Nuance Communications, Inc.

Harman Becker Automotive Systems Gmbh

Harman International Industries, Inc.

Other

Tally Computer Drucker Gmbh

Qnx Software Systems Limited

Cerence Operating Company

Svox Ag

Gerhard Uwe Schmidt

Tim Haulick

Markus Buck

Tobias Wolff

Bernd Iser

Mohamed Krini

Günther Wirsching

Arthur Wolf

Martin Roessler

Michael Tropp

Markus E Christoph

Tobias Herbig

Franz Gerl

Lars König

Guido Kolano

Walter Schnug

Toby Christian Lawin-Ore

Hans-Jörg Köpf

Harald Lenhardt

Olga González Valbuena

Samuel Ngouoko Mboungueng

Shreyas Anand Paranjpe

Johann Stempfle

Thomas Franke

David Giesbrecht

Andreas Löw

Raymond Brückner

Gerhard Nüssle

Clarence Chu

Hans-Joerg Koepf

Matthias Schulz

Gerhard Beurer

Klaus Wegner

Martin Rössler

Wolfgang Warzawa

Harald Loser

Wolfgang Kretschmer

Werner Mascher

Martin Röβler

Ange Tchinda-Pockem

Olga Gonzalez Valbuena

Mathias Roder

Viola Hoff

Gerhard Nuessle

Patrick Hannon

Jochen Withopf

A system used in a loudspeaker-room-microphone environment includes a microphone signal partitioner that divides a signal from a microphone into one or more divided portions. A reverberation energy estimator estimates reverberation energy in some of the divided portions of the microphone signal based on a loudspeaker signal. The estimated reverberation energy is processed to generate a dereverberated output signal.

Dereverberation system for use in a signal processing apparatus

Methods and apparatus for determining phase shift information between the first and second microphone signals for a sound signal, and determining an angle of incidence of the sound in relation to the first and second positions of the first and second microphones from the phase shift information of a band-limited test signal received by the first and second microphones for a frequency range of interest.

Speaker localization

A speech processing device includes an automotive device that filters data that is sent and received across an in-vehicle bus. The device selectively acquires vehicle data related to a user settings or adjustments. An interface acquires the selected vehicle data from in-vehicle sensors in response to a user's articulation of a first code phrase. A memory stores the selected vehicle data with unique identifying data associated with the user and establishes a connection between the selected vehicle data and the user when a second code phrase is articulated. A data interface provides access to the selected vehicle data and stored relationship data and enables the processing of the data to customize the in-vehicle system. The data interface is responsive to the user's articulation of a third code phrase to process the selected vehicle data that enables the setting or adjustment of the in-vehicle system.

Adjusting or setting vehicle elements through speech control

Methods and apparatus to beamform a first plurality of microphone signals using at least one beamforming weight to obtain a first beamformed signal, beamform a second plurality of microphone signals using the at least one beamforming weight to obtain a second beamformed signal, and adjust the at least one beamforming weight so that the power density of at least one perturbation component present in the first or the second plurality of microphone signals is reduced.

Beamforming pre-processing for speaker localization

A system locates a speaker in a room containing a loudspeaker and a microphone array. The loudspeaker transmits a sound that is partly reflected by a speaker. The microphone array detects the reflected sound and converts the sound into a microphone array, the speaker's distance from the microphone array, or both, based on the characteristics of the microphone signals.

Acoustic localization of a speaker

The invention provides a method for determining a noise reference signal for noise compensation and/or noise reduction. A first audio signal on a first signal path and a second audio signal on a second signal path are received. The first audio signal is filtered using a first adaptive filter to obtain a first filtered audio signal. The second audio signal is filtered using a second adaptive filter to obtain a second filtered audio signal. The first and the second filtered audio signal are combined to obtain the noise reference signal. The first and the second adaptive filter are adapted such as to minimize a wanted signal component in the noise reference signal.

Method for determining a noise reference signal for noise compensation and/or noise reduction

Methods and apparatus for beamforming and performing echo compensation for the beamformed signal with an echo canceller including calculating a set of filter coefficients as an estimate for a new steering direction without a complete adaptation of the echo canceller.

Method for determining a set of filter coefficients for an acoustic echo compensator

A signal processing system enhances an audio signal. The audio signal is divided into audio sub-band signals. Some audio sub-band signals are excised. Other audio sub-band signals are processed to obtain enhanced audio sub-band signals. At least a portion of the excised audio sub-band signals are reconstructed. The reconstructed audio sub-band signals are synthesized with the enhanced audio sub-band signals to form an enhanced audio signal.

Efficient audio signal processing in the sub-band regime

Embodiments of the present invention exploit redundancy of succeeding FFT spectra and use this redundancy for computing interpolated temporal supporting points. An analysis filter bank converts overlapped sequences of an audio (ex. loudspeaker) signal from a time domain to a frequency domain to obtain a time series of short-time loudspeaker spectra. An interpolator temporally interpolates this time series. The interpolation is fed to an echo canceller, which computes an estimated echo spectrum. A microphone analysis filter bank converts overlapped sequences of an audio microphone signal from the time domain to the frequency domain to obtain a time series of short-time microphone spectra. The estimated echo spectrum is subtracted from the microphone spectrum. Further signal enhancement (filtration) may be applied. A synthesis filter bank converts the filtered microphone spectra to the time domain to generate an echo compensated audio microphone signal. Computational complexity of signal processing systems can, therefore, be reduced.

Temporal interpolation of adjacent spectra

A system and method for compensating audio signal components is disclosed. The method includes detecting, by a microphone, a sound signal. The sound signal comprises audio signal components resulting from reproducing an audio signal of an audio source and speech signal components corresponding to a speech signal from a person. The sound signal is filtered to whiten the sound signal. The audio signal components in the whitened sound signal are then compensated. The whitening of the compensated sound signal is removed. The filtering of the audio signal is performed using at least two filters in an alternating way, each filter using time-dependent filter coefficients.

Mono and multi-channel echo compensation from selective output

A method for detecting barge-in in a speech dialog system comprising determining whether a speech prompt is output by the speech dialog system, and detecting whether speech activity is present in an input signal based on a time-varying sensitivity threshold of a speech activity detector and/or based on speaker information, where the sensitivity threshold is increased if output of a speech prompt is determined and decreased if no output of a speech prompt is determined. If speech activity is detected in the input signal, the speech prompt may be interrupted or faded out. A speech dialog system configured to detect barge-in is also disclosed.

Detecting barge-in in a speech dialogue system

The invention provides a method for estimating a fundamental frequency of a speech signal comprising the steps of receiving a signal spectrum of the speech signal, filtering the signal spectrum to obtain a refined signal spectrum, determining a cross-power spectral density using the refined signal spectrum and the signal spectrum, transforming the cross-power spectral density into the time domain to obtain a cross-correlation function, and estimating the fundamental frequency of the speech signal based on the cross-correlation function.

Method for estimating a fundamental frequency of a speech signal

A communication system enhances communications in a noisy environment. Multiple input arrays convert a voiced or unvoiced signal into an analog signal. A converter receives the analog signal and generates digital signals. A digital signal processor determines temporal and spatial information from the digital signals. The processed signals are then converted to audible sound.

Noisy environment communication enhancement system

A system enhances speech by detecting a speaker's utterance through a first microphone positioned a first distance from a source of interference. A second microphone may detect the speaker's utterance at a different position. A monitoring device may estimate the power level of a first microphone signal. A synthesizer may synthesize part of the first microphone signal by processing the second microphone signal. The synthesis may occur when power level is below a predetermined level.

System enhancement of speech signals

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Ulm, Germany

Gerhard Uwe Schmidt