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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
01 Jun 1994
TL;DR: Current activity in speech compression is dominated by research and development of a family of techniques commonly described as code-excited linear prediction (CELP) coding, which offer a quality versus bit rate tradeoff that significantly exceeds most prior compression techniques.
Abstract: Speech and audio compression has advanced rapidly in recent years spurred on by cost-effective digital technology and diverse commercial applications. Recent activity in speech compression is dominated by research and development of a family of techniques commonly described as code-excited linear prediction (CELP) coding. These algorithms exploit models of speech production and auditory perception and offer a quality versus bit rate tradeoff that significantly exceeds most prior compression techniques for rates in the range of 4 to 16 kb/s. Techniques have also been emerging in recent years that offer enhanced quality in the neighborhood of 2.4 kb/s over traditional vocoder methods. Wideband audio compression is generally aimed at a quality that is nearly indistinguishable from consumer compact-disc audio. Subband and transform coding methods combined with sophisticated perceptual coding techniques dominate in this arena with nearly transparent quality achieved at bit rates in the neighborhood of 128 kb/s per channel. >

234 citations

Proceedings ArticleDOI
30 Jul 2000
TL;DR: This paper presents a solution to robust watermarking of audio data and reflects the security properties of the technique and shows good robustness of the approach against MP3 compression and other common signal processing manipulations.
Abstract: This paper considers the desired properties and possible applications of audio watermarking algorithms. Special attention is given to statistical methods working in the Fourier domain. It presents a solution to robust watermarking of audio data and reflects the security properties of the technique. Experimental results show good robustness of the approach against MP3 compression and other common signal processing manipulations. Enhancements to the presented methods are discussed.

233 citations

Journal ArticleDOI
TL;DR: Improvement by as much as 71 percentage points was observed for sentence recognition in the presence of a competing voice and the present result strongly suggests that frequency modulation be extracted and encoded to improve cochlear implant performance in realistic listening situations.
Abstract: Different from traditional Fourier analysis, a signal can be decomposed into amplitude and frequency modulation components. The speech processing strategy in most modern cochlear implants only extracts and encodes amplitude modulation in a limited number of frequency bands. While amplitude modulation encoding has allowed cochlear implant users to achieve good speech recognition in quiet, their performance in noise is severely compromised. Here, we propose a novel speech processing strategy that encodes both amplitude and frequency modulations in order to improve cochlear implant performance in noise. By removing the center frequency from the subband signals and additionally limiting the frequency modulation's range and rate, the present strategy transforms the fast-varying temporal fine structure into a slowly varying frequency modulation signal. As a first step, we evaluated the potential contribution of additional frequency modulation to speech recognition in noise via acoustic simulations of the cochlear implant. We found that while amplitude modulation from a limited number of spectral bands is sufficient to support speech recognition in quiet, frequency modulation is needed to support speech recognition in noise. In particular, improvement by as much as 71 percentage points was observed for sentence recognition in the presence of a competing voice. The present result strongly suggests that frequency modulation be extracted and encoded to improve cochlear implant performance in realistic listening situations. We have proposed several implementation methods to stimulate further investigation.

233 citations

Journal ArticleDOI
Y. Medan1, E. Yair1, D. Chazan1
TL;DR: Based on a new similarity model for the voice excitation process, a novel pitch determination procedure is derived that has infinite (super) resolution, better accuracy than the difference limen for F/sub 0/, robustness to noise, reliability, and modest computational complexity.
Abstract: Based on a new similarity model for the voice excitation process, a novel pitch determination procedure is derived. The unique features of the proposed algorithm are infinite (super) resolution, better accuracy than the difference limen for F/sub 0/, robustness to noise, reliability, and modest computational complexity. The algorithm is instrumental to speech processing applications which require pitch synchronous spectral analysis. The computational complexity of the proposed algorithm is well within the capacity of modern digital signal processing (DSP) technology and therefore can be implemented in real time. >

232 citations

Patent
28 Oct 1991
TL;DR: In this paper, a CELP speech processor utilizes an organized, non-overlapping, algebraic codebook containing a predetermined number of vectors, uniformly distributed over a multi-dimensional sphere to generate a remaining speech residual.
Abstract: Apparatus and method for encoding speech using a codebook excited linear predictive (CELP) speech processor and an algebraic codebook for use therewith The CELP speech processor receives a digital speech input representative of human speech and performs linear predictive code analysis and perceptual weighting filtering to produce a short term speech information and a long term speech information The CELP speech processor utilizes an organized, non-overlapping, algebraic codebook containing a predetermined number of vectors, uniformly distributed over a multi-dimensional sphere to generate a remaining speech residual The short term speech information, long term speech information and remaining speech residual are combinable to form a quality reproduction of the digital speech input

230 citations


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