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THE IMPACT OF VOICE PROCESSING ON MODERN TELECOMMUNICATIONS

SPEECH COMMUNICATION 1995年第3-4期17卷 217-226页

作者： RABINER, LR AT&T BELL LABS INFORMAT PRINCIPLES RES LABS 600 MT AVE MURRAY HILL NJ 07974 USA

Research has been conducted in the area of voice processing for over six decades but it has only been in the past few years that the impact of the years of research is starting to be seen in modern telecommunications systems. Virtually every area of voice processing, including speech coding, speech synthesis, speech recognition, and even, to a small extent, speaker verification, has left the research laboratory and now appears in a product or service that is in daily use out in the marketplace, often by millions of customers per day. This revolution in voice processing in telecommunications is fueled by algorithmic advances (which improve the quality of the voice processing systems), hardware advances (which provide high processing power and memory at low cost), and networking advances (which provide high bandwidth pipes to the home, office, and throughout the telecommunications network). In this paper we illustrate the impact of voice processing on modern telecommunications by showing the diverse ways in which speech coding, speech synthesis, speech recognition and speaker verification have become embodied in new products and services.

关键词： Speech coding linear predictive coding Code excited linear prediction Voice messaging systems Voice response systems Telephone security devices Telephone bandwidth coders Wideband speech coders Digital audio broadcast Speech synthesis Text-to-speech synthesis Speech recognition Template systems Hidden Markov models Isolated word recognition Connected word recognition Continuous speech recognition Speaker verification

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A SPEECH ANALYSIS ALGORITHM WHICH ELIMINATES THE INFLUENCE OF PITCH USING THE MODEL-REFERENCE ADAPTIVE SYSTEM

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1982年第1期30卷 88-96页

作者： MIYANAGA, Y MIKI, N NAGAI, N HATORI, K Research Institute of Applied Electricity Hokkaido University Japan

A new adaptive algorithm based upon a least square criterion with a weighting factor is presented and shown to be quite useful for estimating ARMA parameters together with input in speech analysis. The estimator of both the input pulse train for voiced speech and the input white noise for unvoiced speech are easily obtained from the prediction errors by using this new adaptive algorithm. When these estimated inputs are used as the input of the model to be estimated, the influence of the pitch can be eliminated from the estimated ARMA parameters. By using this method the accuracy of formant and antiformant estimators is shown experimentally in comparison with LPC and cepstrum estimators.

关键词： Speech analysis Adaptive systems Parameter estimation Speech enhancement Speech synthesis Adaptive algorithm White noise Speech processing Frequency response linear predictive coding

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Comparison of FPGA and GPU Implementations of LPC Algorithm for Voice Processing

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RECENT ADVANCES IN ELECTRICAL & ELECTRONIC ENGINEERING 2018年第2期11卷 188-193页

作者： Sayadi, Fatma E. Bahri, Haythem Chouchene, Marwa Atri, Mohamed Monastir Univ Fac Sci Lab Elect & Microelect E E Environm St Monastir 5019 Tunisia

Background: Many applications in voice processing have high inherent parallelism. Field programmable gate array (FPGA) has shown very high performance in spite of its low operational frequency by fully extracting the parallelism. Nevertheless, recent CPU and graphic processing unit (GPU) have also an inherent for high performance. Methods: In fact, it becomes possible to utilize the parallelism using multi-cores, which support improved single instruction multiple data (SIMD) instruction. Recent GPUs support a large number of cores, and have a potential for high performance in many applications. Our goals are at first to compare GPU and FPGA implementation of the linear prediction coding (LPC) algorithm, in order to understand the trade-off between the flexibility but relatively low speed of an FPGA and the high speed and fixed architecture of the GPU. Secondly, we try to apply various levels optimization from overlapping data transfers to fine-tuning operation sequences. Results: The experimental results highlight the relative strengths and limitations of the two systems. Conclusion: Our experiments show that, for several samples corresponding to several speeches coding, GPU manages speedups of up to 3x compared to the FPGA and around 35x compared to a sequential execution.

关键词： Compute unified device architecture (CUDA) FPGA GPU linear predictive coding optimization strategies shared memory

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Optimization and performance evaluation of graphic processing units for voice processing

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JOURNAL OF ALGORITHMS & COMPUTATIONAL TECHNOLOGY 2017年第4期11卷 388-394页

作者： Sayadi, Fatma E. Bahri, Haythem Chouchene, Marwa Atri, Mohamed Monastir Univ Fac Sci Lab Elect & Microelect EE Environm St Monastir 5019 Tunisia

With the advancement in the device technology and parallel architecture, field-programmable gate arrays (FPGAs) can well perform the speech processing operation. FPGAs have very impressive results, despite their low operating frequency, by completely extracting the parallelism. Nevertheless, recent central processing unit and graphic processing unit (GPU) have also an inherent feature for high performance. In fact, recent GPUs enable dramatic increases in computing performance by harnessing great number of cores. In this context, we seek to analyze the performance of the linear prediction coding algorithm implementation on two different platforms: one based on the GPU NVIDIA GeForce GTX 480 and another on the FPGA Spartan-6. Subsequently, we try to apply several optimization strategies on those platforms. The experimental results highlight the relative robustness or weakness of both these platforms. The tests prove that, for several samples, GPU manages speedups of up to 4x compared to the FPGA and around 48x compared to a sequential execution.

关键词： linear predictive coding graphic processing unit field-programmable gate arrays optimization strategies shared memory

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Digital acoustic analysis of five vowels in maxillectomy patients

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JOURNAL OF ORAL REHABILITATION 2002年第7期29卷 649-656页

作者： Sumita, YI Ozawa, S Mukohyama, H Ueno, T Ohyama, T Taniguchi, H Tokyo Med & Dent Univ Dept Maxillofacial Prosthet Grad Sch Bunkyo Ku Tokyo 1138549 Japan Tokyo Med & Dent Univ Dept Sports Med Dent Grad Sch Tokyo 1138549 Japan Tokyo Med & Dent Univ Dept Removable Prosthodont Grad Sch Tokyo 1138549 Japan

The aim of the study was to characterize the acoustics of vowel articulation in maxillectomy patients. Digital acoustic analysis of five vowels, /a/, /e/, /i/, /o/ and /u/, was performed on 12 male maxillectomy patients and 12 normal male individuals. A simple set of acoustic descriptions called the first and second formant frequencies, F1 and F2, were employed and calculated based on linear predictive coding. The maxillectomy patients had a significantly lower F2 for all five vowels and a significantly higher F1 for only /i/ vowel. From the data plotted on an F1-F2 plane in each subject, we determined the F1 range and the F2 range, which are the differences between the minimum and the maximum frequencies among the five vowels. The maxillectomy patients had a significantly narrower F2 range than the normal controls. In contrast, there was no significant difference in the F1 range. These results suggest that the maxillectomy patients had difficulty in controlling F2 properly. In addition, the speech intelligibility (SI) test was performed to verify the results of this new frequency range method. A high correlation between the F2 range and the score of SI test was demonstrated, suggesting that the F2 range is effective in evaluating the speech ability of maxillectomy patients.

关键词： vowel maxillectomy digital acoustic analysis linear predictive coding formant

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ESTIMATING THE STEADY-STATE FREQUENCY OF A SINE-WAVE BURST WITH EXTREMELY SHORT RECORD LENGTH

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1986年第1期34卷 213-215页

作者： PLOTKIN, EI ROYTMAN, LM SWAMY, MNS CONCORDIA UNIV DEPT ELECT ENGNMONTREAL H3G 1M8QUEBECCANADA CUNY CITY COLL DEPT ELECT ENGNNEW YORKNY 10031

We present a technique based on modified linear prediction for accurate estimation of the steady-state frequency of a sine-wave burst with extremely short record length. The estimation procedure presented here makes use of automatic windowing of the burst location within the observation interval. Such a windowing technique eliminates the transition effects which accompany the estimators derived using the traditional LPC technique. To implement this windowing, a set of artificial state variables, given by the product of output samples and their delayed versions, is used. The advantages of this method are demonstrated using a computer simulation.

关键词： Frequency estimation Steady-state Delay estimation Signal processing Least squares approximation Signal processing algorithms White noise predictive models Delay effects linear predictive coding

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SIMULATION OF VECTOR TRELLIS-ENcoding SYSTEMS

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IEEE TRANSACTIONS ON COMMUNICATIONS 1986年第3期34卷 214-218页

作者： BEI, CD GRAY, RM STANFORD UNIV INFORMAT SYST LABSTANFORDCA 94305

Most tree or trellis encoding data compression systems use decoders which form scalar outputs as a (possibly nonlinear) function of the contents of a shift register containing received channel symbols. We here develop design algorithms for trellis encoding systems having decoders not constrained to have such a scalar sliding-block code structure. In particular, we consider using the decoder of a finite-state vector quantizer together with a vector trellis search. Simulation results are presented for vector trellis encoding systems for Gauss-Markov sources, sampled speech data, and LPC speech data.

关键词： Encoding Decoding Algorithm design and analysis Image coding Speech Nonlinear distortion Data compression Gaussian processes linear predictive coding Filters

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SPEAKER-INDEPENDENT PHONE RECOGNITION USING HIDDEN MARKOV-MODELS

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1989年第11期37卷 1641-1648页

作者： LEE, KF HON, HW Computer Science Department Carnegie Mellon University Pittsburgh PA USA

Hidden Markov modeling is extended to speaker-independent phone recognition. Using multiple codebooks of various linear-predictive-coding (LPC) parameters and discrete hidden Markov models (HMMs) the authors obtain a speaker-independent phone recognition accuracy of 58.8-73.8% on the TIMIT database, depending on the type of acoustic and language models used. In comparison, the performance of expert spectrogram readers is only 69% without use of higher level knowledge. The authors introduce the co-occurrence smoothing algorithm, which enables accurate recognition even with very limited training data. Since the results were evaluated on a standard database, they can be used as benchmarks to evaluate future systems.< >

关键词： Hidden Markov models Databases Knowledge engineering Speech recognition Maximum likelihood decoding Humans Acoustics Context modeling linear predictive coding Natural languages

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VARIABLE-RATE SPEECH COMPRESSION BY ENcoding SUBSETS OF THE PARCOR COEFFICIENTS

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1983年第3期31卷 706-713页

作者： PAPAMICHALIS, PE BARNWELL, TP GEORGIA INST TECHNOL SCH ELECT ENGN ATLANTA GA 30332 USA

In LPC analysis, the speech signal is divided into frames each of which is represented by a vector of estimated vocal tract parameters, assumed to be constant throughout the frame. For many sounds, these parameters do not change significantly from one frame to the next, and some of them can often be adequately represented by previously transmitted values. In the LPC coding systems described in this paper, a number of alternative representations are considered for each frame. These representations (vectors) are combinations of PARCOR coefficients from the current frame and from previous frames. Several consecutive frames are analyzed at once, and all the possible sequences of PARCOR coefficient vectors are examined. The sequence which minimizes a preselected cost function is chosen for transmission, resulting in a reduced overall data rate. The examination of all the decision sequences is equivalent to a decision tree search, which is most efficiently accomplished through dynamic programming. Using these techniques, LPC encoded speech at 1200 bits/s is demonstrated to be of quality comparable to a constant rate LPC vocoder at 2400 bits/s.

关键词： Encoding linear predictive coding Vocoders Filters Speech analysis Speech coding Feedback Signal analysis Parameter estimation Cost function

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DIRECT SAMPLE INTERPOLATION (DSI) SPEECH SYNTHESIS - AN INTERPOLATION TECHNIQUE FOR DIGITAL SPEECH DATA-COMPRESSION AND SPEECH SYNTHESIS

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1982年第6期30卷 825-832页

作者： BEDDOES, MP CHU, TK University of British Columbia Vancouver BC Canada Department of Communications Government of Canada Ottawa ONT Canada

Direct transcription of the waveform is a potentially simple method to generate speech. However, it has a poor reputation due to the enormous data store required, although many digital encoding techniques have been suggested which reduce the data store significantly. Digital encoding is not invoked but a method called direct sample interpolation (DSI) is described which will compute bridging sections quite simply between, in the first instance, very short vowel sections called here phoneme fragments (PF's). This reduces the data store for vowels. DSI will significantly compress the store for other sounds, notably the fricatives. The method relies on the property in speech that the spectrum changes relatively slowly with time. Perhaps, most importantly, DSI can produce bridging sections of predetermined durations and pitch contouring and can therefore provide facilities for pitch inflexion and stress. Speech generation can be effected using DSI and the hardware is simpler than required for file "straight" LPC method. Speech synthesizers using the method have been built and some details of the data store and the hardware are given.

关键词： Speech synthesis Interpolation Data compression Vocoders Encoding linear predictive coding Communication system control Stress Hardware

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