A mobile communications device contains at least two microphones. One microphone is designated by a selector to provide a voice dominant signal and another microphone is designated to provide a noise or echo dominant signal, for a call or a recording. The selector communicates the designations to a switch that routes the selected microphone signals to the inputs of a processor for voice signal enhancement. The selected voice dominant signal is then enhanced by suppressing ambient noise or canceling echo therein, based on the selected noise or echo dominant signal. The designation of microphones may change at any instant during the call or recording depending on various factors, e.g. based on the quality of the microphone signals. Other embodiments are also described.

Multiple microphone switching and configuration

Systems and methods for providing single microphone noise suppression fallback are provided. In exemplary embodiments, primary and secondary acoustic signals are received. A single microphone noise estimate may be generated based on the primary acoustic signal, while a dual microphone noise estimate may be generated based on the primary and secondary acoustic signals. A combined noise estimate based on the single and dual microphone noise estimates is then determined. Using the combined noise estimate, a gain mask may be generated and applied to the primary acoustic signal to generate a noise suppressed signal. Subsequently, the noise suppressed signal may be output.

System and method for providing single microphone noise suppression fallback

Systems and methods for utilizing inter-microphone level differences (ILD) to attenuate noise and enhance speech are provided. In exemplary embodiments, primary and secondary acoustic signals are received by omni-directional microphones, and converted into primary and secondary electric signals. A differential microphone array module processes the electric signals to determine a cardioid primary signal and a cardioid secondary signal. The cardioid signals are filtered through a frequency analysis module which takes the signals and mimics a cochlea implementation (i.e., cochlear domain). Energy levels of the signals are then computed, and the results are processed by an ILD module using a non-linear combination to obtain the ILD. In exemplary embodiments, the non-linear combination comprises dividing the energy level associated with the primary microphone by the energy level associated with the secondary microphone. The ILD is utilized by a noise reduction system to enhance the speech of the primary acoustic signal.

System and method for utilizing omni-directional microphones for speech enhancement

Systems and methods for adaptive intelligent noise suppression are provided. In exemplary embodiments, a primary acoustic signal is received. A speech distortion estimate is then determined based on the primary acoustic signal. The speech distortion estimate is used to derive control signals which adjust an enhancement filter. The enhancement filter is used to generate a plurality of gain masks, which may be applied to the primary acoustic signal to generate a noise suppressed signal.

System and method for adaptive intelligent noise suppression

Systems and methods for utilizing inter-microphone level differences to attenuate noise and enhance speech are provided. In exemplary embodiments, energy estimates of acoustic signals received by a primary microphone and a secondary microphone are determined in order to determine an inter-microphone level difference (ILD). This ILD in combination with a noise estimate based only on a primary microphone acoustic signal allow a filter estimate to be derived. In some embodiments, the derived filter estimate may be smoothed. The filter estimate is then applied to the acoustic signal from the primary microphone to generate a speech estimate.

System and method for utilizing inter-microphone level differences for speech enhancement

The present invention relates to a telephony device comprising a near-mouth microphone (M1) for picking up an input acoustic signal including the speaker's voice signal (S1) and an unwanted noise signal (N1,D1), a far-mouth microphone (M2) for picking up an unwanted noise signal (N2,D2) in addition to the near-end speaker's voice signal (S2), said speaker's voice signal being at a lower level than the near-mouth microphone, and an orientation sensor for measuring an orientation indication of said mobile device. The telephony device further comprises an audio processing unit comprising an adaptive beam­former (BF) coupled to the near-mouth and far-mouth microphones, including spatial filters for spatially filtering the input signals (zl,z2) delivered by the two microphones, and a spectral post-processor (SPP) for post-processing the signal delivered by the beam-former so as to separate the desired voice signal from the unwanted noise signal so as to deliver the output signal (y).

Telephony Device with Improved Noise Suppression

An acoustic echo cancellation device (1) for canceling an echo in a microphone signal (z) in response to a far-end signal (x) comprises: an adaptive filter unit (10) arranged for filtering the far-end signal (x) so as to produce an echo cancellation signal (y), - a combination unit (12) arranged for combining the echo cancellation signal (y) with the microphone signal (z) so as to produce a residual signal (r), and a post-processor unit (13) arranged for substantially removing any remaining echoes from the residual signal. In accordance with the invention, the device further comprises: - a near-end pitch estimation unit (18) arranged for estimating the pitch of any near-end signal (s) contained in the microphone signal (z) and for controlling the post-processor unit (13) in dependence of the estimated pitch such that harmonics of the estimated pitch may be substantially preserved. The control unit may further be arranged for controlling the post-processor unit (13) in dependence of the estimated pitch of the residual signal in addition to the estimated pitch of the residual signal.

Acoustic Echo Canceller

A two way sound reproduction system, such as a mobile phone system has an input (2) for an incoming far end signal (W), a loudspeaker (5), a D-to-A-converter (3) between the input (2) and the loudspeaker (5), a microphone (7), a A-to-D (8) converter after the microphone (7), an echo cancellation system (AEC) and an output (12) for an outgoing far end signal. The system comprises a pre-processor (30) between the input (2) and the D-to-A converter (3) comprising: an amplifier (32) to amplify the signal to a sufficient sound pressure level, a clipper or compressor or limiter (33) to limit the signal in the digital domain, so that the telephone system between D-to-A converter (3) and A-to-D converter (8) behaves substantially like a linear system.

Loudspeaker-microphone system with echo cancellation system and method for echo cancellation

An apparatus for acoustic beamforming comprises a beamform processor (105) for generating a beamformed signal from two audio inputs. An update processor (107) updates the beamforming filter of the beamform processor (105) if an update criterion is met. An adaptive filter (111) filters the signal from one of the signals and the difference signal between the filtered signal and the signal from the other audio input (101) is generated. An adaptation processor (115) adapts the adaptive filter (111) to minimize the difference signal. A criterion processor (109) modifies the update criterion in response to the (possibly normalized) difference signal. Specifically, the update criterion may be relaxed to improve acquisition performance if the difference signal is indicative of a strong signal outside the beam of the beamform processor ( 105).

Apparatus and method for acoustic beamforming

Disclosed is a communication system having stations mutually coupled through a communication channel, wherein at least one of the stations is provided with echo cancelling (EC) means. The EC means comprise: adaptive EC means arranged for simulating linear echo effects in said station, a subtracter coupled to the adaptive EC means and having a subtracter output, and non linear EC means coupled to the subtracter output for simulating additional echo effects in the communication station. The non linear EC means are arranged as harmonic suppressing post processor means coupled to the subtracter output for effecting non linear echo cancellation based on outputted higher harmonics. It is thus presented a low cost algorithm for full duplex acoustic echo cancellation, wherein a frequency dependent non linearity attenuation deals with non linearly distorted echo components. The system is well suited for robust hands free communication by mobile phones, video phones, conferencing phones etc.

Ivo L. D. M. Merks

Papers

Telephony Device with Improved Noise Suppression

Acoustic Echo Canceller

Loudspeaker-microphone system with echo cancellation system and method for echo cancellation

Apparatus and method for acoustic beamforming

Echo canceller having nonlinear echo suppressor for harmonics calculations