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 1993TL;DR: This chapter describes “QCELP,” a variable rate speech coder which has been selected as the speech coding algorithm for the TIA North American digital cellular standard based on CDMA technology.
Abstract: Digital cellular telephone systems require efficient encoding of speech to achieve capacity improvements required of the next generation of cellular systems. The use of a variable rate speech coder can reduce the average data rate required to transmit conversational speech by a factor of two or more, while providing many other advantages. This reduction in average data rate leads to a factor of two increase in the capacity of a Code Division Multiple Access, or CDMA, based digital cellular telephone system by decreasing the mutual interference among users. This chapter describes “QCELP,” a variable rate speech coder which has been selected as the speech coding algorithm for the TIA North American digital cellular standard based on CDMA technology.
58 citations
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21 Apr 1997TL;DR: A new approach for optimum estimation of the speech codec parameters is developed, which can be applied to any speech codec standard if bit reliability information is provided by the demodulator or by the channel decoder.
Abstract: In digital mobile communication systems there is the need for reducing the subjective effects of residual bit errors which have not been eliminated by channel decoding by the use of error concealment techniques. Due to the fact that most standards do not specify these algorithms bit exactly, there is room for new solutions to improve the speech quality. This article develops a new approach for optimum estimation of the speech codec parameters. It can be applied to any speech codec standard if bit reliability information is provided by the demodulator (e.g. DECT), or by the channel decoder (e.g. soft-output Viterbi algorithm-SOVA in GSM). The proposed method includes an inherent muting mechanism leading to a graceful degradation of speech quality in case of adverse transmission conditions. Particularly the additional exploitation of the residual source redundancy, i.e. some a priori knowledge about the codec parameters gives a significant enhancement of the output speech quality. In the case of an error free channel, bit exactness as required by the standards can be preserved.
55 citations
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TL;DR: Novel techniques for source-controlled variable-rate wideband speech coding are presented including robust pitch tracking algorithm, coding-mode dependent prediction of linear prediction (LP) filter quantization, and novel frame erasure concealment techniques including supplementary information for reconstruction of lost onsets and improving decoder convergence.
Abstract: This paper presents novel techniques for source-controlled variable-rate wideband speech coding. These techniques have been used in the variable-rate multimode wideband (VMR-WB) speech codec recently selected by the Third-Generation Partnership Project 2 (3GPP2) for wideband (WB) speech telephony, streaming, and multimedia messaging services in the cdma2000 third-generation wireless system. The codec utilizes efficient coding modes optimized for different classes of speech signal including generic coding based on AMR-WB for transients and onsets, voiced coding optimized for stable voiced signals, unvoiced coding optimized for unvoiced segments, and comfort noise generation for inactive segments. Several innovations enable very good performance at average bit rates below 8 kb/s for active speech coding. The article presents an overview of the codec and describes in detail some of the codec novel features: Robust pitch tracking algorithm, coding-mode dependent prediction of linear prediction (LP) filter quantization, and novel frame erasure concealment techniques including supplementary information for reconstruction of lost onsets and improving decoder convergence. Selected results from the Selection and Characterization tests of the codec illustrate its performance
49 citations
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TL;DR: A region-based video codec is presented, which is compatible with the H.263+ standard, and its associated rate control algorithm for low variable-bit-rate (VBR) video, which incorporates traditional block DCT coding as well as object-based coding.
Abstract: This paper presents a region-based video codec, which is compatible with the H.263+ standard, and its associated rate control algorithm for low variable-bit-rate (VBR) video. The proposed region-based coding scheme is a hybrid method that incorporates traditional block DCT coding as well as object-based coding. To achieve this, we adopt H.263+ as the platform, and develop a fast macroblock-based segmentation method to implement the new region-based codec. The associated rate control solution includes rate control in three levels: encoding frame selection, frame-layer rate control and macroblock-layer rate control. The goal is to improve human visual perceptual quality at low bit rates. The efficiency of the proposed rate control algorithm applied to the region-based H.263+ codec is demonstrated via several typical test sequences.
47 citations
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TL;DR: A backward-compatible multichannel audio codec that unifies the above-mentioned conditions: backward compatibility and exploitation of both signal and perceptual redundancies and combines a high audio quality and a low parameter bit rate.
Abstract: We propose in this paper a backward-compatible multichannel audio codec. This codec represents a multichannel audio input signal by a down mix and parametric data. In order to enable backward compatibility, it is necessary to have the possibility of exerting control over the down-mixing procedure. At the same time, in order to achieve a high coding efficiency, both signal and perceptual redundancies should be exploited. In this paper, we describe a codec that unifies the above-mentioned conditions: backward compatibility and exploitation of both signal and perceptual redundancies. The codec combines a high audio quality and a low parameter bit rate. Moreover, its design is flexible, examples of which are the scalability of the audio quality to (in principle) transparency and the possibility to preserve the correlation structure of the original input signals by using synthetic signals. A stereo backward compatible version of the proposed codec is used as a component of the recently standardized MPEG Surround multichannel audio codec.
45 citations