C. Quinquis

French National Centre for Scientific Research, Lutetia Parisorum, Île-de-France, France

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Publications (13)1.68 Total impact

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    ABSTRACT: Speech and sound codecs subjective assessment requires anchor signals to allow the comparison of results from different laboratories. The anchor signal presently used, the Modulated Noise Reference Unit (MNRU), is based on the hypothesis that coding technique artifact is only due to quantization noise. Earlier work showed that the impairments of speech codecs could be described by a four-dimensional event and two of these dimensions, namely “Muffled” and “Background noise” dimensions, have been already modeled. In this paper, we propose to design anchor signals for the third dimension mostly characterized by “Echo/Reverberation” attribute.
    Signal Processing Conference (EUSIPCO), 2012 Proceedings of the 20th European; 01/2012
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    ABSTRACT: For subjective assessment of speech quality in codecs, a reference impairment system is required to introduce controlled degradations to calibrate subjective evaluation. A reference system provides a convenient means for making meaningful comparisons between subjective test results across laboratories and can be viewed as a scale on which mean opinion scores are projected, this scale being supposed to cover the whole range of quality. Nowadays, standardized anchor systems do not fit any more the degradations brought by the present codecs. This paper aims at offering new reference signals simulating the defaults of codecs currently used on telecommunication networks. Twenty wideband codecs are compared through dissimilarity tests. A multidimensional scaling technique allows us to define a four-dimensional perceptive space that appears stable for male and female talkers. A verbalization task suggests qualifying the degradations perceived by the listeners with the following attributes: muffle, background noise, noise on speech, and hiss, each conveyed by one dimension. These dimensions are correlated with objective measures such as spectral centroid, energy in the silent part in the high frequency sub-band, ratio of brightness between deterministic part and residual part of the signal and spectral correlation coefficient. New reference signals are produced and a phase of validation suggests a perceptive space quite coherent with the original one.
    IEEE Transactions on Audio Speech and Language Processing 08/2011; · 1.68 Impact Factor
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    ABSTRACT: This paper aims to produce a reference system that can simu-late and calibrate degradations of conversational codecs which are currently used on telecommunications networks, for subjective assessment tests of voice quality. At first, 20 wideband codecs are evaluated through subjective tests in order to produce the multidimensional perceptual space un-derlying the perception of current degradations. Then, a ver-balization task helps to identify attribute for each dimension: clear/muffle, background noise, noise on speech and hiss. These dimensions are characterized with correlates such as spectral centroid, energy in the silent part in the high fre-quency sub-band, ratio of brightness between deterministic part and residual part of the signal and correlation coeffi-cient. This 4-dimension perceptual space is stable for male and female talkers. A new reference system is proposed. The phase of validation leads to a perceptual space with five di-mensions including the four previous ones.
    01/2009;
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    ABSTRACT: The introduction of packet switched (PS) networks (e.g., IMS) creates a need to evaluate speech transmission quality when using the 3GPP default speech codecs. 3GPP SA4 has created a work item in 3GPP Release 6 on "Performance characterization of default codecs for PS conversational application". France Telecom R&D and Siemens proposed a test framework consisting of a UMTS simulator for the air interface and an IP network simulator. ITU-T recommendation P.800 (1996) is used for the quality estimation of the transmission. The real-time conversational test results show that the AMR-NB (adaptive multirate narrow band) and AMR-WB (AMR wide band) speech codecs are well suited for PS conversational applications. Furthermore, the results clearly show a higher understanding when using AMR-WB rather than AMR-NB.
    Multimedia and Expo, 2004. ICME '04. 2004 IEEE International Conference on; 07/2004
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    ABSTRACT: This paper describes an adaptive multi-rate wideband (AMR-WB) speech codec proposed for the GSM system and also for the evolving third generation (3G) mobile speech services. The speech codec is based on SB-CELP (splitband-code-excited linear prediction) with five modes operating bit rates from 24 kbit/s down to 9.1 kbit/s. The respective channel coding schemes are based on RSC (recursive systematic code) and UEP (unequal error protection). Both, source and channel codec are designed as homogenous as possible to guarantee robust transmission on current and future mobile radio channels.
    Proceedings - ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing 10/2001; 2:757-760.
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    ABSTRACT: This paper describes an adaptive multi-rate wideband (AMR-WB) speech codec proposed for the GSM system and also for the evolving third generation (3G) mobile speech services. The coder is a multi rate SB-CELP (subband-code excited linear prediction) with five modes operating at bit rates from 24 kbit/s down to 9.1 kbit/s. Our basic approach consists of an unequal band-splitting of the input signal into two subbands (SB). A variable rate, multi-mode ACELP coder is applied to the lower subband (0-6 kHz). The various bit rates are integrated in a common structure where the scalability is realized by exchanging the fixed excitation codebooks while leaving all other codec parameters invariant. For the GSM related modes (9.1-17.8 kbit/s), the upper subband (6-7 kHz) is coded using a very low bit rate representation based on bandwidth expansion techniques. In case of the 3G application (24 kbit/s) the upper band is coded using a 4 kbit/s ADPCM coding scheme. In addition the analysis by synthesis (AbS) coder of the lower band employs a novel closed loop gain re-quantization technique controlled by the character of the speech signal. Thereby the codec achieves an enhanced performance for background noise while maintaining its clean speech quality
    Speech Coding, 2000. Proceedings. 2000 IEEE Workshop on; 02/2000
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    ABSTRACT: This paper describes a combined adaptive transform codec (ATC) and code-excited linear prediction (CELP) algorithm, called ATCELP, for the compression of wideband (7 kHz) signals. The CELP algorithm applies mainly to speech, whereas the ATC mode is selected for music and noise signals. We propose a switching scheme between CELP and ATC mode and describe a frame erasure concealment technique. Subjective listening tests have shown that the ATCELP codec at bit rates of 16, 24 and 32 kbit/s achieved performances close to those of the CCITT G.722 at 48, 56 and 64 kbit/s, respectively, at most operating conditions
    Acoustics, Speech, and Signal Processing, 1999. ICASSP '99. Proceedings., 1999 IEEE International Conference on; 04/1999
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    ABSTRACT: An algorithm for coding 20 Hz-15 kHz speech signals at 64 kbit/s with a very low delay (frame of 0.16 ms) is presented. To achieve a quality near to transparency, the authors propose adapting the Low-Delay CELP coder to the 15 kHz bandwidth and suggest a new noise shaping method based on a psychoacoustic model. In this way they take advantage of linear predictive coding and masking properties of the human perception system. Finally, an algebraic codebook is proposed, allowing an important reduction of coder computational complexity, without decreasing the perceived quality of signals
    Spoken Language, 1996. ICSLP 96. Proceedings., Fourth International Conference on; 11/1996
  • C. Murgia, Gang Feng, Catherine Quinquis, Alain Le Guyader
    Fourth European Conference on Speech Communication and Technology, EUROSPEECH 1995, Madrid, Spain, September 18-21, 1995; 01/1995
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    ABSTRACT: La manière la plus fiable d'évaluer la qualité des codecs consiste à réaliser des tests d'écoute. Ceux-ci nécessitent la présence de conditions de référence afin de permettre la comparaison des résultats d'un test à l'autre. De nouvelles techniques de codage apparaissent pour lesquelles le système de référence MNRU (Modulated Noise Reference Unit ou appareil de référence à bruit modulé) n'est plus adapté. L'objectif de ce papier est de proposer un système de référence adapté aux dégradations générées par les nouveaux schémas de compression. Pour 19 codeurs de qualité voisine mais présentant des défauts différents et sélectionnés par une analyse ACR (Absolute Category Rating), des tests de dissimilarité sont conduits dont les résultats font l'objet d'une analyse multidimensionnelle. A partir des attributs perceptifs caractérisant les quatre dimensions extraites, nous avons proposé quatre modèles de signaux d'ancrage. Une deuxième analyse sur un ensemble constitué de signaux d'ancrage et de signaux codés a permis de valider trois des attributs perceptifs.
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    ABSTRACT: Ce travail vise à fournir les signaux de référence représentatifs des nouvelles techniques de codage pour l'évaluation subjective de la qualité de parole codée. La démarche adoptée consiste à caractériser d'un point de vue perceptif les dégradations apportées par les nouveaux codecs, puis à les relier aux techniques de codage afin de pouvoir les recréer artificiellement. La MDS (Multidimensional Scaling ou analyse multidimensionnelle des proximités) non métrique pondérée est utilisée pour générer l'espace perceptif de dégradations de codecs wideband. L'analyse révèle un espace perceptif à quatre dimensions interprétable vis-à-vis des techniques de codage.
    21° Colloque GRETSI, 2007 ; p. 1189-1192.
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    ABSTRACT: Dans cette communication, nous proposons un algorithme de compression de haute qualité des signaux de la bande 20 Hz - 15 kHz avec un débit de 64 kbit/s et à très faible retard (trame de 0,16 ms). Pour atteindre une qualité proche de la transparence, nous proposons une adaptation de la technique de codage Low-Delay CELP à la bande 20 Hz - 15 kHz et nous introduisons une nouvelle technique de mise en forme du bruit de quantification. De cette manière, nous associons les avantages des techniques de codage par prédiction linéaire aux propriétés de masquage fréquentiel du système auditif humain.
    15° Colloque sur le traitement du signal et des images, 1995 ; p. 345-348.
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    ABSTRACT: This paper describes an adaptive multi rate wideband (AMR-WB) speech codec proposed for the GSM system and also for the evolving Third Generation (3G) mobile speech services. The coder is a multi rate SB-CELP (Subband-Code-Excited Linear Prediction) with five modes operating at bit rates from 24 kbit/s down to 9.1 kbit/s. Our basic approach consists of an unequal bandsplitting of the input signal into two subbands (SB). A variable rate, multi-mode ACELP coder is applied to the lower subband (0-6kHz). The various bit rates are integrated in a common structure where the scalability is realized by exchanging the fixed excitation codebooks while leaving all other codec parameters invariant. For the GSM re- lated modes (9.1-17.8 kbit/s), the upper subband (6-7 kHz) is coded using a very low bit rate representation based on bandwidth expansion techniques. In case of the 3G application (24kbit/s) the upper band is coded using a 4 kbit/s ADPCM coding scheme. In addition the analysis by synthesis (AbS) coder of the lower band employs a novel closed loop gain re-quantization technique controlled by the character of the speech signal. Thereby the codec achieves an enhanced performance for background noise while maintaining its clean speech quality.