Abstract: Methods, systems, and apparatuses are described for determining relative locations of wireless loudspeakers and performing channel mapping thereof. An audio processing component utilizes sounds produced by wireless loudspeakers during setup/installation procedures, which are received by a microphone at locations in an acoustic space, to determine an amount of time between when the audio signal is initially transmitted and when the microphone signal is received. The audio processing component also utilizes wireless timing signals provided by a wireless transceiver, at locations in the acoustic space, to wireless loudspeakers and then back to the wireless transceiver to determine an amount of time between transmission and reception by the wireless transceiver. The timing delays are used to determine the locations of the wireless loudspeakers in the acoustic space.

Abstract: Methods, systems, and apparatuses are described for determining relative locations of wireless loudspeakers and performing channel mapping thereof. An audio processing component utilizes sounds produced by wireless loudspeakers during setup/installation procedures, which are received by a microphone at locations in an acoustic space, to determine an amount of time between when the audio signal is initially transmitted and when the microphone signal is received. The audio processing component also utilizes wireless timing signals provided by a wireless transceiver, at locations in the acoustic space, to wireless loudspeakers and then back to the wireless transceiver to determine an amount of time between transmission and reception by the wireless transceiver. The timing delays are used to determine the locations of the wireless loudspeakers in the acoustic space.

Abstract: Methods, systems, and apparatuses are described for determining relative locations of wireless loudspeakers and performing channel mapping thereof. An audio processing component utilizes sounds produced by wireless loudspeakers during setup/installation procedures, which are received by a microphone at locations in an acoustic space, to determine an amount of time between when the audio signal is initially transmitted and when the microphone signal is received. The audio processing component also utilizes wireless timing signals provided by a wireless transceiver, at locations in the acoustic space, to wireless loudspeakers and then back to the wireless transceiver to determine an amount of time between transmission and reception by the wireless transceiver. The timing delays are used to determine the locations of the wireless loudspeakers in the acoustic space.

Abstract: Methods, systems, and apparatuses are described for determining relative locations of wireless loudspeakers and performing channel mapping thereof ?n audio processing component utilizes sounds produced by wireless loudspeakers during setup/installation procedures, which are received by a microphone at locations in an acoustic space, to determine an amount of time between when the audio signal is initially transmitted and when the microphone signal is received. The audio processing component also utilizes wireless timing signals provided by a wireless transceiver, at locations in the acoustic space, to wireless loudspeakers and then back to the wireless transceiver to determine an amount of time between transmission and reception by the wireless transceiver. The timing delays are used to determine the locations of the wireless loudspeakers in the acoustic space.

Abstract: Methods, systems, and apparatuses are described for improved multi-microphone source tracking and noise suppression. In multi-microphone devices and systems, frequency domain acoustic echo cancellation is performed on each microphone input, and microphone levels and sensitivity are normalized. Methods, systems, and apparatuses are also described for improved acoustic scene analysis and source tracking using steered null error transforms, on-line adaptive acoustic scene modeling, and speaker-dependent information. Switched super-directive beamforming reinforces desired audio sources and closed-form blocking matrices suppress desired audio sources based on spatial information derived from microphone pairings. Underlying statistics are tracked and used to updated filters and models. Automatic detection of single-user and multi-user scenarios, and single-channel suppression using spatial information, non-spatial information, and residual echo are also described.

Abstract: A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.

Abstract: A system and method is described that improves the intelligibility of a far-end telephone speech signal to a user of a telephony device in the presence of near-end background noise. As described herein, the system and method improves the intelligibility of the far-end telephone speech signal in a manner that does not require user input and that minimizes the distortion of the far-end telephone speech signal. The system is integrated with an acoustic echo canceller and shares information therewith.

Abstract: Methods, systems, and apparatuses are described for improved multi-microphone source tracking and noise suppression. In multi-microphone devices and systems, frequency domain acoustic echo cancellation is performed on each microphone input, and microphone levels and sensitivity are normalized. Methods, systems, and apparatuses are also described for improved acoustic scene analysis and source tracking using steered null error transforms, on-line adaptive acoustic scene modeling, and speaker-dependent information. Switched super-directive beamforming reinforces desired audio sources and closed-form blocking matrices suppress desired audio sources based on spatial information derived from microphone pairings. Underlying statistics are tracked and used to updated filters and models. Automatic detection of single-user and multi-user scenarios, and single-channel suppression using spatial information, non-spatial information, and residual echo are also described.

Abstract: A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.

Abstract: Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this tact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described.

Abstract: Methods, systems, and apparatuses are described for performing speaker-identification-assisted speech processing. In accordance with certain embodiments, a communication device includes speaker identification (SID) logic that is configured to identify a user of the communication device and/or the identity of a far-end speaker participating in a voice call with a user of the communication device. Knowledge of the identity of the user and/or far-end speaker is then used to improve the performance of one or more speech processing algorithms implemented on the communication device.

Abstract: A hybrid instantaneous/differential encoding technique is described herein that may be used to reduce the bit rate required to encode a pitch period associated with a segment of a speech signal in a manner that will result in relatively little or no degradation of a decoded speech signal generated using the encoded pitch period. The hybrid instantaneous/differential encoding technique is advantageously applicable to any speech codec that encodes a pitch period associated with a segment of a speech signal.

Abstract: Methods, systems, and apparatuses are described for performing speaker-identification-assisted speech processing in an uplink path of a communication device. In accordance with certain embodiments, a communication device includes speaker identification (SID) logic that is configured to identify the identity of a near-end speaker. Knowledge of the identity of the near-end speaker is then used to improve the performance of one or more uplink speech processing algorithms implemented on the communication device.

Abstract: Systems and methods are described for enhancing the audio quality of an FM receiver. In embodiments described herein, a stop band noise signal is extracted from an L+R or L?R signal produced by an FM stereo decoder. A channel quality measure is calculated based on the stop band noise signal and is used to control whether a pop suppression technique is applied to the L+R signal. The channel quality measure and the stop band noise signal are also leveraged to perform single-channel noise suppression in the frequency domain on the L?R signal and on the L+R signal. The channel quality measure is also used to control the application of a fast fading compensation process that replaces noisy segments of the L?R and L+R signal with replacement waveforms generated via waveform extrapolation.

Abstract: A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.

Abstract: A loudness enhancement system and method is described that increases the loudness of an audio signal being played back by an audio device that places limits on the dynamic range of the audio signal. In an embodiment, the loudness enhancement system and method compresses the audio signal to an adaptively-determined compression limit that is greater than or equal to a maximum desired output level and then applies an adaptively-determined degree of soft clipping to the compressed audio signal. The compression limit and degree of soft clipping may be determined based on an overload measure that is calculated for successive portions of the audio signal. The loudness enhancement system and method advantageously operates in a manner that generates less distortion than the method of simply over-driving the audio signal such that hard-clipping occurs.

Abstract: Systems and methods are described for enhancing the audio quality of an FM receiver. In embodiments described herein, a stop band noise signal is extracted from an L+R or L?R signal produced by an FM stereo decoder. A channel quality measure is calculated based on the stop band noise signal and is used to control whether a pop suppression technique is applied to the L+R signal. The channel quality measure and the stop band noise signal are also leveraged to perform single-channel noise suppression in the frequency domain on the L?R signal and on the L+R signal. The channel quality measure is also used to control the application of a fast fading compensation process that replaces noisy segments of the L?R and L+R signal with replacement waveforms generated via waveform extrapolation.

Abstract: Systems and methods are described for enhancing the audio quality of an FM receiver. In embodiments described herein, quadrature L?R demodulation is applied to a composite baseband signal output by an FM demodulator to obtain an L?R noise signal. A channel quality measure is calculated based on the L?R noise signal and is used to control whether a pop suppression technique is applied to an L+R signal obtained from the composite baseband signal to detect and remove noise pulses therefrom. The channel quality measure and the L?R noise signal are also leveraged to perform single-channel noise suppression in the frequency domain on an L?R signal obtained from the composite baseband signal and on the L+R signal. The channel quality measure is also used to control the application of a fast fading compensation process that replaces noisy segments of the L?R and L+R signal with replacement waveforms generated via waveform extrapolation.

Abstract: A system and method for processing of audio and speech signals is disclosed, which provide compatibility over a range of communication devices operating at different sampling frequencies and/or bit rates. The analyzer of the system divides the input signal in different portions, at least one of which carries information sufficient to provide intelligible reconstruction of the input signal. The analyzer also encodes separate information about other portions of the signal in an embedded manner, so that a smooth transition can be achieved from low bit-rate to high bit-rate applications. Accordingly, communication devices operating at different sampling rates and/or bit-rates can extract corresponding information from the output bit stream of the analyzer. In the present invention embedded information generally relates to separate parameters of the input signal, or to additional resolution in the transmission of original signal parameters.

Abstract: A system that utilizes closed-form solutions to perform echo cancellation is described. The system includes a filter, filter parameter determination logic and a combiner. The filter is configured to process a far-end audio signal in accordance with one or more filter parameters to generate an estimated echo signal. The filter parameter determination logic is configured to update estimated statistics associated with the far-end audio signal and a microphone signal based on instantaneous statistics associated with the far-end audio signal and the microphone signal, and calculate the one or more filter parameters based upon the updated estimated statistics. The combiner is configured to generate an estimated near-end audio signal by subtracting the estimated echo signal from the microphone signal.