Avian’s Blog

A while ago I was doing some fairly trivial repairs on a Philips portable radio/CD/cassette player. It had a sticker on it claiming that this particular product features something called Dynamic Bass Boost.

From the outside this means an extra switch, that when turned on, causes the music to sound better (well, different). It appears that this is just another marketing name for a filter that tries to compensate for the frequency response of the human ear (sometimes it's called Loudness, my Sony calls it Dynamic Sound Generator).

The idea is that the ear is less sensitive to low and high end of the spectrum and that the sensitivity varies with the volume. So to compensate you insert a filter into the audio amplifier with a transfer function that matches the inverted ear sensitivity function.

I was curious how exactly that filter is implemented, so I disassembled the radio a bit further than it was strictly necessary for repairs and had a look at the circuits. Judging by the patent application I was expecting something complicated, with active filters and such.

Well, it turned out that in practice it is a lot simpler than that:

This little passive filter circuit is all there is behind that button. It's implemented on a small circuit board that's attached directly to the rotary logarithmic potentiometer for volume adjustment (R1 and R2 on the picture) and is duplicated for each channel. Input is marked Ui and output is Uo. The button merely shorts C2 to the ground.

Unfortunately, the capacitors and resistors didn't have any recognizable markings on them, so I wasn't able to simply read their values. Instead of measuring each component separately, I measured the circuit's frequency response using a signal generator with frequency sweep and a digital oscilloscope with a Fourier transform:

(DBB turned off)

(DBB turned on)

The potentiometer has a tap at around on third of its range. As you can see from the circuit, with volume between zero and the tap position the shape of the transfer function is constant while at higher volume settings the transfer function gradually flattens out (compare that with the equal-loudness contours - the curves are flatter at higher volumes).

The measurements above were done with the volume knob near the tap, so that the filter was most effective.

Interestingly, the circuit attenuates lower frequencies even when it's "turned off". I wonder if this is intentional, to make the difference in sound more noticeable when you turn it on.

From the viewpoint of circuit analysis, it's pretty complicated to calculate the transfer function analytically without any simplifications. However if you assert that responses of the two capacitors are independent, you get a Bode plot with two poles and two zeros:

The first pole and first zero are responsible for the decreased attenuation at the lower frequency range (they disappear when the switch shorts C3), while the second pole and the second zero result in the steady slope towards higher frequencies.

Here's the simulated filter attenuation versus frequency, this time in the more familiar logarithmic scale. I chose the values of the components so that the shape roughly matches the one I measured on the actual circuit (these are also the values written at the schematic above)