Abstract: 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.

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: 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 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.

Abstract: 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.

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: 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 cancels echo or refrains from canceling echo based on the activity of the other functional components.

Abstract: 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.

Abstract: 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.

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 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.

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: 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.

Abstract: A packet voice communication system having a jitter buffer external to a voice processor. The jitter buffer stores voice packets received from a packet network. The voice processor processes the voice packets from the jitter buffer. A jitter buffer processor may place an indicator in each voice packet it holds. The indicator can indicate a length of time the voice packet was held. The rate at which packets come from the jitter buffer may be based upon the indicator, a higher rate if holding times are high and a slower rate if low. The voice processor can store the voice packets in a packet queue prior to processing the voice packets. The rate voice packets come to the voice processor may be based upon how full the packet queue is, a higher rate if the packet queue is relatively empty and a slower rate if relatively full.

Abstract: A system and method for provide a stable gain from an adaptive gain control device in a signal path. An echo canceller is also located in the signal path, and is used to provide performance information regarding losses in the signal. This performance information is fed to the automatic gain control device via a connection. The automatic gain control device thereafter uses the performance information to determine a maximum gain that might be provided based upon losses cause by echo conditions. The gain however is limited in order to provide for a stable system. The performance information includes a loss rate that includes a combination of the echo return loss and the echo return loss enhancement.

Abstract: 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 cancels echo or refrains from canceling echo based on the activity of the other functional components.

Abstract: During playing of a streamed audio while engaging in a voice communication session, a communication device may be operable to generate a background audio during an echo cancellation operation for the voice communication session, using known information associated with the streamed audio. The streamed audio may comprise audio content that is received from one or more external sources and/or locally generated within the communication device. The known information associated with the streamed audio may comprise an echo replica corresponding to an echo associated with the streamed audio and/or a modified version of the streamed audio. In instances when a residual echo suppression is operational during the echo cancellation operation, the communication device may be operable to suppress a residual echo based on a signal level of the echo replica corresponding to the echo associated with the streamed audio and/or a signal level of the modified version of the streamed audio.

Abstract: 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 cancels echo or refrains from canceling echo based on the activity of the other functional components.