Submitted by **MEGA_AEOIU792** t3_10kgr8r
in **headphones**

## Comments

#
**No_Analysis6187**
t1_j5qus88 wrote

Everything else is snake oil and placebo, obviously.

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**Marathalayan**
t1_j5smw7s wrote

An yet some people believe it is true

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**pinkcunt123**
t1_j5sq7m7 wrote

Well, objectively this is kinda true tho.

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**attlo996**
t1_j5squfo wrote

Why then you own two expensive headphones, since you can EQ them to whatever sound you like? One should do the job.

#
**pinkcunt123**
t1_j5ss4q4 wrote

Closed backs provide noise isolation, open backs don't.

The Noire is more comfortable than the AKG K317 and like it does NOT need EQ, so I bought it. Back when Poweramp did not have PEQ stock tonality was important for my closed backs, which I take with me on business trips and on the train.

I own the 660S and not the 600 or 650, because I got them cheaper. I don't own a DT 880 and use EQ because the DT 880 is uncomfortable for me with its small cup diameter. The 560S was not available back then, otherwise that would have probably been an even cheaper option.

I EQed the 660S to Harman but anything above 10k is basically a free for all and EQing soundstage doesn't work, so obviously the don't sound 100% identical but close enough. Also, the noire has a mid bass hump that adss some punch (I don't get reviewers talking lack of dynamics. Unit variation maybe?). I could EQ that hump away but I don't mind it, gives em some extra character, like the excessive energy above 10k.

There is nothing wrong with buying expensive headphones, just don't expect them to be better because of it. "Detail" and "resolution" being tied to price in a way is awfully suspicious. Also, as long as no one can explain those things properly, thus telling other what to hear for and focus on, they are placebo/bias kicking in.

Soundstage is real and can be a valid purchase criteria if you are into that, but then again, desktop speakers do it better for cheaper, without taking up much room at all.

The Apple dongle has more than enough power for my listening levels, I use my phone with half as much output voltage without any issue. I bought a Fiio K5 for its volume knon, pre out and 6.35mm jack. Cheapest device that ticks all those boxes.

So, any more questions?

#
**attlo996**
t1_j5t9fw7 wrote

Soundstage is real? I don't trust you. It is only your perception of a headphone. I don't see it on graph then it does not exist. It is your brain generating that. Again, your perception. Detail, resolution, layering, soundstage, imaging? BULLSHIT. We trust only in the Holy Harman Target.

#
**AnOldMoth**
t1_j5tmhpk wrote

Nope, not quite. We don't have the ability to measure the minute differences accurately enough to make the amount of extremely small changes to do that.

You also can't really fix resonance peaks with EQ, when you try it messes with something about the sound, though I'm way too sleep-deprived it remember.

Otherwise, if we had good enough measurements, we could actually do what you described with a convolution filter, assuming the headphone had low enough distortion.

#
**audioen**
t1_j5r8ty5 wrote

It is actually the same information as the (complex) frequency response. This is just the time-domain representation of it. More specifically, the Fourier transform of the impulse response is the (complex) frequency response, and the inverse Fourier transform of the (complex) frequency response is the impulse response.

I guess usual smoothed magnitude spectrum has elided phase information, while phase information is in some sense still visible in the impulse response. It is thus the more complete record of the system's behavior. That is why I put the word "complex" in parenthesis above, it means that the ratio of the imaginary and real part of the complex number gives the phase angle. In my opinion, the phase should be processed to group delay plot which shows how much the system delays sound across the response's frequency range. I agree in that I don't think the raw impulse response is easy to read at all.

Group delay is not often an interesting plot with headphones because they are not supposed to have much group delay to begin with. Group delay is more of a property of electronics and digital filters. However, sometimes group delay plots of headphones have big spikes that show that phase is wildly inconsistent at some frequency, and this is often something like a resonating structure in the headset cup. It would also be apparent in frequency response as a narrow spike at that location, but often frequency response plots are heavily smoothed which hides these defects.

As an example, Hifiman Ananda has something wrong in its group delay plot: https://www.audiosciencereview.com/forum/index.php?attachments/hifiman-ananda-group-delay-measurements-open-back-planar-headphone-png.122835/ -- the plot can be quite noisy which probably comes partially from how the headset sits on the fixture and how reflections go inside the headset cup. However, curve fragments going up and down all over, especially in low frequencies above 200 Hz just isn't normal. So this is example of headset with messed up phase that indicates a sound quality problem.

#
**No-Tune-9435**
t1_j5rzc8c wrote

This is an online audio fallacy / myth. Time domain and frequency domain are only equivalent if both are infinite.

I wrote out a longer response in replay to a different comment, but an easy way to see the issue with what you’re starting is to ask how bass response would appear on the above chart (which is completely flat past ~0.002 seconds. One cycle at 200 hz is 0.005 seconds. One cycle at 20 hz is 0.05 seconds! How could you possibly infer anything about the bass response of that unit from that graph? How then can you say that graph is telling us only and exactly what the FR is?

If we want to get real technical, you’d also have to address how certain time domain translations do NOT alter the frequency domain (see shift property of the Fourier transform). That is, time delays do not alter the frequency domain. Relative timing information is very likely lost due to these two effects (representing the freq domain on a finite spectrum and not accounting for time delays).

Please stop propagating this misunderstanding that time and frequency are 100% equivalent

#
**audioen**
t1_j5soc56 wrote

I think you are just wrong, and you do not seem to know what flat frequency response looks like in impulse response graph. It is one sample long singular spike, followed by perfect silence afterwards, forever. You could say that studio monitor speaker systems strive to reproduce just such an impulse, and the closer it is to a very narrow spike, the better the acoustic system. Even this headset looks like it is not that far from perfect impulse apart from some ringing afterwards which suggests it has some resonance peaks and probably highpass filtering because the impulse goes below zero, and that would indicate it cancels some of the sound it produced earlier after a time delay, which is how highpass filters generally work. I don't know, it is really hard to try to read the frequency response off impulse response.

It is completely obvious to me that time delays do not have impact on the magnitude spectrum. They have an effect on the complex spectrum because phase (and group delay) are different and are encoded in the ratio of the imaginary and real parts of the complex number which is usually not shown because phase angle is difficult to relate to anything we actually hear. In that case, group delay would show the added fixed delay just fine, though.

I can only assure you that from mathematical point of view, the impulse response and complex frequency response can be converted to each other without loss. Whether Fourier analysis is good model of human hearing is perhaps a thornier question, as this isn't quite how our ears work, but I think it is still plenty useful as a construct.

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**[deleted]**
t1_j5sqmgp wrote

[removed]

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**SoNic67**
t1_j5tfqxd wrote

Limit the mathematical model only to a couple of terms of the series, and see how "lossless" your transformation is.

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**TaliskerBay22**
t1_j5tvung wrote

I do not understand, your comment can you explain? Only thing I am saying is that the impulse response of the headphone in this graph is known to infinity as the assumption that it will continue to be 0 after the end of the plotted data is certainly a valid one. So we can pad with 0s this graph as much as we want. On the other hand the time 0 in this impulse wrong seems to be arbitrary. If somebody can pass me the data of the graph I can certainly do an FFT and get the Freq response.

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**SoNic67**
t1_j5uqz7w wrote

I am talking about people that say the FFT is equal to impulse response because... math.

FFT implies infinite bandwidth.

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**TaliskerBay22**
t1_j5vuc8d wrote

The maximum frequency of a fast Fourier transform is 1 over the sampling interval or the sampling frequency. As shown in this example https://uk.mathworks.com/help/matlab/ref/fft.html

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**SoNic67**
t1_j5vxaxg wrote

Nope.

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**blutfink**
t1_j5v6ctl wrote

> Time domain and frequency domain are only equivalent if both are infinite.

We’re talking about discrete FTs here. There is nothing infinite about those.

> One cycle at 200 hz is 0.005 seconds.

This is completely irrelevant. You can easily pack a signal that contains components below 200 Hz into a time window much shorter than 1/200 s.

If your understanding of the “myth” is based on this flawed intuition, you may want to reconsider your argument.

#
**No-Tune-9435**
t1_j5vfwi9 wrote

I’m getting bombarded with a lot of “nuh uh”s, your comment included, despite my mathematical explanation above. Would you care to mathematically demonstrate your claim that they are linked?

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**blutfink**
t1_j5vmj53 wrote

What exactly is it you doubt? That the impulse response and the frequency response are linked? This is taught in any undergrad course on the subject. For instance, see this online textbook, last sentence on the page. (“Decayed” here is often named “windowed” in other texts.)

It’s also very easy to convince oneself of the fact. Just fire up MATLAB, Octave, Mathematica, etc. Manipulate a vector to emulate an impulse response, calculate its DFT (typically using the FFT algorithm), then calculate the element-wise magnitude — *et voilà*, that’s the associated frequency response. If you apply the inverse DFT on the output of the DFT (before you calculate the magnitude), you get the original impulse again.

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**No-Tune-9435**
t1_j5vrj9e wrote

Have you done that with a 20-20k FR and looked at the time domain?

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**blutfink**
t1_j5vvd3w wrote

Of course. As in professionally, for decades. Note that for that to work, you need the full, complex-valued FR of a system. The amplitude response does not contain phase information.

That’s why people add the impulse response as supporting information: A plot of the complex-valued response (either 3D or as two real-valued graphs) is not intuitive for humans.

Here is a quick explanation of how the responses are calculated in REW.

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**No-Tune-9435**
t1_j5vwfa2 wrote

Perfect. What ~time gate would that signal have sufficiently decayed? (per your DTFT link)

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**blutfink**
t1_j5vz4wj wrote

Proper decaying/windowing is performed via element-wise multiplication with a window function. All we want is that the signal is smoothly approaching zero at the edges. Since this is approximately the case for typical impulse responses of audio systems, it won’t change the result that much if you don’t window it at all.

To help your intuition, convince yourself that the FT of a centered Gaussian bell curve is itself a Gaussian bell curve. Note that the “low frequencies” in this graph are near the center.

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**No-Tune-9435**
t1_j5vzhjj wrote

You didn’t answer my question at all though. If we want to represent an FR from 20-20khz, what time window would be appropriate for the impulse response? Or have you actually not done this math before?

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**TaliskerBay22**
t1_j5w0hju wrote

Look here is an example, the time domain lasts for a ms and the Fourier response extends from 0 to some tens of KHz. Plug it in Matlab and tweak it as you like

% Define time-domain signal

dt = 0.000001; % time step (s)

t = 0:dt:0.001; % time vector (s)

x = cos(50000.*t-0.0004).*exp(-(t-0.0003).^2/0.00006^2); % time-domain signal

% Perform FFT

X = fft(x); % FFT of time-domain signal

f = (0:length(X)-1)/(dt*length(X)); % frequency vector (Hz)

% Plot results

figure;

subplot(2,1,1);

plot(t,x);

xlabel('Time (s)');

ylabel('Amplitude');

title('Time-domain signal');

subplot(2,1,2);

Amplitude=abs(X);

semilogy(f(1:length(X)/20),Amplitude(1:length(X)/20));

xlabel('Frequency (Hz)');

ylabel('Amplitude');

title('Frequency-domain signal (FFT)');

#
**blutfink**
t1_j5w0fke wrote

In the extreme case? Infinitesimally short time. The FT of a Dirac pulse is a flat, constant response from 0 Hz to infinity Hz (or, in the case of DFT, Nyquist frequency).

As I said, the FT of a Gaussian is a Gaussian. Really sharp in the time domain means really wide in the frequency domain. If you do not understand why that is, your intuition about FTs will be flawed.

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**[deleted]**
t1_j5w2sn4 wrote

[deleted]

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**blutfink**
t1_j5w2dc2 wrote

Maybe this helps: The impulse response in OPs post image basically does not have to be windowed at all, it’s decently close to zero at the edges. Plug it into a FT and you’ll get a decently accurate FR.

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**Physx32**
t1_j5sopan wrote

Can you explain what you mean by "time delays do not alter frequency domain". A delay in time domain causes a phase shift in frequency domain. So it does alter the Fourier transform.

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**SoNic67**
t1_j5tfktx wrote

Only if you have infinite bandwidth.

limit the Fourier series to just a couple of terms (2-3 harmonics) and you will see that's not at all relevant.

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**No-Tune-9435**
t1_j5t87ir wrote

Sure, but like you said, it’s a phase shift in the frequency domain. It doesn’t alter the magnitude of the FR spectrum, just the imaginary vs real distribution. For the purposes of measuring audio equipment, we can pretty safely ignore that distinction you raise, and it doesn’t negate my point that time and frequency domain are not equivalent in the OP.

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**chargedcapacitor**
t1_j5tbga0 wrote

Ha, found the non engineer. You are most certainly wrong. This is a basic math/engineering principle.

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**No-Tune-9435**
t1_j5urhv7 wrote

Not a fan of the snide remark. Would you care to at least substantiate this “basic principle” with a link, or did you just intend to negate my position via name calling?

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**SoNic67**
t1_j5tfsd9 wrote

Ivory tower.

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**SoNic67**
t1_j5tfgwh wrote

I have argued with people about that. They think that theoretical considerations about Fourier transformations are all they need to understand.

They lack the understanding that real systems are not of infinite bandwidth. Especially headphones.

So that response is more important than they think, and it cannot be derived by just looking at 20Hz-30kHz response.

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**blutfink**
t1_j5v8018 wrote

> real systems are not of infinite bandwidth

I don’t see anyone making that assumption here. Could you elaborate?

> that response […] cannot be derived by just looking at [frequency] response.

Of course not from the magnitude response as it is typically plotted. But from the complex-valued frequency response, of course it can, no assumption of infinite bandwidth required.

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**SoNic67**
t1_j5vo2ux wrote

>I don’t see anyone making that assumption here. Could you elaborate?

Every single person that has a fetish about FFT math and assumes it applies in real world 100%.

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**blutfink**
t1_j5vp0v8 wrote

Well, I’m allowed to have that fetish. Digital signal processing was literally my job for decades. Let me tell you that the math and the methods very much apply to the real world. That’s how they were conceived; to analyze the filter properties of physical systems.

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**Rasyad95**
t1_j611nft wrote

Your comment makes me assume that you do not understand the use cases of FT. Do you?

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**NearlyCompressible**
t1_j5rhwo0 wrote

This is a side point, I agree with everything you've said, but the Ananda does not measure like that when you're getting a proper seal.

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**SupOrSalad**
t1_j5rtl56 wrote

Are you claiming that Amir doesn't take extra care to have a proper seal and make sure that variables like pad wear are accounted for? /s

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**audioen**
t1_j5sp8ch wrote

Hmm, okay. I couldn't find any other measurement of this headset, so I will leave it up to the air whether it is representative of the headset, then. Certainly the noisy/spiky character of it should be discarded as a measurement artifact, but the other curve parts going up and down in it might be real. The ideal group delay plot is just a flat line at 0 ms that gradually rises towards the bass due to inevitable high-pass filtering somewhere in the amplifier or such.

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**aknudskov**
t1_j5qy47h wrote

This measures how fast a membrane moves in response to a signal, right?

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**blargh4**
t1_j5r11n4 wrote

Well, moreso how the sound decays. You stimulate the driver with a very short electrical pulse, and see how it responds and how fast different frequencies decay.

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**aknudskov**
t1_j5r1s62 wrote

Yeah that is where I was going with it at first but I guess the first part shows the initial response too. Interesting. I would say this implies to show how fast/responsive a speaker is. I wager a planar magnetic would have a much tighter graph than an hd-600 as an example. Something that sounds "muddy" would have a much longer timeframe shown right ?

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**FastGecko5**
t1_j5r7f2n wrote

Isn't "mud" as we usually talk about it a product of the frequency response? Namely too much boost in the 200-500Hz range

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**aknudskov**
t1_j5r9k9n wrote

I always thought it was the membrane vibrating too long /shrug

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**audioen**
t1_j5srj6c wrote

I answered earlier to OP directly at top-level comment, but I want to answer this one directly. The impulse response is the same thing as the frequency response, it is just time-domain characterization of the system, whereas frequency response is the frequency domain equivalent, though we are usually not shown the complex number nature of the frequency spectrum where phase information of the sound is encoded because we do not hear phase directly and the phase plot doesn't relate to anything we can intuitively understand.

If the speaker membrane moves slowly back to neutral position after an impulse has excited it, that would show up as a decaying plot, and in frequency response would look like a low-pass filter. One way to understand it is to think that system isn't fast enough to reproduce waveform that cycles in and out of phase within that decay region, so if the wavelength is short relative to the impulse's decay time, it cancels with prior versions of itself that are decaying in the impulse, resulting in little output.

In this case, the impulse drops gradually rather than instantly, suggesting that there is some low-pass filtering effect, but also overshoots and goes below zero, which suggest to me that it has could have high pass filtering characteristic, too. For low frequencies, whose wavelength is long relative to the impulse, the negative parts of the impulse subtract from the positive side, and reduce output for low frequency. The fact impulse also returns to the positive side suggests it also contains a resonating component, though. If wavelength is the same as the impulse's ringing around zero level, then it will be amplified by the impulse response.

Finally, if impulse response is ideal, and system's frequency response is perfectly flat, and phase is linear, the entire impulse response is just a single spike with perfect silence surrounding it forever. The ideal impulse indicates that whatever the signal wants to do, the system can reproduce without altering it.

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**SupOrSalad**
t1_j5rc0mt wrote

This is a good article on the topic https://www.soundstagesolo.com/index.php/features/313-is-it-possible-for-headphones-to-sound-fast-or-slow

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**lr_science**
t1_j5sor2a wrote

interesting article, but for some reason it stops right before the point where I would EQ the cheap cans to bring back the frequencies they’re lacking. isn’t it the most obvious next step to do this and see if that makes them sound faster? and then the next question would be: do all headphones sound identical once the FR is equalized between them? my guess would be no, and the next question is what measurements or graphs would explain the remaining differences.

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**Titouan_Charles**
t1_j5sja2p wrote

The membrane will only vibrate at the speed of the signal it is given, so a Planar will replay 1000Hz the same way a dynamic Wil lvibrate 1000Hz. They both play the same source. What we can observe here are the decay, reflections, resonances involved with the headphon'es design (closed back, ported back, good seal/sound leak from the pads, etc) and this affects the sound signature significantly.

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**shaledecimal**
t1_j5r52lc wrote

https://www.stereophile.com/content/innerfidelity-headphone-measurements-explained "You can also think of step response as a measure of frequency response where the leading edge slew rate indicates the high-frequency limit, and the length of time it can keep the step at the new level an indication of its low-frequency limit. At every point between, you can think of the level of the top of the step response as related to the frequency response at the frequency whose quarter wavelength is equal to the elapsed time since the leading edge of the step/square wave."

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**Physx32**
t1_j5smj1t wrote

OP wanted to know about impulse response, not step response.

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**blargh4**
t1_j5qmwp6 wrote

I’ve never had much luck correlating it to any audible quality of a headphone, personally.

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**mvw2**
t1_j5rs3zn wrote

Isn't it nicer to see a full spectrum waterfall plot for decay?

For example, at what frequency is this? Or is this like the aggregate average?

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**SupOrSalad**
t1_j5ru9bq wrote

Waterfall plots are misleading when it comes to headphones, since headphones are mostly minimum phase, the "decay" from a Waterfall plot is directly linked to the FR. You can see this if you EQ the headphone or just shift the headphone on the measurment rig so the FR slightly changes, the "decay" will also change equally with the FR change.

Waterfall plots are designed for speakers and room treatment, and they work for that since it's measuring in different conditions, but for use in headphone measurements they can be more misleading rather than helpful if viewed the same way as waterfall plots for loudspeakers

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**MDZPNMD**
t1_j5vqy4o wrote

Waterfall plots are nice because you find out that the noise of your fan is ruining your measurements.

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**The_D0lph1n**
t1_j5zdrq9 wrote

Is the decay linked to the absolute SPL at a frequency, or to the relative SPL of that frequency relative to the rest? If you EQ down a 6K peak by 3 dB to get rid of that decay trail, but then raise the entire signal level by 3 dB so that 6K is back at the original SPL, does the trail return? Does the rest of the signal now exhibit the same trail?

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**[deleted]**
t1_j5xueev wrote

[deleted]

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**lorentz_217**
t1_j5zfnpl wrote

Well I don’t know much abt this specific graph but an impulse response will contain a lot of frequency components cuz the shorter a pulse gets (hence the impulse) the wider the bandwidth of the signal (you can think of it like a bunch of sine waves at various frequencies adding up to create the pulse, this is just a Fourier series). But yeah that’s the cool science part, I don’t get the utility of the impulse response for headphones either

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**Physx32**
t1_j5smg0x wrote

Impulse function, δ(t) is defined as function whose value is infinite at t=0 and 0 elsewhere. So, Impulse function can be thought of as a signal which has a uniform amplitude of 1 throughout the frequency range. We give this signal as an input to the system (headphones) to find it's response for all the frequencies. The output given by the system is called the impulse response (the plot you posted). Now, if you compute the Fourier transform of the Impulse response, you'll get a complex function. The magnitude of this function is the frequency response of the system. Audiophiles are usually more interested in the frequency response than its time domain counterpart.

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**KuroFafnar**
t1_j5sp1me wrote

Now explain like I’m five

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**audioen**
t1_j5sux24 wrote

No ELI5 for this. Impulse response is the convolution that system applies to its input to produce its output. Convolution is a description of how prior data seen by the system alters the signal that is being produced right now. It is computed as integral of the input at past points multiplied with convolution function's value at corresponding time point. In sampled digital systems, both the impulse response and the signal are arrays of numbers, and you approximate the integral by placing the impulse response and input signal side by side, and you multiply input and impulse together at their corresponding positions, sum all the values together, and write the sum as the output sample. Then you move convolution function forwards along the input by one sample, and redo this massive calculation again to get the next output sample, and so on, and this is the convolution of the input with the impulse response.

As an example, guitarists may have amplifier cabinet simulators which are the sampled impulse responses of the speaker in the cabinet, which is usually open in the back, or possibly they have sampled room impulse responses and their effect processor actually convolves their playing with these impulses -- possibly multiple seconds long -- in real time to mimic the sound of these amplifiers and rooms. They actually perform the equivalent of the convolution computation in frequency domain because of the massive amount of multiplications that are needed for long impulses in time domain.

The ideal impulse response is infinitely narrow spike, because it means that the input at that one position is the only thing that affects the output and the convolution's output is thus the same as the input.

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**Physx32**
t1_j5t2u64 wrote

Impulse function is a signal that contains information for all frequencies from 0 to infinity. Since it contains this information, we give this as input to the headphone. The headphone then gives an output (which the OP posted). Now, we transform this output from time domain to frequency domain to see the frequency response of the headphone.

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**notomnis**
t1_j5rbntl wrote

Being put on the spot like this made me completely forget how to read a graph.

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**[deleted]**
t1_j5t86dc wrote

[deleted]

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**TheHelpfulDad**
t1_j5txjgs wrote

What is the y-axis?

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**MEGA_AEOIU792**
OP
t1_j5ubz9r wrote

I was asking the same question as you

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**xMoRioPLx**
t1_j5u8or9 wrote

I understand it as delay made by reference sound bouncing back to input from some kind of wall.

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**xMoRioPLx**
t1_j5u90sx wrote

Oh and tbh no computer or anything can create sound instantly and make it instantly disappear so there is always some imperfections.

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**TaliskerBay22**
t1_j5uys2g wrote

This does not exactly matter as long as the instrument that creates the impulse and the instrument that measures the impulse response are much faster than the system that is under test. In this case a function generator can easily go to 100 KHz so it is much faster than the headphone, also the microphone that measures the response is much faster than the headphone so you are ok.

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**Xypton**
t1_j5xud4i wrote

From the perspectives of control systems and signal processing: Impulse/step response tells you the rise time, settling time, damping and natural frequency of a system

From the perspective of just headphones: not much

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**RubenRag**
t1_j5raprf wrote

It’s a visual representation of the impulse response

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**gonomon**
t1_j5slcsl wrote

This is movement of the magnet of the headphone when you give it an impulse signal. Normally if everything were perfect you should be seeing a straight vertical line, but it takes time for magnet to return to its original position. And this graph shows that and with how many oscillations it takes to get there.

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**KuroFafnar**
t1_j5qvsqs wrote

I think it is an indirect measure of distortion / harmonics. If a sound / impulse lingers it is like echoes.

Happy if somebody more knowledgeable can explain it better

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**Physx32**
t1_j5snmcw wrote

No, impulse response is the time domain counterpart of frequency response. They both convey the same information but in different domains. See my comment in the thread for a more detailed explanation.

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**pinkcunt123**
t1_j5qk7fg wrote

It shows you how damped the driver is. Cool too look at, but irrelevant.

The only two relevant measurements for headphones are frequency response and distortion.

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**No-Tune-9435**
t1_j5qrqfg wrote

Has this claim actually been tested? This sounds a lot like conjecture

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**AtomikPi**
t1_j5r5hbl wrote

The reason why is that headphones are nearly entirely minimum phase. This means that phase domain and time domain are directly related, and the above impulse response is implied by the frequency response. (This isn’t true of room issues with speakers by the way.) This is not entirely the case at low frequencies and with certain weird designs. There is a Crinacle video and some Oratory1990 posts explaining this better that I can’t find right now. Fwiw I subjectively sometimes think impulse response corresponds with my sense of speed and decay (stats sound dry/ fast decay), but don’t claim that is objectively the case.

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**No-Tune-9435**
t1_j5rety5 wrote

Two quick fallacies to call out here A) the one everyone misses: time domain = frequency domain if you have infinite time and infinite frequencies. But if I showed you a frequency response of a song on only 1-40,000hz (or whatever, limits exaggerated to make the point), there would be infinite different songs that have that exact same FR. Simplest way to understand this is imagine if I played a song in reverse. It’d have the same FR from 1-40khz. Imagine I took the first second of a song and moved it to the end of that song. Also same FR. You could absolutely take the Fourier transform of a song and convert it to frequency domain. But you’d need to go into the microhertz to fully represent it in FR. People like to cite the frequency & time domain equivalencies to say that the time domain doesn’t matter at all just because you have an FR graph that goes from 20hz-20khz. This concept gets misquoted and abused in lots of arguments about interpreting FR plots. It doesn’t conclude what people want it to, and if people want to claim time domain plots are 100% irrelevant, the onus is still on them to demonstrate this with controlled studies. We can do some fancier math to put some constraints around my argument, but the original point needs to be made that time and frequency domains are only equivalent if both are infinite. Source: I am a mathematician who studied signal processing

B) Nobody said anything about subjective listening or hearing impulse response. I know you cite it as your own experience, but that feels a bit like a straw man argument. Original post made a conclusive assertion that impulse response is irrelevant

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**AtomikPi**
t1_j5rhl45 wrote

Hi there. I don’t claim to be an expert on this branch of math and will believe you here. I would be curious to hear from someone like Oratory on the topic who has made similar arguments.

I’m only offering my subjective experience since I often find it hard to reconcile the subjective and objective side of things in audio. In the case of electronics and certainly cables, I’d rather largely ignore my subjective experience (also given the tons of failed ABT with speakers, which make it hard to believe electronics make any audible difference if not faulty). As you’ve stated, there have not been any time domain AB tests to my knowledge so the answer is ??? and people are trying to reason from the math rather than trials.

- also soundstage (partially not driven by FR) and FR smoothness (important in Harman’s experiments) are two reasons to consider higher end headphones.

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**Titouan_Charles**
t1_j5sjnk1 wrote

This is legit on paper, thanks to your Fourier transform but it doesn't help with actually understanding headphones here. The graph we see helps us seeing if something is wrong with the resonances/reflections inside the earcup, or if the open/closed back design manages what it intended to. No need to expand into theoretical Fourier transforms of "whole song FR" and Hz ranges outside of human hearing.

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**No-Tune-9435**
t1_j5t7nd7 wrote

I don’t understand your comment that there is no need to get into the math. Do you realize the original comment was misunderstanding the exact math I was talking about?

To make sure you’ve read my post… I’m not saying the math tells us anything. I replied to a comment that claims that impulse response is exactly the same as frequency response, and therefore we could safely ignore impulse response graphs. The math says that isn’t true. Therefore I claim we need controlled studies to understand what an impulse response can or cannot tell us before anyone knows what to interpret from them.

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**pinkcunt123**
t1_j5so9dk wrote

Well, if damping is odd and affects sound you will see that in the frequency response graph. Therefore, just skip the damping measurement.

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**Physx32**
t1_j5snhcl wrote

Clearly you have no clue about signal processing. The frequency response (which is one of the only relevant measurement according to you) is actually computed from the impulse response by applying Fourier transform and selecting its magnitude.

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**pinkcunt123**
t1_j5soopz wrote

Oh, ok. You clearly have no idea how headphones work.

All you need to see is the frequency response. Over damping and under damping might affect it, but why even bother looking at it, when the result i.e. changed FR can be measured directly?

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**Physx32**
t1_j5sp1uf wrote

Omegalol. There's no way of measuring FR "directly". See my comment in the main thread to learn how frequency response is computed and what Impulse response actually means. FR is calculated from impulse response only.

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**pinkcunt123**
t1_j5sr1zx wrote

Again...

Why brother looking ag impulse response rather than the result????

Are you visually impaired or just unable to understand english?

attlo996t1_j5qljos wroteIn this subreddit everything is irrelevant as long as you use an Apple Dongle and you can EQ...