What characteristics help main speakers integrate with a sub?
Major factors would be;
1. transient characteristics/behaviors of your sub(s) (SWs) and main woofer(s) (WOs) around the crossover frequency (XO Fq),
2. determination of XO Fq,
3. lower-side filter (low-pass = high-cut) selection and slope (can be done in upstream DSP as well as in sub's built-in filter),
4. higher-side filter (high-pass = low-cut) selection and slope,
5. polarity (phase) continuity selection (to be optimized at your listening position),
6. time alignment (partly phase continuity) between SWs and WOs (in mSec level precision),
7. optimization of relative gains for SWs and WOs,
of course, all of them should be hopefully measured and tuned with your own SWs and WOs in your home acoustic environments including your actual room modes.
Using 8-wave and/or 3-wave rectangular sine tone burst signal of Fq around the XO, and simple recording of the tone burst sound at your listening position, then analysis of the recorded tone burst sound by suitable audio software like Adobe Audition and Audacity,
you can visually observe the transient behavior (tightness/compactness of the sound) of your SWs and WOs at your listening position, especially by 3D (Fq-Gain-Time) color spectrum of Adobe Audition (I use Adobe Audition 3.0.1). I believe this would be very reliable and reproducible semi-objective tuning method which you may easily perform/repeat using suitable single measurement microphone set at your listening position.
Such provisionally determined sub and woofer configurations should be subjectively checked and further optimized using your preferable music playlist.
As I recently wrote
here...,
My simple suggestion for your first step is to prepare two 8-sine-wave rectangular tone-burst signals of about -15 dB gain, one is 40 Hz, another one is 90 Hz. Using typical measurement microphone set at your listening position, you would first record
SWs-only sound of 40 Hz tone-burst and 90 Hz tone-burst by using suitable audio-interface connected to your PC or Mac in 96 kHz 24 bit PCM format; no low-pass (LP) (high-cut) filter should be used.
Next, you would record
WOs-only sound of 40 Hz tone-burst and 90 Hz tone-burst, with no high-pass (HP) (low-cut) filter at all.
Then, you would record
SWs+WOs (singing together) sound of 40 Hz tone-burst and 90 Hz tone-burst, with no LP nor HP filter at all.
Such recorded three PCM tracks should be loaded to
e.g. Adobe Audition (ver.3.0.1 in my case) for 3D time-gain-Fq color spectrum analysis which gives visual representation of sound energy distribution in 3D (time-gain-Fq) space; you can see/observe tightness of the tone-burst air sound, yes, it represents "transient behavior" of your SWs and WOs as well as SWs+WOs for 40 Hz and 90 Hz tone-burst. Of course, you can see/observe the actual air sound wave shapes given by SWs, WOs and SWs+WOs.
Generally speaking, HiFi 30 cm WOs in excellent cabinet driven by high-damping-factor amp has better transient behavior than any SWs around 50 Hz - 120 Hz zone, and the Adobe Audition 3D color spectrum will give you visual observation/confirmation for selection of rather lower XO Fq, say XO at around 55 Hz, in this example case.
Now you can easily understand what would/should be the subsequent tests and measurements.
You may now apply LP filter for SWs,
e.g. -24 dB/Oct Linkwitz-Riley (LR) at 60 Hz, HP filter for WOs
e.g. -24 dB LR at 50 Hz, and record the room air sound of SWs-only, WOs-only, SWs+WOs, analyze them by Adobe Audition's wave spectrum and 3D-color spectrum. The wave-shape spectrum will give you visual info on phase continuity (for time-alignment), 3D-color spectrum will give you transient behaviors/characteristics (tightness/compactness of low-Fq sound energy distribution).
In this way, you can semi-objectively optimize (or best tune) XO-Fq, filter slopes, polarity, relative gains, time-alignment,
etc. between your own specific SWs and WOs in your own acoustic environment. (Of course, final decision should be done by subjective listening to various tracks of your preferred music!)
If you would be interested in such approach, you can find some example cases of spectrum analyses using Adobe Audition 3.0.1 on my project thread as follows;
- Measurement of transient characteristics of Yamaha 30 cm woofer JA-3058 in sealed cabinet and Yamaha active sub-woofer YST-SW1000: #495,
#497,
#503,
#507
- Precision measurement and adjustment of time alignment for speaker (SP) units:
Part-1_ Precision pulse wave matching method: #493
Part-2_ Energy peak matching method: #494
Part-3_ Precision single sine wave matching method in 0.1 msec accuracy: #504,
#507
**************
Edit: If you would be seriously interested in these measurements and tunings, I will be happy to share the test tone signals I prepared for the measurements. If this would be the case, please simply PM me writing your wish.
**************
These posts would be also of your reference and interests;
- Perfect (0.1 msec precision) time alignment of all the SP drivers greatly contributes to amazing disappearance of SPs, tightness and cleanliness of the sound, and superior 3D sound stage: #520
- Not only the precision (0.1 msec level) time alignment over all the SP drivers but also SP facing directions and sound-deadening space behind the SPs plus behind our listening position would be critically important for effective (perfect?) disappearance of speakers: #687
- Reproduction and listening/hearing/feeling sensations to 16 Hz (organ) sound with my DSP-based multichannel multi-SP-driver multi-amplifier fully active stereo audio system having big-heavy active L&R sub-woofers: #782
Finally, just for your possible reference, you would please find
post #931, on my project thread, the latest total system setup of
my PC-DSP-based multichannel multi-SP-driver multi-amplifier stereo audio rig with large and heavy L&R SWs (YAMAHA YST-SW1000).