Abstract: An audio controller device and method of operation thereof. The audio controller device includes a memory for storing audio data and a plurality of primary audio control interfaces. An electronic processor at the audio controller device is configured to identify a plurality of audio segments from the audio data based on a plurality of contextual parameters associated with the audio data, associate each of the plurality of audio segments to a respective one of the primary audio control interfaces, and control a playback operation associated with the respective one of the audio segments when an input is received at one or more of the primary audio control interfaces. The electronic processor controls the playback operation by varying a playback speed rate associated with the respective one of the audio segments when the input is received at one or more of the primary audio control interfaces.

Abstract: A microphone apparatus including a MEMS transducer, an acoustic activity detector, a local oscillator, and an external-device interface standardized for compatibility with devices from different manufacturers is disclosed. The microphone apparatus has a first mode of operation during which the apparatus is clocked by the internal clock signal when the acoustic activity detector processes digital data for acoustic activity, and a second mode of operation during which the microphone apparatus is clocked by an external clock signal received at the external-device interface after voice activity is detected by the acoustic activity detector.

Abstract: An apparatus for air flow generation for scent output is described. The device includes a scent system, a speaker, and an output channel. The scent system and the speaker share the output channel, and the speaker is to move air to release scent from the device. A membrane of the speaker may vibrate to produce the air flow.

Abstract: The signal processing device comprises the adaptive gain calculation unit configured to calculate and output the adaptive gain for processing the voice reception output signal, by utilizing at least the voice transmission input signal and the relative volume value indicating the predetermined relative sound volume ratio between the voice transmission input signal and the voice reception output signal; and the multiplication unit configured to calculate the voice reception output signal by multiplying the signal based on the voice reception input signal by the value based on the adaptive gain.

Abstract: A microphone assembly includes an acoustic sensor configured to produce an electrical signal and an electrical circuit including a voice activity detector coupled to an output of the acoustic sensor. The microphone assembly further includes an internal clock signal generator and a host interface coupled to the electrical circuit. The host interface has external connections including an external clock signal and a data connection. The microphone assembly includes an acoustic activity detection mode of operation when the electrical circuit is clocked by the internal clock signal generator in the absence of an external clock signal at the host interface. The microphone assembly has a plurality of operating modes controlled by an external clock signal received in response to an interrupt signal provided by the electrical circuit. The electrical circuit is configured to provide data representing the electrical signal at the host interface when the external clock signal is received.

Abstract: The system allows a consumer to shortcut through a call queue to reach a human agent faster. For phone or voice-based calls that require a consumer wait on hold, the system can give that consumer an accurate estimate of the wait time and allow them to skip all or some of it. It does this by using automated processes to proactively dial and wait in those call queues, measuring their time in queue. A new consumer making an incoming call can then join the call queue occupied by of those automated processes, shortcutting the consumer past all or some of the wait time.

Abstract: The present disclosure provides a miniature speaker, including a vibration unit, a magnetic unit and a housing, the vibration unit includes a first diaphragm, a centering support, a second diaphragm and a voice coil; the centering support includes a first centering support and a second centering support having a conductive portion electrically connected with an external circuit; the second diaphragm includes a first half-diaphragm assembled with the first centering support and a second half-diaphragm assembled with the second centering support, the first half-diaphragm and the second half-diaphragm are asymmetrical configured with respect to each other. Under such structure, it is only necessary to adjust the material and shape of the first half-diaphragm and the second half-diaphragm, swing of the miniature speaker at low frequency can be effectively suppressed, a maximum output sound pressure of the miniature speaker can be effectively improved, and low-frequency sound quality can be improved accordingly.

Abstract: Embodiments of the disclosure may provide an apparatus and method of controlling and altering the acoustic output of audio devices that are used in conjunction with a computing device. In some embodiments, the apparatus and methods include a wireless speaker communication method and computing device software application that are configured to work together to more easily setup and deliver audio information from an audio source to one or more portable audio speakers.

Abstract: The invention relates to an adaptive filter unit, in particular for being used as an echo canceller, comprising a first filter input, configured to receive a first electric audio signal, indicative of a first audio signal A(t), a second filter input, configured to receive a second electric audio signal, indicative of a second audio signal B(t), a processor and a filter output. The processor is configured to calculate and provide audio estimation data X(fn, A(t1, . . .

Abstract: Audio objects are associated with positional metadata. A received downmix signal comprises downmix channels that are linear combinations of one or more audio objects and are associated with respective positional locators. In a first aspect, the downmix signal, the positional metadata and frequency-dependent object gains are received. An audio object is reconstructed by applying the object gain to an upmix of the downmix signal in accordance with coefficients based on the positional metadata and the positional locators. In a second aspect, audio objects have been encoded together with at least one bed channel positioned at a positional locator of a corresponding downmix channel. The decoding system receives the downmix signal and the positional metadata of the audio objects. A bed channel is reconstructed by suppressing the content representing audio objects from the corresponding downmix channel on the basis of the positional locator of the corresponding downmix channel.

Abstract: Techniques are provided for vector noise cancellation. Different value combinations for a plurality of weighting factors may be established for a plurality of selection regions. Each value combination for the plurality of weighting factors may correspond to a different combination of a set of input signals. One or more characteristics of input signals may be used to select a particular selection region. A particular value combination of the set of weighting factors may be chosen to attenuate or amplify the input signals to generate one or more output signals.

Abstract: Systems and methods are provided for conducting conference calls using doppler-based, i.e., reverberation-based techniques. The embodiments comprise a call device performing operations to join a call session hosted on a session server; receive sensor data comprising an audio signal from a first microphone and location information associated with the first microphone; determine a reverberation parameter associated with the location information; generate a first processed audio signal based on the audio signal and the reverberation parameter; and transmit the first processed audio signal to the session server. The session server may perform operations to receive a respective processed audio signal; determine a sound quality parameter of the respective processed audio signal; generate a balanced audio signal based on the sound quality parameter and the received processed audio signal; and transmit the balanced audio signal to a remote call device belonging to a second party.

Abstract: The present disclosure provides a miniature speaker, including a frame, and a vibrating unit and a magnetic unit which are fixedly connected with the frame; the vibrating unit includes a voice diaphragm, a voice coil located below the voice diaphragm and configured to drive the voice diaphragm to vibrate, and an elastic supporting member located below the voice coil and configured to elastically support the voice coil; the elastic supporting member includes a diaphragm and a flexible circuit board fixedly connected below the diaphragm; the diaphragm includes a first connecting portion fixedly connected with the frame, a second connecting portion fixedly connected with the voice diaphragm and a suspension connecting the first connecting portion with the second connecting portion; the suspension is an upward protruded structure protruding toward the voice diaphragm, and the voice coil transmits an electric signal to an external circuit through the flexible circuit board.

Abstract: A request is received from a communications device to execute an interaction site. A request is transmitted to the automated response system. First instructions that provide one or more steps of the multi-step communication flow between the communications device and the automated response system are received from the automated response system. In response to determining that the request is for the voice-interaction with the interaction site, second instructions that provide the one or more steps through a voice interaction with a user of the communications device are determined and transmitted to the communications device. In response to determining that the request is for the visual-interaction with the interaction site, third instructions that provide the one or more steps through a visual interaction with the user of the communications device are determined and transmitted to the communications device.

Abstract: The present disclosure discloses an echo cancellation method and apparatus. The method includes: obtaining a far-end signal and a near-end signal; detecting whether a speech signal exists in the near-end signal; processing the far-end signal according to a normalized least mean square adaptive filtering (NLMS) algorithm to obtain an estimated signal, and subtracting the estimated signal from the near-end signal to obtain a residual signal; after it is detected that the speech signal exists in the near-end signal, detecting whether the residual signal meets a preset output condition; and when it is detected that the residual signal does not meet the preset output condition, multiplying the residual signal by a first attenuation factor to obtain an output signal.

Abstract: Embodiments relate to a method for detecting a broken binder in an access network including a plurality of telecommunication lines grouped in respective binders, executed by a detection device, including identifying a plurality of telecommunication lines for which a transmission interruption has been detected within a given time period obtaining line data associated with the respective identified telecommunication lines, for respective pairs of identified telecommunication lines, determining a virtual distance between the two telecommunication lines, in function of the associated line data, determining at least one cluster of telecommunication lines in function of the virtual distances, and detecting a broken binder in function of the determined at least one cluster.

Abstract: Systems and methods are provided for conducting conference calls using doppler-based, i.e., reverberation-based techniques. The embodiments comprise a call device performing operations to join a call session hosted on a session server; receive sensor data comprising an audio signal from a first microphone and location information associated with the first microphone; determine a reverberation parameter associated with the location information; generate a first processed audio signal based on the audio signal and the reverberation parameter; and transmit the first processed audio signal to the session server. The session server may perform operations to receive a respective processed audio signal; determine a sound quality parameter of the respective processed audio signal; generate a balanced audio signal based on the sound quality parameter and the received processed audio signal; and transmit the balanced audio signal to a remote call device belonging to a second party.

Abstract: An audio media visualization method and system. The method includes receiving by a computing processor, mood description data describing different human emotions/moods. The computer processor an audio file comprising audio data and generates a mood descriptor file comprising portions of the audio data associated with specified descriptions of the mood description data. The computer processor receives a mood tag library file comprising mood tags mapped to animated and/or still objects representing various emotions/moods and associates each animated and/or still object with an associated description. The computer processor synchronizes the animated and/or still objects with the portions of said audio data and presents the animated and/or still objects synchronized with the portions of said audio data.

Abstract: Disclosed is a method for determining filter coefficients of an audio precompensation controller for the compensation of an associated sound system, including N?2 loudspeakers, including estimating, for each one of at least a pair of the loudspeakers, a model transfer function at each of M control points distributed in Z?2 spatially separated listening zones in a listening environment of the sound system. The method also includes determining, for each of the M control points, a zone-dependent target transfer function at least based on the zone affiliation of the control point; and determining the filter coefficients of the audio precompensation controller at least based on the model transfer functions and the target transfer functions of the M control points. Consequently, an audio precompensation controller for an associated sound system can be obtained that enables improved and/or customized sound reproduction in two or more listening zones simultaneously.

Abstract: A call and recorded information management server transmits a requested call identification ID for recognition and an incoming call number corresponding to the ID, to a voice recognition server, which searches for the recognition engine corresponding to the number of the received call, and adds the received ID to a recognition queue. The voice recognition server requests the call and recorded information management server to obtain recorded data by the received ID. The call and recorded information management server transfers the recorded data corresponding to the ID to the voice recognition server to perform voice recognition by the corresponding recognition engine on the recorded data corresponding to the call identification ID stored in the recognition queue, and stores the result as text data. Thus, the most appropriate voice recognition engine can be used and the recognition rate of voice recognition can be increased.