Showing papers on "Noise published in 2014"

STFT phase reconstruction in voiced speech for an improved single-channel speech enhancement

Abstract: The enhancement of speech which is corrupted by noise is commonly performed in the short-time discrete Fourier transform domain. In case only a single microphone signal is available, typically only the spectral amplitude is modified. However, it has recently been shown that an improved spectral phase can as well be utilized for speech enhancement, e.g., for phase-sensitive amplitude estimation. In this paper, we therefore present a method to reconstruct the spectral phase of voiced speech from only the fundamental frequency and the noisy observation. The importance of the spectral phase is highlighted and we elaborate on the reason why noise reduction can be achieved by modifications of the spectral phase. We show that, when the noisy phase is enhanced using the proposed phase reconstruction, instrumental measures predict an increase of speech quality over a range of signal to noise ratios, even without explicit amplitude enhancement.

Apparatus and Method for Beamforming to Obtain Voice and Noise Signals

Abstract: One method of operation includes beamforming a plurality of microphone outputs to obtain a plurality of virtual microphone audio channels. Each virtual microphone audio channel corresponds to a beamform. The virtual microphone audio channels include at least one voice channel (135) and at least one noise channel (136). The method includes performing voice activity detection (151) on the at least one voice channel (135) and adjusting a corresponding voice beamform until voice activity detection (151) indicates that voice is present on the at least one voice channel (135). Another method beamforms the plurality of microphone outputs to obtain a plurality of virtual microphone audio channels, where each virtual microphone audio channel corresponds to a beamform, and with at least one voice channel (135) and at least one noise channel (136). The method performs voice recognition on the at least one voice channel (135) and adjusts the corresponding voice beamform to improve a voice recognition confidence metric (159).

Microphone partial occlusion detector

Abstract: Digital signal processing for microphone partial occlusion detection is described. In one embodiment, an electronic system for audio noise processing and for noise reduction, using a plurality of microphones, includes a first noise estimator to process a first audio signal from a first one of the microphones, and generate a first noise estimate. The electronic system also includes a second noise estimator to process the first audio signal, and a second audio signal from a second one of the microphones, in parallel with the first noise estimator, and generate a second noise estimate. A microphone partial occlusion detector determines a low frequency band separation of the first and second audio signals and a high frequency band separation of the first and second audio signals to generate a microphone partial occlusion function that indicates whether one of the microphones is partially occluded.

Systems and methods for multi-mode adaptive noise cancellation for audio headsets

Abstract: An integrated circuit for implementing at least a portion of a personal audio device may include an output and a processing circuit. The output may provide an output signal to a transducer including both a source audio signal for playback to a listener and an anti-noise signal for countering the effect of ambient audio sounds in an acoustic output of the transducer. The processing circuit may implement an adaptive noise cancellation system that generates the anti-noise signal to reduce the presence of the ambient audio sounds heard by the listener by adapting, based on a presence of the source audio signal, a response of the adaptive noise cancellation system to minimize the ambient audio sounds at the acoustic output of the transducer, wherein the adaptive noise cancellation system is configured to adapt both in the presence and the absence of the source audio signal.

Human Responses to Simulated Motorized Noise in National Parks

Abstract: Natural sounds contribute to high-quality experiences for visitors to protected areas. This study investigated the effects of three common sources of recreational motorized noise on laboratory participants’ evaluations of landscape scenes. Seventy-five study participants completed landscape assessments along eight aesthetic and experiential dimensions while listening to audio clips of natural sounds, propeller planes, motorcycles, and snowmobiles. The change from the natural sound baseline for each motorized source of noise was calculated. Results indicated that all motorized sources of noise had detrimental impacts on evaluations of landscape quality compared with natural sounds. Motorcycle noise was demonstrated to have the largest negative impact on landscape assessments. In addition to confirming that noise from motorized recreation has significant impacts on the experiences of potential park visitors, this simulation suggests that the specific source of the noise is an important factor in determining...

Ambient noise impact on accuracy of automated hearing assessment

Abstract: Objective: The aim of this study was to determine the effect of ambient noise on the accuracy of thresholds obtained using the KUDUwave portable clinical audiometer as compared to those obtained using a GSI-61 clinical audiometer in a sound booth. Design: Pure-tone air conduction thresholds were obtained in three conditions: (1) with a clinical audiometer in a quiet sound booth, (2) with the KUDUwave in a quiet sound booth, and (3) with the KUDUwave with 40 dBA of background noise. Study sample: A total of 31 individuals ranging in age from 15 to 80 years participated in the study, 21 with normal hearing and ten with hearing loss. Results: Eighty-nine percent of thresholds obtained with the KUDUwave in quiet, and 92% of thresholds obtained with the KUDUwave in background noise were within 5 dB of those obtained with the clinical audiometer. Accuracy was poorer at 250 Hz and 8000 Hz. Conclusion: Ambient noise typical of that found in a non-sound-treated room, did not affect the accuracy of air cond...

Sound effect switching method and system for mobile terminal

Abstract: The invention discloses a sound effect switching method and system for a mobile terminal. The method comprises the steps that the stream label of an output scene of an audio stream is defined and serves as a mark for distinguishing output scene types, the matching relations between output scene types and the sound modes supported by the mobile terminal are established, and a sound effect comparison table is formed; when an audio output instruction is detected, an audio management service of the mobile terminal identifies the output scene type of the current audio stream by reading the stream label; the sound mode is switched to be matched with the output scene of the audio stream according to the sound mode comparison table. According to the sound effect switching method and system for the mobile terminal, the output scene of the audio stream is identified by means of the audio management service, the matched sound mode is obtained through automatic switching under different scenes, the problem that due to the fact that the global sound effect is not applicable to certain scenes, noise is caused or expected audio enjoyment can not be reached is solved, it is guaranteed that suitable sound effects can be obtained under different application scenes, manual switching conducted by users is not needed, and experience is improved.

Does traffic noise alter calling time in frogs and toads? A case study of anurans in Eastern Ontario, Canada

Abstract: In habitats disturbed by anthropogenic noise, acoustically communicating species may develop behavioral responses that help them transmit information and overcome signal masking. We studied four anuran species breeding in wetlands, ponds, and ditches near a highway in eastern Ontario, Canada, to test whether they called more often when traffic noise intensity was lower, and stopped calling when the noise intensity increased (i.e., gap calling behavior). We made call recordings between April and July 2011, and compared the traffic noise intensity (sound pressure level) between times when the anurans were calling and times when they were not calling. We found that the two species with the highest call peak frequency (American toad, gray treefrog) called randomly with regard to traffic noise intensity. In contrast, the two species with the lowest call peak frequency (green frog, bullfrog) called more often when traffic noise intensity was low. The behavioral response in the two latter species likely represents a short-term strategy that enhances their signal-to-noise ratio thereby increasing the chance of effective communication. Our results support predictions derived from the acoustic adaptation hypothesis: low-frequency signals are more prone to be masked by anthropogenic noise and therefore require behavioral adjustments (in this study gap-calling behavior) to ameliorate this effect.

Single microphone wind noise PSD estimation using signal centroids

Abstract: This contribution presents an efficient technique for the enhancement of speech signals disturbed by wind noise. In almost all noise reduction systems an estimate of the current noise power spectral density (PSD) is required. As common methods for background noise estimation fail due to the non-stationary characteristics of wind noise signals, special algorithms are required. The proposed estimation technique consists of three steps: a feature extraction followed by a wind noise detection and the calculation of the current wind noise PSD. For all steps we exploit the different spectral energy distributions of speech and wind noise. In this context, the so-called signal centroids are introduced. Investigations with measured audio data show that our method can cope with the non-stationary characteristics and enables a sufficient reduction of wind noise. In contrast to other wind noise reduction schemes the proposed algorithm has low complexity and low memory consumption.

The effects of aging on speech perception in noise: comparison between normal-hearing and cochlear-implant listeners.

Abstract: BACKGROUND Speech understanding in noise is comparatively more problematic for older listeners with and without hearing loss, and age-related changes in temporal resolution might be associated with reduced speech recognition in complex noise. PURPOSE The purpose of this study was to investigate the effects of aging on temporal processing and speech perception in noise for normal-hearing (NH) and cochlear-implant (CI) listeners. RESEARCH DESIGN All participants completed three experimental procedures: (1) amplitude modulation (AM) detection thresholds, (2) sentence recognition in quiet, and (3) speech recognition in steady or modulating noise. STUDY SAMPLE Four listener groups participated in the study: 11 younger (≤ 30 yr old, YNH) listeners and 12 older (> 60 yr old, ONH) listeners with NH and 7 younger ( 60 yr old, OCI) CI users. CI listeners have been wearing their device either monaurally or binaurally at least 1 yr. DATA COLLECTION AND ANALYSIS For speech recognition testing, there were eight listening conditions in noise (4 modulation frequencies × 2 signal-to-noise ratios) and one in quiet for each listener. For modulation detection testing, a broadband noise with a duration of 500 msec served as the stimuli at three temporal modulation frequencies of 2, 4, and 8 Hz, which were used to modulate the noise in the speech recognition experiment. We measured AM detection thresholds using a two-interval, two-alternative, forced-choice adaptive procedure. We conducted a series of analysis of variance tests to examine the effect of aging on each test result and measured the correlation coefficient between speech recognition in noise and modulation detection thresholds. RESULTS Although older NH and CI listeners performed similar to the younger listeners with the same hearing status for sentence recognition in quiet, there was a significant aging effect on speech recognition in noise. Regardless of modulation frequency and signal-to-noise ratio, speech recognition scores of the older listeners were poorer than those of the younger listeners when hearing status was matched. We also found a significant effect of aging on AM detection at each modulating frequency and a strong correlation between speech recognition in modulating noise and AM detection thresholds at 2 and 4 Hz. CONCLUSIONS Regardless of differences in hearing status, the degree and pattern of aging effect on auditory processing of the NH listener groups were similar to those of the CI listener groups. This result suggests that age-related declines in speech understanding are likely multifactorial, including peripheral and central factors. Although the age cutoff of the current older age group was 10 yr less than in previous studies (Dubno et al, 2002; Lin et al, 2011), we still found the age-related differences on two auditory tasks. This study extends the knowledge of age-related auditory perception difficulties to CI listeners.

Evaluation of the noise exposure of symphonic orchestra musicians.

Abstract: For musicians, the impact of noise exposure is not yet fully characterized. Some inconsistencies can be found in the methodology used to evaluate noise exposure. This study aims to analyze the noise exposure of musicians in a symphonic orchestra to understand their risk for hearing loss, applying the methodology proposed by ISO 9612:2009. Noise levels were monitored among musicians during the rehearsal of eight different repertoires. Test subjects were selected according to their instrument and position in the orchestra. Participants wore noise dosimeters throughout the rehearsals. A sound meter was used to analyze the exposure of the conductor. The results showed that musicians are exposed to high noise levels that can damage hearing. Brass, woodwind and percussion and timpani musicians were exposed to noise levels in excess of the upper exposure action level of 85 dB (A), while the other instrumental groups had a lower exposure action level of 80 dB (A). Percussion musicians were exposed to high peak noise levels of 135 dB (C). Sound levels varied by instrument, repertoire and position. Octave frequency analyses showed differences among musicians. This study suggests that musicians are at risk for hearing loss. There is a need for more effective guidelines applicable to all countries, which should define standardized procedures for determining musician noise exposure and should allow exposure level normalization to the year, including different repertoires.

Vehicle hvac noise control system

Abstract: A vehicle HVAC noise control system includes an air handler, a blower motor, an audio event device and a controller. The air handler is connected to a passenger compartment. The blower motor is operably connected to the air handler to provide a plurality of airflow rates for air flowing into a passenger compartment. The audio event device produces an audio event to the passenger compartment. The controller is operatively connected to the blower motor and the audio event device. The controller controls operation of the blower motor initially at an operating airflow rate and subsequently switches to a noise reducing airflow rate in response to a signal from the audio event device indicating that the audio event device will subsequently produce the audio event.

Method and device for controlling audio output volume in playing device

Abstract: The invention aims at providing a method and a device for controlling audio output volume in a playing device. The method includes the following steps that the comfortable volume of a current user is obtained; playing environment information of the playing device is obtained; the output volume of the audio to be played is determined based on the playing environment information and the comfortable volume of the user, and the audio to be played is played. The method and the device have the advantage that by obtaining the comfortable volume selected by the user, the output volume of the audio to be played is determined based on the comfortable volume of the user, and the user can hear the audio with the comfortable volume, noise in the current environment and effects of distance between the user and the playing device on the output volume can be determined, corresponding adjustment is conducted on the volume of the audio to be played, and the user can hear the audio to be played clearly.

Dim Down the Lights: Implementing Quiet Time in the Coronary Care Unit

Abstract: To purchase electronic and print reprints, contact the American Association of Critical-Care Nurses, 101 Columbia, Aliso Viejo, CA 92656. Phone, (800) 809-2273; fax, (949) 362-2049; e-mail, reprints@aacn.org. setting. After exclusions, 10 articles remained to serve as a framework for our Quiet Time development. The literature review was presented at a unit-based shared governance meeting. A Quiet Time subcommittee was created (comprising nurses from day and night shifts), and members were designated as “Quiet Time champions.” It was decided by the nursing staff that Quiet Time would be most beneficial from 2 to 3 PM and 2 to 4 AM daily. Education and information about the new Quiet Time initiative was disseminated to physicians, physical therapists, and other potential persons who might be affected. Nurses were encouraged to advocate for the rescheduling of routine procedures that occurred during our designated Quiet Time when appropriate. All admitted patients and their families are educated about Quiet Time during unit orientation. Families are encouraged to bring in music and other items that will specifically relax their family member (eg, a special blanket, soothing music). Before Quiet Time periods, nurses, if appropriate, premedicate for pain, pretoilet, and reposition patients to ensure a period of uninterrupted rest. When Quiet Time begins, a one-on-one announcement is made to every Critical care units can be hectic, chaotic, and generally overstimulating. The interventions that critical care patients require can interrupt normal sleep/wake cycles, causing cognitive and physiological disturbances such as delirium and hemodynamic instability. Providing a healing environment can often be a challenge, one that the coronary care unit (CCU) at The Valley Hospital has confronted head on with our Quiet Time initiative. Quiet Time was devised to promote adequate rest for our vulnerable critical care population. Two blocks of time (2-3 PM and 2-4 AM) have been designated during which lights are dimmed, noise-reduction strategies are implemented, and procedures are minimized. Our main goal is to implement strategies that will lower the noise level, promote rest and healing, and increase patient satisfaction. Our stated objectives are as follows: 1. Identify common noise sources that interfere with a patient’s ability to rest. 2. Measure sound levels in the CCU to which patients are often exposed. 3. Minimize procedures and disruptions during the designated time frame. 4. Increase patient satisfaction. We began by using the CINAHL database to perform a literature review, searching for the key words “quiet time,” “noise,” and “intensive care unit” and excluding articles that were not relevant to the CCU

Systems and methods for automatic signal attenuation

Abstract: The systems, devices, and methods provided herein can be used with an audio source to automatically attenuate a signal from the audio source in response to information about a local environment. In an example, an attenuation network circuit can be used to provide automatic or “hands-free” adjustment of a headphone or headset volume level to compensate for changing background sound or noise levels. In relatively noisy environments, the attenuation network can provide little or no attenuation of an audio source signal, and in less noisy environments, the attenuation network can provide greater attenuation of the audio source signal.

Analysis of instrument self-noise and microseismic event detection using power spectral density estimates

Abstract: Summary Apart from the event magnitude, hypocentral distance, and background noise level, the instrument self-noise can also act as a major constraint for the detection of weak microseismic events in particular for deployments in quiet environments such as below 1.5-2km depths. Instrument self-noise levels that are comparable or above background noise levels may not only complicate detection of weak events at larger distances but also challenge methods such as seismic interferometry which aim at analysis of coherent features in the noise wavefields to reveal subsurface structure. In this article we use power spectral densities to estimate the instrument self-noise for a sample dataset and show how it could affect further analysis. We also suggest that the variations of the spectral powers in a timefrequency representation can be used as a new criterion for event detection and therefore propose a new event time picking algorithm. Compared to the common Short-time average and Long-time average (STA/LTA) method, our suggested technique requires easier parameter settings and detects small events with anomalous spectral powers with respect to an estimated background noise spectrum.

Measurement, analysis and simulation of wind noise signals for mobile communication devices

Abstract: In this contribution, we study the characteristics of sound generated by wind and a signal model for the synthesis of wind noise signals is derived. An analysis of the statistics of wind noise recorded in a laboratory setup is carried out with respect to the spectral and temporal properties of the signals. In particular, an autoregresive model is developed for the spectral shape description and the temporal statistics are modeled by a Markov chain. These two components are combined in a model which synthesizes reproducible artificial wind noise signals. Furthermore, a database of measured wind noise signals is provided. The aim of this model and the measured audio data is to provide wind signals for the evaluation of speech enhancement and noise reduction systems.

Limiting active noise cancellation output

Abstract: In general, techniques are described for limiting active noise cancellation output. As one example, an apparatus comprising one or more processors may perform the techniques. The one or more processors may be configured to, when an estimated noise level increases, dynamically lowering application of active noise cancellation to at least a portion of an audio signal to obtain at least a portion of an active noise cancelled version of the audio signal.

Automatic Segmentation of Audio Signals for Bird Species Identification

Abstract: The identification of bird species from their audio recorded songs are nowadays used in several important applications, such as to monitor the quality of the environment and to prevent bird-plane collisions near airports. The complete identification cycle involves the use of: (a) recording devices to acquire the songs, (b) audio processing techniques to remove the noise and to select the most representative elements of the signal, (c) feature extraction procedures to obtain relevant characteristics, and (d) decision procedures to make the identification. The decision procedures can be obtained by Machine Learning (ML) algorithms, considering the problem in a standard classification scenario. One key element is this cycle is the selection of the most relevant segments of the audio for identification purposes. In this paper we show that the use of short audio segments with high amplitude - called pulses in our work - outperforms the use of the complete audio records in the species identification task. We also show how these pulses can be automatically obtained, based on measurements performed directly on the audio signal. The employed classifiers are trained using a previously labeled database of bird songs. We use a database that contains bird song recordings from 75 species which appear in the Southern Atlantic Coast of South America. Obtained results show that the use of automatically obtained pulses and a SVM classifier produce the best results, all the necessary procedures can be installed in a dedicated hardware, allowing the construction of a specific bird identification device.

Adapting audio based upon detected environmental acoustics

Abstract: Technologies pertaining to improving an auditory experience of a listener are described. Audio is modified based upon noise generated by noise sources in an environment. A microphone generates a signal that is representative of noise in the environment, and the signal is processed to identify peak frequencies therein. When a key frequency of the audio is proximate to a peak frequency in the noise, the audio is modified to improve the listener's perception of the audio.

Quad-based audio fingerprinting robust to time and frequency scaling

Abstract: We propose a new audio fingerprinting method that adapts findings from the field of blind astrometry to define simple, efficiently representable characteristic feature combinations called quads. Based on these, an audio identification algorithm is described that is robust to large amounts of noise and speed, tempo and pitch-shifting distortions. In addition to reliably identifying audio queries that are modified in this way, it also accurately estimates the scaling factors of the applied time/frequency distortions. We experimentally evaluate the performance of the method for a diverse set of noise, speed and tempo modifications, and identify a number of advantages of the new method over a recently published distortioninvariant audio copy detection algorithm.

Speech enhancement based on a modified spectral subtraction method

Abstract: In this paper, a noisy speech enhancement method based on modified spectral subtraction performed on short time magnitude spectrum is presented. Here the cross-terms containing spectra of noise and clean signals are taken into consideration which are neglected in the traditional spectral subtraction method on the basis of the assumption that clean speech and noise signals are completely uncorrelated which is not true for most of the noises. In this method, the noise estimate to be subtracted from the noisy speech spectrum is proposed to be determined exploiting the low frequency regions of noisy speech of the current frame rather than depending only on the initial silence frames. We argue that this approach of noise estimation is capable of tracking the time variation of the non-stationary noise. By employing the noise estimates thus obtained, a procedure is formulated to reduce noise from the magnitude spectrum of noisy speech signal. The noise reduced magnitude spectrum is then recombined with the unchanged phase spectrum to produce a modified complex spectrum prior to synthesizing an enhanced frame. Extensive simulations are carried out using NOIZEUS database in order to evaluate the performance of the proposed method. It is shown in terms of objective measures, spectrogram analysis and subjective listening tests that the proposed method consistently outperforms one of the state-of-the-art methods of speech enhancement from noisy speech corrupted by babble or car noise at high as well as very low levels of SNR.

Effect of Noise Intensity and Illumination Intensity on Visual Performance

Abstract: The results of Experiment 1 indicated that noise and illumination intensity have a significant effect on character identification performance, which was better at 30 dBA than at 60 and 90 dBA, and better at 500 and 800 lux than at 200 lux. However, the interaction of noise and illumination intensity did not significantly affect visual performance. The results of Experiment 2 indicated that noise and illumination intensity also had a significant effect on reading comprehension performance, which was better at 30 dBA than at 60 and 90 dBA, and better at 500 lux than at 200 and 800 lux. Furthermore, reading comprehension performance was better at 500 lux lighting and 30 dBA noise than with 800 lux and 90 dBA. High noise intensity impaired visual performance, and visual performance at normal illumination intensity was better than at other illumination intensities. The interaction of noise and illumination had a significant effect on reading comprehension. These results indicate that noise intensity lower than 30 dBA and illumination intensity approximately 500 lux might be the optimal conditions for visual work.

Spatial audio database based noise discrimination

Abstract: Methods, systems, and computer-readable and executable instructions for spatial audio database based noise discrimination are described herein. For example, one or more embodiments include comparing a sound received from a plurality of microphones to a spatial audio database, discriminating a speech command and a background noise from the received sound based on the comparison to the spatial audio database, and determining an instruction based on the discriminated speech command.

Adaptive microphone sampling rate techniques

Abstract: An apparatus for adjusting a microphone sampling rate, the apparatus including an input to receive an audio signal from a microphone and a front-end processing module. The front-end processing module is to generate a plurality of frames from the audio signal received by the microphone, determine a noise profile using the plurality of frames, and adjust a sampling rate of the microphone based on the determined noise profile.

Exploiting visual information for enhancing audio signals via source separation and beamforming

Abstract: A system for exploiting visual information for enhancing audio signals via source separation and beamforming is disclosed. The system may obtain visual content associated with an environment of a user, and may extract, from the visual content, metadata associated with the environment. The system may determine a location of the user based on the extracted metadata. Additionally, the system may load, based on the location, an audio profile corresponding to the location of the user. The system may also load a user profile of the user that includes audio data associated with the user. Furthermore, the system may cancel, based on the audio profile and user profile, noise from the environment of the user. Moreover, the system may include adjusting, based on the audio profile and user profile, an audio signal generated by the user so as to enhance the audio signal during a communications session of the user.

A Power Mask based audio fingerprint

Abstract: The Philips audio fingerprint[1] has been used for years, but its robustness against external noise has not been studied accurately. This paper shows the Philips fingerprint is noise resistant, and is capable of recognizing music that is corrupted by noise at a -4 to -7 dB signal to noise ratio. In addition, the drawbacks of the Philips fingerprint are addressed by utilizing a “Power Mask” in conjunction with the Philips fingerprint during the matching process. This Power Mask is a weight matrix given to the fingerprint bits, which allows mismatched bits to be penalized according to their relevance in the fingerprint. The effectiveness of the proposed fingerprint was evaluated by experiments using a database of 1030 songs and 1184 query files that were heavily corrupted by two types of noise at varying levels. Our experiments show the proposed method has significantly improved the noise resistance of the standard Philips fingerprint.