I am not sure what you mean. A transducer whether it's a speaker or a microphone works the same way. If a speaker can reproduce, a microphone can pick it up, If all a speaker can do is an impulse, you don't have the harmonic content, or the timbre, of a square wave, therefore you can't hear it. That's the very basic, I don't know if you where looking for a more low level mechanics answer, this one I don't have. But What about PWM in the audio band? Should it all measure all the same? what should this measurement look like in this case?
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High-resolution Speaker Frequency Response & Distortion Measurements and Why We Need Them (video)
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dc655321
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Based on levels and/or transitions between levels, yes.
Levels themselves are DC. It’s in the name.
That was my point. Lost in translation probably.
Yes of course, but in the audio band the concept of frequency remains the same one. A X Hz square wave has a fundamental frequency X is heard at a pitch of X and has harmonics at all odd factors of X.
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"... If a speaker can reproduce, a microphone can pick it up..."
But a speaker cant produce any spl at a stady state. And a microphone pics up spl. And pwm changes nothing on that. You wont hear a real rectangel, you wont measure it on the mic. You hear what the speaker does with the input signal, that changes with the PW or the PWM but it wont be a rectangel. Couse the top of the rectangel is a stady state signal, and staty state means no movement, and no movement means no sound. The speaker couse its imperfect wobbles around this staty state for some time. If you make the switching fast enough this may look like a rectangle, but it is not, couse if it would be stady state the mic would not pic up anything.
A stady state makes sense in the electronic world, or in the mechanic world but not in the acustic world. No change, no sound.
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Thanks for the reply but that didn't really answer my questions. I don't understand that graph and also the other questions I posed in relation to that in this post:
High-resolution Speaker Frequency Response & Distortion Measurements and Why We Need Them (video)
"..A speaker system or driver is a high-pass filter. That is why they cannot reproduce DC..." Yes but i think its much more easy to understand, that a speaker that not moves cant produce sound. In case of DC on a speaker. Yes. SPL is proportional to acceleration. No acceleration, no SPL.
www.audiosciencereview.com
I'm not trying to be a pain or cause you extra work, I'm just trying to understand, I'm sure there's plenty of people that got stuck at around this point in your video if they were really trying to understand exactly how & what that graph was showing....I think it's probably the most complicated part of your video.
Thanks for the reply but that didn't really answer my questions. I don't understand that graph and also the other questions I posed in relation to that in this post:
and the pic of the graph I don't understand again:High-resolution Speaker Frequency Response & Distortion Measurements and Why We Need Them (video)
"..A speaker system or driver is a high-pass filter. That is why they cannot reproduce DC..." Yes but i think its much more easy to understand, that a speaker that not moves cant produce sound. In case of DC on a speaker. Yes. SPL is proportional to acceleration. No acceleration, no SPL.
View attachment 216515
I'm not trying to be a pain or cause you extra work, I'm just trying to understand, I'm sure there's plenty of people that got stuck at around this point in your video if they were really trying to understand exactly how & what that graph was showing....I think it's probably the most complicated part of your video.
Did you ever heard about the complex plane or complex numbers?
tuga
Major Contributor
The Devil's advocate has emailed the following questions:
Why do we need high-resolution speaker measurements if smoothed frequency response correlates better with human thresholds of audibility?
Why measure speaker harmonic (and intermodulation) distortion if most speakers only distort in the low- and sub-bass and it seems we have very low sensitivity to distortion in that range?
Why all the fuss about Klippel's ability to capture "anechoic" response down to the bottom of the audio range if rooms mess up the bass anyway, and Toole says that we can hear through the room anyway?
Why plot CSDs if "frequency response trumps everything"?
Same with electronics, why plot the measurements' level down to -160dBFS when our hearing ability is limited to some 90dB DR (frequency dependent) and most rooms have the noise-floor above 30-40dB (and all adequately-designed electronics sound the same anyway)?
Why do we need high-resolution speaker measurements if smoothed frequency response correlates better with human thresholds of audibility?
Why measure speaker harmonic (and intermodulation) distortion if most speakers only distort in the low- and sub-bass and it seems we have very low sensitivity to distortion in that range?
Why all the fuss about Klippel's ability to capture "anechoic" response down to the bottom of the audio range if rooms mess up the bass anyway, and Toole says that we can hear through the room anyway?
Why plot CSDs if "frequency response trumps everything"?
Same with electronics, why plot the measurements' level down to -160dBFS when our hearing ability is limited to some 90dB DR (frequency dependent) and most rooms have the noise-floor above 30-40dB (and all adequately-designed electronics sound the same anyway)?
You keep referring about DC and steady states. Where do you see a steady state in this measurments posted earlier by a member?
If a square wave is not an audio signal, what does it do has one of the few main basic wave shape available in the very first analog audio synthesisers?
Where does this well known and documented theory about square waves are coming from if it produce no sound? I am really trying to see what is your point and why you keep reffering to DC, A square wave is not DC, it's alternating DC, It produce movement of a driver, it produce a fundamental and harmonics, in the audio band. Speakers reproduce them if not we wouldn't hear them, not sure how by now you don't seem to get that. (source apple)
Again you reference to the electronic world.
Last time.
We both accept that a speaker that not moves produces no sound?
Lets define t1= max. V reached, t2 begin off swich off
We both accept that a square wave is defined that the voltage off a square wave not varies at the top? Or the voltage between t1 and t2 is a constand. A stady state.
Ergo. cant the speaker broduce sound between t1 and t2 or we not talk about a scuare wave. We can max. talk about something that looks a littel like a square wave.
No. I am talking of what we can hear.
No.
The audible part of a square wave, the one that reside in the audio band, varies at the top. It is band limited, both in digital and analog audio, equipment have a limited frequency response. Even in the electronic world, a square wave does not contain DC, not in the audible band, not in the acoustic world, the only place a square wave is a DC signal is in fundamental mathematics. I am here to talk audio reproduction. I am interested in knowing how well a transducer is able to reproduce the spectrum of a square wave in the audio band.
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Sorry if we dissagre that a speaker that not moves produces no sound. I stop conversation.
I never said I disagree with that... My to Nos where about the underlined statements.
The Devil's advocate has emailed the following questions:
Why do we need high-resolution speaker measurements if smoothed frequency response correlates better with human thresholds of audibility?
Why measure speaker harmonic (and intermodulation) distortion if most speakers only distort in the low- and sub-bass and it seems we have very low sensitivity to distortion in that range?
Why all the fuss about Klippel's ability to capture "anechoic" response down to the bottom of the audio range if rooms mess up the bass anyway, and Toole says that we can hear through the room anyway?
Why plot CSDs if "frequency response trumps everything"?
Same with electronics, why plot the measurements' level down to -160dBFS when our hearing ability is limited to some 90dB DR (frequency dependent) and most rooms have the noise-floor above 30-40dB (and all adequately-designed electronics sound the same anyway)?
Very good questions, why not open a thread for them?
Hi, i'm baffled by your comments that a speaker can't produce square waves....
Do you realize the screen shots I posted of square waves in #55 and were reposted in #109, are microphone measurements of a speakers output?
In term of audio: An oscillating signal that contain all the odd harmonics to the limit of either hearing, or nyquist frequency, or the frequency response of the electronic producing it. In other terms:
dc655321
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Thanks for the reply but that didn't really answer my questions. I don't understand that graph and also the other questions I posed in relation to that in this post:
and the pic of the graph I don't understand again:High-resolution Speaker Frequency Response & Distortion Measurements and Why We Need Them (video)
"..A speaker system or driver is a high-pass filter. That is why they cannot reproduce DC..." Yes but i think its much more easy to understand, that a speaker that not moves cant produce sound. In case of DC on a speaker. Yes. SPL is proportional to acceleration. No acceleration, no SPL.
View attachment 216515
I'm not trying to be a pain or cause you extra work, I'm just trying to understand, I'm sure there's plenty of people that got stuck at around this point in your video if they were really trying to understand exactly how & what that graph was showing....I think it's probably the most complicated part of your video.
Would things click for the negative/positive time components if an analogy were offered to pre/post ringing, as from linear-phase impulse responses?
I think it was part of my algebra in Maths class decades ago, so I can't remember it.
I had a thought though re the interpretation of the graph, and correct me if you know I'm wrong, assuming you understand it:
Is the Harmonic Distortion occuring at minus numbers on the X-axis there because the microphone is capturing the Harmonic Distortion, which is obviously at a higher frequency than the initial excitation frequency that caused that harmonic distortion, and therefore the microphone is capturing a reading at that higher frequency that hasn't been yet played in the chirp - if that's the case then I can understand why the harmonic distortions are listed in that graph in ever decreasing minus values on the x-axis because as you go up the different harmonic distortions from 2nd through 3rd through 4th (etc) then the microphone is picking up a response at those higher frequencies that have yet to be played during the chirp. So the minus time relevance is how much earlier the microphone picks up a significant reading for a given frequency earlier than expected in comparison to the fundamental chirp, and therefore that's how it recognises harmonic distortion from the fundamental chirp.
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