Abstract: Voice communication method and apparatus and method and apparatus for operating jitter buffer are described. Audio blocks are acquired in sequence. Each of the audio blocks includes one or more audio frames. Voice activity detection is performed on the audio blocks. In response to deciding voice onset for a present one of the audio blocks, a subsequence of the sequence of the acquired audio blocks is retrieved. The subsequence precedes the present audio block immediately. The subsequence has a predetermined length and non-voice is decided for each audio block in the subsequence. The present audio block and the audio blocks in the subsequence are transmitted to a receiving party. The audio blocks in the subsequence are identified as reprocessed audio blocks. In response to deciding non-voice for the present audio block, the present audio block is cached.

Abstract: An audio signal with a temporal sequence of blocks or frames is received or accessed. Features are determined as characterizing aggregately the sequential audio blocks/frames that have been processed recently, relative to current time. The feature determination exceeds a specificity criterion and is delayed, relative to the recently processed audio blocks/frames. Voice activity indication is detected in the audio signal. VAD is based on a decision that exceeds a preset sensitivity threshold and is computed over a brief time period, relative to blocks/frames duration, and relates to current block/frame features. The VAD and the recent feature determination are combined with state related information, which is based on a history of previous feature determinations that are compiled from multiple features, determined over a time prior to the recent feature determination time period. Decisions to commence or terminate the audio signal, or related gains, are outputted based on the combination.

Abstract: A method, an apparatus, and logic to post-process raw gains determined by input processing to generate post-processed gains, comprising using one or both of delta gain smoothing and decision-directed gain smoothing. The delta gain smoothing comprises applying a smoothing filter to the raw gain with a smoothing factor that depends on the gain delta: the absolute value of the difference between the raw gain for the current frame and the post-processed gain for a previous frame. The decision-directed gain smoothing comprises converting the raw gain to a signal-to-noise ratio, applying a smoothing filter with a smoothing factor to the signal-to-noise ratio to calculate a smoothed signal-to-noise ratio, and converting the smoothed signal-to-noise ratio to determine the second smoothed gain, with smoothing factor possibly dependent on the gain delta.

Abstract: Embodiments of method and apparatus for acoustic echo control are described. According to the method, an echo energy-based doubletalk detection is performed to determine whether there is a doubletalk in a microphone signal with reference to a loudspeaker signal. A spectral similarity between spectra of the microphone signal and the loudspeaker signal is calculated. It is determined that there is no doubletalk in the microphone signal if the spectral similarity is higher than a threshold level. Adaption of an adaptive filter for applying acoustic echo cancellation or acoustic echo suppression on the microphone signal is enabled if it is determined that there is no doubletalk in the microphone signal through the echo energy-based doubletalk detection, or there is no doubletalk through the spectral similarity-based doubletalk detection.

Abstract: Methods and apparatuses for convolutive blind source separation are described. Each of a plurality of input signals is transformed into frequency domain. Values of coefficients of unmixing filter corresponding to frequency bins are calculated by performing a gradient descent process on a cost function at least dependent on the coefficients of the unmixing filters. In each iteration of the gradient descent process, gradient terms for calculating the values of the same coefficient of the unmixing filters are adjusted to improve smoothness of gradient terms across the frequency bins. With respect to each of the frequency bins, source signals are estimated by filtering the transformed input signals through the respective unmixing filter configured with the calculated values of the coefficients. The estimated source signals on the respective frequency bins are transformed into time domain. The cost function is adapted to evaluate decorrelation between the estimated source signals.

Abstract: A method of compensating for noise in a receiver having a first receiver unit and a second receiver unit, the method includes receiving a first transmission at the first receiver unit, the first transmission having a first signal component and a first noise component; receiving a second transmission at the second receive unit, the second transmission having a second signal component and a second noise component; determining whether the first noise component and the second noise component are incoherent and; only if it is determined that the first and second noise components are incoherent, processing the first and second transmissions in a first processing path, wherein the first processing path is configured to compensate for incoherent noise.

Abstract: A method and apparatus for decoding portions of a data stream, wherein each portion comprises a plurality of samples. The method comprises storing portions of the data stream, decoding portions of the data stream to form decoded portions, and storing the decoded portions. The method further comprises identifying that a portion of the data stream is degraded. Following identifying that a portion of the data stream is degraded, the method generates a decoded portion for the degraded portion of the data stream using the stored decoded portions. The method also updates a state of a decoder by: estimating a pitch period of the degraded portion; selecting a group of successive samples of the stored portions of the data stream, the group of successive samples offset from the degraded portion in the data stream by a multiple of the estimated pitch period; and decoding the selected samples at the decoder.

Abstract: A method and apparatus for selectively replacing damaged portions of a data stream. The method comprises analyzing the data stream to identify damaged portions therein; selecting a damaged portion for replacement; and replacing the selected damaged portion. The selected damaged portion is selected for replacement in dependence on a rate of replacement, the rate of replacement being that at which previous portions of the data stream have been replaced.

Abstract: A method of suppressing wind noise in a voice signal determines an upper frequency limit that lies within the frequency spectrum of the voice signal, and for each of a plurality of frequency bands below the upper frequency limit, compares the average power of signal components in a first portion of the signal to the average power of signal components in a second portion of the signal, where the second portion is successive to the first portion. Signal components are identified in at least one of the plurality of frequency bands as containing impulsive wind noise in dependence on the comparison, and the identified signal components are attenuated.

Abstract: A method of enhancing an audio signal includes the steps of: a) receiving a primary audio input signal, b) receiving a detected audio signal which comprises: A) an echo component derived from play-out of the primary audio input signal and B) a noise component, and c) estimating from the primary audio input signal and the detected audio signal: 1) a set of frequency-specific lower bound gains, such that each frequency-specific lower bound gain, when applied to a respective frequency of the primary audio input signal, would cause the noise component to just mask the echo component at that respective frequency and 2) a set of frequency-specific upper bound gains, such that each frequency-specific upper bound gain, when applied to a respective frequency of the primary audio input signal, would cause the echo component to just mask the noise component at that respective frequency; d) estimating a set of frequency-specific gains in such a way that each frequency-specific gain falls between the respective frequency-

Abstract: A method and apparatus for selectively replacing damaged portions of a data stream. The method comprises analyzing the data stream to identify damaged portions therein; selecting a damaged portion for replacement; and replacing the selected damaged portion. The selected damaged portion is selected for replacement in dependence on a rate of replacement, the rate of replacement being that at which previous portions of the data stream have been replaced.

Abstract: Methods and apparatuses for convolutive blind source separation are described. Each of a plurality of input signals is transformed into frequency domain. Values of coefficients of unmixing filter corresponding to frequency bins are calculated by performing a gradient descent process on a cost function at least dependent on the coefficients of the unmixing filters. In each iteration of the gradient descent process, gradient terms for calculating the values of the same coefficient of the unmixing filters are adjusted to improve smoothness of gradient terms across the frequency bins. With respect to each of the frequency bins, source signals are estimated by filtering the transformed input signals through the respective unmixing filter configured with the calculated values of the coefficients. The estimated source signals on the respective frequency bins are transformed into time domain. The cost function is adapted to evaluate decorrelation between the estimated source signals.

Abstract: In some embodiments, a method for processing output of at least one microphone of a device (e.g., a headset) to identify at least one touch gesture exerted by a user on the device, including by distinguishing the gesture from input to the microphone other than a touch gesture intended by the user, and by distinguishing between a tap exerted by the user on the device and at least one dynamic gesture exerted by the user on the device, where the output of the at least one microphone is also indicative of ambient sound (e.g., voice utterences). Other embodiments are systems for detecting ambient sound (e.g., voice utterences) and touch gestures, each including a device including at least one microphone and a processor coupled and configured to process output of each microphone to identify at least one touch gesture exerted by a user on the device.

Abstract: A method and apparatus for selectively replacing damaged portions of a data stream. The method comprises analyzing the data stream to identify damaged portions therein; selecting a damaged portion for replacement; and replacing the selected damaged portion. The selected damaged portion is selected for replacement in dependence on a rate of replacement, the rate of replacement being that at which previous portions of the data stream have been replaced.

Abstract: A method of estimating a pitch period of a first portion of a signal wherein the first portion overlaps a previous portion. The method comprises computing a first autocorrelation value for part of the first portion not overlapping the previous portion. The method further comprises retrieving a stored second autocorrelation value for part of the first portion overlapping the previous portion, the second autocorrelation value having been computed during estimation of a pitch period of the previous portion. The method further comprises forming a combined autocorrelation value using the first and second autocorrelation values, and selecting the estimated pitch period in dependence on the combined autocorrelation value.

Abstract: A method of refining a pitch period estimation of a signal, the method comprising: for each of a plurality of portions of the signal, scanning over a predefined range of time offsets to find an estimate of the pitch period of the portion within the predefined range of time offsets; identifying the average pitch period of the estimated pitch periods of the portions; determining a refined range of time offsets in dependence on the average pitch period, the refined range of time offsets being narrower than the predefined range of time offsets; and for a subsequent portion of the signal, scanning over the refined range of time offsets to find an estimate of the pitch period of the subsequent portion.

Abstract: A method of compensating for noise in a receiver having a first receiver unit and a second receiver unit, the method includes receiving a first transmission at the first receiver unit, the first transmission having a first signal component and a first noise component; receiving a second transmission at the second receive unit, the second transmission having a second signal component and a second noise component; determining whether the first noise component and the second noise component are incoherent and; only if it is determined that the first and second noise components are incoherent, processing the first and second transmissions in a first processing path, wherein the first processing path is configured to compensate for incoherent noise.

Abstract: A method and apparatus for estimating the pitch period of a signal. The method includes identifying a first candidate pitch period by performing a search only over a first range of potential pitch periods. The method further includes determining a second candidate pitch period by dividing the first candidate pitch period by an integer, wherein the second candidate pitch period is outside the first range of potential pitch periods. The method further includes selecting as the estimate of the pitch period of the signal the smaller of the candidate pitch periods that is such that portions of the signal separated by that candidate pitch period are well correlated.

Abstract: A method of enhancing an audio signal includes the steps of: a) receiving a primary audio input signal, b) receiving a detected audio signal which comprises: A) an echo component derived from play-out of the primary audio input signal and B) a noise component, and c) estimating from the primary audio input signal and the detected audio signal: 1) a set of frequency-specific lower bound gains, such that each frequency-specific lower bound gain, when applied to a respective frequency of the primary audio input signal, would cause the noise component to just mask the echo component at that respective frequency and 2) a set of frequency-specific upper bound gains, such that, each frequency-specific upper bound gain, when applied to a respective frequency of the primary audio input signal, would cause the echo component to just mask the noise component, at that respective frequency; d) estimating a sat of frequency-specific gains in such a way that each frequency-specific gain falls between the respective frequenc

Abstract: A method of suppressing wind noise in a voice signal determines an upper frequency limit that lies within the frequency spectrum of the voice signal, and for each of a plurality of frequency bands below the upper frequency limit, compares the average power of signal components in a first portion of the signal to the average power of signal components in a second portion of the signal, where the second portion is successive to the first portion. Signal components are identified in at least one of the plurality of frequency bands as containing impulsive wind noise in dependence on the comparison, and the identified signal components are attenuated.