Abstract: In one embodiment the present invention includes a method of improving audibility of speech in a multi-channel audio signal. The method includes comparing a first characteristic and a second characteristic of the multi-channel audio signal to generate an attenuation factor. The first characteristic corresponds to a first channel of the multi-channel audio signal that contains speech and non-speech audio, and the second characteristic corresponds to a second channel of the multi-channel audio signal that contains predominantly non-speech audio. The method further includes adjusting the attenuation factor according to a speech likelihood value to generate an adjusted attenuation factor. The method further includes attenuating the second channel using the adjusted attenuation factor.

Abstract: The invention relates to audio signal processing and speech enhancement. In accordance with one aspect, the invention combines a high-quality audio program that is a mix of speech and non-speech audio with a lower-quality copy of the speech components contained in the audio program for the purpose of generating a high-quality audio program with an increased ratio of speech to non-speech audio such as may benefit the elderly, hearing impaired or other listeners. Aspects of the invention are particularly useful for television and home theater sound, although they may be applicable to other audio and sound applications. The invention relates to methods, apparatus for performing such methods, and to software stored on a computer-readable medium for causing a computer to perform such methods.

Abstract: A method and system for filtering a multi-channel audio signal having a speech channel and at least one non-speech channel, to improve intelligibility of speech determined by the signal. In typical embodiments, the method includes steps of determining at least one attenuation control value indicative of a measure of similarity between speech-related content determined by the speech channel and speech-related content determined by the non-speech channel, and attenuating the non-speech channel in response to the at least one attenuation control value. Typically, the attenuating step includes scaling of a raw attenuation control signal (e.g., a ducking gain control signal) for the non-speech channel in response to the at least one attenuation control value. Some embodiments are a general or special purpose processor programmed with software or firmware and/or otherwise configured to perform filtering in accordance the invention.

Abstract: The invention relates to audio signal processing. More specifically, the invention relates to enhancing multichannel audio, such as television audio, by applying a gain to the audio that has been smoothed between portions of the audio. The invention relates to methods, apparatus for performing such methods, and to software stored on a computer-readable medium for causing a computer to perform such methods.

Abstract: The invention relates to audio signal processing. More specifically, the invention relates to enhancing multichannel audio, such as television audio, by applying a gain to the audio that has been smoothed between segments of the audio. The invention relates to methods, apparatus for performing such methods, and to software stored on a computer-readable medium for causing a computer to perform such methods.

Abstract: The invention relates to audio signal processing. More specifically, the invention relates to enhancing multichannel audio, such as television audio, by applying a gain to the audio that has been smoothed between segments of the audio. The invention relates to methods, apparatus for performing such methods, and to software stored on a computer-readable medium for causing a computer to perform such methods.

Abstract: Metadata comprising a set of gain values for creating a dominance effect is automatically generated. Automatically generating the metadata includes receiving multiple audio streams and a dominance criterion for at least one of the audio streams. A set of gains is computed for one or more audio streams based on the dominance criterion for the at least one audio stream and metadata is generated with the set of gains.

Abstract: The invention relates to audio signal processing. More specifically, the invention relates to enhancing entertainment audio, such as television audio, to improve the clarity and intelligibility of speech, such as dialog and narrative audio. The invention relates to methods, apparatus for performing such methods, and to software stored on a computer-readable medium for causing a computer to perform such methods.

Abstract: A personal sound system is described that includes a wireless network supporting an ear-level module, a companion module and a phone. Other audio sources are supported as well. A configuration processor configures the ear-level module and the companion module for private communications, and configures the ear-level module for a plurality of signal processing modes, including a hearing aid mode, for a corresponding plurality of sources of audio data. The ear module is configured to handle variant audio sources, and control switching among them.

Abstract: A personal sound system is described that includes a wireless network supporting an ear-level module, a companion module and a phone. Other audio sources are supported as well. A configuration processor configures the ear-level module and the companion module for private communications, and configures the ear-level module for a plurality of signal processing modes, including a hearing aid mode, for a corresponding plurality of sources of audio data. The ear module is configured to handle variant audio sources, and control switching among them.

Abstract: At least one segment is identified in an audio signal. The audio segment is associated with an artifact within the audio signal and has a time duration. At least one stored sound clip is retrieved, which has a time duration that exceeds the time duration associated with the audio segment. The retrieved sound clip is mixed with the audio signal and the retrieved sound clip audibly compensates for the audio artifact.

Abstract: In one embodiment the present invention includes a method of improving audibility of speech in a multi-channel audio signal. The method includes comparing a first characteristic and a second characteristic of the multi-channel audio signal to generate an attenuation factor. The first characteristic corresponds to a first channel of the multi-channel audio signal that contains speech and non-speech audio, and the second characteristic corresponds to a second channel of the multi-channel audio signal that contains predominantly non-speech audio. The method further includes adjusting the attenuation factor according to a speech likelihood value to generate an adjusted attenuation factor. The method further includes attenuating the second channel using the adjusted attenuation factor.

Abstract: The invention relates to audio signal processing. More specifically, the invention relates to enhancing entertainment audio, such as television audio, to improve the clarity and intelligibility of speech, such as dialog and narrative audio. The invention relates to methods, apparatus for performing such methods, and to software stored on a computer-readable medium for causing a computer to perform such methods.

Abstract: The invention relates to audio signal processing and speech enhancement. In accordance with one aspect, the invention combines a high-quality audio program that is a mix of speech and non-speech audio with a lower-quality copy of the speech components contained in the audio program for the purpose of generating a high-quality audio program with an increased ratio of speech to non-speech audio such as may benefit the elderly, hearing impaired or other listeners. Aspects of the invention are particularly useful for television and home theater sound, although they may be applicable to other audio and sound applications. The invention relates to methods, apparatus for performing such methods, and to software stored on a computer-readable medium for causing a computer to perform such methods.

Abstract: Methods and devices for dynamically adjusting a multi-band signal-modification profile based on a psychoacoustic model are disclosed. In one arrangement, the encoding parameter side information is used to estimate encoding noise of an encoded signal. The signal spectrum of the signal is estimated. Adjustments to the multi-band signal-modification profile are determined using the estimated noise and signal spectrum and a psychoacoustic profile.

Abstract: A personal sound system is described that includes a wireless network supporting an ear-level module, a companion module and a phone. Other audio sources are supported as well. A configuration processor configures the ear-level module and the companion module for private communications, and configures the ear-level module for a plurality of signal processing modes, including a hearing aid mode, for a corresponding plurality of sources of audio data. The ear module is configured to handle variant audio sources, and control switching among them.

Abstract: The present invention includes methods of preprocessing soundtracks adapted to hearing impairment, environmental noise or other factors and making the preprocessed soundtracks available as alternate soundtracks. Particular aspects of the present invention are described in the claims, specification and drawings.

Abstract: The present invention includes methods of and devices for signal modification during decoding of an audio signal and for dynamically adjusting a signal-modification profile based on a psychoacoustic model. Particular aspects of the present invention are described in the claims, specification and drawings.