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Journal ArticleDOI

The adaptive multirate wideband speech codec (AMR-WB)

TLDR
In this paper, the adaptive multirate wideband (AMR-WB) speech codec was selected by the Third Generation Partnership Project (3GPP) for GSM and the third generation mobile communication WCDMA system for providing wideband speech services.
Abstract
This paper describes the adaptive multirate wideband (AMR-WB) speech codec selected by the Third Generation Partnership Project (3GPP) for GSM and the third generation mobile communication WCDMA system for providing wideband speech services. The AMR-WB speech codec algorithm was selected in December 2000 and the corresponding specifications were approved in March 2001. The AMR-WB codec was also selected by the International Telecommunication Union-Telecommunication Sector (ITU-T) in July 2001 in the standardization activity for wideband speech coding around 16 kb/s and was approved in January 2002 as Recommendation G.722.2. The adoption of AMR-WB by ITU-T is of significant importance since for the first time the same codec is adopted for wireless as well as wireline services. AMR-WB uses an extended audio bandwidth from 50 Hz to 7 kHz and gives superior speech quality and voice naturalness compared to existing second- and third-generation mobile communication systems. The wideband speech service provided by the AMR-WB codec will give mobile communication speech quality that also substantially exceeds (narrowband) wireline quality. The paper details AMR-WB standardization history, algorithmic description including novel techniques for efficient ACELP wideband speech coding and subjective quality performance of the codec.

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Citations
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PatentDOI

Methods and devices for low-frequency emphasis during audio compression based on acelp/tcx

TL;DR: In this paper, a method for low-frequency emphasizing the spectrum of a sound signal transformed in a frequency domain and comprising transform coefficients grouped in a number of blocks, in which a maximum energy for one block is calculated and a position index of the block with maximum energy is determined, a factor is calculated for each block having a position Index smaller than the position Index of the Block with maximum Energy, and for each blocks a gain is determined from the factor and is applied to the transform coefficients of the blocks.
Journal ArticleDOI

ASVspoof: The Automatic Speaker Verification Spoofing and Countermeasures Challenge

TL;DR: A review of postevaluation studies conducted using the same dataset illustrates the rapid progress stemming from ASVspoof and outlines the need for further investigation.
Proceedings ArticleDOI

Unified speech and audio coding scheme for high quality at low bitrates

TL;DR: This new codec forms the basis of the reference model in the ongoing MPEG standardization activity for Unified Speech and Audio Coding, which results in a codec that exhibits consistently high quality for speech, music and mixed audio content.
Proceedings ArticleDOI

A harmonic bandwidth extension method for audio codecs

TL;DR: This paper exposes the origin of the roughness and proposes a bandwidth extension method, which does not introduce roughness into the reconstructed audio signal, and demonstrates the advantage of the proposed method compared to a standard bandwidth extension.
References
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Journal ArticleDOI

Design and description of CS-ACELP: a toll quality 8 kb/s speech coder

TL;DR: The coder structure is described in detail and the reasons behind certain design choices are discussed and a summary of the subjective test results based on a real-time implementation of this version are presented.
Journal ArticleDOI

A toll quality 8 kb/s speech codec for the personal communications system (PCS)

TL;DR: A toll quality speech codec at 8 kb/s suitable for the future personal communications system and can support a frame erasure rate up to 3% with a degradation in its performance that is still worse than the ITU-T requirements.
Proceedings ArticleDOI

Immittance spectral pairs (ISP) for speech encoding

TL;DR: In quantization experiments ISP has been found to compare favorably with LSP, and a study of interframe differentiation coding for ISP and LSP demonstrates the respective performances of the two sets.
Proceedings ArticleDOI

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

TL;DR: Various approaches for link adaptation with respect to varying radio channel conditions are described and the method of inband signaling that is standardized is discussed and motivated.
Proceedings ArticleDOI

Low-delay code-excited linear-predictive coding of wideband speech at 32 kbps

TL;DR: An enhanced noise weighting technique is proposed and demonstrated its efficiency via subjective listening tests and was essentially equal to that of the 65 kb/s standard (G.722) CCITT wideband coder.