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#11
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![]() "Ashhar Farhan" wrote in message om... alan, the polyphase network used by hans (http://www.hanssummers.com/radio/polyphase/index.htm) is a very good idea if you _have_ to use passive components. although hans has used 1% tolerance components, you can get away with even lesser tolerence and use ordinary capacitors with 1% resistors. Thanks Farhan. I actually used 0.1% resistors, and matched the capacitors also to 0.1% by adding parallel capacitance. Of course, all that could change with temperature due to the different parallel capacitances drifting different amounts. I believe the passive polyphase network to be superior to the active phase shifting networks which use op-amps. For any given level of component tolerance, a passive network will give much better opposite sideband suppression (I recall seeing 10-20dB reported somewhere but can't provide references). Provided attention is paid to the values used, the network can be made lossless which overcomes any concerns about gain distribution and harming the overall receiver dynamic range. So the way I tend to put it, is if you _have_ to use active components ... 73 de Hans G0UPL http://www.HansSummers.com |
#12
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Alan Peake wrote:
Hi all, I'm looking for the element values of the 2Q4 phase shifter as described in the 1992 ARRL Handbook. Alternatively, can anyone recommend an active (analog) all-pass that would give the same or better results. I have some precision capacitors so that's no problem. Thanks, Alan VK2TWB These days the way to do it would be to use a DSP. If this is for a receiver the DSP would have two A/D converters and a single D/A unit. The DSP would combine the two quadature signals and apply bandwidth filtering. The quadature RF drive to the mixer would come from a DDS circuit with both sine and cosine waveform outputs (AD9853/54). For a transmitter the DSP would have a single A/D and two D/A stages, and would split the audio into two quadature signals after applying bandwidth limiting and compression filtering. Having said this, I wish I knew how to write the required DSP software! However I'm sure there are some reading this list with the required skill (talent?). (My software experience lies in other areas, such as embedded controllers, NOT math with imaginary numbers!). |
#13
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Alan Peake wrote:
Hi all, I'm looking for the element values of the 2Q4 phase shifter as described in the 1992 ARRL Handbook. Alternatively, can anyone recommend an active (analog) all-pass that would give the same or better results. I have some precision capacitors so that's no problem. Thanks, Alan VK2TWB These days the way to do it would be to use a DSP. If this is for a receiver the DSP would have two A/D converters and a single D/A unit. The DSP would combine the two quadature signals and apply bandwidth filtering. The quadature RF drive to the mixer would come from a DDS circuit with both sine and cosine waveform outputs (AD9853/54). For a transmitter the DSP would have a single A/D and two D/A stages, and would split the audio into two quadature signals after applying bandwidth limiting and compression filtering. Having said this, I wish I knew how to write the required DSP software! However I'm sure there are some reading this list with the required skill (talent?). (My software experience lies in other areas, such as embedded controllers, NOT math with imaginary numbers!). |
#14
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Alan Peake wrote:
Having said this, I wish I knew how to write the required DSP software! However I'm sure there are some reading this list with the required skill (talent?). (My software experience lies in other areas, such as embedded controllers, NOT math with imaginary numbers!). alan, you don't really have to know a lot of DSP to play around with this particular beast. very simply, you collect the audio samples in a first-in first-out buffer of about 250 slots. Each time a new sample is added at one end, a sample is retired at the other end. each of the 90 degree phase shift-ed samples is generated by simpy multiplying all the samples in the pipe with a individual 'magic' constants and adding them all up together. pretty basic stuff as far as programming goes. the magic constants are themselves quite complex to calculated, but that work has alread been done for you. The CD accompanying EMRFD has those constants in a text file under the DSP folder. it is really simple. all the controls are soft and you can play with a bunch of things. if you were considering the analog route, i think polyphase approach is simply the best : it is simple and without any tune-up and the results are on par with the best DSP can offer. - farhan |
#15
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Alan Peake wrote:
Having said this, I wish I knew how to write the required DSP software! However I'm sure there are some reading this list with the required skill (talent?). (My software experience lies in other areas, such as embedded controllers, NOT math with imaginary numbers!). alan, you don't really have to know a lot of DSP to play around with this particular beast. very simply, you collect the audio samples in a first-in first-out buffer of about 250 slots. Each time a new sample is added at one end, a sample is retired at the other end. each of the 90 degree phase shift-ed samples is generated by simpy multiplying all the samples in the pipe with a individual 'magic' constants and adding them all up together. pretty basic stuff as far as programming goes. the magic constants are themselves quite complex to calculated, but that work has alread been done for you. The CD accompanying EMRFD has those constants in a text file under the DSP folder. it is really simple. all the controls are soft and you can play with a bunch of things. if you were considering the analog route, i think polyphase approach is simply the best : it is simple and without any tune-up and the results are on par with the best DSP can offer. - farhan |
#16
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![]() Thanks for the suggestions. I'm reluctant to use the computer as I'm on solar power and can't use the machine for too long, particularly on cloudy days in winter - such as today. I do have some DSP stuff though - ADSP2100- but as it's fixed point, I'm not sure if it has the precision needed. I really want to use this an exciter and while filters are probably easier, the phasing method always struck me as more "elegant". 73s Alan |
#17
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![]() Thanks for the suggestions. I'm reluctant to use the computer as I'm on solar power and can't use the machine for too long, particularly on cloudy days in winter - such as today. I do have some DSP stuff though - ADSP2100- but as it's fixed point, I'm not sure if it has the precision needed. I really want to use this an exciter and while filters are probably easier, the phasing method always struck me as more "elegant". 73s Alan |
#18
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![]() Thanks Farhan. I actually used 0.1% resistors, and matched the capacitors also to 0.1% by adding parallel capacitance. Of course, all that could change with temperature due to the different parallel capacitances drifting different amounts. This may be why the 2Q4 network is enclosed in a can. The can may also be filled with something like thermal grease to keep all the components at the same temperature. Alan |
#19
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![]() Thanks Farhan. I actually used 0.1% resistors, and matched the capacitors also to 0.1% by adding parallel capacitance. Of course, all that could change with temperature due to the different parallel capacitances drifting different amounts. This may be why the 2Q4 network is enclosed in a can. The can may also be filled with something like thermal grease to keep all the components at the same temperature. Alan |
#20
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![]() The 2Q4 was an 8-pin plug-in (octal?) and this is how it is shown in the 51SB-B sideband generator schematic. Pin 1 - 680pF - 487k - Pin 2 - 770k||430pF - Pin 3 Pin 5 - 680pF - 125k - Pin 6 - 198k||430pF - Pin 7 Pins 1 & 5 were strapped and fed with one side of a balanced, band- limited audio input and 3 & 7 (also strapped) with the other. Phase- shifted outputs were then taken from 2 & 6. I guess that 4 or 8 could have been a grounded shell. I haven't worked it out but wouldn't be surprised if these are not just Wein Bridge values for a certain frequency. Cheers - Joe Many thanks Joe - I'll put those values into my simulator and see what comes out. Cheers, Alan |
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