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DSP First

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DSP First

ویرایش: 2nd 
نویسندگان: , ,   
سری:  
ISBN (شابک) : 9780136019251 
ناشر: Pearson 
سال نشر: 2016 
تعداد صفحات: 581 
زبان: English 
فرمت فایل : PDF (درصورت درخواست کاربر به PDF، EPUB یا AZW3 تبدیل می شود) 
حجم فایل: 8 مگابایت 

قیمت کتاب (تومان) : 36,000



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توجه داشته باشید کتاب DSP اول نسخه زبان اصلی می باشد و کتاب ترجمه شده به فارسی نمی باشد. وبسایت اینترنشنال لایبرری ارائه دهنده کتاب های زبان اصلی می باشد و هیچ گونه کتاب ترجمه شده یا نوشته شده به فارسی را ارائه نمی دهد.


توضیحاتی در مورد کتاب DSP اول

برای دوره های مقدماتی (دوره های سال اول و دوم) پردازش سیگنال دیجیتال و سیگنال ها و سیستم ها. ممکن است قبل از گذراندن یک دوره آموزشی در مدارها، از متن استفاده شود. DSP First و دارایی های دیجیتال همراه آن نتیجه بیش از 20 سال کار است که از این فرضیه که پردازش سیگنال بهترین نقطه شروع برای مطالعه مهندسی برق و کامپیوتر است، سرچشمه گرفته و هدایت شده است. رویکرد "DSP First" استفاده از ریاضیات را به عنوان زبانی برای تفکر در مورد مسائل مهندسی معرفی می کند، زمینه را برای دوره های بعدی فراهم می کند و به دانش آموزان تجربیات عملی با MATLAB می دهد. نسخه دوم دارای سه فصل جدید در سری فوریه، تبدیل فوریه گسسته، و تبدیل فوریه گسسته و همچنین آزمایشگاه های به روز شده، نمایش های تصویری، به روز رسانی به فصل های موجود و صدها مشکل و راه حل تکالیف جدید است.


توضیحاتی درمورد کتاب به خارجی

For introductory courses (freshman and sophomore courses) in Digital Signal Processing and Signals and Systems. Text may be used before the student has taken a course in circuits. DSP First and it's accompanying digital assets are the result of more than 20 years of work that originated from, and was guided by, the premise that signal processing is the best starting point for the study of electrical and computer engineering. The "DSP First" approach introduces the use of mathematics as the language for thinking about engineering problems, lays the groundwork for subsequent courses, and gives students hands-on experiences with MATLAB. The Second Edition features three new chapters on the Fourier Series, Discrete-Time Fourier Transform, and the The Discrete Fourier Transform as well as updated labs, visual demos, an update to the existing chapters, and hundreds of new homework problems and solutions.



فهرست مطالب

Content: Introduction    1-1 Mathematical Representation of Signals  1-2 Mathematical Representation of Systems  1-3 Systems as Building Blocks1-4 The Next Step Sinusoids   2-1 Tuning Fork Experiment   2-2 Review of Sine and Cosine Functions2-3 Sinusoidal Signals2-3.1 Relation of Frequency to Period2-3.2   Phase and Time Shift2-4 Sampling and Plotting Sinusoids2-5 Complex Exponentials and Phasors2-5.1 Review of Complex Numbers 2-5.2 Complex Exponential Signals2-5.3   The Rotating Phasor Interpretation2-5.4   Inverse Euler Formulas Phasor Addition2-6 Phasor Addition2-6.1   Addition of Complex Numbers2-6.2   Phasor Addition Rule2-6.3   Phasor Addition Rule: Example2-6.4   MATLAB Demo of Phasors2-6.5   Summary of the Phasor Addition Rule Physics of the Tuning Fork2-7.1   Equations from Laws of Physics2-7.2   General Solution to the Differential Equation2-7.3   Listening to Tones2-8 Time Signals: More Than FormulasSummary and LinksProblemsSpectrum Representation  3-1 The Spectrum of a Sum of Sinusoids3-1.1   Notation Change3-1.2   Graphical Plot of the Spectrum3-1.3   Analysis vs. SynthesisSinusoidal Amplitude Modulation3-2.1   Multiplication of Sinusoids3-2.2   Beat Note Waveform3-2.3   Amplitude Modulation3-2.4   AM Spectrum3-2.5   The Concept of BandwidthOperations on the Spectrum3-3.1   Scaling or Adding a Constant3-3.2   Adding Signals3-3.3   Time-Shifting x.t/ Multiplies ak by a Complex Exponential3-3.4   Differentiating x.t/ Multiplies ak by .j 2nfk/3-3.5   Frequency ShiftingPeriodic Waveforms3-4.1   Synthetic Vowel3-4.3   Example of a Non-periodic SignalFourier Series3-5.1   Fourier Series: Analysis3-5.2   Analysis of a Full-Wave Rectified Sine Wave3-5.3   Spectrum of the FWRS Fourier Series3-5.3.1  DC Value of Fourier Series3-5.3.2  Finite Synthesis of a Full-Wave Rectified SineTime-Frequency Spectrum3-6.1   Stepped Frequency3-6.2   Spectrogram AnalysisFrequency Modulation: Chirp Signals3-7.1   Chirp or Linearly Swept Frequency3-7.2   A Closer Look at Instantaneous FrequencySummary and LinksProblemsFourier Series Fourier Series Derivation4-1.1   Fourier Integral DerivationExamples of Fourier Analysis4-2.1   The Pulse Wave4-2.1.1  Spectrum of a Pulse Wave4-2.1.2  Finite Synthesis of a Pulse Wave4-2.2   Triangle Wave4-2.2.1  Spectrum of a Triangle Wave4-2.2.2  Finite Synthesis of a Triangle Wave4-2.3   Half-Wave Rectified Sine4-2.3.1  Finite Synthesis of a Half-Wave Rectified SineOperations on Fourier Series4-3.1   Scaling or Adding a Constant4-3.2   Adding Signals4-3.3   Time-Scaling4-3.4   Time-Shifting x.t/ Multiplies ak by a Complex Exponential4-3.5   Differentiating x.t/ Multiplies ak by .j!0k/4-3.6   Multiply x.t/ by SinusoidAverage Power, Convergence, and Optimality4-4.1   Derivation of Parseval's Theorem4-4.2   Convergence of Fourier Synthesis4-4.3   Minimum Mean-Square ApproximationPulsed-Doppler Radar Waveform4-5.1   Measuring Range and VelocityProblemsSampling and Aliasing  Sampling5-1.1   Sampling Sinusoidal Signals5-1.2   The Concept of Aliasing5-1.3   Spectrum of a Discrete-Time Signal5-1.4   The Sampling Theorem5-1.5   Ideal ReconstructionSpectrum View of Sampling and Reconstruction5-2.1   Spectrum of a Discrete-Time Signal Obtained by Sampling5-2.2   Over-Sampling5-2.3   Aliasing Due to Under-Sampling5-2.4   Folding Due to Under-Sampling5-2.5   Maximum Reconstructed FrequencyStrobe Demonstration5-3.1   Spectrum InterpretationDiscrete-to-Continuous Conversion5-4.1   Interpolation with Pulses5-4.2   Zero-Order Hold Interpolation5-4.3   Linear Interpolation5-4.4   Cubic Spline Interpolation5-4.5   Over-Sampling Aids Interpolation5-4.6   Ideal Bandlimited InterpolationThe Sampling TheoremSummary and LinksProblemsFIR Filters     6-1 Discrete-Time Systems6-2 The Running-Average Filter6-3 The General FIR Filter6-3.1   An Illustration of FIR FilteringThe Unit Impulse Response and Convolution6-4.1   Unit Impulse Sequence6-4.2   Unit Impulse Response Sequence6-4.2.1  The Unit-Delay System6-4.3   FIR Filters and Convolution6-4.3.1  Computing the Output of a Convolution6-4.3.2  The Length of a Convolution6-4.3.3  Convolution in MATLAB6-4.3.4  Polynomial Multiplication in MATLAB6-4.3.5  Filtering the Unit-Step Signal6-4.3.6  Convolution is Commutative6-4.3.7  MATLAB GUI for ConvolutionImplementation of FIR Filters6-5.1   Building Blocks6-5.1.1  Multiplier6-5.1.2  Adder6-5.1.3  Unit Delay6-5.2   Block Diagrams6-5.2.1  Other Block Diagrams6-5.2.2  Internal Hardware DetailsLinear Time-Invariant (LTI) Systems6-6.1   Time Invariance6-6.2   Linearity6-6.3   The FIR CaseConvolution and LTI Systems6-7.1   Derivation of the Convolution Sum6-7.2   Some Properties of LTI SystemsCascaded LTI SystemsExample of FIR FilteringSummary and LinksProblemsFrequency Response of FIR Filters7-1 Sinusoidal Response of FIR Systems7-2 Superposition and the Frequency Response7-3 Steady-State and Transient Response7-4 Properties of the Frequency Response7-4.1   Relation to Impulse Response and Difference Equation7-4.2   Periodicity of H.ej !O /7-4.3   Conjugate Symmetry Graphical Representation of the Frequency Response7-5.1   Delay System7-5.2   First-Difference System7-5.3   A Simple Lowpass Filter Cascaded LTI SystemsRunning-Sum Filtering7-7.1   Plotting the Frequency Response7-7.2   Cascade of Magnitude and Phase7-7.3   Frequency Response of Running Averager7-7.4   Experiment: Smoothing an ImageFiltering Sampled Continuous-Time Signals7-8.1   Example: Lowpass Averager7-8.2   Interpretation of DelaySummary and LinksProblemsThe Discrete-Time Fourier Transform DTFT: Discrete-Time Fourier Transform8-1.1   The Discrete-Time Fourier Transform8-1.1.1  DTFT of a Shifted Impulse Sequence8-1.1.2  Linearity of the DTFT8-1.1.3  Uniqueness of the DTFT8-1.1.4  DTFT of a Pulse8-1.1.5  DTFT of a Right-Sided Exponential Sequence8-1.1.6  Existence of the DTFT8-1.2   The Inverse DTFT8-1.2.1  Bandlimited DTFT8-1.2.2  Inverse DTFT for the Right-Sided Exponential8-1.3   The DTFT is the SpectrumProperties of the DTFT8-2.1   The Linearity Property8-2.2   The Time-Delay Property8-2.3   The Frequency-Shift Property8-2.3.1  DTFT of a Complex Exponential8-2.3.2  DTFT of a Real Cosine Signal8-2.4   Convolution and the DTFT8-2.4.1  Filtering is Convolution8-2.5   Energy Spectrum and the Autocorrelation Function8-2.5.1  Autocorrelation FunctionIdeal Filters8-3.1   Ideal Lowpass Filter8-3.2   Ideal Highpass Filter8-3.3   Ideal Bandpass FilterPractical FIR Filters8-4.1   Windowing8-4.2   Filter Design 8-4.2.1  Window the Ideal Impulse Response 8-4.2.2  Frequency Response of Practical Filters8-4.2.3  Passband Defined for the Frequency Response8-4.2.4  Stopband Defined for the Frequency Response8-4.2.5  Transition Zone of the LPF8-4.2.6  Summary of Filter Specifications8-4.3   GUI for Filter DesignTable of Fourier Transform Properties and PairsSummary and LinksProblemsThe Discrete Fourier Transform   Discrete Fourier Transform (DFT)9-1.1   The Inverse DFT9-1.2   DFT Pairs from the DTFT9-1.2.1  DFT of Shifted Impulse9-1.2.2  DFT of Complex Exponential9-1.3   Computing the DFT9-1.4   Matrix Form of the DFT and IDFTProperties of the DFT9-2.1   DFT Periodicity for XOEk]9-2.2   Negative Frequencies and the DFT9-2.3   Conjugate Symmetry of the DFT9-2.3.1  Ambiguity at XOEN=2]9-2.4   Frequency Domain Sampling and Interpolation9-2.5   DFT of a Real Cosine SignalInherent Periodicity of xOEn] in the DFT9-3.1   DFT Periodicity for xOEn]9-3.2   The Time Delay Property for the DFT9-3.2.1  Zero Padding9-3.3   The Convolution Property for the DFTTable of Discrete Fourier Transform Properties and PairsSpectrum Analysis of Discrete Periodic Signals9-5.1   Periodic Discrete-time Signal: Fourier Series9-5.2   Sampling Bandlimited Periodic Signals9-5.3   Spectrum Analysis of Periodic SignalsWindows9-6.0.1  DTFT of WindowsThe Spectrogram9-7.1   An Illustrative Example9-7.2   Time-Dependent DFT9-7.3   The Spectrogram Display9-7.4   Interpretation of the Spectrogram9-7.4.1  Frequency Resolution9-7.5   Spectrograms in MATLABThe Fast Fourier Transform (FFT)9-8.1   Derivation of the FFT9-8.1.1  FFT Operation CountSummary and LinksProblemsz-Transforms   Definition of the z-TransformBasic z-Transform Properties10-2.1  Linearity Property of the z-Transform10-2.2  Time-Delay Property of the z-Transform10-2.3  A General z-Transform FormulaThe z-Transform and Linear Systems10-3.1  Unit-Delay System10-3.2  z-1 Notation in Block Diagrams10-3.3   The z-Transform of an FIR Filter10-3.4   z-Transform of the Impulse Response10-3.5  Roots of a z-transform PolynomialConvolution and the z-Transform10-4.1  Cascading Systems10-4.2  Factoring z-Polynomials10-4.3  DeconvolutionRelationship Between the z-Domain and the !O -Domain10-5.1   The z-Plane and the Unit CircleThe Zeros and Poles of H.z/10-6.1  Pole-Zero Plot10-6.2   Significance of the Zeros of H.z/10-6.3  Nulling Filters10-6.4  Graphical Relation Between z and !O10-6.5  Three-Domain MoviesSimple Filters10-7.1   Generalize the L-Point Running-Sum Filter10-7.2  A Complex Bandpass Filter10-7.3  A Bandpass Filter with Real CoefficientsPractical Bandpass Filter DesignProperties of Linear-Phase Filters10-9.1  The Linear-Phase Condition10-9.2  Locations of the Zeros of FIR Linear-Phase SystemsSummary and LinksProblemsIIR Filters  The General IIR Difference EquationTime-Domain Response11-2.1  Linearity and Time Invariance of IIR Filters11-2.2  Impulse Response of a First-Order IIR System11-2.3  Response to Finite-Length Inputs11-2.4  Step Response of a First-Order Recursive SystemSystem Function of an IIR Filter11-3.1  The General First-Order Case11-3.2  H.z/ from the Impulse Response11-3.3  The z-Transform MethodThe System Function and Block-Diagram Structures11-4.1  Direct Form I Structure11-4.2  Direct Form II Structure11-4.3  The Transposed Form StructurePoles and Zeros11-5.1  Roots in MATLAB11-5.2  Poles or Zeros at z D 0 or 111-5.3  Output Response from Pole LocationStability of IIR Systems11-6.1  The Region of Convergence and StabilityFrequency Response of an IIR Filter11-7.1  Frequency Response using MATLAB11-7.2  Three-Dimensional Plot of a System FunctionThree DomainsThe Inverse z-Transform and Some Applications11-9.1  Revisiting the Step Response of a First-Order System11-9.2  A General Procedure for Inverse z-TransformationSteady-State Response and StabilitySecond-Order Filters11-11.1 z-Transform of Second-Order Filters11-11.2 Structures for Second-Order IIR Systems11-11.3 Poles and Zeros11-11.4 Impulse Response of a Second-Order IIR System11-11.4.1  Distinct Real Poles11-11.5 Complex PolesFrequency Response of Second-Order IIR Filter11-12.1 Frequency Response via MATLAB11-12.23-dB Bandwidth11-12.3 Three-Dimensional Plot of System FunctionsExample of an IIR Lowpass FilterSummary and LinksProblems




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