Abstract: Example embodiments disclosed herein relate to spatial congruency adjustment. A method for adjusting spatial congruency in a video conference is disclosed. The method includes detecting spatial congruency between a visual scene captured by a video endpoint device and an auditory scene captured by an audio endpoint device that is positioned in relation to the video endpoint device, the spatial congruency being a degree of alignment between the auditory scene and the visual scene, comparing the detected spatial congruency with a predefined threshold and in response to the detected spatial congruency being below the threshold, adjusting the spatial congruency. Corresponding system and computer program products are also disclosed.

Abstract: In one embodiment, a pickup system includes a wind detector and a wind suppressor. The wind detector has a plurality of analyzers each configured to analyze first and second input signals, and a combiner configured to combine outputs of the plurality of analyzers and issue, based on the combined outputs, a wind level indication signal indicative of wind activity. The analyzers can be selected from a group of analyzers including a spectral slope analyzer, a ratio analyzer, a coherence analyzer, a phase variance analyzer and the like. The wind suppressor has a ratio calculator configured to generate a ratio of the first and second input signals, and a mixer configured to select one of the first or second input signals and to apply to the selected input signal one of first or second panning coefficients based on the wind level indication signal and on the ratio.

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: A method of facilitating the re-joining of a virtual meeting by a participant. The method comprises determining at what time the participant was disconnected from the virtual meeting; and determining whether or not the participant was talking within a time period containing the time at which the participant was disconnected from the virtual meeting. If it is determined that the participant was talking within the time period, then, consequently, the method comprises deciding not to enable the participant to replay stored data representative of a missed portion of the virtual meeting. As well as the method, related apparatuses, systems and computer readable storage media are disclosed.

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 method for altering an audio signal of interest in a multi-channel soundfield representation of an audio enviroment, the method including the steps of: (a) extracting the signal of interest from the soundfield representation; (b) determining a residual soundfield signal; (c) inputting a further associated audio signal, which is associated with the signal of interest; (d) transforming the associated audio signal into a corresponding associated soundfield signal compatable with the residual soundfield; and (e) combining the residual soundfield signal with the associated soundfield signal to produce an output soundfield signal.

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 (4) comprises apparatus (10) to deliver the audio soundfields to the listener (4). The delivery apparatus (10) delivers the audio soundfields to the listener (4) with first and second audio elements perceived by the listener (4) as emanating from first and second virtual source locations (1), respectively, and with the first audio element and/or the second audio element delivered to the listener (4) from a third virtual source location (3). The first virtual source location (1) and the second virtual source location (2) are perceived by the listener (4) as being located to the front of the listener (4), and the third virtual source location (3) is located to the rear or the side of the listener (4).

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: Voice communication method and apparatus and method and apparatus for operating jitter buffer are described. Audio blocks are acquired in sequence. Each of the audio blocks includes one or more audio frames. Voice activity detection is performed on the audio blocks. In response to deciding voice onset for a present one of the audio blocks, a subsequence of the sequence of the acquired audio blocks is retrieved. The subsequence precedes the present audio block immediately. The subsequence has a predetermined length and non-voice is decided for each audio block in the subsequence. The present audio block and the audio blocks in the subsequence are transmitted to a receiving party. The audio blocks in the subsequence are identified as reprocessed audio blocks. In response to deciding non-voice for the present audio block, the present audio block is cached.

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: In an audio conferencing mixing system of the type taking a plurality of audio input streams of input audio information of conference participants, including mixing transition events and outputting a plurality of audio output streams including output audio information, a method of mixing the audio output streams so as to reduce the detectability of the mixing transition events, the method including the steps of (a) determining that a transition event is to occur; (b) determining that a masking trigger is to occur; (c) scheduling the transition event to substantially occur when the masking event occurs. Change blindness mechanism to mask changes in audio conference mix and maintain perceptual continuity.

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: Voice communication method and apparatus and method and apparatus for operating jitter buffer are described. Audio blocks are acquired in sequence. Each of the audio blocks includes one or more audio frames. Voice activity detection is performed on the audio blocks. In response to deciding voice onset for a present one of the audio blocks, a subsequence of the sequence of the acquired audio blocks is retrieved. The subsequence precedes the present audio block immediately. The subsequence has a predetermined length and non-voice is decided for each audio block in the subsequence. The present audio block and the audio blocks in the subsequence are transmitted to a receiving party. The audio blocks in the subsequence are identified as reprocessed audio blocks. In response to deciding non-voice for the present audio block, the present audio block is cached.

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: 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: 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: Embodiments of method and apparatus for acoustic echo control are described. According to the method, an echo energy-based doubletalk detection is performed to determine whether there is a doubletalk in a microphone signal with reference to a loudspeaker signal. A spectral similarity between spectra of the microphone signal and the loudspeaker signal is calculated. It is determined that there is no doubletalk in the microphone signal if the spectral similarity is higher than a threshold level. Adaption of an adaptive filter for applying acoustic echo cancellation or acoustic echo suppression on the microphone signal is enabled if it is determined that there is no doubletalk in the microphone signal through the echo energy-based doubletalk detection, or there is no doubletalk through the spectral similarity-based doubletalk detection.

Abstract: An encoding system (100) encodes a first (E1) and further (E2, E3) audio signals as a layered bitstream (B), wherein a quantizer for each frequency band of each signal is selected using a rate allocation rule based on signal-specific rate allocation data, a spectral envelope of the signal and a reference level (EnvE1Max), which is determined based on the spectral envelope of the first signal and is not necessarily included in the bitstream. Further disclosed is a decoding system for reconstructing the audio signals based on the bitstream. In embodiments, the bitstream has a basic layer (BE1), which contains data that enable decoding of the first audio signal, and a spatial layer (Bspatial) facilitating decoding of the further audio signal(s). In embodiments, the encoding system prepares the bitstream subject to a basic-layer bitrate constraint and a total bitrate constraint.

Abstract: A method, an apparatus, and a computer-readable medium configured with instructions that when executed carry out the method for determining a measure of harmonicity. In one embodiment the method includes selecting candidate fundamental frequencies within a range, and for candidate determining a mask or retrieving a pre-calculated mask that has positive value for each frequency that contributed to harmonicity, and negative value for each frequency that contributes to inharmonicity. A candidate harmonicity measure is calculated for each candidate fundamental by summing the product of the mask and the magnitude measure spectrum. The harmonicity measure is selected as the maximum of the candidate harmonicity measures.