Class-D or switching amps, any opinions on??


Does anybody have experience on Class-d or switching amps vs either a/b or traditional amps?? I have heard people knock them for limited ability at the low frequencies. However, I listened to a Linn amp not long ago and could not hear it wanting for anything. I want to hear a Rotel switching amp to compare. Why buy a massive 90lb amp thats a space heater if you dont have to, right???
128x128bobrock
Pure digital might not be bad? It's not that good- yet. As for HDTV- I saw a demonstration of Sony's analog HDTV at the Chicago CES in 1989. It was amazing. Today's digital HDTV has a ways to go yet. Just look at the lack of low level detail in dark scenes. I also see pixelation in fast moving scenes in HDTV broadcasts, but not in 1080p blu-ray material.
Al - I'm not sure what they do to obtain decent resolution. In order to get 16 bit resolution and 20kHz bandwidth with traditional PWM clock has to be 65536x20e3=1.3GHz. It is way to fast and no room to filter out carrier. They might play games with modulating power supply at the same time or creating more states (more Mosfets and more voltages). It is getting very complicated to get real digital amp. Pure digital might be not bad - just look at HDTV.
That's an excellent question, Kijanki. I'm not particularly knowledgeable in that area, but I believe that what is done to provide adequate resolution is oversampling + noise shaping, conceptually similar to what is done in the so-called 1-bit cd players. I found the following paragraph here:
The PWM resolution is typically much more coarse (than that of the incoming pcm), in order to keep switching frequency above maximum audio sample rates of 192 kHz, whilst limiting MCK frequency. Typical ratios are between 32 (5 bits) and 256 (8 bits), giving a best dynamic range of 50 dB. Clearly, this means that other techniques are required to provide sufficient dynamic range, and the answer is oversampling with noise shaping. This technique essentially allows a time averaging of the PWM pulse widths to be applied, to deliver fractional resolution in the amplitude of the output signal.
The theory behind that is esoteric and also somewhat counter-intuitive, and I'm not sufficiently familiar with it to be able to explain it well, but I believe that is basically the answer. Note that the Tact manual I referenced indicates an 8-bit pwm resolution, within each pulse period (corresponding approximately to the 50db dynamic range mentioned in the quote above); a pulse rate of 384kHz; and a master clock rate of 98MHz.

Best regards,
-- Al
Here is a pretty good Wikipedia article on noise shaping. Very basically, oversampling + noise shaping allows the quantization noise resulting from the limited resolution to be mostly shifted up to frequencies that are higher than the audio information, allowing it to be digitally filtered without significantly affecting the audio.

Regards,
-- Al
Al - Thank you for the link. My problem is with resolution. As you said TACT stated resolution from PWM is 8 bit. Where do they get required extra resolution (possibly by adjusting power supply?). 384kHz carrier sounds about right (Icepower is 400-500kHz).

Class D is here to stay. Some people don't like the concept of switching in audio but delta-sigma converters, SACD and DDS recordings are exactly that (PWM).
My problem is with resolution. As you said TACT stated resolution from PWM is 8 bit. Where do they get required extra resolution?
Kijanki, though it may seem counter-intuitive, reducing the quantization noise that results from limited resolution is mathematically the SAME as increasing the resolution, apart from any possible side-effects of the digital filtering processes. So they get the required extra resolution by the oversampling + noise shaping + dither processes that I referred to.

Keep in mind that limitation of resolution to a finite number of bits is mathematically identical to summing a noise component ("quantization noise") into the signal. If I recall correctly, assuming random error distribution (which is pretty much assured by applying proper dither), the rms quantization noise amplitude is equal to the lsb increment (in volts) divided by the square root of 12.

The oversampling allows most of the quantization noise to be shifted to higher frequencies than the audio information, where it can be filtered out with minimal impact on the audio. Dither randomizes the process to eliminate "deterministic errors," as the article indicates.

I believe that the other pwm applications you mentioned, delta-sigma converters, sacd, and dds, do the same thing.

Regards,
-- Al