Speech Coding System Research Articles

Analytical Model for Estimating the Bounds in Predictive Entropy Coding of Digital Speech Signals

A detailed analysis of estimating the upper bounds in predictive entropy coding of digital speech signal is carried out in this paper. Lossless speech compression is exploited to measure the useful information content of the data. The bit rate achieved by the reversible compression takes into account both the contribution of the observation noise i.e. information regarding to statistical uncertainty, which is not relevant to the user, and the intrinsic information of hypothetically noise-free speech. An entropy model of the speech source is defined in order to estimate the upper bound on SQNR gain in entropy coding of digital speech signals. The gain is calculated for speech predictive coding system with non-adaptive quantizer for white and correlated noise, respectively.

Unsupervised noise reduction scheme for voice-based information retrieval in mobile environments

This study proposes an unsupervised noise reduction scheme that improves the performance of voice-based information retrieval tasks in mobile environments. Various types of noises could interfere with speech processing tasks, and noise reduction has become an essential technique in this field. In particular, noise reduction needs to be carefully processed in mobile environments based on the speech coding system and the client-server architecture. In this study, we propose an effective noise reduction scheme that employs the adaptive comb filtering technique. A way of directly using several codec parameters during the filtering process is also investigated. In particular, we modify the conventional comb filter using line spectral pair parameters. To verify the efficiency of the proposed noise reduction approach, we conducted speech recognition experiments using the Aurora2 database. Our approach provided superior recognition performance under various noise conditions compared to the conventional techniques.

A Study on LMS-MPC Method Considering Low Bit Rate

In a speech coding system using excitation source of voiced and unvoiced, it would be a distortion of speech waveform in case of exist a voiced and an unvoiced consonants in a frame. To solve this problem, this paper present a method of LMS-MPC uses individual pitch and LMS(Least Mean Square). I evaluate the MPC and LMS-MPC using LMS. As a result, SNRseg of LMS-MPC was improved 1.5dB for female voice and 1.3dB for male voice respectively. Compared to the MPC, SNRseg of LMS-MPC has been improved that I was able to control the distortion of the speech waveform finally. And so, I expect to be able to this method for cellular phone and smart phone using excitation source of low bit rate.

Realtime Joint Speech Coding and Transmission Algorithm for High Packet Loss Rate Wireless Channels

In digital speech communication over noisy high packet loss rate wireless channels, improving the overall performance of the realtime speech coding and transmission system is of great importance. A novel joint speech coding and transmission algorithm is proposed by fully exploiting the correlation between speech coding, channel coding and the transmission process. The proposed algorithm requires no algorithm delay and less bandwidth expansion while greatly enhancing the error correcting performance and the reconstructed speech quality compared with conventional algorithms. Simulations show that the residual error rate is reduced by 84.36% and the MOS (Mean Opinion Score) is improved over 38.86%.

Application of the wavelet transform to the low‐bit‐rate speech coding system

AbstractIn the analysis‐synthesis coding of speech signals, realization of high quality in the low‐bit‐rate coding depends on the extraction of its characteristic parameters in the preprocessing. The precise extraction of the fundamental frequency, one of the parameters of the source information, guarantees quality in speech synthesis. But its extraction is difficult because of the influence of the consonant, nonperiodicity of vocal cord vibrations, the wide range of the fundamental frequency, and so on. In this paper we propose a new fundamental frequency extraction with the criterion based on its harmonics structure and low‐bit‐rate speech coding system using the wavelet transform. © 2004 Wiley Periodicals, Inc. Electr Eng Jpn, 148(3): 62–71, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/eej.10309

The Application of The Wavelet Transform to The Low Bit Rate Speech Coding System

In the analysis-synthesis coding of speech signals, realization of the high quality in the low bit rate coding depends on the extraction of its characteristic parameters in the pre-processing. The precise extraction of the fundamental frequency, one of the parameters of the source information, guarantees the quality in the speech synthesis. But its extraction is difficult because of the influence of the consonant, non-periodicity of vocal cords vibration, wide range of the fundamental frequency, etc..In this paper, we will propose a new fundamental frequency extraction with the criterion based on its harmonics structure and low bit rate speech coding system using the Wavelet transform.

최소 자승법을 이용한 TSIUVC 근사합성법에 관한 연구

유성음원과 무성음원을 사용하는 음성부호화 방식에 있어서, 같은 프레임 안에 모음과 무성자음이 있는 경우에 음성 파형에 일그러짐이 나타난다. 본 논문에서는 최소 자승법을 적용한 새로운 TSIVUC 근사합성법을 제시하였다. 이 방법에서 TSlUVC의 추출에는 zero crossing rate과 FIR-STREAK 필터의 잔차신호를 이용한 개별피치 추출 알고리즘을 사용하였다. 실험결과, 최소 자승법을 적용하여 양호한 TSIUVC 근사합성 파형을 얻을 수 있었다. 또한 최대 오타신호에 있어서의 주파수 신호가 일그러짐이 적은 근사합성 파형에 중요한 역할을 한다는 것이다. 이 방법은 음성합성, 음성분석, 새로운 Voiced/Silence/TSIUVC의 음성부호화 방식에 활용할 수 있을 것으로 기대된다. In a speech coding system using excitation source of voiced and unvoiced, it would be involves a distortion of speech waveform in case coexist with a voiced and an unvoiced consonants in a frame. This paper present a new method of TSIUVC (Transition Segment Including Unvoiced Consonant) approximate-synthesis by using Least Mean Square. The TSIUVC extraction is based on a zero crossing rate and IPP (Individual Pitch Pulses) extraction algorithm using residual signal of FIR-STREAK Digital Filter. As a result, This method obtain a high Quality approximation-synthesis waveform by using Least Mean Square. The important thing is that the frequency signals in a maximum error signal can be made with low distortion approximation-synthesis waveform. This method has the capability of being applied to a new speech coding of Voiced/Silence/TSIUVC, speech analysis and speech synthesis.

Complete optimization of excitation and model parameters in parametric speech coders

In a speech coding system, synthesis error (the difference between the original speech at the encoder input and reproduced speech at the decoder output) is a more relevant measure of signal distortion than linear prediction (LP) error. By minimizing the synthesis error instead of the linear prediction error, the analysis and synthesis stages become more compatible. In a previous paper (J. Acoust. Soc. Am. 109, 2491), an Analysis-by-Synthesis (AbS) technique for joint optimization of the excitation and filter parameters in parametric speech coders were presented. Using a gradient search in the root domain, synthesis error is minimized by reoptimizing the filter parameters for a given excitation. This paper provides a major improvement over the previous work, resulting in more than 3 dB of gain in the segmental signal-to-noise ratio. The improvement is due to a new algorithm for computing the gradient vector in the gradient search procedure. The paper describes the new algorithm and report on the improved results. By adding an extra minimization step, this technique can be incorporated into the existing LPC, multipulse LPC, and CELP-type speech coding standards.

Predictive LSF computation

This paper deals with a key component involved in the short time spectrum coding in a speech coding system, namely the computation of line spectral frequencies (LSF). The method consists in: (1) the use of some predictions of the LSF as starting points for the search of the current LSF, (2) the fast bracketing of the LSF restricted on a grid of points and (3) the refinement of LSF by a variable number of bisections. The experimental results show an improvement of the average performance with more than 30% with respect to one frequently used method of LSF computation—the Kabal–Ramachandran method.

A new technique for joint optimization of excitation and model parameters in parametric speech coders

In a speech coding system, synthesis error (the difference between the original speech at the encoder input and the reproduced speech at the decoder output) is a more relevant measure of signal distortion than linear prediction (LP) error. By minimizing the synthesis error instead of the linear prediction error, the analysis and synthesis stages become more compatible. While LP error is linear in filter parameters, synthesis error is a highly nonlinear function of these parameters, making it computationally intractable for real-time applications. This paper presents a computationally feasible solution for minimizing the synthesis error applicable for joint optimization of the excitation and model parameters in real time. Using a gradient search in the root domain, synthesis error is minimized by re-optimizing the filter parameters for a given excitation. Starting the gradient search from the LPC solution, the resultant synthesis error produced by the proposed technique is guaranteed to be lower than the synthesis error using the LPC filter. By adding an extra minimization step, this technique can be incorporated into any parametric speech coder including LPC, multipulse LPC and CELP speech coders. The new technique and results of application to real speech will be presented.

Combined Speech and Channel Coding for Wireless Communications

In this paper, we propose a combined variable-rate code-excited linearly predictive (QCELP) speech coding and unequal error protection (UEP) channel coding system for wireless communications. In contrast to the conventional schemes, our system employs a concatenated super-imposed rate-compatible punctured convolutional (SI-RCPC) channel coding scheme which can provide UEP with respect to not only the bit-significance of speech packets but also the speech activity and local channel characteristics. Verified by the simulation results, the combined system achieves an average transmission rate less than 8 kb/s as well as an average 2 dB signal-to-noise ratio (SNR) gain over the conventional equal error protection system.

Time-frequency plane wiener filtering for robust processing of speech signals

This paper uses discrete wavelet transforms together with Wiener filtering to enhance noisy speech for further processing. The speech signal is mixed with various levels of white and coloured noises and decomposed into several levels using wavelet analysis. The output signal at each level is processed separately using iterative Wiener filtering. An efficient speech coding system is developed for the enhanced speech bands.

A PATTERN RECOGNITION APPROACH TO ROBUST VOICED/UNVOICED SPEECH CLASSIFICATION USING FUZZY LOGIC

In the field of mobile communications new robust Voiced/Unvoiced (V/UV) classification algorithms are required in that correct voicing detection is a crucial point in the perceived quality and naturalness of a very low bit-rate speech coding system. The paper shows that a valid and more convenient alternative to deal with the problem of voicing decision is to use methodologies like fuzzy logic which are suitable for problems requiring approximate rather than exact solutions, and which can be represented through descriptive or qualitative expressions. The Fuzzy Voicing Detector proposed is based on a pattern recognition approach in which the matching phase is performed using three fuzzy rules. The rules have been obtained using FuGeNeSys, a new hybrid learning tool based on Genetic Algorithm and Neural Networks. The fuzzy classifier is computationally very simple and more efficient than traditional methods, which are affected by misclassification errors, above all in the presence of background noise.

Speech coding system and method using voicing probability determination

A modular system and method is provided for encoding and decoding of speech signals using voicing probability determination. The continuous input speech is divided into time segments of a predetermined length. For each segment the encoder of the system computes the signal pitch and a parameter which is related to the relative content of voiced and unvoiced portions in the spectrum of the signal, which is expressed as a ratio Pv, defined as a voicing probability. The voiced portion of the signal spectrum, as determined by the parameter Pv, is encoded using a set of harmonically related amplitudes corresponding to the estimated pitch. The unvoiced portion of the signal is processed in a separate processing branch which uses a modified linear predictive coding algorithm. Parameters representing both the voiced and the unvoiced portions of a speech segment are combined in data packets for transmission. In the decoder, speech is synthesized from the transmitted parameters representing voiced and unvoiced portions of the speech in a reverse order. Boundary conditions between voiced and unvoiced segments are established to ensure amplitude and phase continuity for improved output speech quality. Perceptually smooth transition between frames is ensured by using an overlap and add method of synthesis. Also disclosed is the use of the system in the generation of a variety of voice effects.

Speech coding and decoding system

Speech coding and decoding system

Speech coding parameter smoothing method

A decoding method and apparatus for speech coding systems which takes into account the fact that the human auditory system is sensitive to changes in signal characteristics. For example, a sustained distortion of the spectral characteristic of reconstructed speech is usually less perceptible than an objectively smaller distortion which changes as a function of time. This property of the auditory system is advantageously exploited in the design of a speech coding system receiver in accordance with the present invention, by selecting the sequence of decoded speech parameter values on a perceptual basis. Illustratively, the sequence of decoded speech parameter values is selected so as to describe a smooth path through the sequence of Voronoi regions. The distance between successive parameter values is advantageously minimized, under the constraint that the resultant parameter values fall within, or nearly within, the appropriate Voronoi regions. In this manner, a smoother trajectory will result, thereby enabling the receiver to produce a perceptually superior reconstructed speech signal.

Digital transmission of acoustic signals over a noisy communication channel

The performance of digital communication over a noisy communication channel is improved. An encoder combines bit modulation with error control encoding to allow the decoder to use the redundancy in the error control codes to detect uncorrectable bit errors. This method improves the efficiency of the communication system since fewer bits are required for error control, leaving more bits available for data. In the context of a speech coding system, speech quality is improved without sacrificing robustness to bit errors. A bit prioritization method further improves performance over noisy channels. Individual bits in a set of quantizer values are arranged according to their sensitivity to bit errors. Error control codes having higher levels of redundancy are used to protect the most sensitive (highest priority) bits, while lower levels of redundancy are used to protest less sensitive bits. This method improves efficiency of the error control system, since only the highest priority data is encoded with the highest levels of redundancy. The effect of uncorrectable bit errors is reduced by adaptively smoothing the spectral parameters in a speech decoder. The amount of smoothing is varied depending upon the number of errors detected during the error control decoding of the received data. More smoothing is used when a large number of errors are detected, thereby reducing the perceived effect of any uncorrectable bit errors which may be present.

Sampling rate versus quantisation in speech coders

The use of Kalman filtering techniques to reconstruct down-sampled speech signals is investigated. We compare a full-sampled, coarsely quantised speech coding system with a down-sampled, finely quantised speech coding system transmitting at the same bit-rate. Testing reveals that the down-sampled system can outperform the full-sampled system. Sample rate, sample accuracy and decoding delay are traded to achieve improvements at a fixed bit-rate.

Quantization of harmonic amplitudes representing speech

In a speech coding and decoding system, in which a timewise segment of an acoustic speech signal is represented by a frame of a data signal characterized by a fundamental frequency and spectral harmonics, a current frame is reconstructed using a set of prediction signals based on the number of spectral harmonics for the current frame and a preceding frame and reconstructed signal parameters characterizing the preceding frame. The number of spectral harmonics for the current and preceding frames are reconstructed from at least a pair of digitally encoded signals that are generated using error protection codes for all of their bits.

Speech coding system having codebook storing differential vectors between each two adjoining code vectors

A speech coding system is provided where input speech is coded by finding via an evaluation computation a code vector giving a minimum error between reproduced signals obtained by linear prediction analysis filter processing, simulating speech path characteristics, on code vectors successively read out from a noise codebook storing a plurality of noise trains as code vectors and an input speech signal and by using a code specifying the code vector. In the speech coding system, the noise codebook includes a delta vector codebook which stores an initial vector and a plurality of delta vectors having difference vectors between adjoining code vectors. In addition, provision is made in the computing unit for the evaluation computation of a cyclic adding unit for cumulatively adding the delta vectors to virtually reproduce the code vectors.