Impulse Response:

https://dsp.stackexchange.com/questions/536/what-is-meant-by-a-systems-impulse-response-and-frequency-
response

The impulse that is referred to in the term impulse response is generally a short-duration time-domain signal. For
continuous-time systems, this is the Dirac delta function δ(t)δ(t), while for discrete-time systems, the Kronecker delta
function δ[n]δ[n] is typically used. A system's impulse response (often annotated as h(t)h(t) for continuous-time
systems or h[n]h[n] for discrete-time systems) is defined as the output signal that results when an impulse is applied
to the system input.

Why is this useful? It allows us to predict what the system's output will look like in the time domain. Remember the
linearity and time-invariance properties mentioned above? If we can decompose the system's input signal into a sum
of a bunch of components, then the output is equal to the sum of the system outputs for each of those components.
What if we could decompose our input signal into a sum of scaled and time-shifted impulses? Then, the output would
be equal to the sum of copies of the impulse response, scaled and time-shifted in the same way.

For discrete-time systems, this is possible, because you can write any signal x[n]x[n] as a sum of scaled and time-
shifted Kronecker delta functions:

Each term in the sum is an impulse scaled by the value of x[n]x[n] at that time instant. What would we get if we
passed x[n]x[n] through an LTI system to yield y[n]y[n]? Simple: each scaled and time-delayed impulse that we put
in yields a scaled and time-delayed copy of the impulse response at the output. That is:

where h[n]h[n] is the system's impulse response. The above equation is the convolution theorem for discrete-time LTI
systems. That is, for any signal x[n]x[n] that is input to an LTI system, the system's output y[n]y[n] is equal to the
discrete convolution of the input signal and the system's impulse response.
For continuous-time systems, the above straightforward decomposition isn't possible in a strict mathematical sense
(the Dirac delta has zero width and infinite height), but at an engineering level, it's an approximate, intuitive way of
looking at the problem. A similar convolution theorem holds for these systems:

where, again, h(t)h(t) is the system's impulse response. There are a number of ways of deriving this relationship (I
think you could make a similar argument as above by claiming that Dirac delta functions at all time shifts make up an
orthogonal basis for the L2L2 Hilbert space, noting that you can use the delta function's sifting property to project any
function in L2L2 onto that basis, therefore allowing you to express system outputs in terms of the outputs associated
with the basis (i.e. time-shifted impulse responses), but I'm not a licensed mathematician, so I'll leave that aside).
One method that relies only upon the aforementioned LTI system properties is shown here.
In summary: For both discrete- and continuous-time systems, the impulse response is useful because it allows us to
calculate the output of these systems for any input signal; the output is simply the input signal convolved with the
impulse response function.

Frequency response:
An LTI system's frequency response provides a similar function: it allows you to calculate the effect that a system will
have on an input signal, except those effects are illustrated in the frequency domain. Recall the definition of the Fourier
transform:

In essence, this relation tells us that any time-domain signal x(t)x(t) can be broken up into a linear combination of
many complex exponential functions at varying frequencies (there is an analogous relationship for discrete-time
signals called the discrete-time Fourier transform; I only treat the continuous-time case below for simplicity). For a
time-domain signal x(t)x(t), the Fourier transform yields a corresponding function X(f)X(f) that specifies, for each
frequency ff, the scaling factor to apply to the complex exponential at frequency ff in the aforementioned linear
combination. These scaling factors are, in general, complex numbers. One way of looking at complex numbers is in
amplitude/phase format, that is:

Looking at it this way, then, x(t)x(t) can be written as a linear combination of many complex exponential functions,
each scaled in amplitude by the function A(f)A(f) and shifted in phase by the function ϕ(f)ϕ(f). This lines up well
with the LTI system properties that we discussed previously; if we can decompose our input signal x(t)x(t) into a
linear combination of a bunch of complex exponential functions, then we can write the output of the system as the
same linear combination of the system response to those complex exponential functions.
Here's where it gets better: exponential functions are the eigenfunctions of linear time-invariant systems. The idea is,
similar to eigenvectors in linear algebra, if you put an exponential function into an LTI system, you get the same
exponential function out, scaled by a (generally complex) value. This has the effect of changing the amplitude and
phase of the exponential function that you put in.
This is immensely useful when combined with the Fourier-transform-based decomposition discussed above. As we
said before, we can write any signal x(t)x(t) as a linear combination of many complex exponential functions at
varying frequencies. If we pass x(t)x(t) into an LTI system, then (because those exponentials are eigenfunctions of
the system), the output contains complex exponentials at the same frequencies, only scaled in amplitude and shifted
in phase. These effects on the exponentials' amplitudes and phases, as a function of frequency, is the
system's frequency response. That is, for an input signal with Fourier transform X(f)X(f) passed into system HH to
yield an output with a Fourier transform Y(f)Y(f),

In summary: So, if we know a system's frequency response H(f)H(f) and the Fourier transform of the signal that we
put into it X(f)X(f), then it is straightforward to calculate the Fourier transform of the system's output; it is merely the
product of the frequency response and the input signal's transform. For each complex exponential frequency that is
present in the spectrum X(f)X(f), the system has the effect of scaling that exponential in amplitude by A(f)A(f) and
shifting the exponential in phase by ϕ(f)ϕ(f) radians.

Bringing them together:

So, given either a system's impulse response or its frequency response, you can calculate the other. Either one is
sufficient to fully characterize the behavior of the system; the impulse response is useful when operating in the time
domain and the frequency response is useful when analyzing behavior in the frequency domain.