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).

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: 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: Embodiments of systems and methods are described for estimating a position of a loudspeaker and notifying a listener if an abnormal condition is detected, such as an incorrect loudspeaker orientation or an obstruction in a path between the loudspeaker and a microphone array. For example, a front component of a multi-channel surround sound system may include the microphone array and a position estimation engine. The position estimation engine may estimate the distance between the loudspeaker and the microphone array. In addition, the position estimation engine may estimate an angle of the loudspeaker using a first technique. The position estimation engine may also estimate an angle of the loudspeaker using a second technique. The two angles can be processed to determine whether the abnormal condition exists. If the abnormal condition exists, a listener can be notified and be provided with suggestions for resolving the issue in a graphical user interface.

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: 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: 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: A bass management system and method for mitigating bass management errors by using explicit information available in the object audio rendering process and deriving the correct subwoofer contribution for each audio object. Embodiments of the bass management system and method are used to maintain the correct balance of the bass reproduced by the subwoofer relative to the sound coming out of the other speakers. The system and method are useful for a variety of different speaker configurations, including speaker configurations having different speaker sub-zones. Power-normalized gain coefficients for each speaker are combined and the power of the combined gain coefficients is computed and used to obtain a power-preserving subwoofer contribution coefficient. This subwoofer contribution coefficient is applied to the bass portion of the audio signal and audio objects to determine the contribution of a particular subwoofer.

Abstract: An active audio output detection system and method that automatically determines the type of audio output being used on an audio playback device. Embodiments of the system and method include playing back audio content on the audio playback device (such as a mobile phone) and capturing sound during playback using an audio capture device (such as a microphone). The captured sound is analyzed to determine whether the audio playing back on the audio playback device is contained therein. If the captured sound matches the audio playing back on the audio playback device, then the active audio output is a loudspeaker. On the other hand, if the captured sound does not match the audio playing back on the audio playback device, then the active audio output is headphones.

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: Loudness control systems or methods may normalize audio signals to a predetermined loudness level. If the audio signal includes moderate background noise, then the background noise may also be normalized to the target loudness level. Noise signals may be detected using content-versus-noise classification, and a loudness control system or method may be adjusted based on the detection of noise. Noise signals may be detected by signal analysis in the frequency domain or in the time domain. Loudness control systems may also produce undesirable audio effects when content shifts from a high overall loudness level to a lower overall loudness level. Such loudness drops may be detected, and the loudness control system may be adjusted to minimize the undesirable effects during the transition between loudness levels.

Abstract: Certain embodiments enable improved encoding and decoding of a vector of coefficients by associating a vector element of a signed pyramid with an encoded value that includes a first portion and a second portion, where the first portion identifies a corresponding vector element of an unsigned pyramid and a second portion characterizes sign values for nonzero components of the vector element of the signed pyramid. As a result, computational constraints such as word size apply to the unsigned pyramid instead of the signed pyramid. The smaller size of the unsigned pyramid enables extending the range of signed pyramid parameters that are operable within the computational constraints.

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: Embodiments of systems and methods are described for estimating a position of a loudspeaker and notifying a listener if an abnormal condition is detected, such as an incorrect loudspeaker orientation or an obstruction in a path between the loudspeaker and a microphone array. For example, a front component of a multi-channel surround sound system may include the microphone array and a position estimation engine. The position estimation engine may estimate the distance between the loudspeaker and the microphone array. In addition, the position estimation engine may estimate an angle of the loudspeaker using a first technique. The position estimation engine may also estimate an angle of the loudspeaker using a second technique. The two angles can be processed to determine whether the abnormal condition exists. If the abnormal condition exists, a listener can be notified and be provided with suggestions for resolving the issue in a graphical user interface.

Abstract: A system can be provided for increasing loudness of an audio signal to present a perceived loudness to a listener that is greater than a loudness provided natively by a loudspeaker. The system can include one or more of the following: a frequency suppressor, a loudness adjuster, an equalizer, and a distortion control module. The frequency suppressor can increase headroom in the audio signal by filtering out low and/or high frequencies. The loudness adjuster can calculate a loudness of the audio signal and apply a gain to the audio signal to increase the loudness. The equalizer can further increase headroom by attenuating portions of a passband of the loudspeaker's frequency response. The distortion control module can induce partial harmonic distortion in the audio signal to further increase loudness.

Abstract: Systems, devices, and methods are described herein for adjusting a relationship between dialog and non-dialog signals in an audio program. In an example, information about a long-term dialog balance for an audio program can be received. The long-term loudness dialog balance can indicate a dialog-to-non-dialog loudness relationship of the audio program. A dialog loudness preference can be received, such as from a user, from a database, or from another source. A desired long-term gain or attenuation can be determined according to a difference between the received long-term dialog balance for the audio program and the received dialog balance preference. The long-term gain or attenuation can be applied to at least one of the dialog signal and the non-dialog signal of the audio program to render an audio program that is enhanced according to the loudness preference.

Abstract: Systems and methods to provide an audio spectral correction can be implemented in a variety of applications. Such systems and methods can include an equalizer design that uses efficient filter sections tuned with a closed form algorithm to give an accurate and intuitive frequency response with low complexity and minimal processing overhead. Embodiments can enable efficient and accurate control of reverberation time vs. frequency in artificial reverberators. Embodiments can enable efficient realization of a cascade combination of multi-band equalization functions as a single multi-band equalizer function, in which such functions can be expressed as graphic equalization functions using the same set of center frequencies as the cascade combination of multi-band equalization functions. Additional apparatus, systems, and methods are disclosed.

Abstract: The methods and apparatus described herein optimally represent full 3D audio mixes (e.g., azimuth, elevation, and depth) as “sound scenes” in which the decoding process facilitates head tracking. Sound scene rendering can be modified for the listener's orientation (e.g., yaw, pitch, roll) and 3D position (e.g., x, y, z). This provides the ability to treat sound scene source positions as 3D positions instead of being restricted to positions relative to the listener. Sound scene rendering can be augmented by encoding depth to a source directly. This provides the ability to modify the transmission format and panning equations to support adding depth indicators during content production. Unlike typical methods that apply depth cues such as loudness and reverberation changes in the mix, this method would enable recovering the distance of a source in the mix so that it can be rendered for the final playback capabilities rather than those on the production side.

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: The present invention provides for methods and apparatuses for processing audio data. In one embodiment, there is provided a method for achieving bitstream scalability in a multi-channel audio encoder, said method comprising receiving audio input data; organizing said input data by a Code Excited Linear Predictor (CELP) processing module for further encoding by arranging said data according to significance of data, where more significant data is placed ahead of less significant data; and providing a scalable output bitstream; a higher bitrate bitstream is scaled to lower bitrate by discarding less significant data from frame ends. The organized CELP data comprises of a first part and a second part. The first part comprises a frame header, sub frame parameters and innovation vector quantization data from the first frame from all channels. The innovation vector quantization data from the first frames from all channels is arranged according to channel number.