Skype Limited

Qualcomm Incorporated

Global IP Solutions, Inc.

Global IP Solutions (gips) Ab

Microsoft Technology Licensing, LLC

Koen Bernard Vos

Karsten Vandborg Sorensen

Ananthapadmanabhan Aasanipalai Kandhadai

Jon Bergenheim

Soren Skak Jensen

Mattias Nilsson

Soren Vang Andersen

Ananthapadmanabhan Arasanipalai Kandhadai

Christoffer Asgaard Rödbro

Jan K Skoglund

Sriram Srinivasan

Soren Andersen

Karsten Vandborg Sørensen

Stefan Strommer

Jan Plasberg

Julian Spittka

Henrik Astrom

Soren Vang Anderson

Henrik Aström

Søren Skak Jensen

Stefan Kurt Olof Strömmer

Karsten Vandborg SØRENSEN

A method of regenerating wideband speech from narrowband speech, the method comprising: receiving samples of a narrowband speech signal having a first range of frequencies; identifying, based on a characteristic of the narrowband speech signal, frequencies in the first range of frequencies to translate into a target band of a regenerated speech signal; modulating the identified frequencies in the first range of frequencies of the received samples of the narrowband speech signal with a modulation signal, the modulation signal having a modulating frequency adapted to upshift the identified frequencies in the first range of frequencies into the target band; filtering the modulated samples, using a target band filter, to form the regenerated speech signal in the target band; and combining the narrowband speech signal with the regenerated speech signal to produce a new wideband speech signal.

Regeneration of wideband speech

Some embodiments describe methods, programs, and systems for speech encoding. Among other things, a received input signal representing a property of speech is quantized to generate a quantized output signal. Prior to the quantization, a version of the input signal is supplied to a first noise shaping filter having a first set of filter coefficients effective to generate a first filtered signal. Following the quantization, the quantized output signal is supplied to a second noise shaping filter having a second set of filter coefficients, thus generating a second filtered signal. A noise shaping operation is performed to control a frequency spectrum of a noise effect in the quantized output signal caused by the quantization, wherein the noise shaping operation is based on both the first and second filtered signals. Finally, the quantized output signal is transmitted in an encoded signal.

Speech encoding utilizing independent manipulation of signal and noise spectrum

A method of regenerating wideband speech from narrowband speech, the method comprising: receiving samples of a narrowband speech signal in a first range of frequencies; modulating received samples of the narrowband speech signal with a modulation signal having a modulating frequency adapted to upshift each frequency in the first range of frequencies by an amount determined by the modulating frequency wherein the modulating frequency is selected to translate into a target band a selected frequency band within the first range of signals; filtering the modulated samples using a target band filter to form a regenerated speech signal in the target band; and combining the narrow band speech signal with the regenerated speech signal in the target band to regenerate a wideband speech signal, the method comprising the step of controlling the modulated samples to lie in a second range of frequencies identified by determining a signal characteristic of frequencies in the first range of frequencies.

A method, system and program for encoding and decoding speech according to a source-filter model whereby speech is modelled to comprise a source signal filtered by a time-varying filter. The method comprises: receiving a speech signal comprising successive frames. For each of a plurality of frames of the speech signal: adding a predetermined noise signal generated by a quantization gain multiplied by 0.5 times an inverse of a pitch correlation to the speech signal to generate a simulated signal, determining linear predictive coding coefficients based on the simulated signal frame, and determining a linear predictive coding residual signal based on the linear predictive coding coefficients and one of the speech signal and the simulated signal. Then forming an encoded signal representing said speech signal, based on the linear predictive coding coefficients and the linear predictive coding residual signal.

Speech encoding by determining a quantization gain based on inverse of a pitch correlation

A terminal for transmitting a voice signal, comprising: a transmitter having a first transmission module arranged to transmit signals over a packet-switched data network and a second transmission module arranged to transmit signals over a circuit-switched telephony network; a microphone for generating a voice signal; and signal processing apparatus configured to generate a first signal and a second signal from the voice signal, each representing information from the voice signal over a same portion of time. The signal processing apparatus is further configured to supply the first signal to the first transmission module for transmission to a receiver via the packet-switched data network, and to supply the second signal to the second transmission module for transmission to the receiver via the circuit-switched telephony network. There is also provided a terminal for receiving such signals and reconstructing the voice signal, and corresponding methods and program products.

Enhanced circuit-switched calls

A first user terminal, host terminal, method and program. The first terminal comprises: a transceiver for communicating with a plurality of other user terminals over a communication network; and communications processing apparatus, coupled to the transceiver, and arranged to participate in a call with a selected number of the other user terminals via the transceiver and communication network, the call including transmission of a voice signal from the first user terminal. The communications processing apparatus is operable in a mode whereby it temporarily discontinues transmission of the voice signal in response to detecting less than a predetermined level of activity on said voice signal, and the communications processing apparatus is further configured to selectively enable that mode in dependence on the selected number of other user terminals in the call.

Controlling multi-party communications

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.

Gain control for an audio signal

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.

Noise suppression

A method, system and program for decoding a speech signal. In some embodiments, the method comprises: receiving an encoded speech signal having quantization values; transforming the quantization values by adding simulated random-noise samples; and from the encoded speech signal, determining a parameter of the transformation that is usable to control the transformation of the quantization values.

Speech coding by quantizing with random-noise signal

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.

Systems, methods, and apparatus for gain factor attenuation

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.

Systems, methods, and apparatus for gain factor smoothing

Some embodiments describe methods, programs, and systems for speech encoding. Among other things, a received input signal representing a property of speech is quantized to generate a quantized output signal. Prior to the quantization, a version of the input signal is supplied to a first noise shaping filter having a first set of filter coefficients effective to generate a first filtered signal. Following the quantization, the quantized output signal is supplied to a second noise shaping filter having a second set of filter coefficients, thus generating a second filtered signal. A noise shaping operation is performed to control a frequency spectrum of a noise effect in the quantized output signal caused by the quantization, wherein the noise shaping operation is based on both the first and second filtered signals. Finally, the quantised output signal is transmitted in an encoded signal.

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.

Method of estimating noise levels in a communication system

A method, system and program for encoding and decoding speech according to a source-filter model whereby speech is modelled to comprise a source signal filtered by a time-varying filter. The method comprises: receiving a speech signal comprising successive frames, for each of a plurality of frames of the speech signal, deriving a first line spectral frequency vector for a first portion of the frame, and a second line spectral frequency vector for a second portion of the frame, and determining a transmit line spectral frequency vector and an interpolation factor based on the first and second line spectral frequency vectors, and on the transmit line spectral frequency vector for a preceding one of the frames.

Growing community of inventors

San Francisco, CA, United States of America

Koen Bernard Vos