thanksPetr, if you click on 'Quote' in the post you want to react to, the quoted post is automagically highlighted in your new post. Much easier to read.
best regards
Petr
You could precise the order of the filter.Of course I know it. It is not the slew rate of the audio signal. But you can compare by yourself the sound different if you add low pass filter at input amplifier about 15kHz and limit the full power bandwidth (slew rate) of the amplifier about 15kHz.
What you could eventually hear with a low pass filter is the slightly lower level in the high end of the audio band.
Nothing related to the slew-rate performance of the amplifier.
You could precise the order of the filter.
What you could eventually hear with a low pass filter is the slightly lower level in the high end of the audio band.
Nothing related to the slew-rate performance of the amplifier.
Is the distortion not affected by slew limiting amplifier? It is the same effect of passive low pass filter?
Have you measure it or have you compare the different of the sound? May be you can not design an amplifier 😀 😀 😀
I'm looking to gamble and win some money on a double-blind ABX test on this.we are sure that the second one will sound better.
Look at my 2600 posts since 2004 and you'll have an idea if I can or cannot design an amplifier.May be you can not design an amplifier 😀 😀 😀
you can not design an amplifier 😀 😀 😀
To be honest, this is an advice:
In my opinion you should stop this nonsense you are doing all the time about saying that "you don't know how to design an amplifier / you can not design an amplifier".
You keep saying this to almost everybody that have another opinion than you or when you simply don't understand what they are talking about.
Also please note that the people that you are attacking with your nonsense knows many hundredfolds more than you when it comes to amplifier design and electronics.
Your comments simply makes you look like an idiot.
Stein
Is the distortion not affected by slew limiting amplifier? It is the same effect of passive low pass filter?
Have you measure it or have you compare the different of the sound? May be you can not design an amplifier 😀 😀 😀
Low pass filtering does not the same as slew rate limiting. Low pass filtering cannot cause distortion, it is a linear process.
Jan
Low pass filtering does not the same as slew rate limiting. Low pass filtering cannot cause distortion, it is a linear process.
Jan

Low pass filtering does not the same as slew rate limiting. Low pass filtering cannot cause distortion, it is a linear process.
Jan
+1000
To be honest, this is an advice:
In my opinion you should stop this nonsense you are doing all the time about saying that "you don't know how to design an amplifier / you can not design an amplifier".
You keep saying this to almost everybody that have another opinion than you or when you simply don't understand what they are talking about.
Also please note that the people that you are attacking with your nonsense knows many hundredfolds more than you when it comes to amplifier design and electronics.
Your comments simply makes you look like an idiot.
Stein
He think I don't know about slew rate. Now, I think he don't know about amplifier designing. It is even. Maybe he is your friend so you are in his side? 😀😀😀
An audio amplifier designer who do not know about the sound different that cause by low pass filter and slew limiting amplifier... 😀
Last edited:
Enough with the disrespect and arrogance towards other members on this board Bimo, please take some time read the forum rules, after that maybe you could humbly teach us how to design an audio amplifier able to handle a square wave with rise/fall time on the order of a few ns without exhibiting "speed distortion", ok?
I never found anything on this question from Graham"Speed distortion" seems to be just another name and spin on Graham Maynard's (RIP) "first cycle distortion", or whatever he called it, but such sudden discontinuous change doesn't exist in the real world of audio, for example a CD audio is heavily LP filtered above 20 kHz so sudden discontinuities would be smoothed out such there wouldn't exist any components above 20 kHz anyway, at least as far as I can see.
Attachments
Fourier analysis
Indeed. An amplifier with feedback should have a low pass filter on the input that removes any high amplitude fast signals that could push the amp into slew limiting.
Perhaps not everyone appreciates that any rapid change is a modulation that creates sideband frequencies, and when those sideband exceed 20KHz, they should be removed to avoid slew limiting. The "first cycle" of a sine burst that abruptly leaves zero is a high frequency spike of unlimited bandwidth that no amplifier can precisely reproduce. Such frequencies do not exist in the output of a real microphone.
Human awareness operates at a rate of about 20Hz/50mS and above that our perception is based on the frequency analysis of the inner ear, sampled at about 10Hz.
It is true that occasionally the un-initiated point out overlooked issues but some DIYA members obviously operate with very limited background in math and there are some names that I don't read so that I'm not temped to try and educate them, given my own limitations.
Just as artists do not attempt photographic perfection, we should appreciate that music is an art and unlimited perfect reproduction is simplistic and unrealistic, or should I say that reproducing frequencies above 20KHz is a flaw and is not desirable. Sophisticated audio professionals understand what is required and desirable and what is pointless waste of time.
Low pass filtering does not the same as slew rate limiting. Low pass filtering cannot cause distortion, it is a linear process.
Jan
Indeed. An amplifier with feedback should have a low pass filter on the input that removes any high amplitude fast signals that could push the amp into slew limiting.
Perhaps not everyone appreciates that any rapid change is a modulation that creates sideband frequencies, and when those sideband exceed 20KHz, they should be removed to avoid slew limiting. The "first cycle" of a sine burst that abruptly leaves zero is a high frequency spike of unlimited bandwidth that no amplifier can precisely reproduce. Such frequencies do not exist in the output of a real microphone.
Human awareness operates at a rate of about 20Hz/50mS and above that our perception is based on the frequency analysis of the inner ear, sampled at about 10Hz.
It is true that occasionally the un-initiated point out overlooked issues but some DIYA members obviously operate with very limited background in math and there are some names that I don't read so that I'm not temped to try and educate them, given my own limitations.
Just as artists do not attempt photographic perfection, we should appreciate that music is an art and unlimited perfect reproduction is simplistic and unrealistic, or should I say that reproducing frequencies above 20KHz is a flaw and is not desirable. Sophisticated audio professionals understand what is required and desirable and what is pointless waste of time.
Originally posted by Jan.Didden
Low pass filtering cannot cause distortion? It is a linear process
You're right! But any filtering leads to additional distortions: amplitude, phase and speed, no matter how we call them linear. In any case, these distortions are heard by sophisticated listeners. Do a simple experiment in the studio: close your eyes and listen to the person's voice amplified by the microphone, amplifier, and reproduced by the speaker system. Then turn off the microphone and listen live. Feel the difference. And this is without recording and subsequent playback of "canned" sound
Originally posted by steveu
Just as artists do not attempt photographic perfection, we should appreciate that music is an art and unlimited perfect reproduction is simplistic and unrealistic
That's how we got to music in mp3 format with a bit rate of 128 and below. When approaching amplifier design from your perspective, it is really unrealistic to get the sound close to real.
Last edited:
I never found anything on this question from Graham
Funny that people keep explaining to you what's going on and you just ignore it and continue on this nonsense.
This forum is great to learn and understand things that one didn't know before; it has helped me numerous times. I thought I would return the favor, but this is disappointing.
Jan
I really hope that it will help this time too, and if you are unable to understand what the speech is about, then I have nothing to do with itThis forum is great to learn and understand things that one didn't know before; it has helped me numerous times.
see a small addition at the end
Attachments
A triangle wave has again infinitely fast changing waveform, so again you test the amp with an infinitely fast, infinite wideband signal. Clearly, the amp cannot handle that so will show output differences with the input signal which you call 'speed distortion'.
Never thought that people would be so creative to call a well-known phase shift or transition delay 'speed distortion'.
Jan
I also protest against copying my posts here to a document and then add your own opposition.
The dishonesty of using my posts without the opportunity to argue my reply is appalling.
Off to the ignore list, I am done with such dishonesty.
Jan
Never thought that people would be so creative to call a well-known phase shift or transition delay 'speed distortion'.
Jan
I also protest against copying my posts here to a document and then add your own opposition.
The dishonesty of using my posts without the opportunity to argue my reply is appalling.
Off to the ignore list, I am done with such dishonesty.
Jan
Last edited:
Excuse me, but isn't the elephant in the room that nobody is taking about input filters used in real world amplifiers (1-pole? 2-poles?), their actual attenuation verses frequency, the maximum frequency at which the filter remains effective as a consequence of parasitic reactance, and the effect of layout in real world amplifiers that may allow very high frequencies to find a way to at least partially sneak around an input filter that looks fine on a schematic? Then of course there is the other antenna input at the speaker terminals to think about.
Sure, if theory is followed perfectly then fast amplifiers shouldn't sound any different than slower ones. If they do sound different, it may simply mean that in the real world no one or almost no one has mastered the practical implementation of theoretical perfection.
Sure, if theory is followed perfectly then fast amplifiers shouldn't sound any different than slower ones. If they do sound different, it may simply mean that in the real world no one or almost no one has mastered the practical implementation of theoretical perfection.
Last edited:
You have a mess: the phase shift as it was a phase shift has remained, the group delay time of the signal passage has not been canceled either, the GDT is the GDT. You still do not understand what speed distortion is. Here are the drawings from the book of Jiri Dostal, carefully understand what is whatNever thought that people would be so creative to call a well-known phase shift or transition delay 'speed distortion'.
Jan
Attachments
Last edited by a moderator:
The experiment you described has a bunch of non-excluded factors and can't be cited as evidence. Even timbral differences will allow you to hear the difference, not to mention the rest.You're right! But any filtering leads to additional distortions: amplitude, phase and speed, no matter how we call them linear. In any case, these distortions are heard by sophisticated listeners. Do a simple experiment in the studio: close your eyes and listen to the person's voice amplified by the microphone, amplifier, and reproduced by the speaker system. Then turn off the microphone and listen live. Feel the difference. And this is without recording and subsequent playback of "canned" sound .
- Home
- Amplifiers
- Solid State
- Bob Cordell's Power amplifier book