Dynamic Bass Boost

04.04.2009 0:21

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:

Dynamic Bass Boost circuit

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:

Measured frequency response of disabled DBB circuit

(DBB turned off)

Measured frequency response of enabled DBB circuit

(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:

\omega_{p1} = \frac{1}{(R_1\|R_2 + R_3) \cdot C_3}
\omega_{z1} = \frac{1}{R_3 \cdot C_3}
\omega_{z2} = \frac{1}{R_1 \cdot C_1}
\omega_{p2} = \frac{1}{R_1\|R_2\|R_3 \cdot C_1}

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)

Simulated frequency response of DBB circuit

Posted by Tomaž | Categories: Analog

Comments

Great work! Probably I will use this circuit in some of my lower quality projects. I think you can upgrade this circuit by adding an operational amplifier. And I think it is oalso possible to put a potentiometer instead of switch to control the level of the effect. By adding theese improvenents this circuit will work beter and may be used in some higher quality sound devices.
Good luck!
P.S. Sorry for poor English.

Man, I know nothing about circuits but you really laid this out intuitively. I feel like I understand what the DBB on my Sony amp does now and you answered my question about it too: Is it better to use DBB w equalizer or just manually equalize it? It seems from what I could comprehend here that if i'm using a detailed enough equalizer, it would be counter-productive to have the DBB active and interfering since its pretty much an equalizer function itself (I think). I should study this stuff.

Posted by tom

Tom, I don't think that Philips player even had an equalizer. You are correct though: you can think of DBB as just another equalizer with a preset setting - it adds to the effects of any manually adjustable equalizer.

Posted by Tomaž

Tomaž, this would be true for a simple bass boost, but DBB should emphasize the lower freqs more for low volume that for high volume...

Posted by Gert

Gert, amplifying a low volume signal more than a high volume one would mean a non-linear response. I'm sure that is not something you would want in audio equipment since non-linearities mean signal distortion. Think about how a saturated microphone input sounds.

The circuit I analyzed above is most definitely a linear circuit (as are equalizers), which means that the frequency response is the same regardless of the volume.

Posted by Tomaž

You are right in the sense that this circuit cannot achieve dynamic bass boost, because it's passive.
But DBB does not introduce non-linearities, because the time constant of the boosting dynamics should be very slow, increasing the flatness of the frequency response with the average amplitude. Here is a circuit, very linear but passive, that does something similar in the opposite direction:

http://www.redcircuits.com/Page38.htm

Posted by Gert

Are R1 and R2 fixed values ? Why did you draw them in one component ? Otherwise I'm confused cause it states "volume".

Posted by Andy

R1 and R2 are two parts of a single 10 kΩ logarithmic potentiometer that is used for setting the volume. It has a fixed tap (connected to C1 and R3) as well as a sliding contact (marked U0).

Posted by Tomaž

So this is no standard component with 3 pins ? I would like to test this circuit.

Posted by Andy

No, it's not a standard component. It has 4 pins. Unless you can find such a potentiometer you can't directly replicate this circuit.

Posted by Tomaž

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