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.

Capacity Estimation for Cellular LTE Using AMR Codec with Semi-persistent Scheduling

Abstract: Long-term evolution (LTE) network is a fully IP-based and does not include a circuit-switched domain for voice communication as known from GSM and UMTS networks. Continuous switching from active to inactive state is one of the challenging tasks for VoIP in LTE. Because of these switching and retransmissions, control channels come into function. Control channel is one of the major limitations for capacity in VoIP. In this paper, we analyzed different scheduling techniques to reduce the number of control channels in the network. We estimated the maximum number of LTE users can be supported over enhanced node B (eNodeB) using different bandwidth levels. From the numerical results, we observed that AMR codec with semi-persistent scheduling scheme is utilizing less number of control channels and accommodates more number of users.

Multi-bus architecture for a video codec

Abstract: Embodiments of the present invention relate to a multi-bus architecture within a video codec that discretely and efficiently transports video components within the codec. This multi-bus architecture provides a relatively more efficient transport mechanism because the various buses are designed to specifically address unique characteristics of the video components or parameters being processed within the codec.

A motion vector replenishment video codec for ATM networks

Abstract: A low bit-rate video codec for ATM networks is described. It is based on two-layer coding principles. The base layer comprises the motion vectors plus a strip of interframe coded video data. The remaining video data are coded by a second layer. Transmission of the base layer cells is assumed to be guaranteed. The required guaranteed channel rate can be as low as that needed for a speech signal, such that networks like Orwell can handle them equally. The second layer cells may be lost, if congestion arises. Simulation results demonstrate the performance of the codec for a range of cell loss rates from the second layer.

Algorithm for achieving adaptive tiling of time axis for audio coding purposes

Abstract: A new algorithm for achieving flexible tiling of the time axis for audio coding purposes is presented. It is based on the calculus of the distances among a predetermined number of time-frequency pairs. From the computed distances, a clustering process determines the final subdivision of each audio frame. Experimental results demonstrates the good performance of the proposed algorithm, which provides high coding efficiency with a reduced complexity.

A real-time QoE methodology for AMR codec voice in mobile network

Abstract: This paper studies a general strategy to predict voice Quality of Experience (QoE) for various mobile networks. Particularly, based on data-mining for Adaptive Multi-Rate (AMR) codec voice, a novel QoE assessment methodology is proposed. The proposed algorithm consists of two parts. The first part is devoted to assessing speech quality of fixed rate codec mode (CM) of AMR while in the other one a adaptive rate CM is designed. Measuring basic network parameters that have much impact on speech quality, QoE can be monitored in real time for operators. Meanwhile, based on the measurement data sets from real mobile network, the QoE prediction strategy can be implemented and QoE assessment model for AMR codec voice is trained and tested. Finally, the numerical results suggest that the correlation coefficient between predicted values and true values is greater than 90% and root mean squared error is less than 0.5 for fixed and adaptive rate CM.