Abstract: Disclosed are technologies for processing a surround sound audio signal. The center channel is processed to generate a processed center channel. For each pair of corresponding left and right audio channels, mid and side channel signals are generated by taking the sum and difference of the pair, respectively. The side channel is outputted to a first signal pathway that features a dynamic range compressor. The mid channel is outputted to a second signal pathway and spectrally decomposed into a plurality of sub-band signals. Each sub-band signal is compressed by a dynamic range compressor and transmitted to a gain stage. The compressed sub-band signals are outputted from the gain stage and recombined. The outputs of the first and second signal pathways are then recombined and decoded to produce processed left and right audio signals. Each pair or processed left and right audio signals are output along with the processed center channel.

Abstract: Systems and methods are provided for modifying an audio signal using custom psychoacoustic models. A user's hearing profile is first obtained. Subsequently, an audio processing function is parameterized so as to optimize the user's perceptually relevant information. The method for calculating the user's perceptually relevant information comprises first processing audio signal samples using the parameterized processing function and then transforming samples of the processed audio signals into the frequency domain. Next, masking and hearing thresholds are obtained from the user's hearing profile and applied to the transformed audio sample, wherein the user's perceived data is calculated. Once perceptually relevant information is optimized, the resulting parameters are transferred to the audio processing function and an output audio signal is processed.

Abstract: Systems and methods are provided for modifying an audio signal using custom psychoacoustic methods, for encoding the audio signal. A user's hearing profile is first obtained. Subsequently, a sample of the audio signal is split into frequency components. Next, masking and hearing thresholds are obtained from the user's hearing profile and applied to the frequency components of the audio sample, wherein the user's perceived data is calculated. User's imperceptible audio signal data is then disregarded. The audio sample is quantized and the resulting transformed audio sample encoded.

Abstract: Systems and methods are provided for modifying an audio signal using custom psychoacoustic models. A user's hearing profile is first obtained. Subsequently, an audio processing function is parameterized so as to optimize the user's perceptually relevant information. The method for calculating the user's perceptually relevant information comprises first processing audio signal samples using the parameterized processing function and then transforming samples of the processed audio signals into the frequency domain. Next, masking and hearing thresholds are obtained from the user's hearing profile and applied to the transformed audio sample, wherein the user's perceived data is calculated. Once perceptually relevant information is optimized, the resulting parameters are transferred to the audio processing function and an output audio signal is processed.

Abstract: Systems and methods for processing an audio signal are provided for replay on an audio device. An audio signal is spectrally decomposed into a plurality of subband signals using and pass filters. Each of the subband signals are provided to a respective modulator and subsequently, from the modulator output, provided to a respective first processing path that includes a first dynamic range compressor, DRC. Each subband signal is feedforward compressed by the respective first DRC to obtain a feedforward-compressed subband signal, wherein the first DRC is slowed relative to an instantaneous DRC. Subsequently, each feedforward-compressed subband signal is provided to a second processing path that includes a second DRC, wherein the feedforward-compressed subband signal is compressed by the respective second DRC and outputted to the respective modulator. Modulation of the subband signals is then performed in dependence on the output of the second processing path.

Abstract: Systems and methods for processing an audio signal are provided for replay on an audio device. An audio signal is spectrally decomposed into a plurality of subband signals using band pass filters. Each of the subband signals are provided to a respective modulator and subsequently, from the modulator output, provided to a respective first processing path that includes a first dynamic range compressor, DRC. Each subband signal is feedforward compressed by the respective first DRC to obtain a feedforward-compressed subband signal, wherein the first DRC is slowed relative to an instantaneous DRC. Subsequently, each feedforward-compressed subband signal is provided to a second processing path that includes a second DRC, wherein the feedforward-compressed subband signal is compressed by the respective second DRC and outputted to the respective modulator. Modulation of the subband signals is then performed in dependence on the output of the second processing path.

Abstract: A method is provided for generating a three-dimensional morphable model of an element from an initial database of examples of such elements providing data allowing, for each of the elements of the initial database, a three-dimensional meshed surface based on points and on a triangular network connecting the points to be determined.

Abstract: An ambisonic encoder for a sound wave has a plurality of reflections. The ambisonic encoder makes it possible to improve the sensation of immersion in a 3D audio scene. The complexity of encoding of the reflections of sound sources for an ambisonic encoder is less than the complexity of encoding of the reflections of sound sources of previously known ambisonic encoders. The ambisonic encoder makes it possible to encode a greater number of reflections of a sound source in real time, and makes it possible to reduce the power consumption related to ambisonic encoding, and to increase the life of a battery of a mobile device used for said application.

Abstract: Systems and methods are provided for modifying an audio signal using custom psychoacoustic models. A user's hearing profile is first obtained. Subsequently, a multiband dynamic processor is parameterized so as to optimize the user's perceptually relevant information. The method for calculating the user's perceptually relevant information comprises first processing audio signal samples using the parameterized multiband dynamic processor and then transforming samples of the processed audio signals into the frequency domain. Next, masking and hearing thresholds are obtained from the user's hearing profile and applied to the transformed audio sample, wherein the user's perceived data is calculated. Once perceptually relevant information is optimized, the resulting parameters are transferred to a multiband dynamic processor and an output audio signal is processed.

Abstract: A method for generating an individual-specific head-related transfer function from a database containing 3D or 2D ear data and corresponding head-related transfer functions, the method comprises the steps of: performing a statistical analysis of the 3D or 2D ear space of the database; performing a statistical analysis of the head-related-transfer-function space of the data base; performing an analysis of the relationships between the statistical parameters of the statistical analysis of the 3D or 2D ear space and the statistical parameters of the head-related-transfer-function space; and determining, from the relationship analysis and the statistical analysis of the 3D or 2D ear space, a function for calculating a head-related transfer function from data representative of at least one ear.

Abstract: A method of enhancing an audio signal from an audio output device is provided. For a frequency band, a user masking contour curve covering at least a part of said frequency band is obtained, a target masking contour curve is derived from the user masking contour curve, and a multi-band digital compression system is parameterized based on the sound level of the target masking contour curve at a given frequency and the sound level of the user masking contour curve at the same given frequency. The obtained parameters are outputted to provide an enhanced audio signal.

Abstract: Method for estimating a pure tone hearing threshold or a pure tone audiogram of a person with a non-calibrated audio-system, comprising the steps of performing a supra-threshold test at least over a portion of the audible frequency spectrum, wherein the audible frequency spectrum ranges particularly from 16 Hz to 20.

Abstract: Disclosed are systems and methods for user-dependent conditioning of stimuli in tests to elicit user responses to variations of one or more adaptive parameters of a user stimulus signal. The user stimulus signal is generated based on first and second adaptive parameters. The first adaptive parameter is modified to thereby generate a plurality of successive variations in the user stimulus signal, over one or more ranges of values of the second adaptive parameter. In response to modifying the first adaptive parameter, a plurality of user responses from a given user are received. Each user response indicates that one of the plurality of successive variations in the user stimulus signal has occurred. Based on an expected user response curve for the given user and a calculated time interval between successive user responses, an instantaneous rate of change for modifying the first adaptive parameter is adjusted such that the user responses are steered toward a neutral state around the expected user response curve.

Abstract: Systems and methods for processing an audio signal are provided for replay on an audio device. An audio signal is spectrally decomposed into a plurality of subband signals using and pass filters. Each of the subband signals are provided to a respective modulator and subsequently, from the modulator output, provided to a respective first processing path that includes a first dynamic range compressor, DRC. Each subband signal is feedforward compressed by the respective first DRC to obtain a feedforward-compressed subband signal, wherein the first DRC is slowed relative to an instantaneous DRC. Subsequently, each feedforward-compressed subband signal is provided to a second processing path that includes a second DRC, wherein the feedforward-compressed subband signal is compressed by the respective second DRC and outputted to the respective modulator. Modulation of the subband signals is then performed in dependence on the output of the second processing path.