The frequency response and the impulse response are correlated. You can just get one from the other through fourier transformation. Therefore it's perfectly fine to measure whatever is easier to measure, which is frequency response through sine waves for audio equipment. Phase shift can also be measured through this approach (and is required for calculating impulse reponse).
Sure, nonlinear distortions are not covered and also sure that the measurements won't cover everything which will influence the perceived sound. But nevertheless frequency response measurements are a very solid tool which allows objective comparions between various systems.
In theory, theory and practice are the same. In practice, however...
Yes, mathematically, you can get impulse response via Fourier transform. In practice, you're using a current to push a coil against a magnet into a spring. A real current, a real magnet, a real spring. You get all sorts of non-linear behaviors. On top of that, the spring (that is, the driver) retains impulse energy as mechanical energy, and then emits it back in a nonlinear way, by moving that coil in the magnetic field - turning a motor into a generator, producing seriously nonlinear signal back into the amplifier that produced the original signal, where it's picked up by the distortion-correcting negative feedback loop, and...
Yeah, that's not a simple Fourier transform anymore. If you had enough data points, it could be modeled mathematically. We don't have all the data, and don't yet have a good way to measure all that stuff.
That's really valid only in the case when the frequency response and the impulse response completely characterize a system--that is, only for linear and time-invariant systems. I know you covered this base by mentioning nonlinear distortions, but I think that's the big issue. Otherwise you could just correct the frequency response with equalization (assuming the frequency response isn't literally 0 ever).
For example, consider a system that just squares or cubes its input. The unit impulse response will look the same in both cases. The frequency response isn't even defined, since the steady state response to a sinusoid is not itself a sinusoid.
Came here to respond with something similar to the first paragraph. I was wondering if perhaps audio engineers mean something different when they're talking about impulse response, since my expertise is in signal processing for electromagnetic signals (vs. audio).
Yeah, I think we do. When audio engineers talk about impulse response, we're really talking about how accurately components can reproduce sharp-attack sounds. Snare drums, acoustic guitar strings, some synth sounds - these are examples of fast impulse response.
Microphones really struggle with this. So do mic preamps, recording media, amplifiers, and especially speakers. Recorded instruments sound wildly different from live-in-a-room instruments.
edit: I should add here that fast impulse response isn't necessarily what we want. Consider the common use of the relatively sluggish Shure SM57 on snare drums, rather than the much faster response of a small diaphragm condenser. The condenser usually just sounds harsh. The SM57 smooths out the sound. This is good, because nobody in their right mind actually sticks their ear one inch from a snare drum.
No, you're not talking about different things. You're talking about exactly the same thing in two domains: a signal which is transduced from air pressure to voltage in real time, and then later goes back in the other direction. Audio engineers should know that, and many of them do.
"Recorded instruments sound wildly different from live-in-a-room instruments." Yes, they do.
In the best case, you have two nearly perfectly accurate transducers located inside the not-quite-ear-canals of a dummy head doing the recording, and you play those two signals back to transducers located similarly in the head of your listener. This is the best case for accurate reproduction of instruments. You can get a recording that sounds nearly the same as being in the room where it happened.
In the most common "purist" method, recording engineers are using two or three mics arrayed in free space somewhere near where the audience would be, and hoping to capture the sensation of being in the room. This is problematic because the instruments are complex, moving three-dimensional shapes producing sound in a field which interacts with all sorts of things before being sampled by those 2-3 points. That throws away nearly all the available information. Then playback emanates from two not-really-point sources in a completely different room.
In the normal non-purist methods, engineers close-mike each of the instruments, take direct input from some electronic instruments, and may or may not try to capture some room ambience. Then they process everything to the point where they are as much a performer as the recorded players, and work on it over and over again until (hopefully) everyone is happy with it as a work of art. There should be no pretense that this is going to get you an accurate rendition of the feeling of being there, though.
Good points. Purist recording methods are almost unused in popular music; they're for classical or solo instruments only. There are few rational musical situations where an acoustic guitar is as loud as a drum kit, but we hear it all the time on records. Likewise, sounds aren't panned hard left or hard right, we don't record in giant churches, etc. Not to mention multiple overdubs by the same performer...
I don't really care if a recording is accurate. I care if it's enjoyable.
When I think of "the feeling of being there", I don't think of the feeling of being in the room with the musicians. I think of the feeling of what you're doing when you experience that music - where does it take you? I'll never be able to separate Pink Floyd's The Wall from making out with the girlfriend who introduced me to Pink Floyd, for example. Music has strong sense-memory, almost as strong as smell. That's what I want to engage - I want to make music you remember.
It's sometimes amazing to hear the albums which didn't have all that processing work done at the end. I love the sound of the DMB Lillywhite sessions which was leaked after recoding but not turned into an official album. I much prefer those versions to the later official ones.
> The frequency response and the impulse response are correlated. You can just get one from the other through fourier transformation.
It's true. It should be added, however, that when the frequency response is complex (and it always is with transducers) and the measurement conditions are not very repeatable, it's probably beneficial to also get an impulse response.
> Phase shift can also be measured through this approach
I thought headphones are zero-phase-shift devices? This is because of the very small volume of air between the membrane and the eardrum - they both move in lockstep, hence zero phase shift.
Human beings are remarkably good at compensating for deviations in frequency response. You can very quickly acclimatise to deviations much greater than +-10dB. The main shortcoming of mid-priced headphones is transient response, which we are acutely sensitive to. Heavy, flexible and poorly-damped drivers in highly resonant enclosures might look reasonably accurate in the frequency domain, but they have absolutely dismal real-world performance.
Audio engineers routinely choose transient response over frequency response, hence the popularity of Yamaha NS10 and Auratone 5C monitors in professional environments.
Sure, nonlinear distortions are not covered and also sure that the measurements won't cover everything which will influence the perceived sound. But nevertheless frequency response measurements are a very solid tool which allows objective comparions between various systems.