9689 results about "Noise" patented technology

A power tool having one or more sound damping members which reduce sound and / or vibration from one or more parts of a power tool. The sound damping member can reduce sound and / or vibration from static or dynamic parts of a power tool. The sound damping member can reduce noise and / or vibration from one or more rotating or moving parts of a power tool and its housing or internal structure. Methods, means, controls, systems and practices for reducing or eliminating undesired sound from a power tool are disclosed.

Ear protecting device with a sealing section for acoustically sealing the meatus of a human, includes a sound generator with a sound outlet for being directed toward the user meatus; an inner microphone with a sound inlet from the meatus, arranged for measuring the resulting sound field in the meatus; connected to an electronics unit including a sound analyser coupled to the inner microphone, for analyzing sound characteristics of the resulting sound field in the meatus, producing analyzed sound characteristics; storing part in the electronics unit for storing measured predetermined sound characteristics of a properly functioning ear protecting device; a comparing part in the electronics unit for comparing the inner microphone analyzed sound characteristics with the stored measured predetermined sound characteristics; indicating part coupled to the comparing part for being activated if the analyzed sound characteristics differ significantly from the predetermined sound characteristics.

Spatial noise suppression for audio signals involves generating a ratio of powers of difference and sum signals of audio signals from two microphones and then performing noise suppression processing, e.g., on the sum signal where the suppression is limited based on the power ratio. In certain embodiments, at least one of the signal powers is filtered (e.g., the sum signal power is equalized) prior to generating the power ratio. In a subband implementation, sum and difference signal powers and corresponding the power ratio are generated for different audio signal subbands, and the noise suppression processing is performed independently for each different subband based on the corresponding subband power ratio, where the amount of suppression is derived independently for each subband from the corresponding subband power ratio. In an adaptive filtering implementation, at least one of the audio signals can be adaptively filtered to allow for array self-calibration and modal-angle variability.

A digital wavetable audio synthesizer is described. The synthesizer can generate up to 32 high-quality audio digital signals or voices, including delay-based effects, at either a 44.1 KHz sample rate or at sample rates compatible with a prior art wavetable synthesizer. The synthesizer includes an address generator which has several modes of addressing wavetable data. The address generator's addressing rate controls the pitch of the synthesizer's output signal. The synthesizer performs a 10-bit interpolation, using the wavetable data addressed by the address generator, to interpolate additional data samples. When the address generator loops through a block of data, the signal path interpolates between the data at the end and start addresses of the block of data to prevent discontinuities in the generated signal. A synthesizer volume generator, which has several modes of controlling the volume, adds envelope, right offset, left offset, and effects volume to the data. The data can be placed in one of sixteen fixed stereo pan positions, or left and right offsets can be programmed to place the data anywhere in the stereo field. The left and right offset values can also be programmed to control the overall volume. Zipper noise is prevented by controlling the volume increment. A synthesizer LFO generator can add LFO variation to: (i) the wavetable data addressing rate, for creating a vibrato effect; and (ii) a voice's volume, for creating a tremolo effect. Generated data to be output from the synthesizer is stored in left and right accumulators. However, when creating delay-based effects, data is stored in one of several effects accumulators. This data is then written to a wavetable. The difference between the wavetable write and read addresses for this data provides a delay for echo and reverb effects. LFO variations added to the read address create chorus and flange effects. The volume of the delay-based effects data can be attenuated to provide volume decay for an echo effect. After the delay-based effects processing, the data can be provided with left and right offset volume components which determine how much of the effect is heard and its stereo position. The data is then stored in the left and right accumulators.

A personal audio device, such as a wireless telephone, includes an adaptive noise canceling (ANC) circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone is also provided proximate the speaker to measure the ambient sounds and transducer output near the transducer, thus providing an indication of the effectiveness of the noise canceling. A processing circuit uses the reference and / or error microphone, optionally along with a microphone provided for capturing near-end speech, to determine whether the ANC circuit is incorrectly adapting or may incorrectly adapt to the instant acoustic environment and / or whether the anti-noise signal may be incorrect and / or disruptive and then take action in the processing circuit to prevent or remedy such conditions.

Audio frames are classified as either speech, non-transient background noise, or transient noise events. Probabilities of speech or transient noise event, or other metrics may be calculated to indicate confidence in classification. Frames classified as speech or noise events are not used in updating models (e.g., spectral subtraction noise estimates, silence model, background energy estimates, signal-to-noise ratio) of non-transient background noise. Frame classification affects acceptance / rejection of recognition hypothesis. Classifications and other audio related information may be determined by circuitry in a headset, and sent (e.g., wirelessly) to a separate processor-based recognition device.

Apparatus and methods for including a code having at least one code frequency component in an audio signal are provided. The abilities of various frequency components in the audio signal to mask the code frequency component to human hearing are evaluated and based on these evaluations an amplitude is assigned to the code frequency component. Methods and apparatus for detecting a code in an encoded audio signal are also provided. A code frequency component in the encoded audio signal is detected based on an expected code amplitude or on a noise amplitude within a range of audio frequencies including the frequency of the code component.

Methods and apparatus for encoding and decoding information in broadcast or recorded segment signals are described. In certain embodiments, an audience monitoring system encodes identification information in the audio signal portion of a broadcast or recorded segment using spread spectrum encoding. A personal monitoring device receives an acoustically reproduced version of the broadcast or recorded signal via a microphone, decodes the identification information from the audio signal portion despite significant ambient noise, and stores this information, automatically providing a diary for the audience member which is later uploaded to a centralized facility. A separate monitoring device decodes additional information from the broadcast signal, which is matched with the audience diary information at the central facility. This monitor may simultaneously send data to the centralized facility using a dial-up telephone line, and receive data from the centralized facility through a signal encoded using a spread spectrum technique and modulated with a broadcast signal from a third-party.

A source-controlled Variable bit-rate Multi-mode WideBand (VMR-WB) codec, having a mode of operation that is interoperable with the Adaptive Multi-Rate wideband (AMR-WB) codec, the codec comprising: at least one Interoperable full-rate (I-FR) mode, having a first bit allocation structure based on one of a AMR-WB codec coding types; and at least one comfort noise generator (CNG) coding type for encoding inactive speech frame having a second bit allocation structure based on AMR-WB SID_UPDATE coding type. Methods for i) digitally encoding a sound using a source-controlled Variable bit rate multi-mode wideband (VMR-WB) codec for interoperation with an adaptative multi-rate wideband (AMR-WB) codec, ii) translating a Variable bit rate multi-mode wideband (VMR-WB) codecsignal frame into an Adaptive Multi-Rate wideband (AMR-WB) signal frame, iii) translating an Adaptive Multi-Rate wideband (AMR-WB) signal frame into a Variable bit rate multi-mode wideband (VMR-WB) signal frame, and iv) translating an Adaptive Multi-Rate wideband (AMR-WB) signal frame into a Variable bit rate multi-mode wideband (VMR-WB) signal frame are also provided.

In one embodiment, a directional microphone array having (at least) two microphones generates forward and backward cardioid signals from two (e.g., omnidirectional) microphone signals. An adaptation factor is applied to the backward cardioid signal, and the resulting adjusted backward cardioid signal is subtracted from the forward cardioid signal to generate a (first-order) output audio signal corresponding to a beampattern having no nulls for negative values of the adaptation factor. After low-pass filtering, spatial noise suppression can be applied to the output audio signal. Microphone arrays having one (or more) additional microphones can be designed to generate second- (or higher-) order output audio signals.

A highly versatile interface that is capable of digital and audio signal coupling is provided. The interface comprises contacts (122, 124, 216, 218) that are used to couple both audio and digital signals, and separate contacts (126, 220) that are used initiate and negotiate signaling mode transitions. Transitions can be effected without creating glitches, e.g., audible noise, in audio signals that are being coupled through the interface.

A desired acoustic signal is extracted from a noisy environment by generating a signal representative of the desired signal with a processor. The processor receives aural signals from two sensors each at a different location. The two inputs to the processor are converted from analog to digital format and then submitted to a discrete Fourier transform process to generate discrete spectral signal representations. The spectral signals are delayed by a number of time intervals in a dual delay line to provide a number of intermediate signals, each corresponding to a different spatial location relative to the two sensors. Locations of the noise source and the desired source are determined and the spectral content of the desired signal is determined from the intermediate signal corresponding to the noise source locations. Inverse transformation of the selected intermediate signal followed by digital to analog conversion provides an output signal representative of the desired signal. Techniques to localize multiple acoustic sources are also disclosed. Further, a technique to enhance noise reduction from multiple sources based on two-sensor reception is described.

Methods and apparatus for echo cancellation in a system having a speaker and a microphone are disclosed. The speaker receives a speaker signal x(t). The microphone receives a microphone signal d(t) containing a local signal s(t) and an echo signal x1(t) that is dependent on the speaker signal x(t). The microphone signal d(t) is filtered in parallel with first and second adaptive filters having complementary echo cancellation properties relative to each other. A minimum echo output e3(t) is determined from an output e1(t) of the first adaptive filter and an output e2(t) of the second adaptive filter. The minimum echo output has a smaller energy and less correlation to the speaker signal x(t). A microphone output is then generated using the minimum echo output e3(t).

An electronic pillow including a pillow unit encasing at least one error microphone and at least one loudspeaker in electrical connection with a controller unit, the pillow unit also including a power source, and a reference sensing unit including at least one reference microphone in electrical connection with the controller unit, the controller unit including an algorithm for controlling interactions between the error microphone, loudspeaker, and reference microphone. A method of abating unwanted noise, by detecting an unwanted noise with a reference microphone, analyzing the unwanted noise, producing an anti-noise corresponding to the unwanted noise in a pillow, and abating the unwanted noise. Methods of hands-free communication, recording and monitoring sleep disorders, providing real-time response to emergencies, and playing audio sounds.