Abstract: Digital methods and systems for signal processing and filtering are provided. The methods and corresponding systems provide asynchronous conversion of sampling rate frequencies and utilize advanced multistage phasor filters for converting an input signal having a first sampling rate into an output signal sampled in an arbitrary sequence of sampling times. The conversion process provides a sequence of sets of complex numbers representing a filtered version of the input signal. More specifically, the conversion process includes the calculation of values of the output signal by multiplying (e.g., scaling) the sets of complex numbers by a corresponding set of complex phasors, the complex phasors corresponding to the timing of the arbitrary time sequence to obtain a corresponding set of real results with the value of the output signal being the sum of the real results.

Abstract: Provided are computer-implemented systems and methods for image correction after combining images from multiple cameras. While combining multiple images may significantly enhance image quality without a need for a longer focal length lens, combined images may have some perspective distortion caused by different imaging angles used by different cameras. For example, when two cameras provided on the same device are used to capture images of a face and these images are combined by cross-fading, a combined image may have an unnaturally widened center portion. Specifically, proportions of nose and ears may be distorted resulting in a wider nose and generally a wider face. The combined image may be compressed either uniformly or non-uniformly (e.g., more compression in the center) to yield a processed image with more realistic and/or natural proportions. The compression ratio and/or compression distribution may depend on imaging angles, various distances, and other factors described herein.

Abstract: Disclosed are methods and corresponding systems for audio processing of audio signals after applying a noise reduction procedure such as noise cancellation and/or noise suppression, according to various embodiments. A method may include calculating spectral envelopes for corresponding samples of an initial audio signal and the audio signal transformed by application of the noise cancellation and/or suppression procedure. Multiple spectral envelope interpolations may be calculated between these two spectral envelopes. The interpolations may be compared to predetermined reference spectral envelopes associated with predefined clean reference speech. One of the generated interpolations, which is the closest to one of the predetermined reference spectral envelopes, may be selected. The selected interpolation may be used for restoration of the transformed audio signal such that at least a part of the frequency spectrum of the transformed audio signal is modified to the levels of the selected interpolation.

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: Robust feedforward active noise cancellation is provided which can overcome or substantially alleviate problems associated with the diverse and dynamic nature of the surrounding acoustic environment. A multi-faceted analysis is performed to determine the direction (or directions) of propagation of noise in the surrounding acoustic environment. The direction of propagation is then utilized to determine direction-dependent characteristics of the acoustic path between a reference position where the noise is captured and a desired position where the noise is to be cancelled. These characteristics are used to form a feedforward signal adapted to cancel the noise at the desired position. By forming the feedforward signal based on direction-dependent characteristics of the acoustic path, the techniques described herein can achieve optimal noise cancellation at the desired location, regardless of the direction of propagation of the noise.

Abstract: Systems and methods described herein provide for low latency active noise cancellation, which alleviates the problems associated with analog filter circuitry. The present technology utilizes low latency digital signal processing techniques that overcome the high latency conventionally associated with conversion between the analog and digital domains. As a result, low latency active noise cancellation is performed utilizing digital filter circuitry which is not subject to the inaccuracies and drift of analog filter components. In doing so, the present technology provides robust, high quality active noise cancellation.

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 noise reduction of an acoustic signal using a configurable classification threshold which provides a sophisticated level of control to balance the tradeoff between positional robustness and noise reduction robustness. The configurable classification threshold corresponds to a configurable spatial region, such that signals arising from sources within the configurable spatial region are preserved, and signals arising from sources outside it are rejected. In embodiments, the configurable classification threshold can be automatically and dynamically adjusted in real-time based on evaluated environmental conditions surrounding an audio device implementing the noise reduction techniques described herein.

Abstract: Provided are methods and systems for digital signal processing by applying an improved SDFT technique for eliminating ripple in side lobes of a resulting spectrum. An exemplary method may comprise receiving an input signal which includes a number of discrete samples taken at regular time intervals. The input signal may then be filtered by a modified comb filter according to the sliding DFT technique to generate first and second filtered signals, the second filtered signal being filtered using a different filter coefficient. The first and second filtered signals may be processed by a first and second plurality of resonators respectively, according to the sliding DFT technique, to generate respective first and second SDFT output signals. The first and second SDFT output signals may then be selectively summed to generate a resulting output signal.

Abstract: A robust noise reduction system may concurrently reduce noise and echo components in an acoustic signal while limiting the level of speech distortion. The system may receive acoustic signals from two or more microphones in a close-talk, hand-held or other configuration. The received acoustic signals are transformed to frequency domain sub-band signals and echo and noise components may be subtracted from the sub-band signals. Features in the acoustic sub-band signals are identified and used to generate a multiplicative mask. The multiplicative mask is applied to the noise subtracted sub-band signals and the sub-band signals are reconstructed in the time domain.

Abstract: Systems and methods described herein provide for low latency active noise cancellation, which alleviates the problems associated with analog filter circuitry. The present technology utilizes low latency digital signal processing techniques that overcome the high latency conventionally associated with conversion between the analog and digital domains. As a result, low latency active noise cancellation is performed utilizing digital filter circuitry which is not subject to the inaccuracies and drift of analog filter components. In doing so, the present technology provides robust, high quality active noise cancellation.

Abstract: A method, system, and computer program for enhancing a signal are presented. The signal is received, and energy estimates of the signal may be determined. At least one characteristic of the signal may be inferred based on the energy estimates. A mask may be generated based, in part, on the at least one characteristic. In turn, the mask may be applied to the signal to produce an enhanced signal, which may be outputted.

Abstract: Background audio can be provided during telephonic communication. Telephonic communication can be established via a network, such as between a user of a telephony device and a communication partner having a second telephony device. A voice signal may be received from the user via a microphone integral with the telephony device. An audio track can be retrieved, for example, from memory integral with the telephony device or from a third-party service provider via the communications network. Noise reduction is performed on the voice signal to produce a clean voice signal. The clean voice signal may be combined with the audio track to produce a combined signal, such that the audio track provides background audio to the clean voice signal. The combined signal can then be transmitted from the telephony device to the second telephony device via the communications network.

Abstract: Systems and methods for envelope-based acoustic echo cancellation in a communication device are provided. In exemplary embodiments, a primary acoustic signal is received via a microphone of the communication device, and a far-end signal is received via a receiver. Frequency analysis is performed on the primary acoustic signal and the far-end acoustic signal to obtain frequency sub-bands. An echo mask based on magnitude envelopes of the primary and far-end acoustic signals for each frequency sub-band is generated. A noise mask based on at least the primary acoustic signal for each frequency sub-band may also be generated. A combination of the echo mask and noise mask may then be applied to the primary acoustic signal to generate a masked signal. The masked signal is then output.

Abstract: The present technology provides adaptive noise reduction of an acoustic signal using a sophisticated level of control to balance the tradeoff between speech loss distortion and noise reduction. The energy level of a noise component in a sub-band signal of the acoustic signal is reduced based on an estimated signal-to-noise ratio of the sub-band signal, and further on an estimated threshold level of speech distortion in the sub-band signal. In embodiments, the energy level of the noise component in the sub-band signal may be reduced to no less than a residual noise target level. Such a target level may be defined as a level at which the noise component ceases to be perceptible.

Abstract: The present technology provides adaptive noise reduction of an acoustic signal using a sophisticated level of control to balance the tradeoff between speech loss distortion and noise reduction. The energy level of a noise component in a sub-band signal of the acoustic signal is reduced based on an estimated signal-to-noise ratio of the sub-band signal, and further on an estimated threshold level of speech distortion in the sub-band signal. In embodiments, the energy level of the noise component in the sub-band signal may be reduced to no less than a residual noise target level. Such a target level may be defined as a level at which the noise component ceases to be perceptible.

Abstract: The present technology provides systems and methods for robust feedforward active noise cancellation which can overcome or substantially alleviate problems associated with the diverse and dynamic nature of the surrounding acoustic environment. A multi-faceted analysis decouples the background noise within the earpiece from the acoustic wave (e.g. the anti-noise and desired audio) generated by an audio transducer within the earpiece. A difference signal is formed utilizing monitoring signals captured by array of monitoring microphones within the earpiece. The difference signal is formed such that contributions due to the acoustic wave generated by the audio transducer are selectively attenuated. As a result, the difference signal indicates an acoustic energy level of the background noise within the earpiece.

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: Systems and methods for asynchronous sample rate conversion are provided that allow time-varying arbitrary sample ratios. An uncorrected ratio between two arbitrary sample rates is corrected and subsequently used to perform an efficient sample rate conversion on the samples in a data stream. Coefficients of a (polyphase) finite impulse response filter are interpolated based on a current time register value. Additional computational efficiency (and a smaller finite impulse response filter) may be used due to oversampling the input signal to the finite impulse response filter.

Abstract: Provided are methods and systems for performing binary data conversion in digital-to-analog converters (DACs) having sigma-delta modulators and involving data directed scrambling (DDS) algorithms to effectively shape the element mismatch errors to first-order shaped wideband noise. An example method may commence with receiving binary data, which comprises positive binary data and negative binary data. The method may further comprise converting, by a DDS unit, the binary data into binary thermometer code data. The conversion is performed by selectively assigning one or more logic ones in a data array from a current location of a positive (negative) pointer when positive (negative) binary data is received. A number of the one or more logic ones may be associated with the binary data. The method may further comprise circularly shifting the positive (negative) pointer to a new location within the array by the number of the one or more logic ones.