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.

Hybrid codec for video signals

Abstract: Motion compensation can be used for improving the prediction within a hybrid codec. As a rule, great circuit complexity is required for determining the displacement vectors. If the data block to be coded or decoded in the hybrid codec, and a particular search area accommodated in a frame buffer of the hybrid codec are subjected to a coarse adaptive quantization, the mathematical effort for the motion compensation can be considerably reduced in a simple manner.

Error Concealment of EVRC Speech Decoder Using Residual Redundancy

Abstract: In digital mobile communication systems, speech coding is very important to increase the bandwidth efficiency Usually, speech coding algorithms determine speech parameters, which are highly sensitive to transmission errors In this paper, we use the residual redundancy remaining after using the Enhanced Variable Rate Codec (EVRC) algorithm for error concealment Average residual redundancies of the quantized parameters are exploited in the error concealment process as a priori knowledge of the source The simulation results show that the use of error concealment improves the parameter SNRs of these parameters, especially for Pitch delay, Delta delay and Fixed Codebook (FCB) gain The results also show that the more redundancy exists in the encoded parameter, the more improvement we could obtain by using the error concealment scheme

Simulasi global system for mobile communication adaptive multi rate speech codec

Abstract: As the wireless technology continuing developing enormously, within it the development of speech signal compression joining too, which is implemented in digital communication system. Among the aim is privileged to minimize the channel capacity or bandwidth usage and the speed of transmission. So the use of smaller bits, which brings maximised voice quality, are the most important factor to this technology. One of the speech signal compression techniques which is implemented in GSM network is called Adaptive Multi Rate (AMR) speech codec or known as GSM-AMR speech codec. There are 8 (eight) bit rates 12.2, 10.2, 7.95, 7.40, 6.70, 5.90, 5.15 and 4.75 kbps which is used by GSM-AMR. The bit rates have abilities to adapt to the situation and condition of the network and also ability to tailor the operator needs and requirement. The result of a non real-time simulation is shown that GSM-AMR has a good enough quality of speech signal reconstruction with range of SNR between 18.78 and 20.35 dB and also range of MSE between 0.053 and 0.079. The smaller the bitrate, the better the SNR and the MSE. It was also proven with the comparison of original signal and constructed signal which is shown that the reconstructed signal has similarities figure with the original signal. Overall, the performance of the speech codec gives a good speech quality. Keyword: Speech Codec, GSM-AMR, Adaptive Multi Rate, SNR, MSE.

A New Video Codec Based on Compressed Sensing

Abstract: In this paper, we propose a new video codec which combines a new compressed sensing theory and some critical ideas of traditional video codecs. Samples needed in the new codec are fewer than in traditional codecs, and the number of samples can be changed according to the encoding mode. The encoder works as a projector, which projects a high dimensional signal to a low dimensional domain, while the decoder works for the solution of underdetermined equations. The decoder is more complex than the encoder, and the process of decoding is no longer the inverse process of encoding. Compared with traditional codecs, the new codec is simpler, but high compression ratio and good quality of reconstructed video are obtained.

Optimization and Implementation of Speech Codec Based on Symmetric Multi-processor Platform

Abstract: This paper proposes an efficient implementation scheme of the speech codec based on the embedded symmetric multiprocessor data signal processing platform. The base codec uses the G.729 algorithm. To satisfy the requirements of real time applications for efficient processing both audio and video simultaneously on multi-channels, this paper present some implementation techniques to optimize the speech coder. To exploit the characteristics of the embedded symmetric multiprocessor platform, we propose a coarse-grained functional partitioning method to balance the load of the encoder among the cores with small overhead of synchronization. In order to reduce the computational burdens, we propose a fast voice activity detection method. The experimental results show that the proposed scheme improves the performance of the speech codec greatly in terms of computational complexity.