Help designing analog filter designs for an audio activated 8x8x8 LED cube

Discussion in 'The Projects Forum' started by mp4nerd, Dec 1, 2012.

  1. mp4nerd

    Thread Starter New Member

    Dec 1, 2012
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    0
    I am working on designing a music visualization system using LEDs. I would like to create an analog design that uses op amps to filter, amplify, and add DC bias to an audio signal so it can be read into the A/D of a 5V microprocessor. The input signal is a +/- 1.5V volt audio signal coming from an MP3 player. From this signal I need to produce three output signals. This means that I need to design three filters (a lowpass, a bandpass, and a highpass). These filters also need to amplify and bias the input signal so it will be a 0 to 5V signal that can be read by the processor's A/D at the maximum resolution. So basically what I have pictured is to first bias the signal, then amplify it, then filter it:


    MP3 Player Audio signal --> Bias --> Amplify --> Filter --> uP


    I am familiar with building filters with amplification, but I am not sure about the correct way to go about adding a DC bias to the filters with amplification.


    Lowpass (bass range): cutoff f @ ~500 Hz
    Bandpass (mid range): center f @ ~2250 Hz
    Highpass (high range): cutoff f @ ~4kHz


    The parts I have available for use are: LT1632 dual op-amps, resistors, 100nF capacitors
    Measuring the output signal from the iPhone: V(peak-to-peak): ~2.85V centered around 0V
    Required Gain: ~1.75
    Available Power supply units: +5V and +12V DC

    I've tried designing these filters by hand, and using TI's FilterPro software, however I have not been able to achieve the DC bias affect. I have been simulating the filters using LTSpice. I have the design files for the three filters though if it would help in describing my current problem. Any help in this area would be greatly appreciated.
     
  2. #12

    Expert

    Nov 30, 2010
    16,343
    6,828
    My crystal ball doesn't support FilterPro. You should post a schematic in .PNG format so we can see what you have so far.
     
  3. mp4nerd

    Thread Starter New Member

    Dec 1, 2012
    4
    0
    Okay, here is the schematic along with transient and AC analysis from LTSpice for the lowpass filter @ Input(f) = 300 Hz:

    [​IMG]

    [​IMG]

    [​IMG]
     
  4. mp4nerd

    Thread Starter New Member

    Dec 1, 2012
    4
    0
    Here are my schematics and transient and AC analysis for the highpass filter @ Input(f) = 20kHz:

    [​IMG]

    [​IMG]

    [​IMG]
     
  5. Audioguru

    New Member

    Dec 20, 2007
    9,411
    896
    The resistor values in your differential amplifier are EXTREMELY low. The signal source and the opamp you are using will might not be able to driver them.
    The biasing resistor values are also EXTREMELY low which is not needed for biasing the opamp since its input bias current is extremely low. Oh, why are you also biasing the signal source?

    Your lowpass filter response is EXTREMELY gradual since it has only one RC. If it cuts 500Hz by -3dB then 1kHz is down only a little more and there is still plenty of 2500Hz there.
    A third-order Butterworth filter will have a much sharper cutoff where 1000Hz will be at -18dB and 2kHz will be fairly low at -36dB.
     
  6. mp4nerd

    Thread Starter New Member

    Dec 1, 2012
    4
    0
    Okay, well I am trying to bias the signal source because the ADC on the Atmega32 uP can only take in positive voltage values. I was also trying to minimize the number of stages of the filter since the output on the LED side doesn't require that much precision and the amount of space left within our project container is decreasing.

    Mainly I am trying to match the phase response of the output signal with the input signal and set the DC offset of the output signal to 2.5V. Any suggestions on how I should alter/calculate the appropriate RC values to achieve this using one (if possible) or two stages?
     
  7. Audioguru

    New Member

    Dec 20, 2007
    9,411
    896
    Why are you trying to match the phase of th input and the output?
    A filter chages the phase.
    You cannot hear a phase change.

    Decide which is more important, the filter or the phase.
     
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