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I am designing a class D amplifier for a pair of bookshelf speakers (Sony SS-MB150H).
I guess inside each speaker there is a crossover for each driver but… am I right to assume that I just need to feed a PWM to the speaker and it will work just fine? Or do I need to put a band pass filter (from 20HZ to 20kHz) on the output of the class D amplifier before feeding it to the speakers?
Thank you so much for your attention!
None of the options is good. The speaker produces a sound pressure level for a certain wattage inputted and this is the criteria you need to look at. Some speakers are more effective than others and this effectiveness is largely determined by their enclosure. Luckily, for any given size and shape of speaker the enclosure is roughly constant.
Decide what SPL you need and look at speaker specs to shortlist a handful then design your enclosure to suit the shortlisted items then do comparative studies. You are obviously looking for bandwidth, loudness and distortion as the key ingredients of the selected speaker. Choose also, the speaker that is capable of withstanding the power output that can be thrown at it. Choose an amplifier that won't melt down ie has protection built in.
This amplifier will not be able to drive large 8 ohm speakers very loud. I conclude this from the two 5 ohm resistors present at the emitters of the output transistors. The amplifier will be sufficient to drive a small speaker like the ones most people use with their PC.
It will be able to drive three 8 ohm speakers in parallel but it will not be able to drive these very loud. Most power will be lost through the two 5 ohm resistors !
But why build your own amplifier from discrete components ? If you want the learning expercience then EXCELLENT :-) go ahead and build. If you want something a bit easier, there are ready-to-use modules on ebay like this class D type or this TDA2030 based one. These can only drive one speaker per amplifier but they are cheap. Also, these don't require the +/- 15 V but only a single 5 or 12 V supply.
Oh, and volume control is done like this:
This uses an LM386 audio amplifier, you can just replace it with the amplifier circuit from your question. Use a LOGARITHMIC potmeter (marked in red) to get nice volume control. For fun temporarily replace it with a standard LINEAIR potmeter and notice how the volume control is "crazy" ! :-) Why is that ? Because our ears are also sort of "logarthmic".
Best Answer
Filterless design is well possible. If you have say 200kHz pulse frequency, the slowness of the speakers do the needed lowpass filtering.
But your modulation principle must be different than the simplest possible PWM. If you have bipolar square wave and your modulator pumps that to the speakers with 50% duty cycle for 0V output, your speakers probably burn if you have no filter in front of them. The needed filter is a lowpass filter. Bandpass filter to kill DC is needed if you make it even more simple by having unipolar pulses.
The proper filterless amp has 3 output voltages, say +20V, -20V and zero. If the intended audio output voltage is positive, the amp gives 0V and +20V with appropriate duty cycle. If the intended audio output voltage is negative the amp gives 0V and -20V with appropriate duty cycle. Output 0V means that the amp gives constant 0V to the speaker.
Actually a filter is still needed. But that's not for smoothing audio signal. Speaker's inertia does it. The filter is needed to kill radio interference. Too sharp pulses would jam radio signals. In addition the circuit itself must have metal shield like switchmode power supplies. Many D-class amp ICs operate at so high frequency, say 2MHz that they have no effect on AM radio, but at higher bands the EMI can be potentially disastrous.
Read this for more info: https://www.eetimes.com/filterless-class-d-simplifies-audio-amplifier-design/
The actual design isn't a trivial task. Like pulse circuits generally, the circuit layout and supply voltage decoupling is critical. In addition high quality audio amplification must have closed loop feedback design, which is especially tricky and to get it right, one must have a strong on experience based intuition or brilliant application skills of the feedback control theory (=math).
As a shortcut find ICs which are designed for this and read their application notes. This is Texas Instrument's starting page for the subject: http://www.ti.com/audio-ic/amplifiers/speaker-amplifiers/overview.html