CONVERSION OF DISCRETE TIME SIGNAL INTO FREQUENCY DOMAIN: We have 3 options: DTFT: it applies on aperiodic signal and will

produce continuous output (Inverse will be done using integration). DFT: it applies on aperiodic signal and produces discrete frequency signal. DFS: it applies on periodic signal and produces discrete frequency signal Important: when one domain is continuous then result in other domain will be aperiodic and vice versa. Similarly when one domain is discrete then result in other domain will be periodic and vice versa. Cross correlation: When it applies two different signal (more dissimilar signals) produces greater output than those of output which are resulted when it applies to more similar signals. This is why cross correlation is normalized. Antialiasing filter: it is analog filter and normally low pass. It is miss conception that we can not use band pass instead of low pass actually we can use it but then we have to use band pass sampling techniques those are rarely used. Sampling of Sine: it is mandatory to sample the sine signal greater than Nyquist rate because other wise samples will be taken only on zero crossings thus results nothing. Important : the spectrum between fs/2 to fs/2 has the real information all others are periodic extention in the spectrum of digital signal. There are two method of performing FFT decimation in time and decimation in frequency .,the word decimation refers to the fact that there will be separation of odd and even bins. Conversion : from foureir transform to laplace transform just by replacing jw to s Conversion : z transform of any signal x is basically fourier transform of r^-n times the x signal. Pole and zeros in Z plane : poles are those point where z plane diverges and zeros are those point where z plane converges. ROC for infinite duration signal: Region of convergence for causal signal is exterior of the circle and for the anticausal signa is an interior of circle. ROC for finite duration of signal: ROC is everywhere poles. Stabliity of Digital signals: stable if poles of transfers function are inside the unit circle and unstable if outside the circle and marginally stable if lies on the circle.

One sided z transform is for the causal signal. Increasing the sampling rate:up sampling +interpolation (L*fs) Decreasing the sampling rate: Antialiasing filteration+downsampling/decimation. (fs/M) Adaptive Filter: it is the filter with time variant coefficient its well known applications are system identification and noise cancellation. Filters: Bands: there are three types of bands found in filters 1) pass band 2) stop band 3) transition band. Types: band pass, low pass, high pass, band stop/elimination Important: Must be aware of their corresponding amplitude and phase responses of each type. In analog we have two types of filter approximation Butterworth, (in respect with no ripple in pass band), chebyshev (ripples in pass band), In digital we have two types : Fir (time limited impulse response) also called all zero or non-recursive filters.(no feedback). Its phase response is linear and simple design. Gibbs phenomenon: the truncation of time domain signal results in the band un-limitedness of band (frequency) which results as oscillatory in magnitude response. The ripples are due to abrupt change in impulse response. We can decrease these ripples by producing the transition in impulse response. Method of windows : I) II) III) IV) V) Consider the ideal magnitude response Take idft of the magnitude response to convert into impulse response. Sample the impulse response Truncate the impulse response using window Take DTFT to finde magnituderesponse

Method of Frequency Sampling: 1) 2) 3) 4) Consider the ideal magnitude response Sampe the magnitude response Take inverse DFT to convert into impuse response which will be discrete and periodic. Take DTFT to find magnitude response.

IIR (Time unlimited impulse response) also called recursive filters.(feedback).

Thought it is complex but requires fewer coefficients as compared to FIR filters to design same filters. So these are fasters from input to the ouput and they are relatively used in real time applications with high sampling rate. Four important methods of IIR filter designing; I)Impuls variance method ii) bilinear transformation iii) matched z transform iv) approximation of derivatives Impulse variant methods: first we take transfer function in analog then we apply inverse laplace transform to find impulse response then we sample it to form discrete impulse response then we take z transform to form digital transfer function of digital filter. Issues: we need to sample at the higher rate otherwise it will not accurate according to approximaions and very low sampling rate can results in aliasing in frequency domain. They are suitable for only band pass but they are stable. Approximation of derivatives: in this we map left side of s-plane into the circle at centre(0.5,0) and radius 0.5. thus not suitable for the higher frequencies. Or high pass filters. Billinear transformation: It Is nonlinear mapping of s plane into the z plane . we mapped all of the left side of s plane into the unit circle with centre(0,0) thus provides the approximation any angular frequencies or provides the way designing any type of filters. Minimum phase filters: if all zeros of lies in left side of plane for analog: if all zeros lies inside the unit circle for digital: