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:
Methods, systems, and apparatuses are described for performing speaker-identification-assisted speech processing in a downlink 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 far-end speaker participating in a voice call with a user of the communication device. Knowledge of the identity of the far-end speaker is then used to improve the performance of one or more downlink speech processing algorithms implemented on the communication device.
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:
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:
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:
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:
Methods, systems, and apparatuses are described for determining relative locations of wireless loudspeakers and performing channel mapping thereof Δn 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. Based on the determined locations, the audio processing component generates indications of correct or incorrect wireless loudspeaker placements, and performs audio channel mapping.
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:
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 system and method for modifying the size of data packets transmitted over a packet network in a manner that avoids overloading of the network. The system and method involves monitoring one or more parameters indicative of an amount of bandwidth being utilized on the packet network, responsive to the monitoring, determining that a level of bandwidth utilization on the packet network has changed, responsive to the determination that the level of bandwidth utilization on the packet network has changed, issuing a command to change the size of packets used for carrying data from a first packet size to a second packet size.