When you add a sub to a system, you need to:
1. Volume match the sub to the mains. Failure to do this will result in thin or overwhelming bass.
2. Smooth out the frequency peaks and dips in the room because ALL bass does this, not just subs. Failure to do this will result in lumpy sounding bass.
3. Low pass the sub and high pass the mains. Failure to do this will result in misaligned phase over the band where freqs overlap, producing cancellation at many points in the FR.
4. Align the phase at the XO region. Failure to do this may produce cancellation at the XO point.
5. Time align the sub to the mains. Failure to do this will result in "flabby" sounding bass, where the upper freqs are heard first, and the bass freqs 10-20ms later. It will sound disjointed.
And all this is assuming that the subwoofers are properly placed in the first place. These are your friend's options, from worst to best:
1. Do not use any type of crossover. Use the mains full range, low pass the sub only.
This is what the majority of people seem to do (the subjective side of audio anyway). All they do is adjust the controls on the SW plate amp until it sounds right - i.e. rough volume and phase matching and maybe some PEQ. Most people do this subjectively, but it is extremely difficult to do without measurements, and it is impossible to achieve a satisfactory result with any degree of precision. This is REL's recommended approach. I think it is utterly stupid, it is really disappointing to see a subwoofer manufacturer mislead their customers this way. REL surely knows better, yet they spread misinformation.
2. High pass the mains, low pass the sub with an external analog crossover, or built-in XO in the sub.
This is what the remaining majority of the subjectivists do, usually without measurements. This is better, because the overlap region is smaller, and therefore less potential for phase cancellation. But - still no time alignment nor phase alignment, and no DSP for the natural peaks and dips in the FR. Furthermore, if the subs have DSP but the mains don't (e.g. as seen in Velodyne subwoofers), the DSP itself introduces additional latency, up to 30ms. This is HUGE and it will definitely worsen the time misalignment and produce flabby sounding bass. This DSP is usually low resolution and IIR because of the requirement to keep latency as low as possible whilst running on hardware with minimal processing power, so it will not correct bass freq peaks/dips in a rough fashion.
3. Use a hardware based DSP to high pass the mains, and low pass the subs
Examples of hardware based DSP: MiniDSP, built-in bass management of AVR's, Genelec GLM, DLBC/Dirac Art, etc. This is the most common recommendation you will see on ASR but it is still not the ultimate. This is better, because it ticks nearly all the boxes of the requirements laid out above. Nearly all hardware based DSP use automated software algorithms that removes decision from the user and these algorithms usually get it wrong, usually with no way to over-ride the algorithm. For e.g. some AVR's calculate the delay by directing you to enter the distance of the sub from the listener. This is wrong, the delay should be measured with a microphone with a timing impulse (we can discuss why). Also, processing is done on low powered hardware, and the low latency requirement again means IIR filters or mixed phase filters and USB microphones (or even worse - cheap uncalibrated microphones). So while there is some bass management, it is not the best - better precision, but not great precision. It may correct the bass up to 80-90% of what needs to be done if the algorithm gets it right. If you use this approach, you need to be aware of the shortcomings and do what you can to mitigate it.
4. Use software based DSP to high pass the mains, and low pass the subs
Examples: Acourate, Audiolense, Focus Fidelity, REW/RePhase, Matlab, Octave. These software packages vary in the degree of automation (i.e. use of a software algorithm), but most let you over-ride the defaults and give you more control over what you need to do. Some (like Acourate, REW/RePhase) are completely manual and relies on you to interpret graphs and make decisions. Octave and Matlab are even more manual in that you need to enter mathematical equations. It is my belief that manual correction is superior to "one button DSP" if you know what you are doing and most people don't, even ASR readers. The disadvantage of this approach is the steep learning curve, requirement for a certain type of hardware configuration (i.e. some kind of computer must always be in the signal chain), lack of convenience, and difficulty of use.
At the end of this, you need to explain to your friend that better correction comes at the cost of greater difficulty. He needs to choose how serious he is about bass correction, and how much inconvenience he is prepared to put up with.
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