Abstract: A method and computing system for suppressing noise in an audio signal, comprising: receiving the audio signal at signal processing means; determining that another signal is input to the signal processing means, the input signal resulting from an activity which generates noise in the audio signal; and selectively suppressing noise in the audio signal in dependence on the determination that the input signal is input to the signal processing means to thereby suppress the generated noise in the audio signal.

Abstract: A method and system for modifying an audio signal, the method comprising: receiving the audio signal at signal processing means; analysing the received audio signal to identify characteristic signal components in the audio signal; determining that another signal is input to the signal processing means, the input signal resulting from an activity which generates noise in the audio signal; and selectively applying an adjusted gain to the audio signal based on the determination that the input signal is input to the signal processing means, wherein the adjusted gain is generated in dependence upon the signal strength of the identified signal components.

Abstract: The disclosure relates to encoding and decoding of digital data, and in particular to lossless arithmetic encoding and decoding of digital data representing audio, image or video data. A probability density function used for lossless arithmetic encoding of digital data is controlled by employing one or more parameters that changes over the set of data to be encoded. A parametric model in the form of an envelope function describes the spread of quantization indices derived from the data in a transform domain. By transmitting the one or more parameters together with the arithmetically encoded data, a receiving decoder may decode the data by exploiting the same parametric model as used by the encoder.

Abstract: The invention provides a method of controlling transmission of packets from a transmitter to a receiver via a channel, and a corresponding transmitter and receiver. The method comprises: transmitting packets from a queue, each packet having a packet size based on data in the packet; determining a transmission time for each packet, based on a transmission clock; determining a reception time of each packet, based on a reception clock; supplying to an estimation function successive sets of observations including in each set transmission time, reception time and packet size, the estimate function being arranged to provide an estimate of bandwidth for the channel using the relationship between the bandwidth, the amount of data in the queue, packet size and the delay between transmitting successive packets from the queue; and using the estimated bandwidth to control transmission of packets.

Abstract: In an embodiment, a method for removing an echo component is a first signal received at an audio input device includes outputting a second signal from an audio output device, receiving at the audio input device the first signal wherein the echo component in the first signal is the result of the second signal traversing an echo path; detecting if the first signal only comprises the echo component; determining an estimate indicative of the echo path by comparing the first signal and the second signal when it is detected that the first signal only comprises echo; applying the estimate indicative of the echo path to the first signal to determine an equalised first signal; comparing the equalised first signal with the second signal to determine an estimate indicative of the echo component; and removing the echo component from the first signal in dependence on the estimate indicative of the echo component; wherein the step of detecting if the first signal only comprises the echo component comprises comparing a char

Abstract: A method for removing an echo component in a first signal received at an audio input device, said method comprising the steps of outputting a second signal from an audio output device, receiving at the audio input device the first signal wherein the echo component in the first signal is the result of the second signal traversing an echo path having a frequency response which defines the relationship between the echo component and the second signal; determining an echo path model indicating an estimate of the frequency response of the echo path; removing an estimate of the echo component from the first signal using the echo path model; detecting if the first signal is overloaded; and wherein if it is detected that the first signal is overloaded the step of determining an echo path model comprises determining an overload echo path model such that the estimate of the frequency response indicated by the overload echo path model is only allowed to increase over time regardless of whether the frequency response of

Abstract: A method for reducing ringing in a signal output from a filter comprising inputting a signal into a filter; filtering a first portion of the input signal to generate a filtered portion of the output signal; analyzing the filtered portion of the output signal; detecting if ringing is present in the filtered portion of the output signal based on said analysis; and adjusting the filter characteristics to reduce ringing in a subsequent filtered portion of the output signal if it is determined that ringing is present.

Abstract: A method of estimating noise in data containing voice information and noise includes receiving the data as a sequence of input values; transforming the data by applying a first non linear mapping to the input values wherein the derivative function of the mapping decreases in magnitude as the input values increase in magnitude smoothing the transformed data; and transforming the smoothed transformed data by applying a second non linear mapping that is opposite to the first non linear mapping, to determine an estimate of the noise in the inputted data.

Abstract: The disclosure relates to encoding and decoding of digital data, and in particular to lossless arithmetic encoding and decoding of digital data representing audio, image or video data. A probability density function used for lossless arithmetic encoding of digital data is controlled by employing one or more parameters that changes over the set of data to be encoded. A parametric model in the form of an envelope function describes the spread of quantization indices derived from the data in a transform domain. By transmitting the one or more parameters together with the arithmetically encoded data, a receiving decoder may decode the data by exploiting the same parametric model as used by the encoder.

Abstract: A wideband speech encoder according to one embodiment includes a narrowband encoder and a highband encoder. The narrowband encoder is configured to encode a narrowband portion of a wideband speech signal into a set of filter parameters and a corresponding encoded excitation signal. The highband encoder is configured to encode, according to a highband excitation signal, a highband portion of the wideband speech signal into a set of filter parameters. The highband encoder is configured to generate the highband excitation signal by applying a nonlinear function to a signal based on the encoded narrowband excitation signal to generate a spectrally extended signal.

Abstract: A wideband speech encoder according to one embodiment includes a filter bank having a lowband processing path and a highband processing path. The processing paths have overlapping frequency responses. A first encoder is configured to encode a speech signal produced by the lowband processing path according to a first coding methodology. A second encoder is configured to encode a speech signal produced by the highband processing path according to a second coding methodology that is different than the first coding methodology.

Abstract: In one embodiment, a highband burst suppressor includes a first burst detector configured to detect bursts in a lowband speech signal, and a second burst detector configured to detect bursts in a corresponding highband speech signal. The lowband and highband speech signals may be different (possibly overlapping) frequency regions of a wideband speech signal. The highband burst suppressor also includes an attenuation control signal calculator configured to calculate an attenuation control signal according to a difference between outputs of the first and second burst detectors. A gain control element is configured to apply the attenuation control signal to the highband speech signal. In one example, the attenuation control signal indicates an attenuation when a burst is found in the highband speech signal but is absent from a corresponding region in time of the lowband speech signal.

Abstract: In one embodiment, a method of signal processing includes encoding a low-frequency portion of a speech signal into at least an encoded narrowband excitation signal and a plurality of narrowband filter parameters; and generating a highband excitation signal based on a narrowband excitation signal. The narrowband excitation signal is based on the encoded narrowband excitation signal. The method also includes encoding a high-frequency portion of the speech signal into at least a plurality of highband filter parameters according to at least the highband excitation signal. The encoded narrowband excitation signal includes a time warping, and the method includes applying a time shift to the high-frequency portion based on information related to the time warping.

Abstract: A method of signal processing according to one embodiment includes calculating an envelope of a first signal that is based on a low-frequency portion of a speech signal, calculating an envelope of a second signal that is based on a high-frequency portion of the speech signal, and calculating a plurality of gain factor values according to a time-varying relation between the envelopes of the first and second signal. The method includes attenuating, based on a variation over time of a relation between the envelopes of the first and second signals, at least one of the plurality of gain factor values. In one example, the variation over time of a relation between the envelopes is indicated by at least one distance among the plurality of gain factor values.

Abstract: In one embodiment, a method of signal processing includes calculating an envelope of a first signal that is based on a low-frequency portion of a speech signal, calculating an envelope of a second signal that is based on a high-frequency portion of the speech signal, and calculating a first plurality of gain factor values according to a time-varying relation between the envelopes of the first and second signals. The method includes, based on the first plurality of gain factor values, calculating a plurality of smoothed gain factor values. In one example, each of the plurality of smoothed gain factor values is based on a weighted sum of at least one of the first plurality of gain factor values and at least one smoothed gain factor value. In another example, the sum is adaptively weighted based on at least one distance among the plurality of gain factor values.

Abstract: In one embodiment, a method of generating a highband excitation signal includes generating a spectrally extended signal by extending the spectrum of a signal that is based on an encoded lowband excitation signal; and performing anti-sparseness filtering of a signal that is based on the encoded lowband excitation signal. In this method, the highband excitation signal is based on the spectrally extended signal, and the highband excitation signal is based on a result of the anti-sparseness filtering.

Abstract: In one embodiment, a method of generating a highband excitation signal includes harmonically extending the spectrum of a signal that is based on a lowband excitation signal; calculating a time-domain envelope of a signal that is based on the lowband excitation signal; and modulating a noise signal according to the time-domain envelope. The method also includes combining (A) a harmonically extended signal based on a result of the harmonically extending and (B) a modulated noise signal based on a result of the modulating. In this method, the highband excitation signal is based on a result of the combining.

Abstract: A quantizer according to an embodiment is configured to quantize a smoothed value of an input value (e.g., a vector of line spectral frequencies) to produce a corresponding output value, where the smoothed value is based on a scale factor and a quantization error of a previous output value.