Topic
Linear predictive coding
About: Linear predictive coding is a research topic. Over the lifetime, 6565 publications have been published within this topic receiving 142991 citations. The topic is also known as: Linear predictive coding, LPC.
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07 May 1996
TL;DR: A novel multi-pulse excitation signal quantization method is proposed, where the pulse amplitudes are vector-quantized (VQ), which remarkably enhances the performance and drastically reduces the position search complexity.
Abstract: This paper proposes a speech codec, named MP-CELP (multi-pulse-based CELP), with a 10 msec frame length, which has been developed for the GSM EFR (enhanced full-rate) codec standardization. A novel multi-pulse excitation signal quantization method is proposed, where the pulse amplitudes are vector-quantized (VQ). The combination search of the pulse position and the amplitude VQ remarkably enhances the performance. By restricting the pulse positions based on the algebraic-type structure, the search complexity and the bits are reduced. The divided pulse position search drastically reduces the position search complexity. The speech quality for MP-CELP is higher than that for G.728 LD-CELP. MP-CELP also satisfies all the speech quality requirements of the GSM EFR standardization except for the background noise condition.
36 citations
03 Oct 2001
TL;DR: An in-depth look at the influence of different speech and audio codecs on the performance of the continuous speech recognition engine and a new strategy is proposed to cope with degradation due to low bitrate coding.
Abstract: This paper proposes an in-depth look at the influence of different speech and audio codecs on the performance of our continuous speech recognition engine. GSM full rate, G711, G723.1 and MPEG coders are investigated. It is shown that MPEG transcoding degrades the speech recognition performance for low bitrates whereas performance remains acceptable for specialized speech coders like GSM or G711. A new strategy is proposed to cope with degradation due to low bitrate coding. The acoustic models of the speech recognition system are trained with transcoded speech (one acoustic model for each speech/audio codec). First results show that this strategy allows one to recover acceptable performance.
36 citations
21 Oct 2001
TL;DR: It is shown how a parametrization of L stationary sinusoids in the complex ODFT spectrum can lead to the effective subtraction, in the real MDCT spectrum, of 3L spectral lines.
Abstract: Recent research in high-quality audio coding seeks not only improved coding gains but also new functionalities such as easy semantic access to compressed audio material and audio modification in the compressed domain. These objectives imply the decomposition of the audio signal into several components of specific semantic value, such as sinusoidal components, that take advantage of selective coding and parametrization tools. We presume an MDCT based audio coding environment and present a new technique combining spectral envelope normalization with accurate subtraction of sinusoidal components in the MDCT frequency domain. It is shown how a parametrization of L stationary sinusoids in the complex ODFT spectrum can lead to the effective subtraction, in the real MDCT spectrum, of 3L spectral lines. A demonstration of the implementation of the technique is available on the Internet (see http://www.inescn.pt//spl sim/ajf/waspaa01/flattening.html.).
36 citations
Patent•
16 Aug 2000TL;DR: In this article, the authors provided speech coding methods and systems for estimating a plurality of speech parameters of a speech signal for coding the speech signal using one or more speech coding algorithms.
Abstract: There are provided speech coding methods and systems for estimating a plurality of speech parameters of a speech signal for coding the speech signal using one of a plurality of speech coding algorithms, the plurality of speech parameters includes pitch information, the plurality of speech parameters is calculated using a plurality of thresholds. An example method includes estimating a background noise level in the speech signal to determine a signal to noise ratio (SNR) for the speech signal, adjusting one or more of the plurality of thresholds based on the SNR to generate one or more SNR adjusted thresholds, analyzing the speech signal to extract the pitch information using the one or more SNR adjusted thresholds, and repeating the estimating, the adjusting and the analyzing to code the speech signal using one the plurality of speech coding algorithms.
36 citations
TL;DR: The authors propose an adaptively weighted Itakura distortion measure, which they studied its effects on the performance of a conventional dynamic time-warping (DTW)-based speech recognizer in a series of speaker-independent, isolated-digit-recognition experiments.
Abstract: The authors propose an adaptively weighted Itakura distortion measure. They studied its effects on the performance of a conventional dynamic time-warping (DTW)-based speech recognizer in a series of speaker-independent, isolated-digit-recognition experiments. The equivalent SNR improvement achieved by using the proposed weighted Itakura distortion at low SNRs is about 5-7 dB. >
36 citations