Updated: May 8
The idea of using electromagnetic force to create sound by moving a membrane is almost 100 years old. In the early days, audio amplifiers were weak and only used a few watts of power. Therefore, speakers had to be huge to create adequate sound pressure. As for quality – let’s just say that everyone’s jaws dropped if they could discern that music was playing or understand what recorded voices said. Nowadays, amplifiers are tiny and deliver more power than ever. Advances in material sciences mean even small loudspeakers can move a lot of air with a strong enough amp. But what about quality? Sadly, the expectations haven’t moved much since the old days. But that shouldn’t be the case – read on to learn why!
Traditionally, speakers consist of three components:
The loudspeaker units - which turn electricity into audible moving air
The enclosure - which keeps everything in its right place and helps play bass
The filtering circuit - prepares the audio signal so that it works correctly with the other components
During the 100 years since the inception of the moving membrane loudspeaker, all three have seen innovation.
Loudspeaker units are now computer-modeled, and membranes often use space-age materials like metal alloys, ceramics, or composite material sandwiches. The result is that even small speakers can resonate enough to generate low-frequency sounds.
The enclosure seemingly has the least amount of innovation. It’s still essentially a box that holds the other components together. Computer-assisted modeling offers the largest leap forward in enclosure design, allowing us to determine what shapes create the least sound interference.
Modern materials like cast aluminum, carbon, or wood composites are made into all sorts of shapes. However, manufacturing these shapes is rather expensive, so the mainstream speaker usually uses an MDF box with some surface finishing to look nice.
Filters are where things get interesting. A simple filter consists of electronic components that divide the signal so high-frequency content reaches the tweeter, and everything else gets sent to the woofer.
In a three-way system, extra circuitry feeds the mids to the mid-driver. Very high-quality loudspeaker drivers that play close to the mathematical ideal are required to get the best sound from this approach. In reality, ideal drivers only exist in manufacturer sales brochures. Designers must wrangle with physics to keep production within budget.
Even then, effects like manufacturing tolerances are rarely accounted for except in top-of-the-line speaker systems.
With the advent of digital audio, we discovered digital filtering techniques. With enough processing power, we can make filters fairly complex and close to their mathematical ideals without using more electrical components.
Initially, this technology was picked up and widely used in live sound for cinema and concerts. They have few limitations regarding processors and other gear, and traveling speaker kits always need fine-tuning once moved somewhere new.
Next came studio monitors. Sound quality is paramount in serious studios, as engineers need to hear precisely what they’re doing without speaker distortion standing in the way.
Microprocessors have become so powerful and affordable that digital audio processing has reached consumer sound systems. Every Klear LAYLA soundbar has a digital signal processing (DSP) unit which divides the signal to each speaker. However, there’s another secret ingredient rarely found even in high-end speaker systems.
Early DSP units just copied what their analog predecessors did to a letter. This approach isn’t a bad start, but DSP can do much more! The key is actually measuring before trying to fix stuff. Here are three key aspects of getting it right:
Know WHAT to measure
Know WHAT to correct
Use enough processing power
High-resolution measurement of a speaker’s performance means you can correct the major flaws of all three main components – the speaker drivers, the enclosure, and the filter. What’s more – measuring every speaker from the assembly line, as with the LAYLA soundbar, removes any deviation from imperfect manufacturing tolerances. Therefore, you get performance similar to companies that cherry-pick parts for extra tight matching.
This technique isn’t uncommon in high-end studio monitors like Genelec, Kii, or Dutch & Dutch. However, it’s quite labor-intensive and rarely used in consumer audio. Even high-end audiophile speakers often rely on manufacturing tolerances to deliver great sound and forego custom calibration.
Here at Klear, we decided to let listeners know what they’re missing out on.
Tune Up for What?
So, technically it’s all impressive, but what about, you know… the sound?
The main aspect of any speaker system is the frequency response, the primary characteristic that digital calibration corrects. Frequency or tonal response determines whether the speaker changes its loudness depending on the tone. Ideally, a speaker should treat all sounds equally. Calibration ensures this happens.
An imperfect tonal response messes up a speaker's timbre – sounds become skewed, bass booms and overpowers mids, and treble causes unpleasant ear fatigue.
Get rid of coloration, and all that’s left is music or the sound of whatever media you’re playing. This calibration also tends to extend the speakers' bass response, so don’t be surprised when you get more oomph than you expect!
Calibration is done on both the left and right channels of the speaker for superb channel matching, making the stereo image distinct and stable. Sounds won’t float around the phantom stage unless the recording calls for it. If something’s centered, it’ll sound like it’s coming from the center channel, even if there are only two speakers.
The overall effect is such that people rediscover their favorite music and prefer watching movies at home now that their sound system matches the TV clarity.
All it took was 100 years of innovation!