Topic
Code-excited linear prediction
About: Code-excited linear prediction is a research topic. Over the lifetime, 2025 publications have been published within this topic receiving 28633 citations. The topic is also known as: CELP.
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01 Oct 2016TL;DR: The objective of this paper is to implement SPEEX decoding on ARM microprocessor, based on the voice compression algorithm technology of Code Excited Linear Prediction (CELP), which can effectively compress voice and retain the integrity of speech.
Abstract: The objective of this paper is to implement SPEEX decoding on ARM microprocessor. SPEEX [1] is based on the voice compression algorithm technology of Code Excited Linear Prediction (CELP) [2], which can effectively compress voice and retain the integrity of speech. For hardware part, we give up the high-cost, high-power consumption digital signal processor, and select the STM32 series ARM microprocessor produced by STMicroelectronics. Through coding at PC end, Bluetooth wireless transmission to the ARM processor, and SPEEX decoding, the voice is then played back. Finally, voice quality verification is carried out through Perceptual Evaluation of Speech Quality (PESQ).
4 citations
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01 Jan 2003
TL;DR: Comparisons have been performed between the estimated vocal noise and the perceived hoarseness in steady vowel fragments, as well as between the Estimated vocal noise in connected speech and sustained vowels produced by the same speakers.
Abstract: The presentation concerns forward and backward double linear prediction of speech with a view to the characterization of vocal noise due to voice disorders. Bi-directional double linear prediction consists in a conventional short-term prediction followed by a distal inter-cycle prediction that enables removing inter-cycle correlations owing to voicing. The long-term prediction is performed forward and backward. The minimum of the forward and backward prediction error is a cue of vocal noise. The minimum backward and forward prediction error has been calculated for corpora involving connected speech and sustained vowels. Comparisons have been performed between the estimated vocal noise and the perceived hoarseness in steady vowel fragments, as well as between the estimated vocal noise in connected speech and sustained vowels produced by the same speakers.
4 citations
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03 Oct 1996TL;DR: An algorithms for coding 20 Hz-15 kHz speech signals at 64 kbit/s with a very low delay and an algebraic codebook is proposed, allowing an important reduction of coder computational complexity, without decreasing the perceived quality of signals.
Abstract: An algorithm for coding 20 Hz-15 kHz speech signals at 64 kbit/s with a very low delay (frame of 0.16 ms) is presented. To achieve a quality near to transparency, the authors propose adapting the Low-Delay CELP coder to the 15 kHz bandwidth and suggest a new noise shaping method based on a psychoacoustic model. In this way they take advantage of linear predictive coding and masking properties of the human perception system. Finally, an algebraic codebook is proposed, allowing an important reduction of coder computational complexity, without decreasing the perceived quality of signals.
4 citations
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01 Oct 1985
TL;DR: This paper describes two systems that use VQ and transmit intelligible speech in the range of 300 to 600 b/s and presents the quantization algorithms and bit allocation for the two vocoders and compares their performance for varying bit rates and different noisy speech conditions.
Abstract: Vector quantization (VQ) has been used recently for developing vocoders operating below 800 b/s. We describe in this paper two systems that use VQ and transmit intelligible speech in the range of 300 to 600 b/s. The frame vocoder which uses VQ for quantizing the spectral parameters of a single frame of speech was found to be most effective at the higher rate of 600 b/s. The segment vocoder which uses VQ for quantizing the spectral parameters of a sequence of frames yielded better intelligibility at the lower 300 b/s rate. We present the quantization algorithms and bit allocation for the two vocoders and compare their performance for varying bit rates and different noisy speech conditions.
4 citations
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02 May 2001TL;DR: A new spectral modeling approach is introduced to refine the LP coefficients obtained by the autocorrelation method and use the refined parameters for CELP coding with refined LP parameters.
Abstract: During the coding of speech with the code excited linear prediction (CELP) method, the linear prediction (LP) filter coefficients are usually calculated by standard autocorrelation or covariance methods. These methods minimize the mean squared error between the speech signal and predicted value. The perceptual quality of the coded speech is indicated by the spectral distortion measured over a set of discrete frequencies related to the formant of the speech. We introduce a new spectral modeling approach to refine the LP coefficients obtained by the autocorrelation method and use the refined parameters for CELP coding. Simulation results show that enhanced harmonic structure is obtained in the speech spectrum coded by CELP coding with refined LP parameters. Discussions of informal listening tests are also given.
4 citations