Showing papers on "Enhanced Variable Rate Codec published in 2014"

Dynamic VoIP codec selection on smartphones

Abstract: Voice over IP (VoIP) applications can choose a plethora of different speech codecs, which differ in bandwidth, listening speech quality, and resilience to quality degradation under packet loss. However, VoIP Codecs also exhibit differences in facets such as computational complexity or traffic generated that impact on the energy consumption of smartphones due to the use of processor. In this work deals with the study of energy consumption differences among VoIP codecs. We compare the execution time required to encode/decode reference conversations. Our results show that computational complexity has a significant impact on battery consumption (a factor of up to 10 was found between different codecs). Based on our results, we provide a ranking of energy efficiency. We also propose a simple algorithm for codec dynamic selection considering the dimensions of quality, energy and bandwidth. Our algorithm reacts to network conditions choosing the codec that provides less battery consumption constrained to user-defined targets for minimum quality and maximum codec bitrate.

Performance Evaluation for Voice over LTE by using G.711 as a Codec

Abstract: LTE is the advanced technology for mobile network that operates completely in packet domain with less component compared to previous technology like HSPA or HSPA+ and it’s provide a very high data throughput in both direction upload or download. This development from network side was flowed by development from user side, actually the mobile phone has been changed to the Smart-Phone with a high number of application that use network technology like Video application, VoIP application, Game. This new technology forces the researcher and the developer to optimize the network settings to have a good performance. This paper perform a study and analysis of the Quality of Experience (QoE) at the end user for Voice over the LTE network (VoLTE), after the implementation of VoLTE by using G.711 as a voice codec and will use IP as a core network, will analyze the network performance in order to sort out if this codec is the most recommended codec for VoLTE environment.as per IUT recommendation. Keywords-component: LTE, QoE, IP, VoIP, VoLTE, G.711.

Performance enhanced scalable wideband speech coding for IP networks

Abstract: The scalable wideband speech coding scheme based on the internet low bitrate codec (iLBC) was previously presented and achieved speech quality equivalent to ITU-T G.729.1 at high bit rates. However, the performance was limited at low bit rates. In this paper, we present various approaches to improve performance especially at low bit rates. In particular, the time-domain bandwidth extension (TDBWE) is employed for higher-band coding, and the efficient coding structure is employed in enhancement layers. The performance evaluation results show that significant improvement is achieved at low bit rates and the proposed codec outperforms G.729.1 at most bit rates.

AVS2 speech and audio coding scheme for high quality at low bitrates

Abstract: With the increasing number of portable and wireless devices and the expanding of wireless network bandwidth, there is a growing demand for high quality and low bitrate speech and audio codec. This paper presents a speech and audio codec, which is based upon AVS P10. In our presented codec, ACELP and TCX technology is adopted, and also presents a hybrid bandwidth extension to coding high band signal. The paper gives an overview of the codec architecture and presents results of formal listening tests, objective test, and complicity of algorithm comparing this new codec with ITU-T G.718 and AVS P10. This result shows a comparable performance and forms the basis of the reference model in the ongoing AVS standardization activity for AVS2 speech and audio coding.

Rate Quality Factor Optimization for VP8 Video Codec for Mobile Communication Applications

Abstract: In this paper, mainly, we are proposing a better rate control algorithm optimization technique; it is applying for VP8 video coding for improving the performance and also quality of video codec data for mainly mobile communication applications. Actually, rate control plays an better role in video coding and transmission to provide the best video quality at the receiver end, and our proposed algorithm technique mainly exploits the existing constant quality control, which is governed by a parameter called quality factor (QF) to give a constant bit rates. For this purpose, a new mathematical model called the rate–quality factor(R–Q′) is derived to generate optimum QF for the current coding frame using the bitrates resulting from the encoding of the previous frame in order to meet the target bitrates. The process of calculating the QF is simple, and further calculation is not required for each coded frame. It also provides the rate control solution for both intra-frame-only and inter-frame coding modes. Our all experimental results show that the proposed scheme generates coding bits very close to target bits and provides improved coding efficiency at low bit rates also introducing simple complexity to measure for the video content. So that, in this proposed method considering previous results and comparing with our results then confirm by comprehensive experiments results.. The quality control parameter can be derived from Lagrangian multiplier and hence can be used in any type of encoder that uses RDO. Experimental results have shown that the proposed algorithm can control the bitrates within 1 % of the target bitrates on average, and it has better bitrates regulation over each GOP than the rate control algorithm of H.264. This is an advantage that is crucial in real-time multimedia data streaming in preventing buffer overflow or underflow. Performance of this video data is analyzed using PSNR and bitrates which needs to calculate once this new technique is implemented on specified plat form.

Study the Voice QoE for Speech Codec in Chinese Environment

Abstract: A lot of study has been done on the voice QoE (Quality of Experience). However, QoE study on the background of Chinese language is scarce. PESQ (Perceptual evaluation of speech quality) is a well known objective method for the voice QoE evaluation, it is approved as ITU-T P.862 Recommendations. This paper research the accuracy of PESQ in evaluating speech codec in Chinese environment. Two 3G speech codecs are chosen, including Adaptive Multi-Rate codec (AMR) and Enhanced Variable Rate Codec (EVRC), which all used in 3G communication systems. In this paper, we firstly construct a Chinese speech database, which is used to the study of voice QoE. Then, do lots of experimental test, to get the accuracy of PESQ for evaluating the two codecs under Chinese database. And analyze the reason of the experiment result. Finally, in order to improve the accuracy of PESQ for speech codec, we proposed the improved of the PESQ algorithm.

Enhanced variable rate code resistant voice end-to-end encryption and decryption method

Abstract: The invention provides an enhanced variable rate code resistant voice end-to-end encryption method, and aims at a voice end-to-end encryption device designed for CDMA2000 mobile phones. The device provides a selectable independent voice encryption hardware module and corresponding voice input/output equipment for the mobile phone so that a voice signal end-to-end encryption/decryption function resisting 8K rate EVRC coding compression is realized. Encryption and decryption equipment is composed of three parts of an FPGA module (1), an A/D conversion module (2) and a power supply management module (3). The FPGA module (1) is a hardware platform for performance of the encryption and decryption process and is the core of the whole device, and an encryption and decryption algorithm stored by the FPGA module (1) is the key of realization of the encryption and decryption process. Analog-to-digital conversion is performed on an encrypted and decrypted EVRC voice signal by the A/D conversion module (2). The power supply management module (3) is used for providing a power supply with the required specific voltage to the whole device.

The method of codec selection in the audio transmission process in ICT systems

Abstract: The object of the invention is a method for selecting a codec which is optimal in terms of the properties of the communication channel in a sound transmission system that uses packet-switched data communications. The method involves continuous measurement of the properties of communication channel in each direction and the selection of a codec optimal for the transmission in a given direction from a set of available codecs.