switching(class d?) audio amp schematic

If you don't want an all-digital solution, you can use analog feedback to improve the performance. Sense the output voltage right at the output of the full-bridge switching stage (so, before the low-pass passive filter) and use it to correct the input voltage to the analog PWM generator. If you do it right, the benefit can be impressive.

Also, make sure that the dead times at the switching stage are not too long. Supplies and PCB should be carefully designed, too. If stereo, use separate supplies (or at least separate supply filters) for each channel.

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What i don't understand is how can i sense the voltage without integrate the PWM with another low pass filter Should i measure it directly as a time?I mean with stuff like a microcontroller or other digital circuit? I'm filling the registration module for that forum that Genome suggested me,looks very interesting! Thank you again

Sense the output voltage between the switching stage and the low-pass filter, scale it down, convert it to asymmetric (if the input is so, and you are using a full-bridge), low-pass filter it, subtract it from the input voltage, and add some amount of the resulting error to your input voltage, to create the signal that will drive your analog PWM generator.

No, you don't measure the pulse widths, and you don't need any digital part. The LPF takes care of going back from PWM to a continuous analog signal that you can use for the comparison (subtraction).

Best,

^ ^ ^ ^

So, you originally said to take the signal before the integrator? And you say to LPF it anyway???

Tim

-- Deep Fryer: a very philosophical monk. Website:

I didn't mention any integrator.

Don't know if I understood you. The LPF for the feedback path does not need inductors, and can be more linear and with tighter tolerances than the high-power, passive LPF at the output of the switching stage.

Best,

It seems to me a LPF integrates. Whatever you're doing, if you view it in the time domain, you're integrating PWM'd up and down pulses into a smooth, slower waveform. In the frequency domain, you're cutting out the carrier and sidebands, leaving the low frequency band of interest. Same thing.

Well, the goal is to get excruciatingly linear *output* voltage, isn't it? So wouldn't the idea be to correct the output voltage itself?

BTW, if you drop inductors, you can't get as sharp a cutoff response in the loop, correct? So to get the same amount of filtering of the carrier, you'll really slow down the loop, possible causing oscillation? Fast op-amp active filters aside.

Tim

-- Deep Fryer: a very philosophical monk. Website:

False. Try dc input.

An integrator has infinite dc gain. An LPF does not.

That can yield better performance, but it is more complex to design the feedback so that the system is stable, and I don't recommend it to someone that wants to do it himself. Besides, this is analog. To aim at the highest performance, I would recommend an all-digital solution (upsampling, uniform-to-natural sampling conversion, noise shaping, digital pwm generation, MOSFET driver, full-bridge switching stage, passive low-pass filter and some elaborate feedback).

You don't need op-amps, but it is not that you cannot use them. You can, if you want, and know how to.

Best,

Well obviously, it fails for very slow rates. But I was hoping you had the intelligence to integrate over a period somewhat shorter than the audio you're trying to amplify in the first place......

Fair enough. How about frequencies of interest? Hmmmm?

Tim

-- Deep Fryer: a very philosophical monk. Website:

Tim, what's your problem? Tell us.

Integrator : -20 dB/decade everywhere. LPF : approx. flat in its passband.

Do they still look like the same to you?

Besides, which are the "frequencies of interest" when you work with an integrator? What's its bandwidth? What's its cutoff frequency?