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: Encoding and decoding devices for encoding the channels of an audio system having at least four channels are disclosed. The decoding device has a first stereo decoding component which subjects a first pair of input channels to a first stereo decoding, and a second stereo decoding component which subjects a second pair of input channels to a second stereo decoding. The results of the first and second stereo decoding components are crosswise coupled to a third and a fourth stereo decoding component which each performs stereo decoding on one channel resulting from the first stereo decoding component, and one channel resulting from the second stereo decoding component.

Abstract: Methods, systems, and computer-readable media for admitting conferees into a conference call. Methods to achieve the foregoing can comprise: transmitting an admission request from a user of a communications device to a server for admission to a conference call, the admission request including a unique identifier; storing data unique to the communications device in a data store; relaying the unique identifier from the server to the data store; comparing the unique identifier with the data unique to the communications device; and, when the unique identifier and the previously stored data match, connecting the user directly to the conference call which includes bypassing entry of at least one conference pass code by the user.

Abstract: In a method for use in Interactive Voice Response (IVR) systems, an IVR is configured such that voice prompts provided to a caller vary systematically and caller hang-up activity is attributed to the particular version of a prompt played to the caller. Voice prompt modifications are chosen based on how long the caller willingly listens to the voice prompt. A relatively short time listening to a prompt before hang-up, indicates dissatisfaction with that prompt. The system compares caller hang-up rates for each of the phrases in a prompt and chooses the optimum solution, which is the variant of the prompt with the longest caller listening time. Optionally, the system compares hang up rates to a threshold and contingent on that comparison chooses an alternative version of the prompt.

Abstract: The invention relates to an acoustic device capable of producing active noise reduction, which may be positioned on the head of a user, comprising at least one microphone capable of sensing a sound signal representative of ambient noise, including at least one active noise reduction acoustic module comprising an osteophonic transducer, capable of being positioned on a side flank of the head of the user and of transmitting a vibratory signal transformed by bone conduction into an acoustic signal which may be perceived by the user, connected to said microphone, while said at least one acoustic module includes an electronic circuit capable of generating a vibratory signal giving the possibility of attenuating perception of said ambient noise by the user.

Abstract: Audio characteristics of audio data corresponding to a plurality of audio channels may be determined. The audio characteristics may include spatial parameter data. Decorrelation filtering processes for the audio data may be based, at least in part, on the audio characteristics. The decorrelation filtering processes may cause a specific inter-decorrelation signal coherence (“IDC”) between channel-specific decorrelation signals for at least one pair of channels. The channel-specific decorrelation signals may be received and/or determined. Inter-channel coherence (“ICC”) between a plurality of audio channel pairs may be controlled. Controlling ICC may involve at receiving an ICC value and/or determining an ICC value based, at least partially, on the spatial parameter data. A set of IDC values may be based, at least partially, on the set of ICC values. A set of channel-specific decorrelation signals, corresponding with the set of IDC values, may be synthesized by performing operations on the filtered audio data.

Abstract: A matching system receives probe audio samples for comparison to references of a data store. Comparisons are generated between a first segment of a probe audio sample and corresponding time segments of a plurality of reference audio samples to identify a plurality of sufficiently matching reference audio samples based upon a first set of consistency scores. Matching references are retained, unless they meet a score threshold. Comparisons are continually generated with a second segment of the probe audio sample and corresponding time segments of the sufficiently matching reference audio samples to generate a second set of consistency scores. The retained results are outputted based on the first and second set of consistency scores.

Abstract: The invention provides an efficient implementation of cross-product enhanced high-frequency reconstruction (HFR), wherein a new component at frequency Q?+r?0 is generated on the basis of existing components at ? and ?+?0. The invention provides a block-based harmonic transposition, wherein a time block of complex subband samples is processed with a common phase modification. Superposition of several modified samples has the net effect of limiting undesirable intermodulation products, thereby enabling a coarser frequency resolution and/or lower degree of oversampling to be used. In one embodiment, the invention further includes a window function suitable for use with block-based cross-product enhanced HFR. A hardware embodiment of the invention may include an analysis filter bank, a subband processing unit configurable by control data and a synthesis filter bank.

Abstract: The invention provides a headset control method, comprising the following steps: A) a capacitive sensing circuit electrode layer detects sliding coordinate signals and trajectory signals; B) a processing unit receives the coordinate signals and trajectory signals detected by the capacitive sensing circuit electrode layer; C) the processing unit analyzes whether the received coordinate signals and the trajectory signals meet control rules that are already defined and converts the coordinate signals and trajectory signals meeting the control rules into instructions to be transmitted to a main control chip; D) the main control chip is connected with a playback equipment terminal and the main control chip transmits the instructions to the playback equipment terminal, and the playback equipment terminal executes the instructions.

Abstract: A system, method, and apparatus for drone detection and classification are disclosed. An example method includes receiving a sound signal in a microphone and recording, via a sound card, a digital sound sample of the sound signal, the digital sound sample having a predetermined duration. The method also includes processing, via a processor, the digital sound sample into a feature frequency spectrum. The method further includes applying, via the processor, broad spectrum matching to compare the feature frequency spectrum to at least one drone sound signature stored in a database, the at least one drone sound signature corresponding to a flight characteristic of a drone model. The method moreover includes, conditioned on matching the feature frequency spectrum to one of the drone sound signatures, transmitting, via the processor, an alert.

Abstract: A system and method for facilitating a personalizing settings in a driving apparatus using user's biometric information are provided. Individual users of an automobile are facilitated to set personal settings through a biometric information device in the driving apparatus. In one embodiment, a given user of the driving apparatus is facilitated to provide his/her fingerprint in a first mode through the fingerprint scanner to store settings of a set of currently configured components as the personalized settings for those components. The given user is facilitated to subsequently provide his/her fingerprint in a second mode through the fingerprint scanner to retrieve the personalized settings for the set of configured components and have the set of components configured in accordance with those settings. In some examples, security is provided to determine whether a given user is authorized to personalize certain customizable components in the driving apparatus.

Abstract: A system, method, and apparatus for drone detection and classification are disclosed. An example method includes receiving a sound signal in a microphone and recording, via a sound card, a digital sound sample of the sound signal, the digital sound sample having a predetermined duration. The method also includes processing, via a processor, the digital sound sample into a feature frequency spectrum. The method further includes applying, via the processor, broad spectrum matching to compare the feature frequency spectrum to at least one drone sound signature stored in a database, the at least one drone sound signature corresponding to a flight characteristic of a drone model. The method moreover includes, conditioned on matching the feature frequency spectrum to one of the drone sound signatures, transmitting, via the processor, an alert.

Abstract: A method and system for providing automated call center services, operating by identifying a caller, accessing the caller's account information, determining the likely reason for the call, identifying an agent skill group based on the likely reason for the call, and providing an available agent with access to the caller's account information, together with a script based on the likely reason for the call. Data and voice access to agents is combined with specialized methods and systems for determining agent optimization capabilities, thereby allowing multiple call centers located in geographically different areas to operate as a seamless virtual call center, having the capability to dynamically reorganize the structure of available agents, regardless of where located. Real time statistics allow determination of how many agents should be available and what skill set those agents should have, thereby facilitating management decisions.

Abstract: Apparatuses and methods for encoding/decoding scalable digital audio are disclosed. An apparatus for decoding scalable digital audio according to the present invention includes: an audio packet reception unit configured to receive a digital audio packet, including a plurality of pieces of direct audio channel data mapped to respective physical channels and a plurality of pieces of indirect audio channel data indirectly mapped to respective physical channels; an indirect audio channel extraction unit configured to extract the digital audio packet from the pieces of indirect audio channel data; an indirect audio channel decoding unit configured to generate pieces of audio channel data corresponding to a number of physical channels more than the number of pieces of indirect audio channel data using the pieces of indirect audio channel data; and an audio channel output unit configured to match the pieces of audio channel data to the respective physical channels and perform output.

Abstract: A small-signal power acquisition unit defines a threshold for each of a plurality of input signals based on a long-term average of a minimum power of the corresponding input signal, each of the input signals being obtained by a microphone, sets a power value of each input signal that is smaller than the corresponding defined threshold to be a small-signal power of the input signal. A correction amount setting unit obtains each correction amount for correcting the input signals based on the small-signal power of the corresponding input signal. A correction unit corrects each of the input signals based on each of the correction amounts for correcting the input signals.

Abstract: The present disclosure provides methods, devices and computer program products for non-uniform quantization of parameters relating to parametric spatial coding of audio signals. The disclosure further relates to a method and apparatus for reconstructing an audio object in an audio decoding system taking the non-uniformly quantized parameters into account. According to the disclosure, such an approach renders it possible to reduce bit consumption without substantially reducing the quality of the reconstructed audio object.

Abstract: An audio apparatus including a circuit for applying a dynamic range adjustment processing to an input signal to generate an output signal for rendering by a loudspeaker. The dynamic range adjustment processing is dependent on a set of dynamic range adjustment parameters. A first linear echo-cancellation filter generates a first compensated signal from the input signal and a first adapter determines a set of filter parameters for the first linear echo-cancellation filter in response to the first compensated signal and a microphone signal. An audio circuit generates a second compensated signal by applying the dynamic range adjustment processing and a second echo-cancellation filter to the input signal where the second echo-cancellation filter corresponding to the first echo-cancellation filter. A second adapter determines the set of dynamic range adjustment parameters in response to the second compensated signal and the microphone signal.

Abstract: A first example playback device includes one or more processors and a non-transitory computer-readable medium storing instructions that, when executed by the one or more processors, cause the first playback device to perform functions. The functions include generating a mixed audio signal comprising one or more components of a first audio stream and one or more components of a second audio stream; and playing the mixed audio signal, where the one or more components of the second audio stream of the mixed audio signal played by the first playback device are played in synchrony with a second playback device configured to play the second audio stream. An example non-transitory computer-readable medium and an example method, both related to the first example playback device, are also disclosed herein.

Abstract: A chamber that penetrates vertically is formed in a silicon substrate. A diaphragm is arranged on the upper surface of the silicon substrate so as to cover the upper opening of the chamber. The diaphragm is divided by slits into a region located above the chamber (first diaphragm) and a region located above the upper surface of the silicon substrate (second diaphragm). A fixed electrode plate is arranged above the first diaphragm, and a low-volume first acoustic sensing portion is formed by the first diaphragm and the fixed electrode plate. Also, a high-volume second acoustic sensing portion is formed by the second diaphragm and the upper surface (electrically conducting layer) of the silicon substrate.

Abstract: Audio conferencing systems and methods with mutual adaptation control of adaptive beamforming and adaptive acoustic echo cancellation are disclosed. Mutual adaptation control may be achieved in a system including an adaptive beamforming module, an adaptive acoustic echo cancellation module, and an adaptation control module. The adaptive beamforming module has a controllable beamforming adaptivity and a beamforming adaptation state. The adaptive acoustic echo cancellation module has a controllable AEC adaptivity and an AEC adaptation state. The adaptation control module is configured and/or operates (i) to modify the beamforming adaptivity when the AEC adaptation state is unsettled (adapting to changed conditions), (ii) to modify the AEC adaptivity when the beamforming adaptation state is unsettled (adapting to changed conditions), (iii) to restore the beamforming adaptivity when the AEC adaptation state is settled, and (iv) to restore the AEC adaptivity when the beamforming adaptation state is settled.