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Patent

Speech analyzer/synthesizer using recursive filters

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TLDR
In this paper, a speech analyzer and synthesizer features a digital adaptive linear predictor, using a recursive (rather than transversal) filter in a negative feedback loop which develops both feedforward and feedback filter coefficients.
Abstract
A speech analyzer and synthesizer features a digital adaptive linear predictor, using a recursive (rather than transversal) filter in a negative feedback loop which develops both feedforward and feedback filter coefficients. An input circuit is responsive to an input speech signal and to a first synthesized speech signal for developing an error signal. An output circuit is responsive to the error signal and to first state signals for developing multiplexed speech data signals. The multiplexed speech data signals are fed back, demultiplexed and applied to a first recursive filter to control the development of the first synthesized speech signal and the first state signals by the first recursive filter. The multiplexed speech data signals from the output circuit are also transmitted to a receiver which demultiplexes and applies the demultiplexed received speech data signals to a second recursive filter to control the development of a second synthesized speech signal by the second recursive filter. This second synthesized speech signal is then converted into an output speech signal which substantially sounds like the input speech signal.

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Citations
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Multi-pulse excitation linear-predictive speech coder

TL;DR: In this paper, a multi-pulse excitation linear-predictive speech coder was proposed, which operates in accordance with an analysis-by-synthesis method for determining the excitation.
PatentDOI

Speech information transmission method and system

TL;DR: In this paper, a speech information communicating method and apparatus for transmitting speech information through a transmission system having a low bit rate transmission capacity such as Telex to a receiver station for reproduction is described.
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Apparatus and method for lip-synching animation

TL;DR: In this article, the authors used linear prediction to obtain speech parameters which can be used to identify phonemes from a limited set corresponding to visually distinctive mouth positions, and then stored the determined representations in time correspondence with the sound track.
PatentDOI

Speech analysis and synthesis apparatus

TL;DR: In this article, a speech analysis and synthesis apparatus has been proposed to quantize the characterizing parameters of a speech signal at mutually different quantization steps depending on whether the speech signal represent a voiced sound or an unvoiced sound, and a synthesis unit adapted to decode the quantization output from the analysis unit to repro-duce the characterising parameters.
Patent

Method and means of determining coefficients for linear predictive coding

TL;DR: In this paper, an improved method and means of determining reflection coefficients that characterize an electrical signal that obtains characteristics of an all-zero inverse lattice filter was proposed, where the reflection coefficients were obtained by filtering the signal, sample the filtered signal, obtaining the elements of a correlation array from the samples, initializing values of arrays forward residuals, backward residuals and cross correlation of residuals.
References
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Patent

Speech analyzer-synthesizer system employing improved formant extractor

TL;DR: In this paper, the lowest three formants of a voiced speech signal are estimated from a smoothed spectral envelope using constraints on formant frequency ranges and relative levels of spectral peaks at the formant frequencies.
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Speech analysis and synthesis by the use of the linear prediction of a speech wave

TL;DR: In this article, a speech wave is represented by the output of a linear filter which simulates an acoustic tube and which is excited by a combination of a quasi-periodic pulse train and white noise.
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Speech signal pitch detector using prediction error data

TL;DR: In this article, the pitch periods in a complex speech signal are determined by evaluating the error in predicting the value of a sample of the signal on the basis of past sample values, and by locating samples for which the prediction error is large.
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Adaptive equalizer for digital data systems

TL;DR: In this paper, the automatic time-domain equalizer is modified to improve its signal-to-noise and impulse-response dispersal performance by supplementing the basic prior art equalizer with a final equalizer whose tap attenuators are controlled by correlations of the output of the basic EQ with a distortion error signal.
Journal ArticleDOI

The digital filter and speech communication

TL;DR: The paper concludes with a cursory look at some of the advantages and disadvantages of the use of digital filters in speech processing systems.