Abstract: A method, a device, and a non-transitory storage medium are described in which a full duplex system may use information from a user of the device as a basis to configure parameters of a dynamic range controller system. The parameters may include high pass filter values, compression values, and masking values. The full duplex system may use a pre-distortion filter for generating an estimated echo feedback reference signal. For generating the estimated echo feedback reference signal, the pre-distortion filter uses an echo coupled reference signal via microphone path as a reference signal, and an echo feedback reference signal via speaker feedback path as the input. An echo canceller of the full duplex system may use the estimated echo feedback reference signal, which takes echo path distortion into consideration, for cancelling both linear and non-linear echo.

Abstract: A method, a device, and a non-transitory storage medium are described in which a full duplex system may use information from a user of the device as a basis to configure parameters of a dynamic range controller system. The parameters may include high pass filter values, compression values, and masking values. The full duplex system may use a pre-distortion filter for generating an estimated echo feedback reference signal. For generating the estimated echo feedback reference signal, the pre-distortion filter uses an echo coupled reference signal via microphone path as a reference signal, and an echo feedback reference signal via speaker feedback path as the input. An echo canceller of the full duplex system may use the estimated echo feedback reference signal, which takes echo path distortion into consideration, for cancelling both linear and non-linear echo.

Abstract: A device may provide a baseline uplink audio signal and a baseline downlink audio signal to create a double talk communication for a user device. The device may receive an audio signal, transmitted by the user device, based on providing the baseline uplink audio signal and the baseline downlink audio signal. The device may determine a score based on the audio signal. The score may indicate an effectiveness of the user device with regard to processing audio associated with the double talk communication. The device may provide information that identifies the score to indicate the effectiveness of the user device with regard to processing audio associated with the double talk communication.

Abstract: A communication device may determine total gain for a downlink processing path via the communication device. The total gain may be determined based on determination of audio requirements for each of a plurality of audio processing devices supported via the communication device, during the downlink processing path. The communication device is operable to determine a default mix signal gain, a downlink gain adjustment and a calibration gain based on the determined audio requirements, and to calculate the total gain for the downlink processing path based on the determined default mix signal gain, the downlink gain adjustment and the calibration gain. The determination of the total gain may be based on determination of a telephony state in the communication device. The calibration gain may be determined during calibration of the communication device, whereas the downlink gain adjustment may be calculated based on an adjustment step-size and/or a mode based gain index.

Abstract: A telecommunication system including a fall duplex speakerphone, comprising a first microphone to generate a coupled signal including uplink information and non-linear distortion sensed by the first microphone in a speaker phone mode, a second microphone to generate a reference signal including downlink information and the non-linear distortion sensed by the second microphone in the speaker phone mode, and an acoustic echo canceller (AEC) to receive the coupled signal from the first microphone, to receive the reference signal from the second microphone, and to cancel out the non-linear distortion included in the coupled signal based on the non-linear distortion included in the reference signal.

Abstract: A VoIP access point may be operable to provide VoIP servicing to a plurality of wireless audio endpoint devices. The VoIP access point may extract VoIP audio data received via IP backbone and communicating the extracted audio data as non-VoIP formatted data to the wireless audio endpoint devices. In the uplink direction, the VoIP access point may receive non-VoIP formatted audio data from the wireless audio endpoint devices and pack the received data into IP packets for VoIP communication. The VoIP access point may also be operable to perform PCM encoding/decoding operations during VoIP servicing operations. The wireless audio endpoint devices may perform audio processing during VoIP communications via the VoIP access point. One or more intermediary communication devices may be utilized to route messaging between the VoIP access point and at least some of the wireless audio endpoint devices.

Abstract: Methods and systems for a dual echo canceller (EC) are disclosed and may include cancelling echo in utilizing a dual echo canceller, wherein said dual echo canceller includes an active echo canceller and an adaptive echo canceller. Filter coefficients may be copied from the adaptive echo canceller to the active echo canceller for the cancellation, based on whether said adaptive echo canceller has converged. The coefficients may be copied utilizing copy logic, which may comprise divergence detection and/or echo path change detection. The coefficients may be reset to default settings utilizing the copy logic. The coefficients may be calculated utilizing normalized block least mean squares (NBLMS), and may be calculated when the NBLMS is enabled by update logic. The coefficients may be calculated utilizing linear predictive coefficient (LPC) filtered uplink and downlink signals.

Abstract: Methods and systems for dynamic range control in an audio processing system are disclosed and may include controlling a dynamic range of an audio signal by expanding the dynamic range utilizing a dynamic expander, and dividing the audio signal into a plurality of frequency bands. Each of the bands may be individually compressed utilizing a multi-band compressor. A sum of the individually compressed frequency bands may be compressed utilizing a full-band compressor. The audio signal may be filtered utilizing a pre-emphasis filter, such as an infinite impulse response filter and may be divided into frequency bands utilizing one or more finite impulse response filters and/or delay modules. The dynamic expander may include adaptive thresholds and an envelope detector. Each of the frequency bands may be compressed utilizing syllabic compression in the multi-band compressor. The compressed sum of compressed plurality of bands may be processed utilizing an audio CODEC.

Abstract: In a method and apparatus for measuring voice quality of an endpoint in a Voice-over-IP (VoIP) network, a plurality of network quality parameters are determined for the endpoint. A voice quality factor for the endpoint is determined in real time based on the network quality parameters.

Abstract: A telecommunication system including a fall duplex speakerphone, comprising a first microphone to generate a coupled signal including uplink information and non-linear distortion sensed by the first microphone in a speaker phone mode, a second microphone to generate a reference signal including downlink information and the non-linear distortion sensed by the second microphone in the speaker phone mode, and an acoustic echo canceller (AEC) to receive the coupled signal from the first microphone, to receive the reference signal from the second microphone, and to cancel out the non-linear distortion included in the coupled signal based on the non-linear distortion included in the reference signal.

Abstract: A communication device may determine total gain for a downlink processing path via the communication device. The total gain may be determined based on determination of audio requirements for each of a plurality of audio processing devices supported via the communication device, during the downlink processing path. The communication device is operable to determine a default mix signal gain, a downlink gain adjustment and a calibration gain based on the determined audio requirements, and to calculate the total gain for the downlink processing path based on the determined default mix signal gain, the downlink gain adjustment and the calibration gain. The determination of the total gain may be based on determination of a telephony state in the communication device. The calibration gain may be determined during calibration of the communication device, whereas the downlink gain adjustment may be calculated based on an adjustment step-size and/or a mode based gain index.

Abstract: Methods and systems for echo estimation and cancellation are disclosed and may include estimating combined echo return loss and echo return loss enhancement (ERL+ERLE). A subband gain vector may be calculated utilizing non-linear processing, subband analysis, and the estimated ERL+ERLE to mitigate residual echo. ERL+ERLE may be estimated by averaging a difference in DL and UL signals. A maximum value of ERL+ERLE may be determined over a period of time. A non-linear distortion adjustment factor may be estimated for ERL+ERLE. An ERL+ERLE estimation error may be calibrated specific to the wireless device. The estimating of ERL+ERLE may be suspended briefly after a transition in the DL or UL signals. Comfort noise may be added to mask the residual echo, which may be mitigated following a dual echo canceller. The estimating may be suspended when the DL signal is not present. A noise level may be included in the gain calculation.

Abstract: Methods and systems for a dual echo canceller (EC) are disclosed and may include cancelling echo in utilizing a dual echo canceller, wherein said dual echo canceller includes an active echo canceller and an adaptive echo canceller. Filter coefficients may be copied from the adaptive echo canceller to the active echo canceller for the cancellation, based on whether said adaptive echo canceller has converged. The coefficients may be copied utilizing copy logic, which may comprise divergence detection and/or echo path change detection. The coefficients may be reset to default settings utilizing the copy logic. The coefficients may be calculated utilizing normalized block least mean squares (NBLMS), and may be calculated when the NBLMS is enabled by update logic. The coefficients may be calculated utilizing linear predictive coefficient (LPC) filtered uplink and downlink signals.

Abstract: A VoIP access point may be operable to provide VoIP servicing to a plurality of wireless audio endpoint devices. The VoIP access point may extract VoIP audio data received via IP backbone and communicating the extracted audio data as non-VoIP formatted data to the wireless audio endpoint devices. In the uplink direction, the VoIP access point may receive non-VoIP formatted audio data from the wireless audio endpoint devices and pack the received data into IP packets for VoIP communication. The VoIP access point may also be operable to perform PCM encoding/decoding operations during VoIP servicing operations. The wireless audio endpoint devices may perform audio processing during VoIP communications via the VoIP access point. One or more intermediary communication devices may be utilized to route messaging between the VoIP access point and at least some of the wireless audio endpoint devices.

Abstract: Methods and systems for dynamic range control in an audio processing system are disclosed and may include controlling a dynamic range of an audio signal by expanding the dynamic range utilizing a dynamic expander, and dividing the audio signal into a plurality of frequency bands. Each of the bands may be individually compressed utilizing a multi-band compressor. A sum of the individually compressed frequency bands may be compressed utilizing a full-band compressor. The audio signal may be filtered utilizing a pre-emphasis filter, such as an infinite impulse response filter and may be divided into frequency bands utilizing one or more finite impulse response filters and/or delay modules. The dynamic expander may include adaptive thresholds and an envelope detector. Each of the frequency bands may be compressed utilizing syllabic compression in the multi-band compressor. The compressed sum of compressed plurality of bands may be processed utilizing an audio CODEC.

Abstract: In a method and apparatus for measuring voice quality of an endpoint in a Voice-over-IP (VoIP) network, a plurality of network quality parameters are determined for the endpoint. A voice quality factor for the endpoint is determined in real time based on the network quality parameters.