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•
30 Dec 2014
TL;DR: In this article, 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 is presented.
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.
01 Dec 2015
TL;DR: This paper examines a method for controlling the energy of decoded signal at the recovery frame from a packet loss, implemented in the enhanced voice services (EVS) codec which is the latest 3GPP speech and audio codec standard.
Abstract: This paper examines a method for controlling the energy of decoded signal at the recovery frame from a packet loss. Our observation unveiled that a packet loss before speech onset causes sudden increase in the amplitude of the decoded signal at the recovery frame when predictive quantization of line spectral frequency is used. To mitigate the artifact caused by the overshoot, a detector of the overshoot is proposed as well as a method that controls the amplitude of the decoded signal by adjusting distances of adjacent line spectral frequencies. This technology is implemented in the enhanced voice services (EVS) codec which is the latest 3GPP speech and audio codec standard.
01 Jan 2005
TL;DR: An image compression codec based on WHT is proposed and implemented and test results for wireless handsets show that the proposed codec has a better performance than the IJG JPEG codec.
Abstract: An image compression codec based on WHT is proposed and im- plemented in wireless handset. Considering the low processing power of wireless handset, fast decoding codec is proposed and implemented. The proposed codec consists of RCT, WHT transform, quantization using R-D optimization and lossless coding. The test results for wireless handsets show that the proposed codec has a better performance than the IJG JPEG codec.
Patent•
04 May 2016
TL;DR: In this paper, a VOIP (Voice Over Internet Protocol) based device based on a high-pass platform for voice communication is described, in which a function of voice communications is integrated to the AT command of the communication module to be provided for the user to use directly, and meanwhile, voice sampling utilizes the EVRC data with less data to transmit, so that the voice communication more convenient.
Abstract: The invention discloses a voice connection method and device based on a VOIP (Voice Over Internet Protocol) The method comprises the following steps of receiving a server connection command which is initiated by a user CPU (Central Processing Unit) through an AT (Attention) command; establishing a TCP (Transmission Control Protocol) connection used for sending the command or calling and called information and a UDP (User Datagram Protocol) connection used for sending or receiving voice data; sending a calling command to the server through the TCP or receiving called information of the server through a TCP; periodically converting recorded analog voice to EVRC (Enhanced Variable Rate Codec) audio data and sending the EVRC audio data to the server through a UDP; or receiving the EVRC audio data sent by the server through the UDP, decoding the EVRC audio data and converting and outputting the analog voice According to the voice connection method and device based on the VOIP, voice communication is realized on a communication module of a high-pass platform through adopting an EVRC encoding technology, a function of voice communication is integrated to the AT command of the communication module to be provided for the user to use directly, and meanwhile, voice sampling utilizes the EVRC data with less data to transmit, so that the voice communication is more convenient
TL;DR: A real-time MPEG-2 software CODEC for full-duplex transmission applications, and it provides sufficiently good performance for use as a real- time full NTSC-size C ODEC on a PC of at least 1.2-GHz CPU.
Abstract: This paper proposes a real-time MPEG-2 software CODEC for full-duplex transmission applications, and evaluates its performance and usefulness. The CODEC consists of a high-speed encodersdecoder, an IP sendersreceiver, and an error recovery controller. Each encodersdecoder is accelerated and optimized by exploiting fast algorithms and instruction-level parallelism. The IP sendersreceiver combination achieves low delay owing to the direct translating of each elementary stream of video and audio into UDPsIP packets. The error recovery controller carries out simple but powerful error tolerance against packet loss over IP networks. This CODEC attains low delay of 99 ms (M = 1, N = 1) to 165 ms (M = 3, N = 3) including input, encoding, transmitting, decoding, and output delays, and maintains a normal frame rate of 30 fps (frames per second) and more than 20 fps even under a fairly heavy network load. It provides sufficiently good performance for use as a real-time full NTSC-size CODEC on a PC of at least 1.2-GHz CPU. © 2005 Wiley Periodicals, Inc. Syst Comp Jpn, 36(2): 33–41, 2005; Published online in Wiley InterScience (). DOI 10.1002sscj.20151