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: 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.

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: 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 adaption of telephonic signal are provided. A telephonic listener's demographic information is first received. Next, a set of preset parameters are retrieved from a database based on the demographic information. The preset parameters are calculated using hearing data, which is representative of a demographic group. These preset parameters are then outputted to a sound personalization digital signal processing (DSP) algorithm. A voice signal is then processed using the sound personalization DSP and the processed voice signal is then outputted at the listener's telephonic device.

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: Systems and methods for adaption of telephonic signal are provided. A telephonic listener's demographic information is first received. Next, a set of preset parameters are retrieved from a database based on the demographic information. The preset parameters are calculated using suprathreshold and/or threshold hearing data, which is representative of a demographic group. These preset parameters are then outputted to a sound personalization digital signal processing (DSP) algorithm. A voice signal is then processed using the sound personalization DSP and the processed voice signal is then outputted at the listener's telephonic device.

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: 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.

Abstract: Systems and methods for adaption of telephonic signal are provided. A telephonic listener's demographic information is first received. Next, a set of preset parameters are retrieved from a database based on the demographic information. The preset parameters are calculated using suprathreshold and/or threshold hearing data, which is representative of a demographic group. These preset parameters are then outputted to a sound personalization digital signal processing (DSP) algorithm. A voice signal is then processed using the sound personalization DSP and the processed voice signal is then outputted at the listener's telephonic device.

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: Disclosed are systems and methods for processing an audio signal on a stereo audio device. Left and right audio signals are encoded as mid and side channel signals by taking the sum and difference of the left and right audio signals, 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 which is then spectrally decomposed into a plurality of sub-band signals using one or more bandpass filters. Each sub-band signal is then provided to a dynamic range compressor, compressed and outputted to a gain stage. Subsequently, each compressed sub-band signal is outputted from the gain stage and recombined. The outputs of the first and second signal pathways are then recombined and decoded in order to produce left and right audio signals. The left and right audio signals are then outputted.

Abstract: Disclosed are systems and methods for processing an audio signal on a stereo audio device. Left and right audio signals are encoded as mid and side channel signals by taking the sum and difference of the left and right audio signals, 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 which is then spectrally decomposed into a plurality of sub-band signals using one or more bandpass filters. Each sub-band signal is then provided to a dynamic range compressor, compressed and outputted to a gain stage. Subsequently, each compressed sub-band signal is outputted from the gain stage and recombined. The outputs of the first and second signal pathways are then recombined and decoded in order to produce left and right audio signals. The left and right audio signals are then outputted.

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 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 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: 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.