Abstract: Systems and methods are described for detecting and remedying potential incongruence in a video conference. A camera of a video conferencing system may capture video images of a conference room. A processor of the video conferencing system may identify locations of a plurality of participants within an image plane of a video image. Using face and shape detection, a location of a center point of each identified participant's torso may be calculated. A region of congruence bounded by key parallax lines may be calculated, the key parallax lines being a subset of all parallax lines running through the center points of each identified participant. When the audio device location is not within the region of congruence, audio captured by an audio device may be adjusted to reduce effects of incongruence when the captured audio is replayed at a far end of the video conference.

Abstract: A computer implemented system for rendering captured audio soundfields to a listener comprises apparatus to deliver the audio soundfields to the listener. The delivery apparatus delivers the audio soundfields to the listener with first and second audio elements perceived by the listener as emanating from first and second virtual source locations, respectively, and with the first audio element and/or the second audio element delivered to the listener from a third virtual source location. The first virtual source location and the second virtual source location are perceived by the listener as being located to the front of the listener, and the third virtual source location is located to the rear or the side of the listener.

Abstract: In an audio processing system (300), a filtering section (350, 400): receives subband signals (410, 420, 430) corresponding to audio content of a reference signal (301) in respective frequency subbands; receives subband signals (411, 421, 431) corresponding to audio content of a response signal (304) in the respective subbands; and forms filtered inband references (412, 422, 432) by applying respective filters (413, 423, 433) to the subband signals of the reference signal.

Abstract: Teleconference audio data including a plurality of individual uplink data packet streams, may be received during a teleconference. Each uplink data packet stream may corresponding to a telephone endpoint used by one or more teleconference participants. The teleconference audio data may be analyzed to determine a plurality of suppressive gain coefficients, which may be applied to first instances of the teleconference audio data during the teleconference, to produce first gain-suppressed audio data provided to the telephone endpoints during the teleconference. Second instances of the teleconference audio data, as well as gain coefficient data corresponding to the plurality of suppressive gain coefficients, may be sent to a memory system as individual uplink data packet streams. The second instances of the teleconference audio data may be less gain-suppressed than the first gain-suppressed audio data.

Abstract: A method of processing data comprises processing first frequency-domain audio or video data using signal processing of a first type, and transforming the processed first frequency-domain audio or video data to processed time-domain audio or video data using a transform which is the inverse of the first transform, and transforming the processed time-domain audio or video data using a second transform which is matched to a second type of signal processing. The method further comprises identifying time-domain audio or video data for which signal processing of the first type, after transformation using the second transform, would yield satisfactory results. The method further comprises transforming the identified time-domain audio or video data to frequency-domain identified audio or video data using the second transform, instead of using the first transform, and processing the identified frequency-domain audio or video data using signal processing of the first type.

Abstract: Described herein is a method and audio capture system for suppressing noise due to mechanical disturbances in an audio capture system. In one embodiment a method includes the steps of: a) receiving an input audio signal from a microphone; b) receiving an input reference signal separate from the input audio signal; c) processing the input reference signal to generate a control signal for identifying a noise event; d) in response to the control signal, estimating a contribution of the noise event to the input audio signal by applying an adaptive filter to a sliding temporal window of the control signal in the frequency domain; and e) based on the estimated contribution, selectively modifying the input audio signal to generate an output audio signal in which the noise arising from the noise event is at least partially suppressed.

Abstract: An audio communication endpoint receives a bitstream containing spectral components representing spectral content of an audio signal, wherein the spectral components relate to a first range extending up to a first break frequency, above which any spectral components are unassigned. The endpoint adapts the received bitstream in accordance with a second range extending up to a second break frequency by removing moving spectral components or adding neutral-valued spectral components relating to a range between the first and second break frequencies. The endpoint then attenuates spectral content in a neighbourhood of the least of the first and second break frequencies for thereby achieving a gradual spectral decay. After this, reconstructing the audio signal is reconstructed by an inverse transform operating on spectral components relating to said second range in the adapted and attenuated received bitstream.

Abstract: Systems and methods are described for determining orientation of an external audio device in a video conference, which may be used to provide congruent multimodal representation for a video conference. A camera of a video conferencing system may be used to detect a potential location of an external audio device within a room in which the video conferencing system is providing a video conference. Within the detected potential location, a visual pattern associated with the external audio device may be identified. Using the identified visual pattern, the video conferencing system may estimate an orientation of the external audio device, the orientation being used by the video conferencing system to provide spatial audio video congruence to a far end audience.

Abstract: Example embodiments disclosed herein relate to assessment and adjustment for an audio environment. A computer-implemented method is provided. The method includes obtaining a first audio signal captured by a device located in an environment. The method also includes analyzing a characteristic of the first audio signal to determine an acoustic performance metric for the environment. The method further includes, in response to the acoustic performance metric being below a threshold, providing a first task for a user to perform based on the characteristic of the first audio signal. The first task is related to an adjustment to a setting of the environment. Embodiments in this regard further provide a corresponding computer program product. Corresponding system and computer program product are also disclosed.

Abstract: Systems and methods are described for concealing packet loss in a received audio stream. Packets of the audio stream may be received in a non-lapped transform domain format, where at least one packet is missing in the stream. The received packets are decoded, and each missing packet in the decoded stream is replaced by a reduced-energy signal block. Each reduced-energy signal block may also be modified at a beginning or ending boundary, and shifted such that a start or end of each missing packet does not coincide with a peak of a transform window of a lapped transform domain format. The raw audio signal may then be encoded into transform windows having the lapped transform domain format. Packet loss concealment may then be performed for selected transform windows that include modified reduced-energy blocks, either prior to transmission or after transmission by the receiving endpoint.

Abstract: A method, an apparatus, and logic to post-process raw gains determined by input processing to generate post-processed gains, comprising using one or both of delta gain smoothing and decision-directed gain smoothing. The delta gain smoothing comprises applying a smoothing filter to the raw gain with a smoothing factor that depends on the gain delta: the absolute value of the difference between the raw gain for the current frame and the post-processed gain for a previous frame. The decision-directed gain smoothing comprises converting the raw gain to a signal-to-noise ratio, applying a smoothing filter with a smoothing factor to the signal-to-noise ratio to calculate a smoothed signal-to-noise ratio, and converting the smoothed signal-to-noise ratio to determine the second smoothed gain, with smoothing factor possibly dependent on the gain delta.

Abstract: Described herein is a method for creating an object-based audio signal from an audio input, the audio input including one or more audio channels that are recorded to collectively define an audio scene. The one or more audio channels are captured from a respective one or more spatially separated microphones disposed in a stable spatial configuration. The method includes the steps of: a) receiving the audio input; b) performing spatial analysis on the one or more audio channels to identify one or more audio objects within the audio scene; c) determining contextual information relating to the one or more audio objects; d) defining respective audio streams including audio data relating to at least one of the identified one or more audio objects; and e) outputting an object-based audio signal including the audio streams and the contextual information.

Abstract: In an audio processing system (300), a filtering section (350, 400): receives subband signals (410, 420, 430) corresponding to audio content of a reference signal (301) in respective frequency subbands; receives subband signals (411, 421, 431) corresponding to audio content of a response signal (304) in the respective subbands; and forms filtered inband references (412, 422, 432) by applying respective filters (413, 423, 433) to the subband signals of the reference signal.

Abstract: A method of processing a series of microphone inputs of an audio conference, the method including the steps of: (a) conducting a spatial analysis and feature extraction of the audio conference based on current audio activity; (b) aggregating historical information to obtain information about the approximate relative location of recent sound objects relative to the series of microphone inputs; (c) utilizing the relative location or distance of the sound objects from the series of microphone inputs to determine if beam forming should be utilized to enhance the audio reception from recent sound objects.

Abstract: Example embodiments disclosed herein relate to a estimation of reverberant energy components from audio sources. A method of estimating a reverberant energy component from an active audio source (100) is disclosed. The method comprises determining a correspondence between the active audio source and a plurality of sample sources by comparing one or more spatial features of the active audio source with one or more spatial features of the plurality of sample sources, each of the sample sources being associated with an adaptive filtering model (101); obtaining an adaptive filtering model for the active audio source based on the determined correspondence (102); and estimating the reverberant energy component from the active audio source over time based on the adaptive filtering model (103). Corresponding system (800) and computer program product (900) are also disclosed.

Abstract: Systems and methods for providing forward error correction for a multi-channel audio signal are described. Blocks of an audio stream are buffered into a frame. A transformation can be applied that compacts the energy of each block into a plurality of transformed channels. The energy compaction transform may compact the most energy of a block into the first transformed channel and to compact decreasing amounts of energy into each subsequent transformed channel. The transformed frame may be encoded using any suitable codec and transmitted in a packet over a network. Improved forward error correction may be provided by attaching a low bit rate encoding of the first transformed channel to a subsequent packet. To reconstruct a lost packet, the low bit rate encoding of the first channel for the lost packet may be combined with a packet loss concealment version of the other channels, constructed from a previously-received packet.

Abstract: Systems and methods are described for determining orientation of an external audio device in a video conference, which may be used to provide congruent multimodal representation for a video conference. A camera of a video conferencing system may be used to detect a potential location of an external audio device within a room in which the video conferencing system is providing a video conference. Within the detected potential location, a visual pattern associated with the external audio device may be identified. Using the identified visual pattern, the video conferencing system may estimate an orientation of the external audio device, the orientation being used by the video conferencing system to provide spatial audio video congruence to a far end audience.

Abstract: A computer implemented system for rendering captured audio soundfields to a listener comprises apparatus to deliver the audio soundfields to the listener. The delivery apparatus delivers the audio soundfields to the listener with first and second audio elements perceived by the listener as emanating from first and second virtual source locations, respectively, and with the first audio element and/or the second audio element delivered to the listener from a third virtual source location. The first virtual source location and the second virtual source location are perceived by the listener as being located to the front of the listener, and the third virtual source location is located to the rear or the side of the listener.

Abstract: Systems and methods are described for analyzing and resolving feedback caused by having multiple audio links in a conference room. An audio system may detect the presence of duplicated audio caused by multiple audio links in the conference room. Marker signals may be injected into the conference room or over the network in response to detecting the duplicated audio. Echoes of the marker signals may be received, and the system may determine which case corresponds to the detected duplicated audio based on the received echo of the marker signals. Based on the determined case, operation of at least one of the speaker and the microphone may be modified. After the modification, audio playback in the conference room may be monitored to verify that far end audio playback is taking place and that the duplicated audio has been resolved.

Abstract: Example embodiments disclosed herein relate to impulsive noise suppression. A method of impulsive noise suppression in an audio signal is disclosed. The method includes determining an impulsive noise related feature from a current frame of the audio signal. The method also includes detecting an impulsive noise in the current frame based on the impulsive noise related feature, and in response to detecting the impulsive noise in the current frame, applying a suppression gain to the current frame to suppress the impulsive noise. Corresponding system and computer program product of impulsive noise suppression in an audio signal are also disclosed.