Topic
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.
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Patent•
14 Oct 2003
TL;DR: In this article, an AGC (Automatic Gain Control) preprocessing is performed to the audio data having low dynamic range to reduce the number of pauses in the audio signal.
Abstract: Recently, with the wider use of cellular phones, more and more users listen to music via their cellular phones, and thus, the sound quality of music provided via the cellular phones became more critical. Since music signals are encoded by a voice encoding method optimized to human voice signals such as EVRC (Enhanced Variable Rate Coding) in a cellular communication system, the music signals are often distorted by such encoding method, and listeners experience pauses in music caused by such voice-optimized encoding method. To improve the sound quality of music, a method for preprocessing audio data is provided in order to prevent the problem of pause in music signals in a cellular phone. In particular, AGC (Automatic Gain Control) preprocessing is performed to the audio data having low dynamic range. By this method, the number of pauses in music signal is reduced, and the sound quality of the music is improved.
24 citations
07 Oct 2001
TL;DR: Subjective and objective quality evaluation have confirmed that the speech quality produced by the proposed transcoding algorithm was equivalent to, or better than, the tandem coding, while it had shorter processing delay and less computational complexity.
Abstract: Interoperability is one the most important factors for a successful integration of the speech network To operate speech networks employing different speech coders, but integrated as one, bitstreams generated by one coder should be translated seamlessly to those of the other coders Connecting two coders in tandem may be the simplest way to accomplish this However, coders in tandem connection often produce problems such as poor speech quality, high computational load, and additional transmission delay We propose an efficient transcoding algorithm that can provide interoperability to the networks employing ITU-T G7231 and TIA IS-127 EVRC speech coders Subjective and objective quality evaluation have confirmed that the speech quality produced by the proposed transcoding algorithm was equivalent to, or better than the tandem coding, while it had shorter processing delay and less computational complexity
24 citations
TL;DR: The H.264 Baseline Projile encoder and decoder is implemented on an HP IPAQ hx4700 PDA which has a powerful embedded processor PXA27x produced by Intel and the speed of the codec increases significantly after optimization, and in the QCIF (176times144) resolution, the codec can run in real time on the mobile device.
Abstract: This paper introduces the performance and technical features of the new generation international video coding standard H.264, as well as discusses the implementation and optimization of H.264 on mobile device. Our work is to implement H.264 Baseline Projile encoder and decoder on an HP IPAQ hx4700 PDA (Personal Digital Assistant) which has a powerful embedded processor PXA27x produced by Intel. By using the way of system optimization, algorithm optimization and instruction optimization, we have successfully implemented the H.264 codec on mobile device and made great improvement in terms of coding speed with very little performance degradation in terms of PSNR and bit-rate. Experimental results show that the speed of our codec increases significantly after optimization (much more than 25 frames per second). In the QCIF (176times144) resolution, our codec can run in real time on the mobile device. Based on our optimized real-time H.264 codec, a wireless video conferencing system is implemented and is described in this paper.
22 citations
Patent•
09 Jul 2002TL;DR: In this article, a technique to change the codec or MAC (message authentication code) size in a packet security unit for PacketCable communications during real-time voice transmissions is described.
Abstract: A technique is described to change the codec or MAC (message authentication code) size in a packet security unit for PacketCable communications during realtime voice transmissions is described. An algorithm that provides fast RC4 key advancing to prevent MIPS (millions of instruction cycles per second) overflow is used to perform codec or MAC size changes. The invention is performed without changing the keying material, where the sender and receiver must continue the RC4 encryption process from its state prior to the codec or MAC size change. A sender needs to preserve continuity of the timestamp across a codec change, since the timestamp reflects realtime. Changing the codec or MAC size is likely to change the frame parameters. To preserve continuity of the RC4 state and the timestamp across the codec/MAC size change, the sender TX and receiver RX generates a new frame number. The new frame number is applied to the first frame generated by the new codec or MAC size.
22 citations
19 Apr 2015
TL;DR: Comparison to Opus, IETF driven open source codec as well as industry standard voice codecs: 3GPP AMR and AMR-WB, and ITU-T G.718B, G.1C and G.719 as wellAs direct signals at varying bandwidths was made.
Abstract: This paper discusses the voice and audio quality characteristics of EVS, the recently standardized 3GPP codec. Comparison to Opus, IETF driven open source codec as well as industry standard voice codecs: 3GPP AMR and AMR-WB, and ITU-T G.718B, G.722.1C and G.719 as well as direct signals at varying bandwidths was made. Voice and audio quality was evaluated with three subjective listening tests containing clean and noisy speech in Finnish language as well as a mixed condition test containing both speech and music intermixed. Nine-scale subjective mean opinion score was calculated for all tested conditions.
22 citations