Topic
Enhanced Variable Rate Codec
About: Enhanced Variable Rate Codec is a research topic. Over the lifetime, 357 publications have been published within this topic receiving 4842 citations. The topic is also known as: EVRC.
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01 Jan 1988
TL;DR: A comparison ofgregated and integrated communications networ, a comparison of local and wide area networks, and an overview of the contents of the thesis.
Abstract: CHAPTER 1 : INTRODUCTION 1.1 Background to the thesis 1.1.1 Segregated and integrated communications networ 1.1.2 Local and wide area networks 1.1.3 Problems associated with the addition of voice data LAN 1.1.4 The need for a special speech codec 1.2 Aims of the thesis 1.3 An overview of the thesis contents 1.4 Original contributions made by the thesis 1.5 Publications by the author related to the thesis CHAPTER 2 : THE NETWORK AND WORKSTATIONS 2.
5 citations
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10 May 1992TL;DR: The simulation results show that under the condition that there is no interleaving delay, the resulting codec can provide good quality speech at a channel bit error rate as high as 10/sup -2/.<>
Abstract: The performance of an error-protected speech codec for mobile radio applications is investigated. The speech codec is a 4 kb/s variation of the proposed Federal Standard 1016 CELP. The major difference is the use of a trained codebook (as opposed to a stochastic codebook in the proposed standard). The channel codec on the other hand, consists of a bank of rate-compatible punctured Reed-Solomon (RS) codes. The two subsystems are combined in an optimal fashion, according to the sensitivity of the speech elements. This implies that the most sensitive bits are protected by the most powerful RS codes, while the least sensitive bits are (either uncoded or) protected by the least powerful code. The performance of such a combined codec under different sets of system parameters and channel conditions is studied. In all cases, though, the aggregate rate is fixed at about 6.4 kb/s. The simulation results show that under the condition that there is no interleaving delay, the resulting codec can provide good quality speech at a channel bit error rate as high as 10/sup -2/. >
5 citations
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13 Oct 1993TL;DR: A variable rate speech codec with seven operating rates ranging from 400 bit/s to 16 kbit/s and a 10 ms algorithmic delay is presented in this paper.
Abstract: A variable rate speech codec with seven operating rates ranging from 400 bit/s to 16 kbit/s and a 10 ms algorithmic delay is presented in this paper. The current rate is chosen according to an open-loop speech classification followed by a closed-loop quality evaluation. The rate selection can be source-controlled or network-controlled. The codec is based on a CUP algorithm with a variable number of excitations. The innovations have a deterministic structure allowing a very efficient search procedure and are orthogonalized to the long-term contribution. The algorithm will be implemented in the testbed of one RACE II project: CODIT (COde DIvision Testbed) [1].
5 citations
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18 Nov 2004TL;DR: The adaptive multi-rate (AMR) speech codec offers substantial improvement over previous GSM speech codecs in error robustness by adapting speech and channel coding depending on channel conditions as mentioned in this paper.
Abstract: The adaptive multi-rate (AMR) speech codec offers substantial improvement over previous GSM speech codecs in error robustness by adapting speech and channel coding depending on channel conditions In GSM AMR, the trade-off between speech quality and average bit rate can be further improved by source signal based rate adaptation (SBRA) Together with fast power control, SBRA GSM AMR can be used as a variable rate codec, bringing reduced average bit rate and contributing to an increase in system capacity SBRA GSM AMR was tested against a currently standardised SMV (selectable mode vocoder) variable rate speech codec The paper also presents the general descriptions of both SBRA GSM AMR and SMV codecs
5 citations
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07 May 2001TL;DR: An integrated acoustic echo and noise cancellation system for hands-free telephony is presented that includes a new residual echo cancellation scheme based on spectral analysis and a new double-talk detector suitable for real-time implementation.
Abstract: An integrated acoustic echo and noise cancellation system for hands-free telephony is presented. The proposed system includes a new residual echo cancellation scheme based on spectral analysis and a new double-talk detector suitable for real-time implementation. Residual echo is whitened via AR analysis during no near-end-talk period and is cancelled by noise reduction. Removing speech characteristics of the residual echo signal, noise reduction successfully reduces the power of the residual echo as well as the ambient noise. For further integration with commercial low-bit rate speech coders, noise reduction in IS-127 (EVRC) was considered. For the hands-free situation in the moving car, the proposed system attenuated the interferences more than 30 dB at a constant speed of 90 km/h. The proposed system was implemented on a low-cost DSP with 16-bit fixed-point arithmetic.
5 citations