Abstract: Embodiments of systems and methods are described for reducing undesired leakage energy produced by a non-front-facing speaker in a multi-speaker system. For example, the multi-speaker system can include an array of forward-facing speakers, one or more upward-facing speakers, and/or one or more side-facing speakers. Filters coupled to any two of the speakers in the multi-speaker system can generate audio signals output by the coupled speakers to reduce, attenuate, or cancel a portion of an audio signal output by one or more non-front-facing speakers that acoustically propagates along a direct path from the respective non-front-facing speaker to a listening position in a listening area in front of the multi-speaker system.

Abstract: Media is selected for video playback through a first device and audio playback through one or more separate devices connected through a wireless network. Different techniques for synchronizing the audio and video can be selected based on one or more factors to improve media playback.

Abstract: When the volume is adjusted in a multi-speaker system, it is desirable that one speaker does not change volume disproportionately with respect to another speaker. A method is presented for adjusting a volume level of one or more speakers. Each speaker can have a non-standardized relationship between logical volume level that is input to the speaker and sound pressure level that is produced by the speaker. A selected volume level, corresponding to a sound pressure level, can be received via a user interface. A stored lookup table can be accessed to convert the sound pressure level to a first product-specific logical volume level for each speaker. The stored lookup table can tabulate the non-standardized relationship between logical volume level and sound pressure level for each speaker. Data corresponding to the first product-specific logical volume level can be transmitted to each speaker.

Abstract: There are disclosed automatic mixers and methods for creating a surround audio mix. A set of rules may be stored in a rule base. A rule engine may select a subset of the set of rules based, at least in part, on metadata associated with a plurality of stems. A mixing matrix may mix the plurality of stems in accordance with the selected subset of rules to provide three or more output channels.

Abstract: Embodiments of systems and methods for user enrollment in speaker authentication and speaker identification systems are disclosed. In some embodiments, the enrollment process includes collecting speech samples that are examples of multiple speech types spoken by a user, computing a speech representation for each speech sample, and aggregating the example speech representations to form a robust overall representation or user voiceprint of the user's speech.

Abstract: Deep-learning classifier training systems and methods for training a classification system based on machine learning are disclosed. In some embodiments the training is configured to form classification regions of a specified shape. In some embodiments the training is configured to form classification regions in accordance with specified classification criteria. The systems and methods disclosed for training a classification system lead to improved inference performance.

Abstract: A graphical user interface (GUI) can specify a three-dimensional position. The GUI can include a puck that is user-positionable in two dimensions. A distance between the puck and a reference point on the graphical user interface can determine a radius of the three-dimensional position. An azimuthal orientation of the puck with respect to the reference point can determine an azimuthal angle of the three-dimensional position. The GUI can include an elevation controller, separate from the puck, to specify an elevation of the three-dimensional position. The elevation controller can specify an elevation angle, which can determine a position along the surface of a virtual sphere as a function of elevation angle. In addition, or alternatively, the elevation controller can specify an elevation height, which can determine a position along a line, such as directly upward or downward, as a function of elevation height.

Abstract: Circuitry can separate a multi-channel input signal into a center signal and a residual signal, apply time-varying center and residual gains to the center and residual signals, and combine the gain-adjusted center and residual signals to form a multi-channel audio signal. The center and residual gains are automatically determined in response to the center and residual signals so as to prevent each channel of the multi-channel audio signal from exceeding a target volume. During first times, the center and residual gains can vary synchronously so as to ensure that amplitude and phase relationships among channels in the multi-channel input signal are retained into the multi-channel audio signal. During second times, the center and residual gains can vary independently so as to reduce the energy of the residual signal compared to the center signal in the multi-channel audio signal.

Abstract: The present subject matter provides a technical solution to the technical problems facing sound localization by separating sounds and reproducing the separated sounds using a set of loudspeakers and a set of headphones. A general soundtrack that is meant to be experienced throughout the room would play through the loudspeakers, and specific sounds that are meant to be experienced near the listener would be played through a binaural representation in the headphones. The headphones may be selected to avoid occluding the ear, allowing sound produced at the loudspeakers to be heard clearly. This separation and reproduction of sounds using a combination of a loudspeaker and headphone provides a technical solution to the technical problem facing typical surround sound systems by localizing sounds for listeners in any location within a room. This improves reproduction accuracy of location-specific audio objects, including audio objects above or below a coplanar speaker configuration.

Abstract: An audio signal processing system can be configured to provide virtualized audio information in a three-dimensional soundfield using at least a pair of loudspeakers or headphones. The system can include an audio input configured to receive audio program information that includes at least N discrete audio signals, a first virtualization processor circuit configured to generate intermediate virtualized audio information by filtering M of the N audio signals, and a second virtualization processor circuit configured to generate further virtualized audio information by differently filtering K of the N audio signals, wherein K, M, and N are integers. The system can include an audio signal combination circuit to combine the intermediate virtualized audio information with at least one of the N audio signals, other than the M audio signals, to render fewer than N audio signals for transmission to a second virtualization processor circuit.

Abstract: An air-cooled heat exchanger is provided. A reinforcement beam is laterally attached to an upper surface of a guide housing which is formed to have a dome shape having an area gradually increasing in a downward direction from the stack to guide air discharged to the stack. In another aspect, an air-cooled heat exchanger may further include an enclosure surrounding a power transmission part configured to transmit a rotation force of a motor to the fan to rotate the fan.

Abstract: Systems and methods include audio encoders having improved coding of harmonic signals. The audio encoders can be implemented as transform-based codecs with frequency coefficients quantized using spectral weights. The frequency coefficients can be quantized by use of the generated spectral weights applied to the frequency coefficients prior to the quantization or by use of the generated spectral weights in computation of error within a vector quantization that performs the quantization. Additional apparatus, systems, and methods are disclosed.

Abstract: A system comprises an Internet network interface, and a first server. The first server include a first port operatively coupled to the Internet network interface, a memory, a processor, and a service application for execution by the processor. The service application is configured to receive a digital audio file and associated radio broadcast information via the Internet network interface, obtain an audio file identifier using a segment of the digital audio file, forward the digital audio file to a radio broadcast system according to the radio broadcast information, receive the segment of the digital audio file and associated radio reception information via the internet network interface, and identify the digital audio file and record the radio reception information for the identified digital audio file.

Abstract: Embodiments of systems and methods are described for reducing undesired leakage energy produced by a non-front-facing speaker in a multi-speaker system. For example, the multi-speaker system can include an array of forward-facing speakers, one or more upward-facing speakers, and/or one or more side-facing speakers. Filters coupled to any two of the speakers in the multi-speaker system can generate audio signals output by the coupled speakers to reduce, attenuate, or cancel a portion of an audio signal output by one or more non-front-facing speakers that acoustically propagates along a direct path from the respective non-front-facing speaker to a listening position in a listening area in front of the multi-speaker system.

Abstract: A user device can be used to correct for a latency of an audio chain, which extends inclusively between an input and a speaker. The user device can communicate an indication to the speaker to play a sound at a first time, and record a second time at which a microphone on the user device detects the sound. The user device can compare the first and second times to determine a latency of the audio chain. The user device can communicate adjustment data corresponding to the determined latency to a component in the audio chain. The component can use the adjustment data to correct for the determined latency. In some examples, the user device can display instructions to position the user device a specified distance from the speaker, and can account for a time-of-flight of sound to propagate along the specified distance.

Abstract: A method and a system for calibrating a surround sound system are disclosed. The calibration system can provide a graphical user interface for display comprising a visual representation of the room hosting a multichannel surround sound system. The graphical user interface can permit user input of gestures to place or make changes to the placement of icons representing one or more loudspeakers and a listener. The calibration system can estimate the positions of the one or more loudspeakers or the listener based on the placement of the icons in the model room. A spatial calibration based on the estimated positions can then be performed such that the multichannel surround sound system can render sound scenes more accurately.

Abstract: An audio control system can automatically determine an occupancy of an audience seating area, and can automatically adjust a spectral content of an audio signal to at least partially compensate for spectral effects caused by the occupancy in the audience seating area. Such an audio control system can ensure that the sound in a particular room remains constant, or nearly constant, from performance to performance. The occupancy can include a number of people in attendance in the audience seating area, and/or locations of attendees in the audience seating area. The audio control system can automatically adjust the spectral content a single time, at the start of a performance, or can optionally automatically dynamically update the adjustment throughout the performance based on subsequent measurements of the occupancy. The system can determine the occupancy through ticket sales, and/or analysis of imaging of the seats in the audience seating area.

Abstract: The present subject matter provides technical solutions to technical problems facing audio virtualization. To reduce the technical complexity and computational intensity facing audio virtualization, a technical solution includes rendering audio objects binaurally with differing quality levels, where the quality level for each audio source may be selected based on their position relative to the user's field of view. In an example, this technical solution reduces technical complexity and computational intensity by reducing the audio quality for audio sources outside of a user's central field of vision. In an example, high quality audio rendering may be applied to sound objects within this strong central visual acuity area. These technical solutions reduce processing over higher complexity systems, and provides potential for much higher quality rendering at a reduced technical and computational cost.

Abstract: The technology relates to determining speaker positioning in a multichannel sound system. In one example, one or more controllers perform time of flight measurements between one or more pairs of speakers of the plurality of speakers. A distance between each of the one or more pairs of speakers based on the time of flight measurements is determined A location of each speaker in the one or more pairs of speakers relative to the other speakers in the one or more pairs of speakers is determined. Based on the determined location of each speaker the multichannel sound system is calibrated.

Abstract: An audio signal encoding method is provided comprising: receiving first and second audio signal frames; processing a second portion of the first audio signal frame and a first portion of the second audio signal frame using an orthogonal transformation to determine in part a first intermediate encoding result; and processing the first intermediate encoding result using an orthogonal transformation to determine a set of spectral coefficients that corresponds to at least a portion of the first audio signal frame.