After the estimation of channel impulse response (using the training sequence in burst), to cancel the effect of channel on the received signal, I think, we should convolve the received signal with the inverse of channel impulse response.

But,in GSM, it convolves the received signal with the conjugate of channel impulse response and calls this process as "matched filtering". Could anyone help me with this?

  • $\begingroup$ have you done the math? The formula for discrete convolution isn't that complicated. Make sure to use complex numbers! Add your calculation to your question; that will give us something precise to react to; otherwise, this question might be a little too broad. $\endgroup$ Oct 15 '16 at 11:03
  • $\begingroup$ I dont know why in GSM, the received signal (r) is convolved with the conjugate of h (h= impulse response of channel) instead of inverse of h? in other words, I think, in matched filtering it should be: y=r* inv(h) , instead of y=r*conj(h) $\endgroup$
    – Sarah
    Oct 15 '16 at 11:43
  • $\begingroup$ Write down the formula that explains why you do matched filtering, and expand the convolution operator $*$ to the the sum it actually is. $\endgroup$ Oct 15 '16 at 12:01
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    $\begingroup$ I have to think what they are calling the "Channel Impulse Response" is really the channel compensation to minimize error, as done with a MMSE type equalizer, for example. Further, you would not want to convolve with the inverse of the actual channel response as that will cause noise enhancement wherever any frequency nulls occur (such as in a frequency selective fading channel). $\endgroup$ Oct 16 '16 at 11:55

This answer is based on the contents of "pages 8 and 9 of file below: "GSM Channel Equalization, Decoding, and SOVA on the MSC8126 Viterbi Coprocessor (VCOP)" and not upon the OP's interpretation of what has been said on those pages or the incorrect naming of various mathematical operations therein.

Consider the cross-correlation function $$R_{u,v}(t) = \int_{-\infty}^\infty u(\tau+t)v(\tau)\,\mathrm d\tau \tag{1}$$ of finite-energy signals $u(t)$ and $v(t)$. Its Fourier transform is \begin{align} S_{u,v}(f) &= \int_{-\infty}^\infty R_{u,v}(t)\exp(-j2\pi ft) \,\mathrm dt\\ &= \int_{-\infty}^\infty \left[\int_{-\infty}^\infty u(\tau+t)v(\tau)\,\mathrm d\tau\right]\exp(-j2\pi ft)\,\mathrm dt\\ &= \int_{-\infty}^\infty \left[\int_{-\infty}^\infty u(\tau+t)\exp(-j2\pi ft)\,\mathrm dt\right]v(\tau)\,\mathrm d\tau\\ &= \int_{-\infty}^\infty \left[\int_{-\infty}^\infty u(\lambda)\exp(-j2\pi f(\lambda-\tau))\,\mathrm d\lambda \right]v(\tau)\,\mathrm d\tau\\ &= \int_{-\infty}^\infty \left[\int_{-\infty}^\infty u(\lambda)\exp(-j2\pi f\lambda)\,\mathrm d\lambda \right] v(\tau)\exp(j2\pi f\tau)\,\mathrm d\tau\\ &= \int_{-\infty}^\infty U(f)v(\tau)\exp(-j2\pi f\tau)\,\mathrm d\tau\\ &= U(f)V^*(f) \tag{2}. \end{align} Now, we apply this result to the GSM system under consideration in which a known training signal $x(t)$ is transmitted across the channel which is modeled as an LTI system whose impulse response $h(t)$ is unknown. We wish to estimate $h(t)$ from knowledge of the transmitted training signal $x(t)$ (which we have chosen very carefully to have various desirable properties) and the corresponding channel output $y(t) = (x \star h)_t$ whose Fourier transform $Y(f)$ equals $X(f)H(f)$.

Since $x(t)$ is known to the receiver, it can generate a replica of $x(t)$ at the receiver. Now suppose that we compute the cross-correlation function $R_{y,x}(t)$ of the received signal $y(t)$ and the local replica $x(t)$. The Fourier transform of this cross-correlation is $$\mathcal F\{R_{y,x}(t)\} = Y(f)X^{*}(f) = X(f)H(f)X^{*}(f) = |X(f)|^2 H(f)\tag{3}$$ which shows that $$R_{y,x} = R_{x,x}\star h\tag{4}$$ where $R_{x,x}$ is the autocorrelation function of the signal $x(t)$. (This is Equation (4) on page 8 of the document cited in the first sentence of thus answer.

If the autocorrelation function $R_{x,x}(t)$ is a Dirac delta or impulse $\delta(t)$, then $(4)$ shows that the cross-correlation function $R_{y,x}(t)$ that we have just computed is just $h(t)$, the channel impulse response that we are trying to estimate! Of course, no deterministic signal can have $\delta(t)$ as its autocorrelation function, but there do exist signals whose autocorrelation function resembles the proverbial "inverted thumbtack" function: a large very narrow spike at $t=0$ and very small (close to $0$) values for $t \neq 0$. Binary Barker sequences are one such class, but since the longest known Barker sequence is of length $13$, lots of people have expended lots of computer time searching for longer binary sequences whose autocorrelation functions look like inverted thumbtacks. (If arbitrary amplitude levels are permissible, then Huffman's impulse-equivalent sequences can be considered. Using such a sequence (actually, the corresponding pulse train) for $x(t)$ leads to $$h(t) \approx K\cdot R_{y,x}(t) \tag{5}$$ where $K$ is a constant whose value can be determined once we have chosen $x(t)$.

Finally, we come to the computation of the cross-correlation function $R_{y,x}(t)$. Set $\hat{x}(t) = x(-t)$ and note that we can write \begin{align} R_{y,x}(t) &= \int_{-\infty}^\infty y(\tau+t)x(\tau)\,\mathrm d\tau\\ &= \int_{-\infty}^\infty y(\lambda)x(\lambda-t)\,\mathrm d\lambda\\ &= \int_{-\infty}^\infty y(\lambda)\hat{x}(t-\lambda)\,\mathrm d\lambda\\ &= \left(y\star \hat{x}\right)_t \end{align} that is,

we can compute the desired cross-correlation $R_{y,x}(t)$ by filtering the received signal $y(t)$ through a filter whose impulse response is $\hat{x}(t) = x(-t)$.

But, a filter whose impulse response is $x(-t)$ is what I have called in this answer as the matched filter for $x(t)$ and so

we can compute the desired cross-correlation $R_{y,x}(t)$ by filtering the received signal $y(t)$ through **the* matched filter for $x(t)$.

This is what the document cited above says (just above Equation 4):

The received training sequence in the digital domain $\ldots$ is fed into a digital matched filter $\ldots$ with an impulse response that is matched to $\ldots$ (the transmitted sequence)

It is to be hoped that this will clarify the confused discussion in the comments on the main question as well as on my other answer between the OP, @DanBoschen and myself.

  • $\begingroup$ This answer is correct. Thank you very much. Of course I think we should say: "we can compute the desired cross-correlation by filtering the received signal y(t)through the conjugate of matched filter for x(t)" $\endgroup$
    – Sarah
    Oct 20 '16 at 17:06
  • $\begingroup$ This answer is correct. Thank you very much. Of course I think we should say: "we can compute the desired cross-correlation by filtering the received signal y(t) through the conjugate of matched filter for x(t)". Because in defenition of cross correlation R(x,y), there should be a conjugate mark on one of the signals. Is this true? $\endgroup$
    – Sarah
    Oct 20 '16 at 17:10

First, please read this answer of mine for a detailed description of matched filters for real-valued signals. In particular, note that what I called the matched filter for a signal $x(t)$ is a(n LTI) filter with impulse response $h(t) = x(-t)$ which is better described as the time-reversed signal rather than the "inverse" of the impulse response as you call it: to many people, inverse would suggest $[x(t)]^{-1}$ or $-x(t)$ depending on whether they were thinking of multiplicative or additive inverse.. Note also that convolving the channel output with the time-reversed channel impulse response does not "cancel" the effect of the channel on the signal at the channel input. In general, matched filtering distorts the signal being filtered tremendously because the goal of matched filtering is to produce a peak response at the desired sampling time. This is quite different from the audio enthusiast's desire of reproducing the input signal with the highest possible fidelity, and thus using an "inverse" filter to remove all traces of the distortion caused by the channel filter. See the last part of my answer cited above for some pictorial illustrations of how matched filtering distorts the signal so as create a peak response at the sampling instant.

Second, there are no complex signal in real life; they exist only in the diseased imaginations of DSP engineers. What we have in real life is a real-valued bandpass signal $$x(t) = x_i(t)\cos(2\pi f_0 t) - x_q(t)\sin(2\pi f_0 t) = \Re\left(x_c(t) e^{j2\pi f_0t}\right)$$ where $x_i(t)$ and $x_q(t)$ are slowly varying signals compared to the carrier signal of frequency $f_0$ Hz, and $x_c(t) = x_i(t) + jx_q(t)$ is called the complex baseband equivalent of the RF signal $x(t)$. The filter matched to $x(t)$ thus has impulse response \begin{align} x(-t) &= x_i(-t)\cos(-2\pi f_0 t) - x_q(-t)\sin(-2\pi f_0 t)\\ &= x_i(-t)\cos(2\pi f_0 t) - [-x_q(-t)]\sin(2\pi f_0 t)\\ &= \Re\left([x_i(-t) -j x_q(-t)]e^{j2\pi f_0t}\right)\\ &= \Re\left([x_c(-t)]^* e^{j2\pi f_0t}\right) \end{align} The complex baseband equivalent of the matched filter impulse response is thus $[x_c(-t)]^*$ and thus when the DSP engineer proceeds to match-filter the complex baseband equivalent input $x_c(t)$, she must convolve $x_c(t)$ not with $x_c(-t)$ but rather with $[x_c(-t)]^*$, cos doing the former instead of the latter would result in her being in a state of sin.

  • $\begingroup$ cot of the previous answer was quite impor tan :-) $\endgroup$
    – msm
    Oct 16 '16 at 4:00
  • $\begingroup$ Dilip, Matched filtering is done with the time-reversed signal as you describe, but the poster is asking specifically about cancelling the effects of channel distortion. Does it make sense to you that convolving the signal with the conjugate of the channel impulse response would be called matched filtering? $\endgroup$ Oct 16 '16 at 11:51
  • $\begingroup$ Also as I believe we discussed before, there are no complex signals in real life just as much as there are no real-valued bandpass signals in real life: both are just mathematical representations- I can use a one channel oscilloscope to measure a sinusoidal voltage on a single wire that represents a real signal, but I can also use a two channeled scope probe to measure the I and Q voltages on two wires that represent a complex signal. Both seem equally "real" or in my "imagination" just the same; I don't see why one is more real or not just because I need two reals to represent the complex? $\endgroup$ Oct 16 '16 at 11:52
  • $\begingroup$ @DanBoschen In the frequency domain, channel input $X(f)$ gets transformed into channel output $X(f)H(f)$. So I don't see how filtering the channel output through a filter with transfer function $[H(f)]^*$ or $[H(f)]^{-1}$ or $[[H(f)]^*]^{-1}$ is matched filtering in any sense of the phrase with which I am familiar. The closest is $[H(f)]^{-1}$ which gives us back $X(f)$ at the receiver (and futrther matched filtering would be required) but this, as you correctly state in your comment on the question, plays havoc with the noise. $\endgroup$ Oct 16 '16 at 14:31
  • $\begingroup$ I agree with you. I was just commenting that his question was not really how to perform matched filtering but why his source called the equalization process matched filtering. Matched filtering would never be done with the channel response, it would be done with the time-reversed signal itself. $\endgroup$ Oct 16 '16 at 14:36

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