Topic
Adaptive Multi-Rate audio codec
About: Adaptive Multi-Rate audio codec is a research topic. Over the lifetime, 1467 publications have been published within this topic receiving 19736 citations. The topic is also known as: AMR & Adaptive Multi-Rate.
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01 Dec 2015
TL;DR: A frame interleaving algorithm is developed to reorder the stereo video frames into a monocular video, such that the proposed codec can gain advantage from inter-views and temporal correlations to improve its coding performance.
Abstract: Development of stereo video codecs in latest multi-view extension of HEVC (MV-HEVC) with higher compression efficiency has been an active area of research. In this paper, a frame interleaved stereo video coding scheme based on MVHEVC standard codec is proposed. The proposed codec applies a reduced layer approach to encode the frame interleaved stereo sequences. A frame interleaving algorithm is developed to reorder the stereo video frames into a monocular video, such that the proposed codec can gain advantage from inter-views and temporal correlations to improve its coding performance. To evaluate the performance of the proposed codec; three standard multi-view test video sequences, named “Poznan_Street”, “Kendo” and “Newspaper1”, were selected and coded using the proposed codec and the standard MV-HEVC codec at different QPs and bitrates. Experimental results show that the proposed codec gives a significantly higher coding performance to that of the standard MV-HEVC codec at all bitrates.
5 citations
19 Apr 2015
TL;DR: A novel technique is presented to efficiently mix traditional ACELP time domain coding with a frequency domain coding model to improve the quality of generic audio signals coded at low bitrates without additional delay.
Abstract: In this paper a novel technique is presented to efficiently mix traditional ACELP time domain coding with a frequency domain coding model to improve the quality of generic audio signals coded at low bitrates without additional delay. The paper discusses how to integrate parts of a traditional Algebraic Code Excited Linear Prediction (ACELP) speech codec to create a time-domain contribution which coexists with a frequency based coding model. A mechanism to determine the value of the time-domain contribution is proposed and a method is described how the frequency-domain contribution might be added without increasing the overall delay of the codec. The proposed method forms part of the recently standardised 3GPP EVS codec.
5 citations
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
10 May 1992
TL;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
IBM1
TL;DR: It is found that the proposed efficient subband image/video codec performs comparable with the leading techniques like embedded zero-tree wavelet (EZW) coding reported in the literature and is incorporated into a motion compensated video coding algorithm.
Abstract: Subband coding has been successfully used for image and video applications. This paper emphasizes two design issues of a subband codec. Namely, the selection of the filterbank family and the width of zero-zones in zero-zone linear quantizers. It is shown that the shift only filters (multiplierless PR-QMF) performs as good as other popular filterbank families. Finally, the significance of the quantizers' zerozones in the codec's rate-distortion performance is presented. It is found that the proposed efficient subband image/video codec performs comparable with the leading techniques like embedded zero-tree wavelet (EZW) coding reported in the literature. We also incorporated this efficient subband codec into a motion compensated video coding algorithm. It is observed that the superior performance of zero-zone linear quantizers is also valid for subband video coding. The subjective performance of the proposed efficient codec in image and video coding are found consistent with the objective performance results obtained.
5 citations