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.

Adaptive rate control algorithm to improve the performance of h.264 video codec in varying network conditions

Abstract: This thesis involves proposing an efficient algorithm, which takes into consideration the NALU (Network abstraction Layer Unit) and tries to improve the H.264 codec performance over a network. Along with the bandwidth instabilities in the Internet and other issues like the packet drop cause the video frames to be dropped, which makes the video, appear distorted. The main idea behind the algorithm is to vary the bitrate by varying the Quantization parameter, both of which are stored in the reference table. The reference table is calculated based on the analysis of the sequences of different video clips. By changing the bitrate based on the bandwidth available the codec performs better and produces good results of (Signal to Noise Ratio)SNR. The research is based on the development of the reference table which when referred helps to change the quantization parameter of the codec for coding a Group of Frames in a video by periodically checking the network statistics. At the cost of reduction in the resolution due to the network conditions the video frame drop is avoided which helps improve the video quality at the decoder.. The entire simulations for this research were carried out using the JM Reference Software H.264 Encoder and decoder Version JM 10.2. Packet drop was simulated by dropping the individual NAL units from the video stream.

Optimization of signal processing system for precision measurements using maximum length sequences

Abstract: The paper describes some new research results from designing a high precision audio measurement system using MLS (Maximum Length Sequence) algorithm. Evaluation version of USB audio measurement interface optimized for MLS signal processing has been presented in the paper. Comparison was subjected to commercial professional audio measurement system. Superior performance of the optimized measurement system has been shown.

An Improved Speech and Channel Coding for GSM System

Abstract: In the paper an improved speech and channel coding/decoding for the GSM system has been described. It enables to decrease the source rate and to increase the channel error protection of the speech and thus to achieve better performance of the reconstructed speech than the original speech and channel coding/decoding in the presence of fading and noise.

An Efficient Transcoding Scheme in Parameters for ITU-T G.729 and G.723.1 Codec

Abstract: We proposed an efficient method of converting speech codec formats between the G.729 and G.723.1 speech codec. The G.729 three frames is converted to one frame of the G.723.1, transcoding is completed through four processes： LSP and pitch conversions are used to linear interpolation processing, respectively, fast adaptive-codebook search processing adopt to predict the range of adaptive-codebook gain in the G.723.1, and a fast stochastic excitation pulses estimates method. Simulation results show that the proposed method is able to reduce the computation complexity about 84.87% with shorter delay than the tandem. And the perceptual speech quality (PESQ) evaluated better than the latter.

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.