This paper describes the application of CVSD with delayed decision to improve the quality of LPC encoded speech processed by a CVSD coder. In such tandem configuration, the peaky LPC waveform tends to cause severe slo...
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This paper describes the application of CVSD with delayed decision to improve the quality of LPC encoded speech processed by a CVSD coder. In such tandem configuration, the peaky LPC waveform tends to cause severe slope overload in the subsequent CVSD coder resulting in muffled and unclear speech. By storing a few input signal samples in advance, the delayed-decision CVSD coder generates a more optimum transmitter bit stream that better describes the incoming LPC waveform yielding higher signal-to-noise ratios in the overall LPC/CVSD tandem speech.
This paper describes an approach to modifying the waveshape of LPC synthetic speech, which involves a combination of spectral flattening prior to estimation of the LPC parameters and non-minimum phase de-emphasis foll...
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This paper describes an approach to modifying the waveshape of LPC synthetic speech, which involves a combination of spectral flattening prior to estimation of the LPC parameters and non-minimum phase de-emphasis following conventional LPC synthesis. Results indicate that significant modifications of the synthetic waveform can be effected.
We present a discussion of two studies on the use of linear predictive coding (LPC) techniques for deriving an objective measure of intelligibility over voice communications channels. The first study is a feasibility ...
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ISBN:
(纸本)9798331500191
We present a discussion of two studies on the use of linear predictive coding (LPC) techniques for deriving an objective measure of intelligibility over voice communications channels. The first study is a feasibility study, and the second study an extension of this. The techniques used in the two are similar, but the detailed differences are significant. The results of the feasibility study support the suitability of LPC techniques for the objective measurement of intelligibility.
The analysis of speech using linear Prediction is reformulated to account for the presence of acoustically added noise and a technique is presented for reducing its effect on parameter estimation. The method, called P...
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The analysis of speech using linear Prediction is reformulated to account for the presence of acoustically added noise and a technique is presented for reducing its effect on parameter estimation. The method, called predictive Noise Cancellation (PNC), modifies the noisy speech autocorrelations using an estimate of present background noise which is adaptively updated from an average all-pole noise spectrum. The all-pole noise spectrum is calculated by averaging autocorrelations during non-speech activity. The method uses procedures which are already available to the LPC analyzer, and thus is well suited for real time analysis of noisy speech. Preliminary results show signal to noise improvements on the order of 10 to 20 db.
Variable data rate LPC speech compression schemes are employed to transmit LPC parameters only when speech characteristics have changed sufficiently since the last transmission, yielding improved speech quality relati...
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Variable data rate LPC speech compression schemes are employed to transmit LPC parameters only when speech characteristics have changed sufficiently since the last transmission, yielding improved speech quality relative to fixed-rate schemes for a given average transmission rate. Transmission of variable-rate LPC speech over fixed-rate channels is accomplished using transmit and receive buffers, with resulting transmission delays. Development of proper buffer control strategy is essential to minimize losses caused by exhausting either buffer, or by corrective actions, namely, forced or suppressed transmission. Certain aspects of such strategy and their impact on speech quality and data rate are discussed for a narrowband (2400 bps) speech transmission system.
This paper discusses the identification of vowels in connected speech for a recognition system in which no prior knowledge about the input speakers is assumed. Portions of vowels in ten sentence utterances produced by...
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This paper discusses the identification of vowels in connected speech for a recognition system in which no prior knowledge about the input speakers is assumed. Portions of vowels in ten sentence utterances produced by male and female speakers were labeled on the basis of both acoustic analysis and listening. Formant frequencies and vocal tract area functions were extracted for these vowel portions by the LPC method. Also, the vocal tract length for each analysis frame was estimated acoustically and utilized to normalize the above parameters. The effectiveness of the normalization was investigated by conducting identification experiments based on reference data obtained from 26 speakers. Several factors affecting the identification are discussed.
We present a new quantization method for coding reflection coefficients in linear predictive coding (LPC) of speech. It employs piece-wise linear quantization and requires statistical properties of the LPC reflection ...
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We present a new quantization method for coding reflection coefficients in linear predictive coding (LPC) of speech. It employs piece-wise linear quantization and requires statistical properties of the LPC reflection coefficients. Although the quantization scheme is based on the density of the frequencies of the coefficient values, it does not neglect the importance of spectral sensitivity. In our informal subjective listening tests it was observed that the quality of synthetic speech with the transmission rate of 2.4 kbits/s coded by the piecewise linear quantization method was equivalent to the quality with the rate of 3 kbits/s coded by a linear quantization method.
A speech recognition system has been implemented which accepts reasonably natural English sentences spoken as isolated words. The major components of the system are a speaker dependent word recognizer and a syntax ana...
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A speech recognition system has been implemented which accepts reasonably natural English sentences spoken as isolated words. The major components of the system are a speaker dependent word recognizer and a syntax analyzer. The set of sentences selected for investigation is intended for use as requests in an automated flight information and reservation system. Results are presented of evaluations for speakers using their own stored reference patterns, the reference patterns of other speakers and reference patterns averaged over several speakers. For speakers using their own reference pattern the median word recognition error rate fell from 11.7% to 0.4% with the use of syntax analysis.
We have been attempting to produce further bandwidth reduction in LPC based analysts-synthesis techniques by used the segmentation and labeling algorithms used in the Harpy and Hearsay-II systems. Preliminary results ...
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We have been attempting to produce further bandwidth reduction in LPC based analysts-synthesis techniques by used the segmentation and labeling algorithms used in the Harpy and Hearsay-II systems. Preliminary results indicate that a factor of 3 to 5 further reduction in bandwidth might be possible using segmentation and labeling in conjunction with LPC vocoders.
A new and unique voice modulation system that utilizes the natural time division and frequency division multiplexed characteristics of voiced and unvoiced sounds in human speech has been developed theoretically and pr...
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A new and unique voice modulation system that utilizes the natural time division and frequency division multiplexed characteristics of voiced and unvoiced sounds in human speech has been developed theoretically and proven experimentally in the laboratory. This system, NBVM (Narrow Band Voice Modulation), allows human speech to be transmitted within a 1500 Hz bandwidth and then on reception demodulated or expanded to a full bandwidth of 3000 Hz. The reduction in bandwidth is an advantage whenever the transmission bandwidth is limited. It may also potentially be used to allow more than one speech signal to be communicated over the same channel where previously only one was possible. The speech quality is nearly comparable to telephone speech and is significantly better than current LPC and vocoder techniques. (A demonstration tape will be played). A possible hard-ware implementation and potential applications for this new technique to alleviate spectral crowding for Amateur and CB users are discussed.
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