When it comes to digital signal processing, there’s one puzzle that remains even once processing is complete. How do you fit the processed signal back inside the permissible limits of the digital number format? This post describes the “why” and the “how” of two different approaches you can take to get around this obstacle and finish the operation.
The two weakest components of a HiFi system are typically the loudspeaker and the room the music is playing in— the second of which is most often overlooked. Even if you’ve invested in a best-in-class HiFi system, the listening room can still have a tremendous effect on the overall sound experience. Both a sound system’s frequency response* and impulse response** are profoundly altered by everything from standing wave patterns to wall reflections.
When you’re listening to music and something feels off, it can usually be attributed to at least one of two factors. Either something is out of key— for instance, an instrument isn’t tuned properly or a singer can’t sing. Or someone is missing a beat. If each musician in an orchestra were to play at their own tempo it would sound differently than intended, and likely pretty bad. The first of these factors is a question of frequency for a single sinusoid (does each note sound like it should?). The second is a property of time (does each note arrive when it should?).
As a former HI FI and car stereo dealer I know people spend a lot of money trying to get the perfect sound. To be honest I haven’t looked into the business so much since the 90’s, but after a few months back in the segment, and a few exhibitions later, I see that little has changed since then. People are still spending just as much money on cables, contacting, racks, and turntable weights as ever.