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: A communication system, method, and computer-readable medium therefor comprise a media server configured to receive a plurality of audio streams from a corresponding plurality of client devices, the media server including circuitry configured to rank the plurality of audio streams based on a predetermined metric, group a first portion of the plurality of audio streams into a first set, the first portion of the plurality of audio streams being the N highest-ranked audio streams, group a second portion of the plurality of audio streams into a second set, the second portion of the plurality of audio streams being the M lowest-ranked audio streams, forward respective audio streams of the first set to a receiver device, and discard respective audio streams of the second set, wherein N and M are independent integers.

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: 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: 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 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: A noise estimation method including steps of generating gap confidence values in response to microphone output and playback signals, and using the gap confidence values to generate an estimate of background noise in a playback environment. Each gap confidence value is indicative of confidence of presence of a gap at a corresponding time in the playback signal, and may be a combination of candidate noise estimates weighted by the gap confidence values. Generation of the candidate noise estimates may but need not include performance of echo cancellation. Optionally, noise compensation is performed on an audio input signal using the generated background noise estimate. Other aspects are systems configured to perform any embodiment of the noise estimation method.

Abstract: A noise estimation method including steps of generating gap confidence values in response to microphone output and playback signals, and using the gap confidence values to generate an estimate of background noise in a playback environment. Each gap confidence value is indicative of confidence of presence of a gap at a corresponding time in the playback signal, and may be a combination of candidate noise estimates weighted by the gap confidence values. Generation of the candidate noise estimates may but need not include performance of echo cancellation. Optionally, noise compensation is performed on an audio input signal using the generated background noise estimate. Other aspects are systems configured to perform any embodiment of the noise estimation method.

Abstract: A pervasive listening method including steps of inserting at least one forced gap in a playback signal (thus generating a modified playback signal), and during playback of the modified playback signal, monitoring non-playback content (e.g., including by generating an estimate of background noise) in a playback environment using output of a microphone in the playback environment. Optionally, the method includes generation of the playback signal, including by processing of (e.g., performing noise compensation on) an input signal using a result (e.g., a background noise estimate) of the monitoring of non-playback content. Other aspects are systems configured to perform any embodiment of the pervasive listening method.

Abstract: An audio processing method may involve receiving media input audio data corresponding to a media stream and headphone microphone input audio data, determining a media audio gain for at least one of a plurality of frequency bands of the media input audio data and determining a headphone microphone audio gain for at least one of a plurality of frequency bands of the headphone microphone input audio data. Determining the headphone microphone audio gain may involve determining a feedback risk control value, for at least one of the plurality of frequency bands, corresponding to a risk of headphone feedback between at least one external microphone of a headphone microphone system and at least one headphone speaker and determining a headphone microphone audio gain that will mitigate actual or potential headphone feedback in at least one of the plurality of frequency bands, based at least partly upon the feedback risk control value.

Abstract: A pervasive listening method including steps of inserting at least one forced gap in a playback signal (thus generating a modified playback signal), and during playback of the modified playback signal, monitoring non-playback content (e.g., including by generating an estimate of background noise) in a playback environment using output of a microphone in the playback environment. Optionally, the method includes generation of the playback signal, including by processing of (e.g., performing noise compensation on) an input signal using a result (e.g., a background noise estimate) of the monitoring of non-playback content. Other aspects are systems configured to perform any embodiment of the pervasive listening method.

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: Apparatus comprising an interface for receiving a respective uplink data stream from each of three or more further apparatuses, and for transmitting a respective downlink data stream to each of the further apparatuses; and a logic system in communication with the interface. The logic system is configured: to receive first data in the uplink data stream received from a first one of the further apparatuses; and in a first mode, to include at least some of the first data in the respective downlink data streams transmitted to every other one of the further apparatuses, or, in a second mode, to include at least some of the first data in the downlink data stream transmitted to a second one of the further apparatuses and to omit or attenuate substantially all of the first data in the downlink data stream transmitted to at least a third one of the further apparatuses. Corresponding methods and computer readable media are disclosed.

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: 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: Systems and methods are described for compensating for inaccurate echo prediction in audio systems. A signal may be received at a microphone of an audio system, the signal including audio rendered using a spatial audio renderer across a multi-channel audio output. A signal may be received from the spatial audio renderer that indicates a change in rendering of audio. The audio system may then determine if there is echo power within the received signal greater than an expected echo power. After the signal from the spatial audio renderer has been received, the echo suppression applied to the received signal may be modified in response to a determination that the echo power is greater than the expected echo power, the echo suppression attenuating pre-selected frequency bands of the received signal.

Abstract: A noise estimation method including steps of generating gap confidence values in response to microphone output and playback signals, and using the gap confidence values to generate an estimate of background noise in a playback environment. Each gap confidence value is indicative of confidence of presence of a gap at a corresponding time in the playback signal, and may be a combination of candidate noise estimates weighted by the gap confidence values. Generation of the candidate noise estimates may but need not include performance of echo cancellation. Optionally, noise compensation is performed on an audio input signal using the generated background noise estimate. Other aspects are systems configured to perform any embodiment of the noise estimation method.

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: An apparatus and method of transmission control for an audio device. The audio device uses sources other than the microphone to determine nuisance, and uses this to calculate a gain as well as to make the transmit decision. Using the gain results in a more nuanced nuisance mitigation than using the transmit decision on its own.

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.