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: 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 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: 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 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: A digital signal is processed by splitting it into at least two frequency subbands and the two subband signals are downsampled. A filter is applied in at least one of the subband signals. At least one of the phase and magnitude of the subband filtered signals is matched in the transition frequency band between the two subbands.

Abstract: Methods for voice sensing and keyword analysis are provided. An example method allows for causing a mobile device to transition to a second power mode, from a first power mode, in response to a first acoustic signal. The method includes authenticating a user based at least in part on a second acoustic signal. While authenticating the user, the second acoustic signal is compared to a spoken keyword. The spoken keyword is analyzed for authentication strength based on the length of the spoken keyword, quality of a series of phonemes used to represent the spoken keyword, and likelihood of the series of phonemes to be detected by a voice sensing. While receiving the first and second acoustic signals, a signal to noise ratio (SNR) is determined. The SNR is used to adjust sensitivity of a detection threshold of a voice sensing.

Abstract: A digital signal is processed by splitting it into at least two frequency subbands and the two subband signals are downsampled. A filter is applied in at least one of the subband signals. At least one of the phase and magnitude of the subband filtered signals is matched in the transition frequency band between the two subbands.

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: Provided are systems and methods for generating clean speech from a speech signal representing a mixture of a noise and speech. The clean speech may be generated from synthetic speech parameters. The synthetic speech parameters are derived based on the speech signal components and a model of speech using auditory and speech production principles. The modeling may utilize a source-filter structure of the speech signal. One or more spectral analyzes on the speech signal are performed to generate spectral representations. The feature data is derived based on a spectral representation. The features corresponding to the target speech according to a model of speech are grouped and separated from the feature data. The synthetic speech parameters, including spectral envelope, pitch data and voice classification data are generated based on features corresponding to the target speech.

Abstract: Systems and methods for estimating and tracking multiple attributes of multiple objects from multi-sensor data are provided. An exemplary method includes identifying features associated with sensor data. The sensor data represents data captured by at least one of a plurality of acoustic and non-acoustic sensors. Identification of the features associated with the sensor data may be based variously on detected sounds, motions, images, and the like. The exemplary method further includes determining, in parallel, multiple probable objects based at least in part on the identified features. Various embodiments of the method also include forming hypotheses based at least in part on associating identified features with the multiple probable objects and attributing the formed hypotheses to channels. Sequence of the formed hypotheses are constructed.

Abstract: A digital signal is processed by splitting it into at least two frequency subbands and the two subband signals are downsampled. A filter is applied in at least one of the subband signals. At least one of the phase and magnitude of the subband filtered signals is matched in the transition frequency band between the two subbands.

Abstract: The present technology provides a robust noise suppression system that may concurrently reduce noise and echo components in an acoustic signal while limiting the level of speech distortion. A time-domain acoustic signal may be received and be transformed to frequency-domain sub-band signals. Features, such as pitch, may be identified and tracked within the sub-band signals. Initial speech and noise models may be then be estimated at least in part from a probability analysis based on the tracked pitch sources. Speech and noise models may be resolved from the initial speech and noise models and noise reduction may be performed on the sub-band signals. An acoustic signal may be reconstructed from the noise-reduced sub-band signals.

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: The present technology provides adaptive noise and echo reduction of an acoustic signal which can overcome or substantially alleviate problems associated with mistaken adaptation of speech and noise models to acoustic echo. The present technology carries out a multi-faceted analysis to identify echo within the near-end acoustic signal to derive an echo model. Echo classification information regarding the derived echo model is then utilized to build near-end speech and noise models. These echo, speech, and noise models are then used to generate one or more signal modifications applied to the acoustic signal to preserve the desired near-end speech signal and reduce the echo and near-end noise signals. By building near-end speech and noise models utilizing echo classification information, the present technology can prevent adaptation of the speech and noise model to the acoustic echo.

Abstract: Wind noise is detected in and removed from an acoustic signal. Features may be extracted from the acoustic signal. The extracted features may be processed to classify the signal as including wind noise or not. The wind noise may be removed before or during processing of the acoustic signal. The wind noise may be suppressed by estimating a wind noise model, deriving a modification, and applying the modification to the acoustic signal. In audio devices with multiple microphones, the channel exhibiting wind noise (i.e., acoustic signal frame associated with the wind noise) may be discarded for the frame in which wind noise is detected.

Abstract: Systems and methods for estimating and tracking multiple attributes of multiple objects from multi-sensor data are provided. An exemplary method includes identifying features associated with sensor data. The sensor data represents data captured by at least one of a plurality of acoustic and non-acoustic sensors. Identification of the features associated with the sensor data may be based variously on detected sounds, motions, images, and the like. The exemplary method further includes determining, in parallel, multiple probable objects based at least in part on the identified features. Various embodiments of the method also include forming hypotheses based at least in part on associating identified features with the multiple probable objects and attributing the formed hypotheses to channels. Sequence of the formed hypotheses are constructed.

Abstract: An audio signal is processed to determine primary and ambient components by transforming the signal into frequency-domain vectors, and decomposing the left and right channel vectors into ambient and primary components by orthogonal projection.

Abstract: An audio signal is processed in the frequency domain to convert an input signal format to an output signal format. That is, a multichannel audio signal intended for playback over a predefined speaker layout can be formatted to achieve spatial reproduction over a different layout comprising a different number of speakers.

Abstract: Provided are systems and methods for generating clean speech from a speech signal representing a mixture of a noise and speech. The clean speech may be generated from synthetic speech parameters. The synthetic speech parameters are derived based on the speech signal components and a model of speech using auditory and speech production principles. The modeling may utilize a source-filter structure of the speech signal. One or more spectral analyses on the speech signal are performed to generate spectral representations. The feature data is derived based on a spectral representation. The features corresponding to the target speech according to a model of speech are grouped and separated from the feature data. The synthetic speech parameters, including spectral envelope, pitch data and voice classification data are generated based on features corresponding to the target speech.