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.

Speech signal transmitter wherein coding is maintained during speech pauses despite substantial shut down of the transmitter

Abstract: In so-called Code Excited Linear Prediction (CELP) coding methods for speech signal transmission, a codebook look-up method is used which is very processor-intensive. To conserve power, during speech pauses not only the transmitter but also the speech coder is turned off substantially completely. Consequently, when the speech signal resumes there is a transition interval before the filters of the speech coder become adjusted to full operation. For this reason, according to the invention, the filters are not turned off during speech pauses but are directly driven by codebook excitation vectors which correspond to the speech signal then being processed. As a result, there is a smoother and hardly perceptible transition between background noise and the speech signal when the latter resumes. An artificial background noise is produced in the receiver during speech pauses.

A computationally efficient wavelet transform CELP coder

Abstract: A wavelet-transform-based CELP coder design is presented in this paper for high-quality speech coding at about 4.8 kbits/s. The coder quantizes the second residual using a wavelet transform approach instead of the stochastic-codebook-based vector quantization normally used in CELP coders, including the U.S. Federal Standard FS 1016 coder at 4.8 kbits/s. The wavelet coder improves the computational efficiency for encoding the second residual by requiring only 1.2 MIPS instead of 8.3 MIPS required by FS 1016. Subjective speech quality tests involving pairwise comparisons show that the wavelet coder was preferred 61% of the time over FS 1016. >

Real-time implementation and evaluation of variable rate CELP coders

Abstract: The authors describe the main topics related to a real-time implementation of the Proposed Federal Standard 1016 (PFS-1016) 4800 bps CELP (code excited linear prediction) voice coder together with some alternatives to extend real-time CELP schemes over a wide range of bit rates (8000, 6500, and 2400 bps are considered) Several procedures are evaluated in order to select the options in PFS-1016 that provide the highest quality for an implementation based on a single AT&T DSP32C chip From the basic CELP scheme, higher and lower bit rates are derived after evaluation of the efficiency obtained for the different elements in this speech coding algorithm >

Sinusoidal Speech Coding for Low and Very Low Bit Rate Applications.

Abstract: The past decade has seen a very fast growth of the telecommunications industry. Mobile telephony has evolved from a specialist application to being commonplace and affordable, and is now a mass-market industry. A similar evolution is expected from multimedia communications, where voice, video and data are all to be integrated into one device. These services require a large amount of bandwidth, which is a relatively cheap and expandable resource in wire based fixed networks. However it is at a premium in satellite or cellular radio systems. In order to cope with the growing demand and the increasing number of subscribers, it is necessary to make optimal use of the bandwidth available. This implies using efficient source coding technologies, including speech compression algorithms. Many of the recent cellular radio communication systems have used speech coders based upon the Code Excited Linear Prediction (CELP) model. These provide high speech quality at bit rates of 8 kb/s and above, however this reduces significantly when the bit rate is lowered. Vocoders on the other hand have been used for very low bit rate applications, where they provide low quality speech. This usually restricts their use to specialised applications such as private radio or military use. The aim of the research presented here is to improve the speech quality produced by low bit rate vocoders, ideally bringing it close to that of higher bit rates CELP coders while retaining a low bit rate. In order to achieve this it has been necessary to introduce new and refined parameter estimation and quantisation techniques, which were integrated in an improved vocoder model. The resulting coder was then adapted to a range of low and very low bit rate applications, and submitted as candidates to three major standardisation efforts.

Envelope modeling for speech and audio processing using distribution quantization

Abstract: Envelope models are common in speech and audio processing: for example, linear prediction is used for modeling the spectral envelope of speech, whereas audio coders use scale factor bands for perceptual masking models. In this work we introduce an envelope model called distribution quantizer (DQ), with the objective of combining the accuracy of linear prediction and the flexibility of scale factor bands. We evaluate the performance of envelope models with respect to their ability to reduce entropy as well as their correlation to the original signal magnitude. The experiments show that in terms of entropy, distribution quantization and linear prediction are comparable, whereas for correlation, distribution quantization is better. Furthermore the coefficients of distribution quantization are independent and thus more flexible and easier to quantize than linear predictive coefficients.