Cambridge Silicon Radio Limited

Clarity Technologies, Inc.

Csr Technology Inc.

Kuan-Chieh Yen

Rogerio Guedes Alves

Xuejing Sun

Sameer Gadre

Michael Christopher Vartanian

Bryan Neilson

Jeffrey S Chisholm

Lester Soon Huat Ngia

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 estimation unit for estimating noise in the audio signal in dependence on the power values output by the filter unit and the correction factor formed by the correction unit.

Noise reduction

A portable music player for the playback of a digital audio file comprises a memory for storing a plurality of digital audio files; an audio output; a control for setting a desired change in pitch or tempo; and a digital signal processor configured to process a digital audio file and recover an audio signal therefrom, perceptibly alter one of the pitch and the tempo of the audio signal in response to the desired change in pitch or tempo without perceptibly altering the other of the pitch and tempo, and output the altered audio signal to the audio output.

Portable music player

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.

Wind noise mitigation

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.

Audio processing device and method for reducing echo from a second signal in a first signal

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.

Error concealment

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.

Speech intelligibility

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.

Method and apparatus for improving intelligibility of audible speech represented by a speech signal

A method and system for clear signal capture comprehend several individual aspects that address specific problems in improved ways. In addition, the method and system also comprehend a hands-free implementation that is a practical solution to a very complex problem. Individual aspects comprehended related to echo and noise reduction, and divergence control.

Method and system for clear signal capture

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.

Audio usage detection

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.

Two microphone noise reduction system

Improved vocal signals are obtained in headsets and similar devices by including a microphone inside a chamber formed at least in part by the wearer's ear. This second microphone provides a reduced noise input signal. The reduced noise signal is corrected by input from another microphone, located outside the chamber. This correction can include echo cancellation, spectral shaping, frequency extension, and the like.

Growing community of inventors

Northville, MI, United States of America

Kuan-Chieh Yen