Abstract: An array processing system improves the spatial selectivity by forming multiple steered beams and carrying out a spatial analysis of the acoustic scene. The analysis derives a time-frequency mask that, when applied to a reference look-direction beam (or other reference signal), enhances target sources and substantially improves rejection of interferers that are outside of the specified region.

Abstract: Corrupted portions of an audio signal are detected and repaired. An audio signal, including numerous sequential frames, may be received from an audio input device. One or more corrupted frames included in the audio signal may be identified. A frame approximating an uncorrupted frame and corresponding to each corrupted frame may be constructed. Each corrupted frame may be replaced with a corresponding constructed frame to generate a repaired audio signal. The repaired audio signal may be outputted via an audio output device.

Abstract: An audio processing system processes an audio signal that may come from one or more microphones. The audio processing system may use information from one or more non-acoustic sensors to improve a variety of system characteristics, including responsiveness and quality. Those audio processing systems that use spatial information, for example to separate multiple audio sources, are undesirably susceptible to changes in the relative position of any audio sources, the audio processing system itself, or any combination thereof. Using the non-acoustic sensor information may decrease this susceptibility advantageously in an audio processing system.

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: A frequency band of an audio input signal is analyzed to determine if a transient is present. When transients are detected, modifications are made to the intensity levels corresponding to the frequency band for a brief time period.

Abstract: A two-channel phase-amplitude stereo encoding and decoding scheme enabling flexible and spatially accurate interactive 3-D audio reproduction via standard audio-only two-channel transmission. The encoding scheme allows associating a 2-D or 3-D positional localization to each of a plurality of sound sources by use of frequency independent inter-channel phase and amplitude differences. The decoder is based on frequency-domain spatial analysis of 2-D or 3-D directional cues in a two-channel stereo signal and re-synthesis of these cues using any preferred spatialization technique, thereby allowing faithful reproduction of positional audio cues and reverberation or ambient cues over arbitrary multi-channel loudspeaker reproduction formats or over headphones, while preserving source separation despite the intermediate encoding over only two audio channels.

Abstract: An audio processing system processes an audio signal that may come from one or more microphones. The audio processing system may use information from one or more non-acoustic sensors to improve a variety of system characteristics, including responsiveness and quality. Especially those audio processing systems that use spatial information, for example to separate multiple audio sources, are undesirably susceptible to changes in the relative position of any audio sources, the audio processing system itself, or any combination thereof. Using the non-acoustic sensor information may decrease this susceptibility advantageously in an audio processing system.

Abstract: The present technology provides a sophisticated level of control of the spatial pattern of an acoustic field which can overcome or substantially alleviate problems associated with transmitting an acoustic signal within the near-end acoustic environment. The spatial pattern is produced by utilizing an array of audio transducers which generate a plurality of acoustic waves forming an acoustic interference pattern, such that the resultant acoustic energy is constrained (e.g. limited to an acoustic energy level at or below a predetermined threshold level) in one or more regions of the spatial pattern. In doing so, listeners in these region(s) may not receive sufficient acoustic energy to hear the associated acoustic signal, while listeners in other regions can. Similarly, these techniques can suppress echo paths within those region(s).

Abstract: Provided are systems and methods for improving audio signal quality during teleconferencing. Multiple local devices equipped with microphones may be connected to a teleconference hub device that also maintains communication with one or more remote devices. The local devices may be mobile, such as mobile phones, and each may be positioned closer to at least some local teleconference participants compared to a microphone of a traditional central teleconferencing speakerphone. Such mobility allows users to move during the teleconference without disturbing the quality. The local devices may be connected to the hub device using various local protocols. Individual audio signals may first be processed by the local devices and then sent to the hub device for further collective processing of multiple data streams and production of an external data stream, which is sent to remote devices connected to the hub device over a remote protocol.

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: An input signal is converted to a feature-space representation. The feature-space representation is projected onto a discriminant subspace using a linear discriminant analysis transform to enhance the separation of feature clusters. Dynamic programming is used to find global changes to derive optimal cluster boundaries. The cluster boundaries are used to identify the segments of the audio signal.

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. An acoustic signal may be received and transformed to cochlear 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: An audio signal is processed by transforming the signal into a frequency domain representation having a plurality of frequency subbands. A decorrelated signal is derived from the frequency domain representation using a phase rotation.

Abstract: A frequency-domain method for format conversion or reproduction of 2-channel or multi-channel audio signals such as recordings is described. The reproduction is based on spatial analysis of directional cues in the input audio signal and conversion of these cues into audio output signal cues for two or more channels in the frequency domain.

Abstract: A frequency domain method for phase-amplitude matrixed surround decoding of 2-channel stereo recordings and soundtracks, based on spatial analysis of 2-D or 3-D directional cues in the recording and re-synthesis of these cues for reproduction on any headphone or loudspeaker playback system.

Abstract: A stereo audio signal is processed to determine primary and ambient components by transforming the signal into vectors corresponding to subband signals, and decomposing the left and right channel vectors into ambient and primary components by matrix and vector operations. Principal component analysis is used to determine a primary component unit vector, and ambience components are determined according to a correlation-based cross-fade or an orthogonal basis derivation.

Abstract: Allpass arrays of arbitrary order are presented. The transducers in the arrays are configured with weights corresponding to the FIR approximation of an allpass filter such that a nearly uniform array response is provided.

Abstract: A transducer array includes speaker drivers having nonuniform asymmetric spacing. The array includes at least three drivers formed along a line or arc. The first of the drivers is positioned having a first spacing from an adjacent second driver that is different from a second spacing between the second driver and its adjacent third driver.

Abstract: A method of ambience extraction includes analyzing an input signal to determine the time-dependent and frequency-dependent amount of ambience in the input signal, wherein the amount of ambience is determined based on a signal model and correlation quantities computed from the input signals and wherein the ambience is extracted using a multiplicative time-frequency mask. Another method of ambience extraction includes compensating a bias in the estimation of a short-term cross-correlation coefficient. In addition, systems having various modules for implementing the above methods are disclosed.

Abstract: An audio signal is processed to derive primary and ambient components of the signal. The signal is first transformed to generate frequency-domain subband signals. Primary and ambient components are separated by comparing frequency subband content using a complex-valued similarity metric, wherein one of the primary and ambient components is determined to be the residual after the other is identified using the similarity metric.