Abstract: Methods and systems specifically designed to reduce loudspeaker distortion by reducing voice coil excursion are provided. An input audio signal is processed based on a specific linear model of a loudspeaker, and a dynamic filter is generated and applied to this audio signal. The filter changes the relative phases of the spectral components of the input signal to reduce estimated excursion peaks. The quality of the audio signal is not diminished in comparison to traditional filter approaches. The processing of the input signal may involve determining a main frequency which may be used to determine the fundamental frequency. Multiples of the fundamental frequencies provide harmonic frequencies. The phases of all the harmonic frequencies, including the fundamental, may be measured and compared to a target vector of phases. The difference between the measured phases and the target phases may then be used to calculate the poles and zeros and corresponding filter coefficients.

Abstract: The present technology provides robust, high quality expansion of the speech within a narrow bandwidth acoustic signal which can overcome or substantially alleviate problems associated with expanding the bandwidth of the noise within the acoustic signal. The present technology carries out a multi-faceted analysis to accurately identify noise within the narrow bandwidth acoustic signal. Noise classification information regarding the noise within the narrow bandwidth acoustic signal is used to determine whether to expand the bandwidth of the narrow bandwidth acoustic signal. By expanding the bandwidth based on the noise classification information, the present technology can expand the speech bandwidth of the narrow bandwidth acoustic signal and prevent or limit the bandwidth expansion of the noise.

Abstract: Provided are methods and systems for reducing a transition rate in transmitting data between analog and digital chips in Sigma-Delta Modulator (SDM) based Digital to Analog Converters (DACs) and Analog to Digital Converters (ADCs) intended to be used in audio signal processing. An example method may comprise receiving, by a digital chip, SDM binary data, mapping the SDM binary data to transition binary codes, and transmitting the transition binary codes to an analog chip. The mapping can be carried out according to a principle that the more commonly used SDM binary data codes correspond to transition binary data codes that require that fewer transitions occur in the signals between the chips. The methods and systems described provide for lowering the power needed for carrying out the data transmission between digital and analog chips.

Abstract: An audio equivalent of a video zoom feature for video recording and communication applications, as well as video post production processes. The audio zoom may operate in conjunction with a video zoom feature or independently. The audio zoom may be achieved by controlling reverberation effects of a signal, controlling a gain of the signal, as well as controlling the width of a directional beam which is used to select the particular audio component to focus on. The audio zoom may operate in response to user input, such as a user selection of a particular direction, or automatically based a current environment or other factors.

Abstract: A system for audio processing of an acoustic signal captured by at least one microphone based on analysis of a video signal associated with the acoustic signal is provided. The video signal may be dynamically processed and analyzed in real time by the system to locate one or more sound sources and to determine distances between the sound sources and the at least one microphone. This information may be used to perform selective noise suppression, noise cancellation, or selective adjustment of acoustic signal components or energy levels acoustic signal components such that noise or unrelated acoustic signal components associated with sound sources not associated with located sound sources are suppressed or filtered. The video signal analysis may track sound source movement for dynamic steering of an acoustic beam towards the sound source.

Abstract: Systems and methods for noise suppression using noise subtraction processing are provided. The noise subtraction processing comprises receiving at least a primary and a secondary acoustic signal. A desired signal component may be calculated and subtracted from the secondary acoustic signal to obtain a noise component signal. A determination may be made of a reference energy ratio and a prediction energy ratio. A determination may be made as to whether to adjust the noise component signal based partially on the reference energy ratio and partially on the prediction energy ratio. The noise component signal may be adjusted or frozen based on the determination. The noise component signal may then be removed from the primary acoustic signal to generate a noise subtracted signal which may be outputted.

Abstract: Techniques for replacing an advertisement in a webpage from a website with another advertisement are disclosed. A response including the webpage is intercepted in a network device deployed in an in-line fashion, preferably at a data traffic point along a network. When it is determined that the webpage includes an advertisement that is replaceable, a preferable advertisement more correlated to the interests of a user is embedded in data packets to replace those for the original advertisement. Subsequently, the preferable advertisement is served when the webpage is displayed.

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 various 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: Provided are systems and methods for image enhancement based on combining multiple related images, such as images of the same object taken from different imaging angles. This approach allows simulating images captured from longer distances using telephoto lenses. Initial images may be captured using simple cameras equipped with shorter focal length lenses, typically used on camera phones, tablets, and laptops. The initial images may be taken using two different cameras positioned a certain distance from each other. An object or, more specifically, a center line of the object is identified in each image. The object is typically present in the foreground portion of the initial images. The initial images may be cross-faded along the object center line to yield a combined image. Separating of the foreground and background portions of each image may be separated and separately processed, such as blurring the background portion and sharpening the foreground portion.

Abstract: Systems and methods for adaptive power control are provided. A minimum power level of a primary acoustic signal is estimated. The minimum power level may then be compared to at least one power threshold. Subsequently, a large power consuming system is controlled based on the comparison of the minimum power level to the power threshold.

Abstract: Arithmetic units and methods for floating point processing are provided. In exemplary embodiments, data paths to and from multiple multipliers and adders are flexibly combined through crossbars and alignment units to allow a wide range of mathematical operations, including affine and SIMD operations. The micro-architecture for a high-performance flexible vector floating point arithmetic unit is provided, which can perform a single-cycle throughput complex multiply-and-accumulate operation, as well as a Fast Fourier Transform (radix-2 decimation-in-time) Butterfly operation.

Abstract: Systems and methods for providing voice equalization are provided. In exemplary embodiments, acoustic signals are received from both a near-end and a far-end environment. A power spectrum estimate for the far-end signal and a noise estimate based on the near-end signal are determined. A voice equalization mask based on the power spectrum estimate of the far-end signal and the noise estimate based on the near-end signal is generated and applied to the far-end signal to obtain a modified signal. The modified signal may then be dynamically processed to obtain an enhanced audio signal. The enhanced audio signal is then output.

Abstract: Cross-talk reduction and/or cancellation systems and methods are provided herein. In exemplary embodiments, a far-end acoustic signal is delayed by M samples. Additionally, a cross-talk estimate value for the delayed far-end acoustic signal may be subtracted from an input acoustic signal. The cross-talk estimate value is a scaled version of filter outputs generated by a finite impulse response filter that utilizes predetermined filter coefficients. The filter outputs are scaled using a dynamic gain value.

Abstract: Provided are methods and systems for noise suppression within multiple time-frequency points of spectral representations. A multi-feature cluster tracker is used to track signal and noise sources and to predict signal versus noise dominance at each time-frequency point. Multiple features, such as binaural and monaural features, may be used for these purposes. A Gaussian mixture model (GMM) is developed and, in some embodiments, dynamically updated for distinguishing signal from noise and performing mask-based noise reduction. Each frequency band may use a different GMM or share a GMM with other frequency bands. A GMM may be combined from two models, with one trained to model time-frequency points in which the target dominates and another trained to model time-frequency points in which the noise dominates. Dynamic updates of a GMM may be performed using an expectation-maximization algorithm in an unsupervised fashion.

Abstract: Provided are systems and methods for communicating between mobile devices. Such methods involve establishing a communication session between the mobile devices using a local network, such as a Wi-Fi network. The methods also involve transmitting data streams from one mobile device in the group to all other mobile devices using this local network. The data streams may include audio and/or video data generated by the mobile devices. For example, mobile device users may conduct a teleconference through the local network and/or share media. The mobile devices may be interconnected using a star or ring topology, which may depend on distances between participants and/or alarm features that reflect connection losses between the mobile devices. Each mobile device may be equipped with one or more microphones for collecting audio signals from its user. A mobile device may also include an audio processing system for noise suppression and/or echo cancellation.

Abstract: Systems and methods for sample rate tracking are provided. An example method includes computing an actual latency associated with an output sample from an output sample stream. The actual latency is calculated using a phase and a phase increment (conversion rate ratio). A measured latency is determined using an internal clock using a presentation time of the output sample, or an input sample from an input sample stream, or both. The measured latency is compared to the actual latency to generate a latency error. A successive phase increment can be determined based on the latency error by using a low-pass or adaptive filter to adjust the latency error.

Abstract: Null processing noise subtraction is performed per sub-band and time frame for acoustic signals received from multiple microphones. The acoustic signals may include a primary acoustic signal and one or more additional acoustic signals. A noise component signal may be determined for each additional acoustic signal in each sub-band of signals received by N microphones by subtracting a desired signal component within every other acoustic signal weighted by a complex-valued coefficient ? from the secondary acoustic signal. The noise component signals, each weighted by a corresponding complex-valued coefficient ?, may then be subtracted from the primary acoustic signal resulting in an estimate of a target signal (i.e., a noise subtracted signal).

Abstract: Systems and methods for controlling adaptivity of noise cancellation are presented. One or more audio signals are received by one or more corresponding microphones. The one or more signals may be decomposed into frequency sub-bands. Noise cancellation consistent with identified adaptation constraints is performed on the one or more audio signals. The one or more audio signals may then be reconstructed from the frequency sub-bands and outputted via an output device.

Abstract: Systems and methods for reconstructing decomposed audio signals are presented. In exemplary embodiments, a decomposed audio signal is received. The decomposed audio signal may include a plurality of frequency sub-band signals having successively shifted group delays as a function of frequency from a filter bank. The plurality of frequency sub-band signals may then be grouped into two or more groups. A delay function may be applied to at least one of the two or more groups. Subsequently, the groups may be combined to reconstruct the audio signal, which may be outputted accordingly.

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.