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Class A vs AB vs D amplifiers

Modern high-end Class D amps have as good or better noise, distortion, and power than most of the Class A or A/B amps out there, and especially when you compare by price.
Power, yes. But SINAD (THD+N) is only better when AES17 low pass filter is applied.
 
But the problem I see is that our audio measurement methods are still in the dark ages. Seriously measuring amplifiers using sinewaves is dinosaur methods.
Because nobody uses multitone either :facepalm:

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Yes, it's still sinewaves, but the resulting waveform is anything but repeating and static. It very much resembles a music-like waveform, except that there is way too much high frequency in there, which by your logic should actually make it much harder to reproduce.

Your methods are even worse than Dinosaurian and smell like 400-million-year-old fish... I knew there was something fishy going on ;)
 
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If anyone has ever taken higher math,
You need to be bit careful on ASR. Many of us did university-level mathematics. Some post-grad. Others lecture on the subject!
 
Also the Fourier theorem cannot be applied to non repeating signals.
Really. How then did Audacity create this spectrum plot from the song "my name is"?

And there are many far more advanced systems for audio analysis - what do you think they are all using?
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MP3 doesn't work... there I said it!
Hardly surprising given it's a spin-off from a video/TV standard. :p

[Edited to add the smiley]
 
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Hardly surprising given it's a spin-off from a video/TV standard.
Interesting. I wonder what @amirm has to say about all of this seeing as how he was a lead in charge of something like 1000+ engineers figuring this out. :D I find good MP3s to be great.
 
Really. How then did Audacity create this spectrum plot from the song "my name is
That is a song with contious content. Try it on a click. I have a digital oscilloscope. It explains that no fourier transform can be created on a single cycle of a sinewave. Every test I have read about ALWAYS uses a wave train of multiple cycles.

Try to sample, using the CD sampling rate, of a half cycle of a 10kHz sine wave and you will see that the reconstituted output will not resemble the input. I am not saying its possible to hear the difference, just saying with all these flaws it seems completely pointless to design amplifiers that have -140dB of distortion when you can't hear any difference between that and one that has -80dB harmonic distortion.
 
Try to sample, using the CD sampling rate, of a half cycle of a 10kHz sine wave and you will see that the reconstituted output will not resemble the input.
The problem is that if you only draw half a cycle, that’s just a drawing of a cut off sine… it’s not a properly band limited signal. If it were it would sample and reconstruct perfectly well.
 
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Interesting. I wonder what @amirm has to say about all of this seeing as how he was a lead in charge of something like 1000+ engineers figuring this out. :D I find good MP3s to be great.
Sorry, forgot the smiley!
 
The problem is that if you only draw half a cycle, that’s just a drawing of a cut off sine… it’s not a properly band limited signal. If it were it would sample and reconstruct perfectly well.
@Blenderguy

In other words - put the half sine cycle through a 20Khz low pass anti aliasing filter first, and then you can sample it fine. But the band limited signal won't closely represent a half sine wave either.

Doesn't matter though - cos if you try to listen to a very high bandwidth version of the signal - your ears will cut off the high frequencies also - in a similar way to the low pass filter.

For someone who professes to know so much about audio - you seem to have a pretty sketchy understanding of digital audio. Or the human auditory system.
 
I have a digital oscilloscope. It explains that no fourier transform can be created on a single cycle of a sinewave.
If your oscilloscope is explaining things to you, time to switch the meds.

I'm amused that you've moved from patently incorrect understanding to severely incomplete understanding. I guess that's progress. But still, an indication that you need to actually study the math and basics of signal processing before trying to "explain" things on a public forum.
 
Wrong explanations, lack of education and plenty of misinterpretation are the basics of public forum with laymen audience. You should have known and you should have known that all the fight against those facts is hopeless.
 
That seems pretty high. I'd expect around 0.7 to 0.8uH / m of "2 core", so < 3uH for 12ft. Which would push resonance to 50kHz if your estimate of 2uH for amp output is correct. (EDIT - which also sounds high for a low impedance solid state amp)
You have to double the inductance because a 12ft long speaker cable is actually 24ft long. The one amp schematic I saw used a 2uH output inductor as part of the Zobel network.
You can design an amplifier without it, but it either has to have a very high open loop bandwidth, which generally means low open loop gain and hence higher distortion. I don;t know how this applies to class D amps. The have an output filter and inductor is probably much smaller since the switching frequency is high.
 
Multitone is a good test, I'd like to see what happens when you switch off any one of the carriers to see what is actually in it's place. Multitone is what is used to measure composite triple beat.
You need to be bit careful on ASR. Many of us did university-level mathematics. Some post-grad. Others lecture on the subject!
I guess I should be quite on this subject. Maybe ask questions instead.
But it certainly turned out to be a lively subject with lots of postings.
 
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