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Newbie question - I/Q files

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  • Dan
    Hello all, I m new to the group, and have hardly scratched the surface of SL yet. Seems to be a powerful tool, but also somewhat confusing to me to begin with.
    Message 1 of 8 , Jun 18, 2012
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      Hello all,

      I'm new to the group, and have hardly scratched the surface of SL yet. Seems to be a powerful tool, but also somewhat confusing to me to begin with. I have a specific problem which I hope to solve with SL:
      I am using a well-known SDR receiver for recording ultrasonic sounds - bat calls - at a bandwidth of 100 kHz typically (ranging from 0...100 kHz) by connecting a micro to the antenna input. This produces stereo I/Q files written as .wav at a samping rate of, in this case, 125 kHz.
      Is there a way to transform these files to conventional, mono wav audio at full bandwidth for analyzing? The receiver audio output is limited to merely 16 kHz.
      Any help is very welcome. Here's a sample file for downloading if someone likes to get his hands on:

      https://dl.dropbox.com/u/35425729/20120609_100khz.wav

      Thanks for your attention and best regards,
      Dan, Switzerland
    • Graham Mcleod
      Hi Dan, I shall let others reply with regard to SL; however I think you have Another problem. Sampling at 125 KHz will fold all spectrum over half the sample
      Message 2 of 8 , Jun 19, 2012
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        Hi Dan,
        I shall let others reply with regard to SL; however I think you have Another problem.
        Sampling at 125 KHz will fold all spectrum over half the sample frequency (62.5 KHz)
        with a process akin to Mixing. Hence a signal at 80 KHz will appear as if it were at 45 KHz...
        In order to process 0 to 100 KHz without foldover, you need a sampling rate of at least
        200 KHz in order not to violate Nyquists' Theorem. Ideally, 250 KHz or so, will allow for
        a hardware Lowpass Filter, which passes all signals below 100 KHz, but rejects all signals over
        125 KHz, in order to avoid "Aliassing", the effect noted above, wherein signals at one
        frequency appear to have a different frequency, a deficiency of the Sampling process itself.
        Graham G8PHA.
         
      • Dan
        Hi Graham, the maker of this SDR says that at 100 kHz recording BW, the sampling rate is 125 kHz in stereo (I/Q), which results in a 250 kHz rate
        Message 3 of 8 , Jun 19, 2012
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          Hi Graham,

          the maker of this SDR says that at 100 kHz recording BW, the sampling rate is 125 kHz in stereo (I/Q), which results in a 250 kHz rate information-wise. At least that's what it says in the technical handbook... the problem seems to be to extract the I/Q information from the two channels.

          Regards, Dan



          --- In SpectrumLabUsers@yahoogroups.com, Graham Mcleod <g8pha@...> wrote:
          >
          > Hi Dan,
          > I shall let others reply with regard to SL; however I think you have Another problem.
          > Sampling at 125 KHz will fold all spectrum over half the sample frequency (62.5 KHz)
          > with a process akin to Mixing. Hence a signal at 80 KHz will appear as if it were at 45 KHz...
          > In order to process 0 to 100 KHz without foldover, you need a sampling rate of at least
          > 200 KHz in order not to violate Nyquists' Theorem. Ideally, 250 KHz or so, will allow for
          > a hardware Lowpass Filter, which passes all signals below 100 KHz, but rejects all signals over
          > 125 KHz, in order to avoid "Aliassing", the effect noted above, wherein signals at one
          > frequency appear to have a different frequency, a deficiency of the Sampling process itself.
          > Graham G8PHA.
          >
        • Alan
          ... Sent: Tuesday, June 19, 2012 3:50 PM Subject: [SpectrumLabUsers] Re: Newbie question - I/Q files ... Dan, What is this wonderful SDR? What you say does not
          Message 4 of 8 , Jun 19, 2012
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            ----- Original Message -----
            Sent: Tuesday, June 19, 2012 3:50 PM
            Subject: [SpectrumLabUsers] Re: Newbie question - I/Q files


            >the maker of this SDR says that at 100 kHz recording BW, the sampling rate is 125 kHz in stereo (I/Q), which results in a 250 kHz
            >rate information-wise. At least that's what it says in the technical handbook...

            Dan,

            What is this wonderful SDR?
            What you say does not make sense unless it is described in a non-standard way.
            Always the maximum frequency must be less than half the sample rate.

            73 Alan G4ZXFQ



            > Sampling at 125 KHz will fold all spectrum over half the sample frequency (62.5 KHz)
            > with a process akin to Mixing. Hence a signal at 80 KHz will appear as if it were at 45 KHz...
            > In order to process 0 to 100 KHz without foldover, you need a sampling rate of at least
            > 200 KHz in order not to violate Nyquists' Theorem. Ideally, 250 KHz or so, will allow for
            > a hardware Lowpass Filter, which passes all signals below 100 KHz, but rejects all signals over
            > 125 KHz, in order to avoid "Aliassing", the effect noted above, wherein signals at one
            > frequency appear to have a different frequency, a deficiency of the Sampling process itself.
          • Dan
            Alan, this wonderful SDR is the Excalibur. ... All DDC files are stereo (as they are stored in the I/Q form) and 32 bit. The recording sampling rate varies
            Message 5 of 8 , Jun 20, 2012
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              Alan,
              this wonderful SDR is the Excalibur.
              Here's what the helpfile Appendix G says:

              >>... The DDC recorder records in a standard "wav" file format, but saves the files with an extension ".ddc" in order to avoid confusion with audio files.
              All DDC files are stereo (as they are stored in the I/Q form) and 32 bit.
              The recording sampling rate varies according to the selected DDC bandwidth, shown in the following table:

              DDC Bandwidth(kHz / DDC Recording Sampling Rate (kHz)
              20 25
              24 32
              32 40
              40 50
              50 62.5
              64 80
              80 100
              100 125
              125 160
              160 200
              200 250
              250 312.5
              320 400
              400 500
              500 625
              640 800
              800 1000
              1000 1250
              1250 1666.67
              1500 2000
              2000 2500

              Note: The Nyquist theorem is satisfied because there are two values (I and Q) for each sample (stored as two stereo channels), so the effective sampling rate is twice of that shown.
              The Audio recorder records standard ".wav" files with16 bit word length, sampling at 32 kHz, in mono. ...<<


              Has anyone examined my file yet?

              Regards, Dan




              > Dan,
              >
              > What is this wonderful SDR?
              > What you say does not make sense unless it is described in a non-standard way.
              > Always the maximum frequency must be less than half the sample rate.
              >
              > 73 Alan G4ZXFQ
              >
              >
              >
              > > Sampling at 125 KHz will fold all spectrum over half the sample frequency (62.5 KHz)
              > > with a process akin to Mixing. Hence a signal at 80 KHz will appear as if it were at 45 KHz...
              > > In order to process 0 to 100 KHz without foldover, you need a sampling rate of at least
              > > 200 KHz in order not to violate Nyquists' Theorem. Ideally, 250 KHz or so, will allow for
              > > a hardware Lowpass Filter, which passes all signals below 100 KHz, but rejects all signals over
              > > 125 KHz, in order to avoid "Aliassing", the effect noted above, wherein signals at one
              > > frequency appear to have a different frequency, a deficiency of the Sampling process itself.
              >
            • Alan
              ... Subject: [SpectrumLabUsers] Re: Newbie question - I/Q files ... Dan, Ah! I had not thought of it as a stereo recording, but that should have been obvious
              Message 6 of 8 , Jun 21, 2012
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                ----- Original Message -----
                Subject: [SpectrumLabUsers] Re: Newbie question - I/Q files



                > Here's what the helpfile Appendix G says:

                Dan,

                Ah! I had not thought of it as a stereo recording, but that should have been obvious to me...

                But would the Excalibur convert to audio using SSB? Or is the bandwidth greater than the widest setting?
                I really have no idea on your original query. I copy below.

                I think there have been posts here about bat recordings made directly. Maybe that is the better way. But SpecLab can do all sorts of
                things, possibly feed your recording in as an IQ signal.
                Do a search of the message archives. Start another thread with "bat recordings" in the subject.

                Alan


                >>I am using a well-known SDR receiver for recording ultrasonic sounds - bat calls - at a bandwidth of 100 kHz typically (ranging
                >>from 0...100 kHz) by connecting a micro to the antenna input. This produces stereo I/Q files written as .wav at a samping rate of,
                >>in this case, 125 kHz.
                Is there a way to transform these files to conventional, mono wav audio at full bandwidth for analyzing? The receiver audio output
                is limited to merely 16 kHz.
                Any help is very welcome. Here's a sample file for downloading if someone likes to get his hands on:

                https://dl.dropbox.com/u/35425729/20120609_100khz.wav


                >
                >>>... The DDC recorder records in a standard "wav" file format, but saves the files with an extension ".ddc" in order to avoid
                >>>confusion with audio files.
                > All DDC files are stereo (as they are stored in the I/Q form) and 32 bit.
                > The recording sampling rate varies according to the selected DDC bandwidth, shown in the following table:
                >
                > DDC Bandwidth(kHz / DDC Recording Sampling Rate (kHz)
                > 20 25
                > 24 32
              • ehydra
                ... I don t understand what you want with the file but if you can describe the operation mathematical I can use LTspice to read the wavefile, transform it and
                Message 7 of 8 , Jun 24, 2012
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                  Dan schrieb:
                  > Has anyone examined my file yet?

                  I don't understand what you want with the file but if you can describe
                  the operation mathematical I can use LTspice to read the wavefile,
                  transform it and write it out as wavefile... (I do this often).

                  Maybe that helps.

                  I'm not an expert with IQ-receivers, I always prefer direct sampling the
                  input.
                  Is the IQ-process a mixer function, multiplying with phase coupled 0°
                  and 90° reference oscillators. SL has a complex FFT function and you can
                  read the file in...
                  So the base process is to multiply both input channels with the complex
                  oscillator down to DC and then pythagoras both channels into one
                  magnitude channel?

                  - Henry


                  --
                  ehydra.dyndns.info
                • Dan
                  Hello Henry, sorry for my late reply, and thanks for your offer. Things have developed in a different direction here, as I have found a better way of recording
                  Message 8 of 8 , Jul 8, 2012
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                    Hello Henry,
                    sorry for my late reply, and thanks for your offer.
                    Things have developed in a different direction here, as I have found a better way of recording ultrasonic signals (this being the purpose of my question), so I am not pursuing the problem described in my posting.
                    Best regards,
                    Dan


                    --- In SpectrumLabUsers@yahoogroups.com, ehydra <ehydra@...> wrote:
                    >
                    > Dan schrieb:
                    > > Has anyone examined my file yet?
                    >
                    > I don't understand what you want with the file but if you can describe
                    > the operation mathematical I can use LTspice to read the wavefile,
                    > transform it and write it out as wavefile... (I do this often).
                    >
                    > Maybe that helps.
                    >
                    > I'm not an expert with IQ-receivers, I always prefer direct sampling the
                    > input.
                    > Is the IQ-process a mixer function, multiplying with phase coupled 0°
                    > and 90° reference oscillators. SL has a complex FFT function and you can
                    > read the file in...
                    > So the base process is to multiply both input channels with the complex
                    > oscillator down to DC and then pythagoras both channels into one
                    > magnitude channel?
                    >
                    > - Henry
                    >
                    >
                    > --
                    > ehydra.dyndns.info
                    >
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