Nec Corporation

Nippon Electric Co., Ltd.

Other

Tetsu Taguchi

Shigeji Ikeda

Kazuo Ochiai

Tatsukichi Koshio

In a speech signal encoding system comprising a maximum similarity series extracting unit (50) for producing a series of excitation pulses appearing at an equidistant time interval and an identical amplitude, the maximum similarity series extracting unit sums up, as a waveform, a series of autocorrelation coefficients to produce a series of summation result coefficients. The autocorrelation coefficients correspond to polarized pulses which are equal to one another in pulse interval and pulse amplitude. The polarized pulses form a plurality of pulse sequences which have phases different from one another. The maximum similarity series extracting unit extracts a polarity of each of the polarized pulses by the use of a dynamic programming method using a degree of an accumulated similarity as an evaluation measure and selects, as the excitation pulse series, one of the pulse sequences which has a maximum waveform similarity between the summation result coefficient series and a series of cross correlation coefficients.

Low bit rate speech signal transmitting system using an analyzer and

In a multipulse processing device for achieving a high encoding efficiency without using a high sampling frequency for an input signal and with a great degree of freedom given to positions of multipulses, an input speech signal is subjected by an LPC analyzer/processor 3 to LPC analysis of each analysis frame for extraction of LPC coefficients after sampled by an A/D converter 2. Multipulses are retrieved as a result of decision by a multipulse analyzer 20 with a degree of freedom given relative to sampling points of a sampled speech signal supplied through an auditorily weighting filter 4. Encoded by an encoder 41 and together with k parameters used as an example of the LPC coefficients, retrieved multipulses are multiplexed by a multiplexer 42 for delivery to a synthesis side. A multipulse waveform synthesizer 45 synthesizes a waveform by using decoded multipulse data and the LPC coefficients.

Multipulse processing with freedom given to multipulse positions of a

In a speech signal encoding system comprising an analyzer and a synthesizer, the analyzer is supplied with an input analog signal to preliminarily select a sequence of digital signals within an analysis frame, to extract a sequence of excitation pulses, from the analysis frame which is divided into a plurality of time intervals each of which is subdivided into a plurality of phases, correlations are calculated between autocorrelations of impulse responses within the analysis frame and cross correlations between the digital signals and the impulse responses to detect, by a maximum similarity series searching circuit, the sequence of the excitation pulses which has a maximum similarity between the autocorrelation coefficients and the cross correlations. The excitation pulses appear at an equidistant time interval and an identical amplitude and have a selected one of the phases and variable polarities. The excitation pulses are sent to the synthesizer together with a phase signal representative of the selected phase, an amplitude signal determined by the analyzer, and a sequence of LPC parameters calculated in relation to the analysis frame.

In an encoding device (100) operable in response to an input speech signal by means of an adaptive transform coding to produce an output encoded speech signal, the input speech signal is partitioned into data blocks by a partition circuit (113). Each of data blocks is decomposed into a plurality of frequency components by a Fourier transformer (114). A spectral envelope calculator (120) estimates intensity of a spectral envelope of the input speech signal. In cooperation with a scalar spectral calculator (115) and a bit assignment determiner (121), a quantizer (116) quantizes or encodes the frequency components with phase information selectively removed from a part of the frequency components on the basis of the intensity of the spectral envelope. In a decoding device, a phase information assignor assigns pseudo-phase information to each of the frequency components from which the phase information is selectively removed.

Reduction of phase information in coding of speech

A frequency bandwidth of a speech signal is divided into a plurality of partial bandwidths. Formant information is extracted on the basis of LPC information developed for the respective partial bandwidths. At least one partial bandwidth may overlap upon the preceding bandwidth. The boundary frequencies of the partial bandwidths can be determined based on the frequency envelope of the speech signal.

Formant extraction on the basis of LPC information developed for

In a modulated signal transmission system operable in response to an input digital signal of a baseband to produce a modulated signal through a nonlinear amplifier, a nonlinear converter is controlled by a control signal generator to compensate for not only a nonlinear distortion but also a linear distortion. The control signal generator is given a difference signal between a linearly detected signal and the input digital signal to which an estimated linear distortion is added by a linear filter having filter coefficients controlled by a filter coefficient estimator. An additional nonlinear converter is located prior to the linear filter. Alternatively, the linear filter may be located after the nonlinear converter.

Modulated signal transmission system compensated for nonlinear and

A speech signal communication apparatus used in a confidential communication system, which performs blockwise processing to achieve a high level of confidentiality, free from waveform discontinuity at the block boundary in a restored speech signal. This apparatus includes a calculating unit for calculating linear predictive coefficients of the input speech signal and a filter to inversely filter the speech signal by using the linear predictive coefficients. The filter flattens the spectral envelope of the input speech signal and produces a predictive residual signal therefrom. This apparatus also includes a removing unit for adaptively removing a low power frequency component from the frequency components of the predictive residual signal. The removing unit uses the linear predictive coefficients to determine which frequency components are to be removed. The linearly predictive coefficients are then converted into a signal having the removed frequency components therein. This converted signal is combined with the removed predictive residual signal from the removing unit, thereby producing a waveform continuous at block boundaries in a restored speech signal.

Communication apparatus for speech signal

Speech signal encoding at a low bit rate is realized by decreasing the number of arithmetic operations required to determine LPC parameters in a backward processing unit. These parameters are determined and updated by filtering, rather than by impulse response. In this manner, it is not necessary to use previously-calculated parameters in updating.

Multi-pulse type coding system with correlation calculation by

In a data transmission system where a first signal partially leaks as an echo from a first transmission line to a second transmission line through a hybrid circuit to form a mixed signal of the echo and a second signal on the second transmission line, an echo canceller is used for cancelling the echo by producing an echo replica at a transversal filter according to filter coefficients and subtracting the echo replica from the mixed signal. In order to reliably generate the filter coefficient for a reduced time duration, a series of autocorrelation coefficients of the first signal and a series of cross-correlation coefficients between the first and the mixed signals are calculated at calculators and the filter coefficients are determined from both of the autocorrelation and the cross-correlation coefficient series at a coefficient determining circuit. The coefficient determining circuit may be an arithmetic circuit for solving simultaneous linear equations. Another circuit for determining the filter coefficients may be a circuit where a first filter coefficient is determined by detecting the maximum value and the corresponding delay time from the series of cross-correlation coefficients, making a fresh series of cross-correlation coefficients with reference to the maximum values, the delay time and the first filter coefficient, then, determining a second filter coefficient from the fresh series. Then, filter coefficients are determined by repetition of the similar operation.

Echo canceller with means for determining filter coefficients from

A speech processing system such as a variable frame length type vocoder and a pattern matching vocoder of the same type capable of improving the reproduced speech. Representative frames replacing a plurality of frames in a given section are developed from among the frames in the given frame, or the frames in the given frame and the final representative frame developed in the preceding section. First frames to be replaced by the representative frames, and second frames, located between the neighboring different representative frames, which are to be approximated by interpolation between the neighboring different representative frames, are determined under the condition the lengths of the first and second frames be variable. In the pattern matching vocoder, the representative frames are compared with reference pattern frames and the most similar reference pattern frame is selected on the basis of measure which is obtained by summing a time distortion and a quantum distortion caused by the replacement of the frames with the representative frame and the reference pattern frame.

Speech processing system

A linear predictive speech synthesis apparatus which receives a pitch parameter and linear predictive coefficients. The synthesis apparatus includes a device for producing an exciting source signal in response to the pitch parameter, and a device for filtering the exciting source signal in response to the linear predictive coefficients to produce a synthesized speech signal. A cascade frequency characteristic between spectral envelope frequency characteristics of the producing device and the filtering device is designated such as to correspond to a spectral envelope characteristic of an input speech signal.

Linear predictive speech analysis-synthesis apparatus

A pattern matching vocoder includes first and second reference pattern memories, a pattern matching processor, and a frame selector. The first pattern memory stores reference vector patterns clustered by a distribution of the number of times of occurrence for spectral envelope vectors of an input speech signal. The second reference pattern memory stores reference vector patterns clustered by pole frequencies, pole bandwidths and a bandwidth of the input speech signal. The pattern matching processor divides the bandwidth of the speech signal into frequency regions and performs pattern matching using, as spectral envelope vectors, power ratios between the frequency regions. The frame selector performs frame selection using, as an evaluation value, a total distortion consisting of a vector distortion caused by pattern matching and a time distortion caused by frame selection with a DP (Dynamic Programming) scheme.

Pattern matching vocoder

An input speech signal is converted into sampled data with a first sampling frequency in each of a plurality of analysis frames. The sampled data are produced as filtered data through a digital filter having a high cut-off frequency smaller than the highest frequency of the speech signal. The filtered data are decimated into decimated signals which are sampled at a second sampling frequency smaller than the first sampling frequency and which are used to develop multi-pulses representative of an exciting source information of the input speech signal. Each of the analysis frames is divided into a plurality of subframes. At most one multi-pulse is developed in one subframe and the other multi-pulses are subsequently developed for the subframes other than the subframe where the one multi-pulse has been developed.

Multi-pulse coding apparatus with a reduced bit rate

The system utilizes a linear predictive coding (LPC) analyzer, an Attenuator, a line spectrum pair (LSP) analyzer, a reference pattern memory and a pattern matching device. The LPC analyzer derives LPC parameters from an input speech signal. The LPC parameters are attenuated in the attenuator and fed to the LSP analyzer for deriving LSP parameters which are in turn fed to the pattern matching device. The reference pattern memory stores a plurality of reference patterns composed of a sequence of LSP parameters for a variety of predetermined speech samples. The pattern matching device is connected to the LSP analyzer and the reference pattern memory to select the reference pattern which most closely resembles the input pattern from the LSP analyzer and to provide a label code as an output thereof. On the decoding side, a decoder is responsive to the label for generating LPC parameters corresponding to the reference pattern of the label. A residual signal which is also transmitted with the reference label is received and fed with the generated LPC parameters to a synthesis filter for providing a synthesized speech signal which is subsequently converted into an analog signal.

Pattern matching vocoder using LSP parameters

From a speech signal, spectrum information as a plurality of line spectrum data, pitch position data and amplitude data are extracted. Each of the sinusoidal wave signals of different frequencies is allotted to the predetermined line spectrum data. The frequency of the sinusoidal wave signal is changed with the pitch position being the boundary. The plurality of sinusoidal wave signals are added and the added result is modulated by the amplitude data to transmit the modulated signal as the transmission data. The line spectrum data, the pitch position data and amplitude data are extracted from the modulated signal. The replica of the speech is produced on the basis of these extracted data.

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Tokyo, Japan

Tetsu Taguchi