 by Virtual

Hi, this question is concerning DTMF generation again. I am not very experienced with the PIC's and the code that is already out there just confuses me. Since a DTMF tone is just the two tones mixed, I would like to know if its possible for me to make a small program that creates a sine wave on 2 pins, and then just mix the two together. I was unable to find any info on the web about generating sine waves, and how i could figure out what freq they would be at, also how I would make two at the same time, one for the lower portion, and one for the upper portion, each outputting to a seperate pin. I realize this isn't the best way, but I think it will help me to learn a little more, jumping to PWM assembly code just has my mind going in circles :) p.s. I use the 16C84 running off a color burst crystal (about 3.579275 MHZ) Thanks a lot John

'DC level to 1KHz sine wave conversion.'
1998\02\12@124855 by GERRY COX

My latest project involves a PIC16C73A , which amongst other things, drives a serial 16bit DAC. This gives me a control voltage programmable from 0 to 10V. I also need to generate a sinewave of nominally 1KHz. The p-p amplitude must track the DAC voltage and be always 10 percent of its DC voltage. It has to be a reasonably clean and stable sinewave. Input voltage range varies from 0.5 to 10V. Sine wave output will vary 0 to 1V p-p for the above input range. Accuracy better than 3%. Sine wave Total Harmonic Distortion better than 5% Sine wave frequency accuracy 1Khz +/- 100Hz Output current 5mA max. Power supply - I have +15V, -15V and +5V available. For example if I have a DC input of 3V the sinewave generator should output 300mV peak to peak at 1Khz. this 300mV should be accurate to 309mV max or 291 mV minimum. I am considering an MF10 switched capacitor filter with clock 128 times that of the 1KHZ output but am looking for ideas All ideas welcome. Must be reasonably low component count but parts cost is not paramount for a change! Regards, Gerry Cox

1998\02\12@131007 by Peter van Hoof

did you look into magic sinewaves? http://www.tinaja.com/magsn01.html seems pretty interesting to me Peter ---------- > From: GERRY COX <@spam@gcoxRemoveMEEraseMEDEK.COM> > To: EraseMEPICLIST@spam@MITVMA.MIT.EDU > Subject: DC level to 1KHz sine wave conversion. > Date: Thursday, February 12, 1998 12:20 PM > > My latest project involves a PIC16C73A , which amongst other things, drives > a serial 16bit DAC. This gives me a control voltage programmable from 0 to > 10V. > I also need to generate a sinewave of nominally 1KHz. The p-p amplitude > must track the DAC voltage and be always 10 percent of its DC voltage. > It has to be a reasonably clean and stable sinewave. > Input voltage range varies from 0.5 to 10V. > Sine wave output will vary 0 to 1V p-p for the above input range. Accuracy > better than 3%. > Sine wave Total Harmonic Distortion better than 5% > Sine wave frequency accuracy 1Khz +/- 100Hz > Output current 5mA max. > Power supply - I have +15V, -15V and +5V available. > > For example if I have a DC input of 3V the sinewave generator should output > 300mV peak to peak at 1Khz. > this 300mV should be accurate to 309mV max or 291 mV minimum. > > I am considering an MF10 switched capacitor filter with clock 128 times that > of the 1KHZ output but am looking for ideas > > All ideas welcome. Must be reasonably low component count but parts cost is > not paramount for a change! > > Regards, > Gerry Cox

1998\02\12@132225 by John Payson

> My latest project involves a PIC16C73A , which amongst other things, drives > a serial 16bit DAC. This gives me a control voltage programmable from 0 to > 10V. > I also need to generate a sinewave of nominally 1KHz. The p-p amplitude > must track the DAC voltage and be always 10 percent of its DC voltage. Do you need the control voltage for anything else, or are you just looking to have a finely-adjusted sinewave amplitude? If you're trying to produce an amplitude-adjusted sine wave, one approach you might consider would be to generate two 1KHz sine waves in different phases (the phase difference will set the net amplitude). I don't know exactly how much precision you could hammer out of a PIC that way, but it should be possible to do pretty well (I use that trick in my music box, btw). Alternatively, since your sine wave frequency is fixed, you could probably use a simple gating circuit to convert your DC signal into a square wave, and then feed that through a 4 or 5 stage filter. Since the frequency is fixed, you could optimize the RC's so as to give the best possible 1-1.5KHz cutoff and if you use an active feedback filter you could even add some "Q" to make your sinewaves really gorgeous.

1998\02\12@135532 by peter

GERRY COX wrote: > > My latest project involves a PIC16C73A , which amongst other things, drives > a serial 16bit DAC. This gives me a control voltage programmable from 0 to > 10V. > I also need to generate a sinewave of nominally 1KHz. The p-p amplitude > must track the DAC voltage and be always 10 percent of its DC voltage. > It has to be a reasonably clean and stable sinewave. > Input voltage range varies from 0.5 to 10V. > Sine wave output will vary 0 to 1V p-p for the above input range. Accuracy > better than 3%. > Sine wave Total Harmonic Distortion better than 5% > Sine wave frequency accuracy 1Khz +/- 100Hz Use an LM13700 it's a dual transconductance amp and tailor made for your app both as the osc and the attenuator -- Peter Cousens email: @spam@peterspam_OUT.....cousens.her.forthnet.gr phone: + 3081 324450, 380534 snailmail: Folia, Agia Fotini, Karteros, Heraklion Crete, Greece.

1998\02\12@142236 by Mike Keitz

On Thu, 12 Feb 1998 17:20:29 -0000 GERRY COX <spamBeGonegcoxEraseMEDEK.COM> writes: >My latest project involves a PIC16C73A , which amongst other things, >drives >a serial 16bit DAC. This gives me a control voltage programmable from >0 to >10V. >I also need to generate a sinewave of nominally 1KHz. The p-p >amplitude > must track the DAC voltage and be always 10 percent of its DC >voltage. >It has to be a reasonably clean and stable sinewave. >Input voltage range varies from 0.5 to 10V. >Sine wave output will vary 0 to 1V p-p for the above input range. It would go from 0.05 to 1V. if it were linearly controlled. >Accuracy >better than 3%. Why are you using a 16-bit converter then? A 3% requirement is less than 6 bits of accuracy. The first thing that comes to mind is to use an 8-bit DAC chip in multiply mode. By driving a constant level sinewave into the reference pin, the output will be a sine wave wiht level varying according to the digital code input. Unfortunately most of the DACs only multiply in 2-quadrants (positive voltages), so a DC bias needs to be applied to the input, causing the DC level at the output to also vary according to the input code. Another DAC could be operated to ouptut only the DC level expected at the output, then (assuming the 2 DACs are balanced) the two subtracted to leave a sine wave centered around 0 volts. Another possibility is an analog multiplier. This is the obvious choice, though multiplier chips are somewhat expensive (if you already have a 16-bit DAC, this may not be much of an issue). And you have to come up with a constant-amplitude sine wave from somewhere to begin with. One thing that you didn't specify is the time to reach a new amplitude setting after a change is demanded. If this is rather slow (100s of ms), then another realm of possibilities opens up, using feedback. A rectifier or amplitude detector would sample the output of the circuit, and this would be compared to the desired level. The error voltage would control some other part of the circuit to set the amplitude. If the long response time and otherwise poor dynamics are acceptable (the level isn't changed very often, in other words), I would highly recommend this method. The use of feedback makes the amplitude accuracy dependent only on the sampling detector, reference level source, and error amplifier. Variations in other parts of the circuit would automatically cancel out. >Sine wave Total Harmonic Distortion better than 5% >Sine wave frequency accuracy 1Khz +/- 100Hz Since the frequency only has to be within 10%, a self-oscillating circuit should work. Use a filter of relatively high "Q" with feedback from the output back to the input to make it oscillate. In the feedback path, put an adjustable limiter (a diode clamp circuit). I think it would work to clamp only one polarity of the signal. The input to the filter would be almost a square wave but it would filter to a sine wave. The clamping level could be controlled directly (open loop) or through feedback (closed loop, Automatic Level Control). A slightly more or maybe less complicated variation on this would be to generate a variable-level square wave and filter it to a sine wave. As with the oscillator circuit, the filter could be either an RC network or a switched-capacitor filter like an MF10. >Output current 5mA max. >Power supply - I have +15V, -15V and +5V available. This makes it easy. Standard op-amps will meet the current specification. >I am considering an MF10 switched capacitor filter with clock 128 >times that >of the 1KHZ output but am looking for ideas Yes, this is a good start. It could serve as the 1 KHz sinewave generator for the multiplier ideas, or filter or oscillate with a varying square-wave input level. _____________________________________________________________________ You don't need to buy Internet access to use free Internet e-mail. Get completely free e-mail from Juno at http://www.juno.com Or call Juno at (800) 654-JUNO [654-5866]

1998\02\12@155810 by Roger Anderson

I've done similar projects in two different ways. {Quote hidden}> 1. Use an analog multiplier like the AD633 and wire it so > that it multiplies X times Y. Feed your sine wave at 1 Volt p-p > into the X input > and your DC voltage into the Y input and the output will do what you > wish. > > 2. Use a DAC like the MAX530 and wire it for Four Quadrant > Mutiplication. > (look in the data sheet for the circuit ) > Feed your appropriately scaled Sine wave into the the DAC's Vrefin > and the > amplitude will be controlled by the dacs digital code setting. > > Roger > > ---------- > > From: GERRY COX <gcoxspamBeGoneDEK.COM> > > To: RemoveMEPICLIST@spam@spamBeGoneMITVMA.MIT.EDU > > Subject: DC level to 1KHz sine wave conversion. > > Date: Thursday, February 12, 1998 12:20 PM > > > > My latest project involves a PIC16C73A , which amongst other things, > drives > > a serial 16bit DAC. This gives me a control voltage programmable > from 0 > to > > 10V. > > I also need to generate a sinewave of nominally 1KHz. The p-p > amplitude > > must track the DAC voltage and be always 10 percent of its DC > voltage. > > It has to be a reasonably clean and stable sinewave. > > Input voltage range varies from 0.5 to 10V. > > Sine wave output will vary 0 to 1V p-p for the above input range. > Accuracy > > better than 3%. > > Sine wave Total Harmonic Distortion better than 5% > > Sine wave frequency accuracy 1Khz +/- 100Hz > > Output current 5mA max. > > Power supply - I have +15V, -15V and +5V available. > > > > For example if I have a DC input of 3V the sinewave generator should > output > > 300mV peak to peak at 1Khz. > > this 300mV should be accurate to 309mV max or 291 mV minimum. > > > > I am considering an MF10 switched capacitor filter with clock 128 > times > that > > of the 1KHZ output but am looking for ideas > > > > All ideas welcome. Must be reasonably low component count but parts > cost > is > > not paramount for a change! > > > > Regards, > > Gerry Cox

1998\02\13@082348 by paulb

GERRY COX wrote: > I also need to generate a sinewave of nominally 1KHz. The p-p > amplitude must track the DAC voltage and be always 10 percent of its > DC voltage. It has to be a reasonably clean and stable sinewave. I'd be very tempted to just use an 8038CC function generator chip. Or anything that has subsequently rendered it obsolete. I'm not sure of the linearity or voltage range, but an XR2206 may do. Cheers, Paul B.

1998\02\13@124029 by Wayne Foletta

Gerry: Try one of Motorola's or Analog Devices analog multipliers like the AD633. For a oscillator and attenuator try National's LM13700. Try a multipier DAC like Maxim MAX505 and others. If you want a software solution see http://www.tinaja.com/magsn01.html for what they call "magic sinewaves" - if you want more details on this contact me. I was part of the IC design team that produced the first all digital DTMF chip generators (Comdial Telephone Systems) Patents now expired and the technology is "as old as the hills" so I can explain and give details if there is interest. - Wayne Foletta BMI - Saratoga, CA {Quote hidden}> ---------- > From: GERRY COX[SMTP:.....gcox@spam@EraseMEDEK.COM] > Reply To: pic microcontroller discussion list > Sent: Thursday, February 12, 1998 9:20 AM > To: .....PICLISTRemoveMEMITVMA.MIT.EDU > Subject: DC level to 1KHz sine wave conversion. > > My latest project involves a PIC16C73A , which amongst other things, > drives > a serial 16bit DAC. This gives me a control voltage programmable from > 0 to > 10V. > I also need to generate a sinewave of nominally 1KHz. The p-p > amplitude > must track the DAC voltage and be always 10 percent of its DC > voltage. > It has to be a reasonably clean and stable sinewave. > Input voltage range varies from 0.5 to 10V. > Sine wave output will vary 0 to 1V p-p for the above input range. > Accuracy > better than 3%. > Sine wave Total Harmonic Distortion better than 5% > Sine wave frequency accuracy 1Khz +/- 100Hz > Output current 5mA max. > Power supply - I have +15V, -15V and +5V available. > > For example if I have a DC input of 3V the sinewave generator should > output > 300mV peak to peak at 1Khz. > this 300mV should be accurate to 309mV max or 291 mV minimum. > > I am considering an MF10 switched capacitor filter with clock 128 > times that > of the 1KHZ output but am looking for ideas > > All ideas welcome. Must be reasonably low component count but parts > cost is > not paramount for a change! > > Regards, > Gerry Cox >

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1998\04\06@080525 by sajjad.akhtar

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'Sinewave generation'
1998\12\12@075836 by Darren Logan

Hi Pic'ers, Anyone know how to generate a sinewave using PWM on say a PIC 12Cxx ? I want to generate 100Hz (plus or minus 0.1Hz). Any code will do, i.e. BASIC, C or assembly. Thanks in advance. Darren

1998\12\12@103045 by Chris Eddy

Do you need to have a clean high accuracy sine wave? You might try building a simple op amp oscillator, and coupling the PWM signal in as a sync signal. Then you get the benefits of a clean wave with a clock accuracy. Simply low passing the PWM will surely mash the signal, by the time you get all of the harmonics out, then you have to adjust it for height afterwards. Chris Eddy, PE Pioneer Microsystems, Inc. Darren Logan wrote: {Quote hidden}> Hi Pic'ers, > > Anyone know how to generate a sinewave using PWM on say a PIC 12Cxx ? > > I want to generate 100Hz (plus or minus 0.1Hz). > > Any code will do, i.e. BASIC, C or assembly. > > Thanks in advance. > > Darren

1998\12\12@103636 by Quentin

Read AN 655, It will explain it to you. Best is to use a R-2R ladder with 4 to 8 pins, depending on your aplication. Basically what you do is to create a lookup table with your sine steps (0 to 255 or 0 to 16) in it. Then your programs puts these values in sequence on your output pins. I recently wrote a small test program that does a 4 bit sine for tone generation and it worked great. Pity I lost it in a hard drive crash otherwise I would have posted it to you. Quentin Darren Logan wrote: {Quote hidden}> Hi Pic'ers, > > Anyone know how to generate a sinewave using PWM on say a PIC 12Cxx ? > > I want to generate 100Hz (plus or minus 0.1Hz). > > Any code will do, i.e. BASIC, C or assembly. > > Thanks in advance. > > Darren

1998\12\12@191024 by Mike Keitz

On Sat, 12 Dec 1998 17:35:43 +0200 Quentin <spamBeGoneqscKILLspam@spam@ICON.CO.ZA> writes: >Read AN 655, It will explain it to you. Best is to use a R-2R ladder >with >4 to 8 pins, depending on your aplication. Basically what you do is to >create a lookup table with your sine steps (0 to 255 or 0 to 16) in >it. >Then your programs puts these values in sequence on your output pins. There are lots of ways to make sine waves, especially at 100 Hz. A resistor DAC works OK. It is better to use a nonlinear resistor network that is optimized for sine waves. That trick dates back to very early days of digital logic. Five resistors in the ratio 1.5, 3.9, 4.7, 6.8, and 20 makes a nice 16-sample approximate sine wave. They are also standard value resistors. One end of each rresistor goes to a PIC pin which is always an output, the other end of all 5 resistors goes to a common node which is the sine output. The digital sequence applied tot he pins is pretty much what a shift register would make (early implementations did of course use a shift register, but needed a few more resistors.) You end up with a "staircase" looking wave that still needs a little analog filtering before you really can call it a sine. Often just a single capacitor to ground will be sufficient. In the most recent generation of my CTCSS encoders (that is what the guy who wants a 100 Hz sine wave needs it for, right?), I've used "magic sine" techniques described by Don Lancaster (http://www.tinaja.com). Only one PIC pin is required, it is driven either on or tri-state for each sample period. During half of the sine wave, the pin drives high, during the other half it drives low. The sampling rate is quite high, hundreds of samples per output cycle. The sequence of whether to output a pulse for each sample is stored in a table. The table can represent as little as 1/4 of the full wave's samples if you re-use it properly for each quadrant. The data in the table is of course one bit for each sample. The density of "1" bits increases and decreases in a sinusoidal manner. Lancaster's site has many suitable sequences as well as methods to use to derive new ones. An R-C integrator will clean up the pulses to a decent sine wave. Using two integrators in series makes a really nice sine wave. In order to integrate properly, the integrator neds to be set up so the output amplitude is much less than VDD (this causes the voltage across the resistor to be relativley constant during each pulse). An amplifier may be needed if a level of more than 0.5Vpp or so is required. But, the same sine amplitude can be maintained over a wide range of frequencies using the same filter if the output pulses are a constant width (i.e. reduce the frequnecy by increasing a "dead" off-time after each pulse. Then the energy coupled to the integrator per cycle is the same regardless of the frequency). The downside of the "magic sine" is that with 210 or more samples per cycle it takes a lot more PIC time than a DAC based approach. I haven't tried anything with PWM. I think software PWM would need even more CPU than magic sine, but on a PIC with PWM hardware it could work well. Someday I'll put all this on a web page. For right now I'll just tease you. ___________________________________________________________________ You don't need to buy Internet access to use free Internet e-mail. Get completely free e-mail from Juno at http://www.juno.com/getjuno.html or call Juno at (800) 654-JUNO [654-5866]

'Sinewave generation - The way to do it with a PIC.'
1998\12\14@130528 by John Bellini

Suppose you have a PWM available (hardware or software). You can generate a varing duty cycle PWM wave form that cycles from something like 1% to 50%+ and then back down. You filter this waveform with a simple RC filter (2 parts) and you get a sine wave. You can level shift it if you need to. You can use a constant ROM look up table, different tables can give you different output voltage references. Run the PWM on an interrupt. Change the interrupt time to change the frequency. {Original Message removed}

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'sine wave generation (again? :)'
1999\02\21@182057 by Gerhard Fiedler

i need to produce an approx. sine wave (i guess a few % THD is fine, i'm not sure about this requirement), but with a rather precise frequency in the range from 300Hz to 2500Hz, and with a resolution of 0.5%. i'd like to be able to switch it on (ie. be in sync) within at max. 50ms. i'm considering the following options: - vco (like the exar2206 -- ? not sure about the number) with a pll and a simple square from the pic to sync it. this is easiest in firmware, but i've never used a pll and i'm not sure what difficulties i might have to expect in terms of stability. - vco fed by the pic with a pwm signal (i probably will have a hardware pwm in the pic), and getting the vco output back to the pic into a control loop, which outputs to the pwm into the vco. - a resistor adder approach with maybe five points or so. at 5 points and 2500Hz, that's 20microsec per step; for a resolution of 0.5% i need a step increment of 100ns, which i guess is not possible with a pic. but i probably can "cheat" and use step lengths which might differ by 1, which would give me the necessary overall resolution for the frequency at 20MHz or maybe even 10MHz clock. - just thought of a variant of the above: a programmable (digital) oscillator and a shift register (with the resistors at the parallel outputs). needs more points (since i don't have three output states as with the pic) or some logic to create positive/negative outputs, but maybe is easier. the oscillator would have to bring 4..40kHz with the 0.5% resolution. the magic sine approach or output the (stepped) sine directly from the pic's hardware pwm seem not to be viable because of the frequency resolution. if somebody has any comments to/experiences with the above options, or knows of a digitally controlled oscillator with something resembling a sine at the output, that would be great. thanks, ge

1999\02\21@191051 by Scott Dattalo

On Sun, 21 Feb 1999, Gerhard Fiedler wrote: {Quote hidden}> > i need to produce an approx. sine wave (i guess a few % THD is fine, i'm > not sure about this requirement), but with a rather precise frequency in > the range from 300Hz to 2500Hz, and with a resolution of 0.5%. i'd like to > be able to switch it on (ie. be in sync) within at max. 50ms. > > i'm considering the following options: > > - vco (like the exar2206 -- ? not sure about the number) with a pll and a > simple square from the pic to sync it. this is easiest in firmware, but > i've never used a pll and i'm not sure what difficulties i might have to > expect in terms of stability. > > - vco fed by the pic with a pwm signal (i probably will have a hardware pwm > in the pic), and getting the vco output back to the pic into a control > loop, which outputs to the pwm into the vco. > > - a resistor adder approach with maybe five points or so. at 5 points and > 2500Hz, that's 20microsec per step; for a resolution of 0.5% i need a step > increment of 100ns, which i guess is not possible with a pic. but i > probably can "cheat" and use step lengths which might differ by 1, which > would give me the necessary overall resolution for the frequency at 20MHz > or maybe even 10MHz clock. > > - just thought of a variant of the above: a programmable (digital) > oscillator and a shift register (with the resistors at the parallel > outputs). needs more points (since i don't have three output states as with > the pic) or some logic to create positive/negative outputs, but maybe is > easier. the oscillator would have to bring 4..40kHz with the 0.5% resolution. > > the magic sine approach or output the (stepped) sine directly from the > pic's hardware pwm seem not to be viable because of the frequency resolution. > > if somebody has any comments to/experiences with the above options, or > knows of a digitally controlled oscillator with something resembling a sine > at the output, that would be great. > Check out http://www.interstice.com/~sdattalo/technical/software/pic/picsine.html http://www.brouhaha.com/~eric/pic/sine.html For mine and Eric Smith's pic sine generators. Eric's routine is faster, mine has more frequency resolution (i.e. phase accumulators). The phase accumulators allow you to create sine waves whose frequencies are not limited to an integer multiple of PIC instruction cycles. I suspect that this coupled with the hardware pwm will provide a quite decent sine wave (with the appropriate low-pass filter). If you want to move the solution into a smaller part, then check out this non-interrupt driven, single-cycle-resolution, 256 level, all software PWM routine: http://www.interstice.com/~sdattalo/technical/software/pic/pwm256.txt Scott

1999\02\22@044007 by Gerhard Fiedler

At 16:09 02/21/99 -0800, Scott Dattalo wrote: >On Sun, 21 Feb 1999, Gerhard Fiedler wrote: >> i need to produce an approx. sine wave (i guess a few % THD is fine, i'm >> not sure about this requirement), but with a rather precise frequency in >> the range from 300Hz to 2500Hz, and with a resolution of 0.5%. i'd like to >> be able to switch it on (ie. be in sync) within at max. 50ms. > >Check out > >http://www.interstice.com/~sdattalo/technical/software/pic/picsine.html thanks. i've seen that before when i did some research, but i didn't understand what i would do with a sine function in this application, but now i guess i got it :) >(with the appropriate low-pass filter). If you want to move the solution >into a smaller part, then check out this non-interrupt driven, >single-cycle-resolution, 256 level, all software PWM routine: that's an intriguing idea: a reduced i2c slave and the sine generator in an 8pinner... but for now i guess i leave it in the bigger one as a software module, with the hardware pwm. thanks again, ge

1999\02\22@162618 by Gerhard Fiedler

At 01:37 02/22/99 -0800, Gerhard Fiedler wrote: >At 16:09 02/21/99 -0800, Scott Dattalo wrote: >>On Sun, 21 Feb 1999, Gerhard Fiedler wrote: >>> i need to produce an approx. sine wave (i guess a few % THD is fine, i'm >>> not sure about this requirement), but with a rather precise frequency in >>> the range from 300Hz to 2500Hz, and with a resolution of 0.5%. i'd like to >>> be able to switch it on (ie. be in sync) within at max. 50ms. >> >>Check out >> >>http://www.interstice.com/~sdattalo/technical/software/pic/picsine.html > >thanks. i've seen that before when i did some research, but i didn't >understand what i would do with a sine function in this application, but >now i guess i got it :) just in case anybody made/is going to make the same mistake i made, here's how to get a pretty good frequency resolution for a sine output. the resistor-output approach (and also the magic sine waves) works (and my head worked :) with fixed parts (angles) of a sine wave. therefore you have to work with sample rates which are integer multiples of the desired frequency. using an approach that outputs arbitrary voltages (as opposed to fixed steps with the resistor ladder approach), you're not longer stuck with integer multiples of the frequency, since the sample points don't have to be at the same angles in each period. so my solution looks like this (the terms in [] after a variable gives the unit and scale for it, as an example for the scales of the different values): 20MHz pic w/ 8 bit hardware pwm: f_sample=40kHz, t_sample=25us (this gives me at least 16 samples per period, for frequencies up to 2500Hz) i calculate all times with a resolution of 0.5us, which gives me a frequency resolution of 3Hz at 2500Hz (and 0.1Hz at 300Hz) t_sine [0.5us] = 2*10^7 / f_sine [0.1Hz] the actual time is calculated also in [0.5us] resolution and reduced into the interval 0..t_sine: t_(n) [0.5us] = t_(n-1) + t_sample if ( t_(n) >= t_sine ) t_(n) = t_(n) - t_sine then i use this time and the frequency to calculate the angle of the next sample for the sine function (which has to be scaled to fit the input range of the used sine function, which here is 14 bit for 2*pi, ie. 16384): phi [2*pi/16384] = ( t_(n) [0.5us] * (f_sine [0.1Hz] * scale) ) / (2^16) with scale = (0.5us * 0.1Hz * 16384) * 2^16 = 53.69 output = sin ( phi ) this output value gives me, appropriately scaled (eg. scott's sine function returns -127..+127, so i'd have to add 128 to get an 8bit pwm range of 0..255), the pwm on-time for the next sample period. at 20MHz, there's 128 instruction cycles between samples, which seems to be enough, considering that scott's function needs only 54 IIRC. the rest is a 16bit add, a 16bit subtract, a 16x16bit multiplication and an 8bit add plus some "glue" -- not too heavy. ge

1999\02\22@214835 by Scott Dattalo

On Mon, 22 Feb 1999, Gerhard Fiedler wrote: {Quote hidden}> just in case anybody made/is going to make the same mistake i made, here's > how to get a pretty good frequency resolution for a sine output. the > resistor-output approach (and also the magic sine waves) works (and my head > worked :) with fixed parts (angles) of a sine wave. therefore you have to > work with sample rates which are integer multiples of the desired > frequency. using an approach that outputs arbitrary voltages (as opposed to > fixed steps with the resistor ladder approach), you're not longer stuck > with integer multiples of the frequency, since the sample points don't have > to be at the same angles in each period. > > so my solution looks like this (the terms in [] after a variable gives the > unit and scale for it, as an example for the scales of the different values): > > 20MHz pic w/ 8 bit hardware pwm: f_sample=40kHz, t_sample=25us (this gives > me at least 16 samples per period, for frequencies up to 2500Hz) > > i calculate all times with a resolution of 0.5us, which gives me a > frequency resolution of 3Hz at 2500Hz (and 0.1Hz at 300Hz) > > t_sine [0.5us] = 2*10^7 / f_sine [0.1Hz] > > the actual time is calculated also in [0.5us] resolution and reduced into > the interval 0..t_sine: > > t_(n) [0.5us] = t_(n-1) + t_sample > if ( t_(n) >= t_sine ) > t_(n) = t_(n) - t_sine > > then i use this time and the frequency to calculate the angle of the next > sample for the sine function (which has to be scaled to fit the input range > of the used sine function, which here is 14 bit for 2*pi, ie. 16384): > > phi [2*pi/16384] = ( t_(n) [0.5us] * (f_sine [0.1Hz] * scale) ) / (2^16) > with scale = (0.5us * 0.1Hz * 16384) * 2^16 = 53.69 > > output = sin ( phi ) > > this output value gives me, appropriately scaled (eg. scott's sine function > returns -127..+127, so i'd have to add 128 to get an 8bit pwm range of > 0..255), the pwm on-time for the next sample period. at 20MHz, there's 128 > instruction cycles between samples, which seems to be enough, considering > that scott's function needs only 54 IIRC. the rest is a 16bit add, a 16bit > subtract, a 16x16bit multiplication and an 8bit add plus some "glue" -- not > too heavy. It took me a couple of passes to understand this, but you're (mostly) describing a phase accumulator. The way I usually think about it is: Imagine a counter that rolls over after 2^N counts. At evenly spaced time intervals, add a constant (delta phi) to the counter. The rollover frequency of the counter is (obviously) determined by the constant you add at each of the time intervals. What's not so obvious is that you can use fix point arithmetic to get rollover frequencies that are not an integer number of cpu cycles. You don't get something for nothing though - in the trade off you introduce some frequency jitter. The equation for the frequency is: delta phi 1 f = --------- * ------ delta t 2^N You know delta t - that's dictated by how fast some isocronous routine increments the phase accumulator. Usually you want this as small as possible since that gives better frequency resolution. You also know the frequency, f (that's what you're trying to generate). So you can solve for delta phi: delta phi = f * delta t * 2^N If you had a 20Mhz crystal and used the isochronous software pwm routine I mentioned in the last post, then delta t = 51.2uS. For N=14 bits you get a really simple equation (if I did the arithmetic correctly): delta phi = 0.83886 * f So to generate 2500Hz, delta phi would be 2097. This multiplication, btw only needs to be performed once per frequency. So the 16-bit multiply you mentioned doesn't need to be in the isochronous loop. Also, pre-computing all of the constants removes the need for the addition and subtraction. BTW The sine wave routine executes in 65 cycles. However, there are two ways to speed it up. One is to unroll a loop (it only loops four times). The other is to completely rewrite the routine and take advantage of the fact that the sine values are generated consecutively - this will save on average one table-look-up per iteration. Scott

1999\02\23@034143 by Gerhard Fiedler

At 18:48 02/22/99 -0800, Scott Dattalo wrote: >you're (mostly) describing a phase accumulator. if i had heard that term before, maybe i wouldn't have taken such a long way around it! :) >If you had a 20Mhz crystal and used the isochronous software pwm routine I >mentioned in the last post, then delta t = 51.2uS. For N=14 bits you get a >really simple equation (if I did the arithmetic correctly): > >delta phi = 0.83886 * f > >So to generate 2500Hz, delta phi would be 2097. i still see one problem here. 51.2us=19.5kHz, that's less than 8 points per period at 2500Hz. if i rewrite the pwm for a resolution of 128 (and double the frequency), i guess the free cycles would be less than half, too, and that's probably not enough. what i was trying to figure out is how to get an interrupt out of the hardware pwm module (like when the counter is reset), but i guess that's not possible. or only with feeding the pwm output back to RB0 and make sure the pwm register never gets 0. any ideas on that? i'd like to use the hardware pwm, because it gives me 4 times the frequency. thanks a lot, ge

'programmable sinewave generator chip?'
1999\07\20@171502 by Madis Kaal

I am thinking about building a digitally controlled sinewave generator and after considering the obvious brute force, lots of components approach and after that briefly considering CD player DAC chip I decided that there must be an easier way. Is anyone here aware of a chip that would allow me to tell it a frequency of a sine wave to output, in 1Hz steps would be nice? It sould need no precision analog components.

1999\07\20@173153 by l.allen

> I am thinking about building a digitally controlled sinewave generator > and after considering the obvious brute force, lots of components > approach and after that briefly considering CD player DAC chip I decided > that there must be an easier way. Is anyone here aware of a chip that > would > allow me to tell it a frequency of a sine wave to output, in 1Hz steps > would > be nice? It sould need no precision analog components. You should use DDS For up to 50kHz the MicroLinear ML2036 is very easy to use. Supply +/- 5v, one xtal and a PIC to talk to it over a simple 3 wire interface... youre away and laughing. Analog Devices make higher frequency DDS chips like AD9850. There is a complete 40MHz generator published at http://www.geocities.com/CapeCanaveral/LAb/9595/ddssg.html Lance Allen Uni of Auckland New Zealand

1999\07\20@181147 by Dan Tye

I recently had a similar problem finding an ic to output audio frequency tones. The solution I found was to use a device from MX-com part number 803A. This device does single frequency output and also DTMF. The frequency output is programmable to output non standard tones. +++++++++++++++++++++++++++++++++++++++ ++ Dan Tye ++ Chief Engineer ++ Tel-Tron Technologies Corporation ++ 220 Fentress Boulevard ++ Daytona Beach, FL 32114 ++ Phone: 904-255-1921 Fax: 904-258-3782 +++++++++++++++++++++++++++++++++++++++ {Original Message removed}

'DTMF Tones? [Idea]How to produce sine wave'
1999\11\12@140821 by Nikolai Golovchenko

part 0 3906 bytes {Quote hidden}>--- Dave VanHorn <RemoveMEdvanhornEraseMEKILLspamCEDAR.NET> wrote: >> > I would like to write some code to make DTMF >> tones. However I can't >> figure >> > out even how to produce noise. I have to hook up >> a capaciter or >> > something... I can't find any sample circuits and >> then even if I did. >> What >> > do I send? A pulse of some sort? >> >> >> This is one that might be better done in an external >> chip. > >I disagree. It might be easier if you have no idea how >to do it, but that's why people ask questions... > >I can think of several ways. Probably the easiest >would be to generate two sinusoidally modulated PWM >wave forms on two I/O pins, add them together through >a resistor network and low-pass filter them. Eric >Smith has a fast algorithm for generating sine waves, >while Scott Dattalo has one that based on phase >accumulators (and consequently easier to control the >frequency). Either of these may be used to generate >the two sine waves corresponding to the two tones. >Generating two pwm wave forms in software is more >challenging, but it CAN be done. I believe Lawrence >and again Scott have posted software pwm algorithms. > >I suspect (but don't have the time to show) that one >could combine the sinusoids digitally (as opposed to >an analog based summer). For example, the pwm duty >cycle could be formed by adding the sine waves >together. In other words, for every PWM cycle the duty >cycle is the proportional to the sum of the amplitudes >of the two sinusoids comprising the DTMF tone. > >You'll want to use phase accumulators to minimize >frequency errors and you'll need low-pass filters to >filter the PWM signal to remove the carrier. IIRC, a >single cycle resolution, 256 level PWM waveform on a >20MHZ pic has a carrier frequeny just under 20khz. >This would give you about 12 samples for the highest >DTMF frequency. > >.lo > >===== > >__________________________________________________ >Do You Yahoo!? >Bid and sell for free at http://auctions.yahoo.com > > I may be inventing the wheel... Anyway, there was an interesting idea in Scenix application notes on how to generate sine waveform not using table or multiply, just addition. Basically, they generate triangle waveform, then integrate it and, vou la, something very similar to sine is produced. Triangle waveform is easy to implement - its just addition and substraction. So, the frequency of sine waveform is equal to the primary triangle frequency. BUT, a problem appears when triangle frequency doesn't exactly match sampling frequency divided by an integer. Typical way to make triangle is something like: add to the current value fixed step untill the wave reaches maximum, then substract the step untill the wave reaches minimum, and again. What if the triangle reaches its maximums and minimums BETWEEN sampling points? In this case, an error will appear, and frequency is corrupted. A simple solution is: when triangle waveform goes over maximum, substract the number it went over from the maximum (see appendix); and same for minimum. As a result, the frequency is jitter free. The drawback is that the resulting sine amplitude is varying slightly depending on the step chosen. For DTMF it must be okay. 16bit resolution for step, accumulators of triangle and sine will be enough. For output, only 8 most significant bits can be taken. Is this the same as phase accumulators? _ Nikolai Golovchenko, Electrical Engineering Student National Mining University of Ukraine http://www.nmuu.dp.ua Dnepropetrovsk, Ukraine E-mail: spamBeGonegolovchenkospam_OUTRemoveMEmail.ru P.S. Tracy, may I ask you what "dot lo" means? :-) Attachment converted: wonderland:solution.gif (GIFf/JVWR) (0000F660)

1999\11\12@162527 by Tracy Smith

--- Nikolai Golovchenko <.....golovchenkoRemoveMEMAIL.RU> wrote: > I may be inventing the wheel... Anyway, there was an > interesting idea in > Scenix application notes on how to generate sine > waveform not using table or > multiply, just addition. Basically, they generate It's fun to reinvent the wheel (except when it's on your nickel). > triangle waveform, then > integrate it and, vou la, something very similar to > sine is produced. Are you sure this is not the Goertzel algorithm? Scott has a page that discusses this algorithm. Also, here's a dtmf generator that uses Goertzel http://www.ti.com/sc/docs/psheets/abstract/apps/spra066.htm http://www.interstice.com/~sdattalo/technical/theory/sinewave.html But again, I think Eric Smith's solution is the best. > Is this the same as phase accumulators? No. Phase accumulators are really, really cool. I just don't know how to adequately explain them. A quick alta vista search produced: http://www.g4dvj.demon.co.uk/dds.htm > P.S. > Tracy, may I ask you what "dot lo" means? > > :-) I let that cat out of the bag when I'm ready. :) .lo ===== __________________________________________________ Do You Yahoo!? Bid and sell for free at http://auctions.yahoo.com

1999\11\12@181518 by Paul B Webster

Nikolai Golovchenko wrote: > Is this the same as phase accumulators? I beg to differ from Tracy. I think it is. Phase accumulators are based on the principle of - as the name suggests - adding up little pieces of the cycle and "rolling over" when the sum of these amounts to or in fact exceeds, a full cycle. That appears to be exactly what is suggested here; you would be using a phase accumulator to control the up/ down ramp of the wave, then filtering it further in analogue circuitry. It begs the question though of using a full-blown phase accumulator/ DDS with (a number of stages of) integration. Once you go the DDS path, you are in a position to vary the slope of the waveform during the cycle, by using even a simple resistive DAC, (or perhaps PWM) which *greatly* eases the requirements on your integration/ low-pass filtering. I would think that even a 3-bit (six resistor) ADC would considerably improve performance, and perhaps allow output of both tones via the one channel (though Dave Van Horn's "twist" would still be a difficulty). -- Cheers, Paul B. (away from home for a day or so!).

1999\11\12@181743 by Harold M Hallikainen

On Fri, 12 Nov 1999 13:24:37 -0800 Tracy Smith <dot_lo@spam@YAHOO.COM> writes: >--- Nikolai Golovchenko <EraseMEgolovchenkoRemoveMESTOPspamMAIL.RU> wrote: > >> I may be inventing the wheel... Anyway, there was an >> interesting idea in >> Scenix application notes on how to generate sine >> waveform not using table or >> multiply, just addition. Basically, they generate > >It's fun to reinvent the wheel (except when it's on >your nickel). > >> triangle waveform, then >> integrate it and, vou la, something very similar to >> sine is produced. > Of course, a triangle wave is just a square wave run through an integrator (which is sorta like a 0 Hz low pass filter with a 6 dB/octave or 20 dB/decade roll-off). I think an additional stage of LPF is cheaper than doing a D/A of a triangle wave, then filtering it. I think phase accumulation and DDS is really neat, though I haven't used it yet in anything. By having different tables before the D/A, you can do arbitrary waveform generation. The anti-alias filter needs to be somewhere above the maximum output frequency and below the Nyquist frequency (50% of sample rate). With a perfect filter, you'd get a perfect sine wave out of the thing (assuming you are doing sine wave synthesis). You can vary the frequency and the output level remains the same. Anything that uses filters to remove harmonics (instead of alias components) will have the output level vary as the frequency is varied unless you have the filter track the output frequency (kind of a pain). One thing I'd like to try some time is to have a look up table that simulates a linear mix of several sinusoidal signals (stuff around 1 MHz), then drive the D/A. I wonder how much intermod there'd be... Harold Harold Hallikainen RemoveMEharoldKILLspamTakeThisOuThallikainen.com Hallikainen & Friends, Inc. See the FCC Rules at http://hallikainen.com/FccRules and comments filed in LPFM proceeding at http://hallikainen.com/lpfm ___________________________________________________________________ Get the Internet just the way you want it. Free software, free e-mail, and free Internet access for a month! Try Juno Web: dl.http://www.juno.com/dynoget/tagj.

1999\11\12@194116 by Tracy Smith

--- S Webster <spamBeGones.webster@spam@STUDENT.UNSW.EDU.AU> wrote: A.K.A. Paul > Nikolai Golovchenko wrote: > > > Is this the same as phase accumulators? > > I beg to differ from Tracy. I think it is. Phase > accumulators are > based on the principle of - as the name suggests - > adding up little > pieces of the cycle and "rolling over" when the sum > of these amounts to > or in fact exceeds, a full cycle. True, but that's not what Nikolai described (I thought). OTOH, I'm hardly the Phase Accumulator expert. My understanding of the phase accumulator is: You start off with a single cycle of something you wish to generate. This may be a square wave, sine wave, or a hyperbolic paraboloid wave (what in the hell is that?). The phase accumulator simply determines which portion of the cycle you wish to access. In its simplest form, the single cycle of the wave form is in an array and the phase accumulator is the index. The array is accessed at a constant rate, however the index is advanced according to the frequency of the wave form. Here's a simple C example: #define TABLE_SIZE 256 char wave_table[TABLE_SIZE]; float phase,delta_phase; void phase_accumulator(void) { int i; while(1) { phase += delta_phase; if(phase > TABLE_SIZE) phase -= TABLE_SIZE i = phase; output(wave_table[i]); routine_that_updates_delta_phase(&delta_phase); } } void main(void) { delta_phase = .... init_wave_table(wave_table); } I'd definite do this in assembly instead of C (but that's another thread). In each application, there are always ways to optimize this approach. For example, for square waves you don't need a table at all nor do you need floating point arithmetic for the phase accumulator. Fixed point arithmetic works fine and the most significant bit of the phase accumulator is the output. For most pics, you're forced to optimize because 256 element arrays (in ram aren't possible). 256 element arrays in code are possible, but are always a pain in the ass (because of page boundaries). Who knows, perhaps it's only semantic and S. and I are on the same page. But like I said, I'm not the expert... > I would think that even a 3-bit (six resistor) ADC > would considerably > improve performance, and perhaps allow output of > both tones via the one > channel (though Dave Van Horn's "twist" would still > be a difficulty). Borrowing a little from the best of both worlds is often the best solution. .lo ===== __________________________________________________ Do You Yahoo!? Bid and sell for free at http://auctions.yahoo.com

1999\11\13@122130 by Nikolai Golovchenko

-----Original Message----- From: Tracy Smith <RemoveMEdot_lospam_OUTYAHOO.COM> To: PICLISTspamMITVMA.MIT.EDU <spam_OUTPICLISTspam_OUTspam_OUTMITVMA.MIT.EDU> Date: Friday, November 12, 1999 11:26 PM Subject: Re: DTMF Tones? [Idea]How to produce sine wave {Quote hidden}>--- Nikolai Golovchenko <golovchenkospam_OUTMAIL.RU> wrote: > >> I may be inventing the wheel... Anyway, there was an >> interesting idea in >> Scenix application notes on how to generate sine >> waveform not using table or >> multiply, just addition. Basically, they generate > >It's fun to reinvent the wheel (except when it's on >your nickel). > >> triangle waveform, then >> integrate it and, vou la, something very similar to >> sine is produced. > >Are you sure this is not the Goertzel algorithm? Scott >has a page that discusses this algorithm. Sure it is not. The Goertzel algorithm calculates sine value by a recursive formula, which uses two previous values. Moreover, there is a multiply to do and Scott mentions round off errors accumulation. The method I suggest uses only one previous sample, no multiply, no frequency error, no ... :-) Should Scott Dattalo add yet another, 11th method? > >Also, here's a dtmf generator that uses Goertzel >http://www.ti.com/sc/docs/psheets/abstract/apps/spra066.htm > Are you kidding? This is a decoder algorithm. {Quote hidden}>http://www.interstice.com/~sdattalo/technical/theory/sinewave.html > >But again, I think Eric Smith's solution is the best. > >> Is this the same as phase accumulators? > >No. Phase accumulators are really, really cool. I just >don't know how to adequately explain them. A quick >alta vista search produced: > >http://www.g4dvj.demon.co.uk/dds.htm > Thanks for the link. Explanation is perfect. I think that my method resembles phase accumulator and table method except there is no table. The table is generated on the run and this is triangle. In my case error of triangle generation is lower, because there is no restriction to use only table values. Finally, this reduced error makes the sine wave frequency more stable. Certainly, the phase accumulator method is more straight-forward and simple, at the cost of frequency jitter. In case of DTMF this is mainly the frequency what matters, because DTMF frequencies are very close. {Quote hidden}>> P.S. >> Tracy, may I ask you what "dot lo" means? >> >> :-) > >I let that cat out of the bag when I'm ready. :) > >.lo > >===== > >__________________________________________________ >Do You Yahoo!? >Bid and sell for free at http://auctions.yahoo.com > > ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ -----Original Message----- From: Harold M Hallikainen <RemoveMEharoldhallikainenKILLspam@spam@JUNO.COM> To: PICLISTspamBeGone.....MITVMA.MIT.EDU <KILLspamPICLIST.....MITVMA.MIT.EDU> Date: Saturday, November 13, 1999 1:17 AM Subject: Re: DTMF Tones? [Idea]How to produce sine wave > Of course, a triangle wave is just a square wave run through an >integrator (which is sorta like a 0 Hz low pass filter with a 6 dB/octave >or 20 dB/decade roll-off). I think an additional stage of LPF is cheaper >than doing a D/A of a triangle wave, then filtering it. Thanks Harold, I wonder why it didn't come to my head... In this particular case, triangle of fixed frequency seems easier to produce than square. _ Nikolai Golovchenko, Electrical Engineering Student National Mining University of Ukraine http://www.nmuu.dp.ua Dnepropetrovsk, Ukraine E-mail: spam_OUTgolovchenkoKILLspammail.ru

1999\11\14@110103 by Tracy Smith

> >Are you sure this is not the Goertzel algorithm? > Scott > >has a page that discusses this algorithm. > > > Sure it is not. The Goertzel algorithm calculates > sine value by a recursive Yeah, I misunderstood what you were describing. Sorry > > > >Also, here's a dtmf generator that uses Goertzel > >www.ti.com/sc/docs/psheets/abstract/apps/spra066.htm > > > > > Are you kidding? This is a decoder algorithm. No, I'm not kidding, just wrong. > >http://www.g4dvj.demon.co.uk/dds.htm > > > > > Thanks for the link. Explanation is perfect. 1 out of 3 pieces of information distributed on the pic list is decent. :) {Quote hidden}> I think that my method resembles phase accumulator > and table method except > there is no table. The table is generated on the run > and this is triangle. > In my case error of triangle generation is lower, > because there is no > restriction to use only table values. Finally, this > reduced error makes the > sine wave frequency more stable. Certainly, the > phase accumulator method is > more straight-forward and simple, at the cost of > frequency jitter. In case > of DTMF this is mainly the frequency what matters, > because DTMF frequencies > are very close. True, butt, if you incorporate the proper interpolation along with your look up table, then the jitter effects are mitigated. Without any interpolation, the sinewave output from the look-up table is a stair-stepped waveform. The higher the update frequency, the smaller the steps and the closer to a sine wave approximation. This 'stair-stepped' wave form however is a form of interpolation; specifically zero-order interpolation. You may have heard of a "zero-order hold" in your studies on signals and systems (and perhaps z-transforms). According to the 'theory', as long as you satisfy the Nyquist sampling criterion, you can exactly reproduce the synthesized wave form. But I couldn't look at you with a straight face and say it's possible. The caveat is that a perfect low-pass filter must be constructed. DTMF competition is much too cheap for us to insist on perfect analog filters. So the next step is first order interpolation. For our application, this is linear interpolation between consecutive samples in the table. (For sampling theory, there are some subtle but important differences on the interpretation of a first-order hold.) Which is to say, if you need a sample that is not (exactly) at one of the sample points in your table, then you approximate it by using linear interpolation. For example, suppose you had a table with 10 samples: f(i) is the function defined for i =0..9 . Now suppose you need f(2.25)? Well, using linear interpolation you'd approximate this: f(2.25) - f(2) f(3) - f(2) -------------- = ------------- 2.25 - 2 3 - 2 f(2.25) ~= f(2) + 0.25*(f(3) -f(2)) >From a frequency or spectral point of view, linear interpolation's effect on higher harmonics are approximately the square of the harmonics introduced by zero-order interpolation (sinc vs. sinc^2). And since the magnitude of the higher harmonics is less than one, this is a good thing (e.g. 1/9 is much smaller than 1/3). The direct implication is that the stringent low-pass filter constraints associated with the stair stepped waveform can be relaxed. >From an intuitive view, you can see how first-order interpolation improves the table values. For example, if you had 64 samples of sine wave over one cycle, then the deviation between linear interpolation and the 'true value' of the sine wave is quite small. In fact, an error bound may be placed on it. Specifically, as I alluded to in another thread, you can use the Cauchy Remainder Theorem for interpolation. In essence, this theorem states that the error is bounded by the n'th derivative of the function your approximating, where n is the order of the interpolation (1 for linear). So in the above example, The error in approximating: f(2.25) ~= f(2) + 0.25*(f(3) -f(2)) error <= abs(f'(3) - f'(2)) Now there are additional caveats, like you still have to satisfy Nyquist. .lo ps. Please forgive any mistakes, this is all from memory... ===== __________________________________________________ Do You Yahoo!? Bid and sell for free at http://auctions.yahoo.com

1999\11\15@021719 by Nikolai Golovchenko

-----Original Message----- From: Tracy Smith <RemoveMEdot_loRemoveMEEraseMEYAHOO.COM> To: KILLspamPICLISTspamBeGoneMITVMA.MIT.EDU <PICLISTspamMITVMA.MIT.EDU> Date: Sunday, November 14, 1999 6:01 PM Subject: Re: DTMF Tones? [Idea]How to produce sine wave >1 out of 3 pieces of information distributed on the >pic list is decent. :) I would say 1 out of 10. ;) {Quote hidden}> >> I think that my method resembles phase accumulator >> and table method except >> there is no table. The table is generated on the run >> and this is triangle. >> In my case error of triangle generation is lower, >> because there is no >> restriction to use only table values. Finally, this >> reduced error makes the >> sine wave frequency more stable. Certainly, the >> phase accumulator method is >> more straight-forward and simple, at the cost of >> frequency jitter. In case >> of DTMF this is mainly the frequency what matters, >> because DTMF frequencies >> are very close. > >True, butt, if you incorporate the proper >interpolation along with your look up table, then the >jitter effects are mitigated. Without any >interpolation, the sinewave output from the look-up >table is a stair-stepped waveform. The higher the >update frequency, the smaller the steps and the closer >to a sine wave approximation. This 'stair-stepped' This means a biggg table. Interesting, what size of table will be sufficient to minimize jitter enough for DTMF application. {Quote hidden}>So the next step is first order interpolation. For our >application, this is linear interpolation between >consecutive samples in the table. (For sampling >theory, there are some subtle but important >differences on the interpretation of a first-order >hold.) Which is to say, if you need a sample that is >not (exactly) at one of the sample points in your >table, then you approximate it by using linear >interpolation. For example, suppose you had a table >with 10 samples: f(i) is the function defined for i >=0..9 . Now suppose you need f(2.25)? Well, using >linear interpolation you'd approximate this: > >f(2.25) - f(2) f(3) - f(2) >-------------- = ------------- > 2.25 - 2 3 - 2 > >f(2.25) ~= f(2) + 0.25*(f(3) -f(2)) > Let me compare two methods of sine generation for DTMF application: 1)Triangle generation with subsequent integration. 2)Phase accumalator&table with linear interpolation; The first method doesn't have table, instead, triangle waveform is generated using addition of a fixed step to the previous sample. No need to use interpolation means faster execution time. To produce sine waveform integration is used - some degradation in the waveform, which is acceptable in DTMF. As a result, the first method generates efficiently jitter free, sine waveform signal. The second method has table, therefore it requires more ROM; needs linear interpolation - more computing time because of multiplication. Result is the same. What do you think? _ Nikolai Golovchenko, Electrical Engineering Student National Mining University of Ukraine http://www.nmuu.dp.ua Dnepropetrovsk, Ukraine E-mail: RemoveMEgolovchenkospamBeGoneRemoveMEmail.ru

1999\11\15@024911 by Dave VanHorn

> What do you think? You want the part number of that chip again? :) Three pins, clocked serial byte for what digit to say, and one pin as /xmit. One external component, a 3.585 mhx xtal or resonator. :o)

1999\11\15@101948 by Tracy Smith

--- Dave VanHorn <KILLspamdvanhornspamBeGoneCEDAR.NET> wrote: > > What do you think? > > > You want the part number of that chip again? :) > Three pins, clocked serial byte for what digit to > say, and one pin as /xmit. > One external component, a 3.585 mhx xtal or > resonator. > > :o) Wouldn't it be better to run the 12C508 off the internal oscillator instead. :) .lo ===== __________________________________________________ Do You Yahoo!? Bid and sell for free at http://auctions.yahoo.com

1999\11\15@105408 by Tracy Smith

--- Nikolai Golovchenko <@spam@golovchenkoSTOPspam@spam@MAIL.RU> wrote: > -----Original Message----- > >True, butt, if you incorporate the proper > >interpolation along with your look up table, then > the > >jitter effects are mitigated. Without any > >interpolation, the sinewave output from the look-up > >table is a stair-stepped waveform. The higher the > >update frequency, the smaller the steps and the > closer > >to a sine wave approximation. This 'stair-stepped' > > > This means a biggg table. Interesting, what size of > table will be sufficient > to minimize jitter enough for DTMF application. Actually, the table is not biggg. That's the beauty of linear interpolation. See below. {Quote hidden}> > >So the next step is first order interpolation. For > our > >application, this is linear interpolation between > >consecutive samples in the table. (For sampling > >theory, there are some subtle but important > >differences on the interpretation of a first-order > >hold.) Which is to say, if you need a sample that > is > >not (exactly) at one of the sample points in your > >table, then you approximate it by using linear > >interpolation. For example, suppose you had a table > >with 10 samples: f(i) is the function defined for i > >=0..9 . Now suppose you need f(2.25)? Well, using > >linear interpolation you'd approximate this: > > > >f(2.25) - f(2) f(3) - f(2) > >-------------- = ------------- > > 2.25 - 2 3 - 2 > > > >f(2.25) ~= f(2) + 0.25*(f(3) -f(2)) > > > > > Let me compare two methods of sine generation for > DTMF application: > 1)Triangle generation with subsequent integration. > 2)Phase accumalator&table with linear interpolation; > > The first method doesn't have table, instead, > triangle waveform is generated > using addition of a fixed step to the previous > sample. No need to use > interpolation means faster execution time. To > produce sine waveform > integration is used - some degradation in the > waveform, which is acceptable > in DTMF. As a result, the first method generates > efficiently jitter free, > sine waveform signal. Actually "some degradation" may not be acceptable. A symmetrical triangle wave has odd harmonics whose strengthes vary inversely to the square of the harmonic number (e.g. the third harmonic is 1/9'th the strength of the fundamental). A low pass filter would be of no use since to generate 440Hz, you'd need to block 1320Hz which would also block a few of the higher frequency DTMF tones as well. Ideally, you want to have to only block some very high frequency carrier. For example, if you were using PWM with a 20kHz carrier to generate DTMF then you'd need to block the 20kHz and pass (a maximum of) ~1600Hz. A relatively simple 2'nd order low pass filter with perhaps a cascaded 2'nd order notch (or band-stop) would be sufficient. > > The second method has table, therefore it requires > more ROM; needs linear > interpolation - more computing time because of > multiplication. Result is the > same. > > > What do you think? I just re-visited Scott's sine wave page: http://www.interstice.com/~sdattalo/technical/software/pic/picsine.html His table contains only 16 entries and (according to the comments) can generate a new sample every 65 instruction cycles. I see a way to knock 10 cycles off. So to generate the two tones you'd need about 110 cycles plus an additional overhead of perhaps 50 more cycles. This'll give you a new sample at about a 30kHz rate (for a 20Mhz pic). But you'd still need a way to convert this to analog. So perhaps you'd could back the generation frequency to 20kHz (or ~256 cycles) and use an 8-bit pwm. --------- Perhaps it's not obvious, but the triangle-to-sine algorithm may also be viewed as a kind of a digital filter, or more specifically the impulse response of a digital filter. Consequently, you may wish to consider other simple algorithms. (e.g. a 4'th order FIR filter with binary coefficients or perhaps a 3'rd order IIR.) .lo ===== __________________________________________________ Do You Yahoo!? Bid and sell for free at http://auctions.yahoo.com

1999\11\16@012723 by Nikolai Golovchenko

-----Original Message----- From: Tracy Smith <dot_lospamBeGonespamBeGoneYAHOO.COM> To: spamBeGonePICLISTMITVMA.MIT.EDU <spam_OUTPICLISTSTOPspamMITVMA.MIT.EDU> Date: Monday, November 15, 1999 5:54 PM Subject: Re: DTMF Tones? [Idea]How to produce sine wave >Actually, the table is not biggg. That's the beauty of >linear interpolation. See below. Sorry, I wasn't clear at this. I meant that interpolation may be omitted if the table is big enough to reduce jitter for DTMF frequencies. The question is how big. >Actually "some degradation" may not be acceptable. A >symmetrical triangle wave has odd harmonics whose >strengthes vary inversely to the square of the >harmonic number (e.g. the third harmonic is 1/9'th the >strength of the fundamental). A low pass filter would >be of no use since to generate 440Hz, you'd need to You forget the integration. It acts like LP filter (6 dB/octave) and is made digitally. Already, third harmonic magnitude A3 = 1/9 A1 (-19 dB, assuming 0 dB for the fundametal harmonic), and fifth A5 = 1/25 A1 (-28 dB). When integration is applied, higher harmonics will be further attenuated by 9.5 dB for the third one and by 14 dB for the fifth. Totally, the relative level of third harmonic is -28 dB and the fifth is -42 dB. As far as I know, required signal to noise ratio in this kind of systems is 20 dB. So integration solves the problem, isn't it? {Quote hidden}>block 1320Hz which would also block a few of the >higher frequency DTMF tones as well. Ideally, you want >to have to only block some very high frequency >carrier. For example, if you were using PWM with a >20kHz carrier to generate DTMF then you'd need to >block the 20kHz and pass (a maximum of) ~1600Hz. A >relatively simple 2'nd order low pass filter with >perhaps a cascaded 2'nd order notch (or band-stop) >would be sufficient. An LP filter should be used, of course, at the output stage of D/A or PWM. It's pass band should include all DTMF frequencies. {Quote hidden}>I just re-visited Scott's sine wave page: > >http://www.interstice.com/~sdattalo/technical/software/pic/picsine.html > >His table contains only 16 entries and (according to >the comments) can generate a new sample every 65 >instruction cycles. I see a way to knock 10 cycles >off. So to generate the two tones you'd need about 110 >cycles plus an additional overhead of perhaps 50 more >cycles. This'll give you a new sample at about a 30kHz >rate (for a 20Mhz pic). But you'd still need a way to >convert this to analog. So perhaps you'd could back >the generation frequency to 20kHz (or ~256 cycles) and >use an 8-bit pwm. I dont have time right now to look at this routine, have to go to school. It seems that this small table will require additional effort on calculation of mid points to get more accurate frequency. >Perhaps it's not obvious, but the triangle-to-sine >algorithm may also be viewed as a kind of a digital >filter, or more specifically the impulse response of a >digital filter. Consequently, you may wish to consider Yeah, this is a first order FIR filter with unitary coefficients. >other simple algorithms. (e.g. a 4'th order FIR filter >with binary coefficients or perhaps a 3'rd order IIR.) Simple? :) Second order FIR may be. But 3'rd order IIR - no, thanks. First order IIR is enough to generate sine waveform. It requires cumbersome multiplication. Bye for now. _ Nikolai Golovchenko, Electrical Engineering Student National Mining University of Ukraine http://www.nmuu.dp.ua Dnepropetrovsk, Ukraine E-mail: RemoveMEgolovchenkospammail.ru

1999\11\16@183333 by Tracy Smith

At the risk of boring everyone with our on-line dialog, > >Actually, the table is not biggg. That's the beauty > of > >linear interpolation. See below. > > > Sorry, I wasn't clear at this. I meant that > interpolation may be omitted if > the table is big enough to reduce jitter for DTMF > frequencies. The question > is how big. Again, Cauchy Remainder Theorem for interpolation may be used. In this instance however, it's zero order interpolation (i.e. NO interpolation). IIRC, the error will be proportional to the differences in consecutive entries in the table (I don't know if or what the proportionality constant is). I think intuitively this makes sense. If the index into the table is a fraction, you'll have to round to the nearest integer. Consequently, the ideal sample which lies somewhere between the two that are in the table will be off. And assuming you're satisfying Nyquist, the error will be bounded by the samples that are available. > You forget the integration. Could you perhaps drag up the details showing how this algorithm really works? > I dont have time right now to look at this routine, > have to go to school. It > seems that this small table will require additional > effort on calculation of > mid points to get more accurate frequency. Compared to Eric's table, you're right. His takes about 15 cycles; a savings of about 40 cycles. OTOH, it has 64 entries in the table. I'd probably choose Eric's sine over Scott's for DTMF (especially since Eric already wrote the damn DTMF dialer). 64 table entries is not that many. Further more since the sample output rate is potentially 3 or 4 times faster you could ease the filter requirements even more (again, this depends on how you actually go from a to d...) {Quote hidden}> > >Perhaps it's not obvious, but the triangle-to-sine > >algorithm may also be viewed as a kind of a digital > >filter, or more specifically the impulse response > of a > >digital filter. Consequently, you may wish to > consider > > > Yeah, this is a first order FIR filter with unitary > coefficients. > > >other simple algorithms. (e.g. a 4'th order FIR > filter > >with binary coefficients or perhaps a 3'rd order > IIR.) > > > Simple? :) Second order FIR may be. But 3'rd order > IIR - no, thanks. First > order IIR is enough to generate sine waveform. It > requires cumbersome > multiplication. The simplest 3'rd order IIR I know is Simpson's 3/8'ths rule. It's not quite as good as the 1/3 rule, but it's much better than the rectangular rule for numeric integration. The only caveat is that even though it is an integration formula, it's transfer function in fact amplifies high frequencies. .lo ===== __________________________________________________ Do You Yahoo!? Bid and sell for free at http://auctions.yahoo.com

1999\11\17@152453 by Nikolai Golovchenko

part 0 3051 bytes content-type:application/octet-stream; >Could you perhaps drag up the details showing how this >algorithm really works? The algorithm is really simple. You can find the original version in Scenix Application Note #11 (http://www.scenix.com). The original version is hard to use for any frequency because of fixed triangle slope and amplitude. So I changed the algorithm a little to adjust for DTMF frequencies generation. First you initialize accumulators. There are two - one is a triangle accumulator and another is a sine accumulator. Then, each new sample is calculated by a common routine. Note, that these accumulators contain current value of sine/triangle, not phase. Sine waveform is generated by adding to the last sine value the last value of triangle. This may be regarded as integration or first-order FIR filtering of triangle signal. As a result, the fundamental harmonic of triangle is attenuated significantly less then other harmonics, and you have almost clean sine wave. The sine wave frequency is equal to that of triangle. Triangle waveform is primary in this case and generated by adding pre-calculated fixed step to the triangle accumulator. When the accumulator reaches top limit (also pre-calculated for the given frequency) the step's sign is changed. A little nuance here. If the triangle peak lies somewhere between current and last sample then a correction required to keep the triangle value correct. After triangle reaches bottom limit, the step's sign changes again. Actually, I wrote the sine waveform generation routine for PIC. It takes 64 instructions, 8 RAM bytes, 24 cycles best case, and 41 cycles worst case. This is 14 cycles savings from your supposed 55. The routine takes input 2 bytes for limit and 2 bytes for step, other 4 bytes are for 16 bit accumulators. For comparison I also simulated the routine of Scott Dattalo in MATLAB at 20 kHz sampling rate for all DTMF frequencies. Results are very good. There is no frequency jitter at all and the frequency error is very small. Sorry for wrong suggestions. The phase steps are: f, Hz Phase step 697 571 770 631 852 698 941 771 1209 990 1336 1094 1477 1210 1633 1338 phaseaccstep = round(16384 * f / f0), where f=needed freq. and f0=sampling freq. If I had to do DTMF dialer I would choose the phase accumulators method and Scott Dattalo's routine, because it can provide close approximation to sine wave, small program size, frequency can be changed smoothly, without phase jerk. I suppose other applications may exist where higher speed of my method is very important. Thanks for productive dialog. I hope this will help somebody else. Best regards. _ Nikolai Golovchenko, Electrical Engineering Student National Mining University of Ukraine http://www.nmuu.dp.ua Dnepropetrovsk, Ukraine E-mail: TakeThisOuTgolovchenkospamRemoveMEmail.ru Content-Type: application/octet-stream; name="sin.asm" Content-Disposition: attachment; filename="sin.asm" Attachment converted: wonderland:sin.asm (????/----) (0000F8FC)

'[OT]Re: DTMF Tones? [Idea]How to produce sine wave'
1999\11\17@162538 by Jim Hartmann

Very cool! But its not a true sine wave, its a parabolic wave if that's a real term. One segment of a triangle wave is a line. The algorithm is an integrator, and the integral of a line like y = x is y = (x^2)/2+c, a parabola (let c be your DC offset). Maybe someone else can tell us what the error magnitude amounts to comparing the "parabolic wave" to a sine? Jim Hartmann

'DTMF Tones? [Idea]How to produce sine wave'
1999\11\17@162744 by Dave VanHorn

> For comparison I also simulated the routine of Scott Dattalo in MATLAB at 20 > kHz sampling rate for all DTMF frequencies. Results are very good. There is > no frequency jitter at all and the frequency error is very small. Sorry for > wrong suggestions. So how do you establish twist?

1999\11\17@191816 by Tracy Smith

--- Nikolai Golovchenko <KILLspamgolovchenkospamspam_OUTMAIL.RU> wrote: {Quote hidden}> Hello Tracy. > > > >Could you perhaps drag up the details showing how > this > >algorithm really works? > > > The algorithm is really simple. You can find the > original version in Scenix > Application Note #11 (http://www.scenix.com). The original > version is hard to use > for any frequency because of fixed triangle slope > and amplitude. So I > changed the algorithm a little to adjust for DTMF > frequencies generation. > > First you initialize accumulators. There are two - > one is a triangle > accumulator and another is a sine accumulator. Then, > each new sample is > calculated by a common routine. Note, that these > accumulators contain > current value of sine/triangle, not phase. > > Sine waveform is generated by adding to the last > sine value the last value > of triangle. This may be regarded as integration or > first-order FIR > filtering of triangle signal. As a result, the > fundamental harmonic of > triangle is attenuated significantly less then other > harmonics, and you have > almost clean sine wave. The sine wave frequency is > equal to that of > triangle. Triangle waveform is primary in this case > and generated by adding > pre-calculated fixed step to the triangle > accumulator. When the accumulator > reaches top limit (also pre-calculated for the given > frequency) the step's > sign is changed. A little nuance here. If the > triangle peak lies somewhere > between current and last sample then a correction > required to keep the > triangle value correct. After triangle reaches > bottom limit, the step's sign > changes again. Without taking the time to read scenix's appnote, I'll take your word at face value - (I have no reason not to since you're obviously a logical thinker), and add just one more comment. If instead of generating triangle waves you wish to generate trapezoidal waves, then it possible to exactly suppress all of the even harmonics AND the third harmonic. I'll leave it as an excercise for the student, but my theoretical analysis proves that if the trapezoid has a rise and fall time that is one third of the period then there'll be no third harmonic. I'm not sure if the scenix app is conducive to generating trapezoids, but it's certainly worth an effort to investigate. I'm not sure what the harmonics of the integrated triangle wave look like either. Perhaps .lo ===== __________________________________________________ Do You Yahoo!? Bid and sell for free at http://auctions.yahoo.com

1999\11\17@193450 by Tracy Smith

--- Dave VanHorn <dvanhornRemoveMECEDAR.NET> wrote: > > For comparison I also simulated the routine of > Scott Dattalo in MATLAB at > 20 > > kHz sampling rate for all DTMF frequencies. > Results are very good. There > is > > no frequency jitter at all and the frequency error > is very small. Sorry > for > > wrong suggestions. > > > So how do you establish twist? > How much twist is needed? 3dB? 6dB? I think the best way to approach this is to write: dtmf = A*sin1 + B*sin2 The twist will be 20*log(B/A), right? Determine how many dB are needed and how much error can be tolerated and then find two 'easy to multiply' integers for A and B. 6dB is easy, because B=2 and A=1 would satisfy the equation. To maximize the output amplitude (and consequently minimize truncation errors) you'd want to weight one term with 1/3 and the other with 2/3 for a 6dB twist. So you'd have one multiplication by 2 and one division by 3. I think division by 3 has beaten to blithering pulp. .lo ===== __________________________________________________ Do You Yahoo!? Bid and sell for free at http://auctions.yahoo.com

'[OT]Re: DTMF Tones? [Idea]How to produce sine wave'
1999\11\18@105937 by Tracy Smith

--- Jim Hartmann <EraseMEJim_hartmannSTOPspamRemoveMESILENTKNIGHT.COM> wrote: > Very cool! But its not a true sine wave, its a > parabolic wave if that's a > real term. One segment of a triangle wave is a > line. The algorithm is an > integrator, and the integral of a line like y = x is > y = (x^2)/2+c, a > parabola (let c be your DC offset). Maybe someone > else can tell us what > the error magnitude amounts to comparing the > "parabolic wave" to a sine? I was wondering who'd drag this [OT] first. I did a quick back-of-the-notebook calculation and discovered what should've been intuitive. If you're familiar with time to frequency transformations like Fourier or Laplace then you'll see why I say this. Let's look at the way we've been discussing how to approximate sine waves and their corresponding harmonics. First we started with square waves. We know that the harmonic strengths fall off inversely to the harmonic number (and for symmetrical square waves, there are no even harmonics). This inverse relation ship, 1/s or 1/(jw), is a direct result of taking the transformation of a step function. The additional factors such as defining the harmonic locations are determined by the periodicity of the wave form. When we went to triangle waves, we noted that the harmonic strengths diminished as the square of the harmonic number, 1/s^2 of 1/(jw)^2 . Again, this is a direct result of taking the transform of a line. Continuing with this line of reason, we should (intuitively) suspect that a wave created with parabolas would have harmonics strengths diminishing with the cube of the harmonic number, 1/s^3 or 1/(jw)^3. Guess what? I suppose one could continue with this reasoning to suppress the harmonics even more. But (especially on the pic) you'll reach a point of diminishing returns. This whole subject btw, falls into the category of polynomial approximation. So far we've been using really simple polynomials. In fact to tie in with an earlier observation, so far these polynomials are like FIR filters. If one wanted to create the analogue (perhaps an inappropriate adjective) IIR filter, then perhaps rational Pade' approximations would be of some use. .lo __________________________________________________ Do You Yahoo!? Bid and sell for free at http://auctions.yahoo.com

1999\11\18@190711 by Sean Breheny

Along the lines of this thread: Can someone please tell me the difference between FIR and IIR filters? I am familiar with impulse response for continuous time systems, and with transfer functions,but I have not yet covered IIR versus FIR in class, nor have I read any explanation of it yet, so I am curious. Is it just that the impulse response is time-limited? Thanks, Sean | | Sean Breheny | Amateur Radio Callsign: KA3YXM | Electrical Engineering Student \--------------=---------------- Save lives, please look at http://www.all.org Personal page: http://www.people.cornell.edu/pages/shb7 spam_OUTshb7RemoveMEEraseMEcornell.edu ICQ #: 3329174

1999\11\19@043235 by Nikolai Golovchenko

-----Original Message----- From: Sean Breheny <TakeThisOuTshb7RemoveME@spam@CORNELL.EDU> To: EraseMEPICLISTRemoveMEMITVMA.MIT.EDU <spamPICLIST.....spamMITVMA.MIT.EDU> Date: Friday, November 19, 1999 2:07 AM Subject: Re: [OT]Re: DTMF Tones? [Idea]How to produce sine wave >Along the lines of this thread: Can someone please tell me the difference >between FIR and IIR filters? I am familiar with impulse response for >continuous time systems, and with transfer functions,but I have not yet >covered IIR versus FIR in class nor have I read any explanation of it yet, >so I am curious. Is it just that the impulse response is time-limited? FIR filters use only input signal samples to compute the next output. The difference equation for FIR filter is: y(n) = b0 * x(n) + b1 * x(n - 1) + b2 * x(n - 2) + ..., where y(n) - n'th sample of output x(n) - n'th sample of input IIR filters use also output signals samples. That is, y(n) = b0 * x(n) + b1 * x(n - 1) + b2 * x(n - 2) + ... - a1 * y(n - 1) - a2 * y(n - 2) - ..., You can see from the FIR formula that its impulse response is indeed time-limited, because its output is based on a number of input samples and if input becomes zero then output will go zero sooner or later. The length of FIR filter buffer defines the frequency and magnitude resolution that the filter can provide. For example, a low pass filter with pass band edge frequency of 100 Hz (0dB) and stop frequency 150 Hz (-60 dB) at 44100 Hz sampling requires about 1400 buffer size. FIR filters can have linear phase on frequency dependence. IIR filter output depends on both input and output samples. IIR filters resemble analog filters in this respect and often synthesized on an analog model. IIR filters generally require less memory size, but harder to implement. They have problems with stability, coeficients resolution. Good idea is break the whole filter into second order sections. Response of a practical IIR filter may be finite because of limited coeficients length. Sometimes, the response is made infinite intentionally to generate for example sine waves. If you want more information on digital filters check out Texas Instruments and Analog Devices web sites. They should have lots of information on this topic. _ Nikolai Golovchenko, Electrical Engineering Student National Mining University of Ukraine http://www.nmuu.dp.ua Dnepropetrovsk, Ukraine E-mail: golovchenkospam_OUT@spam@mail.ru

1999\11\19@091501 by Barry King

Nikolai, Thanks for that information, it was the best short-and-sweet explanation of IIR anf FIR filters I ever heard. I think too many people get lost in the theoretical mathematics which derive the IIR and FIR behaviors from the similar continuous time filters. Many university courses approach this from continuous time circuits, then Fourier series, then Fourier transforms, then sampling, then Z transforms, then digital filters. By that time many are confused! But you can also explain "backwards" by learning about sampling, then talk of averaging samples, (FIR, equal co-efficients, right?!) Then talk of weighted averages (FIR, unequal coefficients), and so on. Then go on to feedback and IIR techniques. This makes the practical digital techniques clear. Of course, understanding the design of a complex (or excellent) digital filter makes us get back to Z transforms some day. But it makes more sense to me when I remember to start at sampled data and averaging! ------------ Barry King, Engineering Manager NRG Systems "Measuring the Wind's Energy" http://www.nrgsystems.com Phone: 802-482-2255 FAX: 802-482-2272

1999\11\20@130120 by Nikolai Golovchenko

Hello Barry. Absolutely true. Digital filters took my imagination long before the Control Theory course and I started to learn how they work "backwards", as you say, from sampling, ADC. DAC, etc. So I can say now it is really easier to study digital signal processing from simple examples like averaging, integration, differentiation... Unfortunately, I hadn't a chance yet to work with DSPs, so I am limited to simulation only. MATLAB is a terrific tool for this. By the way, I'm glad that you liked my very short explanation, because sometimes I'm too short to be understood :o) _ Nikolai Golovchenko, Electrical Engineering Student National Mining University of Ukraine http://www.nmuu.dp.ua Dnepropetrovsk, Ukraine E-mail: .....golovchenkospam.....mail.ru

'Generating a Sine wave with a PIC'
1999\12\27@082039 by aipi Wijnbergen

Hi All, >From time to time, people here on the list need to generate a sine wave with a PIC, here is a solution with an active filter that would convert a PWM output into a sine wave. Not a general purpose solution, but, good for some applications, see: http://www.burr-brown.com/WebObjects/BurrBrown/download/ABs/AB-058.pdf I hope that it help some of you, Chaipi

1999\12\27@172201 by Mark Willis

I forget if anyone mentioned "Magic Sinewaves", http://www.tinaja.com/magsn01.html? Mark Chaipi Wijnbergen wrote: {Quote hidden}> > Hi All, > > >From time to time, people here on the list need to generate a sine wave > with a PIC, here is a solution with an active filter that would convert a > PWM output into a sine wave. Not a general purpose solution, but, good for > some applications, see: > > http://www.burr-brown.com/WebObjects/BurrBrown/download/ABs/AB-058.pdf > > I hope that it help some of you, Chaipi -- I re-ship for small US & overseas businesses, world-wide. (For private individuals at cost; ask.)

'Lookup Tables: Sine Converter'
2000\02\01@165822 by Thomas McGahee

Very often a lookup table or set of lookup tables is the simplest solution. By using a little creativity you can greatly reduce the size lookup table needed. Let's say you want to find the Sine of x. First off, the sine of x only has to be calculated or stored for the first 90 degrees. In this quadrant the value ranges from 0 to 1. In the second quadrant (90 to 180) the value ranges from 1 to 0. In this second quadrant you can get the equivalent 1st quadrant value by using 180-x. For example, 180-110=70, so sin(110)=sin(70). In the 3rd quadrant (180 to 270) the value ranges from 0 to -1. In the 3rd quadrant sin(x)=-sin(x-180). In the 4th quadrant (270 to 360) the value ranges from -1 to 0. In the 4th quadrant Sin(x)=-sin(360-x). The implication of this is that you can use a single table that covers from 0 to 90 degrees. You test the incoming x value and branch to one of four routines, based on the quadrant to which x belongs. Each subroutine would then convert x to the proper value, look up the value in the table, and set a posneg indicator that will tell us whether the last value converted is considered a positive or negative value. (The returned value is in positive format, and the posneg flag tells us whether it is really positive or negative). The lookup table can have the answer stored in either binary or packed decimal format. Which form you use depends on what you are going to do with the answer. If you are going to display the answer on an LCD, then use packed decimal format. If you are going to perform further math on the value, then binary might be best. If the math perfomed is something simple like multiply the value by some constant, then you can even store the table values so that you have the value multiplied by the constant already in the table. In packed decimal format you use the upper 4 bits to store one digit, and the lower 4 bits to store another digit. Two bytes will thus get you 4 decimal digits. You can thus store values from .0000 to .9999 You could get 6 digit accuracy if you use 3 bytes of storage per value. Let's assume you want 6 digits of accuracy and a resolution of .5 degrees. There are 90 degrees times 2 to give us .5 degree resolution, times 3 bytes of storage. 90x2x3=540 bytes of storage. A lookup table is limited to a maximum size of 256. This is due to the fact that the addressing mechanism is limited to a one byte value. The way you can get around this limitation is to use three jump tables each having a length of 180. Let's call these tablehigh, tablemiddle, and tablelow. You would load the tables with the required data: SINE Angle In Degrees | Offset Address | VALUE | High | Middle | Low ---------------------------------------------------------------- 0.0 | 0 |.000000| 00 | 00 | 00 0.5 | 1 |.008727| 00 | 87 | 27 1.0 | 2 |.017452| 01 | 74 | 52 1.5 | 3 |.026177| 02 | 61 | 77 etcetera | etcetera | etc. | etc. | etc. | etc. etcetera | etcetera | etc. | etc. | etc. | etc. 89.5 | 179 |.999962| 99 | 99 | 62 90.0 | 180 |.999999| 99 | 99 | 99 ---------------------------------------------------------------- So let's say you feed the value 179 degrees in to the Sine Converter. First the Sine Converter would determine that 179 is in the second quadrant because it is between 90 and 180 degrees. The SecondQuadrant converter would set the posneg flag to "1" to indicate a positive polarity for the answer. Then 180-179=1. We multiply this by 2 to convert the equivalent angle into an Offset Address of 2. The SecondQuadrant routine now jumps to the TableDecode routine. The Offset Address now points to the proper table element. We call a subroutine that looks at posneg and outputs either a + or a - symbol to indicate polarity. It also prints the decimal point (preceded by a "0" if desired). We access tablehigh and get back 0x01. We call a subroutine that sends "01" to the LCD. We access tablemiddle and get back 0x74. We call a subroutine that sends "74" to the LCD. We access tablelow and get back 0x52. We call a subroutine that sends "52" to the LCD. The above assumes you want the sine value displayed. If you instead want the sine value so it can be used with more math processing, then store the value in binary. Here is a simple method for generating the binary equivalents of numbers such as .017452: Bigger than or equal to 1/2? No .0 Bigger than or equal to 1/4? No .00 Bigger than or equal to 1/8? No .000 Bigger than or equal to 1/16? No .0000 Bigger than or equal to 1/32? No .00000 Bigger than or equal to 1/64? Yes .000001 .017452-.015625=.001827 Bigger than or equal to 1/128? No .0000010 Bigger than or equal to 1/256? No .00000100 (just finished high byte) Bigger than or equal to 1/512? No .00000100 0 (we just began middle) Bigger than or equal to 1/1024? Yes .00000100 01 .001827-.0009765625=.0008504375 Bigger than or equal to 1/2048? Yes .00000100 011 .0008504375-.00048828125=.00036215625 Bigger than or equal to 1/4096? Yes .00000100 0111 .00036215625-.000244140625=.000118015625 Bigger than or equal to 1/8192? No .00000100 01110 Continue this process until you have all 24 bits converted. Yes, you are right, this is a regular bun-buster of a thing to have to do for 180 values (to get 0 to 90 degrees in .5 degree increments). Only a fool would attempt to build such a table manually. Instead, write a simple program on your PC in BASIC, or C, or FOXPRO, or whatever language you like. Then use the program to compile the table values for you. Hint: the output is a "0" for No, and a "1" for Yes. When the answer is Yes you subtract and now use the result as your value to compare against. I use FOXPRO to do all this nitty gritty boring stuff, and I have it actually produce three ASCII files that make up the 3 tables of RETLW instructions (High, Middle, and Low). I Then append these files into my assembler at the proper places. Each line contains something like this: RETLW 00110101b where the "b" means that the number is in binary. Well, I would write some more, but I have some other things to get to right now. I hope that someone out there finds this useful. Is there any great desire out there for a series of tutorials on topics like binary addition, subtraction, division, multiplication, etc.? I could probably also write some stuff on interrupt routines and various other topics that are oriented towards the PIC way of doing things. I am willing to write some tutorial type stuff if it is useful to some and not obnoxious to those who are already experts. I should warn you though that I am more inclined to write about the HOW and WHY aspects and leave a complete and full program listing to the user. It is not just a matter of time, but my mindset as an educator. I am willing to teach and explain and show someone how to go about something. I usually won't hand my students a complete program to accomplish the task, because that is what I want *them* to learn how to do. My objective is to educate, not to be a source of finished product so that the student has nothing to do. I believe we learn by doing more than any other way. Fr. Tom McGahee {Original Message removed}

2000\02\04@165913 by andy howard

From: "Thomas McGahee" <tom_mcgaheeKILLspamEraseMESIGMAIS.COM> Subject: Lookup Tables: Sine Converter > Very often a lookup table or set of lookup tables is the simplest > solution. By using a little creativity you can greatly reduce the > size lookup table needed. <Interesting and pleasantly discursive discussion snipped> > I am willing to write some tutorial type stuff if it is useful to some > and not obnoxious to those who are already experts. Do please. Although familiar with the nuts'n'bolts of PICs and able to knock up code to do all the basics, when I see some of the elegant and/or concise code examples posted here I'm all to aware that there's a big gap in my meagre engineer's coding skills when it comes to programming philosophies, strategy etc. The science of programming as it were, rather than the engineering. {Quote hidden}> I should warn you though that I am more inclined to write about > the HOW and WHY aspects and leave a complete and full program > listing to the user. It is not just a matter of time, but my mindset > as an educator. > I am willing to teach and explain and show someone > how to go about something. I usually won't hand my students > a complete program to accomplish the task, because that is what > I want *them* to learn how to do. My objective is to educate, not > to be a source of finished product so that the student has nothing > to do. I believe we learn by doing more than any other way.

2000\02\04@172533 by andy howard

Original Message From: "Thomas McGahee" <EraseMEtom_mcgahee@spam@@spam@SIGMAIS.COM> Subject: Lookup Tables: Sine Converter > Very often a lookup table or set of lookup tables is the simplest > solution. By using a little creativity you can greatly reduce the > size lookup table needed. <Interesting and pleasantly discursive discussion snipped> > I am willing to write some tutorial type stuff if it is useful to some > and not obnoxious to those who are already experts. Do please! Though familiar with the nuts'n'bolts of PICs and able to knock up code to do all the basics, when I see some of the elegant and/or concise code examples posted by some people here I'm all too aware that there's a big gap in my meagre, self-taught engineer's coding knowlege when it comes to programming philosophies, strategy etc. > I should warn you though that I am more inclined to write about > the HOW and WHY aspects and leave a complete and full program > listing to the user. It is not just a matter of time, but my mindset > as an educator. The science of programming as it were, rather than the engineering. And maybe a touch of art too... Something many of the PIC books I've seen find too little space for, IMHO. > I am willing to teach and explain and show someone > how to go about something. I usually won't hand my students > a complete program to accomplish the task, because that is what > I want *them* to learn how to do. My objective is to educate, not > to be a source of finished product so that the student has nothing > to do. I believe we learn by doing more than any other way. They'll make an excellent learning resource as a website or ebook too. .

'Pic A/D sampling a biased sine wave'
2000\02\18@064824 by Lorick

I need to (hopefully) use the 16F876 to sample a sine wave that rides on a 1.75v reference line, swinging from 0v to 3.5v but before I even attempt this, I'm thinking it won't work because if an average value of the sine wave is what's taken over time, then wouldn't I always get a reading somewhere near that 1.75v reference line? Assuming a maximum swing signal, if a peak is measured in one instance in time at 3.5v, and then a trough is measured the next time at 0v, and then the middle of the signal is measured at 1.75v, the average of the 3 samples is 1.75v. What would be a proper method to sample a biased "ac" signal? Considering that the maximum voltage is 3.5v, I have ruled out the idea of rectifying the signal or something unless I can use a semiconductor junction that doesn't drop very much of that valuable signal level. I also considered looking into peak detectors. I don't know off hand what they really do but if they can be used to sort of sample and hold the most positive peaks of this sine wave, I could sample from that and then just compensate in software for the bias level to get a peak reading from the reference 1.75v. What other options are there?

2000\02\18@073722 by Ken Webster

Lorick wrote: {Quote hidden}>I need to (hopefully) use the 16F876 to sample a sine wave that rides on a >1.75v reference line, swinging from 0v to 3.5v but before I even attempt >this, I'm thinking it won't work because if an average value of the sine >wave is what's taken over time, then wouldn't I always get a reading >somewhere near that 1.75v reference line? > >Assuming a maximum swing signal, if a peak is measured in one instance in >time at 3.5v, and then a trough is measured the next time at 0v, and then >the middle of the signal is measured at 1.75v, the average of the 3 samples >is 1.75v. > >What would be a proper method to sample a biased "ac" signal? Considering >that the maximum voltage is 3.5v, I have ruled out the idea of rectifying >the signal or something unless I can use a semiconductor junction that >doesn't drop very much of that valuable signal level. I also considered >looking into peak detectors. I don't know off hand what they really do but >if they can be used to sort of sample and hold the most positive peaks of >this sine wave, I could sample from that and then just compensate in >software for the bias level to get a peak reading from the reference 1.75v. > >What other options are there? Depending on how fast and/or noisy the signal is and what response time you need, you may be able to sample with the A/D and take the max and min of your samples over a given interval. Then (max - min) would give your peak-to-peak voltage. Just be sure that you are sampling at a rate or phase that catches the peaks (sample much faster than the signal or phase-lock to it so that you can align your samples with the peaks). If you do phase-lock to the signal, you could take the average of the samples aligned with the peaks minus the average of the samples aligned with the troughs and thus reject noise much better (than taking max and min). Phase lock can be accomplished by also sampling at the zero crossings and using the difference between the two zero crossing points (rising edge and falling edge) as an error control signal for the loop. I used this technique to lock to the 60kHz WWVB carrier for a direct-sampling WWVB receiver I designed a few years ago based on a 16C73. It worked like a charm and could measure the carrier amplitude with a lot more precision than one might expect given an 8-bit A/D (the signal can bias the noise and produce a clearly measureable difference in the average even when the signal itself is smaller than one LSB worth of voltage on the A/D input). To get a high degree of precision with a noisy signal you will also need to apply a time-domain windowing function (like the windowing functions used for FFTs and serving exactly the same purpose). The only practical windowing function to use with fast signals on a PIC (because it can be done with a coupple of running sums and thus avoid time-consuming multiplications) is the triangle window. But that's probably more than you needed to know ... Anyway, there are lots of ways to measure the amplitude of a sinusoid with a PIC. The most appropriate method depends on how fast your signal is, how fast you want to measure its amplitude, how noisy it is, how much frequency modulation it has, how much precision you need, how much code (or hardware) you are willing to devote to the task, etc. Cheers, Ken

2000\02\18@083001 by Robert A. LaBudde

<x-flowed>At 06:47 AM 2/18/00 -0500, Lorick wrote: >I need to (hopefully) use the 16F876 to sample a sine wave that rides on a >1.75v reference line, swinging from 0v to 3.5v but before I even attempt >this, I'm thinking it won't work because if an average value of the sine >wave is what's taken over time, then wouldn't I always get a reading >somewhere near that 1.75v reference line? It's not clear what you are doing here. Is this supposed to be a simple RMS measurement, or you need details of the sine wave? In the latter case, you seem to be assuming the mean value (DC component) of the signal is 1.75 V precisely. Are you sure about this? If so, what's the problem? If you simply want to determine the average DC level, that's easy: just use a low-pass digital filter. Even a simple moving-average will work fine. What are the performance requirements? If you need a 'rectified' signal, do it in software: record and analyze samples separately for above and below 1.75 V. To rectify, difference from 1.75 V and flip the sign. If you're worried about clipping at 0 V, add a 'bias' with an op-amp summer before sampling. ================================================================ Robert A. LaBudde, PhD, PAS, Dpl. ACAFS e-mail: @spam@ralspamKILLspamlcfltd.com Least Cost Formulations, Ltd. URL: http://lcfltd.com/ 824 Timberlake Drive Tel: 757-467-0954 Virginia Beach, VA 23464-3239 Fax: 757-467-2947 "Vere scire est per causae scire" ================================================================ </x-flowed>

2000\02\18@083616 by Rich Leggitt

Hi Lorick, are you trying to measure the voltage? Assuming you can sample at many times the actual frequency (like 30), then something like this might work: int peak, trough; // averages are here // call for each sample taken void process(int sample) { static int prior, rising; // stuff to remember if (sample > prior) // voltage is rising { if (!rising) // was it before? { rising = 1; // no, note it is now trough += (prior-trough)/k; // prior sample is trough } } else // else, voltage is falling if (rising) // was it before? { rising = 0; // no, note it is now peak += (prior-peak)/k; // prior sample is peak } prior = sample; // remember prior sample } Hope this looks all pretty on your screen like it does on mine -- tabs suck. The 'peak += (sample-peak)/k' thing is the magic, it works exactly like an RC filter. k is adjusted for your sample rate and jitter, noise floor, etc., I'd start with 16 (rrf rrf rrf rrf, somebody let the dog out). Anyway, peak voltage is (peak-trough)/2, don't trust no stinkin' DC bias or the A/D either. No doubt there's a better way, and hopefully someone will post it here so I can swipe it. -- Rich On Fri, 18 Feb 2000, Lorick wrote: {Quote hidden}> Date: Fri, 18 Feb 2000 06:47:48 -0500 > From: Lorick <spamBeGonelorickRemoveMEEraseMEAIR.ON.CA> > Reply-To: pic microcontroller discussion list <RemoveMEPICLISTKILLspamRemoveMEMITVMA.MIT.EDU> > To: TakeThisOuTPICLISTMITVMA.MIT.EDU > Subject: Pic A/D sampling a biased sine wave > > I need to (hopefully) use the 16F876 to sample a sine wave that rides on a > 1.75v reference line, swinging from 0v to 3.5v but before I even attempt > this, I'm thinking it won't work because if an average value of the sine > wave is what's taken over time, then wouldn't I always get a reading > somewhere near that 1.75v reference line? > > Assuming a maximum swing signal, if a peak is measured in one instance in > time at 3.5v, and then a trough is measured the next time at 0v, and then > the middle of the signal is measured at 1.75v, the average of the 3 samples > is 1.75v. > > What would be a proper method to sample a biased "ac" signal? Considering > that the maximum voltage is 3.5v, I have ruled out the idea of rectifying > the signal or something unless I can use a semiconductor junction that > doesn't drop very much of that valuable signal level. I also considered > looking into peak detectors. I don't know off hand what they really do but > if they can be used to sort of sample and hold the most positive peaks of > this sine wave, I could sample from that and then just compensate in > software for the bias level to get a peak reading from the reference 1.75v. > > What other options are there? >

2000\02\18@093921 by Lorick

> It's not clear what you are doing here. Is this supposed to be a simple RMS > measurement, or you need details of the sine wave? > > In the latter case, you seem to be assuming the mean value (DC component) > of the signal is 1.75 V precisely. Are you sure about this? If so, what's > the problem? I'm not actually sure if I would be looking to make measurements of the peaks or if an rms reading would be fine, but what I meant by 1.75v average is that since that is the bias level, if I accidently took successive readings at the peaks and troughs instead of just at all the peaks, the overall average would look like 1.75v which would be wrong, it would be the "0" level of the signal... I am assuming that what I am after is a way to measure the peaks but here is the whole description if it will help. The signal I am working with is from an electret condenser microphone that will be picking up a definite 1KHz audio signal from some test oscillator circuit and speaker. I am interested in detecting the volume level of the sound source so that I can tell how far away it is based on how strong the signal is. The mic output will be signal conditioned to swing from 0 to 3.5v (working from single supply op amps) and I will be reading this volume level with the a/d and I was trying to fiure out how the a/d would perceive the audio. If it reads a level of 2v, how would it know if this is a maximum of the current waveform, or if it's the normalized zero crossing point, thus it should be interpreted as zero instead of two in terms of amplitude. The pic is being run at 4MHz so far but that's just arbitrary, so I am assuming this means I can sample WAY above the 1KHz signal frequency. Now I need to know exactly what it is I should be looking for to get the "volume" level of the signal...

2000\02\18@094355 by Michael Rigby-Jones

part 0 2603 bytes I need to (hopefully) use the 16F876 to sample a sine wave that rides on a 1.75v reference line, swinging from 0v to 3.5v but before I even attempt this, I'm thinking it won't work because if an average value of the sine wave is what's taken over time, then wouldn't I always get a reading somewhere near that 1.75v reference line? Assuming a maximum swing signal, if a peak is measured in one instance in time at 3.5v, and then a trough is measured the next time at 0v, and then the middle of the signal is measured at 1.75v, the average of the 3 samples is 1.75v. What would be a proper method to sample a biased "ac" signal?  Considering that the maximum voltage is 3.5v, I have ruled out the idea of rectifying the signal or something unless I can use a semiconductor junction that doesn't drop very much of that valuable signal level. </UL> To rectify signal under these conditions you use something called a "precision rectifier".  This scheme uses an op-amp with a diode in the feedback path, so the 0.7 volt odd voltage drop is canceled.  You can even get precision rectifiers with True RMS detection built in, e.g <A HREF="http://www2.national.com/ms/LB/LB-25.pdf" TARGET="_blank">http://www2.national.com/ms/LB/LB-25.pdf</A> Hope this helps Mike </BODY> </HTML> </x-html>

2000\02\18@102431 by Wagner Lipnharski

,,, so if you don't need to measure the absolute millivolts of the senoidal signal captured by the mic, what counts for you is which signal (circuit + speaker) made the max signal... In this situation I would install a simple 1N4148 diode integrating the signal over a cap + resistor and that's all. Wagner

2000\02\18@114949 by Martin McCormick

You might get a pair of resistors of equal value between VCC and Common to bias the A/D at its exact midpoint. The path between the biased sine wave signal and this junction between the A/D converter and the resistors would be made with a DC blocking capacitor so the DC bias on the audio would no longer matter. The resistors should be low enough in value such that the sample-and-hold of the A/D converter will not load the circuit. The blocking cap will need to be large enough that its reactence is not a factor in the frequency range to be used. The output of the A/D converter would read either 0x7F or 0x80 so your 0 crossing would be one or the other of those two values. As long as it doesn't drift much with temperature, you should be able to resolve a sine wave 127 counts in one direction and 128 counts in the other. Martin McCormick

2000\02\18@122618 by Harold M Hallikainen

What are you trying to accomplish, and what is your sample rate? If the sample rate is at least twice the frequency of the sine wave, you can get enough data to accurately reproduce the sine wave. As you point out, the average will indeed by 1.75V. Averaging is the same as running the incoming signal thru a low pass filter. If you want the RMS (which will include the RMS of the DC), square the samples as they come in, accumulate a sum of squares (I accumulate a 16 bit sum of 256 8 bit samples), divide by the number of samples (that's why I use 256! It's easy!), then take the square root using one of the recently posted algorithms. That'll get you RMS. If you want peak to peak, set up a RAM location for minimum and maximum, then sample thru the cycle looking for min and max. Subtract and there's the peak to peak. So... again, it comes down to what are you looking for (wasn't that a line from "Third Rock from the Sun" where theyre spying on a neighbor and one of them says "I wonder what he's looking for?"). Harold On Fri, 18 Feb 2000 06:47:48 -0500 Lorick <spamBeGonelorickKILLspamTakeThisOuTAIR.ON.CA> writes: {Quote hidden}> I need to (hopefully) use the 16F876 to sample a sine wave that rides > on a > 1.75v reference line, swinging from 0v to 3.5v but before I even > attempt > this, I'm thinking it won't work because if an average value of the > sine > wave is what's taken over time, then wouldn't I always get a reading > somewhere near that 1.75v reference line? > > Assuming a maximum swing signal, if a peak is measured in one > instance in > time at 3.5v, and then a trough is measured the next time at 0v, and > then > the middle of the signal is measured at 1.75v, the average of the 3 > samples > is 1.75v. > > What would be a proper method to sample a biased "ac" signal? > Considering > that the maximum voltage is 3.5v, I have ruled out the idea of > rectifying > the signal or something unless I can use a semiconductor junction > that > doesn't drop very much of that valuable signal level. I also > considered > looking into peak detectors. I don't know off hand what they really > do but > if they can be used to sort of sample and hold the most positive > peaks of > this sine wave, I could sample from that and then just compensate in > software for the bias level to get a peak reading from the reference > 1.75v. > > What other options are there? FCC Rules Online at http://hallikainen.com/FccRules Lighting control for theatre and television at http://www.dovesystems.com ________________________________________________________________ YOU'RE PAYING TOO MUCH FOR THE INTERNET! Juno now offers FREE Internet Access! Try it today - there's no risk! For your FREE software, visit: dl.http://www.juno.com/get/tagj.

2000\02\18@123641 by Robert A. LaBudde

<x-flowed>At 09:40 AM 2/18/00 -0500, Lorick wrote: >The signal I am working with is from an electret condenser microphone that >will be picking up a definite 1KHz audio signal from some test oscillator >circuit and speaker. I am interested in detecting the volume level of the >sound source so that I can tell how far away it is based on how strong the >signal is. The mic output will be signal conditioned to swing from 0 to >3.5v (working from single supply op amps) and I will be reading this volume >level with the a/d and I was trying to fiure out how the a/d would perceive >the audio. If it reads a level of 2v, how would it know if this is a >maximum of the current waveform, or if it's the normalized zero crossing >point, thus it should be interpreted as zero instead of two in terms of >amplitude. If you are only interested in the power at a single frequency (1 kHz), you can solve your problem with a hardware active bandpass filter @ 1 kHz, or a software digital bandpass filter centered at the same frequency. If you aren't sure about an exact frequency, a simple rectifier and integrator (either passive or active) will provide a smoothed input to the PIC. You can also do it in software by absolute value + moving average. ================================================================ Robert A. LaBudde, PhD, PAS, Dpl. ACAFS e-mail: EraseMEral.....KILLspamlcfltd.com Least Cost Formulations, Ltd. URL: http://lcfltd.com/ 824 Timberlake Drive Tel: 757-467-0954 Virginia Beach, VA 23464-3239 Fax: 757-467-2947 "Vere scire est per causae scire" ================================================================ </x-flowed>

2000\02\18@183045 by Ken Webster

Lorick wrote: >The signal I am working with is from an electret condenser microphone that >will be picking up a definite 1KHz audio signal from some test oscillator >circuit and speaker. I am interested in detecting the volume level of the >sound source so that I can tell how far away it is based on how strong the >signal is. The mic output will be signal conditioned to swing from 0 to (snip) >assuming this means I can sample WAY above the 1KHz signal frequency. Now I >need to know exactly what it is I should be looking for to get the "volume" >level of the signal... The most precise measurement you could achieve in this case would be using a software PLL (digital signal processing). The signal from the microphone will probably have a lot of background noise. A software PLL combined with an integrator to average the peaks and troughs of the signal (multiplied by a time-domain windowing function) over a selected interval will give you the same results as analysing a single band (the 1kHz band in this case) of a discrete (fast) Fourier transform (FFT) performed using the same windowing function and sampling rate. In other words, this technique will give you an amplitude measurement of an extremely narrow band (bandwidth depends on how long an interval you integrate over) centered on your 1kHz signal. Any noise that is not precisely at 1kHz will be ignored. It is not the simplest technique but it can be implemented on a PIC with plenty of room and processing time to spare (I used this technique on a PIC16C73 to measure a 60kHz signal -- 1kHz should be a piece of cake! I would be hapy to give you tips, pointers, and sample code if you are interested). The results are definitely worth the effort if you want a precise measurement. Cheers, Ken

2000\02\19@155523 by paulb

Lorick wrote: > I need to (hopefully) use the 16F876 to sample a sine wave that rides > on a 1.75v reference line, swinging from 0v to 3.5v but before I even > attempt this, I'm thinking it won't work because if an average value > of the sine wave is what's taken over time, then wouldn't I always get > a reading somewhere near that 1.75v reference line? Yes. The *average* value of any pure AC signal is zero, as it spends equal areas under the signal curve in either direction. So if you feed your signal into a low-pass filter, you will get the same voltage with and without signal. Although you say the reference line is 1.75V, it would probably be as well to implement an averaging algorithm to determine exactly what this is as read by the ADC. You probably want the standard FIR (Finite Impulse Response) filter function of adding the current sample to a 16- bit value and subtracting the "current average" which is 1/256 of the current tally. Of course, 1/256 of the current tally is merely the high byte of the tally plus the high bit of the low byte added (or in this case, both are being subtracted from the tally itself) for rounding correction. Now that you know what the "zero level" is, you would prefer to sum the squares of the instantaneous deviation from this level to measure your signal level as a MS (Mean Squares - it's a waste of time to take the square root again as the MS value is a perfectly good indicator). This however is messy as it involves multiplications and at least 24- bit accumulation, presuming you again use a FIR filter to smooth it. Instead, just sum the modulo of the deviation from the average (do the subtract, test the result and negate if presently negative) using the same averaging procedure. For a 1 kHz frequency, sample at approximately 16 k samples per second to get 16 points per cycle and all the mathematics should be easy on a 4 MHz crystal PIC. This method still measures signal plus noise. If you want to filter out the signal at a (one only) precise frequency of 1 kHz, then you need a different technique. -- Cheers, Paul B.

2000\02\20@133303 by Ken Webster

>I would be interested in more information on this technique and some pic >code to implement it. Does this technique >allow an input signal of wide bandwidth and then discriminate for the center >frequency of choice or does it assume > a pre filtered limited bandwidth? It allows a wide bandwidth. Like any technique that samples a signal it is susceptable to aliasing. It is thus best to low-pass-filter the input so that it does not excede the Nyquist frequency (one-half of the sample rate). If you have some processing time to spare I would recommend sampling at a high rate (20-40 ksps). This way, a cheap RC lowpass filter can be used outside of the PIC and still provide plenty of attenuation at the Nyquist frequency. If you sample slower you may want to consider using an active filter with a steeper cutoff or perhaps an LC bandpass filter. The code I have is written to deal with a 60kHz signal. I would do a few things differently for a 1kHz signal. For a 1kHz signal I would recommend the following: 1). Use CCP2 with TMR1 to generate the "special event" trigger that starts an A/D conversion (every 50 microseconds to sample at 20kHz, 25 microseconds for 40kHz). Write an ISR to receive the A/D value and process it. Here's a chunk of code from my WWVB receiver that sets up the timer and A/D interrupt: movlw 1 movwf t1con ;timer1 internal clk, no prescale movlw 149 movwf ccpr2l ;150 clk cycles (tosc/4) clrf ccpr2h ; = 37.5us with 16MHz osc movlw 0x0b movwf ccp2con ;enable ccp2 as comparator ;CCP2 generates the special event trigger which ;clears timer1 and starts an A/D conversion bsf status,rp0 movlw 2 movwf adcon1 ;RA0..3 analog inputs, Ref = Vdd bsf pie1,adie ;enable A/D peripheral interrupt bcf status,rp0 movlw 0x81 movwf adcon0 ;a/d clk = fosc/32, select ch 0 bcf pir1,adif ;clr A/D int flag bsf intcon,peie ;enable peripheral interrupt bsf intcon,gie ;enable interrupts 2). Lowpass filter the A/D signal at a cutoff frequency just above 1kHz. The filter should produce enough attenuation at 3kHz to reduce aliasing to acceptable levels (the first harmonic that will be aliased to 1kHz is at 3kHz). A really simple lowpass filter stage (doing something similar to an analog RC filter) can be achieved by the following: out = out + ((in - out) / n) Since the processing time per stage is so low (if n is a power of 2 and can be accomplished by a simple right-shift), several such stages can be cascaded to achieve the desired cutoff slope. I'm not sure what the formula is to relate n to a cutoff frequency -- I usually just write a C program to calculate the results and then experiment with n (choosing powers of 2 or other easily calculatable values) until I get the results I want. 3). Resample the lowpass-filtered signal at 4ksps (i.e. pick out every 5th value if your original sample rate was 20ksps or every 10th value if your original sample rate was 40ksps). 4). Add the resampled values sequentially into 4 different accumulators. The accumulators will correspond (if you use a PLL) to the rising zero crossing, the positive peak, the falling zero crossing, and the negative peak of the 1kHz signal. If you don't use a PLL, you can get a complex (phasor representation) signal measurement as follows: r = (ACC2 - ACC4) / n j = (ACC1 - ACC3) / n Where n is the number of samples in the accumulators. The measured amplitude is simply sqrt(r*r + j*j). Every so often (how long depends on how much frequency selectivity you need and how often you need to get an amplitude measurement), you take the results as shown above and then zero the accumulators. You may find that this alone produces adequate results (no PLL and a simple rectangular windowing function). The longer you add samples into the accumulators before using the average and zeroing the accumulators, the more selective the frequency response will be. If you average samples for too long (long enough for the phase of the 1kHz signal to drift a substantial amount relative to the sampling phase) then you will need to use a PLL to precisely match the frequency of the signal. A PLL can be implemented by adding or subtracting clock cycles from the A/D sampling timer (by adjusting CCP2) so that each set of samples that go into ACC1 and ACC3 are as close to the zero-crossing as possible. If these samples were exactly on the zero-crossing then S1-S3 would be zero (where S1 is the sample that gets added into ACC1 and S3 is the sample that gets added into ACC3). If the sampling points were late then S1-S3 would be positive. If the sampling points were early then S1-S3 would be negative. Thus, S1-S3 can be used as an error measurement to steer the phase of the A/D sampling. If you are interested in doing this I could provide an example (but it is pretty long and difficult to explain). For really good noise rejection you may want to use a PLL along with a windowing function to improve the frequency selectivity. For more information about windowing functions, see the following: www.spd.eee.strath.ac.uk/~interact/moved/dft/window.html www.engr.sjsu.edu/~knapp/HCIROD3D/3D_phys/Fou_note.htm http://www.cooper.edu/~donahu/auFilter/auFilter.html A triangle (Bartlett) window can be calculated using a coupple of running sums and taking the difference for the ramp up and just using a running sum for the ramp down. At the slow 4ksps rate, other windowing functions requiring multiplications are also possible. I hope this helps and isn't overly confusing. Cheers, Ken

'Magic Sinewaves demystified'
2000\03\05@114431 by Scott Dattalo

Well, at least partially... It finally dawned on me yesterday how magic sine waves can be sythesized. What's frustrating, is that the answer is so simple that it's obvious; so obvious that it's been overlooked in the discussions we had off and on for the last couple of years (at least by me). Lancaster is probably a chuckling lurker :). There are probably a few ways to go about it, but this is the one I tried. Imagine how you'd go about synthesizing a sine wave if you had an A/D converter. The most common way is to produce `N' samples for one sine wave cycle and repeatedly write these to an A/D converter. The sine wave frequency is determined by the rate at which the samples are written to the A/D converter. Now imagine replacing the A/D converter with a PWM generator that has the same resolution. So for example, if you had a 12-bit A/D converter the PWM generator would be capable of producing 4095 (2^12 - 1) different duty cycles. Theoretically, the low-pass filtered output of the A/D converter and the PWM are the same. Practically however, it's difficult to design a filter that would have the proper cut-off characteristics. So to ease the filter requirements, the PWM synthesized sine wave is usually made to be a much higher frequency than the frequency at which samples are written to the A/D converter. Now when we, or at least I, begin to think about `magic sine waves' the idea that the pwm frequency can be reduced is forgotten. Furthermore, we constrain ourselves to the low-resolution 8 or 10 bit pwm generators that are available in the various pic's. What the magic sine wave algorithm does, AFAICT, is find a suitable time discretization for a high resolution PWM synthesized sine wave. So, there are two parts to the synthesis. Part 1 invovles finding the high resolution PWM representation of the sine wave. Part 2 involves finding an optimum way to make time discrete. The first part is easy and the attached octave (or matlab) program will do this. The second part can be determined using brute force search techniques. I don't have an elegant solution though, but hopefully some others have some insight. Scott PS, this octave program is meant to be tool and otherwise not too useful (in other words, I don't bother to explain hardly anything...) % `magic'sine wave synthesis % The purpose of this program is to find an optimum pulse stream % that when low pass filtered will produce a sine wave. The optimization % criterion is to minimize the harmonics for a given number of pulses. % by T. Scott Dattalo 04MAR00 clear; clg; pulses = 32; T = 1; % Period f = 1/T; % fundamental frequency tp = T/pulses; % Time window for each pulse t = [0:(pulses-1)]/pulses; sine_samples = sin(2*pi*f * t); ts = zeros(1,4*(pulses-1)); p = ts; % create a pwm sine wave with edges based on floating point % numbers: for i=1:pulses ts(4*i) = t(i); ts(4*i+1) = t(i); ts(4*i+2) = t(i) + (1 + sine_samples(i))/2 * tp + 1000*eps; ts(4*i+3) = ts(4*i+2); p(4*i+0) = 0; p(4*i+1) = 1; p(4*i+2) = 1; p(4*i+3) = 0; end % Now calculate the harmonic content harmonics = 20*pulses; mag = zeros(1:harmonics); ph = zeros(1:harmonics); phase = [0:(pulses-1)] * T/(pulses); N = 1000; %Number of samples sqwave = zeros(1:N); u = [1:N]/N * T; %Time vector for n=1:harmonics w = n*2*pi/T; m=2; for m=1:(pulses) tau = (1 + sine_samples(m))/2*tp; d = tau/T; sqwave = sqwave + 2*sin(n*pi*d)/(n*pi)*cos(w*(u-tau/2-phase(m))); mag(n) = mag(n) + 2*sin(n*pi*d)/(n*pi)*cos(w*phase(m)); ph(n) = ph(n) + cos(w*phase(m)); end end sqwave = sqwave + 0.5 + 2; dc = sum(sine_samples); mag(2) = 0; ti = sprintf('PWM Synthesis'); title(ti) subplot(111) %plot(t,sine_samples) %subplot(212) % plot the harmonics: plot([0:harmonics-1]/pulses,abs(mag)) % uncomment to plot the square wave train re-synthesized from its % harmonics %plot(ts,p,u,sqwave)

2000\03\06@072604 by Andrew Kunz

Scott, You mean D/A converter, not A/D, right? Otherwise, I'm confused. Andy

2000\03\06@083018 by Scott Dattalo

On Mon, 6 Mar 2000, Andrew Kunz wrote: > You mean D/A converter, not A/D, right? Otherwise, I'm confused. D/A of course - sorry. The octave (matlab) code posted yesterday had a bug in the harmonic calculation. That's been fixed. In addition, I wrote another similar octave program that will analyze an ascii string of 0's and 1's. Like the other program, it plots the square wave that has been reconstructed from its harmonics and it plots the magnitudes of the first few harmonics. Here's a screen shot of a 12-bit string, "010110100000", produced by the program: http://www.dattalo.com/ms1.gif and the programs: http://www.dattalo.com/ms.m http://www.dattalo.com/ms2.m Scott

'[OT] Contracting business'
2000\03\06@095436 by Juimiin Hong

Hi, Recently the idea of becoming a contractor has crossed my mind. For all the contractors out there, how did you start your business? Where did you find your customers? Thanks, Juimiin

'Magic Sinewaves demystified'
2000\03\06@115039 by Grif\ w. keith griffith

<x-flowed>At 10:44 AM 3/5/00 -0600, you wrote.. >It finally dawned on me yesterday how magic sine waves can be sythesized. Major math snip::: Scott: doesn't your code produce an analog output? I was of the opinion the whole reason for the Magic part of the synthesis was using saturated switching at a much higher freq than the final output and stuffing it thru a low pass filter, getting a somewhat usable, higher power sine wave output? Or did I just learn something this early in the week??? ( that would be good,,, then I wouldn't have to work as hard the rest of the week!) 'Grif' N7IVS </x-flowed>

'[OT] Contracting business'
2000\03\06@115301 by Chris Eddy

#1: Don't quit your day job until your side work is so heavy you can barely keep up. #2: You really should be geared towards business.. dealing with late bills, people skills, negotiating, etc... things that most engineers are not. #3: Your first year or two will be thin on income. REAL thin. Unless you have high quality business lined up already. #4: Don't EVER quit on a project. Word travels fast in our high tech community. Eat the loss if you must. Juimiin Hong wrote: > Hi, > > Recently the idea of becoming a contractor has crossed my mind. For all > the contractors out there, how did you start your business? Where did you > find your customers? > > Thanks, > Juimiin

'Magic Sinewaves demystified'
2000\03\06@120514 by jamesnewton

What language are these programs written in? I am flagging these emails for follow up in a week or so (I'm crunched at the moment) but I'm interested in a general purpose magic sinwave routine code generator for the PIC. --- James Newton spamjamesnewtongeocities.com 1-619-652-0593 http://techref.massmind.org NEW! FINALLY A REAL NAME! Members can add private/public comments/pages ($0 TANSTAAFL web hosting) {Original Message removed}

'[OT] Contracting business'
2000\03\06@124743 by Andrew Kunz

>#3: Your first year or two will be thin on income. REAL thin. Unless you have high quality business lined up already. Sometimes other years, too. Been there, done that. Chris' other advice is good. Personally, I found that if I ever want to go out on my own again, I will have a SALES FORCE who can do the initial contact for me. FWIW, my local FAI guy has been really good about this. But only because I've brought several $10K+ customers (repeaters, not one-time-buys) to him. And be LOYAL to the guy, whoever it is. Never say anything bad about other customers to anybody (even your wife), no matter what jerks they are. We all get to act that way from time to time; be as forgetful of their inadequacies as you'd like them to be of yours. NEVER EVER EVER tell them a lie. "It's almost ready." etc. Andy

'Magic Sinewaves demystified'
2000\03\06@132106 by Scott Dattalo

On Mon, 6 Mar 2000, "Grif" w. keith griffith wrote: > At 10:44 AM 3/5/00 -0600, you wrote.. > >It finally dawned on me yesterday how magic sine waves can be sythesized. > > Major math snip::: > > Scott: doesn't your code produce an analog output? I was of the opinion > the whole reason for the Magic part of the synthesis was using saturated > switching at a much higher freq than the final output and stuffing it thru > a low pass filter, getting a somewhat usable, higher power sine wave > output? Or did I just learn something this early in the week??? ( that > would be good,,, then I wouldn't have to work as hard the rest of the week!) True, the ultimate objective is to produce an analog output. However, the goal with the so called 'magic sinewaves' is to find the MINIMUM number of switch transitions. The resulting bit stream is filtered such that a sine wave is produced. The reason for minimizing the transitions is because some switches, particularly those associated with high power applications like in AC inverters, require a significant amount of energy to switch or cannot be efficiently switched at high frequencies. As an example, you may have a sine wave synthesis circuit where you want to drive the gate of a MOSFET directly from a pic I/O pin. Pic's have fairly beefy outputs, however they're very weak when compared to real MOSFET drivers. But given enough time, the PIC I/O can switch a MOSFET (assuming of course that Voh is above Vgs threshold - i.e. logic level Vgs MOSFETs may be required). If you attempted to drive the MOSFET at too high of frequency with this pic I/O pin, then chances are that it will never switch! In addition, if you can use harmonic cancellation, then the constraints on the analog filter may be relaxed. PWM generated waveforms relax the filter constraints by maximizing the switching frequency. If the PWM carrier is 100 times the sinewave you're trying to generate, then you're analog filter is much easier to design. There aren't any harmonics (to an approximation) between the sine wave frequency you're generating and the carrier frequency of the PWM. Magic sine waves explicitly remove these harmonics by strategically placing the pulses (or edges). The pulses, when viewed individually, contain harmonics at the fundamental (of the sine wave being generated) and at all integer multiples above. But when the pulses are taken together, the 2nd, 3rd and so on harmonics are cancelled. The number of harmonics cancelled depends on the quantization of time for the edges. For example, in the previous post, I had a graph of a 12-bit pulse stream 010110100000. The second and third harmonics were exactly cancelled, but there is a 4th and 5th harmonic present. Doubling the number of pulses to 24 allows a pulse stream to be created that will cancel the 4th harmonic. In general, if you have a pulse stream containing n! pulses, then it's possible to exactly cancel the first n harmonics. However, n! grows rather quickly. It's simply not practical to attempt to cancel the first 10 harmonics (after the fundamental) by creating a pulse stream containing 10! or 3,628,800 pulses. Consequently, if you wish to cancel the first 10 harmonics but with a pulse stream containing only a 1000 pulses, you're not going to achieve your goal. However, you may discover a particular stream that exactly cancels the 2nd, 3rd, and 4th, and attenuates the 5th through 10th harmonics by 20dB. From that you may design an appropriate analog filter. AFAIK, there is no way to apriori determine the appropriate length of the pulse stream to achieve some required harmonic profile. I think Lancaster's contribution to this technology is to discover certain lengths that perform well - or at least better than pulses streams that are slightly shorter or longer. What I'm playing around with is a way you can study the characteristics of one stream versus another. Scott

'[OT] Contracting business'
2000\03\06@140948 by mike

> >#4: Don't EVER quit on a project. Word travels fast in our high tech >community. Eat the loss if you must. To expand on this.... Don't take on a job unless you KNOW you can do a good job of it, and if there are potential problems areas, make sure the customer knows this right from the start. In general, deal with people the way you'd like to be dealt with. If you do a good job you will get repeat business.

'Magic Sinewaves demystified'
2000\03\06@163855 by Wagner Lipnharski

Considering the sine wave PWM generated, I wonder about the totally unpredictable harmonics from the combination of several different pulse width along the stream. Square waves are terrible to deal to avoid harmonics, except when you know exactly the fundamental fo's and the harmonics composition, otherwise, as a sine wave generator with variable frequency, it is a real pain. The most comprehensible approach to avoid them is using PWM cycles in the order of several MHz with prime numbers as the base for level control, so they will not repeat or cause alliasing. Of course, a high "Q" active filter would be the best solution to avoid harmonics, but again, or the fo is fixed or the active filter is programmable too, what increase the cost. In this case, the D/A solution (16 bits at least) turns to be much more economically attractive. There is another technique that I call "Rolling Triangles" (nothing to do with Rolling Stones...:), that is; A triangle face is flat, as flat is a digital signal, or a linear ramp generated by a constant current circuit. Rolling a Triangle, means that your program would set the triangle face angle according to the sine value, or signal rotational angle, and this is done by controlling charging current. Suppose you slice a 90¡ angle in 90 triangles, equally distributed in rotation, 1¡ per triangle, you would have the first quadrant of your sine-wave. To reduce the abrupt angle change between one triangle and another, several other small triangles are inserted at the transition, you can think as interpolation. I use to say that the biggest mistake people do when looking at a sine wave, is thinking about sine(x) functions, because it is related to a flat point of view of who is laid on the floor. According to the wheel point of view, it is just a plain and pure rotation, so the triangle explanation makes effect. The interaction of those triangles with a common panel (the floor, and the witness eyesight) we are able to see only a bi-plane sinewave, when other polarization in different dimensions are not allowed, as the magnetic and electric fields modulating a base electrical signal. It wouldn't be nice to be able to see this multidimensional sine wave interacting with several other signals? I first used the expression "Rolling Triangles" when produced a 3 phase 60Hz sine-wave, based on programmable constant current generators, and high speed processors integrating more than 50 thousands small triangles per cycle. I was never able to do it with PWM, even with oscillators close to MHz. Later on I found out that changing the 60¡ angle of each triangle vertice and connect thousands in a net style, it was possible to draw and plot practically *any* 3 dimensional electrical surface representation. Isn't interesting how many different ways you can do the same thing? I use to say that there are several ways to skin a cat, you can really be a winner if you end up with very few scratches... Wagner. Scott Dattalo wrote: [some snip below] {Quote hidden}> In addition, if you can use harmonic cancellation, then the constraints on > the analog filter may be relaxed. PWM generated waveforms relax the filter > constraints by maximizing the switching frequency. If the PWM carrier is > 100 times the sinewave you're trying to generate, then you're analog > filter is much easier to design. There aren't any harmonics (to an > approximation) between the sine wave frequency you're generating and the > carrier frequency of the PWM. > > Magic sine waves explicitly remove these harmonics by strategically > placing the pulses (or edges). The pulses, when viewed individually, > contain harmonics at the fundamental (of the sine wave being generated) > and at all integer multiples above. But when the pulses are taken > together, the 2nd, 3rd and so on harmonics are cancelled. The number of > harmonics cancelled depends on the quantization of time for the edges. For > example, in the previous post, I had a graph of a 12-bit pulse stream > 010110100000. The second and third harmonics were exactly cancelled, but > there is a 4th and 5th harmonic present. Doubling the number of pulses to > 24 allows a pulse stream to be created that will cancel the 4th harmonic. > In general, if you have a pulse stream containing n! pulses, then it's > possible to exactly cancel the first n harmonics. However, n! grows rather > quickly. It's simply not practical to attempt to cancel the first 10 > harmonics (after the fundamental) by creating a pulse stream containing > 10! or 3,628,800 pulses. > > Consequently, if you wish to cancel the first 10 harmonics but with a > pulse stream containing only a 1000 pulses, you're not going to achieve > your goal. However, you may discover a particular stream that exactly > cancels the 2nd, 3rd, and 4th, and attenuates the 5th through 10th > harmonics by 20dB. From that you may design an appropriate analog filter.

2000\03\06@165758 by David VanHorn

<x-flowed>At 04:37 PM 3/6/00 -0500, you wrote: >Considering the sine wave PWM generated, I wonder about the totally >unpredictable harmonics from the combination of several different pulse >width along the stream. Square waves are terrible to deal to avoid >harmonics, except when you know exactly the fundamental fo's and the >harmonics composition, otherwise, as a sine wave generator with variable >frequency, it is a real pain. The most comprehensible approach to avoid >them is using PWM cycles in the order of several MHz with prime numbers >as the base for level control, so they will not repeat or cause >alliasing. If you used a boxcar integrator, clocked at 2X, then you could get rid of all the evens anyway. I'm still looking for a way to nuke the odds. (I know it's all odds in your app) </x-flowed>

2000\03\06@172323 by David VanHorn

<x-flowed>Actually, I am looking for a generalized technique that will let me strip odd harmonics. I have an application that's using the boxcar as I described, both to indicate distortion level, and to subtract said distortion from a signal, but the odd harmonics have me stumped. I use one boxcar to take the signal, but of course it also passes the 2H, 3H, 4H... right on up, though past the 4th it's pretty well damped since it's an eight cap boxcar :) The second grabs the 2h and 4h components perfectly, but I have no simple answer for 3h and 5h. (Maybe I only need 3h, but you get the idea.) </x-flowed>

2000\03\06@203311 by Scott Dattalo

On Mon, 6 Mar 2000, David VanHorn wrote: > Actually, I am looking for a generalized technique that will let me strip > odd harmonics. > I have an application that's using the boxcar as I described, both to > indicate distortion level, and to subtract said distortion from a signal, > but the odd harmonics have me stumped. > > I use one boxcar to take the signal, but of course it also passes the 2H, > 3H, 4H... right on up, though past the 4th it's pretty well damped since > it's an eight cap boxcar :) The second grabs the 2h and 4h components > perfectly, but I have no simple answer for 3h and 5h. (Maybe I only need > 3h, but you get the idea.) If you had a square wave with a 50% duty cycle, there are no even harmonics. Furthermore, the relative magnitude of the harmonics are inversely proportional to their harmonic numbers. If you vary the duty cycle to say 33%, you'll have no 3rd,6th, 9th harmonic, but some of the evens will pop up. I don't think there's a way to arbitrarily kill the odd harmonics. But you can do (at least) two things: vary the pulse width of individual pulses and vary the number of pulses. The plot: http://www.dattalo.com/ms1.gif cancels the 2nd and 3rd harmonics. You could say that it consists of three pulses; one twice as wide as the other. (But the program generating it had 4 bits for those 3 pulses...). I don't have time right now to explain, but as I mentioned earlier, it's possible to exactly cancel the first N harmonics. However, the number of 'time slots' or quantized positions at which the pulse edges may occur is of the order of N! (factorial). There is technique that involves examining the phases of the harmonics of the individual pulses. I found that pulses can be collected into groups that totally cancel one another at a given harmonic number. The smallest group I experimented with is just a pair. For example, for the second harmonic, the pulses are grouped in pairs. One pulse is 180 degrees out of phase with the other. The pairs for the second harmonic are two pulses separated by one pulse. For the third harmonic, the pulses are separated by two pulses. 2nd harmonic suppressed: 010100000000 000010100000 3rd harmonic suppressed: 010010000000 000100100000 Total pulse stream: 010110100000 So the objective is to discover the pairs that cancel at a given harmonic and then to find a set of pulses that will have complete coverage for each harmonic. I've done this for 24 pulses, but beyond that it gets pretty hairy... Scott

'Software RMS measurement of sine waves again'
2000\03\06@205841 by Lorick

Weeks ago I was looking at ways to determine the RMS value of a sampled waveform with the 16F876 and several suggestions were to take 256 measurements (for easy division later) and accumulate the sum of squares, divide by number of samples, and square root to get the RMS value of the readings. Now I would like to know why this works....what is the mathematical relationship between squaring the readings and obtaining the RMS in the end? I know it's almost stupid to ask since we're talking about Root Mean Square to begin with....but I am indeed missing the in between steps in the story...

2000\03\06@213005 by Jeff Davis

On Mon, 6 Mar 2000, Lorick wrote: > Now I would like to know why this works....what is the mathematical > relationship between squaring the readings and obtaining the RMS in the end? > > I know it's almost stupid to ask since we're talking about Root Mean Square > to begin with....but I am indeed missing the in between steps in the > story... What squaring the values does is effectively eliminates the sign of the number (+/-). If you were to simply take the average of all measurements of a sine wave you would get zero. An RMS value is the average magnitude of each reading without taking into account the sign of the reading. Jeff

2000\03\06@221825 by Jeffrey D Spears

Think about what you are trying to get: The Root of the Mean of all the Squares -> Root Mean Square -> RMS by definition! On Mon, 6 Mar 2000, Lorick wrote: {Quote hidden}> Weeks ago I was looking at ways to determine the RMS value of a sampled > waveform with the 16F876 and several suggestions were to take 256 > measurements (for easy division later) and accumulate the sum of squares, > divide by number of samples, and square root to get the RMS value of the > readings. > > Now I would like to know why this works....what is the mathematical > relationship between squaring the readings and obtaining the RMS in the end? > > I know it's almost stupid to ask since we're talking about Root Mean Square > to begin with....but I am indeed missing the in between steps in the > story... > Jeffrey D. Spears University of Michigan College of Engineering ``Double-E, can't spell gEEk without it!'' -Captain Gerald M. Bloomfield II, USMC (my brother)

'Magic Sinewaves demystified'
2000\03\07@025109 by David VanHorn

> > > If you had a square wave with a 50% duty cycle, there are no even > harmonics. Furthermore, the relative magnitude of the harmonics are > inversely proportional to their harmonic numbers. If you vary the duty > cycle to say 33%, you'll have no 3rd,6th, 9th harmonic, but some of the > evens will pop up. I don't think there's a way to arbitrarily kill the odd > harmonics. But you can do (at least) two things: vary the pulse width of > individual pulses and vary the number of pulses. The plot: No, it's a problem of receiving a waveform that's synchronus to what I'm doing, but the waveshape is essentially random. It may have any population of harmonics, but I'm only interested in the fundamental, and whatever processing I do to it has to preserve the phase over frequency. It can introduce an absolute shift, as long as it's constant. (hence the boxcar filter),

2000\03\07@062440 by Nikolai Golovchenko

Hi Scott, I never heard of "magic sinewaves" before, so I can be wrong. Anyway, if I understand correctly, there are two goals in generating the pulse sequence: 1) Minimize switching 2) Attenuate higher harmonics I can't understand why you choose PWM to achieve this! Okay, PWM has small number of transitions - two per PWM period. Now harmonics. They appear as a result of PWM conversion and can be thought of as quantization error. To reduce the quantization effect the PWM resolution (and therefore frequency) must be increased. The problem is: If you want to reduce errors(or harmonics), you have to increase PWM frequency. On the other hand, increased PWM frequency increases the number of transitions per unit of time. So these two goals conflict with each other, don't they? I can see another approach. A scheme to minimize quantization errors, also known as "Scenix PWM". -------------- 0..1 ----- | 1 +--- | -----|+ |---o--| | |--o------------- --|+ | | | 0 ----+ | | | ----- | | 1 | | | | -------------- | | | Quantizer | | | | | ------- ----------- | | | | -------- | | + - | | | | | ------- -------- | | -1 | | -----| Z |------- ------- Switching is determined by sampling frequency. Choose what you need or can afford. In any case, the system is trying to eliminate quantization error at every sample (unlike PWM), which leads to a higher attenuation of harmonics at the same clock frequency. The benefit is that you don't have to worry about pulse position, it's done automatically. The noise is minimal at fundamental harmonic frequency and increases with frequency. Probably there is a better feedback variant (noise shaping filter) but this is the simplest and the only one I know <G>. Nikolai On Monday, March 06, 2000 Scott Dattalo wrote: {Quote hidden}> On Mon, 6 Mar 2000, "Grif" w. keith griffith wrote: >> At 10:44 AM 3/5/00 -0600, you wrote.. >> >It finally dawned on me yesterday how magic sine waves can be sythesized. >> >> Major math snip::: >> >> Scott: doesn't your code produce an analog output? I was of the opinion >> the whole reason for the Magic part of the synthesis was using saturated >> switching at a much higher freq than the final output and stuffing it thru >> a low pass filter, getting a somewhat usable, higher power sine wave >> output? Or did I just learn something this early in the week??? ( that >> would be good,,, then I wouldn't have to work as hard the rest of the week!) > True, the ultimate objective is to produce an analog output. However, the > goal with the so called 'magic sinewaves' is to find the MINIMUM number of > switch transitions. The resulting bit stream is filtered such that a sine > wave is produced. The reason for minimizing the transitions is because > some switches, particularly those associated with high power applications > like in AC inverters, require a significant amount of energy to switch or > cannot be efficiently switched at high frequencies. As an example, you may > have a sine wave synthesis circuit where you want to drive the gate of a > MOSFET directly from a pic I/O pin. Pic's have fairly beefy outputs, > however they're very weak when compared to real MOSFET drivers. But given > enough time, the PIC I/O can switch a MOSFET (assuming of course that Voh > is above Vgs threshold - i.e. logic level Vgs MOSFETs may be required). If > you attempted to drive the MOSFET at too high of frequency with this pic > I/O pin, then chances are that it will never switch! > In addition, if you can use harmonic cancellation, then the constraints on > the analog filter may be relaxed. PWM generated waveforms relax the filter > constraints by maximizing the switching frequency. If the PWM carrier is > 100 times the sinewave you're trying to generate, then you're analog > filter is much easier to design. There aren't any harmonics (to an > approximation) between the sine wave frequency you're generating and the > carrier frequency of the PWM. > Magic sine waves explicitly remove these harmonics by strategically > placing the pulses (or edges). The pulses, when viewed individually, > contain harmonics at the fundamental (of the sine wave being generated) > and at all integer multiples above. But when the pulses are taken > together, the 2nd, 3rd and so on harmonics are cancelled. The number of > harmonics cancelled depends on the quantization of time for the edges. For > example, in the previous post, I had a graph of a 12-bit pulse stream > 010110100000. The second and third harmonics were exactly cancelled, but > there is a 4th and 5th harmonic present. Doubling the number of pulses to > 24 allows a pulse stream to be created that will cancel the 4th harmonic. > In general, if you have a pulse stream containing n! pulses, then it's > possible to exactly cancel the first n harmonics. However, n! grows rather > quickly. It's simply not practical to attempt to cancel the first 10 > harmonics (after the fundamental) by creating a pulse stream containing > 10! or 3,628,800 pulses. > Consequently, if you wish to cancel the first 10 harmonics but with a > pulse stream containing only a 1000 pulses, you're not going to achieve > your goal. However, you may discover a particular stream that exactly > cancels the 2nd, 3rd, and 4th, and attenuates the 5th through 10th > harmonics by 20dB. From that you may design an appropriate analog filter. > AFAIK, there is no way to apriori determine the appropriate length of the > pulse stream to achieve some required harmonic profile. I think > Lancaster's contribution to this technology is to discover certain lengths > that perform well - or at least better than pulses streams that are > slightly shorter or longer. What I'm playing around with is a way you can > study the characteristics of one stream versus another. > Scott

2000\03\07@093711 by Scott Dattalo

On Tue, 7 Mar 2000, Nikolai Golovchenko wrote: > Hi Scott, > > I never heard of "magic sinewaves" before, so I can be wrong. Anyway, > if I understand correctly, there are two goals in generating the pulse > sequence: > 1) Minimize switching > 2) Attenuate higher harmonics > > I can't understand why you choose PWM to achieve this! > > Okay, PWM has small number of transitions - two per PWM period. Recall though that if you're generating a sinewave using PWM that there are many PWM cycles for one period of the sinewave. In fact, the more the better. The reason is because it becomes easier for an analog filter to remove the PWM carrier frequency while passing the sinewave. Now in some applications you want to minimize the number of transitions because the switching losses are a significant factor. If you were to limit yourself to say only two transitions per cycle, i.e. a square wave, to generate or synthesize a sinewave then an analog filter would have a whole lot of work to do. If you made the duty cycle 50% then the even harmonics are suppressed and your analog filter would need to attenuate the odds. But the third harmonic is so close to the fundamental that it's difficult to effectively filter it without detrimentally affecting the fundamental. So the next step may be to say, well, I'm going to try two pulses per sine wave cycle. If you study this case, you'll discover that it's possible to suppress either the 2nd or the 3rd but not both (at least that's my observation). So then you say, well, how about three pulses? It turns out that this can be done: http://www.dattalo.com/ms1.gif. But the 4th and 5th harmonics are fairly significant - almost as big as the fundamental. Now to suppress the 4th and the 5th and so on harmonics requires more and more pulses. {Quote hidden}> Now harmonics. They appear as a result of PWM conversion and can be > thought of as quantization error. To reduce the quantization effect > the PWM resolution (and therefore frequency) must be increased. > > The problem is: If you want to reduce errors(or harmonics), you have > to increase PWM frequency. On the other hand, increased PWM frequency > increases the number of transitions per unit of time. So these two > goals conflict with each other, don't they? > > I can see another approach. A scheme to minimize quantization errors, > also known as "Scenix PWM". <they hardly were the ones to first invent this...> {Quote hidden}> -------------- > 0..1 ----- | 1 +--- | > -----|+ |---o--| | |--o------------- > --|+ | | | 0 ----+ | | > | ----- | | 1 | | > | | -------------- | > | | Quantizer | > | | | > | ------- ----------- > | | | > | -------- > | | + - | > | | | > | ------- -------- > | | -1 | | > -----| Z |------- > ------- > > Switching is determined by sampling frequency. Choose what you need or > can afford. In any case, the system is trying to eliminate > quantization error at every sample (unlike PWM), which leads to a > higher attenuation of harmonics at the same clock frequency. The > benefit is that you don't have to worry about pulse position, it's > done automatically. The noise is minimal at fundamental harmonic > frequency and increases with frequency. Probably there is a better > feedback variant (noise shaping filter) but this is the simplest and > the only one I know <G>. But it is interesting to look at from this perspective (as opposed to looking at an algorithm). Scott

'[OT] Contracting business'
2000\03\07@100113 by Lawrence Lile

Andy, I have long thought that the tech-head engineer would need a smooth talking salesman as a partner to make it a go. Apparently, you solve this by making good allies of people in the selling business anyway. (smart.) An Mchip rep or FAE would be a really good ally to have! I've been working on ally relationships with a couple of other manufacturer's reps - baiscally "If you need in-depth tech support and programming for my parts, see this guy" It helps if the supplier has some kind of certification program similar to Mchip's "Certified PIC Geek" or whatever they call it program. How else does one go about drumming up business? Advertise in the classifieds next to the used bicycles?<G> Scmoooze with the bigwigs at the Chamber of Commerce dinners? Post a web page under "Inventors and Crackpots - Get your Invention built here"? -- Lawrence "one foot in the independant consulting business" Lile {Original Message removed}

'Software RMS measurement of sine waves again'
2000\03\07@114813 by Harold M Hallikainen

On Mon, 6 Mar 2000 22:29:23 -0400 Jeff Davis <davisjrSTOPspamIS2.DAL.CA> writes: {Quote hidden}> On Mon, 6 Mar 2000, Lorick wrote: > > > Now I would like to know why this works....what is the > mathematical > > relationship between squaring the readings and obtaining the RMS > in the end? > > > > I know it's almost stupid to ask since we're talking about Root > Mean Square > > to begin with....but I am indeed missing the in between steps in > the > > story... > > What squaring the values does is effectively eliminates the sign of > the > number (+/-). If you were to simply take the average of all > measurements > of a sine wave you would get zero. An RMS value is the average > magnitude > of each reading without taking into account the sign of the reading. > > Jeff This would be true IF we took the square root prior to taking the mean (squared, took square root, accumulated sum, divided by number of samples) since the square root of the square is one way of defining the absolute value. However, in RMS, we first square the sample, then accumulate a sum, then divide by the number of samples, THEN take the square root (hence Root Mean Square). The reason for the squaring is because RMS is related to the POWER of the waveform, and power has a squared term (either I^2*R or V^2/R). So, RMS is a method of weighting so the higer amplitude signals have more significance than they would with a straight averaging. Harold FCC Rules Online at http://hallikainen.com/FccRules Lighting control for theatre and television at http://www.dovesystems.com ________________________________________________________________ YOU'RE PAYING TOO MUCH FOR THE INTERNET! Juno now offers FREE Internet Access! Try it today - there's no risk! For your FREE software, visit: dl.http://www.juno.com/get/tagj.

'Magic Sinewaves demystified'
2000\03\07@141828 by Scott Dattalo

On Tue, 7 Mar 2000, David VanHorn wrote: > No, it's a problem of receiving a waveform that's synchronus to what I'm > doing, > but the waveshape is essentially random. > It may have any population of harmonics, but I'm only interested in the > fundamental, and whatever processing I do to it has to preserve the phase over > frequency. It can introduce an absolute shift, as long as it's constant. > (hence > the boxcar filter), ? What's your application? I don't understand the concept of 'constant phase over frequency', when there's only one frequency involved. If there's a frequency band then it does make some sense (but I tend to think in terms of group delays [and bessel filters]). Are you building a tone decoder? What's the bandwidth and center Frequency? Or are you building a PLL? Scott

2000\03\07@154817 by Dan Michaels

Scott wrote: > >Recall though that if you're generating a sinewave using PWM that there >are many PWM cycles for one period of the sinewave. In fact, the more the >better. The reason is because it becomes easier for an analog filter to >remove the PWM carrier frequency while passing the sinewave. > Speaking of sinewaves generated via PWM, I once had the brilliant idea I could use the PWM generated by a PIC73, along with some data stored in EEPROM, to add cheapo voice output to an embedded system. It was gonna be called the "Tiny Talker" [or maybe Tiny Skwauker]. You would probably have to limit the highest voice frequencies to 1-2 Khz or so, but it would probably sound as good as early skwauky devices, and would cost you next to nothing to add to the embedded system. The fundamental frequency of 7-bit PWM on a 20Mhz PIC is 156 Khz. That would give you 78 samples on a 2 Khz sinewave. Stick on a nice 1 Khz low-pass filter, and the PWM fundamental would be down about 44 dB. [I think - the first thing you forget is all that EE math]. Think this could work? Toodle Pip - (is that how it goes?) - Dan Michaels Oricom Technologies http://www.sni.net/~oricom ==========================

2000\03\07@162730 by David VanHorn

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 >What's your application? I don't understand the concept of 'constant phase >over frequency', when there's only one frequency involved. If there's a >frequency band then it does make some sense (but I tend to think in terms >of group delays [and bessel filters]). Are you building a tone decoder? Nope. I'm switching an array of antennas, creating a phase shift on the RF carrier, which is demodulated, and used to give a vector to the transmitter. The problem is that the recovered waveform is rather non-sinusoidal, so it has noise, plus odd products of the fundamental. >What's the bandwidth and center Frequency? Or are you building a PLL? It almost hasn't got one. The center frequency is entirely programmable, and will shift during operation. The current boxcar is equivalent to a filter with a center frequency settable anywhere from 200hZ to 8kHz, with a bandwidth of about 1/2 Hz. The boxcar passes higher harmonics fairly well, so I have a second boxcar running at twice the speed, which causes the fundamental to drop out. This output is inverted and summed with the first boxcar, giving me an output that only has the fundamental, and odd harmonics. Due to using boxcars, I can even make radical shifts in switching frequency during operation, since the phase response of the boxcars is linear (they are clocked right along with the antenna switching) Say 10 "spins" at 200 hz, followed by 10 at 8000, and then 10 at 2000, or set the switching speed per "revolution" according to a PN generator's output, so that there is no way to modulate the transmitted carrier so as to disturb my aquisition of the signal. -----BEGIN PGP SIGNATURE----- Version: PGPfreeware 6.5.2 for non-commercial use <http://www.pgp.com> iQA/AwUBOMWeBYFlGDz1l6VWEQJRHQCeMNf79D+BPSAherGgrdl6r1kLr/EAn1EK kVLcsp/wAgdm+UI8KLRg5BVB =o3+H -----END PGP SIGNATURE-----

2000\03\07@163330 by Andrew Kunz

Who you doing phased array for? akunzSTOPspamKILLspamtdipower.com Andy

'Magic Sinewaves demystified [OT]'
2000\03\07@170318 by Sean Breheny

I knew the black helicopters and the aliens had to have SOMEONE from the PICLIST working for them ;-) Sean At 04:31 PM 3/7/00 -0500, you wrote: >Who you doing phased array for? @spam@akunz.....spamtdipower.com > >Andy > | | Sean Breheny | Amateur Radio Callsign: KA3YXM | Electrical Engineering Student \--------------=---------------- Save lives, please look at http://www.all.org Personal page: http://www.people.cornell.edu/pages/shb7 spamshb7..........cornell.edu ICQ #: 3329174

2000\03\07@171610 by David VanHorn

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 At 05:04 PM 3/7/00 -0500, Sean Breheny wrote: >I knew the black helicopters and the aliens had to have SOMEONE from the >PICLIST working for them ;-) You're not as far off as you might imagine, though it's not a black project either. -----BEGIN PGP SIGNATURE----- Version: PGPfreeware 6.5.2 for non-commercial use <http://www.pgp.com> iQA/AwUBOMWpg4FlGDz1l6VWEQLQ0wCgoSeW26lPqeKKySBFixo04D+jrs8AoLUb k2ZAJCh1+FSj5Ilsv//CwI0Q =PHIY -----END PGP SIGNATURE-----

2000\03\07@192806 by Grif\ w. keith griffith

<x-flowed>Well,,, if he ain't a black box type,,, that'd make him an alien? Do engineers count as aliens? How about hams? At 05:14 PM 3/7/00 -0800, you wrote: {Quote hidden}>-----BEGIN PGP SIGNED MESSAGE----- >Hash: SHA1 > >At 05:04 PM 3/7/00 -0500, Sean Breheny wrote: > >I knew the black helicopters and the aliens had to have SOMEONE from the > >PICLIST working for them ;-) > >You're not as far off as you might imagine, though it's not a black project >either. > >-----BEGIN PGP SIGNATURE----- >Version: PGPfreeware 6.5.2 for non-commercial use <http://www.pgp.com> > >iQA/AwUBOMWpg4FlGDz1l6VWEQLQ0wCgoSeW26lPqeKKySBFixo04D+jrs8AoLUb >k2ZAJCh1+FSj5Ilsv//CwI0Q >=PHIY >-----END PGP SIGNATURE----- 'Grif' N7IVS </x-flowed>

2000\03\07@194926 by David VanHorn

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 At 04:31 PM 3/7/00 -0800, \"Grif\" w. keith griffith wrote: >Well,,, if he ain't a black box type,,, that'd make him an alien? >Do engineers count as aliens? How about hams? I'm both. -----BEGIN PGP SIGNATURE----- Version: PGPfreeware 6.5.2 for non-commercial use <http://www.pgp.com> iQA/AwUBOMXNVYFlGDz1l6VWEQJAbQCgvozdC3bmCw6i+fENeFqe1l1NbOwAoKO6 6XcmFdu/pyK/mhQwWv7Zb4S7 =meB2 -----END PGP SIGNATURE-----

'[OT] Contracting business'
2000\03\08@022233 by Glen Torr

Howdy All, This is a subject close to my situation at the moment, about 2.5 years ago I took a part time job at a local University doing general electronics design / maintenance to fund my private development work. Since then I have taken an additional half time contract for 6 months with the same employer. The worry is that I can earn far more for general electronic work in a wage environment than while contracting for hourly work in which I am working at a considerabley higher and more focussed level. I have to face the possibility that there is not a living to be made in contract PIC programming... at least in Australia ( and I suspect most of the rest of the world) at the present time. In Aus at the moment, where I live, ycan earn considerabley more driving a bus than developing PIC products. I am not angry about this but it is hard to understand why it is so. Glen Torr TorrTech (for the momentt)

'Magic Sinewaves demystified'
2000\03\08@034822 by Graeme Zimmer

Hi Dave, > Nope. I'm switching an array of antennas, creating a phase shift on the RF > carrier, which is demodulated, and used to give a vector to the > transmitter. The problem is that the recovered waveform is rather > non-sinusoidal, so it has noise, plus odd products of the fundamental. An excellent application for a PIC! A Doppler Direction Finder.... I trust that you have seen the original "DopplerScant" article from about 15 years ago? (was it QST or HR?). It was the first time that I saw mention of a Commutating Filter...... Strangely they seem to be rarely used today... I would have a copy here somewhere if you want.... ............... Zim

2000\03\08@063005 by paulb

David VanHorn wrote: > Due to using boxcars, I can even make radical shifts in switching > frequency during operation, since the phase response of the boxcars is > linear (they are clocked right along with the antenna switching) These are in fact digital implementations, are they not? > Say 10 "spins" at 200 hz, followed by 10 at 8000, and then 10 at 2000, > or set the switching speed per "revolution" according to a PN > generator's output, so that there is no way to modulate the > transmitted carrier so as to disturb my aquisition of the signal. Somehow, compensating for the phase shift in the IF and detector has always worried me about that proposal. How do you propose to do that? -- Cheers, Paul B.

'Magic Sinewaves demystified [OT]'
2000\03\08@070621 by Andrew Kunz

My mother was a German alien until she naturalized. Now she's an American. I'm a Native American - I was borneded here. Andy

2000\03\08@130428 by David VanHorn

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 >An excellent application for a PIC! A Doppler Direction Finder.... Well... Actually it's not exactly a pic.. It's an AVR project (8 MHz AVR8535), but this part of the project is certainly applicable to both worlds. (he said, edging twoard the door) >I trust that you have seen the original "DopplerScant" article from about 15 >years ago? (was it QST or HR?). Ancient history. It was a masterpice in it's time, but not anymore. >It was the first time that I saw mention of a Commutating Filter...... That's what they called it, AofE calls it a boxcar integrator, which I think is more descriptive. It works like this: I use one timer's int to step the antennas (eight of them, could be four, could be more) On switching from antenna X to antenna X+1, there is some phase shift imposed on the carrier. Although this is a step function, the receiver integrates it to some degree, resulting in something between a pulse, and a level shift on the audio output (a purist would tap the discriminator, I have a 10.7 MHz IF input) Over one scan cycle, you get a number of phase shifts, which are exactly in sync with the antenna switching. The boxcar integrator connects eight capacitors to the signal line, through a high resistance, each for 1/8 of the scan cycle. If you do this a lot, then the integrator averages up a waveform that represents the phase shifts, which in a perfect world, would be a sine wave. Unfortunately, there are reflections from odd angles, so you end up with something rather less than a sine. My device runs a second boxcar twice as fast, which averages up all the even hamonic content of the waveform, which I level detect, and compare to the level of the fundamental plus evens, (two ADC channels) which tells me how "clean" the waveform is. I then invert the even hamonic waveform, and add it to the main, which gets rid of the evens pretty nicely. Unfortunately, the third and to some degree the fifth, sail right on through. After that is done, the signal is fed to an MF-6 filter, which is clocked synchronusly with the antennas, and this filter effectively converts the remaining waveform into a sine. At this point, I detect the level again, and the zero crossings for phase information. The original "dopplescant" and all it's clones run somewhere between 200 and 400 scans/sec, and do almost nothing to "clean up" the signal, relying mostly on "eyeball integration". I can vary my scan rate, even during aquisition, since all the filters are clocked from the scan rate. My low end is 200 Hz, for practical reasons, and high is 7812 Hz, because of the divisor limitations of the timer I'm using to divide the 8 MHz proc clock. 4 MHz is as fast as I can get to, and by the time I divide that down for the MF-6 and the antenna steppers, we end up at 7812 scans/second. The "dopplescant" simply lights the appropriate LED when it has a signal, and lets you worry about everything else. The other microprocessor units out there, average the data incorrectly, by simply averaging the most recent sample with the average of the previous samples. IOW: 90,90,90,.....90,90,90,270 = 180 in that system. Obviously wrong, but that's what they do, and the nature of the system is that you do get "wild" bearings fairly often, when a reflection is momentarily stronger than the true signal path. On each sample int (zero crossing ^ of the recovered sine), I take the 8 bit sample data, correct for an absolute offset, and current vehicle heading (vampired from compass data or GPS if compass is not available), and store it in an array of 16 samples. If the harmonic content is too high, or the amplitude too low, then the sample is skipped. Then the sample is converted from radial form to polar form, which makes the rest easier. (trust me) Every 16th sample, I run vector calculus on the lot, to properly average the bearing, though at this level it's simpler, since the radius is all "1". The result is stored in another array of 16. Each time that array fills, I run the vector calc again, this time with vector and radius to worry about, and derive a "raw" bearing (though it's hardly raw) and a supersample, which is stored in another array of 16. That process is repeated again, allowing me to produce a good output on 16 scans, and superb output if the carrier stays present for 16^4 scans. (8 seconds at 7812 Hz) For both the "raw" and the cooked samples, I convert back to radial, which now gives me both a vector and a radius, and the radius indicates the quality of the bearing. Without running the math on it (approximate) 90,90,90,90,90,90,90,90,90,270 = something like 94 degrees, with a radius of 0.9 or something similar. You see the difference? If the samples were really ratty, then you might have samples all over the map, and I would still derive a bearing, but the radius would be a lot closer to zero, indicating that I'm not very sure of my bearing. The radius information is used by the PC application to indicate confidence. Each sample output is tagged with the current GPS location and heading information, and put in my outbound sample buffer. In the background, I am serving as half of a stat mux, taking data from four serial ports, and two internal "virtual" ports, and creating variable length packets to be sent out another serial port at 38400 baud, to a PC application, which tears this stream apart, and sees all the data as separate inputs again (important for laptops, which generally have only one serial port!) On the downstream side, I grab the data from the pc, parse the packets, and output each channel's data to it's respective port, real or virtual. I also manage handshaking on each port, signalling the host app through a command channel, which ports can accept additional data. I also handle the cases where packets were sent before I said "enough" by putting them in the "holding bin" until the port frees up. (Command channel packets are not queued, they are jammed into the flow as soon as the current packet completes, since they are time sensitive) In my spare time, I vampire the GPS and compass NMEA inputs for the information I need, and apply that as corrections to the sampling, so that you can take an accurate bearing while driving in a circle for example. One final weakness of the original "dopplescant" was that you could transmit a tone that was exactly at the antenna switching speed, and confuse it pretty easily. Finding the exact tone was a little difficult, but that's the only hard part. My unit normally operates at the 7800 Hz rate (for fast aquisition) but if it has trouble getting a bearing, it will start hopping scan rates, according to the output of a PN generator (internal background task) so the effect is similar to spread spectrum, in that your jamming signal is spread, but my antenna scanning is always in sync, so your jamming dissapears. The display unit talks to the main board by SPI link, (low speed) and has another AVR processor that presents the "raw" and "cooked" data on a ring of bi-color LEDs, (tricky multiplexing, I'm using two lead leds) and a bargraph, and sends me the state of three buttons that I use to configure and control the system. The PC application takes my datastream, and puts it on a moving map, with you at the center, plots each bearing from the point where it was taken. It also talks to other systems, through an RF data link, and plots their reported bearings on the map, from wherever they were taken, and indicates their current position, course, and speed as well. So, if you decide to get on the ham bands without a licence, or make a nuisance of yourself, you better be able to outrun me, because every time you transmit, I'm going to know EXACTLY where you are, moving or not. It only takes me about 2mS to grab a bearing that's better than any other doppler unit on the market (more precise) and the longer the signal, the more accurate I get. The end market for this thing is serious hams like me, and certain governmental bodies that have similar interests. There is also a market for tracking tornadoes, as they emit significant RF noise. I have already shown that I can accurately DF broadband noise sources with this rig. The really fun part is the vector math, which involves a square root calculation for every 16 incoming samples, as part of an ISR. (The sqrt is eleven lines of code) Now, if you'll all don the blindfolds, and stand against the wall please? :) But, I'm still looking for a reasonable way to remove the odd harmonics, like the 2x boxcar does the evens. -----BEGIN PGP SIGNATURE----- Version: PGPfreeware 6.5.2 for non-commercial use <http://www.pgp.com> iQA/AwUBOMa/94FlGDz1l6VWEQIb9ACg5qcujxeGyHdjHgxzK2it1X/R98sAn3fu 6lHZvFzJ7n6C9HVey7u55rtN =D59g -----END PGP SIGNATURE-----

2000\03\08@131052 by David VanHorn

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 > Somehow, compensating for the phase shift in the IF and detector has >always worried me about that proposal. How do you propose to do that? >-- > Cheers, > Paul B. It's a constant offset. Just tune in a known transmitter (NWS on 162.5-ish) and enter the true bearing, or point the vehicle at the transmitter, and tell it to calibrate. It sweeps up the possible sample rates, and enters an offset value for each scan rate, so even the group delay in the audio circuits is taken care of. I do accept audio input as from a speaker jack, but I prefer, and have the circuitry for, a 10.7 MHz IF input, into my own demod, so as to eliminate most of the variables. -----BEGIN PGP SIGNATURE----- Version: PGPfreeware 6.5.2 for non-commercial use <http://www.pgp.com> iQA/AwUBOMbBn4FlGDz1l6VWEQJoTQCdEgBqz4NGeIi7OQXTld0nRnHcwXEAn3aJ pzJFMRwFmdv8w3+ExEwJgLsg =b828 -----END PGP SIGNATURE-----

'[OT] Contracting business'
2000\03\08@144911 by l.allen

> > In Aus at the moment, where I live, ycan earn considerabley more driving a > bus than developing PIC products. I am not angry about this but it is hard > to understand why it is so. > > Glen Torr > > TorrTech (for the momentt) Its no better in New Zealand. Basically what I can see is that the electronics industry sees technically minded folk as cheap labour, geeks that do it for the love of it, easy meat. The only trouble is these same employers, business people etc are bleating they cant get enough skilled people to run their businesses and need cheap foreign migrants to fulfil the shortfall. You want a decent income ... become an accountant or make and sell your product direct to the consumer. _____________________________ Lance Allen Technical Officer Uni of Auckland Psych Dept New Zealand http://www.psych.auckland.ac.nz _____________________________

'Magic Sinewaves demystified'
2000\03\08@155737 by paulb

David VanHorn wrote: > It's a constant offset. Just tune in a known transmitter (NWS on > 162.5-ish) and enter the true bearing, or point the vehicle at the > transmitter, and tell it to calibrate. It sweeps up the possible > sample rates, and enters an offset value for each scan rate, so even > the group delay in the audio circuits is taken care of. The delay between altering scan rate and beginning to take readings was my concern. It would matter if you were truly using a spread spectrum approach but as it seems you are just swapping frequencies and allowing a substantial number of "sweeps" at each, this apparently is diluted out. -- Cheers, Paul B.

2000\03\08@161445 by David VanHorn

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 > The delay between altering scan rate and beginning to take readings >was my concern. It would matter if you were truly using a spread >spectrum approach but as it seems you are just swapping frequencies and >allowing a substantial number of "sweeps" at each, this apparently is >diluted out. I have a holdoff for the MF-6 to stabilize at the new clock, so it ignores scans for a short while after each hop. If I didn't have the MF-6 in there, then each sweep could be at a different rate, but yes, the group delay of the receiver would become important, and somewhat difficult to calibrate to. Maybe not, I haven't tried that. I was thinking about an SS approach earlier (months) but didn't think it would be that useful, and it prevents me from using the MF-6, which definitely helps clean things up. (Nice chip that!) -----BEGIN PGP SIGNATURE----- Version: PGPfreeware 6.5.2 for non-commercial use <http://www.pgp.com> iQA/AwUBOMbsXoFlGDz1l6VWEQI3IwCeK7UoWEAXld9WKoGG0nV9N8x9WGwAoKOh CUQjMS7aOr+qbE3Mqb9rrmSs =qutQ -----END PGP SIGNATURE-----

2000\03\08@171142 by paulb

David VanHorn wrote: > using the MF-6, which definitely helps clean things up. I thought the MF-6 was simply a switched-capacitor "boxcar" filter like you are already using for the primary filter? How does it reject harmonics? -- Cheers, Paul B.

2000\03\08@231550 by David VanHorn

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 > I thought the MF-6 was simply a switched-capacitor "boxcar" filter >like you are already using for the primary filter? How does it reject >harmonics? I'm not really sure of the internals, but it's outputting a nice clean sine. It's clocked a lot faster, 50x (actually 64X in my application) rather than 8x. -----BEGIN PGP SIGNATURE----- Version: PGPfreeware 6.5.2 for non-commercial use <http://www.pgp.com> iQA/AwUBOMdPOoFlGDz1l6VWEQKu+ACgoCT3QnOKQJ6aO861Wr8C7HSQioUAoIov MBxW3BWFo5RKXa6m7LaT3jOb =EwMY -----END PGP SIGNATURE-----

2000\03\09@084059 by Graeme Zimmer

Hi, > I thought the MF-6 was simply a switched-capacitor "boxcar" filter > like you are already using for the primary filter? How does it reject > harmonics? Switched capacitor filters are analogous to conventional multi oppamp filters such as Biquad or State Variable. They merely use the switched capacitor as a (sort of) digitally variable resistor. They don't respond to harmonics. The Commutating Filter is a completely different beast, which DOES respond to harmonics. Personally, I think that Box Car Integrator is the wrong term for these....... cheers ............. Zim

2000\03\09@091810 by paulb

Graeme Zimmer wrote: > Switched capacitor filters are analogous to conventional multi oppamp > filters such as Biquad or State Variable. Ah yes! Now I recall. A bit vague of me. I recall when a local magazine published a telephone modem using switched capacitor filters, quite a while ago. Had a bit of a bad reputation which I take to indicate the difficulty of taming such a beast implemented in discrete components with switching frequencies floating about... In a single IC; a much better proposition. -- Cheers, Paul B.

2000\03\09@120729 by David VanHorn

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 >The Commutating Filter is a completely different beast, which DOES respond >to harmonics. > >Personally, I think that Box Car Integrator is the wrong term for >these....... Well, that's what AofE calls them.. -----BEGIN PGP SIGNATURE----- Version: PGPfreeware 6.5.2 for non-commercial use <http://www.pgp.com> iQA/AwUBOMf/KoFlGDz1l6VWEQJ24QCg0GIZrNL8ieoqAhQqq/f5w+RnbVQAoJFV m4YhhFRcFjtzvK5/+zszxQ28 =sfEX -----END PGP SIGNATURE-----

'[PIC]: Sine table'
2000\08\18@131753 by Andrew Kunz

Anybody have a quick program to build a 128-point (or so) sine table? Thanks. Andy -- http://www.piclist.com hint: PICList Posts must start with ONE topic: [PIC]: PIC only [EE]: engineering [OT]: off topic [AD]: advertisements

2000\08\18@132624 by Simon Nield

Try using Excel or some other spread sheet to give you n values for the first pi/2, export is as a text file and then paste in some retlws ? add a couple of lines of code to invert and add 1 to the result when neccessary and Bob's your mother's brother. Simon p.s. it's the weekend. phew! -- http://www.piclist.com hint: PICList Posts must start with ONE topic: [PIC]: PIC only [EE]: engineering [OT]: off topic [AD]: advertisements

2000\08\18@132851 by M. Adam Davis

Yeah, Excel. Would you like: 128 points of a (full/half/quarter) wave Full scale (0 and 255 are peak, 128 is mid) Any other needs? I can do this now. -Adam Andrew Kunz wrote: {Quote hidden}> > Anybody have a quick program to build a 128-point (or so) sine table? > > Thanks. > > Andy > > -- > http://www.piclist.com hint: PICList Posts must start with ONE topic: > [PIC]: PIC only [EE]: engineering [OT]: off topic [AD]: advertisements -- http://www.piclist.com hint: PICList Posts must start with ONE topic: [PIC]: PIC only [EE]: engineering [OT]: off topic [AD]: advertisements

2000\08\18@133144 by Andrew Kunz

Thanks. I don't know what I was (not) thinking. I got it done on my PC with MSVC. THanks anyway. Andy -- http://www.piclist.com hint: PICList Posts must start with ONE topic: [PIC]: PIC only [EE]: engineering [OT]: off topic [AD]: advertisements

2000\08\18@133754 by Andrew Kunz

part 1 334 bytes content-type:text/plain; charset=us-ascii So nobody needs to ask the same dumb question as I just did... Note that this program gives me 8-bit values to output to a parallel DAC (with appropriate filtering), and that it takes advantage of the fact that the table is symmetrical top to bottom but also on either side of the maxima. Andy (See attached file: sinetbl.c) part 2 413 bytes content-type:application/octet-stream; (decode) part 3 144 bytes -- http://www.piclist.com hint: PICList Posts must start with ONE topic: [PIC]: PIC only [EE]: engineering [OT]: off topic [AD]: advertisements

2000\08\18@174649 by pandersn

Hi Andy: It's is interesting in that we all do it: bring sophisticated tools to simple problems. It is hard to do but we need to try to remember to set back and think of simple solutions to problems. Saw some of the responses. Liked the "use the spread sheet." This tread reminded me of a book on design: The Design of Everyday Things. Can't remember of author. He said he wrote the book because he was tired of working with products - of all kinds - without a simple interface - one that you can "look at and deduce proper operation" - for example the design of the pattern of burners on a stove and the associated design of the panel of switches. My GE at home is a loser! Ha. Have a good weekend. Phil Anderson. On Friday, August 18, 2000 12:30 PM, Andrew Kunz [SMTP:akunz.....TDIPOWER.COM] wrote: {Quote hidden}> Thanks. I don't know what I was (not) thinking. > > I got it done on my PC with MSVC. > > THanks anyway. > > Andy > > -- > http://www.piclist.com hint: PICList Posts must start with ONE topic: > [PIC]: PIC only [EE]: engineering [OT]: off topic [AD]: advertisements -- http://www.piclist.com hint: PICList Posts must start with ONE topic: [PIC]: PIC only [EE]: engineering [OT]: off topic [AD]: advertisements

2000\08\18@220338 by Dan Michaels

Phil Anderson wrote: ........ >This tread reminded me of a book on design: The Design of Everyday Things. >Can't remember of author. He said he wrote the book because he was tired of >working with products - of all kinds - without a simple interface - one >that you can "look at and deduce proper operation" - for example the design >of the pattern of burners on a stove and the associated design of the panel >of switches. My GE at home is a loser! Ha. > author = Donald A. Norman - too bad the guys who designed the insipid flashing 12:00 on the VCR machine never read the book. And note - after 10 years, I still don't know how to set the digital clock on my [used] car radio. - danM -- http://www.piclist.com hint: PICList Posts must start with ONE topic: [PIC]: PIC only [EE]: engineering [OT]: off topic [AD]: advertisements

2000\08\19@061459 by Alan B. Pearce

>export is as a >text file and then paste in some retlws ? you could probably have a column of Retlw in excel so it gets included in the export. It is some time since I tried to do a CSV export from excel, so cannot remember if you can tell it to use spaces instead of commas. -- http://www.piclist.com hint: The list server can filter out subtopics (like ads or off topics) for you. See http://www.piclist.com/#topics

2000\08\20@190127 by Tony Nixon

Simon Nield wrote: > > Try using Excel or some other spread sheet to give you n values for the first pi/2, export is as a > text file and then paste in some retlws ? > add a couple of lines of code to invert and add 1 to the result when neccessary and Bob's your > mother's brother. > > Simon > > p.s. it's the weekend. phew! > > -- > http://www.piclist.com hint: PICList Posts must start with ONE topic: > [PIC]: PIC only [EE]: engineering [OT]: off topic [AD]: advertisements Use the DT table maker. Copy the column of data to the clipboard, paste into DT, press a button and presto, a table of RETLW or DT type instructions is created ready for saving or pasting directly into code. http://www.picnpoke.com/demo/dtimg.html -- Best regards Tony ICmicro's http://www.picnpoke.com KILLspamsalesspam_OUTpicnpoke.com -- http://www.piclist.com hint: The PICList is archived three different ways. See http://www.piclist.com/#archives for details.

2000\08\21@072647 by Andrew Kunz

So who uses assembly any more? I do almost everything in C. Andy "Alan B. Pearce" <spam_OUTA.B.PearceTakeThisOuTRL.AC.UK> on 08/19/2000 06:13:26 AM Please respond to pic microcontroller discussion list <.....PICLIST.....RemoveMEMITVMA.MIT.EDU> To: spam_OUTPICLISTTakeThisOuTEraseMEMITVMA.MIT.EDU cc: (bcc: Andrew Kunz/TDI_NOTES) Subject: Re: [PIC]: Sine table >export is as a >text file and then paste in some retlws ? you could probably have a column of Retlw in excel so it gets included in the export. It is some time since I tried to do a CSV export from excel, so cannot remember if you can tell it to use spaces instead of commas. -- http://www.piclist.com hint: The list server can filter out subtopics (like ads or off topics) for you. See http://www.piclist.com/#topics -- http://www.piclist.com hint: To leave the PICList EraseMEpiclist-unsubscribe-requestspamBeGoneKILLspammitvma.mit.edu>

2000\08\21@073725 by Frank Adlam

Well said Andy! -----Original Message----- From: Andrew Kunz [RemoveMEakunzspamBeGonespamTDIPOWER.COM] Sent: Monday, August 21, 2000 1:26 PM To: @spam@PICLISTspamMITVMA.MIT.EDU Subject: Re: [PIC]: Sine table So who uses assembly any more? I do almost everything in C. Andy "Alan B. Pearce" <TakeThisOuTA.B.PearceKILLspam@spam@RL.AC.UK> on 08/19/2000 06:13:26 AM Please respond to pic microcontroller discussion list <.....PICLISTRemoveMEMITVMA.MIT.EDU> To: KILLspamPICLISTTakeThisOuTMITVMA.MIT.EDU cc: (bcc: Andrew Kunz/TDI_NOTES) Subject: Re: [PIC]: Sine table >export is as a >text file and then paste in some retlws ? you could probably have a column of Retlw in excel so it gets included in the export. It is some time since I tried to do a CSV export from excel, so cannot remember if you can tell it to use spaces instead of commas. -- http://www.piclist.com hint: The list server can filter out subtopics (like ads or off topics) for you. See http://www.piclist.com/#topics -- http://www.piclist.com hint: To leave the PICList TakeThisOuTpiclist-unsubscribe-requestspam_OUTmitvma.mit.edu -- http://www.piclist.com hint: To leave the PICList RemoveMEpiclist-unsubscribe-requestspamSTOPspammitvma.mit.edu>

2000\08\21@091059 by Bob Ammerman

At the risk of starting a flame war... Real (wo)men code in assembly (or if they are extra tough, binary) :-) Just kidding of course. Both asm and "C" have their place. For typical applications "C" is fine, but when you are trying to get the most possible function out of the least possible hardware, you can't beat asm. (see my sig line to figure out where I prefer to be). Bob Ammerman RAm Systems (contract development of high performance, high function, low-level software) {Original Message removed}

2000\08\21@092724 by Alan B. Pearce

>At the risk of starting a flame war... >Real (wo)men code in assembly (or if they are extra tough, binary) :-) >Just kidding of course. Both asm and "C" have their place. For typical >applications "C" is fine, but when you are trying to get the most possible >function out of the least possible hardware, you can't beat asm. (see my sig >line to figure out where I prefer to be). I used to work on a machine where the assembler was known as REAL. It stood for RElocatable Assembly Language. Anyone else here who used to work on QANTEL systems??? -- http://www.piclist.com hint: To leave the PICList .....piclist-unsubscribe-requestEraseMEmitvma.mit.edu>

2000\08\21@094138 by Scott Dattalo

On Mon, 21 Aug 2000, Bob Ammerman wrote: > At the risk of starting a flame war... > > Real (wo)men code in assembly (or if they are extra tough, binary) :-) > > > Just kidding of course. Both asm and "C" have their place. For typical > applications "C" is fine, but when you are trying to get the most possible > function out of the least possible hardware, you can't beat asm. (see my sig > line to figure out where I prefer to be). True. If you need a slow sine, then fine. If you need a fast one then Eric's 6-cycle one can not be beat by any C-compiler (except those that support in-line assembly). Scott -- http://www.piclist.com hint: To leave the PICList spamBeGonepiclist-unsubscribe-requestRemoveMEmitvma.mit.edu>

2000\08\21@095218 by Andrew Kunz

>Just kidding of course. Both asm and "C" have their place. For typical >applications "C" is fine, but when you are trying to get the most possible >function out of the least possible hardware, you can't beat asm. (see my sig >line to figure out where I prefer to be). I agree 100%. "The right tool for the job." Andy -- http://www.piclist.com hint: To leave the PICList .....piclist-unsubscribe-requestEraseMEmitvma.mit.edu>

2000\08\21@095629 by Andrew Kunz

I need to generate a 48-63 Hz sine as part of a PLL for a power system (sync the output to the line, slightly out of phase due to the power-generation lag time). I have LOTS of time to make it do it's thing. Andy Scott Dattalo <spamscottspam_OUT@spam@DATTALO.COM> on 08/21/2000 09:42:11 AM Please respond to pic microcontroller discussion list <spamPICLIST@spam@STOPspamMITVMA.MIT.EDU> To: spamBeGonePICLISTspamBeGone@spam@MITVMA.MIT.EDU cc: (bcc: Andrew Kunz/TDI_NOTES) Subject: Re: [PIC]: Sine table On Mon, 21 Aug 2000, Bob Ammerman wrote: > At the risk of starting a flame war... > > Real (wo)men code in assembly (or if they are extra tough, binary) :-) > > > Just kidding of course. Both asm and "C" have their place. For typical > applications "C" is fine, but when you are trying to get the most possible > function out of the least possible hardware, you can't beat asm. (see my sig > line to figure out where I prefer to be). True. If you need a slow sine, then fine. If you need a fast one then Eric's 6-cycle one can not be beat by any C-compiler (except those that support in-line assembly). Scott -- http://www.piclist.com hint: To leave the PICList RemoveMEpiclist-unsubscribe-requestRemoveMERemoveMEmitvma.mit.edu -- http://www.piclist.com hint: To leave the PICList piclist-unsubscribe-requestKILLspamspammitvma.mit.edu>

2000\08\21@101853 by Scott Dattalo

On Mon, 21 Aug 2000, Andrew Kunz wrote: > I need to generate a 48-63 Hz sine as part of a PLL for a power system (sync the > output to the line, slightly out of phase due to the power-generation lag time). > I have LOTS of time to make it do it's thing. That's the kind of thing I wrote this for: http://www.dattalo.com/technical/software/pic/picsine.html The phase accumulators can be directly fed from a PLL. Scott ps. Beware, it's overly commented :). -- http://www.piclist.com hint: To leave the PICList spam_OUTpiclist-unsubscribe-request@spam@mitvma.mit.edu>

2000\08\21@102932 by Andrew Kunz

Thanks, Scott. That'll be interesting reading. I'm working the PLL from the angle of changing time between steps rather than amplitude in order to keep phase locked (my only input is a zero-crossing detector). I also have to generate multiple waves, since I'm generating "real" power not just a 110V household socket. Andy Scott Dattalo <TakeThisOuTscottspam_OUTDATTALO.COM> on 08/21/2000 10:20:51 AM Please respond to pic microcontroller discussion list <KILLspamPICLIST.....TakeThisOuTMITVMA.MIT.EDU> To: TakeThisOuTPICLISTEraseMERemoveMEMITVMA.MIT.EDU cc: (bcc: Andrew Kunz/TDI_NOTES) Subject: Re: [PIC]: Sine table On Mon, 21 Aug 2000, Andrew Kunz wrote: > I need to generate a 48-63 Hz sine as part of a PLL for a power system (sync the > output to the line, slightly out of phase due to the power-generation lag time). > I have LOTS of time to make it do it's thing. That's the kind of thing I wrote this for: http://www.dattalo.com/technical/software/pic/picsine.html The phase accumulators can be directly fed from a PLL. Scott ps. Beware, it's overly commented :). -- http://www.piclist.com hint: To leave the PICList spam_OUTpiclist-unsubscribe-requestRemoveME.....mitvma.mit.edu -- http://www.piclist.com hint: To leave the PICList spampiclist-unsubscribe-requestKILLspamKILLspammitvma.mit.edu>

2000\08\21@103342 by Dan Michaels

At 07:26 AM 08/21/2000 -0400, you wrote: >So who uses assembly any more? I do almost everything in C. > >Andy > Can you get **reliable** deterministic timing in C? -- http://www.piclist.com hint: To leave the PICList spampiclist-unsubscribe-requestspam_OUTmitvma.mit.edu>

2000\08\21@103531 by Andrew Kunz

Yes. Andy Dan Michaels <STOPspamoricomspam_OUTspamBeGoneLYNX.SNI.NET> on 08/21/2000 10:32:40 AM Please respond to pic microcontroller discussion list <spam_OUTPICLISTspamBeGoneMITVMA.MIT.EDU> To: EraseMEPICLISTKILLspamMITVMA.MIT.EDU cc: (bcc: Andrew Kunz/TDI_NOTES) Subject: Re: [PIC]: Sine table At 07:26 AM 08/21/2000 -0400, you wrote: >So who uses assembly any more? I do almost everything in C. > >Andy > Can you get **reliable** deterministic timing in C? -- http://www.piclist.com hint: To leave the PICList EraseMEpiclist-unsubscribe-requestRemoveMEmitvma.mit.edu -- http://www.piclist.com hint: To leave the PICList .....piclist-unsubscribe-requestspam_OUTmitvma.mit.edu>

2000\08\21@104140 by Dan Michaels

And it's *obvious* to the programmer what it is? At 10:35 AM 08/21/2000 -0400, you wrote: {Quote hidden}>Yes. > >Andy > > > > > > > > >Dan Michaels <@spam@oricomEraseMEspamLYNX.SNI.NET> on 08/21/2000 10:32:40 AM > >Please respond to pic microcontroller discussion list <PICLISTTakeThisOuTKILLspamMITVMA.MIT.EDU> > > > > > > > > > To: RemoveMEPICLISTTakeThisOuTMITVMA.MIT.EDU > > cc: (bcc: Andrew Kunz/TDI_NOTES) > > > > Subject: Re: [PIC]: Sine table > > > > > > > > >At 07:26 AM 08/21/2000 -0400, you wrote: >>So who uses assembly any more? I do almost everything in C. >> >>Andy >> > >Can you get **reliable** deterministic timing in C? > >-- >http://www.piclist.com hint: To leave the PICList >@spam@piclist-unsubscribe-requestSTOPspammitvma.mit.edu > >-- >http://www.piclist.com hint: To leave the PICList >TakeThisOuTpiclist-unsubscribe-requestTakeThisOuTRemoveMEmitvma.mit.edu > > -- http://www.piclist.com hint: To leave the PICList spam_OUTpiclist-unsubscribe-requestspam.....mitvma.mit.edu>

2000\08\21@104557 by Andrew Kunz

Sorry, forgot to add a wink ;-) I use interrupts for long-period stuff (like the period between steps for my sine generation). When I need short items, I pad with NOPs and delay loops (inline assembly works great). Essentially the same way as with assembly code, just easier to comprehend. Andy -- http://www.piclist.com hint: To leave the PICList piclist-unsubscribe-request.....@spam@mitvma.mit.edu>

2000\08\21@104603 by Dan Michaels

Let me re-issue: And it's *obvious* to the programmer what it is? How can you tell *exactly* how many cycles a routine takes without looking at the assembly code? At 10:35 AM 08/21/2000 -0400, you wrote: {Quote hidden}>Yes. > >Andy > > > > > > > > >Dan Michaels <spamBeGoneoricomspam_OUTLYNX.SNI.NET> on 08/21/2000 10:32:40 AM > >Please respond to pic microcontroller discussion list <EraseMEPICLIST.....MITVMA.MIT.EDU> > > > > > > > > > To: spamPICLISTKILLspam@spam@MITVMA.MIT.EDU > > cc: (bcc: Andrew Kunz/TDI_NOTES) > > > > Subject: Re: [PIC]: Sine table > > > > > > > > >At 07:26 AM 08/21/2000 -0400, you wrote: >>So who uses assembly any more? I do almost everything in C. >> >>Andy >> > >Can you get **reliable** deterministic timing in C? > >-- >http://www.piclist.com hint: To leave the PICList >piclist-unsubscribe-requestspamTakeThisOuTmitvma.mit.edu > >-- >http://www.piclist.com hint: To leave the PICList >RemoveMEpiclist-unsubscribe-requestRemoveMEmitvma.mit.edu > > -- http://www.piclist.com hint: To leave the PICList TakeThisOuTpiclist-unsubscribe-request@spam@@spam@mitvma.mit.edu>

2000\08\21@110104 by Scott Dattalo

On Mon, 21 Aug 2000, Andrew Kunz wrote: > Thanks, Scott. > > That'll be interesting reading. I'm working the PLL from the angle of changing > time between steps rather than amplitude in order to keep phase locked (my only > input is a zero-crossing detector). I also have to generate multiple waves, > since I'm generating "real" power not just a 110V household socket. Perhaps, however I don't discuss the PLL part of it at all. I'm not sure how your system is implemented. On one design I had to phase lock to a ~50-60 power sine wave too. I found that zero-crossing information was unreliable. The reason was that this was part of a power quality instrument and (almost) by definition the "sine wave" is suspect. The jitter in zero crossing could be very large. So instead, I wrote a single frequency 'DFT'. In other words, I did something like this: Assume f = f0 for one cycle's worth (delta t = 1/f) of sine wave samples compute the quadrature components: sine_power = sum of (sine samples * sin(t*i)) cosine_power = sum of (sine samples * cos(t*i)) In other words, the integral of the unknown waveform multiplied by sine and cosine functions was computed. You may also view this as two narrow FIR band pass filters center about the frequency 'f'. Now you can find the arc tangent of the ratio: phase ~= arctan(sine_power/cosine_power) to find the phase. This can then server as the control input to a PID algorithm. The control output of course being the frequency. A simple proportional algorithm could go something like: If the phase is positive decrease the frequency slightly else increase the frequency slightly But beware, this can (will) have trouble locking. At the minimum, you'll need a little part of 'D' in PID. As you might expect, computing the division and then following that by arctan is expensive (but doable...). So instead, I might suggest using the raw sine_power and cosine_power terms. You may be lucky to just get away with only the sine_power term. However, if your sine wave amplitude varies then the cosine computation can help normalize the calculation. Scott -- http://www.piclist.com hint: To leave the PICList TakeThisOuTpiclist-unsubscribe-requestspammitvma.mit.edu>

2000\08\21@110304 by Dan Michaels

At 10:45 AM 08/21/2000 -0400, you wrote: >Sorry, forgot to add a wink ;-) > >I use interrupts for long-period stuff (like the period between steps for my >sine generation). When I need short items, I pad with NOPs and delay loops >(inline assembly works great). > >Essentially the same way as with assembly code, just easier to comprehend. > >Andy > This is how you get deterministic timing in C - by using inline assembly? [guess it's difficult to get a straight answer on a monday morning]. - danM -- http://www.piclist.com hint: To leave the PICList KILLspampiclist-unsubscribe-requestKILLspamspamBeGonemitvma.mit.edu>

2000\08\21@110633 by Andrew Kunz

>And it's *obvious* to the programmer what it is? > >How can you tell *exactly* how many cycles a routine takes >without looking at the assembly code? I said I don't WRITE much assembly code, but I still READ it! The HiTech C compiler gives you the C with the assembly that it generates in the .AS file. Andy -- http://www.piclist.com hint: To leave the PICList spamBeGonepiclist-unsubscribe-requestKILLspammitvma.mit.edu>

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