Ohmic Audio

⚙️ ENGINEER LEVEL: Advanced DSP Theory

Minimum Phase vs Linear Phase

Minimum phase system:

Magnitude and phase responses related by Hilbert transform.

Properties: - Minimum group delay for given magnitude response - Causal (doesn't predict future) - Most analog filters are minimum phase

Phase response:

φ(ω) = -imag[H(ln|H(jω)|)]

Where H = Hilbert transform

Linear phase system:

Properties: - Constant group delay: τ_g = constant - No phase distortion - Preserves waveshape - FIR filters can achieve linear phase

Phase response:

φ(ω) = -ω × τ_g

Comparison:

Minimum phase EQ: - Lower latency - Phase shifts with magnitude - Can cause pre-ringing (minor) - IIR implementation (efficient)

Linear phase EQ: - Higher latency - No phase distortion - Can cause pre-ringing (more severe) - FIR implementation (CPU intensive)

For car audio:

Minimum phase generally preferred: - Lower latency - More efficient - Phase effects inaudible for typical EQ

Linear phase for: - Mastering/production - Critical listening - If CPU allows

Psychoacoustic Optimization

Critical bands:

Human hearing analyzes sound in ~24 critical bands (Bark scale).

Bandwidth increases with frequency:

Center Freq Bandwidth Equivalent Q
50 Hz 50 Hz 1.0
500 Hz 100 Hz 5.0
5000 Hz 1100 Hz 4.5
15000 Hz 3500 Hz 4.3

Implication for EQ:

Optimal Q matches critical bandwidth:

At 100 Hz: Q = 1-2 At 1000 Hz: Q = 3-5 At 10 kHz: Q = 3-5

Audibility of EQ:

Just Noticeable Difference (JND): - Midrange: ~1 dB - Bass/Treble: ~2-3 dB

Below JND: Won't hear difference (don't bother) At JND: Subtle but audible 2× JND: Clearly audible

Masking effects:

Loud sound masks nearby frequencies: - ±1/3 octave for narrowband masker - ±1 octave for broadband masker

Implication: EQ changes in masked regions may be inaudible!

Time/Frequency Uncertainty

Heisenberg uncertainty principle applies to audio:

Δt × Δf ≥ 1 / (4π)

Cannot have perfect time AND frequency resolution simultaneously.

Short time window: - Good time resolution - Poor frequency resolution - Good for transients

Long time window: - Poor time resolution - Good frequency resolution - Good for steady-state

For crossovers:

Sharp frequency cutoff (high Q) creates time smearing.

Example: 80 Hz crossover, Q = 15

Frequency resolution: ±5 Hz (very precise) Time resolution: 1/5 = 0.2 seconds (very smeared!)

Trade-off:

Steep crossover: - Excellent frequency separation - Time smearing (minimal impact below ~200 Hz)

Gentle crossover: - More frequency overlap - Better time domain

Practical implication:

Below ~150 Hz: Steep crossovers okay (wavelength long, time resolution less critical)

Above 2 kHz: Consider gentler slopes (time resolution matters more)

FIR Filter Design Methods

Window method:

Process: 1. Define ideal frequency response H_ideal(ω) 2. Inverse FFT to get impulse response h[n] 3. h[n] is infinite - must truncate 4. Apply window function to taper ends

Window functions:

Rectangular: - No tapering - Sharpest frequency response - Worst sidelobes (-13 dB)

Hanning: - Cosine taper - Good compromise - Sidelobes: -31 dB

Blackman: - More gradual taper - Excellent sidelobes (-58 dB) - Wider main lobe

Kaiser: - Adjustable parameter β - Trade-off between main lobe and sidelobes - Optimal for many applications

Frequency sampling method:

Process: 1. Sample desired H(ω) at N points 2. Inverse DFT gives h[n] directly 3. N taps, length N filter

Advantage: Direct control of frequency response

Disadvantage: Frequency resolution = f_s / N

Optimal (Parks-McClellan) method:

Equiripple design: - Minimizes maximum error - Optimal in minimax sense - Remez exchange algorithm

Results: - Shortest filter for given specs - Equal ripple in passband and stopband - Industry standard

Software: MATLAB firpm(), scipy.signal.remez()


4.3 Using Measurement Tools and Software