Abstract: The present invention provides a voice input apparatus which can extract an input voice without interrupting a generated background sound. When a talk switch 50 is pushed down, a LMS algorithm processing part 26 stops an update operation of a filter coefficient of an adaptive filter 24 and stores filter coefficient W1 at this moment in a portion for storing filter coefficients 30. A portion for calculating filter coefficients 44 calculates a filter coefficient W2 of a filter 42 to simulate a sound-transmittal line of an audio sound which is produced from a speaker 102 and is input into microphones 10, 12 by means of these stored filter coefficient W1 and various transmittal characteristics CS1, CS2 and the like. By extracting an output signal Y4 of the filter 42 from an output signal Y3 of a portion for eliminating surrounding noise 20 by means of a computing part 46, audio sound components included with an operating voice collected by the microphones 10, 12 are removed.

Abstract: A portable telephone comprises a case fitted with an antenna and pierced with communications openings in one face associated with a speech microphone and with an earpiece. The telephone also has a directional anti-noise microphone connected to the earpiece via an active anti-noise circuit and associated with a first opening that is separate from the communications openings of the speech microphone and of the earpiece.

Abstract: A method is presented for eliminating an unwanted signal (e.g., background music, interference, etc.) from a mixture of a desired signal and the unwanted signal via time-frequency masking. Given a mixture of the desired signal and the unwanted signal, the goal of the present invention is to eliminate or at least reduce the effects of the unwanted signal to obtain an estimate of the desired signal.

Abstract: A baby monitor system with a parent unit communicable with a baby unit having a soothing unit is provided. In an embodiment for the baby monitor system of the present invention, the baby unit is responsive to receiving a signal representative of an audible sound transmitted from the parent unit and the parent unit is responsive to receiving a signal representative of an audible sound transmitted from the baby unit. In an embodiment for a method of the present invention, a method for monitoring a baby is provided. The method includes the steps of receiving a signal representative of an audible sound at a baby unit from a parent unit, receiving a signal representative of an audible sound at a parent unit from a baby unit, and actuating a soothing unit included in the baby unit.

Abstract: An active noise suppression system for use in noisy environments includes a dual microphone noise suppression system in which the echo between the two microphones is substantially canceled or suppressed. Noise is cancelled by the use of first and second line echo cancellers, which model the delay and transmission characteristics of the acoustic path between the two microphones. In a first embodiment, a noise suppression system acts as an ear protector, canceling substantially all or most of the noise striking the dual microphones of the ear set. In a second embodiment, a noise suppression system in accordance with the present invention acts a noise suppression communication system, suppressing background noise while allowing speech signals to be heard by the wearer.

Abstract: A method of processing an audio signal in order to overcome background noise extraneous to the audio signal. The method includes comparing a measure of the audio signal in a frequency component of the audio signal to a measure of the noise in a frequency component of the background noise, determining a gain responsive to the comparison and amplifying at least a portion of the audio signal, including at least one frequency segment not included in the frequency component of the audio signal, by the determined gain.

Abstract: A system for time delay estimation in a discrete time processing system includes a cross correlator that performs cross correlation on a first signal and a second signal, and provides a cross correlated output signals indicative thereof. A lag smoother receives the cross correlated output signals, and provides lag smoothed output signals indicative thereof. A select logic module selects a pre-defined number of signal values from a respective set indicative of the lag smoothed output signals to compute the time delay estimation associated with the first and second signals.

Abstract: A built-in microphone device for use in an apparatus having a mechanism section generating internal noise inside a housing of the apparatus includes a main microphone for picking up an external sound; a noise reference microphone for picking up the internal noise; an adaptive filter member for generating a control audio signal based on an output signal from the noise reference microphone using a filter coefficient; a signal subtraction section for subtracting the control audio signal generated by the adaptive filter member from an output signal from the main microphone to generate a subtraction result; and a filter coefficient update control section for receiving an operation signal generated at the time of an operation of the mechanism section, and in response to the operation signal, updating the filter coefficient of the adaptive filter member based on the subtraction result generated by the signal subtraction section and an output signal from the noise reference microphone.

Abstract: An adaptive signal processing system for improving a quality of a signal. The system includes an analysis filterbank for transforming a primary information signal in time domain into oversampled sub-band primary signals in frequency domain and an analysis filterbank for transforming a reference signal in time domain into oversampled sub-band reference signals. Sub-band processing circuits process the signals output from the filterbanks to improve a quality of an output signal. A synthesis filterbank can combine the outputs of the sub-band processing circuits to generate the output signal.

Abstract: An apparatus for discriminating between signals produced by a desired acoustic source and an undesired acoustic source includes: a first microphone disposed at a first distance from the desired acoustic source; at least a second microphone disposed at a second distance from the desired acoustic source; a proximity estimation block configured to utilize signals form the first microphone and second microphone to produce a signal representing an estimate of the proximity of the desired acoustic source; and a variable gain block configured to adjust the gain of an output signal from at least one of the microphones by utilizing the estimate of the proximity of the desired acoustic source.

Abstract: An input signal is applied to a notch filter having a transfer function that is the inverse of the expected noise signal. The filtered signal is coupled to a first amplifier and the input signal is coupled to a second amplifier. The outputs of the amplifiers are summed. The gains of the amplifiers are oppositely adjusted in response to the magnitude of the input signal. At low amplitude, the filtered signal is amplified more than the unfiltered signal. At high amplitude, the unfiltered signal is amplified more than the filtered signal.

Abstract: A method for processing a sound signal is proposed, which is applied to a signal processing system including a signal receiving device, a signal recognizing device and a signal processing device. The signal recognizing device is used to distinguish the sound signal received from the signal receiving device into a sound source signal and background interference, so as to eliminate the background interference and transfer the sound source signal to the signal processing device. In this case, the signal processing device merely processes and amplifies the sound source signal, and thus no filtering circuit is required for the elimination of the background interference.

Abstract: A device and a method is presented in which an adjustment to the noise conditions is made for the purpose of controlling the volume and other variables of a desired signal offered in a monitored space, in the course of which, for the purpose of adjustment, a monitoring signal occurring at the monitoring point is picked up and split into a desired-signal component and a noise-signal component. These two components then become the basis for the adjustment.

Abstract: Methods and devices for reducing undesired signals in a communication environment. At least two distinct composite signals (X1 and X2) are transmitted from a first communication environment (260). A noise coefficient of the first communication environment (260) based on the at least two distinct composite signals (X1 and X2) is calculated. At least two noise canceling signals based on the noise coefficient are calculated. The at least two noise canceling signals are added to an incoming signal (Y3) from a second communication environment (280) to produce at least two combined signals. The at least two combined signal are transmitted into the first communication environment (260).

Abstract: A delay and attenuation compensation-based BSS technique for source selection and noise reduction is provided for speech recognition applications. The undesired signals are attenuated in every situation and from every direction as much as possible. For this purpose, the direction of the input signals is determined. Depending on the direction of the undesired signals, the internal delay between the microphones is calculated. This internal delay is then used in subtracting the microphone signals from each other. Therefore, the invention gives not only a good directionality on average, but the best directionality for every listening situation. Furthermore, the channel selection problem is solved in conjunction with the BSS technique.

Abstract: The present invention relates to a method with which speech is captured in a noisy environment with as high a speech quality as possible. To this end, a compact array of, for example, two single microphones is combined to form one system through signal processing methods consisting of adaptive beam formation and spectral subtraction. Through the combination with a spectral subtraction, the reference signal of the beam former is freed from speech signal components to the extent that a reference signal of the interference is formed and the beam former produces high gains.

Abstract: A signal filter includes a first adaptive filter receiving a desired signal containing a noise component and a first noise correlated reference signal, and outputting a first filtered output signal. A second adaptive filter, receives the desired signal containing the noise component and a second noise correlated reference signal and outputs a second filtered output signal. Signal filter parameters of the second adaptive filter are controlled based on the first filtered output signal and the second filtered output signal.

Abstract: A baby monitor system incorporated into a receiver of the baby monitor circuitry that can remotely control an electric or electronic device when the output sound level of the receiver reaches a reference level. The control can be by infrared, radio frequency or hardwire transmission of a volume control command to the electric or electronic device. By incorporating this control system into the baby monitor, a parent is ensured to hear sounds of a baby.

Abstract: For the purpose of spatial reproduction of an audio signal, the latter must be projected onto the positions of the existing loudspeakers. It is desirable in this case not to have to be fixed on a specific loudspeaker configuration for transmitting the audio signal. However, a problem here is that a multiplicity of possible combinations exists. In the method according to the invention, the sound sources (3) are interpreted as acoustic objects for the purpose of projecting them onto an arbitrary loudspeaker configuration (2). Here, an acoustic object consists in that in addition to the audio signal a sound source is assigned an item of spatial information which specifies a virtual, spatial position of the sound source. In order to reproduce an acoustic object, the spatial information of the sound source and the actual position of a loudspeaker are used to calculate the virtual distance from the sound source via the loudspeaker to the hearer (1).

Abstract: Methods and apparatus for automatic gain and time delay computations between multiple sensors. In exemplary embodiments, one sensor is treated as a reference sensor with respect to a measured signal quantity which is assumed to be a narrowband process. The relative gain and relative time delay in the additional sensors with respect to the same measured quantity are then automatically calculated based on an adaptive filtering algorithm. Advantageously, the disclosed embodiments are implemented using standard digital signal processing components.

Abstract: A pickup section 11 transduces sounds to audio signals. A zoom control section 12 outputs a zoom position signal corresponding to a zoom position. A directivity control section 13 alters the directivity characteristics under telescopic operation so as to mainly pickup sounds coming from a frontal direction with an enhancement which is in accordance with the zoom position signal, thereby outputting an R channel audio signal and an L channel audio signal. In accordance with the zoom position signal, a noise suppression section 14 applies a greater degree of suppression to the background noise contained in the respective channel audio signals under telescopic operation than under wide-angle operation. As a result, a target sound from a remote location can be picked up with a sufficient enhancement in accordance with the zoom position under telescopic operation.

Abstract: A process for determining an estimated value for the noise level n of a background noise superimposed on an acoustic useful signal is characterised in that the estimated value n(x) for a sampled input signal x(k) is defined as a value n1(x) which is determined by means of the minimum value of the quantity of all the successive maximum values of the input signal x(k) in each case found within a short time interval ts≧1 ms; that the value n1(x) is adopted as estimated value n(x) for the current noise level n when the dynamic variations of the input signal x(k) undershoot a threshold value &egr;; and that otherwise the estimated value determined in the preceding step is adopted unchanged as new estimated value n(x). In this way it is possible to achieve an extremely exact determination of the current noise level with very fast adaptation times which are considerably shorter than in known processes, with the need for only a relatively small computation outlay.

Abstract: An improved noise canceling microphone is provided including robust design features and advanced noise control and speech discrimination convergence characteristics Two adaptive controllers are used to ensure robust performance in quickly changing acoustic environments ensuring an acceptable minimum performance characteristic. Additionally, a new real-time spectral estimation procedure is applied to a noise canceling communications microphone platform that permits continued and optimal adaptation of non-voice bandwidth frequencies during speech transients.

Abstract: A method and apparatus for the real-time creation of an output audio signal from an input signal with an unwanted or noise portion. The system detects the unwanted portion of the input signal by utilizing an adaptive detection filter and reduces the unwanted portion of the input signal. The reduction of the unwanted portion is performed by compression of the unwanted signal, subtraction of the unwanted portion of the signal, or eliminating the output signal until the unwanted portion is no longer detected. The system is specifically designed to find a high frequency and high amplitude sound such as a sibilant.

Abstract: A method and system are provided for acoustic noise removal from human speech, wherein noise is removed without respect to noise type, amplitude, or orientation. The system includes microphones and a voice activity detection (VAD) data stream coupled among a processor. The microphones receive acoustic signals and the VAD produces a signal including a binary one when speech (voiced and unvoiced) is occurring and a binary zero in the absence of speech. The processor includes denoising algorithms that generate transfer functions. The transfer functions include a transfer functions generated in response to a determination that voicing information is absent from the received acoustic signal during a specified time period. The transfer functions also include transfer functions generated in response to a determination that voicing information is present in the acoustic signal during a specified time period. At least one denoised acoustic data stream is generated using the transfer functions.

Abstract: A noise suppress processing apparatus has a speech input section for detecting speech uttered by the speaker at different positions, an analyzer section for obtaining frequency components in units of channels by frequency-analyzing speech signals in units of speech detecting positions, a first beam former processor section for obtaining target speech components by suppressing noise in the speaker direction by filtering the frequency components in units of channels using filter coefficients, which are calculated to decrease the sensitivity levels in directions other than a desired direction, a second beam former processor section for obtaining noise components by suppressing the speech of the speaker by filtering the frequency components for the plural channels obtained by the analyzer section to set low sensitivity levels in directions other than a desired direction, an estimating section for estimating the noise direction from the filter coefficients of the first beam former processor section, and estimating

Abstract: A dual-processing interference cancelling system and method for processing a broadband input in a computationally efficient manner. Dual processing divides the input into higher and lower frequency bands and applies adaptive filter processing to the lower frequency band while applying non-adaptive filter processing to the higher frequency band. Various embodiments are shown including those based on sub-bands, broadband processing with band-limited adaptation, and broadband processing with an external main-channel generator.

Abstract: An adaptive noise rejection method and apparatus. An information signal having an information component and a repetitive noise component having a varying characteristic frequency is delayed the information signal by a reference delay time that is equal that is equal to a predetermined number of periods of the characteristic frequency to form a delayed information signal. The delayed information signal is subtracted from the information signal to form a processed information signal in which the information component is substantial and the noise component is negligible. The amplitude of the noise component of the processed information signal is constantly monitored. The delay of the information signal is varied as the characteristic frequency of the noise component changes so as to maintain the amplitude of the noise component of the processed information signal at a predetermined level.

Abstract: A signal-to-noise power ratio estimation circuit 22 detects an error signal power of an error signal output from a subtracter 13 and a pseudo noise signal power of a pseudo signal output from an adaptive filter 12 and, based on the detected signal powers, outputs an estimated value SNR1 of a signal-to-noise power ratio. An SNR average circuit 20 calculates an average value SNR4 of the SNR1. A step size output circuit 21 controls the step size of the adaptive filter 12 according to the SNR4 value.

Abstract: A remotely installed transmitter capable of clearly picking up only the voice of a person who is talking despite being installed away from the speaker's mouth and without causing feedback despite increased sensitivity, and a hands-free two-way voice terminal device using such a transmitter.

Abstract: A noise canceler of the present invention is of the type including an adaptive filter for generating a pseudo noise signal, subtracting the pseudo noise signal from a received signal to thereby output an error signal, and sequentially correcting the filter coefficient of the filter in accordance with the error signal. A second adaptive filter produces a second pseudo noise signal and a second error signal. A first and a second power mean circuit each calculates the signal power of the respective signal. A divider performs division with the resulting two kinds of signal power, so that a signal-to-noise power ratio is estimated. A comparator compares the estimated signal-to-noise power ratio and a delayed version of the same and outputs greater one of them as an extended signal-to-noise power ratio.

Abstract: Signal transmission system includes a processor (SEPAR) for isolating an estimate (IL) for at least one wanted signal (XL) contained in at least one mixed signal (Ea). At least one sensor (Ma) detects the mixed signal which includes at least the wanted signal (XL) and at least two correlated interference signals (Pa, Pb) generated in response respectively to two correlated electric signals (CRa, CRb). The processor (SEPAR) receives on the input the detected mixed signal (Ea) and the two correlated electric signals (CRa, CRb). By decorrelating the estimate (IL) relative respectively to the correlated electric signals (CRa, CRb), the processing means extracts the estimate (IL) of the wanted.

Abstract: In recording for example classical music played by an orchestra (O), a stereo pair (1) is used. To enhance say a quiet instrument such as a flute (F) a spot microphone (2) is used close to the flute. However, the different air-path lengths for the flute to the stereo pair (1) and spot microphone (2) creates undesired effects. The signal from the spot microphone is delayed (4), the delay being automatically controlled by an adaptive filter (5). The filter (5) correlates the spot microphone signal with the signal from the stereo pair to establish the delay time. An alternative arrangement uses an adaptive filter trained by the signal from the spot microphone to extract the flute signal from the stereo pair. In other arrangements, similar techniques are used to cancel undesired noise.

Abstract: A dual-processing interference cancelling system and method for processing a broadband input in a computationally efficient manner. Dual processing divides the input into higher and lower frequency bands and applies adaptive filter processing to the lower frequency band while applying non-adaptive filter processing to the higher frequency band. Various embodiments are shown including those based on sub-bands, broadband processing with band-limited adaptation, and broadband processing with an external main-channel generator.