摘要:
A method for calibrating an ANC-enabled portable audio device having microphones plays continuously a calibration sound by a calibrated speaker of a test station separate from the device. For each microphone of all the microphones, a microphone calibration value is computed using a comparison of a predetermined level and a measured level of an audio signal transduced by the microphone in response to the continuously-played calibration sound. The calibration is done without using a microphone of the test station. A processing element of the device may be programmed to make the comparison and computation. The processing element also causes a speaker of the device to generate a second calibration sound, measures a second level while the computed calibration value is applied to one of microphones (e.g., error microphone), and computes a calibration value for the device speaker using a comparison of a predetermined level and the second level.
摘要:
Automatic noise-reduction (ANR) headsets include circuitry that cancels or suppress undesired noises. Recent years have seen the emergence of in-the-ear (ITE) earphones that incorporate ANR technology; however, designing them to function well usually entails many design tradeoffs, such as using larger ear nozzles that are uncomfortable to obtain desired noise reduction or that require added structures to hold the earphones to a user ear. To avoid these tradeoffs, the present inventors devised, among other things, an exemplary ITE ANR earphone that places its error measurement microphone in the ear nozzle that connects the driver front acoustic volume to a user ear canal. This placement allows use of a narrower more comfortable ear nozzle without compromising noise reduction and without requiring added holding structures. Moreover, the narrower ear nozzle also lowers the likelihood that the ANR circuitry will become unstable and produce undesirable noise.
摘要:
Provided are systems and methods for microphone signal fusion. An example method commences with receiving a first and second signal representing sounds captured, respectively, by internal and external microphones. The second signal includes at least a voice component. The first signal and the voice component are modified by at least human tissue. The first and second signals are processed to obtain noise estimates. The first signal is aligned with the second signal. The second signal and the aligned first signal are blended based on the noise estimates to generate an enhanced voice signal. The internal microphone is located inside an ear canal and sealed for isolation from acoustic signals outside the ear canal. The external microphone is located outside the ear canal. All of parts of the processing, blending and aligning of the systems and method may be performed on a subband basis in the frequency domain.
摘要:
Techniques are disclosed for position-robust multiple microphone noise estimation techniques. The position-robust noise estimation techniques can be used when receiving speech including diffuse noise sources, which is commonly encountered in noisy environments. The position-robust noise estimation techniques include detecting speech using the power level difference (PLD) and the coherence statistics (CS) between two microphone input signals. This multi-dimensional approach results in dual microphone noise estimation which is not affected by the position of the audio input device, resulting in more accurate detection of speech periods and more accurate noise estimation results. The position-robust noise estimate obtained from the techniques can then be used as part of a noise reduction system to reduce the levels of noise in noisy speech signals.
摘要:
In accordance with systems and methods of the present disclosure, a hybrid feed-forward/feedback adaptive noise cancellation system may include an alignment filter configured to correct misalignment of a reference microphone signal and an error microphone signal by generating a misalignment correction signal from a playback-corrected error signal.
摘要:
본 발명은 이어셋에 관한 것으로서, 사용자의 귀에 착용되는 이어폰부의 내부에 마련되어, 사용자의 입으로부터 이관을 거쳐 외이도를 통해 전달되는 제1소리를 수신하여 제1소리신호로 변환하는 내부 마이크; 상기 이어폰부의 외부에 마련되어, 사용자의 입으로부터 제공되는 제2소리를 수신하여 제2소리신호로 변환하는 하나 이상의 외부 마이크; 및 상기 제1소리신호를 기준으로 상기 제2소리신호 중의 노이즈를 필터링하여 제거한 제3소리신호를 생성하는 제어부를 포함하는 것을 특징으로 한다.
摘要:
A system includes a laser microphone or laser-based microphone or optical microphone. The laser microphone includes a laser transmitter to transmit an outgoing laser beam towards a human speaker. The laser transmitter acts also as a self-mix interferometry unit that receives the optical feedback signal reflected from the human speaker, and generates an optical self-mix signal by self-mixing interferometry of the laser beam and the received optical feedback signal. Instead of utilizing a single laser beam, multiple laser beams are used, by operating an array of laser transmitters, or by utilizing a laser beam splitter or a crystal to split laser beams or to diffract or scatter laser beams. Optionally, one or more laser beams may temporally scan a target area.
摘要:
A method for generating a first sound filter for suppression of snoring induced sounds comprises recording a first sound signal at a first measurement location in proximity of a source of snoring sounds, recording a second sound signal at a second measurement location in proximity of a person to be isolated from snoring sounds, determining a first snoring sound signal from the source by comparing the first sound signal with the second sound signal and setting first adaptive filters based on the first snoring sound signal. The disclosure further relates to systems for generating sound filters, and methods and systems for suppressing snoring induced sounds.
摘要:
An earpiece adapts to the acoustics characteristics and needs of its user to provide a perceptually transparent hearing protection, apart from uniform loudness reduction. An occlusion effect (OE) active noise control (ANC) system reduces the augmented perception of one's own voice while occluded. This occlusion effect active control adapts to the specific acoustic characteristics of the user to provided better control of the details occlusion effect reduction, and enhanced performances relative to fixed or one-size-fits-all solutions. An isolation effect (IE) filtering algorithm adapts itself to the user's acoustic characteristics to provide a uniform attenuation either in dB or in phons. Additionally, the device may be used as an in-ear monitor that also adapts to its user characteristics to provide in-ear quality sound.
摘要:
In one embodiment, an audio processing system reduces reverberation in an audio signal. A first beamformer generates a first, directional beampattern, and a second beamformer generates a second beampattern. A signal-processing subsystem (i) processes the first and second beampatterns to generate suppression factors corresponding to the reverberation and (ii) applies the suppression factors to one of the first and second beampatterns to reduce the reverberation in the beampattern. In one implementation, the beampatterns are crossed-beam beampatterns, and the signal- processing subsystem generates the suppression factors based on coherence estimates for the beampatterns. In another implementation, the beampatterns are disjoint beampatterns, and the signal-processing subsystem generates the suppression factors based on short-time and long-time envelope estimates for the beampatterns. Depending on the implementation, the beamformers may be co-located with differently shaped beampatterns or non-co-located with differently or equally shaped beampatterns.