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K. Jarvinen

Bio: K. Jarvinen is an academic researcher from Université de Sherbrooke. The author has contributed to research in topics: Codec & Adaptive Multi-Rate audio codec. The author has an hindex of 2, co-authored 2 publications receiving 307 citations.

Papers
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Journal ArticleDOI
TL;DR: 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.

312 citations

Proceedings ArticleDOI
06 Oct 2002
TL;DR: The history and performance of the adaptive multi-rate wideband (AMR-WB) speech codec recently selected by the Third Generation Partnership Project (3GPP) for GSM and the third generation mobile communication WCDMA system for providing wideband speech services is given.
Abstract: This paper gives the history and performance of the adaptive multi-rate wideband (AMR-WB) speech codec recently 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. In July 2001, the AMR-WB codec was also selected by ITU-T in the standardization activity for wideband speech coding around 16 kbit/s. 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 3.4 kHz to 7 kHz and gives superior speech quality and voice naturalness compared to 2/sup nd/ and 3/sup rd/ generation mobile communication systems.

14 citations


Cited by
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PatentDOI
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.
Abstract: An aspect of the present invention relates to 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 block a gain is determined from the factor and is applied to the transform coefficients of the block.

243 citations

Journal ArticleDOI
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.
Abstract: Concerns regarding the vulnerability of automatic speaker verification (ASV) technology against spoofing can undermine confidence in its reliability and form a barrier to exploitation. The absence of competitive evaluations and the lack of common datasets has hampered progress in developing effective spoofing countermeasures. This paper describes the ASV Spoofing and Countermeasures (ASVspoof) initiative, which aims to fill this void. Through the provision of a common dataset, protocols, and metrics, ASVspoof promotes a sound research methodology and fosters technological progress. This paper also describes the ASVspoof 2015 dataset, evaluation, and results with detailed analyses. A review of postevaluation studies conducted using the same dataset illustrates the rapid progress stemming from ASVspoof and outlines the need for further investigation. Priority future research directions are presented in the scope of the next ASVspoof evaluation planned for 2017.

177 citations

Proceedings ArticleDOI
19 Apr 2009
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.
Abstract: Traditionally, speech coding and audio coding were separate worlds. Based on different technical approaches and different assumptions about the source signal, neither of the two coding schemes could efficiently represent both speech and music at low bitrates. This paper presents a unified speech and audio codec, which efficiently combines techniques from both worlds. This results in a codec that exhibits consistently high quality for speech, music and mixed audio content. The paper gives an overview of the codec architecture and presents results of formal listening tests comparing this new codec with HE-AAC(v2) and AMR-WB+. This new codec forms the basis of the reference model in the ongoing MPEG standardization activity for Unified Speech and Audio Coding.

108 citations

Proceedings ArticleDOI
19 Apr 2009
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.
Abstract: Today's efficient audio codecs for low bitrate application scenarios often rely on parametric coding of the upper frequency band portion of a signal while the lower frequency band portion of the same is conveyed by a waveform preserving coding method. At the decoder, the upper frequency signal is approximated from the lower frequency data using the upper frequency band parameters. However, commonly used methods of bandwidth extension almost inevitably suffer from a sensation of unpleasant roughness, which is especially present for tonal music items. In this paper we expose the origin of the roughness and propose a bandwidth extension method, which does not introduce roughness into the reconstructed audio signal. A listening test demonstrates the advantage of the proposed method compared to a standard bandwidth extension.

106 citations