Abstract: Example embodiments disclosed herein relate to audio signal processing. A method of indicating a presence of a nuisance in an audio signal is disclosed. The method includes determining a probability of the presence of the nuisance in a frame of the audio signal based on a feature of the audio signal, the nuisance representing an unwanted sound made by a user, in response to the probability of the presence of the nuisance exceeding a threshold, tracking the audio signal based on a metric over a plurality of frames following the frame, determining, based on the tracking, that the presence of the nuisance is to be indicated to the user, and in response to the determination, presenting to the user a notification of the presence of the nuisance. Corresponding system and computer program product are also disclosed.

Abstract: Methods for echo estimation or echo management (echo suppression or cancellation) on an input audio signal, with at least one of adaptation of a sparse prediction filter set, modification (for example, truncation) of adapted prediction filter impulse responses, generation of a composite impulse response from adapted prediction filter impulse responses, or use of echo estimation and/or echo management resources in a manner determined at least in part by classification of the input audio signal as being (or not being) echo free. Other aspects are systems configured to perform any embodiment of any of the methods.

Abstract: A method for interactive and user guided manipulation of multichannel audio content, the method including the steps of: providing a content preview facility for replay and review of multichannel audio content by a user; providing a user interface for the user selection of a segment of multichannel audio content having an unsatisfactory audio content; processing the audio content to include associated audio object activity spatial or signal space regions, to create a time line of activity where one or more spatial or signal space regions are active at any given time; matching the user's gesture input against at least one of the active spatial or signal space regions; signal processing the audio emanating from selected active spatial or signal space region using a number of differing techniques to determine at least one processed alternative; providing the user with an interactive playback facility to listen to the processed alternative.

Abstract: Described herein are audio capture systems and methods. One embodiment provides an audio capture system (1) including: microphones (9-11) positioned to capture respective audio signals from different directions or locations within an audio environment; a mixing module (7) configured to mix the audio signals in accordance with a mixing control signal to produce an output audio mix, wherein, upon the detection of vibration activity, the mixing control signal controls the mixing module (7) to selectively temporarily modify one or more of the audio signals to reduce the presence of noise associated with vibration activity in the output audio mix.

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: 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: Example embodiments disclosed herein relate to audio signal processing. A method of processing an audio signal is disclosed. The method includes detecting, based on a power distribution of the audio signal, a type of content of a frame of the audio signal, generating a first gain based on a sound level of the frame for adjusting the sound level, processing the audio signal by applying the first gain to the frame; and in response to the type of content being detected to be a breath sound, generating a second gain for mitigating the breath sound and processing the audio signal by applying the second gain to the frame. Corresponding system and computer program product 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: Methods for echo estimation or echo management (echo suppression or cancellation) on an input audio signal, with at least one of adaptation of a sparse prediction filter set, modification (for example, truncation) of adapted prediction filter impulse responses, generation of a composite impulse response from adapted prediction filter impulse responses, or use of echo estimation and/or echo management resources in a manner determined at least in part by classification of the input audio signal as being (or not being) echo free. Other aspects are systems configured to perform any embodiment of any of the methods.

Abstract: Systems and methods are described for modifying one of far-end signal playback and capture of local audio on an audio device. Frames of both a far-end audio stream and a near-end audio stream may be analyzed using a measure of voice activity, the analyzing producing voice data associated with each frame. Based on the voice data, a conference state may be determined, and one of playback of the far-end audio stream and capture of local audio on an audio device may be modified based on the determined conference state. By associating the likely intent with a predefined state, the device may further cull or remove unwanted or unlikely content from the device input and output. This may have a substantial advantage in allowing for full duplex operation in the case of more meaningful and continuing voice activity, particularly in the case where there are many connected endpoints.

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: 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: 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 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 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: Some aspects of the present disclosure involve the recording, processing and playback of audio data corresponding to conferences, such as teleconferences. In some teleconference implementations, the audio experience heard when a recording of the conference is played back may be substantially different from the audio experience of an individual conference participant during the original teleconference. In some implementations, the recorded audio data may include at least some audio data that was not available during the teleconference. In some examples, the spatial characteristics of the played-back audio data may be different from that of the audio heard by participants of the teleconference.

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: 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 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. At small computational expense, the endpoint may to adapt to different sample rates in received bitstreams.

Abstract: Some aspects of the present disclosure involve the recording, processing and playback of audio data corresponding to conferences, such as teleconferences. In some teleconference implementations, the audio experience heard when a recording of the conference is played back may be substantially different from the audio experience of an individual conference participant during the original teleconference. In some implementations, the recorded audio data may include at least some audio data that was not available during the teleconference. In some examples, the spatial characteristics of the played-back audio data may be different from that of the audio heard by participants of the teleconference.