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Rabiner L.R., Schafer R.W. — Digital Processing of Speech Signals
Rabiner L.R., Schafer R.W. — Digital Processing of Speech Signals



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Íàçâàíèå: Digital Processing of Speech Signals

Àâòîðû: Rabiner L.R., Schafer R.W.

Àííîòàöèÿ:

The purpose of this text is to show how digital signal processing techniques can be applied to problems related to speech communication. The book gives an extensive description of the physical basis for speech coding including fourier analysis, digital representation and digital and time domain models of the wave form. It goes on to discuss homomorphic speech processing, linear predictive coding and digital processing for machine communication by voice.


ßçûê: en

Ðóáðèêà: Òåõíîëîãèÿ/

Ñòàòóñ ïðåäìåòíîãî óêàçàòåëÿ: Ãîòîâ óêàçàòåëü ñ íîìåðàìè ñòðàíèö

ed2k: ed2k stats

Ãîä èçäàíèÿ: 1978

Êîëè÷åñòâî ñòðàíèö: 512

Äîáàâëåíà â êàòàëîã: 31.03.2007

Îïåðàöèè: Ïîëîæèòü íà ïîëêó | Ñêîïèðîâàòü ññûëêó äëÿ ôîðóìà | Ñêîïèðîâàòü ID
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Ïðåäìåòíûé óêàçàòåëü
Acoustic admittance of a uniform lossless tube      64
Acoustic admittance, due to thermal conduction      69
Acoustic admittance, due to yielding walls      68
Acoustic impedance of a uniform lossless tube      64
Acoustic impedance, due to viscous friction      69
Adaptive delta modulation with one bit memory      221—223
Adaptive delta modulation, continuously variable slope (CVSD)      223—224
Adaptive differential PCM (ADPCM)      226—232
Adaptive prediction in DPCM      228—232
Adaptive prediction, feed-forward control      232
Adaptive prediction, feedback control      229—232
Adaptive quantization      195—208
Additive modifications of the short-time spectrum      280—281
Affricates      54
Aids-to-the-handicapped      8
Aliasing in computing the cepstrum      364—365 393
Aliasing, definition      26
All-pole model for speech      398
All-pole transfer function      99
All-pole transfer function, cascade realization      100
All-pole transfer function, direct form realization      101
AMDF      149—150
Analysis frames      117
Analysis-by-synthesis      318—323
Analysis-by-synthesis, comparison to LPC      439—440
Analysis-synthesis systems      324—344
APCM      196
Area function      44
Area function, measurements of      61
Autocorrelation function for deterministic signals      141
Autocorrelation function for periodic signals      141
Autocorrelation function for random signals      141
Autocorrelation function, computation of      162—164
Autocorrelation function, properties of      141
Autocorrelation function, short-time      141—149
Autocorrelation method of LPC      401—403
Average magnitude difference function (AMDF)      149—150
Backward prediction error      414
Bandwidth of Hamming window      264
Bandwidth of speech signals      174
Bit-rate      180
Boundary condition at glottis      78—82 86—87
Boundary condition at lips      71—74 85—86 111 113
Burg's method      416
Cascade form implementation      23
Causal systems      19 20
Center clipping      151—154
Center clipping, three level      154
Cepstrum      359
Cepstrum from predictor polynomial      442
Cepstrum, smoothing      378
Cepstrum, window      369—370
Channel vocoder      341—344
Characteristic impedance      64
Characteristic system for homomorphic deconvolution      357—359
Chirp z-transform      379 394—395
Cholesky decomposition      407—411
Clipping level effect of in pitch estimation      153
companding      186—191
Comparison of digital coders      232—235
Complex cepstrum for rational transforms      360
Complex cepstrum of speech      365—372
Complex cepstrum of speech, unvoiced      370—372
Complex cepstrum of speech, voiced      367—370
Complex cepstrum, computation of      363—365
Complex cepstrum, properties of      360—362
Composite frequency response      268
Composite impulse response using real filters      285
Compressor u-law      188
Computational requirements in LPC      417—418
Computer voice response      see "Voice response"
Concatenation of formant-coded words      470—473
Continuant sounds      43
Continuous digit recognition      494—498
Convolution, descrete      13
Correlation function, long term estimate      177—178 see
Covariance method of LPC      403—404
Cross-correlation      148
CVSD      223—224
CVSD, maximum and minimum step sizes      247
Decimation and interpolation      27 273—274
Deconvolution      355
Delta modulation      216—225
Delta modulation, adaptive      221—224
Delta modulation, double integration      225
Delta modulation, linear      216—221
Design of digital filter banks      see "Digital filter bank design"
Differential PCM (DPCM)      225—232
Differential quantization      208—216
Digital code conversion      235—238
Digital coding of formants      382—384
Digital coding of LPC parameters      450—453
Digital coding of the cepstrum      388—389
Digital coding of the time dependent Fourier transform      324—334
Digital coding of the time dependent Fourier transform, using adaptive delta modulation      331—332
Digital coding of the time dependent Fourier transform, using PCM      332
Digital filter bank design      282—302
Digital filter bank design, practical considerations in      282—290
Digital filter bank design, using FIR filters      292—302
Digital filter bank design, using IIR filters      290—292
Digital filters      18—23
Digital filters, causality of      19 20
Digital filters, frequency response of      18
Digital filters, implementation of      23
Digital filters, stability of      19 20
Digital filters, system function of      18
Digital transmission of speech      7
Diphthongs      48
Direct form implementation      23
Discrete Fourier Transform      16—18
Discrete-time model for speech      103—105
Distance measures      484—485 498—500
Dithering      242—243
Double integration delta modulation      225
DPCM      225—232
Durbin's method of solution of the LPC equations      411—413
Dynamic range      187
Dynamic range of u-law quantizer      191
Encoding of quantized samples      179-181
Energy      119
Enhancement of speech quality      8
Error, quantization      182
Expander      188
Exponential sequence      11
Fast Fourier Transform (FFT), definition of      18
Fast Fourier Transform (FFT), use in computing the cepstrum      363—365
Fast Fourier Transform (FFT), use in short-time Fourier analysis      303—306
Fast Fourier Transform (FFT), use in short-time Fourier synthesis      306—310
Feed-forward adaptation      199—203
Feed-forward adaptation in differential PCM      226—227
Feed-forward adaptation, performance of      203
Feedback adaptation      203—207
Feedback quantization in differential PCM      227—228
Feedback quantization, performance of      207
Filter bank summation method      266—274
Filter bank summation method, implementation of      303—310
Finite impulse response      see "FIR"
FIR systems      20—21
FIR systems, design of      20
FIR systems, linear phase      20
Formant frequencies      41 44
Formant frequencies of uniform lossless tube      65—66
Formant frequencies, quantization of      382—384
Formant frequency estimation, using LPC      442—450
Formant frequency estimation, using the cepstrum      378—385
Formant vocoder      382—385
Forward prediction error      414
Fourier transform      15—16
Frequency domain interpretation of LPC      431—440
Frequency resolution dependence on window length      260
Frequency response      18
Frequency response of telephone line      174
Fricative      40
Fricative, excitation model      81—82
Fricative, unvoiced      51—52
Fricative, voiced      52
Gain computation in LPC      404—407
Glottis      39
Glottis, boundary condition at      63 81 87
Granular noise      219
Hamming window, definition of      121
Hamming window, Fourier transform of      122
Homomorphic systems for convolution      356—365
Homomorphic systems for convolution, canonic form for      357
Homomorphic systems for convolution, characteristic system      357
Homomorphic systems for convolution, implementation of      363—365
Homomorphic vocoder      385—390
Idle channel noise      195
IIR systems      21—23
IIR systems, design of      22
IIR systems, implementation, cascade form      23
IIR systems, implementation, direct form      23
IIR systems, implementation, parallel form      23
Infinite duration impulse response      see "IIR"
Information rate      180
Information rate of speech      2
Instantaneous quantization      179—195
Integrator      217
Interpolation      28—29
Interpolation of LPC synthesis parameters      445
Interpolation of short-time Fourier transform      306—310 324—334
Isolated digit recognition      490—493
Laplacian probability density      176 241
Lattice formulation of LPC      413—417
LDM-to-PCM conversion      236—237
Leaky integrator      224
Linear delta modulation      216—221
Linear delta modulation, circuit implementation      236
Linear filtering interpretation of time dependent Fourier analysis      261—263
Linear phase FIR systems      20
Linear predictive analysis, autocorrelation method      401—403
Linear predictive analysis, basic principles      398—401
Linear predictive analysis, covariance method      403—404
Linear predictive analysis, use in speaker identification      485—489
Linear predictive analysis, use in speaker verification      478—479
Linear predictive analysis, use in speech recognition      491—498
Linear predictive coding (LPC)      367
Linear predictive spectrum      433
Linear predictive vocoder      450—453
Linear predictor      210
Linear shift-invariant systems      13
Linguistics      39
Log area ratio      444
logarithmic quantization      186—187
Losses in the vocal tract, due to thermal conduction      69
Losses in the vocal tract, due to viscous friction      69
Losses in the vocal tract, due to yielding walls      66—69
Lossless tube models      82—98
Lossless tube models from LPC      440—441
Lossless tube models, boundary condition at glottis      86—87
Lossless tube models, boundary condition at lips      86
Lossless tube models, equivalent discrete-time system      89—92
Lossless tube models, relationship to digital filters      88—92
Lossless tube models, transfer function of      92—98
Lossless tube models, transfer function of, recursion formula for      96
LPC      see "Linear predictive coding" "Linear
LPC distance measures      498—500
Maximum phase signals      361
Maximum phase signals, complex cepstrum of      361
Median smoothing      158—161
Median smoothing, application to speech processing      160—161
Median smoothing, properties of      158
Mid-riser quantizer      181—182
Mid-tread quantizer      181—182
Minimum phase signals      361
Minimum phase signals, complex cepstrum of      361
Modifications to the short-time spectrum in OLA method      279—280
Modifications to the short-time spectrum, effects of synthesis      277—281
Modifications to the short-time spectrum, effects of synthesis in FBS method      277—279
Nasal tract      39
Nasals      49—50
Nasals, model for production      76—78
Noise, quantization      182
Normalized mean-squared error      424—426
Normalized prediction error      412
Nyquist frequency      25
Optimum quantization      191—195
Overlap addition method of short-time Fourier synthesis      274—277
Oversampling ratio      265
Parallel form implementation      23
Parallel processing pitch detector      135—141
PARCOR coefficients      415 443—444
PARCOR coefficients, quantization of      444 452
PARCOR coefficients, relation to reflection coefficients      441
PARCOR coefficients, stability condition for      419
PARCOR coefficients, synthesis from      446—447
PCM-to-ADPCM conversion      237—238
Pharynx      39
Phase adjustment in digital filterbanks      290
Phase derivative      335
Phase vocoder      334—340
Phoneme      2 42
Phoneme, classes      43
Phonetics      39
Pitch detection      see "Pitch period estimation"
Pitch period estimation, using LPC      447—449
Pitch period estimation, using parallel processing      135—141
Pitch period estimation, using the autocorrelation function      150—157
Pitch period estimation, using the autocorrelation function, algorithm for      156
Pitch period estimation, using the cepstrum      372—378
Pitch period estimation, using the short-time Fourier transform      314—318 352
Pitch synchronous estimation of the glottal wave      322—324
Pitch synchronous LPC      427
Pitch synchronous spectrum analysis      319—321
Plosive sounds      41
Pole-zero analysis      321—322
Power spectrum for speech      177—178
Prediction error in differential PCM      208
Prediction error, definition      399
Prediction error, filter      399
Prediction error, short-time average      400
Prediction error, signal      421—424
Prediction gain      210
Prediction gain, bounds on      230
Prediction gain, optimum      213
Probability density for speech      175—176
Probability density, gamma      175
Probability density, Laplacian      176 241
Quantization error      182
Quantization noise      182—186
Quantization noise, model for      182
Quantization noise, variance of      185
Quantization of LPC parameters      450
Quantization of short-time Fourier transform      324—334
Quantization of the cepstrum      388—389
Quantization, instantaneous      179—195
Quantization, uniform      181—186
Radiation of sound at the lips      71
Radiation, impedance      71
Radiation, load      71
Rectangular window, definition of      121
Rectangular window, Fourier transform of      1
Reflection coefficient, definition of      85
Sampling rate, increase      28—29
Sampling rate, reduction      27
Sampling, aliasing in      26
Sampling, speech signals      173—174
Sampling, the short-time Fourier transform      263—266 329
Sampling, theorem      24
Segmentation in speech recognition      495—498
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