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Proceedings ArticleDOI

Lowdelay combined wideband speech and audio coding with scalable bit rates

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TLDR
In this paper, a single coding algorithm based on backward adaptive linear predictive coding (BA LPC) is proposed for compressing wideband speech and audio signals (0-8kHz) operating with scalable bit rates and low delay.
Abstract
The integration of fixed and wireless networks have expanded the range in which speech and audio coders were designed to operate. As a result, current research is focusing on a number of developments including algorithms which are able to adapt to different transmission environments and to operate under multiple constraints of bit rate, complexity, delay, robustness to bit errors and diversity of input signals. In this paper, we propose a single coding algorithm for compressing wideband speech and audio signals (0-8kHz) operating with scalable bit rates and with low delay. The algorithm is based on the backward-adaptive linear predictive coding (BA LPC) technique in conjunction with an efficient closedloop optimised excitation structure consisting of sparse pulses of ternary values. The output bit rates range from 17 to 68kb/s. The scalability feature is achieved by means of discrete quantisation layers representing various levels of enhancements of the base-line coder and also flexibility in terms of complexity and bit allocation requirements depending on the particular application and on the network resources. An evaluation of the performance of the coder operating at 17kb/s is carried out using the G.722 standard as a reference.

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Citations
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Proceedings ArticleDOI

A multi-stage Levinson-Durbin algorithm

R. Yu, +2 more
TL;DR: This paper proposes an adaptive multi-stage Levinson-Durbin algorithm, which is more numerically robust than the conventional Levison-Durstin algorithm for input signals with high spectral dynamics such as speech or audio signals and can be used in practical linear prediction coding systems for better coding performance.
References
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Journal ArticleDOI

Perceptual coding of digital audio

TL;DR: This paper reviews methodologies that achieve perceptually transparent coding of FM- and CD-quality audio signals, including algorithms that manipulate transform components, subband signal decompositions, sinusoidal signal components, and linear prediction parameters, as well as hybrid algorithms that make use of more than one signal model.
Journal ArticleDOI

G.722: a new CCITT coding standard for digital transmission of wideband audio signals

TL;DR: A tutorial discussion is provided of the adaptive differential PCM (pulse-code modulation) coding method recommended by the group, which covers the subjective performance tests performed, mode initialization and mode switching, data-speed multiplexing, and communication between narrowband and wideband terminals.
Proceedings ArticleDOI

A wideband speech and audio codec at 16/24/32 kbit/s using hybrid ACELP/TCX techniques

TL;DR: A hybrid ACELP/TCX algorithm for coding speech and music signals at 16, 24, and 32 kbit/s is presented, which switches between algebraic code excited linear prediction (ACELP) and transform coded excitation (TCX) modes on a 20-ms frame basis.
Proceedings ArticleDOI

A 16, 24, 32 kbit/s wideband speech codec based on ATCELP

TL;DR: A combined adaptive transform codec (ATC) and code-excited linear prediction (CELP) algorithm for the compression of wideband (7 kHz) signals is described and a switching scheme between CELP and ATC mode is proposed and a frame erasure concealment technique is proposed.
Book ChapterDOI

Wideband Speech Coding

TL;DR: This chapter contains sections titled: Sub-band-ADPCM Wideband Coding at 64 kbps Wideband Transform-coding at 32 kbps Sub- Band CELP Codecs Fullband Wideband ACELP Coding A Turbo-coded Burst-by-burst AdaptiveWideband Speech Transceiver Turbo-detected Unequal Error Protection Irregular Convolutional Coded AMR-WB Transceivers.