CO6021/COA002 Digital Signal Processing Handout 812/1/2009by P.C. ChangDSP Final Projects - 2009Project 1. Audio equalizer - 設(shè)計(jì)一個(gè)可自由調(diào)整frequency response 之 equalizer (等化器), 規(guī)格如下：(1) sample rate : 44.1 kHz(2) 20 kHz frequency response set to 0(3) using IIR filter, order less than 16(4) design a user interface to adjust frequency response(5) 7個(gè)等化中心頻率：125, 250, 500, 1k, 2k, 4k, 8k Hz(6) 可調(diào)範(fàn)圍 10 dB +10 dB at any frequency from 100 to 10k Hz(7) 評比標(biāo)準(zhǔn)：perceptual testing, computational complexity (filter coefficient calculation and audio signal filtering), ripple, adjacent channel response error.使用test audio 1 and audio 2 (mp3 reconstructed)(8) The report must include tables with the following items:1) 是否會(huì)聽到noise ?2) complexity3) ripple4) delay5) 自行開發(fā)?Project 2. DTMF detector design a filter bank with 8 BPFs that can detect correct digits represented by DTMF signals. The specification is listed as follows:(1) DTMF generation interface：referred to DTMF tone generator in “dsptutor” website. /(2) DTMF signal format：input *.wav format signal with sampling rate 44.1 kHz.(3) output interface：display shows the detected digits.(4) Using IIR or FIR filter(5) Filter bank spec.：as used on touch tone phones. (you find it !)(6) Criteria：accurate detection rate.(%)(7) Test pattern：1) DTMF sequence generated by yourself2) DTMF sequence generated by dsptutor website3) DTMF sequence generated by a touch-tone telephone set.The report must include tables with the following items:1) filter type (IIR、FIR?)2) filter order3) overall system complexityProject 3. Sampling rate converter - 設(shè)計(jì) sampling rate converter, 規(guī)格如下：(1) 轉(zhuǎn)換 sampling frequency from 32 kHz or 48 kHz 到 44.1 kHz(2) 評比標(biāo)準(zhǔn)：low complexity (less than 10 MIPS* will have extra credit ), low memory requirement, transparency converter (low distortion, high SNR, by perceptual testing ).Reference(1) “A Course in Digital Signal Processing”, Boaz Porat, pp. 473 481, 1997(2) “A Real-time Method for Sample Rate Conversion from CD to DAT”, Sangil Park, Motorola Inc., Consumer Electronics, 1990. ICCE 90. IEEE 1990 International Conference on , pp. 360 361NOTE:(1) * MIPS : Million Instruction Per Second (2) Test Pattern:Input signal : Pure tone signals (100Hz, 1kHz, and 10kHz) sampled at 32kHz and 48kHz.Output Reference signal : Pure tone signal (100Hz, 1kHz, and 10kHz) sampled at 44.1kHz.SNR= 10 * log (output signal power)/(error signal power)Error signal = output reference signal output signal (after synchronization)Project 4. Image scalar - 設(shè)計(jì) image scalar, 規(guī)格如下：(1) 轉(zhuǎn)換 image size from 352*288 and 320*240 to 352*240(2) 評比標(biāo)準(zhǔn)：low complexity, low memory requirement, transparency converter (low distortion, high PSNR, by perceptual testing ).Reference(1) “A Course in Digital Signal Processing”, Boaz Porat, pp. 473 481, 1997(2) “Image and Video Compression for Multimedia Engineering: Fundamentals, Algorithms, and Standards”, Yun Q. Shi, Huifang Sun, 1999NOTE:(1) Test Pattern:Input image: MTF (modulation transfer function) (given Matlab code) and image with the size 352*288 and 320*240Output Reference signal: MTF (modulation transfer function) (given Matlab code) and image with the size 352*240PSNR= 10 * log (peak signal power)/(error signal power)Error signal = output reference signal output signal (after synchronization)Project 5. Hilbert Transformers (IIR or FIR filter) - design a Hilbert transformer in a single-sideband modulation system (SSB) as in the following figure :H(f) is the Hilbert Transformer. The spec of this system is as follows:Sampling rate: 800 kHzCarrier frequency: 200 KHzMessage bandwidth: 4 kHzCheck point: (A) Whether the frequency response Xc(t) matches the SSB characteristics.(B) adjacent channel interference : case I : -40 dBcase II : -60 dBcase III : -80 dBReference(1) “Principles of Communications Systems, Modulation, and Noise, Third Edition”, Ziemer/Tranter, pp. 152 158, 1997(2) “Principles of Communications Systems, Modulation, and Noise, Third Edition”, Ziemer/Tranter, pp. 99 109, 1997(3) “A Course in Digital Signal Processing”, Boaz Porat, pp. 569 575, 1997The report must include tables with the following items:1) filter type (IIR、FIR?)2) filter order3) overall system complexityProject 6. Digital Comb Filters - design digital comb filters to separate Luminance Y(t) and Chrominance C(t), i.e., I/Q, signals from NTSC TV signals. I, Q are quadrature modulated by fc. fc=455*fH/2=3579545Hz = 3.58MHz, where fH =15734Hz (line-scan-rate) I, Q are quadrature amplitude modulated (QAM), C(t) = I(t)cos(2fc+33o) + Q(t)sin (2fc+33o) Composite video: Sc(t) = Y(t) + C(t) Demodulation Ideal case : comb filter to separate Y(t) and C(t), synchronous demodulation of C(t) to recover I(t) and Q(t). Economic case: simple bandpass filters to separate luminance and chrominance, luminance is 3 MHz bandwidth, I,Q chrominance is 0.5 MHz bandwidth. Digitization: fmax = 4.2MHz, 4fc=14.3MHz is a multiple of line-scan rate. Test procedure:1. obtain NTSC TV signal2. digitize NTSC signal as the input signals of the filters3.