>On Fri, 17 Apr 2015 14:52:32 -0500, "tomb18" <103312@DSPRelated> >wrote: > >>>I'm not going to fix your top-posting, so this is just going to be a >>>mess from here if the thread continues. >>> >>>You're doing things the hard way, IMHO. >>> >>>You don't need to take an FFT and then an iFFT. Take the FFT, use >>>that for your spectrum analysis or whatever. Take the same signal >>>that went into the FFT and mix whatever part of that you want to be >at >>>zero-Hz with a complex mix operation (i.e., multiply sample-by-sample >>>with a complex tone at the appropriate frequency). >>> >>>You can then apply a low-pass filter to isolate the signal that you >>>want before detecting the magnitude. >>> >>>That'll save a ton of computation and avoid all the hassles with >>>overlap-add, etc. A LPF is much easier to design and implement than >>>a BPF. >>> >>> >>> >>>On Fri, 17 Apr 2015 10:41:17 -0500, "tomb18" <103312@DSPRelated> >>>wrote: >>> >>>>Many thanks to all that replied! >>>>Starting here I will explain this a little more. >>>>1)Yes the FFT input buffer is filled with complex samples. >>>>2)I take the complex FFT and output it to the inverse FFT. I also >>>take >>>>the complex FFT and plot it in a spectrum (this is independent of >the >>>>demodulation). >>>>3) I take the complex values in the output buffer and convert to >>>polar. >>>>4) I only take the magnitude. >>>> >>>>The first point is better explained by my overall goal which is to >>>provide >>>>a spectrum scope for a software defined radio. This is functional >and >>>>works very well. I now wish to isolate signals (SSB or AM) from the >>>>spectrum, which can be anywhere over the whole range of 192kHz (I >>>sample >>>>at 192kHz and plot positive and negative frequencies. The radio >>>provides >>>>a quadrature output). >>>>So my test application is just demodulating signals around 0-3kHz >i.e. >>>in >>>>the center of the spectrum. By tuning the radio manually, I can get >>>the >>>>signals down to this frequency range and then demodulate. >>>>So the overall flow is: >>>>Stereo audio (I and Q fed into left and right channels), Sampling at >>>>192kHz, >>>>FFT with blackman, plotting of spectrum (with IQ balance control). >>>>Now I can take the FFT with rectangular windowing, inverse FFT, then >>>for >>>>AM I convert to polar and take the magnitude. Then it's off to the >>>audio >>>>out. >>>>For USB I take the output of the inverse fourier, take the imaginary >>>>values, send them through a hilbert transform and subtract the >shifted >>>>values from the real.Then it's out to the audio. For LSB its the >same >>>>except it's added. >>>>All of this works well. >>>>So my original question is why did the audio become distorted when I >>>used >>>>a blackman window. That has been explained by Randy. >>>> >>>>My ultimate goal however (and I am a beginner here!) is to isolate a >>>>signal in the frequency domain and bring it down to 0 hz. >>>>Next I need a bandpass filter. I then need to pass this filter >through >>>an >>>>inverse FFT, apply windowing to this filter. Next do an FFT on this >>>>filter and multiply it with the original signals FFT. After this I >do >>>an >>>>iFFT with the overlap add procedure. Whew! This is why I started >>>with >>>>the simple case above. >>>>So, my next questions are about the bandpass filter. How does one >>>create >>>>one, and also how does one apply an iFFT to it? >>>>Assuming that there is an equation for the bandpass filter, does one >>>just >>>>create a buffer containing the results of the filter from -0.5 to >+0.5 >>>and >>>>using the value of "1" as the input? Do you then send this buffer >to >>>the >>>>inverse FFT? I think this is the one area where I have not really >seen >>>an >>>>explanation of this. >>>>I really appreciate the help here as things are not always clear. >>>>Regards, Tom >>> >>> >>>>>Your question is missing many details. I assume you are doing the >>>>>following >>>>> >>>>>1) fill the fft input buffer with complex samples >>>>>2) fft, then inverse fft >>>>>3) take the complex values in the output buffer and convert to >>>>>polar. >>>>> >>>>>While you didn't say this, I assume the output you use is just the >>>>>magnitude portion of the polar output (the angle is thrown away so >>>don't >>>>bother >>>>>computing it) >>>>> >>>>>So, a few points. >>>>>The fft/inverse fft just cancel each other so the output buffer >>>equals >>>>>the input buffer. Were you planning to do some filtering in the >>>>frequency >>>>>domain ? If not then the whole scheme is just wasting the >fft/inverse >>>fft >>>>and >>>>>you could just compute the magnitude of the original complex signal >on >>>a >>>>>sample-by-sample basis. >>>>> >>>>>If you ARE planning on doing some filtering in the frequency domain >>>then >>>>>you need to use the conventional overlap-add method of >reconstructing >>>>the >>>>>time-domain signal, followed by taking the magnitude of the >resulting >>>>>samples. >>>>> >>>>>Note that none of the above can be described as AM demodulation in >>>the >>>>>frequency domain, since after the ifft you are back in the time >>>domain. >>>>I've >>>>>never heard of AM demodulation in the frequency domain, perhaps it >>>>exists, >>>>>I'm not a comms expert. >>>>> >>>>>Bob >>>> >>>> >>>>--------------------------------------- >>>>Posted through http://www.DSPRelated.com >>> >>>Eric Jacobsen >>>Anchor Hill Communications >>>http://www.anchorhill.com >> >>Hi, >>Sorry about the top posting....I subscribe to mailing lists often and >the >>ones I subscribe to suggest top posting. >>Well I'll be....you are right. I just tried this taking the IQ signal >off >>before it goes into the FFT and as expected, all the demodulation I >did >>works exactly the same. No need for an inverse FFT. Not only that, >>getting rid of the iFFT means there is now no noticeable delay. Much >>better. >>At the moment, I am just tuning the radio to get the signal to where I >can >>hear it. Next, I need to do the multiplication with the complex tone >but >>I have a question that will really show how little I know about all of >>this, lol. >>The IQ signals come from the sound card. The sampling is done at >192kHz >>and 4096 byte blocks are delivered at a time. I presume I just need >to >>deal with one block at a time? Can I just multiply each signal ( I and >Q) >>with a sine and cosine? Do I need to match the magnitudes in any way? > >>Like I said, here i really don't know what I'm doing. >>Thanks, Tom >>--------------------------------------- >>Posted through http://www.DSPRelated.com > >Assuming the time between the blocks is the same as the time between >the samples, it will be better to keep the phase of the mixer >continuous across the edge of the blocks. In other words, don't >reset the phase at the beginning of each block, or you may hear a pop >or other artifact at the block edge. > >This may be less likely since you're ultimately demodulating AM, but >if you do apply some filtering before detecting the AM you can wind up >with artifacts at the block edges if you're not careful. > > >Eric Jacobsen >Anchor Hill Communications >http://www.anchorhill.comFabulous. in this one day I understood more that I have in weeks of reading. It sure helps to have people answer questions. So, I will be getting 4k blocks of I and Q data which are floats, I will be starting an oscillator at my sampling rate (192kHz)and then for each data point I do the multiplication with the oscillator value. At the next block I continue, without resetting the oscillator if I read this correctly. I have spent much time learning about applying filters in the frequency domain and never thought I should just look at the time domain directly. Much simpler. Thanks for all the help, one day I hope to repay the favour to others. Now unfortunately, I do have another problem that i haven't yet solved with a filter in the frequency domain that cannot be done in the time domain. This one involves frequency dependent IQ gain and phase balance. I'm almost there, but that's another story. Pretty soon I think everything will just click and fall in place. Tom --------------------------------------- Posted through http://www.DSPRelated.com
Audio output distorted with any FFT windowing other than rectangular
Started by ●April 16, 2015
Reply by ●April 17, 20152015-04-17
Reply by ●April 18, 20152015-04-18
>On Fri, 17 Apr 2015 14:52:32 -0500, "tomb18" <103312@DSPRelated> >wrote: > >>>I'm not going to fix your top-posting, so this is just going to be a >>>mess from here if the thread continues. >>> >>>You're doing things the hard way, IMHO. >>> >>>You don't need to take an FFT and then an iFFT. Take the FFT, use >>>that for your spectrum analysis or whatever. Take the same signal >>>that went into the FFT and mix whatever part of that you want to be >at >>>zero-Hz with a complex mix operation (i.e., multiply sample-by-sample >>>with a complex tone at the appropriate frequency). >>> >>>You can then apply a low-pass filter to isolate the signal that you >>>want before detecting the magnitude. >>> >>>That'll save a ton of computation and avoid all the hassles with >>>overlap-add, etc. A LPF is much easier to design and implement than >>>a BPF. >>> >>> >>> >>>On Fri, 17 Apr 2015 10:41:17 -0500, "tomb18" <103312@DSPRelated> >>>wrote: >>> >>>>Many thanks to all that replied! >>>>Starting here I will explain this a little more. >>>>1)Yes the FFT input buffer is filled with complex samples. >>>>2)I take the complex FFT and output it to the inverse FFT. I also >>>take >>>>the complex FFT and plot it in a spectrum (this is independent of >the >>>>demodulation). >>>>3) I take the complex values in the output buffer and convert to >>>polar. >>>>4) I only take the magnitude. >>>> >>>>The first point is better explained by my overall goal which is to >>>provide >>>>a spectrum scope for a software defined radio. This is functional >and >>>>works very well. I now wish to isolate signals (SSB or AM) from the >>>>spectrum, which can be anywhere over the whole range of 192kHz (I >>>sample >>>>at 192kHz and plot positive and negative frequencies. The radio >>>provides >>>>a quadrature output). >>>>So my test application is just demodulating signals around 0-3kHz >i.e. >>>in >>>>the center of the spectrum. By tuning the radio manually, I can get >>>the >>>>signals down to this frequency range and then demodulate. >>>>So the overall flow is: >>>>Stereo audio (I and Q fed into left and right channels), Sampling at >>>>192kHz, >>>>FFT with blackman, plotting of spectrum (with IQ balance control). >>>>Now I can take the FFT with rectangular windowing, inverse FFT, then >>>for >>>>AM I convert to polar and take the magnitude. Then it's off to the >>>audio >>>>out. >>>>For USB I take the output of the inverse fourier, take the imaginary >>>>values, send them through a hilbert transform and subtract the >shifted >>>>values from the real.Then it's out to the audio. For LSB its the >same >>>>except it's added. >>>>All of this works well. >>>>So my original question is why did the audio become distorted when I >>>used >>>>a blackman window. That has been explained by Randy. >>>> >>>>My ultimate goal however (and I am a beginner here!) is to isolate a >>>>signal in the frequency domain and bring it down to 0 hz. >>>>Next I need a bandpass filter. I then need to pass this filter >through >>>an >>>>inverse FFT, apply windowing to this filter. Next do an FFT on this >>>>filter and multiply it with the original signals FFT. After this I >do >>>an >>>>iFFT with the overlap add procedure. Whew! This is why I started >>>with >>>>the simple case above. >>>>So, my next questions are about the bandpass filter. How does one >>>create >>>>one, and also how does one apply an iFFT to it? >>>>Assuming that there is an equation for the bandpass filter, does one >>>just >>>>create a buffer containing the results of the filter from -0.5 to >+0.5 >>>and >>>>using the value of "1" as the input? Do you then send this buffer >to >>>the >>>>inverse FFT? I think this is the one area where I have not really >seen >>>an >>>>explanation of this. >>>>I really appreciate the help here as things are not always clear. >>>>Regards, Tom >>> >>> >>>>>Your question is missing many details. I assume you are doing the >>>>>following >>>>> >>>>>1) fill the fft input buffer with complex samples >>>>>2) fft, then inverse fft >>>>>3) take the complex values in the output buffer and convert to >>>>>polar. >>>>> >>>>>While you didn't say this, I assume the output you use is just the >>>>>magnitude portion of the polar output (the angle is thrown away so >>>don't >>>>bother >>>>>computing it) >>>>> >>>>>So, a few points. >>>>>The fft/inverse fft just cancel each other so the output buffer >>>equals >>>>>the input buffer. Were you planning to do some filtering in the >>>>frequency >>>>>domain ? If not then the whole scheme is just wasting the >fft/inverse >>>fft >>>>and >>>>>you could just compute the magnitude of the original complex signal >on >>>a >>>>>sample-by-sample basis. >>>>> >>>>>If you ARE planning on doing some filtering in the frequency domain >>>then >>>>>you need to use the conventional overlap-add method of >reconstructing >>>>the >>>>>time-domain signal, followed by taking the magnitude of the >resulting >>>>>samples. >>>>> >>>>>Note that none of the above can be described as AM demodulation in >>>the >>>>>frequency domain, since after the ifft you are back in the time >>>domain. >>>>I've >>>>>never heard of AM demodulation in the frequency domain, perhaps it >>>>exists, >>>>>I'm not a comms expert. >>>>> >>>>>Bob >>>> >>>> >>>>--------------------------------------- >>>>Posted through http://www.DSPRelated.com >>> >>>Eric Jacobsen >>>Anchor Hill Communications >>>http://www.anchorhill.com >> >>Hi, >>Sorry about the top posting....I subscribe to mailing lists often and >the >>ones I subscribe to suggest top posting. >>Well I'll be....you are right. I just tried this taking the IQ signal >off >>before it goes into the FFT and as expected, all the demodulation I >did >>works exactly the same. No need for an inverse FFT. Not only that, >>getting rid of the iFFT means there is now no noticeable delay. Much >>better. >>At the moment, I am just tuning the radio to get the signal to where I >can >>hear it. Next, I need to do the multiplication with the complex tone >but >>I have a question that will really show how little I know about all of >>this, lol. >>The IQ signals come from the sound card. The sampling is done at >192kHz >>and 4096 byte blocks are delivered at a time. I presume I just need >to >>deal with one block at a time? Can I just multiply each signal ( I and >Q) >>with a sine and cosine? Do I need to match the magnitudes in any way? > >>Like I said, here i really don't know what I'm doing. >>Thanks, Tom >>--------------------------------------- >>Posted through http://www.DSPRelated.com > >Assuming the time between the blocks is the same as the time between >the samples, it will be better to keep the phase of the mixer >continuous across the edge of the blocks. In other words, don't >reset the phase at the beginning of each block, or you may hear a pop >or other artifact at the block edge. > >This may be less likely since you're ultimately demodulating AM, but >if you do apply some filtering before detecting the AM you can wind up >with artifacts at the block edges if you're not careful. > > >Eric Jacobsen >Anchor Hill Communications >http://www.anchorhill.comWell I have a prototype. I have an oscillator, and I am multiplying each sample with the oscillator step. It works. It allows me to select a frequency for the oscillator and thus tune in to a signal in the spectrum. The low pass filters(one on the real and one on the imaginary) , work. However, it seems that I hear many tones as I tune through the spectrum in AM mode. When I get on a signal I have to tune the radio a bit until the tone goes lower until it disappears but there is a point where it is inaudible but I can hear a beat in the signal. It seems it is acting as a synchronous detector. In USB, I do not hear a tone. But if I just listen to a properly tuned signal, it appears to slowly waver in pitch. I'm not quite sure what is happening here. Any ideas? --------------------------------------- Posted through http://www.DSPRelated.com
Reply by ●April 19, 20152015-04-19
"tomb18" <103312@DSPRelated> writes:>>On Fri, 17 Apr 2015 14:52:32 -0500, "tomb18" <103312@DSPRelated> >>wrote: >> >>>>I'm not going to fix your top-posting, so this is just going to be a >>>>mess from here if the thread continues. >>>> >>>>You're doing things the hard way, IMHO. >>>> >>>>You don't need to take an FFT and then an iFFT. Take the FFT, use >>>>that for your spectrum analysis or whatever. Take the same signal >>>>that went into the FFT and mix whatever part of that you want to be >>at >>>>zero-Hz with a complex mix operation (i.e., multiply sample-by-sample >>>>with a complex tone at the appropriate frequency). >>>> >>>>You can then apply a low-pass filter to isolate the signal that you >>>>want before detecting the magnitude. >>>> >>>>That'll save a ton of computation and avoid all the hassles with >>>>overlap-add, etc. A LPF is much easier to design and implement than >>>>a BPF. >>>> >>>> >>>> >>>>On Fri, 17 Apr 2015 10:41:17 -0500, "tomb18" <103312@DSPRelated> >>>>wrote: >>>> >>>>>Many thanks to all that replied! >>>>>Starting here I will explain this a little more. >>>>>1)Yes the FFT input buffer is filled with complex samples. >>>>>2)I take the complex FFT and output it to the inverse FFT. I also >>>>take >>>>>the complex FFT and plot it in a spectrum (this is independent of >>the >>>>>demodulation). >>>>>3) I take the complex values in the output buffer and convert to >>>>polar. >>>>>4) I only take the magnitude. >>>>> >>>>>The first point is better explained by my overall goal which is to >>>>provide >>>>>a spectrum scope for a software defined radio. This is functional >>and >>>>>works very well. I now wish to isolate signals (SSB or AM) from the >>>>>spectrum, which can be anywhere over the whole range of 192kHz (I >>>>sample >>>>>at 192kHz and plot positive and negative frequencies. The radio >>>>provides >>>>>a quadrature output). >>>>>So my test application is just demodulating signals around 0-3kHz >>i.e. >>>>in >>>>>the center of the spectrum. By tuning the radio manually, I can get >>>>the >>>>>signals down to this frequency range and then demodulate. >>>>>So the overall flow is: >>>>>Stereo audio (I and Q fed into left and right channels), Sampling at >>>>>192kHz, >>>>>FFT with blackman, plotting of spectrum (with IQ balance control). >>>>>Now I can take the FFT with rectangular windowing, inverse FFT, then >>>>for >>>>>AM I convert to polar and take the magnitude. Then it's off to the >>>>audio >>>>>out. >>>>>For USB I take the output of the inverse fourier, take the imaginary >>>>>values, send them through a hilbert transform and subtract the >>shifted >>>>>values from the real.Then it's out to the audio. For LSB its the >>same >>>>>except it's added. >>>>>All of this works well. >>>>>So my original question is why did the audio become distorted when I >>>>used >>>>>a blackman window. That has been explained by Randy. >>>>> >>>>>My ultimate goal however (and I am a beginner here!) is to isolate a >>>>>signal in the frequency domain and bring it down to 0 hz. >>>>>Next I need a bandpass filter. I then need to pass this filter >>through >>>>an >>>>>inverse FFT, apply windowing to this filter. Next do an FFT on this >>>>>filter and multiply it with the original signals FFT. After this I >>do >>>>an >>>>>iFFT with the overlap add procedure. Whew! This is why I started >>>>with >>>>>the simple case above. >>>>>So, my next questions are about the bandpass filter. How does one >>>>create >>>>>one, and also how does one apply an iFFT to it? >>>>>Assuming that there is an equation for the bandpass filter, does one >>>>just >>>>>create a buffer containing the results of the filter from -0.5 to >>+0.5 >>>>and >>>>>using the value of "1" as the input? Do you then send this buffer >>to >>>>the >>>>>inverse FFT? I think this is the one area where I have not really >>seen >>>>an >>>>>explanation of this. >>>>>I really appreciate the help here as things are not always clear. >>>>>Regards, Tom >>>> >>>> >>>>>>Your question is missing many details. I assume you are doing the >>>>>>following >>>>>> >>>>>>1) fill the fft input buffer with complex samples >>>>>>2) fft, then inverse fft >>>>>>3) take the complex values in the output buffer and convert to >>>>>>polar. >>>>>> >>>>>>While you didn't say this, I assume the output you use is just the >>>>>>magnitude portion of the polar output (the angle is thrown away so >>>>don't >>>>>bother >>>>>>computing it) >>>>>> >>>>>>So, a few points. >>>>>>The fft/inverse fft just cancel each other so the output buffer >>>>equals >>>>>>the input buffer. Were you planning to do some filtering in the >>>>>frequency >>>>>>domain ? If not then the whole scheme is just wasting the >>fft/inverse >>>>fft >>>>>and >>>>>>you could just compute the magnitude of the original complex signal >>on >>>>a >>>>>>sample-by-sample basis. >>>>>> >>>>>>If you ARE planning on doing some filtering in the frequency domain >>>>then >>>>>>you need to use the conventional overlap-add method of >>reconstructing >>>>>the >>>>>>time-domain signal, followed by taking the magnitude of the >>resulting >>>>>>samples. >>>>>> >>>>>>Note that none of the above can be described as AM demodulation in >>>>the >>>>>>frequency domain, since after the ifft you are back in the time >>>>domain. >>>>>I've >>>>>>never heard of AM demodulation in the frequency domain, perhaps it >>>>>exists, >>>>>>I'm not a comms expert. >>>>>> >>>>>>Bob >>>>> >>>>> >>>>>--------------------------------------- >>>>>Posted through http://www.DSPRelated.com >>>> >>>>Eric Jacobsen >>>>Anchor Hill Communications >>>>http://www.anchorhill.com >>> >>>Hi, >>>Sorry about the top posting....I subscribe to mailing lists often and >>the >>>ones I subscribe to suggest top posting. >>>Well I'll be....you are right. I just tried this taking the IQ signal >>off >>>before it goes into the FFT and as expected, all the demodulation I >>did >>>works exactly the same. No need for an inverse FFT. Not only that, >>>getting rid of the iFFT means there is now no noticeable delay. Much >>>better. >>>At the moment, I am just tuning the radio to get the signal to where I >>can >>>hear it. Next, I need to do the multiplication with the complex tone >>but >>>I have a question that will really show how little I know about all of >>>this, lol. >>>The IQ signals come from the sound card. The sampling is done at >>192kHz >>>and 4096 byte blocks are delivered at a time. I presume I just need >>to >>>deal with one block at a time? Can I just multiply each signal ( I and >>Q) >>>with a sine and cosine? Do I need to match the magnitudes in any way? >> >>>Like I said, here i really don't know what I'm doing. >>>Thanks, Tom >>>--------------------------------------- >>>Posted through http://www.DSPRelated.com >> >>Assuming the time between the blocks is the same as the time between >>the samples, it will be better to keep the phase of the mixer >>continuous across the edge of the blocks. In other words, don't >>reset the phase at the beginning of each block, or you may hear a pop >>or other artifact at the block edge. >> >>This may be less likely since you're ultimately demodulating AM, but >>if you do apply some filtering before detecting the AM you can wind up >>with artifacts at the block edges if you're not careful. >> >> >>Eric Jacobsen >>Anchor Hill Communications >>http://www.anchorhill.com > > Well I have a prototype. I have an oscillator, and I am multiplying each > sample with the oscillator step. It works. It allows me to select a > frequency for the oscillator and thus tune in to a signal in the spectrum. > The low pass filters(one on the real and one on the imaginary) , work. > However, it seems that I hear many tones as I tune through the spectrum > in AM mode. When I get on a signal I have to tune the radio a bit until > the tone goes lower until it disappears but there is a point where it is > inaudible but I can hear a beat in the signal. It seems it is acting as a > synchronous detector.If the AM modulation has a carrier component, then this would produce the tones. The difference between the actual carrier and your "digital oscillator" frequency will be a tone present in the demodulated output.> In USB, I do not hear a tone. But if I just listen to a properly tuned > signal, it appears to slowly waver in pitch. > I'm not quite sure what is happening here. Any ideas?Your carrier and digital oscillator are slowly varying relative to one another. -- Randy Yates Digital Signal Labs http://www.digitalsignallabs.com
Reply by ●April 19, 20152015-04-19
>"tomb18" <103312@DSPRelated> writes: > >>>On Fri, 17 Apr 2015 14:52:32 -0500, "tomb18" <103312@DSPRelated> >>>wrote: >>> >>>>>I'm not going to fix your top-posting, so this is just going to be >a >>>>>mess from here if the thread continues. >>>>> >>>>>You're doing things the hard way, IMHO. >>>>> >>>>>You don't need to take an FFT and then an iFFT. Take the FFT, >use >>>>>that for your spectrum analysis or whatever. Take the same >signal >>>>>that went into the FFT and mix whatever part of that you want to be >>>at >>>>>zero-Hz with a complex mix operation (i.e., multiply >sample-by-sample >>>>>with a complex tone at the appropriate frequency). >>>>> >>>>>You can then apply a low-pass filter to isolate the signal that you >>>>>want before detecting the magnitude. >>>>> >>>>>That'll save a ton of computation and avoid all the hassles with >>>>>overlap-add, etc. A LPF is much easier to design and implement >than >>>>>a BPF. >>>>> >>>>> >>>>> >>>>>On Fri, 17 Apr 2015 10:41:17 -0500, "tomb18" <103312@DSPRelated> >>>>>wrote: >>>>> >>>>>>Many thanks to all that replied! >>>>>>Starting here I will explain this a little more. >>>>>>1)Yes the FFT input buffer is filled with complex samples. >>>>>>2)I take the complex FFT and output it to the inverse FFT. I also >>>>>take >>>>>>the complex FFT and plot it in a spectrum (this is independent of >>>the >>>>>>demodulation). >>>>>>3) I take the complex values in the output buffer and convert to >>>>>polar. >>>>>>4) I only take the magnitude. >>>>>> >>>>>>The first point is better explained by my overall goal which is to >>>>>provide >>>>>>a spectrum scope for a software defined radio. This is functional >>>and >>>>>>works very well. I now wish to isolate signals (SSB or AM) from >the >>>>>>spectrum, which can be anywhere over the whole range of 192kHz (I >>>>>sample >>>>>>at 192kHz and plot positive and negative frequencies. The radio >>>>>provides >>>>>>a quadrature output). >>>>>>So my test application is just demodulating signals around 0-3kHz >>>i.e. >>>>>in >>>>>>the center of the spectrum. By tuning the radio manually, I can >get >>>>>the >>>>>>signals down to this frequency range and then demodulate. >>>>>>So the overall flow is: >>>>>>Stereo audio (I and Q fed into left and right channels), Sampling >at >>>>>>192kHz, >>>>>>FFT with blackman, plotting of spectrum (with IQ balance control). >>>>>>Now I can take the FFT with rectangular windowing, inverse FFT, >then >>>>>for >>>>>>AM I convert to polar and take the magnitude. Then it's off to >the >>>>>audio >>>>>>out. >>>>>>For USB I take the output of the inverse fourier, take the >imaginary >>>>>>values, send them through a hilbert transform and subtract the >>>shifted >>>>>>values from the real.Then it's out to the audio. For LSB its the >>>same >>>>>>except it's added. >>>>>>All of this works well. >>>>>>So my original question is why did the audio become distorted when >I >>>>>used >>>>>>a blackman window. That has been explained by Randy. >>>>>> >>>>>>My ultimate goal however (and I am a beginner here!) is to isolate >a >>>>>>signal in the frequency domain and bring it down to 0 hz. >>>>>>Next I need a bandpass filter. I then need to pass this filter >>>through >>>>>an >>>>>>inverse FFT, apply windowing to this filter. Next do an FFT on >this >>>>>>filter and multiply it with the original signals FFT. After this >I >>>do >>>>>an >>>>>>iFFT with the overlap add procedure. Whew! This is why I started >>>>>with >>>>>>the simple case above. >>>>>>So, my next questions are about the bandpass filter. How does one >>>>>create >>>>>>one, and also how does one apply an iFFT to it? >>>>>>Assuming that there is an equation for the bandpass filter, does >one >>>>>just >>>>>>create a buffer containing the results of the filter from -0.5 to >>>+0.5 >>>>>and >>>>>>using the value of "1" as the input? Do you then send this buffer >>>to >>>>>the >>>>>>inverse FFT? I think this is the one area where I have not really >>>seen >>>>>an >>>>>>explanation of this. >>>>>>I really appreciate the help here as things are not always clear. >>>>>>Regards, Tom >>>>> >>>>> >>>>>>>Your question is missing many details. I assume you are doing the >>>>>>>following >>>>>>> >>>>>>>1) fill the fft input buffer with complex samples >>>>>>>2) fft, then inverse fft >>>>>>>3) take the complex values in the output buffer and convert to >>>>>>>polar. >>>>>>> >>>>>>>While you didn't say this, I assume the output you use is just >the >>>>>>>magnitude portion of the polar output (the angle is thrown away >so >>>>>don't >>>>>>bother >>>>>>>computing it) >>>>>>> >>>>>>>So, a few points. >>>>>>>The fft/inverse fft just cancel each other so the output buffer >>>>>equals >>>>>>>the input buffer. Were you planning to do some filtering in the >>>>>>frequency >>>>>>>domain ? If not then the whole scheme is just wasting the >>>fft/inverse >>>>>fft >>>>>>and >>>>>>>you could just compute the magnitude of the original complex >signal >>>on >>>>>a >>>>>>>sample-by-sample basis. >>>>>>> >>>>>>>If you ARE planning on doing some filtering in the frequency >domain >>>>>then >>>>>>>you need to use the conventional overlap-add method of >>>reconstructing >>>>>>the >>>>>>>time-domain signal, followed by taking the magnitude of the >>>resulting >>>>>>>samples. >>>>>>> >>>>>>>Note that none of the above can be described as AM demodulation >in >>>>>the >>>>>>>frequency domain, since after the ifft you are back in the time >>>>>domain. >>>>>>I've >>>>>>>never heard of AM demodulation in the frequency domain, perhaps >it >>>>>>exists, >>>>>>>I'm not a comms expert. >>>>>>> >>>>>>>Bob >>>>>> >>>>>> >>>>>>--------------------------------------- >>>>>>Posted through http://www.DSPRelated.com >>>>> >>>>>Eric Jacobsen >>>>>Anchor Hill Communications >>>>>http://www.anchorhill.com >>>> >>>>Hi, >>>>Sorry about the top posting....I subscribe to mailing lists often >and >>>the >>>>ones I subscribe to suggest top posting. >>>>Well I'll be....you are right. I just tried this taking the IQ >signal >>>off >>>>before it goes into the FFT and as expected, all the demodulation I >>>did >>>>works exactly the same. No need for an inverse FFT. Not only that, >>>>getting rid of the iFFT means there is now no noticeable delay. >Much >>>>better. >>>>At the moment, I am just tuning the radio to get the signal to where >I >>>can >>>>hear it. Next, I need to do the multiplication with the complex >tone >>>but >>>>I have a question that will really show how little I know about all >of >>>>this, lol. >>>>The IQ signals come from the sound card. The sampling is done at >>>192kHz >>>>and 4096 byte blocks are delivered at a time. I presume I just need >>>to >>>>deal with one block at a time? Can I just multiply each signal ( I >and >>>Q) >>>>with a sine and cosine? Do I need to match the magnitudes in any >way? >>> >>>>Like I said, here i really don't know what I'm doing. >>>>Thanks, Tom >>>>--------------------------------------- >>>>Posted through http://www.DSPRelated.com >>> >>>Assuming the time between the blocks is the same as the time between >>>the samples, it will be better to keep the phase of the mixer >>>continuous across the edge of the blocks. In other words, don't >>>reset the phase at the beginning of each block, or you may hear a pop >>>or other artifact at the block edge. >>> >>>This may be less likely since you're ultimately demodulating AM, but >>>if you do apply some filtering before detecting the AM you can wind >up >>>with artifacts at the block edges if you're not careful. >>> >>> >>>Eric Jacobsen >>>Anchor Hill Communications >>>http://www.anchorhill.com >> >> Well I have a prototype. I have an oscillator, and I am multiplying >each >> sample with the oscillator step. It works. It allows me to select a >> frequency for the oscillator and thus tune in to a signal in the >spectrum. >> The low pass filters(one on the real and one on the imaginary) , >work. >> However, it seems that I hear many tones as I tune through the >spectrum >> in AM mode. When I get on a signal I have to tune the radio a bit >until >> the tone goes lower until it disappears but there is a point where it >is >> inaudible but I can hear a beat in the signal. It seems it is acting >as a >> synchronous detector. > >If the AM modulation has a carrier component, then this would produce >the tones. The difference between the actual carrier and your "digital >oscillator" frequency will be a tone present in the demodulated output. > >> In USB, I do not hear a tone. But if I just listen to a properly >tuned >> signal, it appears to slowly waver in pitch. >> I'm not quite sure what is happening here. Any ideas? > >Your carrier and digital oscillator are slowly varying relative to one >another. >-- >Randy Yates >Digital Signal Labs >http://www.digitalsignallabs.comThat's what I thought. However, on further investigation I believe it's images from the IQ imbalance. If I have a fixed oscillator frequency, as I tune a signal, I see an image in the spectrum. As I tune, the image gets closer to the carrier, the pitch lowers, and when it completely overlaps the carrier the pitch is gone. Well, this brings me right back to working in the frequency domain. My spectrum scope requires IQ imbalance correction in the frequency domain. The reason for this is that the imbalance is frequency dependent and there is a quadratic curve used in a digital filter to correct it. Thus I have to do demodulation after the IQ correction. Tom --------------------------------------- Posted through http://www.DSPRelated.com
Reply by ●April 19, 20152015-04-19
>>"tomb18" <103312@DSPRelated> writes: >> >>>>On Fri, 17 Apr 2015 14:52:32 -0500, "tomb18" <103312@DSPRelated> >>>>wrote: >>>> >>>>>>I'm not going to fix your top-posting, so this is just going to be >>a >>>>>>mess from here if the thread continues. >>>>>> >>>>>>You're doing things the hard way, IMHO. >>>>>> >>>>>>You don't need to take an FFT and then an iFFT. Take the FFT, >>use >>>>>>that for your spectrum analysis or whatever. Take the same >>signal >>>>>>that went into the FFT and mix whatever part of that you want to >be >>>>at >>>>>>zero-Hz with a complex mix operation (i.e., multiply >>sample-by-sample >>>>>>with a complex tone at the appropriate frequency). >>>>>> >>>>>>You can then apply a low-pass filter to isolate the signal that >you >>>>>>want before detecting the magnitude. >>>>>> >>>>>>That'll save a ton of computation and avoid all the hassles with >>>>>>overlap-add, etc. A LPF is much easier to design and implement >>than >>>>>>a BPF. >>>>>> >>>>>> >>>>>> >>>>>>On Fri, 17 Apr 2015 10:41:17 -0500, "tomb18" <103312@DSPRelated> >>>>>>wrote: >>>>>> >>>>>>>Many thanks to all that replied! >>>>>>>Starting here I will explain this a little more. >>>>>>>1)Yes the FFT input buffer is filled with complex samples. >>>>>>>2)I take the complex FFT and output it to the inverse FFT. I >also >>>>>>take >>>>>>>the complex FFT and plot it in a spectrum (this is independent of >>>>the >>>>>>>demodulation). >>>>>>>3) I take the complex values in the output buffer and convert to >>>>>>polar. >>>>>>>4) I only take the magnitude. >>>>>>> >>>>>>>The first point is better explained by my overall goal which is >to >>>>>>provide >>>>>>>a spectrum scope for a software defined radio. This is >functional >>>>and >>>>>>>works very well. I now wish to isolate signals (SSB or AM) from >>the >>>>>>>spectrum, which can be anywhere over the whole range of 192kHz (I >>>>>>sample >>>>>>>at 192kHz and plot positive and negative frequencies. The radio >>>>>>provides >>>>>>>a quadrature output). >>>>>>>So my test application is just demodulating signals around 0-3kHz >>>>i.e. >>>>>>in >>>>>>>the center of the spectrum. By tuning the radio manually, I can >>get >>>>>>the >>>>>>>signals down to this frequency range and then demodulate. >>>>>>>So the overall flow is: >>>>>>>Stereo audio (I and Q fed into left and right channels), Sampling >>at >>>>>>>192kHz, >>>>>>>FFT with blackman, plotting of spectrum (with IQ balance >control). >>>>>>>Now I can take the FFT with rectangular windowing, inverse FFT, >>then >>>>>>for >>>>>>>AM I convert to polar and take the magnitude. Then it's off to >>the >>>>>>audio >>>>>>>out. >>>>>>>For USB I take the output of the inverse fourier, take the >>imaginary >>>>>>>values, send them through a hilbert transform and subtract the >>>>shifted >>>>>>>values from the real.Then it's out to the audio. For LSB its the >>>>same >>>>>>>except it's added. >>>>>>>All of this works well. >>>>>>>So my original question is why did the audio become distorted >when >>I >>>>>>used >>>>>>>a blackman window. That has been explained by Randy. >>>>>>> >>>>>>>My ultimate goal however (and I am a beginner here!) is to >isolate >>a >>>>>>>signal in the frequency domain and bring it down to 0 hz. >>>>>>>Next I need a bandpass filter. I then need to pass this filter >>>>through >>>>>>an >>>>>>>inverse FFT, apply windowing to this filter. Next do an FFT on >>this >>>>>>>filter and multiply it with the original signals FFT. After this >>I >>>>do >>>>>>an >>>>>>>iFFT with the overlap add procedure. Whew! This is why I >started >>>>>>with >>>>>>>the simple case above. >>>>>>>So, my next questions are about the bandpass filter. How does >one >>>>>>create >>>>>>>one, and also how does one apply an iFFT to it? >>>>>>>Assuming that there is an equation for the bandpass filter, does >>one >>>>>>just >>>>>>>create a buffer containing the results of the filter from -0.5 to >>>>+0.5 >>>>>>and >>>>>>>using the value of "1" as the input? Do you then send this >buffer >>>>to >>>>>>the >>>>>>>inverse FFT? I think this is the one area where I have not >really >>>>seen >>>>>>an >>>>>>>explanation of this. >>>>>>>I really appreciate the help here as things are not always clear. >>>>>>>Regards, Tom >>>>>> >>>>>> >>>>>>>>Your question is missing many details. I assume you are doing >the >>>>>>>>following >>>>>>>> >>>>>>>>1) fill the fft input buffer with complex samples >>>>>>>>2) fft, then inverse fft >>>>>>>>3) take the complex values in the output buffer and convert to >>>>>>>>polar. >>>>>>>> >>>>>>>>While you didn't say this, I assume the output you use is just >>the >>>>>>>>magnitude portion of the polar output (the angle is thrown away >>so >>>>>>don't >>>>>>>bother >>>>>>>>computing it) >>>>>>>> >>>>>>>>So, a few points. >>>>>>>>The fft/inverse fft just cancel each other so the output buffer >>>>>>equals >>>>>>>>the input buffer. Were you planning to do some filtering in the >>>>>>>frequency >>>>>>>>domain ? If not then the whole scheme is just wasting the >>>>fft/inverse >>>>>>fft >>>>>>>and >>>>>>>>you could just compute the magnitude of the original complex >>signal >>>>on >>>>>>a >>>>>>>>sample-by-sample basis. >>>>>>>> >>>>>>>>If you ARE planning on doing some filtering in the frequency >>domain >>>>>>then >>>>>>>>you need to use the conventional overlap-add method of >>>>reconstructing >>>>>>>the >>>>>>>>time-domain signal, followed by taking the magnitude of the >>>>resulting >>>>>>>>samples. >>>>>>>> >>>>>>>>Note that none of the above can be described as AM demodulation >>in >>>>>>the >>>>>>>>frequency domain, since after the ifft you are back in the time >>>>>>domain. >>>>>>>I've >>>>>>>>never heard of AM demodulation in the frequency domain, perhaps >>it >>>>>>>exists, >>>>>>>>I'm not a comms expert. >>>>>>>> >>>>>>>>Bob >>>>>>> >>>>>>> >>>>>>>--------------------------------------- >>>>>>>Posted through http://www.DSPRelated.com >>>>>> >>>>>>Eric Jacobsen >>>>>>Anchor Hill Communications >>>>>>http://www.anchorhill.com >>>>> >>>>>Hi, >>>>>Sorry about the top posting....I subscribe to mailing lists often >>and >>>>the >>>>>ones I subscribe to suggest top posting. >>>>>Well I'll be....you are right. I just tried this taking the IQ >>signal >>>>off >>>>>before it goes into the FFT and as expected, all the demodulation I >>>>did >>>>>works exactly the same. No need for an inverse FFT. Not only >that, >>>>>getting rid of the iFFT means there is now no noticeable delay. >>Much >>>>>better. >>>>>At the moment, I am just tuning the radio to get the signal to >where >>I >>>>can >>>>>hear it. Next, I need to do the multiplication with the complex >>tone >>>>but >>>>>I have a question that will really show how little I know about all >>of >>>>>this, lol. >>>>>The IQ signals come from the sound card. The sampling is done at >>>>192kHz >>>>>and 4096 byte blocks are delivered at a time. I presume I just >need >>>>to >>>>>deal with one block at a time? Can I just multiply each signal ( I >>and >>>>Q) >>>>>with a sine and cosine? Do I need to match the magnitudes in any >>way? >>>> >>>>>Like I said, here i really don't know what I'm doing. >>>>>Thanks, Tom >>>>>--------------------------------------- >>>>>Posted through http://www.DSPRelated.com >>>> >>>>Assuming the time between the blocks is the same as the time between >>>>the samples, it will be better to keep the phase of the mixer >>>>continuous across the edge of the blocks. In other words, don't >>>>reset the phase at the beginning of each block, or you may hear a >pop >>>>or other artifact at the block edge. >>>> >>>>This may be less likely since you're ultimately demodulating AM, but >>>>if you do apply some filtering before detecting the AM you can wind >>up >>>>with artifacts at the block edges if you're not careful. >>>> >>>> >>>>Eric Jacobsen >>>>Anchor Hill Communications >>>>http://www.anchorhill.com >>> >>> Well I have a prototype. I have an oscillator, and I am multiplying >>each >>> sample with the oscillator step. It works. It allows me to select a >>> frequency for the oscillator and thus tune in to a signal in the >>spectrum. >>> The low pass filters(one on the real and one on the imaginary) , >>work. >>> However, it seems that I hear many tones as I tune through the >>spectrum >>> in AM mode. When I get on a signal I have to tune the radio a bit >>until >>> the tone goes lower until it disappears but there is a point where >it >>is >>> inaudible but I can hear a beat in the signal. It seems it is acting >>as a >>> synchronous detector. >> >>If the AM modulation has a carrier component, then this would produce >>the tones. The difference between the actual carrier and your "digital >>oscillator" frequency will be a tone present in the demodulated >output. >> >>> In USB, I do not hear a tone. But if I just listen to a properly >>tuned >>> signal, it appears to slowly waver in pitch. >>> I'm not quite sure what is happening here. Any ideas? >> >>Your carrier and digital oscillator are slowly varying relative to one >>another. >>-- >>Randy Yates >>Digital Signal Labs >>http://www.digitalsignallabs.com > >That's what I thought. However, on further investigation I believe it's >images from the IQ imbalance. If I have a fixed oscillator frequency, as >I >tune a signal, I see an image in the spectrum. As I tune, the image >gets >closer to the carrier, the pitch lowers, and when it completely overlaps >the carrier the pitch is gone. >Well, this brings me right back to working in the frequency domain. My >spectrum scope requires IQ imbalance correction in the frequency domain. > >The reason for this is that the imbalance is frequency dependent and >there >is a quadratic curve used in a digital filter to correct it. Thus I >have >to do demodulation after the IQ correction. >Tom > >--------------------------------------- >Posted through http://www.DSPRelated.comHi, No, its not the images! I recorded the IQ signal with audacity and fed it into a third party prodcut that will do demodulation of an IQ signal and it works perfect. So, something in my code. Good to know. --------------------------------------- Posted through http://www.DSPRelated.com
Reply by ●April 19, 20152015-04-19
>>>"tomb18" <103312@DSPRelated> writes: >>> >>>>>On Fri, 17 Apr 2015 14:52:32 -0500, "tomb18" <103312@DSPRelated> >>>>>wrote: >>>>> >>>>>>>I'm not going to fix your top-posting, so this is just going to >be >>>a >>>>>>>mess from here if the thread continues. >>>>>>> >>>>>>>You're doing things the hard way, IMHO. >>>>>>> >>>>>>>You don't need to take an FFT and then an iFFT. Take the FFT, >>>use >>>>>>>that for your spectrum analysis or whatever. Take the same >>>signal >>>>>>>that went into the FFT and mix whatever part of that you want to >>be >>>>>at >>>>>>>zero-Hz with a complex mix operation (i.e., multiply >>>sample-by-sample >>>>>>>with a complex tone at the appropriate frequency). >>>>>>> >>>>>>>You can then apply a low-pass filter to isolate the signal that >>you >>>>>>>want before detecting the magnitude. >>>>>>> >>>>>>>That'll save a ton of computation and avoid all the hassles with >>>>>>>overlap-add, etc. A LPF is much easier to design and implement >>>than >>>>>>>a BPF. >>>>>>> >>>>>>> >>>>>>> >>>>>>>On Fri, 17 Apr 2015 10:41:17 -0500, "tomb18" <103312@DSPRelated> >>>>>>>wrote: >>>>>>> >>>>>>>>Many thanks to all that replied! >>>>>>>>Starting here I will explain this a little more. >>>>>>>>1)Yes the FFT input buffer is filled with complex samples. >>>>>>>>2)I take the complex FFT and output it to the inverse FFT. I >>also >>>>>>>take >>>>>>>>the complex FFT and plot it in a spectrum (this is independent >of >>>>>the >>>>>>>>demodulation). >>>>>>>>3) I take the complex values in the output buffer and convert to >>>>>>>polar. >>>>>>>>4) I only take the magnitude. >>>>>>>> >>>>>>>>The first point is better explained by my overall goal which is >>to >>>>>>>provide >>>>>>>>a spectrum scope for a software defined radio. This is >>functional >>>>>and >>>>>>>>works very well. I now wish to isolate signals (SSB or AM) from >>>the >>>>>>>>spectrum, which can be anywhere over the whole range of 192kHz >(I >>>>>>>sample >>>>>>>>at 192kHz and plot positive and negative frequencies. The radio >>>>>>>provides >>>>>>>>a quadrature output). >>>>>>>>So my test application is just demodulating signals around >0-3kHz >>>>>i.e. >>>>>>>in >>>>>>>>the center of the spectrum. By tuning the radio manually, I can >>>get >>>>>>>the >>>>>>>>signals down to this frequency range and then demodulate. >>>>>>>>So the overall flow is: >>>>>>>>Stereo audio (I and Q fed into left and right channels), >Sampling >>>at >>>>>>>>192kHz, >>>>>>>>FFT with blackman, plotting of spectrum (with IQ balance >>control). >>>>>>>>Now I can take the FFT with rectangular windowing, inverse FFT, >>>then >>>>>>>for >>>>>>>>AM I convert to polar and take the magnitude. Then it's off to >>>the >>>>>>>audio >>>>>>>>out. >>>>>>>>For USB I take the output of the inverse fourier, take the >>>imaginary >>>>>>>>values, send them through a hilbert transform and subtract the >>>>>shifted >>>>>>>>values from the real.Then it's out to the audio. For LSB its the >>>>>same >>>>>>>>except it's added. >>>>>>>>All of this works well. >>>>>>>>So my original question is why did the audio become distorted >>when >>>I >>>>>>>used >>>>>>>>a blackman window. That has been explained by Randy. >>>>>>>> >>>>>>>>My ultimate goal however (and I am a beginner here!) is to >>isolate >>>a >>>>>>>>signal in the frequency domain and bring it down to 0 hz. >>>>>>>>Next I need a bandpass filter. I then need to pass this filter >>>>>through >>>>>>>an >>>>>>>>inverse FFT, apply windowing to this filter. Next do an FFT on >>>this >>>>>>>>filter and multiply it with the original signals FFT. After >this >>>I >>>>>do >>>>>>>an >>>>>>>>iFFT with the overlap add procedure. Whew! This is why I >>started >>>>>>>with >>>>>>>>the simple case above. >>>>>>>>So, my next questions are about the bandpass filter. How does >>one >>>>>>>create >>>>>>>>one, and also how does one apply an iFFT to it? >>>>>>>>Assuming that there is an equation for the bandpass filter, does >>>one >>>>>>>just >>>>>>>>create a buffer containing the results of the filter from -0.5 >to >>>>>+0.5 >>>>>>>and >>>>>>>>using the value of "1" as the input? Do you then send this >>buffer >>>>>to >>>>>>>the >>>>>>>>inverse FFT? I think this is the one area where I have not >>really >>>>>seen >>>>>>>an >>>>>>>>explanation of this. >>>>>>>>I really appreciate the help here as things are not always >clear. >>>>>>>>Regards, Tom >>>>>>> >>>>>>> >>>>>>>>>Your question is missing many details. I assume you are doing >>the >>>>>>>>>following >>>>>>>>> >>>>>>>>>1) fill the fft input buffer with complex samples >>>>>>>>>2) fft, then inverse fft >>>>>>>>>3) take the complex values in the output buffer and convert to >>>>>>>>>polar. >>>>>>>>> >>>>>>>>>While you didn't say this, I assume the output you use is just >>>the >>>>>>>>>magnitude portion of the polar output (the angle is thrown away >>>so >>>>>>>don't >>>>>>>>bother >>>>>>>>>computing it) >>>>>>>>> >>>>>>>>>So, a few points. >>>>>>>>>The fft/inverse fft just cancel each other so the output buffer >>>>>>>equals >>>>>>>>>the input buffer. Were you planning to do some filtering in the >>>>>>>>frequency >>>>>>>>>domain ? If not then the whole scheme is just wasting the >>>>>fft/inverse >>>>>>>fft >>>>>>>>and >>>>>>>>>you could just compute the magnitude of the original complex >>>signal >>>>>on >>>>>>>a >>>>>>>>>sample-by-sample basis. >>>>>>>>> >>>>>>>>>If you ARE planning on doing some filtering in the frequency >>>domain >>>>>>>then >>>>>>>>>you need to use the conventional overlap-add method of >>>>>reconstructing >>>>>>>>the >>>>>>>>>time-domain signal, followed by taking the magnitude of the >>>>>resulting >>>>>>>>>samples. >>>>>>>>> >>>>>>>>>Note that none of the above can be described as AM demodulation >>>in >>>>>>>the >>>>>>>>>frequency domain, since after the ifft you are back in the time >>>>>>>domain. >>>>>>>>I've >>>>>>>>>never heard of AM demodulation in the frequency domain, perhaps >>>it >>>>>>>>exists, >>>>>>>>>I'm not a comms expert. >>>>>>>>> >>>>>>>>>Bob >>>>>>>> >>>>>>>> >>>>>>>>--------------------------------------- >>>>>>>>Posted through http://www.DSPRelated.com >>>>>>> >>>>>>>Eric Jacobsen >>>>>>>Anchor Hill Communications >>>>>>>http://www.anchorhill.com >>>>>> >>>>>>Hi, >>>>>>Sorry about the top posting....I subscribe to mailing lists often >>>and >>>>>the >>>>>>ones I subscribe to suggest top posting. >>>>>>Well I'll be....you are right. I just tried this taking the IQ >>>signal >>>>>off >>>>>>before it goes into the FFT and as expected, all the demodulation >I >>>>>did >>>>>>works exactly the same. No need for an inverse FFT. Not only >>that, >>>>>>getting rid of the iFFT means there is now no noticeable delay. >>>Much >>>>>>better. >>>>>>At the moment, I am just tuning the radio to get the signal to >>where >>>I >>>>>can >>>>>>hear it. Next, I need to do the multiplication with the complex >>>tone >>>>>but >>>>>>I have a question that will really show how little I know about >all >>>of >>>>>>this, lol. >>>>>>The IQ signals come from the sound card. The sampling is done at >>>>>192kHz >>>>>>and 4096 byte blocks are delivered at a time. I presume I just >>need >>>>>to >>>>>>deal with one block at a time? Can I just multiply each signal ( >I >>>and >>>>>Q) >>>>>>with a sine and cosine? Do I need to match the magnitudes in any >>>way? >>>>> >>>>>>Like I said, here i really don't know what I'm doing. >>>>>>Thanks, Tom >>>>>>--------------------------------------- >>>>>>Posted through http://www.DSPRelated.com >>>>> >>>>>Assuming the time between the blocks is the same as the time >between >>>>>the samples, it will be better to keep the phase of the mixer >>>>>continuous across the edge of the blocks. In other words, don't >>>>>reset the phase at the beginning of each block, or you may hear a >>pop >>>>>or other artifact at the block edge. >>>>> >>>>>This may be less likely since you're ultimately demodulating AM, >but >>>>>if you do apply some filtering before detecting the AM you can wind >>>up >>>>>with artifacts at the block edges if you're not careful. >>>>> >>>>> >>>>>Eric Jacobsen >>>>>Anchor Hill Communications >>>>>http://www.anchorhill.com >>>> >>>> Well I have a prototype. I have an oscillator, and I am >multiplying >>>each >>>> sample with the oscillator step. It works. It allows me to select a >>>> frequency for the oscillator and thus tune in to a signal in the >>>spectrum. >>>> The low pass filters(one on the real and one on the imaginary) , >>>work. >>>> However, it seems that I hear many tones as I tune through the >>>spectrum >>>> in AM mode. When I get on a signal I have to tune the radio a bit >>>until >>>> the tone goes lower until it disappears but there is a point where >>it >>>is >>>> inaudible but I can hear a beat in the signal. It seems it is >acting >>>as a >>>> synchronous detector. >>> >>>If the AM modulation has a carrier component, then this would produce >>>the tones. The difference between the actual carrier and your >"digital >>>oscillator" frequency will be a tone present in the demodulated >>output. >>> >>>> In USB, I do not hear a tone. But if I just listen to a properly >>>tuned >>>> signal, it appears to slowly waver in pitch. >>>> I'm not quite sure what is happening here. Any ideas? >>> >>>Your carrier and digital oscillator are slowly varying relative to >one >>>another. >>>-- >>>Randy Yates >>>Digital Signal Labs >>>http://www.digitalsignallabs.com >> >>That's what I thought. However, on further investigation I believe >it's >>images from the IQ imbalance. If I have a fixed oscillator frequency, >as >>I >>tune a signal, I see an image in the spectrum. As I tune, the image >>gets >>closer to the carrier, the pitch lowers, and when it completely >overlaps >>the carrier the pitch is gone. >>Well, this brings me right back to working in the frequency domain. >My >>spectrum scope requires IQ imbalance correction in the frequency >domain. >> >>The reason for this is that the imbalance is frequency dependent and >>there >>is a quadratic curve used in a digital filter to correct it. Thus I >>have >>to do demodulation after the IQ correction. >>Tom >> >>--------------------------------------- >>Posted through http://www.DSPRelated.com > >Hi, >No, its not the images! I recorded the IQ signal with audacity and fed >it >into a third party prodcut that will do demodulation of an IQ signal and >it works perfect. >So, something in my code. >Good to know. >--------------------------------------- >Posted through http://www.DSPRelated.comProblem solved....I wasn't doing a complex multiply for one. Second there was an issue with the filters. Now it's working beautifully. --------------------------------------- Posted through http://www.DSPRelated.com
Reply by ●April 19, 20152015-04-19
"tomb18" <103312@DSPRelated> writes:> [...] > > Problem solved....I wasn't doing a complex multiply for one. Second there > was an issue with the filters. Now it's working beautifully. > --------------------------------------- > Posted through http://www.DSPRelated.comGood to hear. -- Randy Yates Digital Signal Labs http://www.digitalsignallabs.com
Reply by ●May 8, 20152015-05-08
On Friday, April 17, 2015 at 10:41:21 AM UTC-5, tomb18 wrote:> Many thanks to all that replied! > Starting here I will explain this a little more. > 1)Yes the FFT input buffer is filled with complex samples. > 2)I take the complex FFT and output it to the inverse FFT. I also take > the complex FFT and plot it in a spectrum (this is independent of the > demodulation). > 3) I take the complex values in the output buffer and convert to polar. > 4) I only take the magnitude.The simplest way to check everything is to keep the phase in. When plotting, color-code the phase and plot intensity as brightness. When cranking out sound, put the (real + imaginary)/2 and (real - imaginary)/2 on left and right stereo channels. BTW, once you see what the phase is doing you may well abandon your entire endeavor, thinking you've stumbled onto a major new discovery. At that point look up "relocation" for DSP.
Reply by ●May 9, 20152015-05-09
On Fri, 8 May 2015 15:27:29 -0700 (PDT), federation2005@netzero.com wrote:>On Friday, April 17, 2015 at 10:41:21 AM UTC-5, tomb18 wrote: >> Many thanks to all that replied! >> Starting here I will explain this a little more. >> 1)Yes the FFT input buffer is filled with complex samples. >> 2)I take the complex FFT and output it to the inverse FFT. I also take >> the complex FFT and plot it in a spectrum (this is independent of the >> demodulation). >> 3) I take the complex values in the output buffer and convert to polar. >> 4) I only take the magnitude. > >The simplest way to check everything is to keep the phase in. When plotting, color-code the phase and plot intensity as brightness. When cranking out sound, put the (real + imaginary)/2 and (real - imaginary)/2 on left and right stereo channels. > >BTW, once you see what the phase is doing you may well abandon your entire endeavor, thinking you've stumbled onto a major new discovery. At that point look up "relocation" for DSP.Could you provide a specific link? "DSP relocation" turns out to be a business name and a service, apparently for DSP professionals to relocate to other companies or countries. That's all the hits Google returns (11 pages). Best regards, Bob Masta DAQARTA v7.60 Data AcQuisition And Real-Time Analysis www.daqarta.com Scope, Spectrum, Spectrogram, Sound Level Meter Frequency Counter, Pitch Track, Pitch-to-MIDI FREE Signal Generator, DaqMusiq generator Science with your sound card!
