Abstract: Described are techniques for audio processing. For instance, a process can include obtaining a first set of digital signal processing (DSP) parameters associated with a first hearing profile and a second set of DSP parameters associated with a second hearing profile different than the first hearing profile. One or more differentially processed audio samples can be output, each including a first audio output signal generated by processing a first audio signal using the first set of DSP parameters and a second audio output signal generated by processing a second audio signal using the second set of DSP parameters. For each differentially processed audio sample, a user input can be obtained indicative of the first or second audio output signal having a lower audio quality. One or more user hearing thresholds can be determined based on the respective user input obtained for each respective differentially processed audio sample.

Abstract: A method for conducting a cross frequency simultaneous masking (xF SM) test begins with generating a signal probe and a masker probe. The center frequencies of the signal and masker probes are separated by a fixed frequency ratio. An xF SM curve is generated by sweeping the signal and masker probes across a given frequency range, while maintaining the fixed frequency ratio between the two. While sweeping, the masker probe is maintained at a pre-determined masker amplitude or a series of pre-determined masker amplitudes. The amplitude of the signal probe is adjusted in response to a series of user inputs, which are then interpolated to generate the xF SM curve. Additionally, while sweeping, the signal probe can be maintained at one or more pre-determine amplitudes and the amplitude of the masker probe adjusted in response to user inputs, which are then interpolated to generate the xF SM curve.

Abstract: 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.

Abstract: 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.

Abstract: 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.

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: A method for conducting a cross frequency simultaneous masking (xF SM) test begins with generating a signal probe and a masker probe. The center frequencies of the signal and masker probes are separated by a fixed frequency ratio. An xF SM curve is generated by sweeping the signal and masker probes across a given frequency range, while maintaining the fixed frequency ratio between the two. While sweeping, the masker probe is maintained at a pre-determined masker amplitude or a series of pre-determined masker amplitudes. The amplitude of the signal probe is adjusted in response to a series of user inputs, which are then interpolated to generate the xF SM curve. Additionally, while sweeping, the signal probe can be maintained at one or more pre-determine amplitudes and the amplitude of the masker probe adjusted in response to user inputs, which are then interpolated to generate the xF SM curve.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: A method for processing an audio signal includes generating a user hearing profile and calculating, based on the user hearing profile, at least one set of audio content-specific DSP (digital signal processing) parameters for one or more sound personalization algorithms. One or more of the calculated sets of content-specific DSP parameters are associated with a content-identifier for their respective specific content. In response to an audio stream on an audio output device, the audio stream is analyzed to determine at least one content type. Based on the determined content type, corresponding content-specific DSP parameters are outputted to the audio output device, based at least in part on their content identifier. An audio signal is processed on the audio output device by using a given sound personalization algorithm parameterized by the corresponding content-specific DSP parameters.

Abstract: Disclosed are systems and methods for ambient sound enhancement on a mobile device. A user hearing profile is generated and a set of ambient sound enhancement digital signal processing (DSP) parameters is calculated for a sound enhancement algorithm, based at least in part on the user hearing profile. In response to a user initiating an ambient sound enhancement function on a mobile computing device, at least one set of calculated ambient sound enhancement DSP parameters is retrieved. Ambient sound is captured with a microphone of the mobile computing device and processed with an ambient sound enhancement DSP. The ambient sound enhancement DSP is parameterized with the retrieved set of calculated ambient sound enhancement DSP parameters and the DSP enhanced processed audio signal is outputted to a transducer of the mobile device.

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: 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.

Abstract: 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.

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: 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.

Abstract: The present invention relates to an ambisonic encoder for a sound wave having a plurality of reflections. The ambisonic encoder according to the invention 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 according to the invention is less than the complexity of encoding of the reflections of sound sources of an ambisonic encoder according to the prior art. The ambisonic encoder according to the invention makes it possible to encode a greater number of reflections of a sound source in real time. The ambisonic encoder according to the invention 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.