The European Telecommunications Standards Institute (ETSI) has just defined an adaptive multi rate (AMR) speech codec standard for the GSM system with a multitude of source and channel coding rates. The standard aims to provide robust high quality speech together with the flexibility to deliver radio network capacity enhancements by means of low bit-rate operation. The codec rates are dynamically selected with respect to the rapidly changing radio conditions and to local capacity requirements. This paper describes various approaches for link adaptation with respect to varying radio channel conditions and puts a focus on the solution in the AMR standard. Moreover the method of inband signaling that is standardized is discussed and motivated.

Concepts and solutions for link adaptation and inband signaling for the GSM AMR speech coding standard

An overall encoding procedure and associated decoding procedure are presented. The encoding procedure involves at least two signal encoding processes operating on signal representations of a set of audio input channels. Local synthesis is used in connection with a first encoding process to generate a locally decoded signal, including a representation of the encoding error of the first encoding process. This locally decoded signal is applied as input to a second encoding process. The overall encoding procedure generates at least two residual encoding error signals from at least one of said encoding processes, including at least said second encoding process. The residual error signals are then subjected to compound residual encoding in a further encoding process, preferably based on correlation between the residual error signals.

Joint enhancement of multi-channel audio

The invention relates to the technical field of audio encoding and/or decoding technologies, and thus concerns an overall encoding procedure and associated decoding procedure. The encoding procedure involves at least two signal encoding processes (S1-S3) operating on signal representations of a set of audio input channels, as well as residual encoding (S7-S8). It also involves a dedicated process (S4-S6) to estimate and encode energies of the audio input channels. Each encoding process is associated with a corresponding decoding process. In the overall decoding procedure the decoded signals from each encoding process are preferably combined such that the output channels are close to the input channels in terms of energy and/or quality. Normally, the combination step also adapts to the possible loss of one or more signal representation in part or in whole, such that the energy and quality is optimized with the signals at hand in the decoder. In this way, the overall quality of the output channels is improved.

Energy conservative multi-channel audio coding

An example of a bandwidth efficient adaptive multi rate (AMR) system for Voice over IP (VoIP) is presented. In VoIP, packet losses cause degradation of the synthesized speech. The distortions may propagate over several consecutive frames, since predictors in the codec exploit inter-frame correlations to gain coding efficiency. To reduce the effects of packet loss, forward error correction (FEC) that adds redundant information to voice packets can be used. However, while FEC can reduce the effects of packet loss, it will increase the amount of bandwidth used by the voice stream, which is not desirable. In this paper we propose FEC methods like partial redundancy, selective redundancy for the most sensitive frames and parameter interpolation in conjunction with AMR codec mode adaptation, which secure the speech quality when using AMR for VoIP without increasing the bandwidth substantially.

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Bandwidth efficient AMR operation for VoIP

In this chapter, we present an overview of the most widely used algorithms, standards, and applications of wideband and narrowband speech coding. Algorithms for speech coding are classified into four broad headings: (1) waveform coding techniques (including PCM, companded PCM, and DPCM), which are typically used for landline telephony, internet telephony, and secure military communications; (2) subband coding, including perceptually transparent multi-rate and embedded coding which is mainly used for internet and digital audio applications; (3) linear predictive analysis by synthesis coding (LPC-AS) algorithms, including multipulse LPC, CELP, SELP, VSELP, and low-delay CELP, which are typically used for digital cellular and telephony; and (4) LPC vocoders, including advanced vocoder algorithms (e.g., MELP, MBE, and PWI) are used for applications such as secure telephony and satellite telephony. Applications in areas such as voiceover IP (VoIP) and digital cellular are emerging and require a speech coder to gracefully adapt to rapidly changing channel conditions—a need that is met by embedded and multirate speech coders associated with joint source-channel coding algorithms. Measures of speech coder perceptual quality include subjective measures of intelligibility (DRT and DALT) and naturalness (MOS and DAM), as well as objective measures such as segmental SNR, Bark spectral distortion, PSQM, and PESQ. Speech coding standards are set by organizations including the ITU (for landline telephony), MPEG (for multimedia applications), ETSI (for European digital cellular), TIA (for U.S. digital cellular), and DDVPC (for United States military applications).


Keywords:

speech coding;
PCM;
subband coding;
CELP;
LPC;
digital cellular;
multimedia;
voiceover IP;
mean opinion score (MOS)

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Speech Coding: Fundamentals and Applications

Adaptive multi-rate (AMR) is an emerging speech service currently being standardized in the ETSI for the GSM system. The new AMR standard will be flexible by adapting the error protection level and the allocated radio resources. A trade-off between speech quality and system capacity can be achieved for a variety of radio channel and operating conditions. The adaptation of the protection level will be fast and speech service specific. Besides the basic source and channel codec for speech signal payload, the AMR system concept further includes channel state tracking and inband transmission of adaptation data. We review the general AMR system concept and discuss the capacity and quality benefits that can be achieved. An example solution for GSM is described including speech coding, channel coding, inband signaling, and the adaptation scheme.

S. Bruhn

Papers

Adaptive multi-rate. A speech service adapted to cellular radio network quality