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IEEE TRANSACTIONS ON CONSUMER ELECTRONICS 1984年第4期30卷 598-603页

作者： LIN, KS FRANTZ, GA TEXAS INSTRUMENTS INC SEMICOND GRP SPEECH MODEM DIGITAL SIGNAL PROC APPLICAT DALLAS TX 75265 USA

Text-to-speech systems available today generate virtually unlimited speech from a prestored component sounds library, which is frequently created from a male voice. A major reason for that is due to the low pitch profile in the male voice. For example, the speech analysis software and commercial synthesizers seem to work more accurately and better with the male voice than the female voice. To overcome the analysis and synthesis problem on the female voice and to provide the choices of having more than one sex of voices output from a text-to-speech system, some means of voice characteristics conversion is needed. To do so, it is important to first understand what parameters in speech define the perception of different sex. An attempt is underway herein to first study these parameters and then to learn to adjust them so that the voice of one sex can be converted to another or from one voice character to another. The voice characteristics conversion technique described herein is best suited for speech systems either a text-to-speech or analysis-synthesis (meaning record-playback) using LPC synthesizers.

关键词： Speech synthesis linear predictive coding Speech analysis Synthesizers Encoding Instruments Software libraries Psychology Frequency Bandwidth

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Rate Control for Entropy-Coded LPC: Application to Packet-Based Fronthauling

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IEEE TRANSACTIONS ON COMMUNICATIONS 2022年第9期70卷 5898-5911页

作者： Ramalho, Leonardo Fortuna, Joary Lu, Chenguang Berg, Miguel Almeida, Igor Klautau, Aldebaro Fed Univ Para Comp & Telecommun Engn Dept BR-66615170 Belem Para Brazil Ericsson Res S-16480 Kista Sweden Ericsson Res Brazil BR-13337300 Indaiatuba Brazil

linear prediction and entropy coding are widely used in speech, audio, video, and other coding systems. Rate-distortion optimization (RDO) is another relevant coding technique, which is especially useful when the encoder performance depends on tunable parameters from distinct coding subsystems. This work presents a RDO strategy for variable-rate encoders that use linear prediction and entropy coding. The method is suitable when the signal to be compressed is short-term stationary and the prediction order varies over signal segments. The second contribution of this paper is to customize the method to the compression of time-domain OFDM-based cellular radio signals. The method enables a 4G / 5G packet-based fronthaul to flexibly trade-off rate and distortion depending on the amount of information to be transmitted and, consequently, benefit from multiplexing gain. Simulation results with LTE signals show that the proposed method achieves improved performance with a relatively low computational cost, while operating with time-domain signals under different traffic scenarios.

关键词： Quantization (signal) Distortion Loading Entropy Power capacitors Encoding Rate distortion theory Data compression linear predictive coding entropy codes fronthaul

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Vector predictive coding for Very Low-Bit-Rate Speech Encoders

Vector Predictive Coding for Very Low-Bit-Rate Speech Encode...

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MILCOM, Military Communications Conference

作者： Y. J. Liu James D. Mosko Defense Communications Division ITT Nutley NJ USA

The application of adaptive vector predictive coding for very low-bit-rate speech encoders was investigated and the performance quantified. The input speech samples were analyzed using the government standard LPC-10 algorithm to generate a vector with 10 LPC coefficients. This LPC vector was used as the input for vector prediction. Three prediction schemes: time-invariant prediction switched prediction and continuous prediction were investigated. The assumption of independence among the LPC coefficients was also studied. The results indicated the performance of time-invariant and independent prediction schemes is satisfactory for very-low-bit rate speech encoding applications.

关键词： predictive coding Speech coding linear predictive coding Vector quantization Speech analysis Algorithm design and analysis Government Pulse modulation Phase change materials Encoding

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IMPROVEMENT OF THE EXCITATION SOURCE IN THE NARROW-BAND linear PREDICTION VOCODER

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1985年第2期33卷 377-386页

作者： KANG, GS EVERETT, SS Naval Research Laboratory Inc. Washington D.C. DC USA

The major weakness of the current narrow-band LPC synthesizer lies in the use of a "canned" invariant excitation signal, The use of such an excitation signal is based on three primary assumptions, namely, 1) that the amplitude spectrum of the excitation signal is flat and time invariant, 2) that the phase spectrum of the voiced excitation signal is a time-invariant function of frequency, and 3) that the probability density function of the phase spectrum of the unvoiced excitation signal is also time invariant. This paper critically examines these assumptions and presents modifications which improve the quality of the synthesized speech without requiring the transmission of additional data. Diagnostic acceptability measure (DAM) tests show an increase of up to five points in overall speech quality with the implementation of each of these improvements. These modifications can also improve the speech quality of LPC-based speech synthesizers.

关键词： Narrowband Vocoders linear predictive coding Speech synthesis Signal processing Frequency Speech processing Synthesizers Speech analysis Speech enhancement

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TRELLIS EXCITATION SPEECH coding AT LOW BIT RATES

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IEEE TRANSACTIONS ON COMMUNICATIONS 1994年第2-4期42卷 1902-1910页

作者： KANG, SW FISCHER, TR WASHINGTON STATE UNIV SCH ELECT ENGN & COMP SCI DEPT ELECT & COMP ENGN PULLMAN WA 99164 USA

Trellis coded vector quantization (TCVQ) and code-excited linear prediction (CELP) coding are combined to form an efficient low-bit rate speech coding system. The resulting system uses a trellis search to select the synthesis filter excitation sequence, and is referred to as trellis excitation coding (TEC). Simulations are performed for encoding rates of 6.4 and 8 kbps. Informal listening tests indicate that the 8 kbps TEC system has quality roughly between that of 6-bit and 7-bit mu-law PCM with mu = 255. The 6.4 kbps TEC system provides speech quality between 5-bit and 6-bit mu-law PCM. A subjective comparison with vector sum excited linear prediction (VSELP) indicates that the 8 kbps TEC and the VSELP reconstructed speech is about equally preferable.

关键词： Speech coding Bit rate Encoding Vector quantization Phase change materials linear predictive coding Delay Speech synthesis Filters System testing

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Voice quality evaluation in wireless packet communication systems: A tutorial and performance results for ROHC

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IEEE WIRELESS COMMUNICATIONS 2005年第1期12卷 60-67页

作者： Rein, S Fitzek, FHP Reisslein, M Aalborg Univ Dept Commun Technol Aalborg Denmark Arizona State Univ Dept Elect Engn Tempe AZ 85287 USA

As wireless systems evolve toward supporting a wide array of services, including traditional voice service, using packet-switched transport, it becomes increasingly important to assess the impact of packet-switched transport protocols on voice quality. In this article we present a tutorial on voice quality evaluation for wireless packet-switched systems. We introduce an evaluation methodology that combines elementary objective voice quality metrics with a frame synchronization mechanism. ne methodology allows networking researchers to conduct effective and accurate quality evaluation of packet voice. To illustrate the use of the described evaluation methodology and interpretation of the results, we conduct a case study of the impact of robust header compression (ROHC) on the voice quality achieved with real-time transmission of GSM encoded voice over a wireless link.

关键词： Tutorial Wireless communication GSM Decoding linear predictive coding Speech analysis Wireless application protocol Performance gain Internet Multimedia systems

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Vector quantization based on Gaussian mixture models

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 2000年第4期8卷 385-401页

作者： Hedelin, P Skoglund, J Chalmers Dept Signals & Syst Informat Theory Grp SE-41296 Gothenburg Sweden

In this paper, we model the underlying probability density function of vectors in a database as a Gaussian mixture (GM) model. The model is employed for high rate vector quantization analysis and for design of vector quantizers. It is shown that the high rate formulas accurately predict the performance of model-based quantizers. We propose a novel method for optimizing GM model parameters for high rate performance, and an extension to the EM algorithm for densities having bounded support is also presented. The methods are applied to quantization of LPC parameters in speech coding and we present new high rate analysis results for band-limited spectral distortion and outlier statistics. In practical terms, we find that an optimal single-stage VQ can operate at approximately 3 bits less than a state-of-the-art LSF-based 2-split VQ.

关键词： Vector quantization Databases Probability density function predictive models Optimization methods linear predictive coding Speech coding Speech analysis Statistical analysis Gaussian distribution

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Modeling and Reproducing Textile Sensor Noise: Implications for Textile-Compatible Signal Processing Algorithms

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IEEE JOURNAL OF BIOMEDICAL AND HEALTH INFORMATICS 2022年第1期26卷 243-253页

作者： Tian, Yupeng Kabir, Muammar Abdizadeh, Mohammad Poursartip, Behnaz Mahnam, Amin Bhattachan, Presish Eskandarian, Ladan Meghrazi, Milad Alizadeh Mellal, Idir Popovic, Milos R. Lankarany, Milad Univ Hlth Network UHN Krembil Res Inst Toronto ON M5T 1M8 Canada Univ Hlth Network UHN KITE Res Inst Toronto ON M5G 2A2 Canada Univ Toronto Inst Biomed Engn Toronto ON M5S 3G9 Canada Myant Inc Toronto ON M9W 1B6 Canada Univ Waterloo Waterloo ON N2L 3G1 Canada Univ Toronto Dept Mat Sci & Engn Toronto ON M5S 3G9 Canada Univ Toronto Inst Biomed Engn Toronto ON M5S Canada Univ Toronto Dept Physiol Toronto ON M5S 1A8 Canada

Smart textiles provide an opportunity to simultaneously record various electrophysiological signals, e.g., ECG, from the human body in a non-invasive and continuous manner. Accurate processing of ECG signals recorded using textile sensors is challenging due to the very low signal-to-noise ratio (SNR). Signal processing algorithms that can extract ECG signals out of textile-based electrode recordings, despite low SNR are needed. Presently, there are no textile ECG datasets available to develop, test and validate these algorithms. In this paper we attempted to model textile ECG signals by adding the textile sensor noise to open access ECG signals. We employed the linear predictive coding method to model different features of this noise. By approximating the linear predictive coding residual signals using Kernel Density Estimation, an artificial textile ECG noise signal was generated by filtering the residual signal with the linear predictive coding coefficients. The synthetic textile sensor noise was added to the MIT-BIH Arrhythmia Database (MITDB), thus creating Textile-like ECG dataset consisting of 108 trials (30 min each). Furthermore, a Python code for generating textile-like ECG signals with variable SNR was also made available online. Finally, to provide a benchmark for the performance of R-peak detection algorithms on textile ECG, the five common R-peak detection algorithms: Pan & Tompkins, improved Pan & Tompkins (in Biosppy), Hamilton, Engelse, and Khamis, were tested on textile-like MITDB. This work provides an approach to generating noisy textile ECG signals, and facilitating the development, testing, and evaluation of signal processing algorithms for textile ECGs.

关键词： Textiles Electrocardiography Feature extraction Electrodes Motion artifacts Signal processing algorithms Signal to noise ratio Kernel density estimation linear predictive coding R-peak detection textile sensor noise (TSN)

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ARCHITECTURE DESIGN OF A HIGH-QUALITY SPEECH SYNTHESIZER BASED ON THE MULTIPULSE LPC TECHNIQUE

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IEEE JOURNAL ON SELECTED AREAS IN COMMUNICATIONS 1985年第2期3卷 377-383页

作者： SHARMA, R AT and T Bell Laboratories Inc. Murray Hill NJ USA

This paper describes an architectural design of a high-quality speech synthesizer based on the multipulse linear predictive coding technique. The algorithms implemented in the design are derived. The main features of the synthesizer are a multiprocessor architecture, real time operation, \mu-/A -law coded PCM output, flexible system interface, relative ease of programming, high quality of synthesized speech, and variable bit rates. The synthesizer is highly testable as most of the internal resources can be addressed externally through a parallel system-controller port and a set of instructions.

关键词： Speech synthesis Synthesizers linear predictive coding Speech coding Bit rate Phase change materials Pulse modulation Information filtering Information filters Very large scale integration

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On 450-600 b/s Natural Sounding Speech coding

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1993年第2期1卷 207-220页

作者： Cheng, Yan-Ming O'Shaughnessy, Douglas INRS Telecommun Verdun PQ H3E 1H6 Canada

This paper studies algorithms to encode speech with good intelligence and naturalness at very low rates. Naturalness is retained by encoding accurately the speech excitation information from an LPC model. We first briefly describe a glottal ARX technique to model the speech signal for high quality. A large reduction in coding rate Is achieved through short-term temporal compression of the speech and vector quantization. After B brief study of short-term temporal decomposition, application of traditional vector quantization to the temporal decomposition output is discussed with a study of distortion measures and codebook generation. Based on properties of short-term temporal decomposition, we introduce finite-state vector quantization to further decrease the coding rate. We then deal with a problem associated with this technique, estimation of a state transition matrix with incomplete data. The general result is that we can design practical coders operating in a range of 450-600 b/s with a delay of about 200 ms and natural-sounding output speech.

关键词： Speech coding linear predictive coding Speech synthesis Vector quantization Distortion measurement Speech analysis Time measurement Testing Bit rate Encoding

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