ISSN:1124-318X    

DOI:10.1002/ett.4460050502

出版商:Wiley Subscription Services, Inc., A Wiley Company    

年代:1994

数据来源: WILEY

ISSN:1124-318X    

DOI:10.1002/ett.4460050503

出版商:Wiley Subscription Services, Inc., A Wiley Company    

年代:1994

数据来源: WILEY

ISSN:1124-318X    

DOI:10.1002/ett.4460050504

出版商:Wiley Subscription Services, Inc., A Wiley Company    

年代:1994

数据来源: WILEY

摘要:

AbstractThis paper considers the optimization of a vector quantizer whose indices are transmitted over a channel with known noise statistics (pdf). A decoder is introduced which uses a linear combination of VQ codebook entries with weights depending on the received signal. Rather than using a received index to select an entry in the codebook (which implies a hard decision on the index), the decoder uses the received signal and the channel statistics to weight the codebook entries in order to produce a good reconstructed signal. Borrowing a term used in digital communications, we will say that the optimal decoder uses a “soft” decision rule. The corresponding approach will be called Soft Decision Vector Quantization (SDVQ). It is shown that SDVQ achieves a significant performance improvement over both the source optimized VQ and the channel optimized VQ (the latter is based on known channel transition probabilities). For correlated sources, it is shown that SDVQ performance can be further improved by using sequential decoding. Results are presented for an AR(1) source and for speech line spectral pair (LSP) parameter quantization at 24 bits/fr

ISSN:1124-318X    

DOI:10.1002/ett.4460050505

出版商:Wiley Subscription Services, Inc., A Wiley Company    

年代:1994

数据来源: WILEY

摘要:

AbstractEfficient quantization of synthesis filter coefficients for CELP (Code Excited Linear Prediction) coders is essential to achieve high quality speech at low rates. Three vector quantizers with good potential for utilization in low rate coders are studied. Each of them is implemented inside the structure of the VSELP (Vector Sum Excited Linear Prediction) coder, an important member in the class of CELP coders. These three vector quantizers are used to encode LSF (Line Spectral Frequency) parameters and are compared in terms of robustness to channel errors, complexity and quality of synthesized speech. Performance of synthesized speech is evaluated considering the objective measure of frequency weighted signal to noise ratio and subjective results obtained from listening tests. With the purpose of improving the robustness to channel errors, the application of simulated annealing to assign binary indices to the output levels of the quantizer is also investigated. A split vector quantization scheme which employs interframe prediction has shown to be an attractive approach to encode the synthesis filter parameters. It provides a performance comparable to the IS‐54 scheme and uses 10 bits less for each LSF fram

ISSN:1124-318X    

DOI:10.1002/ett.4460050506

出版商:Wiley Subscription Services, Inc., A Wiley Company    

年代:1994

数据来源: WILEY

摘要:

AbstractThis paper describes the development of efficient methods to encode the parameters of long‐term (pitch) and short‐term (LPC all‐pole) prediction filters with a limited bit‐rate, such that the design of high quality speech coders in the 3–4 kbit/s range becomes easier. Our results indicate that good performance can be reached with about 800 bit/s to represent both the pitch lags and the coefficients of a three‐tap pitch filter, thus achieving a significant 36% bit‐rate reduction in comparison with conventional quantization techniques, and using 24 bit/frame to encode spectral parameters relative to amultilanguagesp

ISSN:1124-318X    

DOI:10.1002/ett.4460050507

出版商:Wiley Subscription Services, Inc., A Wiley Company    

年代:1994

数据来源: WILEY

摘要:

AbstractAt bit rates between 4 and 16 kbit/s, many state‐of‐the‐art speech coding algorithms fall into the class of linear‐prediction based analysis‐by‐synthesis (LPAS) speech coders. At the lower bit rates the waveform‐matching, on which LPAS coders rely, constrains the speech quality. To overcome this drawback, we present a coder (RCELP) that uses a generalization of the analysis‐by‐synthesis paradigm. This generalization relaxes the waveform‐matching constraints without affecting speech quality. We describe several implementations at bit rates between 4 and 6 kbit/s. MOS tests show that a 6 kbit/s RCELP has a quality similar or better than the 13 kbit/s GSM full‐rate coder, and a 4.4 kbit/s RCELP has a speech quality significantly better than the 4.8 k

ISSN:1124-318X    

DOI:10.1002/ett.4460050508

出版商:Wiley Subscription Services, Inc., A Wiley Company    

年代:1994

数据来源: WILEY

摘要:

AbstractThis paper discusses two speech‐coding algorithms at around 8 kbit/s. These algorithms both use pre‐selection in the codebook search and training for the conjugate structured random codebook. One has short‐delay time that can dispense with echo‐control equipment. This algorithm is robust against channel errors by using LPC analysis separated from pitch excitation, although LPC parameters are derived from the locally decoded signal. The other algorithm has medium‐delay time and meets all the requirements of the ITU‐T (formerly CCITT) 8 kbit/s speech coding standard. The performance of these codecs is evaluated using signal to noise ratio (SNR), pair comparison tests with a modulated noise reference unit (MNRU), and Mean Opinion Score (MOS). The major applications of these codecs are personal handy phone systems

ISSN:1124-318X    

DOI:10.1002/ett.4460050509

出版商:Wiley Subscription Services, Inc., A Wiley Company    

年代:1994

数据来源: WILEY

摘要:

AbstractA variable bit‐rate speech coder intended for digital cellular applications is described. A voice activity detection algorithm is used to distinguish active speech from background noise. Each frame of active speech is further classified to distinguish between three phonetic categories: voiced, unvoiced, and onset. Each input frame is assigned one of five bit rates according to voice activity and phonetic classification and coded using an analysis‐by‐synthesis algorithm tailored to the needs of the class that it belongs to. The resulting coder, calledVariable Rate Phonetic Segmentation, produces good quality speech at an average bit‐rate below 3 kbit/s when operating with a voice activity factor of 0.5. Informal subjective quality assessment for speech in clean and noisy backgrounds indicates a performance that is comparable to the TIA standard QCELP algorithm while operating at a 25% to 40% lower average b

ISSN:1124-318X    

DOI:10.1002/ett.4460050510

出版商:Wiley Subscription Services, Inc., A Wiley Company    

年代:1994

数据来源: WILEY

摘要:

AbstractThis paper describes the general principle of variable rate applied to CELP coding and presents an algorithm conceived for mobile applications based on CDMA technique. The speech coder operates at an average rate of around 6 kbit/s with a peak rate of 16 kbit/s, providing a quality close to the one of the G.728 standard. The algorithm is based on a multi‐rate CELP whose operating mode is selected by a combination of open‐loop speech classification and closed‐loop quality evaluation. The scheme is not fully optimized for the average rate, but attention has mainly been paid to the classification algorithm having the constraint to keep the algorithmic delay equal to

ISSN:1124-318X    

DOI:10.1002/ett.4460050511

出版商:Wiley Subscription Services, Inc., A Wiley Company    

年代:1994

数据来源: WILEY