Measurement & DSP workflow (concrete, actionable)

Below is a workflow you can run, since you want to get deep into DSP and automation.

  1. Measurement setup

    • Measure each speaker individually: on-axis at 1 m, off-axis at ±15°, ±30°, and 60° (if applicable) in both horizontal and vertical planes to capture directivity.
    • Measure at listening position(s) using sweep deconvolution to get impulse response (IR).
    • For nonlinear/algebraic distortion: measure THD vs frequency at relevant SPLs (e.g., 85 dB, 95 dB at listening position or at 1 m) with pure tones or with special test signals.
    • Measure cabinet resonances with good SNR (windowed IR and high FFT resolution).

  2. Analysis

    • Derive magnitude response and phase/group delay from the IR.
    • Compute ETC / energy decay to spot ringing cabinet modes.
    • Compute in-room power response by integrating off-axis responses or by multi-position averaging.
    • Quantify directivity index (DI) or directivity vs frequency to understand how off-axis energy differs across speakers.
    • Build per-speaker metrics: on-axis SPL deviation from target, group delay error in the critical band (500 Hz–5 kHz), THD mask.

  3. Design correction filters

    • Choose a target curve (e.g., flat on-axis or a slight in-room tilt like Harman-style targets if you prefer). Decide whether you optimize for on-axis or power response.
    • Use minimum-phase IIR for coarse correction where you want low latency and modest correction.
    • Use linear-phase FIR for fine phase/group delay alignment and to match timbre across L/C/R. FIR length depends on how much phase correction you need — be prepared for long filters (tens of milliseconds) if you want perfect linear phase.
    • Impose constraints on filter gain and slope to avoid boosting driver-weak bands excessively.
    • Implement time alignment for arrival times (use fractional-sample delays in FIR or allpass/IIR methods).
    • For subs: apply parametric EQ and group delay alignment; consider multiple-band EQ and crossovers with slopes that allow smooth addition.

  4. Optimization

    • Define a cost function that weights (a) on-axis magnitude deviation, (b) interchannel spectral mismatch (e.g., L vs R vs C), (c) group delay mismatch in 500 Hz–5 kHz, and (d) a penalty for excessive EQ gain or filter energy.
    • Use numerical optimization (e.g., convex solvers, gradient methods) to search for per-speaker filter coefficients that minimize cost while respecting limits.
    • Optionally include listening-position spreading: minimize variance across several listening seats.

  5. Perceptual / HRTF stage

    • If you intend to psychoacoustically render object audio or use HRTF-based distribution, use a head-tracked or fixed HRTF stage after speaker compensation to control localization cues.
    • Implement binaural renderers for headphone checks and compare with speaker output to guard against localization anomalies.

  6. Validation

    • Measure the combined system at the listening position(s) and compare to the target curves.
    • Do blind A/B tests (same listener, same levels) to check for imaging, center stability, and coloration.
    • Listen at different seat positions to check robustness.

Practical limits & tradeoffs you should be ready for

  • Latency: long FIR corrections + HRTF processing add latency — acceptable for movies but not ideal for real-time gaming unless you account for it.
  • Filter length & CPU: linear-phase FIRs can be computationally heavy if you do them per speaker at high sample rates.
  • Subjective artifacts: huge spectral changes or phase corrections that are technically “flat” may sound unnatural; always audition.
  • Diminishing returns: at some point the listener’s perception (and room non-idealities) limit how “matched” things can be made; money and time may be better spent improving speakers or room acoustics.

After this you can go one at a time in REW. Before you start running sweeps go to the back of the amp and turn the channel’s volume pot up while playing a test tone in REW (you’ll have to do this with each channel 1 at a time to match volume and get it to a level you want, then start your sweeps measuring the speakers and DSP correcting them with the MiniDSPs. It’s a long process but WELL worth it.
It’s also worth noting that from here on out you’ll be using the volume control on the computer as the master system volume.
(don’t forget to DSP correct your subwoofer also!)

  1. The Loopback software is cakewalk compared to having to time align and EQ each speaker with REW, then applying those settings to the MiniDSPs. We’re talking hours of work. Then you’re going to have to figure out how to run a split of all channel’s to a bus that can then be used as an LFE crossover channel, that then also gets fed into the main .1 LFE channel for the sub(s). This is would be like running your speakers as “Small” aka not full range, which you should do but that’s another topic. For audiophiles they could really fine tune their system this way, but most people want “good enough” and don’t even want to touch REW (although it’s pretty easy to use). This can all be done MUCH faster by using an AVR’s room calibration like Audyssey MultEQ XT32 or Dirac. On top of that, you’ve only calibrated at the Main Listening Position, you haven’t done any spatial calculations, you have to do those by hand, and by telling the software where the speaker is sitting in space. An AVR does this by taking multiple measurements in all 3 Dimensions.

  2. This will only be usable on a PC. For home theater it gets more complicated. If you want to use a separate device for Movies, Music, Gaming, then you’d have to have a capture card in your pc and deal with added latency, and potential HDMI issues with VRR, Resolution and Refresh Rate switching, and potentially not getting the original video quality. If you have multiple devices you’ll also most likely want an HDMI Selector.

  3. I don’t remember you mentioning any upmixing DSP. So Dolby Surround Upmixer, DTS Neural:X, Auro3D, etc won’t be an option. This would really stink for when you want to upmix a Stereo or 5.1 track/movie to 7.1.4 or higher. Applying a global EQ would have to be manually done on each channel as well, where as an AVR has a simple Bass/Treble option. This applies to creating a “room curve” as well. Audyssey/Dirac have simple ways to apply a room curve via an app.

  4. “Auto Lip Sync Correction” aka Speaker/TV alignment would need to be done manually as well on each channel, instead of using HDMI’s automatic function.

  5. The DAC you picked is most likely performs much worse than most AVRs DACs, I know this doesn’t matter as much for your super budget build, but it becomes a consideration when comparing to an AVR.