Abstract: The method and apparatus described herein make use of multiple sets of head related transfer functions (HRTFs) that have been synthesized or measured at various distances from a reference head, spanning from the near-field to the boundary of the far-field. Additional synthetic or measured transfer functions maybe used to extend to the interior of the head, i.e., for distances closer than near-field. In addition, the relative distance-related gains of each set of HRTFs are normalized to the far-field HRTF gains.

Abstract: A method of encoding an audio signal is provided comprising: applying multiple different time-frequency transformations to an audio signal frame; computing measures of coding efficiency across multiple frequency bands for multiple time-frequency resolutions; selecting a combination of time-frequency resolutions to represent the frame at each of the multiple frequency bands based at least in part upon the computed measures of coding efficiency; determining a window size and a corresponding transform size; determining a modification transformation; windowing the frame using the determined window size; transforming the windowed frame using the determined transform size; modifying a time-frequency resolution within a frequency band of the transform of the windowed frame using the determined modification transformation.

Abstract: The methods and apparatus described herein provides technical solutions to the technical problems facing crosstalk cancellation for 3D audio virtualization. One technical solution includes preconditioning audio signals based on crosstalk canceller characteristics and based on characteristics of sound sources at intended locations in 3D space. To provide these technical solutions, the systems and methods described herein include an audio virtualizer and an audio preconditioner. In particular, the audio virtualizer includes a crosstalk canceller, and the audio preconditioner preconditions audio signals based on characteristics of a crosstalk cancellation system and based on characteristics of a binaural synthesis system or intended input source location in space. This solution improves the overall accuracy of virtualization of 3D sound sources and reduces or eliminates audio artifacts such as incorrect localization, inter-channel sound level imbalance, or a sound level that is higher or lower than intended.

Abstract: Systems and methods to provide distortion sensing, prevention, and/or distortion-aware bass enhancement in audio systems can be implemented in a variety of applications. Sensing circuitry can generate statistics based on an input signal received for which an acoustic output is generated. In various embodiments, the statistics can be used such that a multi-notch filter can be used to provide input to a speaker to generate the acoustic output. In various embodiments, the statistics from the sensing circuitry can be provided to a bass parameter controller coupled to bass enhancement circuitry to operatively provide parameters to the bass enhancement circuitry. The bass enhancement circuitry can provide a bass enhanced signal for generation of the acoustic output, based on the parameters. Various combinations of a multi-notch filter and bass enhancement circuitry using statistics from sensing circuitry can be implemented to provide an enhanced acoustic output.

Abstract: A method and system is presented for coordinating the transmission of supplemental digital data to accompany broadcast data, and in particular, analog radio broadcasts, among a plurality of broadcasters. The supplemental digital data may provide information about the particular broadcast data being transmitted (i.e. cut data) or may be supplemental to such data (i.e. news, weather and traffic data). The supplemental digital data to be presented is sorted based on particular algorithms which may take into account broadcaster-specified criteria such as target audience, time of day, type of broadcast data presented, and the like. The supplemental digital data may be audio data, visual data, or audio-visual data for presentation with the broadcast data. The supplemental digital data may further be advertisement data. The advertisement data may be sold by the broadcasters or the party coordinating the IBOC transmission of the supplemental digital data.

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: A method to decode audio signals is provided that includes: receiving an input spatial audio signal; determining directions of arrival of directional audio sources represented in the received input spatial audio signal; determining one of an active input spatial audio signal component and a passive spatial audio signal input component, based upon the determined directions of arrival; determining the other of the active input spatial audio signal component and the passive input spatial audio signal component, based upon the determined one of the active input spatial audio signal component and the passive input spatial audio signal component; decoding the active input spatial audio signal component to a first output format; and decoding the passive input spatial audio signal component to a second output format.

Abstract: A system and method can determine a time-varying position of a sound in a multi-channel audio signal. At least one processor can: receive a multi-channel audio signal representing a sound, each channel of the multi-channel audio signal providing audio associated with a corresponding channel position around a perimeter of a soundstage; determine a time-varying volume level for each channel of the multi-channel audio signal; determine, from the time-varying volume levels and the channel positions, a time-varying position in the soundstage of the sound; and generate a location data signal representing the time-varying position of the sound. The channel positions can be time-invariant. The position magnitude can be scaled to provide a unit magnitude as a sound pans from a channel to an adjacent channel. The position azimuth angle can be scaled to account for center location bias.

Abstract: An audio adjustment system is provided that can output a user interface customized by the provider of the audio system instead of the electronic device manufacturer. Such an arrangement can save both field engineers and manufacturers a significant amount of time. Advantageously, in certain embodiments, such an audio adjustment system can be provided without knowledge of the electronic device's firmware. Instead, the audio adjustment system can communicate with the electronic device through an existing audio interface in the electronic device to enable a user to control audio enhancement parameters in the electronic device. For instance, the audio adjustment system can control the electronic device via an audio input jack on the electronic device.

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: Systems and methods to provide distortion sensing, prevention, and/or distortion-aware bass enhancement in audio systems can be implemented in a variety of applications. Sensing circuitry can generate statistics based on an input signal received for which an acoustic output is generated. In various embodiments, the statistics can be used such that a multi-notch filter can be used to provide input to a speaker to generate the acoustic output. In various embodiments, the statistics from the sensing circuitry can be provided to a bass parameter controller coupled to bass enhancement circuitry to operatively provide parameters to the bass enhancement circuitry. The bass enhancement circuitry can provide a bass enhanced signal for generation of the acoustic output, based on the parameters. Various combinations of a multi-notch filter and bass enhancement circuitry using statistics from sensing circuitry can be implemented to provide an enhanced acoustic output.

Abstract: A method and system for the transmission of digital data (210) over existing analog radio frequencies (230) is presented, wherein the digital data may include audio data, visual data or audio-visual data for presentation either with analog broadcast data or at a selectable time. The digital data may be transmitted over a plurality of sub-channels that have varying degrees or reliability (250). A “quality-of-service” process manages the transmission of digital data over various sub-channels based on the reliability of the sub-channel, the amount of digital data and the type of digital data to be transmitted. The digital data may further be encrypted and authenticated.

Abstract: Systems and methods discussed herein can provide three-dimensional audio virtualization with sweet spot adaptation. In an example, an audio processor circuit can be used to update audio signals for sweet spot adaptation based on information from at least one depth sensor or camera about a listener position in a listening environment.

Abstract: An audio encoder can parse a digital audio signal into a plurality of frames, each frame including a specified number of audio samples, perform a transform of the audio samples of each frame to produce a plurality of frequency-domain coefficients for each frame, partition the plurality of frequency-domain coefficients for each frame into a plurality of bands for each frame, each band having bit data that represents a number of bits allocated for the band, and encode the digital audio signal and difference data to a bit stream (e.g., an encoded digital audio signal). The difference data can produce the full bit data when combined with estimate data that can be computed from data present in the bit stream. The difference data can be compressed to a smaller size than the full bit data, which can reduce the space required in the bit stream.

Abstract: An audio encoder can parse a digital audio signal into a plurality of frames, each frame including a specified number of audio samples, perform a transform of the audio samples of each frame to produce a plurality of frequency-domain coefficients for each frame, partition the plurality of frequency-domain coefficients for each frame into a plurality of bands for each frame, each band having a reshaping parameter that represents a time resolution and a frequency resolution, and encode the digital audio signal to a bit stream that includes the reshaping parameters. For a first band, the reshaping parameter can be encoded using a first alphabet size. For a second band, the reshaping parameter can be encoded using a second alphabet size different from the first alphabet size. Using different alphabet sizes can allow for more compact compression in the bit stream.

Abstract: A user device can be used to correct for a latency of a speaker. The user device can communicate an indication to the speaker to play a sound at a first time. The user device can record a second time at which a microphone on the user device detects the sound. The first and second times can be synchronized to a clock of a computer network. The user device can compare the first and second times to determine a latency of the speaker. The user device can communicate adjustment data corresponding to the determined latency to the speaker. The speaker 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 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: In an audiovisual system, in which video is displayed on a screen that does not permit sound to pass through the screen, such as a light emitting diode panel, a high-frequency speaker positioned above an audience seating area can direct sound toward the screen, so that the screen can reflect the sound toward the audience seating area. The high-frequency speaker can be used with one or more low-frequency speakers positioned at or near the height of the audience seating area. The low-frequency and high-frequency sounds can appear to originate from close to the same height, thereby creating a realistic audio image at the audience seating area. A spectral filter can negate the spectral effects of propagation to and reflection from the screen. Suitable time delays can synchronize the high-frequency sound with the low-frequency sound and with video displayed on the screen.

Abstract: Systems and methods are described for processing data from a sequential series of groups of frames to achieve a target average processing bit rate for a particular group of frames in the series. In an example, a look-ahead buffer circuit can be populated with a number of frames from a particular group of frames, and a bit allocation can be determined for a frame in the look-ahead buffer circuit using bit request information about all of the frames in the buffer. The look-ahead buffer circuit can be populated with streaming frame information in a first-in-first-out manner, and bit allocation processing can be performed for each frame, in a particular group of frames, based on a frame position in the look-ahead buffer circuit and further based on bit requests associated with other frames in the look-ahead buffer circuit.

Abstract: The systems and methods described herein can be configured to identify, manipulate, and render different audio source components from encoded 3D audio mixes, such as can include content mixed for azimuth, elevation, and/or depth relative to a listener. The systems and methods can be configured to decouple depth encoding and decoding to permit spatial performance to be tailored to a particular playback environment or platform. In an example, the systems and methods improve rendering in applications that involve listener tracking, including tracking over six degrees of freedom (e.g., yaw, pitch, roll orientation, and x, y, z position).