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 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: 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: 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: A conferencing server (100) receives incoming bitstreams (I1, I2, I3, I4, I5) carrying media data from respective conferencing endpoints (110, 120, 130, 140, 150); receives a mixing strategy (M) specifying properties of at least one outgoing bitstream (O1, O2, O3, O4, O5) and requiring at least one additive media mixing step; and supplies at least one outgoing bitstream by executing, in a processor (103) and a memory (102) with a plurality of memory spaces, a run list of operations selected from a predefined collection of primitives and realizing the received mixing strategy. A pre-processor (104) in the server derives said run list repeatedly and dynamically while taking into consideration determined momentary activity in each incoming bitstream. In embodiments, the run list may be derived by (a) pruning of an initial run list, (b) constrained or non-constrained minimization of a cost function, or (c) automatic code generation.

Abstract: Embodiments are described for a soundfield system that receives a transmitting soundfield, wherein the transmitting soundfield includes a sound source at a location in the transmitting soundfield. The system determines a rotation angle for rotating the transmitting soundfield based on a desired location for the sound source. The transmitting soundfield is rotated by the determined angle and the system obtains a listener's soundfield based on the rotated transmitting soundfield. The listener's soundfield is transmitted for rendering to a listener.

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: 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: 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: In some embodiments, a method for modifying noise captured at endpoints of a teleconferencing system, including steps of capturing noise at each endpoint, and modifying the captured noise to generate modified noise having a frequency-amplitude spectrum which matches a target spectrum and a spatial property set which matches a target spatial property set. In other embodiments, a teleconferencing method including steps of: at endpoints of a teleconferencing system, determining audio frames indicative of audio captured at each endpoint, each of a subset of the frames indicative of noise but not a significant level of speech; and at each endpoint, generating modified frames indicative of modified noise having a frequency-amplitude spectrum which matches a target spectrum and a spatial property set which matches a target spatial property set, and generating encoded audio including by encoding the modified frames. Other aspects are systems configured to perform any embodiment of the method.

Abstract: A conferencing server (100) receives incoming bitstreams (I1, I2, I3, I4, I5) carrying media data from respective conferencing endpoints (110, 120, 130, 140, 150); receives a mixing strategy (M) specifying properties of at least one outgoing bitstream (O1, O2, O3, O4, O5) and requiring at least one additive media mixing step; and supplies at least one outgoing bitstream by executing, in a processor (103) and a memory (102) with a plurality of memory spaces, a run list of operations selected from a predefined collection of primitives and realizing the received mixing strategy. A pre-processor (104) in the server derives said run list repeatedly and dynamically while taking into consideration determined momentary activity in each incoming bitstream. In embodiments, the run list may be derived by (a) pruning of an initial run list, (b) constrained or non-constrained minimization of a cost function, or (c) automatic code generation.