Abstract: The perceived quality of a narrowband speech signal truncated from a wideband speech signal is improved by generating in a third frequency band third speech components matching first speech components in a first frequency band of the narrowband signal, and generating in a fourth frequency band fourth speech components matching second speech components in a second frequency band of the narrowband signal. A first gain factor is applied to the third speech components to generate adjusted third speech components, and a second gain factor is applied to the fourth speech components to generate adjusted fourth speech components, the gain factors being selected such that the ratios of the average powers of the adjusted third and fourth speech components to the average power of the first speech components are predetermined values.

Abstract: The perceived quality of a speech signal is improved by estimating the average power of first and second signal components and applying a first gain factor to the second signal components to generate adjusted second signal components. The first gain factor is selected such that on application of the first gain factor to the second signal components, the ratio of the average power of the first signal components to the average power of the adjusted second signal components would be a first predetermined value, the first predetermined value being such as to inhibit perceptual distortion of the improved speech signal.

Abstract: An audio handling device comprising: a source of audio data; a microphone; a loudspeaker; a transmitter for transmitting audio data; modification means for modifying the audio data; and a control unit for controlling the operation of the device, the control unit being capable of receiving signals from the microphone and configuring the conveying of audio data from the source to one or both of the loudspeaker and the transmitter; the control unit being capable of configuring the device such that during at least a probing period the modification means modifies audio data from the source and the modified audio data is transmitted by the transmitter, and being arranged to select in dependence on data dependent on signals received from the microphone whether to apply audio data from the source to the loudspeaker.

Abstract: An active noise cancellation controller for performing noise attenuation in a system over a predetermined frequency range, the active noise cancellation controller comprising: a first input for receiving a reference signal indicative of a noise level; a second input for receiving an error signal indicative of a remnant noise level; an output for providing a noise cancellation signal to a system in which noise attenuation is to be performed; a fixed feedback controller having a fixed infinite impulse response filter arranged for operation on an error signal received at the second input; a fixed feedforward controller having a fixed infinite impulse response filter arranged for operation on a reference signal received at the first input; and an adaptive feedforward controller having a digital adaptive finite impulse response filter arranged for operation on a reference signal received at the first input and on an error signal received at the second input, the coefficients of the digital adaptive filter being de

Abstract: A two microphone noise reduction system is described. In an embodiment, input signals from each of the microphones are divided into subbands and each subband is then filtered independently to separate noise and desired signals and to suppress non-stationary and stationary noise. Filtering methods used include adaptive decorrelation filtering. A post-processing module using adaptive noise cancellation like filtering algorithms may be used to further suppress stationary and non-stationary noise in the output signals from the adaptive decorrelation filtering and a single microphone noise reduction algorithm may be used to further provide optimal stationary noise reduction performance of the system.

Abstract: The perceived quality of a narrowband speech signal truncated from a wideband speech signal is improved by generating in a third frequency band third speech components matching first speech components in a first frequency band of the narrowband signal, and generating in a fourth frequency band fourth speech components matching second speech components in a second frequency band of the narrowband signal. A first gain factor is applied to the third speech components to generate adjusted third speech components, and a second gain factor is applied to the fourth speech components to generate adjusted fourth speech components, the gain factors being selected such that the ratios of the average powers of the adjusted third and fourth speech components to the average power of the first speech components are predetermined values.

Abstract: The perceived quality of a speech signal output from a user apparatus is improved by storing ambient noise profiles each indicating a model power distribution of a respective ambient noise type as a function of frequency; the ambient noise profile at the user apparatus is measured, the measured ambient noise profile is correlated with each of the stored ambient noise profiles, the stored ambient noise profile is selected with which the measured ambient noise profile is most highly correlated, and the speech signal is manipulated in dependence on which of the stored ambient noise profiles is selected, so as to form an improved speech signal.

Abstract: The perceived quality of a speech signal is improved by estimating the average power of first and second signal components and applying a first gain factor to the second signal components to generate adjusted second signal components. The first gain factor is selected such that on application of the first gain factor to the second signal components, the ratio of the average power of the first signal components to the average power of the adjusted second signal components would be a first predetermined value, the first predetermined value being such as to inhibit perceptual distortion of the improved speech signal.

Abstract: An audio handling device comprising: a source of audio data; a microphone; a loudspeaker; a transmitter for transmitting audio data; modification means for modifying the audio data; and a control unit for controlling the operation of the device, the control unit being capable of receiving signals from the microphone and configuring the conveying of audio data from the source to one or both of the loudspeaker and the transmitter; the control unit being capable of configuring the device such that during at least a probing period the modification means modifies audio data from the source and the modified audio data is transmitted by the transmitter, and being arranged to select in dependence on data dependent on signals received from the microphone whether to apply audio data from the source to the loudspeaker.

Abstract: A two microphone noise reduction system is described. In an embodiment, input signals from each of the microphones are divided into subbands and each subband is then filtered independently to separate noise and desired signals and to suppress non-stationary and stationary noise. Filtering methods used include adaptive decorrelation filtering. A post-processing module using adaptive noise cancellation like filtering algorithms may be used to further suppress stationary and non-stationary noise in the output signals from the adaptive decorrelation filtering and a single microphone noise reduction algorithm may be used to further provide optimal stationary noise reduction performance of the system.

Abstract: A signal processor for estimating noise power in an audio signal includes a filter unit for generating a series of power values, each power value representing the power in the audio signal at a respective one of a plurality of frequency bands; a signal classification unit for analysing successive portions of the audio signal to assess whether each portion contains features characteristic of speech, and for classifying each portion in dependence on that analysis; a correction unit for estimating a minimum power value in a time-limited part of the audio signal, estimating the total noise power in that part of the audio signal and forming a correction factor dependent on the ratio of the minimum power value to the estimated total noise power, the correction unit being configured to estimate the minimum power value and the total noise power over only those portions of the time-limited part of the signal that are classified by the signal classification unit as being less characteristic of speech; and a noise estim

Abstract: An audio processing device for reducing the effect on a first signal of echo from a second signal, the device comprising: an echo reduction processor for processing the first signal to reduce echo in it, the echo reduction unit having: a first mode of operation for reducing echo of a first function from the first signal; and a second mode of operation for reducing echo of a second function from the first signal, the second function being more complex than the first function and the echo reduction processor being such as to consume more power in the second mode of operation than in the first mode of operation; and an echo reduction controller for controlling the echo reduction processor to operate in a selected one of the first mode of operation and the second mode of operation.