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.

Asynchronous conditional-replenishment video codec for 64-kbits/s channels

Abstract: A low cost and low-bit rate video codec for the transmission of traffic images using 64 kbit/s channels is presented. To minimize hardware complexity and data volume, a predictive coding structure with conditional replenishment has been selected. Input pictures are tessellated into 2-D blocks, compared with a reference picture, and only those blocks with significant differences are transmitted.

A Reconfigurable Multi-Mode Architecture for Reed-Solomon Codec

Abstract: In order to save the hardware cost and reduce the power consuming in Reed-Solomon codec, a reconfigurable multi-mode codec is presented, which provides support to encoder and decoder. The reconfigurable calculation module (RCM) in the codec can be reconfigured to perform as a linear feedback shift register (LFSR), syndrome generator and Chien's search algorithm calculator and can handle variable error correction capability (0

Design for New Kind of Low Rate Speech Codec in VoIP

Abstract: An efficient low rate Speech Codec can get well utilized to solve the problem of network congestion under the circumstance of current network. To the speech characteristic analysed, the voice activity detection (VAD)algorithm that is much fewer operational quantity and very effective is designed based on conjugate structure algebraic code excited linear prediction (CS-ACELP),the Speech Codec in this paper is implemented on the Digital Signal Processor(TMS320VC5410)with low bit rate and good timbre. That will lower down the average bit rate to about 4kb/s. As a result, the demand of real-time full-duplex communication in VoIP is fulfilled.

Suitable packetization interval for using Constrained Energy Lapped Transform (CELT) CODEC in Bi-directional communication over WLAN 802.11g

Abstract: Full audio bandwidth with very low algorithmic delay CODEC is state-of-the-art in CODEC technology. This type of CODEC is expected to support full frequency range for human hearing. It will enable future multipurpose audio applications, especially those which require high quality of audio signal with very low delay. A popular choice for this type of CODEC is the Constrained Energy Lapped Transform (CELT) CODEC since it is an open source and royalty free. This study is concerned with the effect of packetization interval for using CELT CODEC in bidirectional communication over WLAN 802.11g with respect to packet loss and end-to-end delay. Possible bitrates used in CELT were also investigated. The result reveals that the commonly default 20 ms. packetization interval is not always suitable for all network conditions. High packetization interval (more than 60ms) interval are not recommended as packet loss can be quite high and end-to-end delay can exceed the 150 ms. ITU recommended value. Using packetization interval at 30 ms. is recommended, apart from achieving better performance, it allows CODEC bitrate to be varied while packet loss is still acceptable and end-to-end delay is still within the recommended value. A practical application for this study is a synchronous remote music session practice between two persons.

Codec G 723.1 by MATLAB simulation

Abstract: The objective of this research and paper is an introduction to multimedia signals compressions under ISDN. Compression and following expansion, in agreement with standard, will be simulate with the help of programme MATLAB. Author's contribution is codec G.723.1 by MATLAB simulation. The ITU-T block structure is observed. Codec realization in MATLAB is applied on test speech signal and results were indicated in this research and paper. Graphics results are one part of research, in the paper is not enough place for this description. Presentation paper contains results of simulation in MATLAB programme for audiocodec by the recommendation G.723.1. This recommendation is used for ISDN as example and although ISDN is now replaced by xDSL, software base used for source encoding is available also for the future