Abstract: An audio processing device comprises a) at least one input unit for providing time-frequency representation Y(k,n) of an electric input signal representing sound consisting of target speech and noise signal components, where k and n are frequency band and time frame indices, respectively, b) a noise detection and/or reduction system configured to b1) determine an a posteriori signal to noise ratio estimate ?(k,n) of said electric input signal, and to b2) determine an a priori target signal to noise signal ratio estimate ?(k,n) of said electric input signal from said a posteriori signal to noise ratio estimate ?(k,n) based on a recursive decision directed algorithm. The application further relates to a method of of estimating an a priori signal to noise ratio. The invention may e.g. be used for the hearing aids, headsets, ear phones, active ear protection systems, handsfree telephone systems, mobile telephones, etc.

Abstract: An audio processing circuit, audio unit and integrated circuit and method for blending are described. The audio processing circuit comprises: a time-to-frequency domain circuit configured to convert a primary audio signal and a secondary audio signal in a time-domain form and output frequency spectrum representations thereof; a cross-fade magnitude circuit configured to obtain a cross-faded magnitude spectrum representation from the frequency spectrum representations; a cross-fade phase circuit configured to obtain a separate cross-faded phase spectrum representation from the frequency spectrum representations; and a frequency-to-time domain circuit configured to receive the separate cross-faded magnitude and cross-faded phase spectrum representations and output a time domain signal.

Abstract: A content replacement system identifies regular content replacement parts and alternative content replacement parts, each alternative content replacement part being a content item with a duration within one of a plurality of ranges. The system detects a start of a content replacement break in a listener content stream, the listener content stream shifted by a listener delay value from a live content stream. The system inserts the regular content replacement parts into the stream. The system detects an end of the content replacement break for the listener content stream, and selects one of the alternative content replacement parts to insert into the listener content stream, the selected alternative content replacement part having a duration that when inserted into the listener content stream minimizes a difference between the listener delay value and a target delay value.

Abstract: A wearable includes an article of clothing, an intelligent control and a plurality of sets of vibration motors operatively connected to the intelligent control and positioned on the article of clothing to provide tactile feedback wherein each of the sets of vibration motors is associated with a different frequency range of audio which may be communicated to the wearable via Bluetooth or otherwise.

Abstract: The embodiments of the invention disclosed are configured with a voice translation software, and can be readily used after switching off the microphone of the mobile phone and inserted into the charging port of the mobile phone. In some embodiments, the invention comprises two microphones; when the mobile phone is directed at a speaker, the audio pickup is performed by using a “hypercardioid noise-filtering microphone”, and when the mobile phone is directed at oneself, the audio pickup is performed by using a “cardioid noise-filtering microphone”, such that the surrounding noises can be prevented from being picked up in order to achieve an accurate and fluent voice translation. In other embodiments, when the microphone is used to ask for directions or in communication while travelling abroad in particular, this use enables the user to keep an adequate distance from the opposite party in keeping with the international social etiquette.

Abstract: Systems, methods, and computer hardware storage media are provided for variably reducing, but not eliminating, the ambient volume that reaches the eardrum in a particular setting. One or more noise signatures of sound from an ambient environment are received at a mobile device. A profile associated with a suppression or amplification control of the one or more noise signatures of sound from the ambient environment are created and stored at the mobile device. The suppression or amplification control is communicated from the mobile device to a set of variable audio ear plugs. The suppression control causes the variable audio ear plugs to variably reduce, but not eliminate, the sound from the ambient environment and the amplification control does not amplify sound from any source other than the ambient environment.

Abstract: The present invention provides a vacuum tube audio amplifier which includes an audio pre-amplifying portion and an audio output transforming portion. The audio pre-amplifying portion includes an equalizer for enhanced sound quality and amplifies an audio signal, which is subsequently transmitted to the audio output transforming portion. Furthermore, the audio output transforming portion includes an audio output transformer which includes a plurality of stacked E-shaped silicon steel sheets and a plurality of stacked I-shaped silicon steel sheets, wherein the stacked E-shaped silicon steel sheets and the stacked I-shaped silicon steel sheets have a same height which is smaller than or equal to 48 mm.

Abstract: The disclosure is directed to a device that includes a multi-layered porous display that permits the passage of sound through the display, while avoiding dead-spots that produce no light. Embodiments of the invention include a display that has first porous layer overlaying a second porous layer. The pores of each layer do not align so that light cannot pass directly through pores of each layer. In this way, a user may not view the pores of the second layer through the pores of the first layer. Further, because the first layer is backed by a second layer, the image displayed will be viewable through the pores of the first layer and, thus, a user will not perceive the pores of the first layer.

Abstract: An integrated circuit for digital signal routing. The integrated circuit has analog and digital inputs and outputs, including digital interfaces for connection to other integrated circuits. Inputs, including the digital interfaces, act as data sources. Outputs, including the digital interfaces, act as data destinations. The integrated circuit also includes signal processing blocks, which can act as data sources and data destinations. Signal routing is achieved by means of a multiply-accumulate block, which takes data from one or more data source and, after any required scaling, generates output data for a data destination. Data from a data source is buffered for an entire period of a data sample clock so that the multiply-accumulate block can retrieve the data at any point in the period, and output data of the multiply-accumulate block is buffered for an entire period of the data sample clock so that the data destination can retrieve the data at any point in the period.

Abstract: An acoustic pressure reducer acoustically couples to and provides acoustic impedance to attenuate the acoustic box pressure of an acoustic system, such as a loudspeaker system. The pressure reducer may also allow an ambient pressure of the acoustic system to equalize with an ambient pressure of an external environment and the ambient pressure of the acoustic pressure reducer at a certain rate. The attenuation may allow for inexpensive acoustic sensors to be utilized within the pressure reducer to measure one or more acoustic properties of an attenuated acoustic pressure within the pressure reducer. An unattenuated acoustic pressure value of the acoustic system may be estimated using a known transfer function of the pressure reducer and the attenuated acoustic pressure values measured within. A controller coupled to the acoustic system may adjust one or more operating characteristics in response to estimating the unattenuated acoustic pressure.

Abstract: A system capable of self-adjusting both sound level and spectral content to improve audibility and intelligibility of medical device audible cues. Audible cues are stored as sound files. Ambient noise is detected, and the output of the audible cues is altered based on the ambient noise. Various embodiments include processed sound files that are more robust in noisy environments.

Abstract: An earpiece includes an earpiece housing, a processor disposed within the ear piece housing, a speaker operatively connected to the processor, at least one microphone operatively connected to the processor, and a cellular transceiver disposed within the earpiece housing and operatively connected to the processor. The cellular transceiver may be a low energy long term evolution (LTE) cellular transceiver. The ear piece may include an inertial sensor disposed within the earpiece housing and operatively connected to the processor. The earpiece may further include a short range transceiver disposed within the earpiece housing. The short range transceiver may be a Bluetooth transceiver. The earpiece may also include a slot within the earpiece housing for receiving an identity module.

Abstract: A smart-home device may include a recording device configured to record sound during a first time interval and a memory device comprising a plurality of stored sound profiles. The smart-home device may also include a processing system configured to receive an environmental input, select a stored sound profile from the plurality of stored sound profiles based on the environmental input, and perform a noise-cancelation routine on the sound recorded during the first time interval. The stored sound profile may be used as an initial background noise profile for the noise-cancelation routine.

Abstract: An acoustic output device and a control method thereof are provided.

Abstract: A microphone assembly includes a capacitive MEMS transducer and a bias circuit having a DC output coupled to the transducer. The microphone assembly also includes a differential amplifier having a first input coupled to a first output of the transducer, the differential amplifier having a second input coupled to a second output of the transducer. The microphone assembly further includes a first impedance matching network and a second impedance matching network configured to balance a first capacitive load at the first input of the amplifier and a second capacitive load at the second input of the amplifier. Electrical signals applied by the transducer to the first and second inputs of the amplifier are matched or approximately matched in magnitude and 180 degrees or approximately 180 degrees out of phase with respect to each other.

Abstract: Methods, systems, computer-readable media, and apparatuses for gesture recognition are disclosed for audio control. The gestures can be used for audio control in virtualized environments. The audio control can include generating a sound that mimics an effect of a modification of a sound wave that would occur if the gesture were performed in a non-virtualized environment.

Abstract: The present disclosure provides a number of techniques for personalization of audio for communication to an endpoint device. According to one technique, a cloud-based computing device may receive media data from a media platform, and generate depersonalized audio data based on the media data, the depersonalized audio data including at least one audio component associated with the media platform. The technique may further generate user personalized audio data based on the depersonalized audio data. The user personalized audio data may include at least one audio component personalized based on a unique head-related transfer function (HRTF) data associated with a user or default HRTF data. The user personalized audio may be communicated to the endpoint device for consumption by the user. Such techniques reduce the need for advanced audio processing technologies on client systems and makes personalized audio more available to users.

Abstract: A method for reducing the latency period of a filter bank for filtering an audio signal. A large number of signal blocks in the time domain are formed from the audio signal, wherein for at least a plurality of the signal blocks in each instance a filter function is predetermined, at least one partial interval of the signal block is predetermined as a prediction period, signal components of the signal block in the at least one partial interval are estimated for the prediction period, and a predicted signal block is generated from the signal components estimated for the prediction period and from the signal components of the signal block outside the prediction period. The predicted signal block, filtered with the predetermined filter function, is transformed into the frequency domain to form a transformed signal block. Signal components of the transformed signal block are output for further processing.

Abstract: A saddlebag speaker assembly for housing a speaker within a saddlebag and a method of forming a saddlebag speaker assembly are provided. The assembly includes a saddlebag lid opening cover configured to cover the saddlebag lid opening and couple to the saddlebag lid and the speaker.

Abstract: An alert system for generating a signal indicating at least one aspect of sound within a workspace environment, the system comprising at least a first sensor positioned one of within and proximate a workspace, the at least a first sensor sensing at least one aspect of sound generated within the workspace and generating a signal indicating the at least one aspect, a communication device located within the workspace, the communication device configured to generate a signal perceivable within the workspace and a processor receiving signals from the at least a first sensor and linked to the communication device, the processor programmed to perform the steps of identifying when the at least one aspect exceeds a threshold value and driving the communication device to indicate that the at least one aspect exceeds the threshold value.