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 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: 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.

Abstract: A sound field coding system and method that provides flexible capture, distribution, and reproduction of immersive audio recordings encoded in a generic digital audio format compatible with standard two-channel or multi-channel reproduction systems. This end-to-end system and method mitigates any impractical need for standard multi-channel microphone array configurations in consumer mobile devices such as smart phones or cameras. The system and method capture and spatially encode two-channel or multi-channel immersive audio signals that are compatible with legacy playback systems from flexible multi-channel microphone array configurations.

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: A transform-based codec and method with energy smoothing for mitigating vector quantization errors (such as “birdies”) during the encoding process. Embodiments of the codec and method use an encoder to apply in combination an orthogonal transformation and a vector permutation to frequency transform coefficients. In some embodiments the transformation is performed first followed by the permutation and in other embodiments the order is reversed. The order used is reversed at the decoder. A smoothing parameter containing the level of energy smoothing to be applied is passed from the encoder to the decoder and used by both to compute a transform matrix and an inverse transform matrix. In some embodiments the transform matrix is a fraction Hadamard matrix that is invertible, energy preserving, controllable, and stable.

Abstract: A GPEQ filter and tuning methods for asymmetric transaural audio reproduction on mobile consumer devices. The GPEQ filter includes an asymmetric FIR filter bank configured to normalize the magnitude responses of at least two audio speakers to a target response above a low frequency cross-over of the audio speakers as reproduced at the listener position and a symmetric IIR filter bank component configured to high pass filter above the low frequency cross-over and equalize the magnitude responses of the at least two audio speakers. In some embodiments the FIR filter bank is configured to normalize the group delay of the audio channels and in some embodiments the IIR filter bank includes an asymmetric IIR filter bank component configured to correct any residual asymmetry in the magnitude responses of the at least two audio speakers. Phase preservation may be accomplished using phase-balanced or phase-compensated IIR filter sections.

Abstract: A constant-power pairwise panning upmixing system and method for upmixing from a two-channel stereo signal to a multi-channel surround sound (having more than two channels). Each output channel is some combination of the two input channels. Closed-form solutions are used to calculate dematrixing coefficients that are used to weight each input channel. The dematrixing coefficients are computed based on an inter-channel level difference and an inter-channel phase difference between the two input signals. The weighted input channels then are mixed uniquely for each output channel to generate a surround sound output from the stereo input signal. Each dematrixing coefficient has an in-phase component and an out-of-phase component. The phase coefficients for each component vary in time and are based on the phase difference between the input signals. The resultant surround sound output faithfully simulates the audio content as originally mixed.