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.

Invariant codebook fast search algorithm for speech coding

Abstract: The invention is a kind of quick searching algorithm of fixed code book for voice encoding, it carries on four times iterative circular search in full speed, acquires the best code vector after each time of iterative circular search, the four best code vectors are synthesized into four synthesis voices, thus the best code vector can be acquired by judging that the weight mean square error between the four synthesis voice and the original voice is minimal. The steps of each time of circular search are: confirms the position of each pulse of the eight pulses on each tract; confirms a pulse position on one tract, the process doesn't uses iterative circulation. The method reduces the complexity of the EVRC algorithm, the search time be reduced about 25.8%, the efficiency is upgraded greatly.

A joint encryption-compression codec for speech signals using the ITU-T G.711 standard and chaotic map

Abstract: We propose to incorporate encryption procedure into the lossy compression of voice data PCM(Pulse Code Modulation) based on the A-law approximation quantization. The proposed codec CPCM (Chaotic Pulse Code Modulaion) will join the encryption to the compression of the voice data. This scheme provides the same compression ratio given by the PCM codec, but with an unintelligible content. Comparisons with many used schemes have been made to highlight the proposed method in terms of security and rapidity. CPCM codec can be a better alternative to Compress-then-encrypt classical methods which is a time and resource consuming and non suitable for real-time multimedia secure transmission.

Implementation of ITU-T G. 729 speech codec in IP telephony gateway

Abstract: ITU-T G. 729 is the primarily recommended speech codec by H. 323 standard. This paper describes how to implement G. 729 codec in IP telephony gateway, and goes deep into the programming skills on TMS320C6201 DSP and optimizing methods of program code to reduce the speech processing delay time of G. 729 codec. Due to adopting these optimizing methods and programming skills, we have implemented a high-speed speech codec that can process concurrently 20 voice channels with single TMS320C6201 chip in IP telephony gateway. Finally, the paper analyzes the performance results of ITU-T G. 729 codec based on TMS320C6201.

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.

A novel transcoding algorithm for AMR and EVRC speech codecs via direct parameter transformation

Abstract: A novel transcoding algorithm for the adaptive multi rate (AMR) codec and the enhanced variable rate codec (EVRC) is proposed. In contrast to the conventional tandem transcoding algorithm, the proposed algorithm transcodes the parameters of one codec to the other without synthesizing the speech. The proposed algorithm decodes the parameters of source codec from the input bitstream, and based on frame classification and mode decision, it appropriately transforms the parameters of source codec to those of the target codec in the parametric domain. Finally, the transformed parameters are encoded into a bitstream that is decodable by the target codec. The parameters transcoded by the proposed algorithm are line-spectral pair (LSP), pitch delay, fixed codevector, codebook gains, and frame energy. Evaluation results show that while reducing both the computational complexity and delay by 50%, the proposed algorithm produces speech quality equivalent to that of produced by the tandem transcoding algorithm. The general idea is not restricted to the AMR and EVRC but is applicable to various other code-excited linear prediction (CELP) based codecs.