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Showing papers on "Noise published in 2007"


Journal ArticleDOI
TL;DR: The present study demonstrates that the spectral-ripple discrimination test, which is time efficient and nonlinguistic, would be a useful tool to evaluate cochlear implant performance with different signal processing strategies.
Abstract: Speech perception ability in noise is one of the most practical measures of success with a cochlear implant; however, with experience, this ability can change dramatically over time, making it a less than ideal tool for comparing performance among different processing strategies. This study examined performance on a spectral discrimination task and compared it to speech perception in noise. An adaptive procedure was used to determine the spectral-ripple density that subjects could discriminate. A closed-set, forced-choice adaptive procedure was used to determine speech reception thresholds for words in two-talker babble and in speech-shaped, steady-state noise. Spectral-ripple thresholds (ripples/octave) were significantly correlated with speech reception thresholds (dB SNR) in noise for 29 cochlear implant users (r = −0.55, p = 0.002 in two-talker babble; r = −0.62, p = 0.0004 in steady-state noise), demonstrating that better spectral resolution was associated with better speech perception in noise. A significant correlation was also found between the spectral-ripple discrimination ability and word recognition in quiet (r = 0.50, p = 0.009). In addition, test–retest reliability for spectral-ripple discrimination was good, and no learning was observed. The present study demonstrates that the spectral-ripple discrimination test, which is time efficient and nonlinguistic, would be a useful tool to evaluate cochlear implant performance with different signal processing strategies.

233 citations


Journal ArticleDOI
TL;DR: The two versions of the Mandarin Hearing In Noise Test (MHINT) are the first standardized Mandarin sentence speech intelligibility tests, and response variability within list was low, and inter-list reliability was high, indicating that consistent results can be obtained using any list.
Abstract: Objective: To develop two versions of the Mandarin Hearing In Noise Test (MHINT). These tests are adaptive tests that measure the reception threshold for sentences (RTSs) in quiet and in noise. The RTS is the presentation level at which half the sentences are correctly recognized. Design: Four studies were undertaken to (1) develop sentence materials, (2) equalize sentence difficulty, (3) create phonemically balanced sentence lists; and (4) evaluate within-list response variability, inter-list reliability, and produce normative data. A total of 137 native Mandarin (Putonghua) speaking subjects in Mainland China and 89 native Mandarin speakers in Taiwan participated. They had normal hearing thresholds at 25 dB HL or better. RTSs were measured under four headphone test conditions: Quiet, and in noise with noise originating from the 0 degree (Noise Front), 90 degrees to the right (Noise Right), and 90 degrees to the left (Noise Left). The speech originated from the front (0 degree) in all conditions. The noise level was fixed at 65 dBA, and the speech was varied adaptively to find the RTS. Results: Two versions of the test materials, consisting of 24, 20-sentence lists each in Mandarin spoken in the Mainland (the MHINT-M) and the dialect of Mandarin spoken in Taiwan (the MHINT-T), were created from two sets of 240 sentences containing 10 syllables per sentence. The mean Quiet RTS was 14.7 dBA, using the MHINT-M, and 19.4 dBA, using the MHINT-T. Using the MHINT-M, the mean RTS for Noise Front was -4.3 dB signal-to-noise ratio (SNR), -11.7 dB SNR for Noise Right, and -11.7 dB SNR for Noise Left. Using the MHINT-T, the Noise Front RTS was -4.0 dB SNR, -10.7 dB SNR for Noise Right, and -11.0 dB SNR for Noise Left. Results in noise are slightly better than those seen for the English HINT norms. Response variability within list was low, and inter-list reliability was high, indicating that consistent results can be obtained using any list. Confidence intervals are reported. Conclusions: The two versions of the MHINT are the first standardized Mandarin sentence speech intelligibility tests. Similar to other language versions of the HINT, the MHINT was developed using the same rationale as the English HINT, allowing norm-referenced results for the MHINT to be compared directly with results in other languages. The MHINT would benefit from further evaluation of its validity.

193 citations


Journal ArticleDOI
TL;DR: It is the temporal distortion rather than the spectral distortion of the low-frequency components that disrupts word identification, and a simulation of cochlear hearing loss had significantly less temporal distortion than was produced by jittering.

181 citations


Book
15 Aug 2007
TL;DR: In this paper, the authors discuss the role of noise in music and propose an approach to reduce the amount of noise by using free electricity, industry, and Inept 6. Power 7. Japan 8. Electronic 10. Quiet 11.
Abstract: 1. Introduction: Noise in Music 2. Electricity 3. Free 4. Industry 5. Inept 6. Power 7. Japan 8. Merzbow 9. Electronic 10. Quiet 11. Conclusion.

128 citations


Journal ArticleDOI
TL;DR: Overall, PLLs were quite conservative, which would theoretically allow for extended permissible listening durations, and suggest that MP3 listening levels may not be as significant a concern as has been reported recently in the mainstream media.
Abstract: Objectives: The main objective of this study was to determine the influence of listening environment and earphone style on the preferred-listening levels (PLLs) measured in users’ ear canals with a commercially-available MP3 player. It was hypothesized that listeners would prefer higher levels with earbud headphones as opposed to over-the-ear headphones, and that the effects would depend on the environment in which the user was listening. A secondary objective was to use the measured PLLs to determine the permissible listening duration to reach 100% daily noise dose. Design: There were two independent variables in this study. The first, headphone style, had three levels: earbud, over-the-ear, and over-the-ear with noise reduction (the same headphones with a noise reduction circuit). The second, environment, also had 3 levels: quiet, street noise and multi-talker babble. The dependent variable was ear canal A-weighted sound pressure level. A 3 3 within-subjects repeated-measures ANOVA was used to analyze the data. Thirtyeight normal hearing adults were recruited from the Faculty of Rehabilitation Medicine at the University of Alberta. Each subject listened to the same song and adjusted the level until it “sounded best” to them in each of the 9 conditions. Results: Significant main effects were found for both the headphone style and environment factors. On average, listeners had higher preferred listening levels with the earbud headphones, than with the over-the-ear headphones. When the noise reduction circuit was used with the over-the-ear headphones, the average PLL was even lower. On average, listeners had higher PLLs in street noise than in multi-talker babble and both of these were higher than the PLL for the quiet condition. The interaction between headphone style and environment was also significant. Details of individual contrasts are explored. Overall, PLLs were quite conservative, which would theoretically allow for extended permissible listening durations. Finally, we investigated the maximum output level of the MP3 player in the ear canals of authors 1 and 3 of this paper. Levels were highest with the earbud style, followed by the over-the-ear with noise reduction. The overthe-ear headphone without noise reduction had the lowest maximum output. Conclusions: The majority of MP3 players are sold with the earbud style of headphones. Preferred listening levels are higher with this style of headphone compared to the over-the-ear style. Moreover, as the noise level in the environment increases, earbud users are even more susceptible to background noise and consequently increase the level of the music to overcome this. The result is an increased sound pressure level at the eardrum. However, the levels chosen by our subjects suggest that MP3 listening levels may not be as significant a concern as has been reported recently in the mainstream media. (Ear & Hearing 2007;28;290–297)

108 citations



Journal ArticleDOI
TL;DR: The results suggest that alterations of subjective evaluation of sleep were determined by physical parameters of the noise but modified by individual factors like noise sensitivity.
Abstract: In order to determine the influence of noise sensitivity on sleep, subjective sleep quality, annoyance, and performance after nocturnal exposure to traffic noise, 12 women and 12 men (age range, 19-28 years) were observed during four consecutive nights over a three weeks period. After a habituation night, the participants were exposed with weekly permuted changes to air, rail and road traffic noise. Of the four nights, one was a quiet night (32 dBA), while three were noisy nights with exposure to equivalent noise levels of 39, 44, and 50 dBA in a permuted order. The traffic noise caused alterations of most of the physiological parameters, subjective evaluation of sleep, annoyance, and performance. Correlations were found between noise sensitivity and subjective sleep quality in terms of worsened restoration, decreased calmness, difficulty to fall asleep, and body movements. The results suggest that alterations of subjective evaluation of sleep were determined by physical parameters of the noise but modified by individual factors like noise sensitivity.

95 citations


Journal ArticleDOI
TL;DR: It is shown that cepstral smoothing can effectively prevent spectral peaks of short duration that may be perceived as musical noise, and preserves speech onsets, plosives, and quasi-stationary narrowband structures like voiced speech.
Abstract: Many speech enhancement algorithms that modify short-term spectral magnitudes of the noisy signal by means of adaptive spectral gain functions are plagued by annoying spectral outliers. In this letter, we propose cepstral smoothing as a solution to this problem. We show that cepstral smoothing can effectively prevent spectral peaks of short duration that may be perceived as musical noise. At the same time, cepstral smoothing preserves speech onsets, plosives, and quasi-stationary narrowband structures like voiced speech. The proposed recursive temporal smoothing is applied to higher cepstral coefficients only, excluding those representing the pitch information. As the higher cepstral coefficients describe the finer spectral structure of the Fourier spectrum, smoothing them along time prevents single coefficients of the filter function from changing excessively and independently of their neighboring bins, thus suppressing musical noise. The proposed cepstral smoothing technique is very effective in nonstationary noise.

90 citations


Journal ArticleDOI
TL;DR: Over the past 40 years, infrasound and low-frequency noise have attracted a great deal of adverse publicity on their effects on health, based mainly on media exaggerations and misunderstandings, and the public takes a one-dimensional view of infrasounds, concerned only by its presence, whilst ignoring its low levels.
Abstract: Definitions of infrasound and low-frequency noise are discussed and the fuzzy boundary between them described. Infrasound, in its popular definition as sound below a frequency of 20 Hz, is clearly audible, the hearing threshold having been measured down to 1.5 Hz. The popular concept that sound below 20 Hz is inaudible is not correct. Sources of infrasound are in the range from very low-frequency atmospheric fluctuations up into the lower audio frequencies. These sources include natural occurrences, industrial installations, low-speed machinery, etc. Investigations of complaints of low-frequency noise often fail to measure any significant noise. This has led some complainants to conjecture that their perception arises from non-acoustic sources, such as electromagnetic radiation. Over the past 40 years, infrasound and low-frequency noise have attracted a great deal of adverse publicity on their effects on health, based mainly on media exaggerations and misunderstandings. A result of this has been that the public takes a one-dimensional view of infrasound, concerned only by its presence, whilst ignoring its low levels.

81 citations


Journal ArticleDOI
TL;DR: Music exposure has made music exposure the most studied source of excessive sound exposure to children and youths in several countries, and not all evidence available confirms increased risk with increasing exposures, and the possibility of a toughening protective effect of such exposures has been suggested.
Abstract: In the past two decades, the number of publications on the risk of acquired hearing loss among children and young adults has increased substantially. The introduction of MP3 players and a lawsuit t...

80 citations


Journal ArticleDOI
Li Xu1, Yunfang Zheng
TL;DR: There was a trade-off between temporal and spectral cues for phoneme recognition in noise and there was no further improvement in performance for consonant recognition when the number of channels was > or =12 in any of the three conditions.
Abstract: Cochlear implant users receive limited spectral and temporal information. Their speech recognition deteriorates dramatically in noise. The aim of the present study was to determine the relative contributions of spectral and temporal cues to speech recognition in noise. Spectral information was manipulated by varying the number of channels from 2 to 32 in a noise-excited vocoder. Temporal information was manipulated by varying the low-pass cutoff frequency of the envelope extractor from 1to512Hz. Ten normal-hearing, native speakers of English participated in tests of phoneme recognition using vocoder processed consonants and vowels under three conditions (quiet, and +6 and 0dB signal-to-noise ratios). The number of channels required for vowel-recognition performance to plateau increased from 12 in quiet to 16–24 in the two noise conditions. However, for consonant recognition, no further improvement in performance was evident when the number of channels was ⩾12 in any of the three conditions. The contributi...

Patent
22 Jun 2007
TL;DR: In this paper, a system for facilitating conversational communications in an environment with background noise, including a microphone for sensing the background noise and a signal processor configured to process the microphone output and produce an anti-noise electrical output, was presented.
Abstract: A system for facilitating conversational communications in an environment with background noise, the system including a microphone for sensing the background noise, a signal processor configured to process the microphone output and produce an anti-noise electrical output, and a directional speaker array configured to receive the anti-noise electrical output and directionally broadcast anti-noise audio output, the anti noise audio output destructively interfering with the environmental background noise.

Proceedings ArticleDOI
22 Oct 2007
TL;DR: This study used empirical mode decomposition (EMD) for filtering power line noise in electrocardiogram signals by adding a pseudo noise at a frequency higher than the highest frequency of the signal to filter out just the power line Noise in the first IMF.
Abstract: This study used empirical mode decomposition (EMD) for filtering power line noise in electrocardiogram signals. When the signal-to-noise (SNR) is low, the power line noise is separated out as the first intrinsic mode function (IMF), but when the SNR is high, a part of the signal along with the noise is decomposed as the first IMF. To overcome this problem, we add a pseudo noise at a frequency higher than the highest frequency of the signal to filter out just the power line noise in the first IMF. The results are compared with traditional IIR-based bandstop filtering. This technique is also implemented for filtering power line noise during enhancement of stress ECG signals.

Patent
Darwin Rambo1
12 Mar 2007
TL;DR: In this paper, a signal processor receives signals from a spatially dispersed set of directional microphones, processing the microphone signals and the far-end received audio into a signal for transmission to a far end party.
Abstract: Systems and methods that enable high quality audio teleconferencing are disclosed. In one embodiment of the present invention, a signal processor receives signals from a spatially dispersed set of directional microphones, processing the microphone signals and the far-end received audio into a signal for transmission to a far-end party. The processing may comprise the use of one or more algorithms that reduce conference room noise and may selectively increase participant audio levels by processing the microphone signals using beamforming techniques. An embodiment of the present invention may also comprise one or more omni-directional microphones that may be used in cooperation with the directional microphones to adjust for background noise, acoustic echo, and the existence of side conversations.

Journal ArticleDOI
TL;DR: The deleterious effects of high levels of noise on newborns and health professionals show the need for interventions in routines and professionals and families' conduct.
Abstract: OBJECTIVE: Determine noise levels in the Neonatal Intensive Care Unit and identify the sources of these noises. METHODS: Quantitative, descriptive and exploratory study, carried out in Sao Paulo. Data was collected in April and May of 2005. A dosimeter was used to record a total of 96 hours of measurements. Nine hours of observation were also conducted to identify sources of noise. RESULTS: Leq noise levels ranged from of 61.3 to 66.6 dBA and were higher on the weekends. Peak values ranged from 90.8 to 123.4 dBC and the highest values were recorded at night. The sources of the noise were: beeping noises from ventilators and heart rate monitors, conversations between health professional and others. CONCLUSION: The deleterious effects of high levels of noise on newborns and health professionals show the need for interventions in routines and professionals and families' conduct.

Journal ArticleDOI
TL;DR: The present result suggests that the cochlear-implant users can rely on either temporal or spectral cues to performtone recognition in quiet, but need both cues for tone recognition in noise.
Abstract: Objective: The present study was aimed to examine the relationship between psychophysical performance in temporal and spectral resolution and Mandarin tone recognition in noise by cochlear-implant (CI) listeners. Design: Seventeen Nucleus-24 implant users, 10 postlingually deafened and 7 prelingually deafened, participated in the experiments. A 3-interval, forced-choice procedure was used to measure gap detection and pure-tone frequency discrimination at 250 to 4,000 Hz in octave steps. A 4-alternative forced-choice procedure was used to measure Mandarin tone recognition in quiet and in noise. Signal-to-noise ratios (SNRs) varied from +10 to -10 dB. All stimuli were delivered to the clinical processor via a speaker in a sound free field. The obtained data were compared to data collected from normal-hearing control subjects, as well as cochlear-implant users who performed similar tasks using single-electrode stimulation via a research interface. Results: Postlingually-deafened CI subjects generally performed better than prelingually-deafened subjects. The average gap detection threshold was 30 ms with a range from 4 to 128 ms. The average frequency difference limen was 100 Hz with a range from 12 to 192 Hz, regardless of the standard frequency. The average tone recognition was 80% correct in quiet, which dropped to 55% at +10 dB SNR and essentially chance performance at -5 dB SNR. In comparison, the normal-hearing control subjects maintained essentially perfect performance over this SNR range. Only frequency discrimination at 1,000 Hz was significantly correlated with tone recognition in quiet but all psychophysical measures were correlated to tone recognition in noise. Conclusions: The present result suggests that the CI users can rely on either temporal or spectral cues to perform tone recognition in quiet, but need both cues for tone recognition in noise. Future CI processors need to extract and encode these acoustic cues to achieve better performance in tone perception and production.

Journal ArticleDOI
TL;DR: The vocal responses of males of E. emiliopugini under noise exposure and their contrast with the congeneric species unveil different strategies in confronting interference, whose origins and adaptive significance warrant further study.

Patent
14 Sep 2007
TL;DR: In this paper, an audio encoding device capable of adjusting a spectrum inclination of a quantized noise without changing the Formant weight is presented. But the device is not suitable for high frequency audio signals.
Abstract: Disclosed is an audio encoding device capable of adjusting a spectrum inclination of a quantized noise without changing the Formant weight. The device includes: an HPF (131) which extracts a high-frequency component of the frequency region from an input audio signal; a high-frequency energy level calculation unit (132) which calculates an energy level of the high-frequency component in a frame unit; an LPF (133) which extracts a low-frequency component of the frequency region from the input audio signal; a low-energy level calculation unit (134) which calculates an energy level of a low-frequency component in a frame unit; an inclination correction coefficient calculation unit (141) multiplies the difference between SNR of the high-frequency component and SNR of the low-frequency component inputted from an adder (140) by a constant and adds a bias component to the product so as to calculate an inclination correction coefficient ?3. The inclination correction coefficient is used for adjusting the spectrum inclination of a quantized noise.

Patent
04 Jun 2007
TL;DR: In this paper, an audio stream containing crowd noise from an event (e.g., sporting event, political rally, religious gathering, etc) is captured and time coded and normalized based on geography and processed to remove undesired artifacts and to identify a set (at least one) of highlights.
Abstract: The present invention generally provides a way to analyze crowd noise to identify “highlights” or the like. Specifically, an audio stream containing crowd noise from an event (e.g., sporting event, political rally, religious gathering, etc) is captured (e.g., using microphones) and time coded. The audio stream is normalized based on geography and processed to remove undesired artifacts and to identify a set (at least one) of highlights. Based on at least one threshold, at least one highlight is selected from the set of highlights.

Journal ArticleDOI
TL;DR: Time- and place-patterns of noise in both institutions suggest that human factor is a major source of noise pollution and this pollution is, therefore, potentially modifiable.
Abstract: Introduction : Noise pollution is known to cause deleterious effects on human health and may especially affect frail elderly patients with poor mental and physiologic reserve. Aims of the study : (i) to learn levels and time- and place-patterns of noise in an urban community teaching hospital (TH) and affiliated urban nursing home (NH); (ii) to compare levels and patterns of noise in both institutions. Results : Recordings were obtained in three areas of the TH: emergency room (ER), intensive care units (ICU), and medical-surgical floors (HF) - nurses' stations and patients' rooms. On nursing home floors (NHF), noise levels were recorded at nurses' stations and in patients' rooms. In all areas of the hospital and NH, noise levels were in range of 55-70 dB and exceeded the 40-50 dB limit recommended by the EPA. In ER and ICU, noise level was higher on weekdays than weekends. In ICU and on HF, noise level was higher during mid-day hours during mornings and evenings. The highest noise level was recorded in ER followed by ICU and HF. On HF, nurses' stations were noisier than patients' rooms. Noise level was higher in the TH than in the NH. On NHF, noise level was similar on weekdays and weekends. Noise was stronger at nurses' stations than in patients' rooms and stronger in the mornings and evenings than during mid-day hours. Patterns of noise followed the human factor activities observed in both facilities. Conclusions : The level of noise in both facilities was above the recommended limit and presents an environmental stressor for a frail elderly patient. With transfer from NH to TH exposure to this stressor is increased. Time- and place-patterns of noise in both institutions suggest that human factor is a major source of noise pollution. This pollution is, therefore, potentially modifiable.

Patent
23 May 2007
TL;DR: In this article, a remote power supply arrangement for graphical display systems was proposed, which places one or more power supplies in a separate location, some distance apart from the graphical display, rather than within the display cabinet or module itself.
Abstract: A remote power supply arrangement for graphical display systems. The remote power supply places one or more power supplies in a separate location, some distance apart from the graphical display, rather than within the display cabinet or module itself. Power is then transferred from the power supply to the display by electrical cords. The remote power supply provides lighter weight displays, less noise near the displays, and improved maintenance and repair. The invention includes apparatus, methods of providing remote power to displays.

Patent
26 Apr 2007
TL;DR: In this article, a multi-tap pitch filter is used for filtering a first-layer decoded spectrum according to a filter state set by a state setting unit (112), a pitch coefficient outputted from a pitch-coverage set unit (115), and a filter coefficient outputting from the filter coefficient decision unit (119), and calculates an estimated spectrum of the input spectrum.
Abstract: Provided is an audio encoding device capable of preventing audio quality degradation of a decoded signal. In the audio encoding device, a noise analysis unit (118) analyzes a noise characteristic of a higher range of an input spectrum. A filter coefficient decision unit (119) decides a filter coefficient in accordance with the noise characteristic information from the noise characteristic analysis unit (118). A filtering unit (113) includes a multi-tap pitch filter for filtering a first-layer decoded spectrum according to a filter state set by a filter state setting unit (112), a pitch coefficient outputted from a pitch coefficient setting unit (115), and a filter coefficient outputted from the filter coefficient decision unit (119), and calculates an estimated spectrum of the input spectrum. An optimal pitch coefficient can be decided by the process of a closed loop formed by the filter unit (113), a search unit (114), and the pitch coefficient setting unit (115).

PatentDOI
TL;DR: In this paper, an audio outputting device for switching a plurality of processes to perform a process on an audio signal, and acoustically reproducing and outputting the audio signal is presented.
Abstract: Disclosed herein is an audio outputting device for switching a plurality of processes to perform a process on an audio signal, and acoustically reproducing and outputting the audio signal, the audio outputting device including, a control section for, when changing a process performed on an audio signal from one process to another process, stopping the one process on the audio signal, outputting sound based on the audio signal unprocessed by either of the one process and the other process, and performing the other process on the audio signal after passage of a predetermined period of time.

Journal ArticleDOI
TL;DR: This study found the noise levels reached 98.5–107.5 dB in power generator rooms and air-conditioning facilities, and suggests employees use ear plugs, and used noise reduction methods.
Abstract: Hospitals are places that allow patients to rest and recover, and therefore must be quiet inside and in the surrounding neighborhood. One medical center was chosen as a sample hospital. This hospital was a tertiary care center during the 2003 outbreak of the severe acute respiratory syndrome (SARS) in Taiwan. The measurement results show that the noise level in the wards and stations was between 50.3 and 68.1 dB which exceeded the suggested hospital ward sound level. The quietest units were the Surgical Intensive Care Unit and recovery rooms with a noise level lower than 50 dB during the night. The higher noise levels were in the hall and pharmacy which were highly populated areas. This study analyzed the causes of this excessive noise and used noise reduction methods. The paired t test was performed and the results showed improvement methods were successful. This study found the noise levels reached 98.5-107.5 dB in power generator rooms and air-conditioning facilities, and suggests employees use ear plugs.

Patent
David E. Johnston1
23 Oct 2007
TL;DR: In this paper, a method for editing digital audio data is described, where a first and second segment of the audio data are analyzed at a selected frequency band to identify noise, and the second segment is compressed.
Abstract: Systems and methods for editing digital audio data are provided. In one implementation, a method is provided that includes receiving digital audio data. Input is received selecting a noise threshold identifying a level at which one or more segments of audio data are considered to be noise. The noise threshold is associated with a plurality of parameters of the audio data and applicable to a plurality of frequency bands of the audio data. A first segment and second segment of the digital audio data are analyzed at a selected frequency band to identify noise. When the audio data in the first segment exceeds the noise threshold, the first segment is identified as including a first noise and the audio data is compressed. When audio data in the second segment exceeds the noise threshold, the second segment is identified as including a second noise and the audio data is compressed.

Patent
03 Aug 2007
TL;DR: In this paper, a fast and efficient method to derive the rate controlling parameter and can be applied to generic perceptual audio encoders where low computational complexity is required, can be found.
Abstract: Perceptual audio coder refers to audio compression schemes that exploit the properties of human auditory perception. The coder allocates the quantization noise below the masking threshold such that even with the bit rate limitation, the noise is imperceptible to the ear. These distortion and bit rate requirement makes the bit allocation-quantization process a considerable computational effort. One method includes incrementally adjusting a global gain according to a gradient. The gradient could be adjusted each time the number of bits used to represent a quantized value is counted. Another method includes limiting a rate controlling parameter to a predetermined number of loops. The method could also include deriving a global gain to ensure exit from the loop. Accordingly, embodiments of the present disclosure provide a fast and efficient method to derive the rate controlling parameter and can be applied to generic perceptual audio encoders where low computational complexity is required.

Patent
18 Jul 2007
TL;DR: In this paper, an earphone unit having a speaker unit and a microphone to output an audio signal and to generate anti-noise with respect to external noise, and a circuit unit to compensate a frequency characteristic of the antinoise generated by the microphone of the earphones.
Abstract: An apparatus and method of reducing noise in a portable audio reproducing apparatus using earphones. The apparatus includes an earphone unit having a speaker unit and a microphone to output an audio signal and to generate anti-noise with respect to external noise, and a circuit unit to compensate a frequency characteristic of the anti-noise generated by the microphone of the earphone unit, to add the anti-noise to an input audio signal, and to remove background noise using the anti-noise by outputting the audio signal having the anti-noise to the speaker unit.

Journal ArticleDOI
TL;DR: Improved results are obtained by considering both the intra-frame masking properties and inter-frame spectrum smoothing in order to enhance a speech signal corrupted by colored noise.

Proceedings ArticleDOI
01 Nov 2007
TL;DR: Results demonstrate the superiority of the proposed NDLMS algorithm over conventional LMS algorithms in achieving much smaller steady-state excess mean square errors.
Abstract: LMS adaptive noise cancellers are often used to recover signal corrupted by additive noise. A major drawback of conventional LMS algorithms is that the excess mean-square errors increase linearly with the desired signal power. This results in degraded performance when the desired signal exhibits large power fluctuations. In this paper, a normalized difference LMS (NDLMS) algorithm is proposed to deal with the situation when the desired signal is strong, e.g., speech signals. Simulations were carried out using real speech signal with different noise power levels in both stationary and nonstationary noise environments. Results demonstrate the superiority of the proposed NDLMS algorithm over conventional LMS algorithms in achieving much smaller steady-state excess mean square errors.

Patent
30 May 2007
TL;DR: In this paper, a method and device based on environment noise test and volume adjusting, which comprises the following steps: presetting play volume zone or value range; storing play audio wave data and collected outer audio wave Data; comparing the data and then comparing the result with preset play volume Zone value range, adjusting the volume according to compare result.
Abstract: This invention discloses one method and device based on environment noise test and volume adjusting, which comprises the following steps: presetting play volume zone or value range; storing play audio wave data and collected outer audio wave data; comparing the data and then comparing the result with preset play volume zone value range; adjusting the volume according to compare result.