Coding Of Speech Signals Research Articles

Characteristic Extraction of Speech Signal Using Wavelet.

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 diffcult 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 of the speech signals using the Wavelet transform with the criterion based on its harmonics structure.

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.

Perceptual speech coding and enhancement using frame-synchronized fast wavelet packet transform algorithms

This paper presents new wideband speech coding and integrated speech coding-enhancement systems based on frame-synchronized fast wavelet packet transform algorithms. It also formulates temporal and spectral psychoacoustic models of masking adapted to wavelet packet analysis. The algorithm of the proposed FFT-like overlapped block orthogonal wavelet packet transform permits us to efficiently approximate the auditory critical band decomposition in the time and frequency domains. This allows us to make use of the temporal and spectral masking properties of the human auditory system to decrease the average bit rate of the encoder while perceptually hiding the quantization error. The same wavelet packet representation is used to merge speech enhancement and coding in the context of auditory modeling. The advantage of the method presented in this paper over previous approaches is that perceptual enhancement and coding, which is usually implemented as a cascade of two separate systems, are combined. This leads to a decreased computational load. Experiments show that the proposed wideband coding procedure by itself can achieve transparent coding of speech signals sampled at 16 kHz at an average bit rate of 39.4 kbit/s. The combined speech coding-enhancement procedure achieves higher bit rate values that depend on the residual noise characteristics at the output of the enhancement process.

Speech signal processing circuit and method for decoding a coded speech signal by controlling operation of a band synthesis filter

A speech signal processing circuit includes an input buffer for receiving inverse quantization samples and for temporarily storing those samples. The circuit also includes a band synthesis filter for reading the inverse quantization samples stored in the input buffer one by one, and for conducting quadrature conversion processing and sum-of-product operation processing to decode the samples into speech signals. The circuit further includes a control circuit for controlling operation of the band synthesis filter. When the inverse quantization samples are recognized as being stored in the input buffer, the control circuit controls the band synthesis filter to execute, as an initial operation, the quadrature conversion processing of the inverse quantization samples as many times as a number corresponding to an operation delay time of the band synthesis filter. After completion of the initial operation, the control circuit control the band synthesis filter to continue, as a steady operation, the quadrature conversion processing of the inverse quantization samples stored in the input buffer, as well as to conduct the sum-of-product operation processing with respect to the results of the quadrature conversion processing.

System for predictive coding/decoding of a digital speech signal by embedded-code adaptive transform

A system for predictive coding of a digital speech signal with embedded codes used in any transmission system or for storing speech signals. The coded digital signal (Sn) is formed by a coded speech signal and, if appropriate, by auxiliary data. A perceptual weighting filter is formed by a filter for short-term prediction of the speech signal to be coded, in order to produce a frequency distribution of the quantization noise. A circuit makes it possible to perform the subtraction from the perceptual signal of the contribution of the past excitation signal P0 n to deliver an updated perceptual signal Pn. A long-term prediction circuit is formed, as a closed loop, from a dictionary updated by the modelled page excitation r1 n for the lowest throughput and makes it possible to deliver an optimal waveform and an associated estimated gain which make up the estimated perceptual signal P1 n. An orthonormal transform module includes an adaptive transform module and a module for progressive modelling by orthogonal vectors, thus making it possible to deliver indices representing the coded speech signal. A circuit makes it possible to insert auxiliary data by stealing bits from the coded speech signal. Decoding is performed through extraction of datasignal and transmission of indices representing coded speech signal which is modelled at the minimum throughput.

Digital coding of speech signals using analysis filtering and synthesis filtering

A digital speech encoder is constructed to include a short term analyzer for forming a set of prediction parameters a(i), corresponding to an input speech signal, and an encoder for producing an excitation signal. The encoder includes a plurality of serially coupled coding blocks, wherein each coding block includes an analysis filter, a sample selection block, and a synthesizer filter. The analysis filter outputs speech signal sample values to the sample selection block, which selects and outputs K i sample values representing a selected partial excitation signal. The synthesis filter synthesizes a speech signal corresponding to the selected partial excitation signal output by the selection block and outputs a partial excitation synthesis result to an output of the coding block. At the output of each coding block is a subtractor arranged for subtracting a partial excitation synthesis result that is output from the coding block from the speech signal to obtain a difference signal. The difference signal is coupled to the input of an analysis filter of a next serially coupled coding block. A quantizer is also provided for forming the excitation signal in accordance with all of the partial excitation signals generated by the coding blocks.

Pitch synchronous multi-band (PSMB) coding of speech signals

A novel speech coding algorithm, named pitch synchronous multi-band (PSMB), is proposed. The new coding algorithm uses the multi-band excitation (MBE) model to generate a representative pitch-cycle waveform (PCW) for each frame. The representative PCW of a frame is encoded by two out of three codebooks depending upon whether the frame is related or unrelated to the previous frame. When a frame is related to its previous frame, the PCW is encoded by a length-converted-excitation (LCE) codebook and a stochastic codebook. The codevectors of the LCE codebook are derived from the previous PCW. When a frame is unrelated to its previous frame, it is encoded by a bandlimited single pulse excitation (BSPE) codebook and the stochastic codebook. The new speech coder introduces a pitch-period-based coding feature. It overcomes some weaknesses existing in the improved MBE (IMBE) speech coder. The PSMB coder operating at 4 kbps outperforms the Inmarsat 4.15 kbps IMBE coder by a clear margin. Our listening tests also indicate that it is slightly better than the FS1016 4.8 kbps code excited linear predictive (CELP) coder in terms of perceptual quality. Fast search algorithms for the three codebooks used in PSMB are also developed. The fast algorithms render the new speech coder comparable to the FS1016 CELP coder, in terms of computational complexity.

Method of and device for coding speech signals with analysis-by-synthesis techniques

The set of possible excitation signals is subdivided into a plurality of subsets, the first of which provides the contribution to the coded signal necessary to set up a transmission at a minimum rate guaranteed by the network, while the others supply a contribution which, when added to that of the first subset, causes a rate increase by successive steps. At the receiving side, a decoded signal is generated by using the excitation contribution of the first subset alone if the coded signals are received at the minimum rate, while for rates higher than the minimum rate the contributions of the subsets which have allowed such rate increase are also used.

System for embedded coding of speech signals

The set of possible excitation signals is subdivided into a plurality of subsets, the first of which provides the contribution to the coded signal necessary to set up a transmission at a minimum rate guaranteed by the network, whilst the others supply a contribution which, when added to that of the first subset, causes a rate increase by successive steps. At the receiving side, a decoded signal is generated by using the excitation contribution of the first subset alone if the coded signals are received at the minimum rate, whilst for rates higher than the minimum rate the contributions of the subsets which have allowed such rate increase are also used.

Pitch perception physiology and psychophysics as a basis for the design of pitch detection algorithms

The generation of high‐quality speech with a source‐filter vocoder depends to a very great extent on accurate analysis of source parameters. After decades of research, even state‐of‐the‐art pitch detection algorithms tend to make gross errors in the analysis of signals that present no difficulty for the human listener. In this study a review of a broad range of pitch detection algorithms was undertaken, with particular attention to the plausibility of those algorithms in relation to what is currently known about the psychophysics of pitch perception and the neural coding of speech signals. Our principal conclusion from this review is that the most plausible model is a time‐domain pitch perception scheme proposed more than 4 decades ago by Licklider [J. C. R. Licklider, Experientia 7, 128–133 (1951)], and extended in more recent studies. The implications of these findings for source‐filter vocoders will be discussed, and an implementation of the Licklider model using level‐crossing interval histograms will be described.

Speech signal coding and decoding system transmitting allowance range information

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Speech coding apparatus using multimode coding

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A perceptual study of source coding of Fourier phase and amplitude of the linear predictive coding residual of vowel sounds

A practical question in a Fourier transform coding of speech signals is to what accuracy their amplitude and phase spectra have to be represented without perceptible distortions. In this paper a concern is with the audibility of quantization noise signals which are produced by quantizing the amplitude and phase spectra of vowel sounds. Experiments show that the detection of the noise targets with maskers of a low fundamental frequency is determined mainly by the sharpest temporal resolution of the auditory system in the high-frequency region. For maskers of a high fundamental frequency the detection is determined mainly by the sharpest spectral resolution in the low-frequency region. Noise targets with global random phase and amplitude are relatively more difficult to detect than those produced by a local randomization. Local random phase noise targets are generally more detectable than those produced by local amplitude randomization. The relative importance of phase and magnitude spectra in the Fourier transform coding is strongly dependent on the fundamental frequency of the vowel sounds and the window size used in the short-time Fourier analysis.

Study of Predictive Vector Quantizers for Speech Coding

This paper deals with the application of predictive vector quantization (PVQ) techniques for coding of speech signals at low bit rates. Memoryless vector quantizer (VQ) has also been studied for comparison, Computer aided design of VQ and PVQ systems has been done using the algorithms presented in the literature. The results obtained using computer simulation for coding of sampled speech have been presented and discussed. A comparison between various PVQ design algorithms has been carried out. It is concluded that PVQ system results in significant improvement in signal to quantization noise ratio over memoryless VQ. Steepest Descent Algorithm for PVQ design is easier to use in practice and yields consistent results.

Novel efficient approach for the design of equiripple quadrature mirror filters

Quadrature mirror filters have been used extensively in subband coding of speech signals. The authors introduce a novel efficient approach for the design of equiripple quadrature mirror filters. The new approach is more efficient than the previously proposed design method in terms of computer time and memory requirement.

A Block Adaptive Model for Speech Signals

A new approach to the coding of speech signals is presented. Called the Block Adaptive Coder, it reduces the coder complexity and computational burden and increases the transmission rate of the data. In the classical Adaptive Predictive Coding methods, the predictor coefficients along with the residual signal, are to be transmitted to the receiver. In the present work, this residual signal is not transmitted, which in turn reduces the coder complexity at the transmitter. But, the first few sample values have to be transmitted every block. The predictor is updated, if necessary, based on the correlation between the input samples. The technique is verified by simulation and the results prove the simplicity of the coder as well as the reduction in computation.

Method of and device for speech signal coding and decoding by parameter extraction and vector quantization techniques

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Method of and device for speech signal coding and decoding by subband analysis and vector quantization with dynamic bit allocation

According to the coding method, the digital speech signal is split into subbands; each subband is coded independently of the others. The first two subbands undergo an adaptive linear prediction operation yielding a residual, normalized with respect to a quantized r.m.s. value; the residual samples are quantized and coded by a variable bit number. As to the third subband the samples normalized with respect to a quantized gain term are directly quantized and coded by a variable bit number. While splitting the subband signals into sample blocks, for the first two subbands the coefficient vectors of subband linear prediction filters relevant to each block are also determined by vector quantization and linear prediction inverse filtering operations; quantized r.m.s. values are also determined to calculate the numbers of bits permitting subband signals coding. The coded signal consists of subband signal quantization level indices, of r.m.s. value quantization level indices, and of linear prediction filter coefficient vector indices.

A hybrid transform method for analysis/synthesis of speech

Abstract In this paper, a hybrid transform method for analysis/synthesis and coding of speech signals is proposed. A finite number of Fourier functions is used to represent frame-by-frame the selective (narrow-band) components of speech. The broadband residual is characterized by a small number of Walsh functions. An iterative algorithm, to determine the frequency and sequency components in a systematic manner, is derived. This is employed for speech coding at 16 kbit/s and 9.6 kbit/s. The hybrid transform method produces natural sounding speech which is indistinguishable from the original at 16 kbit/s and highly intelligible at 9.6 kbit/s.

Source coding of speech and video signals

Some digital source coding techniques for speech and video are reviewed. Predictive coding of speech, multipulse and code-excited coders and frequency-domain coders are discussed and compared for the coding of speech signals, and intraframe and still image coding and interframe coding are examined for the coding of image and video signals. The emphasis is on those algorithms that offer high compression while maintaining the perceptual quality of the source signals are discussed. Some algorithms that are general waveform coding algorithms and do not strictly depend on the input source are included. >