Xử lý tín hiệu số sampling and reconstruction

Exercises 2a • A sound wave has the form where t is in milliseconds. What is the frequency content of this signal? Which parts of it are audible and why?

pdf27 trang | Chia sẻ: nguyenlam99 | Lượt xem: 740 | Lượt tải: 0download
Bạn đang xem trước 20 trang tài liệu Xử lý tín hiệu số sampling and reconstruction, để xem tài liệu hoàn chỉnh bạn click vào nút DOWNLOAD ở trên
Xử lý tín hiệu số Sampling and Reconstruction Ngô Quốc Cường Ngô Quốc Cường ngoquoccuong175@gmail.com sites.google.com/a/hcmute.edu.vn/ngoquoccuong Sampling and reconstruction • Introduction • Review of analog signal • Sampling theorem • Analog reconstruction 2 1.1 Introduction Digital processing of analog signals proceeds in three stages: 1. The analog signal is digitized, that is, it is sampled and each sample quantized to a finite number of bits. This process is called A/D conversion. 2. The digitized samples are processed by a digital signal processor. 3. The resulting output samples may be converted back into analog form by an analog reconstructor (D/A conversion). 3 1.2 Review of analog signal • An analog signal is described by a function of time, say, x(t). The Fourier transform X(Ω) of x(t) is the frequency spectrum of the signal: • The physical meaning of X(Ω) is brought out by the inverse Fourier transform, which expresses the arbitrary signal x(t) as a linear superposition of sinusoids of different frequencies: 4 1.2 Review of analog signal • The response of a linear system to an input signal x(t): • The system is characterized completely by the impulse response function h(t). The output y(t) is obtained in the time domain by convolution: 5 1.2 Review of analog signal • In the frequency domain by multiplication: • where H(Ω) is the frequency response of the system, defined as the Fourier transform of the impulse response h(t): 6 1.2 Review of analog signal 7 8 CT Fourier Transforms of Periodic Signals Source: Jacob White 9 Fourier Transform of Cosine Source: Jacob White 10 Note: (period in t) T (period in ) 2/T Impulse Train (Sampling Function) Source: Jacob White 1.3 Sampling theorem • The sampling process is illustrated in Fig. 1.3.1, where the analog signal x(t) is periodically measured every T seconds. Thus, time is discretized in units of the sampling interval T: 11 1.3 Sampling theorem • For system design purposes, two questions must be answered: 1. What is the effect of sampling on the original frequency spectrum? 2. How should one choose the sampling interval T? 12 1.3 Sampling theorem • Although the sampling process generates high frequency components, these components appear in a very regular fashion, that is, every frequency component of the original signal is periodically replicated over the entire frequency axis, with period given by the sampling rate: • Let x(t) = xc(t), sampling pulse s(t) 13 1.3 Sampling theorem 14 Source: Zheng-Hua Tan 1.3 Sampling theorem 15 Source: Zheng-Hua Tan 1.3 Sampling theorem 16 Source: Zheng-Hua Tan 1.3 Sampling theorem 17 Source: Zheng-Hua Tan 1.3 Sampling theorem 18 Source: Zheng-Hua Tan 1.3 Sampling theorem 19 Source: Zheng-Hua Tan 1.3 Sampling theorem • The sampling theorem provides a quantitative answer to the question of how to choose the sampling time interval T. • T must be small enough so that signal variations that occur between samples are not lost. But how small is small enough? • It would be very impractical to choose T too small because then there would be too many samples to be processed. 20 1.3 Sampling theorem Hardware limits • In real-time applications, each input sample must be acquired, quantized, and processed by the DSP, and the output sample converted back into analog format. Many of these operations can be pipelined to reduce the total processing time. • In any case, there is a total processing or computation time, say Tproc seconds, required for each sample. where: 21 1.4 Aliasing and reconstructor • The set of frequencies, are equivalent to each other. • Among the frequencies in the replicated set, there is a unique one that lies within the Nyquist interval. It is obtained by reducing the original f modulo-fs. 22 1.4 Aliasing and reconstructor • Antialiasing prefilters 23 1.4 Aliasing and reconstructor • An ideal analog reconstructor extracts from a sampled signal all the frequency components that lie within the Nyquist interval. 24 Exercises 1 • Let x(t) be the sum of sinusoidal signals 25 Exercises 2a • A sound wave has the form where t is in milliseconds. What is the frequency content of this signal? Which parts of it are audible and why? 26 Exercises 2b 27

Các file đính kèm theo tài liệu này:

  • pdfdsp_lec2_4608.pdf