Broadcom Corporation

Avago Technologies General IP (singapore) Pte. Ltd.

Wilfrid Paul LeBlanc

Jes Thyssen

Juin-Hwey Chen

Franck Beaucoup

Elias Nemer

Darwin Rambo

Mohammad Reza Zad-Issa

Aleksander Radisavljevic

Sumit Sanyal

Tommy Kee

Syavosh Zad-Issa

Taro Umezawa

A packet voice transceiver adapted to reside at a first end of a communication network and to send and receive voice packets to and from a second packet voice transceiver residing at a second end of the communication network. The transceiver includes a voice encoder, a packet transmitter, a packet receiver, a voice decoder and a far-end echo canceller. The voice encoder receives an ingress audio signal and encodes the audio signal to produce an ingress packet voice signal. The packet transmitter transmits the ingress packet voice signal over the network to the second packet voice transceiver. The packet receiver receives an egress packet voice signal transmitted over the network by the second packet voice transceiver. The voice decoder decodes the egress packet voice signal to produce an egress audio signal. The far-end echo canceller reduces echo that is present in the egress audio signal and that originated at the second end.

Packet voice system with far-end echo cancellation

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.

Speech intelligibility enhancement system and method

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.

Integrated speech intelligibility enhancement system and acoustic echo canceller

A system performs residual echo suppression on a microphone signal that receives, e.g., voice commands, and that is exposed to echo from multiple speakers. An example is a smartphone that receives voice commands while the smartphone is playing music through stereo speakers. The system estimates residual echo level in different ways, and determines which estimate to use. The technique responds well to the difficult to handle scenario of a spatially quiescent image suddenly transitioning to a spatially rich image. Even in the face of such difficult scenarios, the system detects and removes residual echo from the microphone signal, instead of allowing the undesired residual echo to pass through.

Spatial quiescence protection for multi-channel acoustic echo cancellation

Compression for speech intelligibility enhancement

A technique for suppressing non-stationary noise, such as wind noise, in an audio signal is described. In accordance with the technique, a series of frames of the audio signal is analyzed to detect whether the audio signal comprises non-stationary noise. If it is detected that the audio signal comprises non-stationary noise, a number of steps are performed. In accordance with these steps, a determination is made as to whether a frame of the audio signal comprises non-stationary noise or speech and non-stationary noise. If it is determined that the frame comprises non-stationary noise, a first filter is applied to the frame and if it is determined that the frame comprises speech and non-stationary noise, a second filter is applied to the frame.

Single-microphone wind noise suppression

A beamforming technique for a microphone array is described to attenuate a source of undesired noise that is deemed the most limiting to audio quality in an acoustic environment. Possible sources of undesired noise include echo, background noise (stationary) and other interference signals (non-stationary). The beamforming technique is updated based on long-term evaluations. Once an evaluation occurs and a decision is made, the beamformer adapts with a maximum responsiveness and without intentional delay, and therefore not affecting the beamformer's tracking ability. When fixed beamforming is utilized, one of several fixed beamformers having different attenuation targets are selected to implement noise attenuation. When adaptive beamforming is utilized, the beamformer adapts whenever the selected target is deemed dominant.

Beamforming apparatus and method based on long-term properties of sources of undesired noise affecting voice quality

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.

Loudness enhancement system and method

Spectral shaping for speech intelligibility enhancement

Methods and systems for a subband acoustic echo canceller with integrated voice activity detection are disclosed and may include adjusting transmit and/or receive powers of wirelessly communicated audio signals based on voice activity detection via subband analysis of the wirelessly communicated audio signals. The receive power may be adjusted by utilizing a reduced duty cycle, or by conveying voice activity detection information via an asynchronous control channel in a Bluetooth application. A plurality of subbands may be generated utilizing a fast Fourier transform, and a first subset of the subbands corresponding to voice activity may be selected and a second subset of the subbands may be selected that corresponds to background noise. The processing of the subsets may be dynamically adjusted due to variations in the voice activity or background noise. Comfort noise may be generated and transmitted at a reduced bandwidth utilizing the second subset of the subbands.

Method and system for a subband acoustic echo canceller with integrated voice activity detection

A packet voice transceiver adapted to reside at a first end of a communication network and to send an ingress communication signal comprising voice packets to, and receive an egress communication signal comprising voice packets from, a second packet voice transceiver residing at a second end of the communication network. The packet voice transceiver includes a far-end echo canceller that reduces echo that is present in the egress communication signal. The far-end communicates with other functional components of the transceiver system and adapts its behavior based on the activity of the other functional components.

System and method for operating a packet voice far-end echo cancellation system

Methods, systems, and computer program products for enhancing transmitted audio signals for a video conference call are disclosed. Facial recognition technology and audio processing techniques determine a position of a speaking member of a party on a video conference. Based on the determined position of the speaking member, the directionality of the received audio signals can be adjusted towards the speaking member. The video conference system can transmit the adjusted audio signals.

Audio processing for video conferencing

A technique is provided for limiting distortion of an audio signal being processed for playback by an audio device. In accordance with the technique, the audio signal is compressed to generate a compressed audio signal having a level that does not exceed a compression limit. The compressed audio signal is then soft clipped signal to generate a soft-clipped audio signal having a level that does not exceed a soft clipping limit, wherein the compression limit exceeds the soft clipping limit. The technique may also include passing the audio signal through a shaping filter prior to compressing the audio signal, wherein passing the audio signal through a shaping filter comprises modifying the level of selected frequency components of the audio signal.

Audio signal shaping for playback by audio devices

A method and system are provided in which a device, such as an acoustic echo canceller, may reduce the residual echo that may be heard at the far end of a conversation when an external speaker volume is changed. The device may compute a gain based on an echo estimate produced by a filter and on a near-end signal comprising audio information. The gain may be based on a correlation of the echo estimate and the near-end signal that tracks the changes in volume. Once computed, the gain may be validated to ensure that it is being applied when appropriate. The echo estimate may be adjusted by first applying the gain to an output of the filter and subsequently scaling a value of each of the coefficients of the filter based on the gain. The gain may be smoothed out over consecutive frames based on several adaptation schemes.

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Wilfrid Paul LeBlanc