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.

EVRC-Wideband: The New 3GPP2 Wideband Vocoder Standard

Abstract: In this paper, the latest wideband vocoder standard adopted by the cdma2000 standardization body, 3GPP2, is described. Christened enhanced variable rate codec-wideband (EVRC-WB), the proposed codec encodes wideband speech (16 KHz sampling frequency) at a maximum bit-rate of 8.55 kbit/s. EVRC-WB is based on a split band coding paradigm in which two different coding models are used for the signal components in the low frequency (LF) (0-4 KHz) and the high frequency (HF) (3.5-7 KHz) bands. The coding model used for the former is based on the EVRC-B narrowband (0-4 KHz) codec, modified to encode the LF band signal at a maximum bitrate of 7.75 kbit/s. The HF band coding model is a LPC based coding scheme where the excitation is derived from the coded LF band excitation using non-linear processing. Mean opinion scores from 3GPP2 characterization tests are provided to demonstrate that the EVRC-WB codec (8.55 kbit/s, max.) performs statistically significantly better than the adaptive multirate wideband (12.65 kbit/s, max.).

Audio codec reselection for increased port density

Abstract: An Internet telephony gateway and method for operating a gateway are disclosed. The gateway is designed with a port to support a predefined maximum number of audio data channels. The gateway contains sufficient processing throughput to operate a first, high quality audio codec on a subset of the channels. However, this throughput is sufficient to operate a second, lower quality audio codec on a greater number of the channels, preferably all of them. The first and second codecs are designed to produce compressed audio data streams that are interoperably decompressable. In operation, the gateway host processor assigns new calls to either the first or second codec, depending on the current traffic being handled by the gateway. If new calls would result in the gateway's processing throughput being exceeded, the host processor may reassign a channel from the first codec to the second codec in order to create processing headroom for the addition of a new channel. Because the codecs are interoperably decompressable, no renegotiation need occur with the far end of the communication channel when a codec is reassigned. This gateway offers the potential for high-quality communication over the maximum number of channels possible, with a natural degradation as the gateway reaches its full channel capacity, using modest processing resources.

Digital audio coding for visual communications

Abstract: Current and future visual communications for applications such as broadcasting videotelephony, video- and audiographic-conferencing, and interactive multimedia services assume a substantial audio component. Even text, graphics, fax, still images, email documents, etc. will gain from voice annotation and audio clips. A wide range of speech, wideband speech, and wideband audio coders is available for such applications. In the context of audiovisual communications, the quality of telephone-bandwidth speech is acceptable for some videotelephony and videoconferencing services. Higher bandwidths (wideband speech) may be necessary to improve the intelligibility and naturalness of speech. High quality audio coding including multichannel audio will be necessary in advanced digital TV and multimedia services. This paper explains basic approaches to speech, wideband speech, and audio bit rate compressions in audiovisual communications. These signal classes differ in bandwidth, dynamic range, and in listener expectation of offered quality. It will become obvious that the use of our knowledge of auditory perception helps minimizing perception of coding artifacts and leads to efficient low bit rate coding algorithms which can achieve substantially more compression than was thought possible only a few years ago. The paper concentrates on worldwide source coding standards beneficial for consumers, service providers, and manufacturers. >

Video and aural communications system

Abstract: A dial-up aural and visual communication system includes a telecommunication network with a switch connected thereto, a codec subsystem connected to the switch and video equipment connected to the switch via the codec subsystem with voice communication equipment connected directly to the switch. The codec subsystem is shared between several end users and can be located near the switch. Sharing the codec reduces cost and amount of equipment at end users desk. The codec subsystem can also switch video, including composite video, between local lines and can include frame and image storage. The codec can transmit at 9.6 kbps, p×64 kbps, and via ISDN. The system can be used for video conferencing, remote surveillance or desk-top services, and can include an image grooming system. Images may be stored in switch facilities traditionally used for voice mail.

Extended AMR-WB for high-quality audio on mobile devices

Abstract: This article presents the architecture, performance, and application scenarios of the AMR-WB+ (extended AMR-WB) audio codec, which provides high quality at exceptionally low rates, and consistent quality over all audio types. This codec was recently selected by 3GPP and DVB to support low-bit-rate audio and audiovisual applications on mobile networks