Building an audio phase adjuster to adjust a range of signals

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SUMMARY

The discussion focuses on building a phase adjuster for Low Frequency Noise (LFN) audio signals within the frequency range of 50-400 Hz. A digital solution utilizing a Digital Signal Processor (DSP) and Finite Impulse Response (FIR) filters is recommended over an analog approach due to efficiency and cost considerations. Key components include applying Fast Fourier Transform (FFT) to calculate FIR taps and implementing analog anti-aliasing filters for signal integrity. The design must also incorporate a mechanism for controlling the desired response vector.

PREREQUISITES
  • Understanding of Digital Signal Processing (DSP)
  • Familiarity with Finite Impulse Response (FIR) filters
  • Knowledge of Fast Fourier Transform (FFT) techniques
  • Experience with analog anti-aliasing filters
NEXT STEPS
  • Research Digital Signal Processors (DSP) suitable for audio applications
  • Learn about designing Finite Impulse Response (FIR) filters
  • Study the implementation of Fast Fourier Transform (FFT) in audio processing
  • Explore analog anti-aliasing filter design techniques
USEFUL FOR

Audio engineers, signal processing specialists, and developers working on audio signal manipulation and optimization will benefit from this discussion.

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My colleagues and I are building a phase adjuster for LFN (Low Frequency Noise) audio signals. Since we would like to be able to adjust a range of signals that encompasses a range of around 50-400 Hz, an op amp-driven, analog solution would not be economical, nor efficient. Since we need to be able to adjust a range of frequencies, we are looking at a digitally converted solution utilizing potentiometers. If a digital device is not feasible, can anyone describe how to construct an analog solution for real-time phase adjustment?

Has anyone had any experience in this area?
Does anyone know of any resources one can use to construct the aforementioned device?

Thanks in advance.
 
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Sounds like you need a Digital Signal Processor to apply a complex Finite Impulse Response (FIR) filter to your sound. The taps of the FIR would be calculated by Fast Fourier Transform applied to the desired response, i.e. the gain and phase desired for each frequency in your range. The sampling rate of your system governs the number of amplitude/phase bins that will fall within the desired range of frequencies. An apodizing function must be applied to the desired response vector before applying the FFT. This is required to control undesired energy.

Your system must also have an analog input and output with appropriate (analog) anti-aliasing filters to reject products between the sample frequency and the signals being processed. Some mechanism must be provided to control the desired response vector from the controlling source (humans?).
 

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