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.

Abstract: The present document relates to audio communication systems. In particular, the present document relates to the control of the level of audio signals within audio communication systems. A method for leveling a near-end audio signal (211) using a leveling gain (214) is described. The near-end audio signal (211) comprises a sequence of segments, wherein the sequence of segments comprises a current segment and one or more preceding segments.

Abstract: Improved audio data processing method and systems are provided. Some implementations involve dividing frequency domain audio data into a plurality of subbands and determining amplitude modulation signal values for each of the plurality of subbands. A band-pass filter may be applied to the amplitude modulation signal values in each subband, to produce band-pass filtered amplitude modulation signal values for each subband. The band-pass filter may have a central frequency that exceeds an average cadence of human speech. A gain may be determined for each subband based, at least in part, on a function of the amplitude modulation signal values and the band-pass filtered amplitude modulation signal values. The determined gain may be applied to each subband.

Abstract: A method for encoding sound field signals includes allocating coding rate by application of a uniform criterion to all subbands of all signals in a joint process. An allocation criterion may be based on a comparison, in a given subband, between a spectral envelope of the signals to be encoded and a coding noise profile, wherein the noise profile may be a sum of a noise shape and a noise offset, which noise offset is computed on the basis of the coding bit budget. The rate allocation process may be combined with an energy-compacting orthogonal transform, for which there is proposed a parameterization susceptible of efficient coding and having adjustable directivity. In a further aspect, the invention provides a corresponding decoding method.

Abstract: In a conference call having a plurality of participants interacting in a conference exchange of information in a digital transmission environment, the interaction being across a variable network transmission resource, a method of allocating the level of transmission resource, the methods including the steps of: (a) monitoring predetermined aspects of the participant's behavior during the conference call; (b) determining a divergence of participants behavior from normative values; (c) utilising any divergence as an indicator of aberrant operation of the participants; and (d) allocating the resource determinative on the divergence of participants behavior from normative values.

Abstract: An audio apparatus is configured to switch, when there exists a first audio interface between the audio apparatus and a computer apparatus, to using a second audio interface between the audio apparatus and the computer apparatus, the second audio interface being different from the first audio interface. The switching comprises: receiving, via the first audio interface, combined audio data and non-audio data, the non-audio data comprising a request to switch to using the second audio interface; obtaining the request from the data; and, in response to obtaining the request, transmitting to the computer apparatus a confirmation of switching to using the second audio interface. The audio apparatus and the computer apparatus are described and claimed.

Abstract: In one embodiment, a sound field is mapped by extracting spatial angle information, diffusivity information, and optionally, sound level information. The extracted information is mapped for representation in the form of a Riemann sphere, wherein spatial angle varies longitudinally, diffusivity varies latitudinally, and level varies radially along the sphere. A more generalized mapping employs mapping the spatial angle and diffusivity information onto a representative region exhibiting variations in direction of arrival that correspond to the extracted spatial information and variations in distance that correspond to the extracted diffusivity information.

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: Digital media communication systems and methods are disclosed in which a set of end points and/or clients are in communication with a set of servers. The system is capable of determining which end point is a dominant user and/or speaker. When the system determines that an end point is dominant, then the digital data that emanates from dominant end point is preferred. Such preferred digital data may be forwarded asynchronously and possible ahead of digital data from other, non-dominant end points. Synchronous and/or substantially contemporaneous digital data from other end points may be mixed that such that, when rendered, gives substantially the same effect real life superposition of signals. The system may also toggle on or off between states of asynchronous forward and synchronous-only modes.

Abstract: A system and method for initiating conference calls with external devices are disclosed. Call participants are sent conference invitation and conference information regarding the designated conference call. This conference information is stored on the participant's external device. When the participants arrive at a conference call location having a conferencing device, the conferencing device is capable of communicating with the external device, initiating communications, exchanging conference information. If the participant is verified and/or authorized, the conference system may send the IP address of the conference device to the conference system to initiate the conference call. In one embodiment, the conference device uses an ultrasound acoustic communication band to initiate the call with the external device on a semi-automated basis. An acoustic signature comprising a pilot sequence for communications synchronization may be generated to facilitate the call.

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: An audio signal with a temporal sequence of blocks or frames is received or accessed. Features are determined as characterizing aggregately the sequential audio blocks/frames that have been processed recently, relative to current time. The feature determination exceeds a specificity criterion and is delayed, relative to the recently processed audio blocks/frames. Voice activity indication is detected in the audio signal. VAD is based on a decision that exceeds a preset sensitivity threshold and is computed over a brief time period, relative to blocks/frames duration, and relates to current block/frame features. The VAD and the recent feature determination are combined with state related information, which is based on a history of previous feature determinations that are compiled from multiple features, determined over a time prior to the recent feature determination time period. Decisions to commence or terminate the audio signal, or related gains, are outputted based on the combination.

Abstract: A plurality of acoustic sensors in a non-anechoic environment are calibrated with the aim of removing manufacturing tolerances and degradation over time but preserving position-dependent differences between the sensors, The sensors are excited by an acoustic stimulus which has either time-dependent characteristics or finite duration. The calibration is to be based on diffuse-field excitation only, in which indirect propagation (including single or multiple reflections) dominate over any direct-path excitation. For this purpose, the calibration process considers only a non-initial portion of sensor outputs and/or of an impulse response derived there-from. Based on these data, a frequency-dependent magnitude response function is estimated and compared with a target response function, from which a calibration function is derived.

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: In some embodiments, a method for adaptive control of gain applied to an audio signal, including steps of analyzing segments of the signal to identify audio objects (e.g., voices of participants in a voice conference); storing information regarding each distinct identified object; using at least some of the information to determine at least one of a target gain, or a gain change rate for reaching a target gain, for each identified object; and applying gain to segments of the signal indicative of an identified object such that the gain changes (typically, at the gain change rate for the object) from an initial gain to the target gain for the object. The information stored may include a scene description. Aspects of the invention include a system configured (e.g., programmed) to perform any embodiment of the inventive method.

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: Improved audio data processing method and systems are provided. Some implementations involve dividing frequency domain audio data into a plurality of subbands and determining amplitude modulation signal values for each of the plurality of subbands. A band-pass filter may be applied to the amplitude modulation signal values in each subband, to produce band-pass filtered amplitude modulation signal values for each subband. The band-pass filter may have a central frequency that exceeds an average cadence of human speech. A gain may be determined for each subband based, at least in part, on a function of the amplitude modulation signal values and the band-pass filtered amplitude modulation signal values. The determined gain may be applied to each subband.

Abstract: A method, an apparatus, logic (e.g., executable instructions encoded in a non-transitory computer-readable medium to carry out a method), and a non-transitory computer-readable medium configured with such instructions. The method is to generate and spatially render spatial comfort noise at a receiving endpoint of a conference system, such that the comfort noise has target spectral characteristics typical of comfort noise, and at least one spatial property that at least substantially matches at least one target spatial property. On version includes receiving one or more or more audio signals from other endpoints, combining the received audio signals with the spatial comfort noise signals, and rendering the combination of the received audio signals and the spatial comfort noise signals to a set of output signals for loudspeakers, such that the spatial comfort noise signals are continually in the output signal sin addition to output from the received audio signals.

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 application provides an acoustic echo mitigation apparatus and method, an audio processing apparatus and a voice communication terminal. According to an embodiment, an acoustic echo mitigation apparatus is provided, including: an acoustic echo canceller for cancelling estimated acoustic echo from a microphone signal and outputting an error signal; a residual echo estimator for estimating residual echo power; and an acoustic echo suppressor for further suppressing residual echo and noise in the error signal based on the residual echo power and noise power. Here, the residual echo estimator is configured to be continuously adaptive to power change in the error signal. According to the embodiments of the present application, the acoustic echo mitigation apparatus and method can, at least, be well adaptive to the change of power of the error signal after the AEC processing, such as that caused by change of double-talk status, echo path properties, noise level and etc.