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.

SILK Speech Codec

Abstract: This document describes SILK, a speech codec for real-time, packet- based voice communications. Targeting a diverse range of operating environments, SILK provides scalability in several dimensions. Four different sampling frequencies are supported for encoding the audio input signal. Adaptation to network characteristics is provided through control of bitrate, packet rate, packet loss resilience and use of discontinuous transmission (DTX). And several different complexity levels let SILK take advantage of available processing power without relying on it. Each of these properties can be adjusted during operation of the codec on a frame-by-frame basis.

Standardisation of the adaptive multi-rate codec

Abstract: European Telecommunication Standards Institute (ETSI) initiated a standardisation program in October 1997 to develop an Adaptive Multi-Rate (AMR) codec for GSM. After two competitive selection phases, ETSI chose in October 1998 a codec developed in collaboration between Ericsson, Nokia, and Siemens. The codec standard was finalised, characterised, and formally approved in ETSI during early 1999. The AMR codec provides the next step of speech quality improvement in GSM after the introduction of Enhanced Full-Rate (EFR) codec in 1996. AMR offers substantial improvement in error robustness by adapting speech and channel coding depending on channel conditions. By switching to operate in the GSM half-rate channel during good channel conditions, AMR provides also channel capacity gain over the EFR codec. In April 1999, the Third Generation Partnership Project (3GPP) adopted the AMR codec as the mandatory speech codec for the third generation WCDMA system.

Real-time MPEG-1 audio coding and decoding on a DSP chip

Abstract: The MPEG-1 audio standard (ISO/IEC 11172-3) establishes guidelines for the compression of high-quality digital audio signals. The standard dictates the function of an encoder/decoder pair (codec), leaving the form intentionally vague to allow for competing implementations. A typical approach to real-time operation is to design an application-specific integrated circuit (ASIC) dedicated to encoding, decoding, or both. We present an alternative codec that makes use of the general-purpose digital signal processing (DSP) chips that are now common in multimedia-capable workstations and personal computers. We discuss how selective optimization of the codec structure allows robust performance using limited resources, highlight some of the problems inherent in translating the abstractions of the standard into assembly code, and point towards further investigations of real-time implementations of communications standards.

An adaptive multi-rate speech coder for digital cellular telephony

Abstract: We have developed an adaptive multi-rate (AMR) speech coder designed to operate under the GSM digital cellular full rate (22.8 kb/s) and half rate (11.4 kb/s) channels and to maintain high quality in the presence of highly varying background noise and channel conditions. Within each total rate, several codec modes with different source/channel bit rate allocations are used. The speech coders in each codec mode are based on the CELP algorithm operating at rates ranging from 11.85 kb/s down to 5.15 kb/s, where the lowest rate coder is a source controlled multi-modal speech coder. The decoders monitor the channel quality at both ends of the wireless link using the soft values for the received bits and assist the base station in selecting the codec mode that is appropriate for a given channel condition. The coder was submitted to the GSM AMR standardization competition and met the qualification requirements in an independent formal MOS test.

Hi-BIN: an alternative approach to wideband speech coding

Abstract: In this paper, an encoding technique called Hi-BIN (High Band Injection), which can be combined with any narrowband coder to achieve good quality wideband speech, is described. The principle behind this technique is to model frequencies above 4 kHz by noise with an appropriate spectral shape. This simple way of injecting synthetic noise in the higher frequencies gives surprisingly good quality when compared to very widely used computationally intensive waveform coding techniques such as CELP. We show that Hi-BIN offers a low bit-rate representation of the higher band and is backwards compatible with existing narrowband speech coding systems.