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Speech coding

About: Speech coding is a research topic. Over the lifetime, 14245 publications have been published within this topic receiving 271964 citations.


Papers
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Journal ArticleDOI
TL;DR: A low-complexity algorithm for monitoring the speech quality over a network that can be computed from commonly used speech-coding parameters without explicit distortion modeling is described.
Abstract: Monitoring of speech quality in emerging heterogeneous networks is of great interest to network operators. The most efficient way to satisfy such a need is through nonintrusive, objective speech quality assessment. In this paper, we describe a low-complexity algorithm for monitoring the speech quality over a network. The features used in the proposed algorithm can be computed from commonly used speech-coding parameters. Reconstruction and perceptual transformation of the signal is not performed. The critical advantage of the approach lies in generating quality assessment ratings without explicit distortion modeling. The results from the performed experiments indicate that the proposed nonintrusive objective quality measure performs better than the ITU-T P.563 standard

133 citations

Journal ArticleDOI
Willem Bastiaan Kleijn1
TL;DR: The coding method is easily combined with existing LP-based speech coders, such as CELP, for unvoiced signals and excellent voiced speech quality is obtained at rates between 3.0 and 4.0 kb/s.
Abstract: Voiced speech is interpreted as a concentration of slowly evolving pitch-cycle waveforms. This signal can be reconstructed by interpolation from a downsampled sequence of pitch-cycle waveforms with a rate of one prototype waveform per 20-30 ms interval. The prototype waveform is described by a set of linear-prediction (LP) filter coefficients describing the formant structure and a prototype excitation waveform, quantized with analysis-by-synthesis procedures. The speech signal is reconstructed by filtering an excitation signal consisting of the concatenation of (infinitesimal) sections of the instantaneous excitation waveforms. To obtain the correct level of periodicity, the short-term and the long-term correlations between the instantaneous excitation waveforms can be controlled explicitly. Thus, distortions such as noise, reverberation, and buzziness can be prevented. The coding method is easily combined with existing LP-based speech coders, such as CELP, for unvoiced signals. Excellent voiced speech quality is obtained at rates between 3.0 and 4.0 kb/s. >

133 citations

Journal ArticleDOI
M.R. Schroeder1
01 May 1966
TL;DR: Techniques for analysis and synthesis of speech signals are reviewed with emphasis on vocoders and related devices for more efficient transmission and storage of speech.
Abstract: Techniques for analysis and synthesis of speech signals are reviewed with emphasis on vocoders and related devices for more efficient transmission and storage of speech. Selected applications of speech coding methods as sensory aids to the handicapped are described.

133 citations

Patent
21 Apr 1998
TL;DR: In this article, the authors present a method for dynamic adjustment of audio prompts and speech prompts by switching from a foreground state to a background state of a speech interface in response to a users current interaction modality, by selecting alternative states for speech and audio interfaces that represent users needs for speech prompts.
Abstract: Management of speech and audio prompts, and interface presence, in multimodal user interfaces is provided. A communications device having a multimodal user interface including a speech interface, and a non-speech interface, e.g. a graphical or tactile user interface, comprises means for dynamically switching between a background state of the speech interface and a foreground state of the speech interface in accordance with a users input modality choice. Preferably, in the foreground state speech prompts and speech based error recovery are fully implemented and in a background state speech prompts are replaced by earcons, and no speech based error recovery is implemented. Thus there is provided a device which automatically subdue the speech prompts when a user selects a non-speech input/output mechanism. Also provided is a method for dynamic adjustment of audio prompts and speech prompts by switching from a foreground state to a background state of a speech interface in response to a users current interaction modality, by selecting alternative states for speech and audio interfaces that represent users needs for speech prompts. This type of system and method is particularly useful and applicable to hand held Internet access communication devices.

133 citations

Proceedings ArticleDOI
23 May 1989
TL;DR: The colored-noise prefilter greatly enhances the quality and intelligibility of LPC output speech for noisy inputs, and it is demonstrated that such gains are unavailable with white noise assumption Kalman and Wiener filters.
Abstract: A report is presented on experiments using a colored-noise assumption Kalman filter to enhance speech additively contaminated by colored noise, such as helicopter noise and jeep noise, with a particular application to linear predictive coding (LPC) of noisy speech. The results indicate that the colored-noise Kalman filter provides a significant gain in SNR, a clear improvement in the sound spectrogram, and an audible improvement in output speech quality. The authors demonstrate that such gains are unavailable with white noise assumption Kalman and Wiener filters. The colored-noise prefilter greatly enhances the quality and intelligibility of LPC output speech for noisy inputs. >

132 citations


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Performance
Metrics
No. of papers in the topic in previous years
YearPapers
202338
202284
202170
202062
201977
2018108