The invention relates to the measurement and control of the perceived sound loudness and/or the perceived spectral balance of an audio signal. An audio signal is modified in response to calculations performed at least in part in the perceptual (psychoacoustic) loudness domain. The invention is useful, for example, in one or more of: loudness-compensating volume control, automatic gain control, dynamic range control (including, for example, limiters, compressors, expanders, etc.), dynamic equalization, and compensating for background noise interference in an audio playback environment. The invention includes not only methods but also corresponding computer programs and apparatus.

Calculating and adjusting the perceived loudness and/or the perceived spectral balance of an audio signal

In one disclosed aspect, dynamic gain modification s are applied to an audio signal at least partly in response to auditory events and/or the degree of change in signal characteristics associated with said auditory event boundaries. In another aspect, an audio signal is divided into auditory events by comparing the difference in specific loudness between successive time blocks of the audio signal.

Audio gain control using specific-loudness-based auditory event detection

One or a combination of two or more specific loudness model functions selected from a group of two or more of such functions are employed in calculating the perceptual loudness of an audio signal. The function or functions may be selected, for example, by a measure of the degree to which the audio signal is narrowband or wideband. Alternatively or with such a selection from a group of functions, a gain value G[t] is calculated, which gain, when applied to the audio signal, results in a perceived loudness substantially the same as a reference loudness. The gain calculating employs an iterative processing loop that includes the perceptual loudness calculation.

Method, apparatus and computer program for calculating and adjusting the perceived loudness of an audio signal

We propose a new approach for real-time implementation of the fast affine projection (FAP) algorithm. This is based on exploiting the recently introduced dichotomous coordinate descent (DCD) algorithm, which is especially efficient for solving systems of linear equations on real-time hardware and software platforms since it is free of multiplication and division. The numerical stability of the DCD algorithm allows the new combined DCD-FAP algorithm also to be stable. The convergence and complexity of the DCD-FAP algorithm is compared with that of the FAP, Gauss-Seidel FAP (GS-FAP), and modified GS-FAP algorithms in the application to acoustic echo cancellation. The DCD-FAP algorithm demonstrates a performance close to that of the FAP algorithm with ideal matrix inversion and the complexity smaller than that of the Gauss-Seidel FAP algorithms.

/pdf/coordinate-descent-iterations-in-fast-affine-projection-19b56wau1x.pdf

Coordinate descent iterations in fast affine projection algorithm

Methods and apparatus for embedding a watermark are disclosed. An example method disclosed herein to embed a watermark in a compressed data stream comprises obtaining a set of transform coefficients included in the compressed data stream, the set of transform coefficients having a respective first set of mantissa codes and a respective set of exponents, the first set of mantissa codes associated with a respective set of mantissa step sizes, identifying a first transform coefficient from the set of transform coefficients having a smallest magnitude among the set of transform coefficients, determining a second set of mantissa codes based on the first transform coefficient and the set of step sizes, and replacing the first set of mantissa codes included in the compressed data stream with the second set of mantissa codes to embed the watermark without uncompressing the compressed data stream.

Methods and apparatus for embedding watermarks

In this paper, we present a new approach for adaptive echo cancellation: Pseudo Affine projection algorithm. From the original update equation of the filter coefficients of the Affine Projection algorithms, which are based on multiple dimension projection of the input signal vector, we derive a simplified solution under some realistic hypothesis. We show that our solution possesses a complexity figure close to the NLMS one but with the same convergence characteristics of the original sampleby-sample AP algorithm. Implementation details are presented in the last part of the paper.

/pdf/pseudo-affine-projection-algorithm-new-solution-for-adaptive-4miy2g1x4y.pdf

Pseudo affine projection algorithm new solution for adaptive identication.

Dispositif de traitement de signal audio aopte a recevoir une pluralite de flux audio d'entree et a fournir au moins un flux audio de sortie a partir desdits flux audio d'entree, l'un au moins des flux audio d'entree, dit premier flux, pouvant presenter des donnees manquantes, le dispositif comportant des moyens pour decider ou non de la regeneration des donnees manquantes dudit premier flux a partir d'au moins une caracteristique d'au moins un second flux audio d'entree.

Methode et dispositif de traitement d'une pluralite de flux binaires audio

The normalized LMS (NLMS) algorithm has been successfully used in many system identification problems. However, the NLMS algorithm is known to exhibit low convergence speed especially when the input data covariance matrix is ill-conditioned. In this paper, we consider a sub-optimal implementation of the affine projection (AP) algorithm based on a prewithening mechanism, which renders the convergence characteristics less sensitive to the coloring of the input signal spectrum than is the case for the NLMS algorithm. Comparisons with the AP algorithm are given to validate our approach. Implementation details are discussed in the context of hands-free telephony where echo cancelling and speech coding algorithms are integrated on the same DSP board.

/pdf/on-integrating-speech-coding-functions-into-echo-cancelling-2ya862o15q.pdf

On integrating speech coding functions into echo cancelling filters with decorrelating properties

In packet switched networks (case of IP networks) no minimal threshold on rate is offered for the transport of packets. The available bandwidth depends on conditions of the traffic on the network (best effort strategy). In VoIP (Voice Over Internet Protocol) audio packets must be continuously played. When packets are lost or have arrived too late it gives place to chopped sound and to a degradation of the quality. In this paper the lost case is considered when a packet loss is recognized by the reception of the next packet. Generally concealment techniques do not use this received signal. We propose here an algorithm, which uses this information to improve the missing signal synthesis and allows maintaining an acceptable quality of the speech signal. A comparison between this method and techniques recommended by the ITU-T for coders G.711 [3] and G.723.1 [1] will also be described.

/pdf/packet-loss-concealment-using-audio-morphing-6p9owgnc6f.pdf

Packet loss concealment using audio morphing

Le dispositif recoit un signal d'entree x t et un signal d'observation y t 
dont une composante est la reponse d'un systeme au signal d'entree. Il
determine un signal d'erreur e t a un instant t selon l'expression
e t = y t - X T / t H t-1 , ou H t-1 est un vecteur compose de L coefficients d'un filtre
d'identification (18) a reponse impulsionnelle finie representative de la reponse
impulsionnelle du systeme, et X T / t = (x t ,x t-1 ,···,x t-L+1 ). Une analyse predictive
du signal d'entree est effectuee par un calcul portant sur des trames du signal
d'entree. Les L coefficients du filtre d'identification sont adaptes en ajoutant au
vecteur H t-1 un vecteur proportionnel a e t / X T t U t + λ U t , ou U t est un vecteur
compose des L valeurs d'un residu de prediction du signal d'entree a l'instant t
et aux L-1 instants precedents, et λ un coefficient positif ou nul.

Procédé et disposiif d'identification adaptive, et annuleur d'écho adaptive mettant en oeuvre un tel procédé

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