Showing papers on "Microphone published in 2003"

An Industrial Strength Audio Search Algorithm.

Abstract: We have developed and commercially deployed a flexible audio search engine. The algorithm is noise and distortion resistant, computationally efficient, and massively scalable, capable of quickly identifying a short segment of music captured through a cellphone microphone in the presence of foreground voices and other dominant noise, and through voice codec compression, out of a database of over a million tracks. The algorithm uses a combinatorially hashed time-frequency constellation analysis of the audio, yielding unusual properties such as transparency, in which multiple tracks mixed together may each be identified. Furthermore, for applications such as radio monitoring, search times on the order of a few milliseconds per query are attained, even on a massive music database.

Small array microphone for acoustic echo cancellation and noise suppression

Abstract: Techniques for canceling echo and suppressing noise using an array microphone and signal processing. In one system, at least two microphones form an array microphone and provide at least two microphone input signals. Each input signal may be processed by an echo canceller unit to provide a corresponding intermediate signal having some echo removed. An echo cancellation control unit receives the intermediate signals and derives a first gain used for echo cancellation. A noise suppression control unit provides at least one control signal used for noise suppression based on background noise detected in the intermediate signals. An echo cancellation and noise suppression unit derives a second gain based on the control signal(s), cancels echo in a designated intermediate signal based on the first gain, and suppresses noise in this intermediate signal based on the second gain. The signal processing may be performed in the frequency domain.

Blind estimation of reverberation time.

Abstract: The reverberation time (RT) is an important parameter for characterizing the quality of an auditory space. Sounds in reverberant environments are subject to coloration. This affects speech intelligibility and sound localization. Many state-of-the-art audio signal processing algorithms, for example in hearing-aids and telephony, are expected to have the ability to characterize the listening environment, and turn on an appropriate processing strategy accordingly. Thus, a method for characterization of room RT based on passively received microphone signals represents an important enabling technology. Current RT estimators, such as Schroeder’s method, depend on a controlled sound source, and thus cannot produce an online, blind RT estimate. Here, a method for estimating RT without prior knowledge of sound sources or room geometry is presented. The diffusive tail of reverberation was modeled as an exponentially damped Gaussian white noise process. The time-constant of the decay, which provided a measure of the RT, was estimated using a maximum-likelihood procedure. The estimates were obtained continuously, and an order-statistics filter was used to extract the most likely RT from the accumulated estimates. The procedure was illustrated for connected speech. Results obtained for simulated and real room data are in good agreement with the real RT values.

Micromachined piezoelectric membrane acoustic device

Abstract: This paper reports on a 3 mm ×3 mm ×0.003 mm piezoelectric membrane acoustic device, which works as a microphone and a microspeaker. It has a 0.5 μm thick zinc oxide (ZnO) piezoelectric thin film on a 1.5 μm thick low-stress silicon nitride membrane, made of LPCVD. The maximum deflection in the center of membrane, using laser Doppler vibrometer (LDV), is 1 μm at 7.3 kHz with input drive 15 V0-P (zero-peak). The output sound pressure level (SPL) of microspeaker is 76.3 dB SPL at 7.3 kHz, and 83.1 dB SPL at 13.3 kHz with input drive 15 V0-P. The distance between the reference microphone and piezoelectric microspeaker is 1 cm. The sensitivity of the microphone is 0.51 mV/Pa at 7.3 kHz with noise level of 18 dB SPL.

Speech training method with alternative proper pronunciation database

Abstract: In accordance with a present invention speech training system is disclosed. It uses a microphone to receive audible sounds input by a user into a first computing device having a program with a database consisting of (i) digital representations of known audible sounds and associated alphanumeric representations of the known audible sounds, and (ii) digital representations of known audible sounds corresponding to mispronunciations resulting from known classes of mispronounced words and phrases. The method is performed by receiving the audible sounds in the form of the electrical output of the microphone. A particular audible sound to be recognized is converted into a digital representation of the audible sound. The digital representation of the particular audible sound is then compared to the digital representations of the known audible sounds to determine which of those known audible sounds is most likely to be the particular audible sound being compared to the sounds in the database. In response to a determination of error corresponding to a known type or instance of mispronunciation, the system presents an interactive training program from the computer to the user to enable the user to correct such mispronunciation.

Karaoke device with built-in microphone and microphone therefor

Abstract: A karaoke device with built-in microphone includes a main body microphone, and converts an audio signal from the microphone into audio data by an A/D converter, and writes the audio data into a ring buffer by mixing with the data already stored in the ring buffer. If an echo mode is set, a delay time constant (CD) corresponding to the echo mode is determined, and on the basis thereof, a size of the ring buffer is set. The data is read from the ring buffer, and is inputted in a sound channel. If a voice effect mode is set, a reproduction frequency constant (CF) corresponding to the voice effect mode is determined, and based thereon, an inclement value of a read pointer of the ring buffer is determined, and then, the data is read from an address indicated by the read pointer. When the read pointer reaches the delay time constant, the relevant constant is subtracted from the read pointer value. Furthermore, it becomes possible to simultaneously use a microphone of an additional microphone and the main body microphone by inserting a microphone plug of the additional microphone into a microphone jack of the karaoke device.

Design of broadband beamformers robust against gain and phase errors in the microphone array characteristics

Abstract: Fixed broadband beamformers using small-size microphone arrays are known to be highly sensitive to errors in the microphone array characteristics. The paper describes two design procedures for designing broadband beamformers with an arbitrary spatial directivity pattern, which are robust against gain and phase errors in the microphone array characteristics. The first design procedure optimizes the mean performance of the broadband beamformer and requires knowledge of the gain and the phase probability density functions, whereas the second design procedure optimizes the worst-case performance by using a minimax criterion. Simulations with a small-size microphone array show the performance improvement that can be obtained by using a robust broadband beamformer design procedure.

Robust time delay estimation exploiting redundancy among multiple microphones

Abstract: To find the position of an acoustic source in a room, typically, a set of relative delays among different microphone pairs needs to be determined. The generalized cross-correlation (GCC) method is the most popular to do so and is well explained in a landmark paper by Knapp and Carter. In this paper, the idea of cross-correlation coefficient between two random signals is generalized to the multichannel case by using the notion of spatial prediction. The multichannel spatial correlation matrix is then deduced and its properties are discussed. We then propose a new method based on the multichannel spatial correlation matrix for time delay estimation. It is shown that this new approach can take advantage of the redundancy when more than two microphones are available and this redundancy can help the estimator to better cope with noise and reverberation.

The fusion of distributed microphone arrays for sound localization

Abstract: This paper presents a general method for the integration of distributed microphone arrays for localization of a sound source. The recently proposed sound localization technique, known as SRP-PHAT, is shown to be a special case of the more general microphone array integration mechanism presented here. The proposed technique utilizes spatial likelihood functions (SLFs) produced by each microphone array and integrates them using a weighted addition of the individual SLFs. This integration strategy accounts for the different levels of access that a microphone array has to different spatial positions, resulting in an intelligent integration strategy that weighs the results of reliable microphone arrays more significantly. Experimental results using 10 2-element microphone arrays show a reduction in the sound localization error from 0.9m to 0.08m at a signal-to-noise ratio of 0 dB. The proposed technique also has the advantage of being applicable to multimodal sensor networks.

Microphone and voice activity detection (vad) configurations for use with communication systems

Abstract: Communication systems are described, including both portable handset and headset devices, which use a number of microphone configurations to receive acoustic signals of an environment. The microphone configurations include, for example, a two-microphone array including two unidirectional microphones, and a two-microphone array including one unidirectional microphone and one omnidirectional microphone. The communication systems also include Voice Activity Detection (VAD) devices to provide information of human voicing activity. Components of the communications systems receive the acoustic signals and voice activity signals and, in response, automatically generate control signals from data of the voice activity signals. Components of the communication systems use the control signals to automatically select a denoising method appropriate to data of frequency subbands of the acoustic signals. The selected denoising method is applied to the acoustic signals to generate denoised acoustic signals when the acoustic signal includes speech (101) and noise (102).

A new measurement microphone based on MEMS technology

Abstract: This paper presents a new type of measurement microphone that is based on MEMS technology. The silicon chip design and fabrication are discussed, as well as the specially developed packaging technology. The microphones are tested on a number of key parameters for measurement microphones: sensitivity, noise level, frequency response, and immunity to disturbing environmental parameters, such as temperature changes, humidity, static pressure variations, and vibration. A sensitivity of 22 mV/Pa (-33 dB re. 1 V/Pa), and a noise level of 23 dB(A) were measured. The noise level is 7 dB lower than state-of-the-art 1/4-inch measurement microphones. A good uniformity on sensitivity and frequency response has been measured. The sensitivity to temperature changes, humidity, static pressure variations and vibrations is fully comparable to the traditional measurement microphones. This paper shows that high-quality measurement microphones can be made using MEMS technology, with a superior noise performance.

Transducer arrays with an integrated sensor and methods of use

Abstract: Methods and systems for obtaining ultrasound and sensing other patient characteristics are provided from summary. One or more sensors (16) are provided in a same transducer probe (10) as an array of elements (14). For example, sensors (16) are formed on a same semiconductor substrate (26), such as a silicon substrate (26), as microelectromechanical devices or a capacitive membrane ultrasonic transducer (CMUT). As another example, a sensor (16) is provided separate from or attached to transducer materials. Possible sensors (16) include temperature, pressure, microphone, chemical and/or other sensors (16).

Combining standard and throat microphones for robust speech recognition

Abstract: We present a method to combine the standard and throat microphone signals for robust speech recognition in noisy environments. Our approach is to use the probabilistic optimum filter (POF) mapping algorithm to estimate the standard microphone clean-speech feature vectors, used by standard speech recognizers, from both microphones' noisy-speech feature vectors. A small untranscribed "stereo" database (noisy and clean simultaneous recordings) is required to train the POF mappings. In continuous-speech recognition experiments using SRI International's DECIPHER recognition system, both using artificially added noise and using recorded noisy speech, the combined-microphone approach significantly outperforms the single-microphone approach.

Multi-coil coupling system for hearing aid applications

Abstract: A hearing improvement device using a multi-coil coupling system and methods for operating such a device are disclosed. An embodiment of the present invention may use an array microphone to provide highly directional reception. The received audio signal may be filtered, amplified, and converted into a magnetic field for coupling to the telecoil in a conventional hearing aid. Multiple transmit inductors may be used to effectively couple to both in-the-ear and behind-the-ear type hearing aids, and an additional embodiment is disclosed which may be used with an earphone, for users not requiring a hearing aid.

Air- and bone-conductive integrated microphones for robust speech detection and enhancement

Abstract: We present a novel hardware device that combines a regular microphone with a bone-conductive microphone. The device looks like a regular headset and it can be plugged into any machine with a USB port. The bone-conductive microphone has an interesting property: it is insensitive to ambient noise and captures the low frequency portion of the speech signals. Thanks to the signals from the bone-conductive microphone, we are able to detect very robustly whether the speaker is talking, eliminating more than 90% of background speech. Furthermore, by combining both channels, we are able to remove background speech significantly, even when the background speaker speaks at the same time as the speaker wearing the headset.

Small array microphone for beam-forming and noise suppression

Abstract: Techniques are provided to suppress noise and interference using an array microphone and a combination of time-domain and frequency-domain signal processing. In one design, a noise suppression system includes an array microphone, at least one voice activity detector (VAD), a reference generator, a beam-former, and a multi-channel noise suppressor. The array microphone includes multiple microphones—at least one omni-directional microphone and at least one uni-directional microphone. Each microphone provides a respective received signal. The VAD provides at least one voice detection signal used to control the operation of the reference generator, beam-former, and noise suppressor. The reference generator provides a reference signal based on a first set of received signals and having desired voice signal suppressed. The beam-former provides a beam-formed signal based on a second set of received signals and having noise and interference suppressed. The noise suppressor further suppresses noise and interference in the beam-formed signal.

Acoustic model creating method, speech recognition apparatus, and vehicle having the speech recognition apparatus

Abstract: The invention provides an acoustical model creating method that obtains high recognition performance under various noise environments such as the inside of a car. The present invention can include a noise data determination unit, which receives data representing the traveling state of the vehicle, the surrounding environments of the vehicle, and the operational states of apparatuses mounted in the vehicle, and according to the data, determines which noise data of the previously classified n types of noise data corresponds to the current noise. The invention can also include a noise removal processing unit in which the n types of noise data are superposed on standard speech data to create n types of noise-superposed speech data, and then n types of acoustic models M 1 to Mn, which are created based on the n types of noise-removed speech data from which noise is removed, and noise-superposed speech from a microphone are input together with the result of the noise type determination, and then noise removal is performed on the noise-superposed speech. The invention can also include a speech recognition processing unit in which speech recognition is performed on the noise-removed speech using the acoustic model corresponding to the noise type which is determined by the noise data determination unit among the n types of acoustic models.

Applying scattering theory to robot audition system: robust sound source localization and extraction

Abstract: Robot audition by its own ears (microphones) is essential for natural human-robot communication and interface. Since a microphone is embedded in the head of a robot, the head-related transfer function (HRTF) plays an important role in sound source localization and extraction. Usually, from binaural input, the interaural phase difference (IPD) and interaural intensity difference (IID) are calculated, and then the direction is determined by using IPD and IID with HRTF. The problem of HRTF-based sound source localization is that a HRTF should be measured for each robot in an anechoic chamber, because it depends on the shape of robot's head; HRTF should be interpolated to manipulate a moving talker, because it is available only for discrete azimuth and elevation. To cope with these problems of HRTF, we proposed the auditory epipolar geometry as a continuous function of IPD and IID to dispense with HRTF and have developed a real-time multiple-talker tracking system. This auditory epipolar geometry, however, does not give a good approximation to IID of all range and IPD of peripheral areas. In this paper, the scattering theory in physics is employed to take into consideration the diffraction of sounds around robot's head for better approximation of IID and IPD. The resulting system shows that it is efficient for localization and extraction of sound at higher frequency and from side directions.

An overview of microflown technologies

Abstract: The Microflown is an acoustic sensor measuring particle velocity instead of sound pressure, which is usually measured by conventional microphones. Since its recent invention it is mostly used for measurement purposes (1D and 3D-sound intensity measurement and acoustic impedance). The Microflown is also used for measuring DC-flows, that can be considered as particle velocity with a frequency of 0Hz. Furthermore the Microflown is used in the professional audio as a low frequency add on microphone for pressure gradient microphones (figure of eight; directional microphones). Due to its small dimensions and silicon based production method the Microflown is very suitable for mobile applications like mobile telephones or smartcards. Nowadays sound-energy determination, array applications and three-dimensional impulse response are under investigation. Although the Microflown was invented only some years ago, the device is already commercially available.

Non-audible murmur recognition input interface using stethoscopic microphone attached to the skin

Abstract: We propose a new style of practical input interface for the recognition of non-audible murmur (NAM), i.e., for the recognition of inaudible speech produced without vibration of the vocal folds. We have developed a microphone attachment, which adheres to the skin, applying the principle of a medical stethoscope, found the ideal position for sampling flesh-conducted NAM sound vibration and retrained an acoustic model with NAM samples. Then, using the Julius Japanese Dictation Toolkit, we tested the possibilities for practical use of this method in place of an external microphone for analyzing air-conducted voice sound.

Noise Characteristics of Aircraft High Lift Systems

Abstract: The characteristics of far-field noise from aircraft high lift systems are discussed. Analyses are made of the data from an airframe noise test conducted in the 40 by 80 ft wind tunnel at NASA Ames Research Center, using a 4.7% DC-10 aircraft model. Discussions are given on both far-field free microphone data and measurements from a phased microphone array. Major trends are revealed and discussed from the free microphone data, which include far-field frequency characteristics, dependence of the acoustic radiation on flow Mach numbers, effects of flap and/or slat deployment, and far-field directivity. Data from the phased microphone array are used to identify locations of major noise sources. Four subregions on the wing are identified as important source locations, namely, the leading-edge slat region, the inboard and outboard side edges of the outboard flap, and the inboard flap region close to the hub of the wing. The source strength distributions in these subregions are integrated to reveal dependencies of various noise sources on flow conditions and high lift system configurations. The effects of flap side edge fences on far-field noise are also discussed, which shows a reduction of a few decibels in flap-related noise.

Baby or child monitor incorporating mobile telephone

Abstract: A child or baby monitor comprising a transmitter terminal which monitors an activity of the baby or child, and a receiver for a parent which receives the transmitted signals. The transmitter and receiver communicate via a public radio communications network. The transmitter and/or the receiver may comprise a mobile telephone. The transmitter transmits signals in response to the monitored activity and is only activated when the detected activity exceeds a predefined limit. The transmitter terminal may have a microphone for detecting audio activity or a video camera for detecting visual activity, e.g. movement. The transmitter may also include a buffer which records the detected audio or visual data. The signal transmitted by the transmitter may be in the form of an SMS message, text message or an email, and may also include the data which is stored in the buffer.

Beamforming Algorithm for Distributed Source Localization and Its Application to Jet Noise

Abstract: A new algorithm for estimating noncompact, distributed sources by means of phased array microphone measurements is presented and experimentally implemented to determine the noise source distribution in a subscale jet flow. Conventional beamforming techniques, developed for spatially well-separated point sources, can lead to significant errors when applied to reconstruct continuous source distributions such as jet noise. A new beamforming approach is developed for estimating such continuous source distributions. The objective is to recover the average source strength over a small region around each focus position as opposed to seeking the exact source strength at each spatial location as in conventional approaches. This strategy overcomes the drawbacks of conventional methods and yields a beamformer with uniform spatial resolution and accuracy over a large frequency range. The measurement technique is applied to the localization of broadband noise sources in a high-subsonic, heated, turbulent jet flow and shows good comparisons with prior measurements using other techniques.

Microphone array processing for robust speech recognition

Abstract: Speech recognition performance degrades significantly in distant-talking environments, where the speech signals can be severely distorted by additive noise and reverberation. In such environments, the use of microphone arrays has been proposed as a means of improving the quality of captured speech signals. Currently, microphone-array-based speech recognition is performed in two independent stages: array processing and then recognition. Array processing algorithms designed for signal enhancement are applied in order to reduce the distortion in the speech waveform prior to feature extraction and recognition. This approach assumes that improving the quality of the speech waveform will necessarily result in improved recognition performance. However, speech recognition systems are statistical pattern classifiers that process features derived from the speech waveform, not the waveform itself. An array processing algorithm can therefore only be expected to improve recognition if it maximizes or at least increases the likelihood of the correct hypothesis, relative to other competing hypotheses. In this thesis a new approach to microphone-array processing is proposed in which the goal of the array processing is not to generate an enhanced output waveform but rather to generate a sequence of features which maximizes the likelihood of the correct hypothesis. In this approach, called Likelihood Maximizing Beamforming (LIMABEAM), information from the speech recognition system itself is used to optimize a filter-and-sum beamformer. Using LIMABEAM, significant improvements in recognition accuracy over conventional array processing approaches are obtained in moderately reverberant environments over a wide range of signal-to-noise ratios. However, only limited improvements are obtained in environments with more severe reverberation. To address this issue, a subband filtering approach to LIMABEAM is proposed, called Subband-Likelihood Maximizing Beamforming (S-LIMABEAM). S-LIMABEAM employs a new subband filter-and-sum architecture which explicitly considers how the features used for recognition are computed. This enables S-LIMABEAM to achieve dramatically improved performance over the original LIMABEAM algorithm in highly reverberant environments. Because the algorithms in this thesis are data-driven, they do not require a priori knowledge of the room impulse response, nor any particular number of microphones or array geometry. To demonstrate this, LIMABEAM and S-LIMABEAM are evaluated using multiple array configurations and environments including an array-equipped personal digital assistant (PDA) and a meeting room with a few tabletop microphones. In all cases, the proposed algorithms significantly outperform conventional array processing approaches.

Programmable auditory prosthesis with trainable automatic adaptation to acoustic conditions

Abstract: An auditory prosthesis (30) comprising a microphone (27) for receiving the sound and producing a microphone signal corresponding to the received sound, an output device for providing audio signals in a form receivable by a user of the prosthesis (30), a sound processing unit (33) operable to receive the microphone signal and carry out a processing operation on the microphone signal to produce an output signal in a form suitable to operate the output device, wherein the sound processing unit (33) is operable in a first mode in which the processing operation comprises at least one variable processing factor which is adjustable by a user to a setting which causes the output signal of the sound processing unit (33) to be adjusted according to the preference of the user for the characteristics of the current acoustic environment.

Blind dereverberation of single channel speech signal based on harmonic structure

Abstract: The paper presents a new method for dereverberation of speech signals with a single microphone. For applications such as speech recognition, reverberant speech causes serious problems when a distant microphone is used in recording. This is especially severe when the reverberation time exceeds 0.5 s. We propose a method which uses the fundamental frequency (F/sub 0/) of the target speech as the primary feature for dereverberation. This method initially estimates F/sub 0/ and the harmonic structure of the speech signal and then obtains a dereverberation operator. This operator transforms the reverberant signal to its direct signal based on an inverse filtering operation. Dereverberation is achieved without prior knowledge of either the room acoustics or the target speech. Experimental results show that the dereverberation operator estimated from 5240 Japanese word utterances could effectively reduce the reverberation when the reverberation time is longer than 0.1 s.

Method and apparatus for third heart sound detection

Abstract: A cardiac rhythm management system includes a heart sound detector providing for detection of the third heart sounds (S3). An implantable sensor such as an accelerometer or a microphone senses an acoustic signal indicative heart sounds including the second heart sounds (S2) and S3. The heart sound detector detects occurrences of S2 and starts S3 detection windows each after a predetermined delay after a detected occurrence of S2. The occurrences of S3 are then detected from the acoustic signal within the S3 detection windows.

System and method for selectively coupling hearing aids to electromagnetic signals

Abstract: Systems, devices and methods are provided for selectively coupling hearing aids to electromagnetic fields. One aspect relates to a hearing aid device. In various embodiments, the hearing aid device includes an induction signal receiver for receiving induction signals, a microphone system for receiving acoustic signals, a hearing aid receiver, and a signal processing circuit. The signal processing circuit includes a proximity sensor for detecting an induction source. The signal processing circuit presents a first signal to the hearing aid receiver that is representative of the acoustic signals. When the induction source is detected, the signal processing circuit presents a second signal to the hearing aid receiver that is representative of the induction signals and transmits a third signal representative of the induction signals from the hearing aid device to a second hearing aid device. Other aspects are provided herein.

Fault detection system having audio analysis and method of using the same

Abstract: A fault detection system (20) for determining whether a fault exists with a rotating element (30a, 30b, 30c) of a vehicle. The system includes a transducer (22), a diagnosis sampler (24), and a controller (26). The transducer (22) may be a microphone located in the vehicle for converting sounds to an electrical signal. The electrical signal includes a noise component generated from the rotating element. The diagnosis sampler (24) is connected to the transducer and provides a sample of the electrical signal from the transducer (22) to the controller (26). The controller (26) has functional aspects such as an envelope detect (44), a spectrum analysis (48a, 48b, 48c), and a fault detect (50a, 50b, 50c). There is also a method of detecting a fault associated with a rotating element (30a, 30b, 30c) in a vehicle using the above-described system.