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.

Audio data compression based on AVS-P10

Abstract: AVS-P10 is the first national standard for mobile audio codec with completely independent intellectual property rights. In view of the current situation of the explosive growth of network audio data, According to the principle of AVS-P10 coding and based on AVS-P10 core encoder to make the algorithm compile. And the algorithm is optimized by the self search cycle optimization scheme. Finally, the third National Graduate smart city technology and creative design contest organized by the audio data to do the experiment. And the experimental results show that the algorithm has achieved some results. At the same time, the compression ratio of the optimized algorithm increases with the same code rate.

Noise robust speech coding at very low bit rates

Abstract: We propose a new method for noise robust encoding of speech at very low bit rates. The method constitutes an extension to common speech-recognition/speech-resynthesis schemes, which have become feasible in recent years due to advances in speech recognition and artificial speech synthesis. Most such methods, however, suffer from a significant performance degradation in acoustic environments with background noise. Our proposed procedure is novel since speech enhancement capabilities are built directly into the coding paradigm. Denoising and coding are accomplished jointly by utilizing a statistical description of the parameter space of an underlying speech model (i.e. speech inventory). We conducted experiments with a dedicated speaker in acoustic environments with a signal-to-noise ratio of 10dB. The proposed method was able to improve the perceptual quality of the encoded speech signal by 30% in PESQ measure at an average rate of just under 1.5 kbit/sec.

A Novel Error Mitigation Scheme Based on Replacement Vectors and FEC Codes for Speech Recovery in Loss-Prone Channels

Abstract: In this paper, we propose an error mitigation scheme which combines two different approaches, a replacement super vector technique which provides replacements to reconstruct both the LPC coefficients and the excitation signal along bursts of lost packets, and a Forward Error Code (FEC) technique in order to minimize the error propagation after the last lost frame. Moreover, this FEC code is embedded into the bitstream in order to avoid the bitrate increment and keep the codec working in a compliant way on clean transmissions. The success of our recovery technique deeply relies on a quantization of the speech parameters (LPC coefficients and the excitation signal), especially in the case of the excitation signal where a modified version of the well-known Linde-Buzo-Gray (LBG) algorithm is applied. The performance of our proposal is evaluated over the AMR codec in terms of speech quality by using the PESQ algorithm. Our proposal achieves a noticeable improvement over the standard AMR legacy codec under adverse channel conditions without incurring neither on high computational costs or delays during the decoding stage nor consuming any additional bitrate.