Polycom, Inc.

Eric David Elias

Peter L Chu

Kwan Kin Truong

Steven L Potts

Kiril Ratmanski

Erwin Goesnar

Venugopal Kalluri

Richard J Kos

The amount of far-field noise transmitted by a primary communication device in an open-plan office environment is reduced by defining an acoustic perimeter of reference microphones around the primary device. Reference microphones generate a reference audio input including far-field noise in the proximity of the primary device. The primary device generates a main audio input including the voice of the primary speaker as well as background noise. Reference audio input is compared to main audio input to identify the background noise portion of the main audio signal. A noise reduction algorithm suppresses the identified background noise in the main audio signal. The one or more reference microphones defining the acoustic perimeter may be included in separate microphone devices placed in proximity to the main desktop phone, microphones within other nearby desktop telephone devices, or a combination of both types of devices.

Acoustic perimeter for reducing noise transmitted by a communication device in an open-plan environment

A conferencing endpoint includes a loudspeaker, a base microphone, and a double-talk detection module which allows two-way communication between the conferencing endpoint and a remote endpoint only when participants at both endpoints are speaking at the same time, so as to minimize echo due to feedback. The double-talk detection module adds the energy of any distortion from the loudspeaker to the energy of the signal coming from the remote endpoint, and compares this combined energy with the energy of the base microphone to determine whether double-talk is present. The double-talk detection module is thus prevented from mistaking the feedback for near end talk at the endpoint.

Audio echo cancellation with robust double-talk detection in a conferencing environment

Various techniques are disclosed for improving packet loss concealment to reduce artifacts by using audio character measures of the audio signal. These techniques include attenuation to a noise fill instead of attenuation to silence, varying how long to wait before attenuating the extrapolation, varying the rate of attenuation of the extrapolation, attenuating periodic extrapolation at a different rate than non-periodic extrapolation, and performing period extrapolation on successively longer fill data based on the audio character measures, adjusting weighting between periodic and non-periodic extrapolation based on the audio character measures, and adjusting weighting between periodic extrapolation and non-periodic extrapolation non-linearly.

Artifact reduction in packet loss concealment

Various techniques are disclosed for reducing artifacts generated by time compression. by adapting the time compression based on the state of the received audio. The amount of time compression may be bounded based on audio characteristics. Another feature provides a way of determining the most correlated portions of segments of audio. Voiced speech may be distinguished from unvoiced speech. Another feature provides a way of distinguishing between silence, voiced speech, and unvoiced speech. Time compression may be adapted during periods of lengthy silence. Another feature allows for reducing time compression during sensitive portions of the received audio. One or more of these features may be present in different embodiments.

Artifact reduction in time compression

A communication system that includes multiple conferencing devices connected in a daisy-chain configuration is communicably connected to far-end conference participants. Multiple conferencing devices provide improved sound quality to the far-end participants by reducing aural artifacts resulting from reverberation and echo. The daisy-chain communication system also reduces the processing and transmission time of the near-end audio signal by processing and transmitting the audio signal in frequency domain. Each conferencing device in the daisy chain performs signal conditioning on its audio signal before transmitting it in the frequency domain to a mixer. The output signal of the mixer is converted back to the time domain before being transmitted to the far-end. The daisy-chain configuration also provides a distributed bridge to external communication devices that can be connected to each conferencing device.

Audio signal routing

Distributed bridging

Systems, methods, and devices for mixed voice and spread spectrum data signaling with enhanced concealment of data are disclosed. Audible artifacts may be reduced or suppressed in order to improve the audible quality of the sound. The distortion to the audio may be kept just below perceptibility, yet the data contained within the modulated signal is recoverable on the receiving side. The recovery of the data is robust to the impairments imposed by the communication channel. The disclosed systems, methods, and devices may be implemented in audio conferencing or video conferencing. Various embodiments of the present disclosure provide a data and voice mixer, which includes an improved spread spectrum data hiding transmitter. The transmitter may comprise a spreading encoder, a Phase Shift Keying (PSK) modulator, and a notch filter. A pseudo random switching pattern may be applied to a plurality of chip sequence generators in order to reduce or suppress an undesirable audible artifact. The transmitter may further comprise a phase randomizer operable to rotate the phase of the output of the spreading encoder by a pseudo random increment of approximately 45 degrees.

Mixed voice and spread spectrum data signaling with enhanced concealment of data

A method, a process and an audio system for embedding bit stream in a voice signal is disclosed. A voice/data interference signal is composed and supplied to the mixer of the voice signal and the data signal such that the resulting mixed signal effectively cancels the interference and allows the interference-free signal being transmitted to the audio signal receiver/decoder. The decoder may decode the data without interference errors. A method using profile shaping to mask the noise due to modulated carrier signal is also disclosed.

Voice interference correction for mixed voice and spread spectrum data signaling

Systems, methods, and devices for mixed voice and spread spectrum data signaling with spread spectrum data hiding with CDMA (Code Division Multiple Access) for three or more users. The disclosed systems, methods, and devices may be implemented in audio conferencing or video conferencing. Various embodiments of the present disclosure provide a data and voice mixer, which includes an improved spread spectrum data hiding transmitter. The transmitter may comprise a CDMA chip sequence generator and a spreading encoder. Different users may be accommodated by selecting different initializations for the registers of the CDMA chip sequence generator. Different spreading code may be used for different users which are near orthogonal to each other. In some embodiments, the CDMA chip sequence generator pseudo-randomly selects the chip sequence from one of a plurality of Gold chip sequence generators.

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Somerville, MA, United States of America

Eric David Elias