Mimi Hearing Technologies Gmbh

Nicholas R Clark

Michael Hirsch

Christoph Omid Hohnerlein

Vinzenz H Schönfelder

Philipp Skribanowitz

Tina Baumann

Natalia Delgado

Guido Knook

Robert Pelzer

Ryan Klimczak

Systems and methods for processing an audio signal are provided for server-mediated sound personalization on a plurality of consumer devices. A user hearing test is conducted on one of a plurality of audio output devices. Next, the hearing data of the user's hearing test is outputted to a server and stored on the server's database along with a unique user identifier. Next, a set of DSP parameters for a sound personalization algorithm are calculated from the user's hearing data. The DSP parameter set is then outputted to one of a plurality of audio output devices when the user logs in with their unique identifier on an application on the audio output device.

Systems and methods for providing personalized audio replay on a plurality of consumer devices

Disclosed are systems and methods for processing an audio signal. In particular, there is provided a method for determining dynamic gain values to be applied on a digital input signal. The digital signal may be arranged in blocks. The dynamic gain values may be used for attenuating input signal values exceeding a clipping threshold. More particularly, the method comprising, for each signal block, passing backwards over the next signal block and the current signal block to produce a preliminary gain contour from the input signal; and passing forwards over the current signal block to produce a final gain contour for the current signal block based on the preliminary gain contour, wherein the gain contours are produced by applying an instant gain ascent and a smooth gain decay to the gain contours.

Systems and methods for limiter functions

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.

Systems and methods for user-dependent conditioning of stimuli in tests using a method of continuous adjustment

Disclosed are systems and methods for fitting a sound personalization algorithm using a two-dimensional (2D) graphical fitting interface. A calculated set of initial digital signal processing (DSP) parameters are determined for a given sound personalization algorithm, based on a user hearing profile. The initial DSP parameters are outputted to a 2D graphical fitting interface of an audio personalization application, wherein a first axis represents a level of coloration and a second axis represents a level of compression. A user input specifies a first 2D coordinate selected from a coordinate space presented by the 2D graphical fitting interface. A first set of refined DSP parameters is generated to apply a coloration and/or compression adjustment corresponding to the first 2D coordinate. The given sound personalization algorithm is parameterized with the first set of refined DSP parameters.

Systems and methods for fitting a sound processing algorithm in a 2D space using interlinked parameters

Disclosed is a method and apparatus for determining one or more operation parameters for a dynamic range compression (DRC) system. The method comprises obtaining, as an input, a parameter indicative of a hearing ability of a user, the parameter relating to a first difference in sound intensity between a maskee at a first frequency and a masker at a second frequency, determining a target value for the parameter, and determining the one or more operation parameters such that a second difference in sound intensity after sound intensity modification by the DRC (between sound intensity of the maskee of the masker) corresponds to the target value for the parameter. The operation parameters are determined such that a dependence of the second difference in sound intensity on the sound intensity of the maskee is minimized for a given range of sound intensities of the maskee.

Method to process an audio signal with a dynamic compressive system

Systems and methods are provided for evaluating hearing health of a given user. An input audio signal is transformed into the frequency domain and a first and second hearing profile are applied to the input audio sample. The first hearing profile represents a healthy hearing standard and the second hearing profile is the given user's hearing profile. Using the hearing profiles, first and second perceptually relevant information (PRI) values are generated for the input audio sample. The first and second PRI values are analyzed against each other to generate a PRI index value for the given user, where the PRI index value is a hearing health index value for the given user. The given user's hearing profile may additionally be applied to differently processed audio samples to evaluate the amount of perceptual rescue offered by various digital signal processing algorithms.

Systems and methods for evaluating hearing health

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 20.000 Hz, wherein the supra-threshold test is performed on an unreferenced audio-system at a first relative sound level relative to a predefined output level of the unreferenced audio-system, determining the result of the supra-threshold test for at least a portion of the audible frequency spectrum, wherein the results of the supra-threshold test are provided particularly relative to the predefined output level of the unreferenced audio-system, determining from the progression of the determined result of the supra-threshold test at least one absolute pure tone threshold, wherein the at least one absolute pure tone threshold is provided in absolute physical units, particularly in decibels hearing level or decibels sound pressure level.

Method for accurately estimating a pure tone threshold using an unreferenced audio-system

The present disclosure relates to a method for processing media content. The method comprises receiving a definition of a user specific media corpus, the media corpus comprising a plurality of media files; ranking candidate media segments generated from the media corpus according to one or more signal characteristics derived from the candidate media segments; processing the media signals of a subset of the ranked candidate media segments with different signal processings; presenting the differently processed media signals to a user for a user preference judgement; and determining a preferred signal processing based on user responses to the presentations.

Method of selecting a suitable content for subjective preference judgement

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. Finally, the feedforward-compressed subband signals are recombined.

Systems and methods for processing an audio signal for replay on an audio device

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.

Systems and methods for modifying an audio signal using custom psychoacoustic models

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. Finally, the feedforward-compressed subband signals are recombined.

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.

Systems and methods for encoding an audio signal using custom psychoacoustic models

Disclosed are systems and methods for audio processing and sound personalization signal processing. Information suitable to derive an indication of a user's hearing ability is received, and a representative indication of hearing ability is determined for the user. A set of audio processing parameters are determined for the representative indication of hearing ability and a sound personalization signal processing algorithm is configured based on the set of audio processing parameters. An audio signal is received on an entertainment system and is processed using the configured sound personalization signal processing algorithm. The processed audio signal is then outputted from the entertainment system.

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Royston, United Kingdom

Nicholas R Clark