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How to add a low pass filter to this non-inverting amplifier circuit?


Buffer between high pass and low pass filter when making a bandpass filter?EEG amplifier circuit low and high pass filters. Oscillations, noise, etcDigital Low-Pass Filter When Oversampling AudioWill this passive non inverting anti-log circuit work?Low pass filter for differential signalsLow pass sallen key filter using single supply for audio transmissionCompensating for power loss through resistor in high/low pass filter circuit?Photodiode non-inverting amplifier circuitMATLAB How do I pass a signal into a low-pass filter in matlab?Low-Pass Filter Selection and Placement in Active Noise Cancellation






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1












$begingroup$


I've come up with this simple circuit as an amplifier before an 8bit ADC which receives audio signals. R2 is actually a variable resistor (pot.) of 100k. The circuit also biases the signal to VCC/2 and uses large resistors to set the input impedance is pretty high all over the working audio frequency range. This works well for my application, however I would like to add a single low pass filter stage to this amplifier so I can remove some high freq. content that I would not be able to sample. How to approach this without having to use another op-amp and without changing the characteristics of the circuit in the spectrum I would like to pass?



enter image description here










share|improve this question











$endgroup$











  • $begingroup$
    what is the amplitude of the high frequency content? 8 LSBs?
    $endgroup$
    – analogsystemsrf
    1 hour ago

















1












$begingroup$


I've come up with this simple circuit as an amplifier before an 8bit ADC which receives audio signals. R2 is actually a variable resistor (pot.) of 100k. The circuit also biases the signal to VCC/2 and uses large resistors to set the input impedance is pretty high all over the working audio frequency range. This works well for my application, however I would like to add a single low pass filter stage to this amplifier so I can remove some high freq. content that I would not be able to sample. How to approach this without having to use another op-amp and without changing the characteristics of the circuit in the spectrum I would like to pass?



enter image description here










share|improve this question











$endgroup$











  • $begingroup$
    what is the amplitude of the high frequency content? 8 LSBs?
    $endgroup$
    – analogsystemsrf
    1 hour ago













1












1








1





$begingroup$


I've come up with this simple circuit as an amplifier before an 8bit ADC which receives audio signals. R2 is actually a variable resistor (pot.) of 100k. The circuit also biases the signal to VCC/2 and uses large resistors to set the input impedance is pretty high all over the working audio frequency range. This works well for my application, however I would like to add a single low pass filter stage to this amplifier so I can remove some high freq. content that I would not be able to sample. How to approach this without having to use another op-amp and without changing the characteristics of the circuit in the spectrum I would like to pass?



enter image description here










share|improve this question











$endgroup$




I've come up with this simple circuit as an amplifier before an 8bit ADC which receives audio signals. R2 is actually a variable resistor (pot.) of 100k. The circuit also biases the signal to VCC/2 and uses large resistors to set the input impedance is pretty high all over the working audio frequency range. This works well for my application, however I would like to add a single low pass filter stage to this amplifier so I can remove some high freq. content that I would not be able to sample. How to approach this without having to use another op-amp and without changing the characteristics of the circuit in the spectrum I would like to pass?



enter image description here







operational-amplifier filter low-pass non-inverting






share|improve this question















share|improve this question













share|improve this question




share|improve this question








edited 1 hour ago









K H

2,430315




2,430315










asked 2 hours ago









user733606user733606

103111




103111











  • $begingroup$
    what is the amplitude of the high frequency content? 8 LSBs?
    $endgroup$
    – analogsystemsrf
    1 hour ago
















  • $begingroup$
    what is the amplitude of the high frequency content? 8 LSBs?
    $endgroup$
    – analogsystemsrf
    1 hour ago















$begingroup$
what is the amplitude of the high frequency content? 8 LSBs?
$endgroup$
– analogsystemsrf
1 hour ago




$begingroup$
what is the amplitude of the high frequency content? 8 LSBs?
$endgroup$
– analogsystemsrf
1 hour ago










3 Answers
3






active

oldest

votes


















2












$begingroup$

The correct approach is to choose a LPF and sampling frequency such that the maximum signal at fs/2 is less than your ADC resolution. This means you need a brick wall filter at 3x your -3dB BW or 128 x faster sampling rate than your signal -3dB BW for a 20dB decade filter for an 8bit ADC... not 2x faster with a 1st order filter.






share|improve this answer









$endgroup$








  • 1




    $begingroup$
    OK, so let's assume my sampling freq. is 10KHz, so Nyquist freq. is 5KHz . My input range is 5V so LSB is ~19.53mV. So you are saying I need to select a cut freq. so that I can have about -48 dB attenuation at 5KHz? And the filter order depends on how low (or high) I select the cut freq. Is that correct?
    $endgroup$
    – user733606
    1 hour ago










  • $begingroup$
    Correct so 6dB/octave or 8th order at 2.5kHz for less than telephone quality audio which uses log ADC with 8bits > >72 dB I think
    $endgroup$
    – Sunnyskyguy EE75
    1 hour ago


















0












$begingroup$

Put about 1k in series with C1 and a cap in parallel with R3. 800 ohms and 0.01 uF will give about 20kHz. But this is only a first-order filter, not very useful, as SunnySKyGuy says.



Edit: this assumes that the driving impedance is low. If not and you know what it is, then just put an appropriate cap.






share|improve this answer











$endgroup$




















    0












    $begingroup$


    Add a low pass filter to a non-inverting amplifier circuit.




    Without giving it much thought you already have a high pass filter on the input, and the simplest way to apply a low pass is to use the output of the opamp.



    enter image description here



    This image is from this calculator which may help you pick your filter component values easily. The low pass filter is R2C2 in the image above and is buffered by the opamp reducing the interaction of gain and filter components.






    share|improve this answer











    $endgroup$













      Your Answer






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      3 Answers
      3






      active

      oldest

      votes








      3 Answers
      3






      active

      oldest

      votes









      active

      oldest

      votes






      active

      oldest

      votes









      2












      $begingroup$

      The correct approach is to choose a LPF and sampling frequency such that the maximum signal at fs/2 is less than your ADC resolution. This means you need a brick wall filter at 3x your -3dB BW or 128 x faster sampling rate than your signal -3dB BW for a 20dB decade filter for an 8bit ADC... not 2x faster with a 1st order filter.






      share|improve this answer









      $endgroup$








      • 1




        $begingroup$
        OK, so let's assume my sampling freq. is 10KHz, so Nyquist freq. is 5KHz . My input range is 5V so LSB is ~19.53mV. So you are saying I need to select a cut freq. so that I can have about -48 dB attenuation at 5KHz? And the filter order depends on how low (or high) I select the cut freq. Is that correct?
        $endgroup$
        – user733606
        1 hour ago










      • $begingroup$
        Correct so 6dB/octave or 8th order at 2.5kHz for less than telephone quality audio which uses log ADC with 8bits > >72 dB I think
        $endgroup$
        – Sunnyskyguy EE75
        1 hour ago















      2












      $begingroup$

      The correct approach is to choose a LPF and sampling frequency such that the maximum signal at fs/2 is less than your ADC resolution. This means you need a brick wall filter at 3x your -3dB BW or 128 x faster sampling rate than your signal -3dB BW for a 20dB decade filter for an 8bit ADC... not 2x faster with a 1st order filter.






      share|improve this answer









      $endgroup$








      • 1




        $begingroup$
        OK, so let's assume my sampling freq. is 10KHz, so Nyquist freq. is 5KHz . My input range is 5V so LSB is ~19.53mV. So you are saying I need to select a cut freq. so that I can have about -48 dB attenuation at 5KHz? And the filter order depends on how low (or high) I select the cut freq. Is that correct?
        $endgroup$
        – user733606
        1 hour ago










      • $begingroup$
        Correct so 6dB/octave or 8th order at 2.5kHz for less than telephone quality audio which uses log ADC with 8bits > >72 dB I think
        $endgroup$
        – Sunnyskyguy EE75
        1 hour ago













      2












      2








      2





      $begingroup$

      The correct approach is to choose a LPF and sampling frequency such that the maximum signal at fs/2 is less than your ADC resolution. This means you need a brick wall filter at 3x your -3dB BW or 128 x faster sampling rate than your signal -3dB BW for a 20dB decade filter for an 8bit ADC... not 2x faster with a 1st order filter.






      share|improve this answer









      $endgroup$



      The correct approach is to choose a LPF and sampling frequency such that the maximum signal at fs/2 is less than your ADC resolution. This means you need a brick wall filter at 3x your -3dB BW or 128 x faster sampling rate than your signal -3dB BW for a 20dB decade filter for an 8bit ADC... not 2x faster with a 1st order filter.







      share|improve this answer












      share|improve this answer



      share|improve this answer










      answered 2 hours ago









      Sunnyskyguy EE75Sunnyskyguy EE75

      73.9k228104




      73.9k228104







      • 1




        $begingroup$
        OK, so let's assume my sampling freq. is 10KHz, so Nyquist freq. is 5KHz . My input range is 5V so LSB is ~19.53mV. So you are saying I need to select a cut freq. so that I can have about -48 dB attenuation at 5KHz? And the filter order depends on how low (or high) I select the cut freq. Is that correct?
        $endgroup$
        – user733606
        1 hour ago










      • $begingroup$
        Correct so 6dB/octave or 8th order at 2.5kHz for less than telephone quality audio which uses log ADC with 8bits > >72 dB I think
        $endgroup$
        – Sunnyskyguy EE75
        1 hour ago












      • 1




        $begingroup$
        OK, so let's assume my sampling freq. is 10KHz, so Nyquist freq. is 5KHz . My input range is 5V so LSB is ~19.53mV. So you are saying I need to select a cut freq. so that I can have about -48 dB attenuation at 5KHz? And the filter order depends on how low (or high) I select the cut freq. Is that correct?
        $endgroup$
        – user733606
        1 hour ago










      • $begingroup$
        Correct so 6dB/octave or 8th order at 2.5kHz for less than telephone quality audio which uses log ADC with 8bits > >72 dB I think
        $endgroup$
        – Sunnyskyguy EE75
        1 hour ago







      1




      1




      $begingroup$
      OK, so let's assume my sampling freq. is 10KHz, so Nyquist freq. is 5KHz . My input range is 5V so LSB is ~19.53mV. So you are saying I need to select a cut freq. so that I can have about -48 dB attenuation at 5KHz? And the filter order depends on how low (or high) I select the cut freq. Is that correct?
      $endgroup$
      – user733606
      1 hour ago




      $begingroup$
      OK, so let's assume my sampling freq. is 10KHz, so Nyquist freq. is 5KHz . My input range is 5V so LSB is ~19.53mV. So you are saying I need to select a cut freq. so that I can have about -48 dB attenuation at 5KHz? And the filter order depends on how low (or high) I select the cut freq. Is that correct?
      $endgroup$
      – user733606
      1 hour ago












      $begingroup$
      Correct so 6dB/octave or 8th order at 2.5kHz for less than telephone quality audio which uses log ADC with 8bits > >72 dB I think
      $endgroup$
      – Sunnyskyguy EE75
      1 hour ago




      $begingroup$
      Correct so 6dB/octave or 8th order at 2.5kHz for less than telephone quality audio which uses log ADC with 8bits > >72 dB I think
      $endgroup$
      – Sunnyskyguy EE75
      1 hour ago













      0












      $begingroup$

      Put about 1k in series with C1 and a cap in parallel with R3. 800 ohms and 0.01 uF will give about 20kHz. But this is only a first-order filter, not very useful, as SunnySKyGuy says.



      Edit: this assumes that the driving impedance is low. If not and you know what it is, then just put an appropriate cap.






      share|improve this answer











      $endgroup$

















        0












        $begingroup$

        Put about 1k in series with C1 and a cap in parallel with R3. 800 ohms and 0.01 uF will give about 20kHz. But this is only a first-order filter, not very useful, as SunnySKyGuy says.



        Edit: this assumes that the driving impedance is low. If not and you know what it is, then just put an appropriate cap.






        share|improve this answer











        $endgroup$















          0












          0








          0





          $begingroup$

          Put about 1k in series with C1 and a cap in parallel with R3. 800 ohms and 0.01 uF will give about 20kHz. But this is only a first-order filter, not very useful, as SunnySKyGuy says.



          Edit: this assumes that the driving impedance is low. If not and you know what it is, then just put an appropriate cap.






          share|improve this answer











          $endgroup$



          Put about 1k in series with C1 and a cap in parallel with R3. 800 ohms and 0.01 uF will give about 20kHz. But this is only a first-order filter, not very useful, as SunnySKyGuy says.



          Edit: this assumes that the driving impedance is low. If not and you know what it is, then just put an appropriate cap.







          share|improve this answer














          share|improve this answer



          share|improve this answer








          edited 2 hours ago

























          answered 2 hours ago









          Mattman944Mattman944

          75117




          75117





















              0












              $begingroup$


              Add a low pass filter to a non-inverting amplifier circuit.




              Without giving it much thought you already have a high pass filter on the input, and the simplest way to apply a low pass is to use the output of the opamp.



              enter image description here



              This image is from this calculator which may help you pick your filter component values easily. The low pass filter is R2C2 in the image above and is buffered by the opamp reducing the interaction of gain and filter components.






              share|improve this answer











              $endgroup$

















                0












                $begingroup$


                Add a low pass filter to a non-inverting amplifier circuit.




                Without giving it much thought you already have a high pass filter on the input, and the simplest way to apply a low pass is to use the output of the opamp.



                enter image description here



                This image is from this calculator which may help you pick your filter component values easily. The low pass filter is R2C2 in the image above and is buffered by the opamp reducing the interaction of gain and filter components.






                share|improve this answer











                $endgroup$















                  0












                  0








                  0





                  $begingroup$


                  Add a low pass filter to a non-inverting amplifier circuit.




                  Without giving it much thought you already have a high pass filter on the input, and the simplest way to apply a low pass is to use the output of the opamp.



                  enter image description here



                  This image is from this calculator which may help you pick your filter component values easily. The low pass filter is R2C2 in the image above and is buffered by the opamp reducing the interaction of gain and filter components.






                  share|improve this answer











                  $endgroup$




                  Add a low pass filter to a non-inverting amplifier circuit.




                  Without giving it much thought you already have a high pass filter on the input, and the simplest way to apply a low pass is to use the output of the opamp.



                  enter image description here



                  This image is from this calculator which may help you pick your filter component values easily. The low pass filter is R2C2 in the image above and is buffered by the opamp reducing the interaction of gain and filter components.







                  share|improve this answer














                  share|improve this answer



                  share|improve this answer








                  edited 1 hour ago

























                  answered 2 hours ago









                  Jack CreaseyJack Creasey

                  15.9k2824




                  15.9k2824



























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