Abstract: A method in a soundfield-capturing endpoint and the capturing endpoint that comprises a microphone array capturing soundfield, and an input processor pre-processing and performing auditory scene analysis to detect local sound objects and positions, de-clutter the sound objects, and integrate with auxiliary audio signals to form a de-cluttered local auditory scene that has a measure of plausibility and perceptual continuity. The input processor also codes the resulting de-cluttered auditory scene to form coded scene data comprising mono audio and additional scene data to send to others. The endpoint includes an output processor generating signals for a display unit that displays a summary of the de-cluttered local auditory scene and/or a summary of activity in the communication system from received data, the display including a shaped ribbon display element that has an extent with locations on the extent representing locations and other properties of different sound objects.

Abstract: Sound source localization apparatuses and methods are described. A frame amplitude difference vector is calculated based on short time frame data acquired through an array of microphones. The frame amplitude difference vector reflects differences between amplitudes captured by microphones of the array during recording the short time frame data. Similarity between the frame amplitude difference vector and each of a plurality of reference frame amplitude difference vectors is evaluated. Each of the plurality of reference frame amplitude difference vectors reflects differences between amplitudes captured by microphones of the array during recording sound from one of a plurality of candidate locations. A desired location of sound source is estimated based at least on the candidate locations and associated similarity. The sound source localization can be performed based at least on amplitude difference.

Abstract: A method (800) for determining an estimate (215, 261) of an echo path property of an electronic device (200, 250, 300, 600). The electronic device is configured to render a total audio signal using a loudspeaker (102), and the electronic device is configured to record an echo of the rendered audio signal using a microphone (103), thereby yielding a recorded audio signal (112). The method comprises inserting (801), in an inaudible manner, an auxiliary audio signal (212) into the total audio signal to be rendered; wherein the auxiliary audio signal (212) comprises a tonal audio signal at a first frequency; isolating (803) the echo of the auxiliary audio signal (212) from the recorded audio signal (112); and determining (804) the estimate (215, 261) of the echo path property based on the inserted auxiliary audio signal (212) and based on the isolated echo of the auxiliary audio signal (212).

Abstract: An audio processing apparatus and an audio processing method are described. In one embodiment, the audio processing apparatus include an audio masker separator for separating from a first audio signal an audio material comprising a sound other than stationary noise and utterance meaningful in semantics, as an audio masker candidate. The apparatus also includes a first context analyzer for obtaining statistics regarding contextual information of detected audio masker candidates, and a masker library builder for building a masker library or updating an existing masker library by adding, based on the statistics, at least one audio masker candidate as an audio masker into the masker library, wherein audio maskers in the maker library are used to be inserted into a target position in a second audio signal to conceal defects in the second audio signal.

Abstract: In an conferencing system in which a plurality of communication devices electronically connect respective participants to one another, a method for mitigating the effects of substantially concurrent audio bursts from two or more of the participants includes identifying a priority attribute associated with each of multiple substantially concurrent audio bursts, comparing the identified priority attributes, and electronically suppressing at least one audio burst as a function of the comparison.

Abstract: A system, a method, logic embodied in a computer-readable medium, and a computer-readable medium comprising instructions that when executed carry out a method. The method processes: (a) a plurality of input signals, e.g., signals from a plurality of spatially separated microphones; and, for echo suppression, (b) one or more reference signals, e.g., signals from or to be rendered by one or more loudspeakers and that can cause echoes. The method processes the input signals and one or more reference signals to carry out in an integrated manner simultaneous noise suppression and out-of-location signal suppression, and in some versions, echo suppression.

Abstract: A system and method for initiating conference calls with external devices are disclosed. Call participants are sent conference invitation and conference information regarding the designated conference call. This conference information is stored on the participant's external device. When the participants arrive at a conference call location having a conferencing device, the conferencing device is capable of communicating with the external device, initiating communications, exchanging conference information. If the participant is verified and/or authorized, the conference system may send the IP address of the conference device to the conference system to initiate the conference call. In one embodiment, the conference device uses an ultrasound acoustic communication band to initiate the call with the external device on a semi-automated basis. An acoustic signature comprising a pilot sequence for communications synchronization may be generated to facilitate the call.

Abstract: The present document relates to audio conference systems. In particular, the present document relates to the mapping of soundfields within an audio conference system. A conference multiplexer (110, 175, 210, 400) configured to place a first input soundfield signal (402) originating from a first soundfield endpoint (120, 170) within a 2D or 3D conference scene (300) to be rendered to a listener (301) is described. The first input soundfield signal (402) is indicative of a soundfield captured by the first soundfield endpoint (120, 170).

Abstract: The present document relates to audio conference systems. In particular, the present document relates to improving the perceptual continuity within an audio conference system. According to an aspect, a method for multiplexing first and second continuous input audio signals is described, to yield a multiplexed output audio signal which is to be rendered to a listener. The first and second input audio signals (123) are indicative of sounds captured by a first and a second endpoint (120, 170), respectively. The method comprises determining a talk activity (201, 202) in the first and second input audio signals (123), respectively; and determining the multiplexed output audio signal based on the first and/or second input audio signals (123) and subject to one or more multiplexing conditions.

Abstract: Embodiments of client device and method for audio or video conferencing are described. An embodiment includes an offset detecting unit, a configuring unit, an estimator and an output unit. The offset detecting unit detects an offset of speech input to the client device. The configuring unit determines a voice latency from the client device to every far end. The estimator estimates a time when a user at the far end perceives the offset based on the voice latency. The output unit outputs a perceivable signal indicating that a user at the far end perceives the offset based on the time estimated for the far end. The perceivable signal is helpful to avoid collision between parties.

Abstract: In some embodiments, a method for adaptive control of gain applied to an audio signal, including steps of analyzing segments of the signal to identify audio objects (e.g., voices of participants in a voice conference); storing information regarding each distinct identified object; using at least some of the information to determine at least one of a target gain, or a gain change rate for reaching a target gain, for each identified object; and applying gain to segments of the signal indicative of an identified object such that the gain changes (typically, at the gain change rate for the object) from an initial gain to the target gain for the object. The information stored may include a scene description. Aspects of the invention include a system configured (e.g., programmed) to per form any embodiment of the inventive method.

Abstract: A method for encoding sound field signals includes allocating coding rate by application of a uniform criterion to all subbands of all signals in a joint process. An allocation criterion may be based on a comparison, in a given subband, between a spectral envelope of the signals to be encoded and a coding noise profile, wherein the noise profile may be a sum of a noise shape and a noise offset, which noise offset is computed on the basis of the coding bit budget. The rate allocation process may be combined with an energy-compacting orthogonal transform, for which there is proposed a parameterization susceptible of efficient coding and having adjustable directivity. In a further aspect, the invention provides a corresponding decoding method.

Abstract: A method of post-processing banded gains for applying to an audio signal, an apparatus to post-processed banded gains, and a tangible computer-readable storage medium comprising instructions that when executed carry out the method. The banded gains are determined by input processing one or more input audio signals. The method includes post-processing the banded gains to generate post-processed gains, generating a particular post-processed gain for a particular frequency band including percentile filtering using gain values from one or more previous frames of the one or more input audio signals and from gain values for frequency bands adjacent to the particular frequency band.

Abstract: The present document relates to audio communication systems. In particular, the present document relates to the control of the level of audio signals within audio communication systems. A method for leveling a near-end audio signal (211) using a leveling gain (214) is described. The near-end audio signal (211) comprises a sequence of segments, wherein the sequence of segments comprises a current segment and one or more preceding segments.

Abstract: A method (400) for determining that an audio conference is in a first phase of a plurality of phases. The method comprises determining (401) a plurality of sequences (301) of events (305) for a plurality of terminals (120, 170); wherein an event from the sequences (301) of events (305) indicates that a talker activity at a terminal (120, 170) has been detected; determining (403) a sequence (350) of feature vectors (330) based on the plurality of sequences (301) of events (305); wherein a feature vector (330) of the sequence (350) of feature vectors (330) is indicative of the talker activity (332) of at least one of the plurality of terminals (120, 170) relative to the talker activity at least another one of the plurality of terminals (120, 170); and determining (406) that the audio conference is in the first phase based on the sequence (350) of feature vectors (330).

Abstract: Described are: a method, an apparatus, and a tangible computer-readable storage medium comprising instructions to instruct one or more processors to carry out a method. One set of methods is for the transmit side of a communication link and another set of methods is for the receive side. A transmit side method includes assigning one of a set of classifications to media, e.g., voice/audio packets transmitted in a sequence, different classifications impacting differently a measure of perceptual quality calculated at the receive side if packets of the respective classifications are lost. A present packet is sent to the receive side containing the classification of a previous packet.

Abstract: One or more audio signals are processed using a multi-stage (hierarchical) voice and/or signal activity detector (VAD/SAD). A first stage is capable of reducing the workload bandwidth by employing an inexpensive VAD/SAD processor. One or more subsequent stages may further process the audio signals from the first stage. Other implementations may include a first stage that also performs continuity preservation between last blocks of audio signal and the first blocks of audio after it is detected that relevant audio signals are resumed. In yet other implementations, the first stage may extract features from audio signals when they are presented in their coded domain, and possibly with little or no decoding of the audio signal.

Abstract: Digital media communication systems and methods are disclosed in which a set of end points and/or clients are in communication with a set of servers. The system is capable of determining which end point is a dominant user and/or speaker. When the system determines that an end point is dominant, then the digital data that emanates from dominant end point is preferred. Such preferred digital data may be forwarded asynchronously and possible ahead of digital data from other, non-dominant end points. Synchronous and/or substantially contemporaneous digital data from other end points may be mixed that such that, when rendered, gives substantially the same effect real life superposition of signals. The system may also toggle on or off between states of asynchronous forward and synchronous-only modes.

Abstract: A method (400) for determining that an audio conference is in a first phase of a plurality of phases. The method comprises determining (401) a plurality of sequences (301) of events (305) for a plurality of terminals (120, 170); wherein an event from the sequences (301) of events (305) indicates that a talker activity at a terminal (120, 170) has been detected; determining (403) a sequence (350) of feature vectors (330) based on the plurality of sequences (301) of events (305); wherein a feature vector (330) of the sequence (350) of feature vectors (330) is indicative of the talker activity (332) of at least one of the plurality of terminals (120, 170) relative to the talker activity at least another one of the plurality of terminals (120, 170); and determining (406) that the audio conference is in the first phase based on the sequence (350) of feature vectors (330).

Abstract: In some embodiments, a method for processing output of at least one microphone of a device (e.g., a headset) to identify at least one touch gesture exerted by a user on the device, including by distinguishing the gesture from input to the microphone other than a touch gesture intended by the user, and by distinguishing between a tap exerted by the user on the device and at least one dynamic gesture exerted by the user on the device, where the output of the at least one microphone is also indicative of ambient sound (e.g., voice utterences). Other embodiments are systems for detecting ambient sound (e.g., voice utterences) and touch gestures, each including a device including at least one microphone and a processor coupled and configured to process output of each microphone to identify at least one touch gesture exerted by a user on the device.