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At least 1,261 records · Page 70Linked to original sources
PubMed2006

Respiratory patterns in oscine birds during normal respiration and song production.

In this work we study the generation of respiratory patterns by oscine birds. We present a model capable of generating realistic respiratory patterns, during normal respiration and song production. The model accounts for the interaction between neural nuclei and air sac dynamics. We performed experiments in vivo in order to test the predictions of the model, measuring air sac pressure during song and normal respiration in canaries (Serinus canaria).

Air Sacs↗
PubMed2006

Dynamical systems techniques reveal the sexual dimorphic nature of motor patterns in birdsong.

In this work we analyze the pressure motor patterns used by canaries (Serinus canaria) during song, both in the cases of males and testosterone treated females. We found a qualitative difference between them which was not obvious from the acoustical features of the uttered songs. We also show the diversity of patterns, both for males and females, to be consistent with a recently proposed model for the dynamics of the oscine respiratory system. The model not only allows us to reproduce qualitative features of the different pressure patterns, but also to account for all the diversity of pressure patterns found in females.

Animals↗
PubMed2004

Two-tone suppression and combination tone generation as computations performed by the Hopf cochlea.

Recent evidence suggests that the compressive nonlinearity responsible for the extreme dynamic range of the mammalian cochlea is implemented in the form of Hopf amplifiers. Whereas Helmholtz's original concept of the cochlea was that of a frequency analyzer, Hopf amplifiers can be stimulated not only by one, but also by neighboring frequencies. To reduce the resulting computational overhead, the mammalian cochlea is aided by two-tone suppression. We show that the laws governing two-tone suppression and the generation of combination tones naturally emerge from the Hopf-cochlea concept. Thus the Hopf concept of the cochlea reproduces not only local properties like the correct frequency response, but additionally accounts for more complex hearing phenomena that may be related to auditory signal computation.

Animals↗
PubMed2006

Song of the dunes as a self-synchronized instrument.

Since Marco Polo it has been known that some sand dunes have the peculiar ability to emit a loud sound with a well-defined frequency, sometimes for several minutes. The origin of this sustained sound has remained mysterious, partly because of its rarity in nature. It has been recognized that the sound is not due to the air flow around the dunes but to the motion of an avalanche, and not to an acoustic excitation of the grains but to their relative motion. By comparing singing dunes around the world and two controlled experiments, in the laboratory and the field, we prove that the frequency of the sound is the frequency of the relative motion of the sand grains. Sound is produced because moving grains synchronize their motions. The laboratory experiment shows that the dune is not needed for sound emission. A velocity threshold for sound emission is found in both experiments, and an interpretation is proposed.

Colloids↗
PubMed1992

Detection of laryngeal function using speech and electroglottographic data.

The purpose of this research was to develop quantitative measures for the assessment of laryngeal function using speech and electroglottographic (EGG) data. We developed two procedures for the detection of laryngeal pathology: 1) a spectral distortion measure using pitch synchronous and asynchronous methods with linear predictive coding (LPC) vectors and vector quantization (VQ) and 2) analysis of the EGG signal using time interval and amplitude difference measures. The VQ procedure was conjectured to offer the possibility of circumventing the need to estimate the glottal volume velocity wave-form by inverse filtering techniques. The EGG procedure was to evaluate data that was "nearly" a direct measure of vocal fold vibratory motion and thus was conjectured to offer the potential for providing an excellent assessment of laryngeal function. A threshold based procedure gave 75.9 and 69.0% probability of pathological detection using procedures 1) and 2), respectively, for 29 patients with pathological voices and 52 normal subjects. The false alarm probability was 9.6% for the normal subjects.

Adult↗
PubMed1993

Dynamic formant tracking of noisy speech using temporal analysis on outputs from a nonlinear cochlear model.

In this paper we take a modeling approach to studying representation of formant frequencies of spoken speech and speech in noise in the temporal responses of the peripheral auditory system. On the basis of the properties of the representation, we have devised and evaluated a cross-channel correlation algorithm and an interpeak interval analysis for automatic formant extraction of speech which is strongly dynamic in acoustic characteristics and is embedded in noise. The basilar membrane model used in this study contains laterally coupled damping elements, which are made monotonically dependent on the spatial distribution of the short-term power in the outputs of the model. Efficient digital implementation and the related salient numerical properties of the model are described. Simulation results from the model in response to speech and speech in noise illustrate temporal response patterns that are tonotopically organized in relation to speech formant parameters with little influence by the noise level. By utilizing such relations the devised cross-channel correlation algorithm is shown to be capable of accurately tracking formant movements in spoken syllables and sentences.

Algorithms↗
PubMed2004

Prediction of ultrasound-mediated disruption of cell membranes using machine learning techniques and statistical analysis of acoustic spectra.

Although biological effects of ultrasound must be avoided for safe diagnostic applications, ultrasound's ability to disrupt cell membranes has attracted interest as a method to facilitate drug and gene delivery. This paper seeks to develop "prediction rules" for predicting the degree of cell membrane disruption based on specified ultrasound parameters and measured acoustic signals. Three techniques for generating prediction rules (regression analysis, classification trees and discriminant analysis) are applied to data obtained from a sequence of experiments on bovine red blood cells. For each experiment, the data consist of four ultrasound parameters, acoustic measurements at 400 frequencies, and a measure of cell membrane disruption. To avoid over-training, various combinations of the 404 predictor variables are used when applying the rule generation methods. The results indicate that the variable combination consisting of ultrasound exposure time and acoustic signals measured at the driving frequency and its higher harmonics yields the best rule for all three rule generation methods. The methods used for deriving the prediction rules are broadly applicable, and could be used to develop prediciton rules in other scenarios involving different cell types or tissues. These rules and the methods used to derive them could be used for real-time feedback about ultrasound's biological effects.

Animals↗
PubMed2004

Time-frequency detection and analysis of wheezes during forced exhalation.

The objective of the present work was to detect and analyze wheezes by means of a highly sensitive time-frequency algorithm. Automatic measurements were compared with clinical auscultation for forced exhalation segments from 1.2 to 0 liters/second (l/s). Sensitivities between 100% and 71%, as a function of flow level related to wheezing segments detection, were achieved. Time-frequency wheeze parameters were measured for the flow range from 1.2 to 0.2 l/s. Wheezes were detected in both analyzed groups; asthmatics (N = 16) and control subjects (N = 15). Significant differences between groups were found for the mean number of wheezes detected at basal condition (p = 0.0003). Frequency parameter differences were also significant (0.0112 < p < 0.0307). All these parameters were also studied after applying a bronchodilator drug (Terbutaline). Significant differences between patient groups were found when studying the changes in the number of wheezes for each patient (p = 0.0195). Finally, limited bandwidth parameters, which measure the bronchodilator response, were also studied.

Algorithms↗
PubMed2004

Investigation of vocal jitter and glottal flow spectrum as possible cues for depression and near-term suicidal risk.

Among the many clinical decisions that psychiatrists must make, assessment of a patient's risk of committing suicide is definitely among the most important, complex, and demanding. When reviewing his clinical experience, one of the authors observed that successful predictions of suicidality were often based on the patient's voice independent of content. The voices of suicidal patients judged to be high-risk near-term exhibited unique qualities, which distinguished them from nonsuicidal patients. We investigated the discriminating power of two excitation-based speech parameters, vocal jitter and glottal flow spectrum, for distinguishing among high-risk near-term suicidal, major depressed, and nonsuicidal patients. Our sample consisted of ten high-risk near-term suicidal patients, ten major depressed patients, and ten nondepressed control subjects. As a result of two sample statistical analyses, mean vocal jitter was found to be a significant discriminator only between suicidal and nondepressed control groups (p < 0.05). The slope of the glottal flow spectrum, on the other hand, was a significant discriminator between all three groups (p < 0.05). A maximum likelihood classifier, developed by combining the a posteriori probabilities of these two features, yielded correct classification scores of 85% between near-term suicidal patients and nondepressed controls, 90% between depressed patients and nondepressed controls, and 75% between near-term suicidal patients and depressed patients. These preliminary classification results support the hypothesized link between phonation and near-term suicidal risk. However, validation of the proposed measures on a larger sample size is necessary.

Algorithms↗
PubMed2004

Detection of respiratory sounds at the external ear.

Several clinical and ambulatory settings necessitate respiratory monitoring without a mouthpiece or facemask. Several studies have demonstrated the utility of breathing sound measurements performed on the chest or neck to detect airflow. However, there are limitations to skin surface measurements, including susceptibility to external noise and transducer motion. Thus, this two-part study investigated a novel location for breathing sound measurements: the external ear. The first study investigated characteristics of sound transmission from the oropharynx to the external ear in 19 adults (nine males). Broadband noise was directed into the oropharynx through a tube and mouthpiece and measured indirectly via an accelerometer affixed to the cheek. Resultant transmission to the external ear was measured with a microphone inserted into an earplug that provided acoustic isolation from ambient noise. Near-unity coherence estimates (> 0.9) between the sounds recorded at the external ear and the oropharynx were observed up to approximately 800 Hz, indicating a low-frequency region of preferred transmission. In the second study, each of 20 subjects (nine males) breathed through a pneumotachograph at targeted shallow (3.0 mL/s/kg) and tidal (7.5 mL/s/kg) flows normalized to body mass, and the resulting sounds were recorded at the external ear. Recordings during breath hold measured background noise. Shallow and tidal expiratory flows, respectively, produced signal-plus-noise-to-noise [(S + N)/N] ratios of 6.7 +/- 4.1 dB and 14.0 +/- 5.3 dB (mean +/- standard deviation) across all subjects between 150 and 300 Hz. Concurrent inspiration demonstrated (S + N)/N ratios of 6.6 +/- 3.9 dB and 14.9 +/- 6.3 dB. Thus, the external ear shows promise as an anatomic site to detect and monitor breathing in a relatively noninvasive and unobtrusive manner.

Adolescent↗
PubMed2005

Encoding frequency modulation to improve cochlear implant performance in noise.

Different from traditional Fourier analysis, a signal can be decomposed into amplitude and frequency modulation components. The speech processing strategy in most modern cochlear implants only extracts and encodes amplitude modulation in a limited number of frequency bands. While amplitude modulation encoding has allowed cochlear implant users to achieve good speech recognition in quiet, their performance in noise is severely compromised. Here, we propose a novel speech processing strategy that encodes both amplitude and frequency modulations in order to improve cochlear implant performance in noise. By removing the center frequency from the subband signals and additionally limiting the frequency modulation's range and rate, the present strategy transforms the fast-varying temporal fine structure into a slowly varying frequency modulation signal. As a first step, we evaluated the potential contribution of additional frequency modulation to speech recognition in noise via acoustic simulations of the cochlear implant. We found that while amplitude modulation from a limited number of spectral bands is sufficient to support speech recognition in quiet, frequency modulation is needed to support speech recognition in noise. In particular, improvement by as much as 71 percentage points was observed for sentence recognition in the presence of a competing voice. The present result strongly suggests that frequency modulation be extracted and encoded to improve cochlear implant performance in realistic listening situations. We have proposed several implementation methods to stimulate further investigation. Index Terms-Amplitude modulation, cochlear implant, fine structure, frequency modulation, signal processing, speech recognition, temporal envelope.

Algorithms↗
PubMed2005

Discrimination of pathological voices using a time-frequency approach.

Acoustical measures of vocal function are routinely used in the assessments of disordered voice, and for monitoring the patient's progress over the course of voice therapy. Typically, acoustic measures are extracted from sustained vowel stimuli where short-term and long-term perturbations in fundamental frequency and intensity, and the level of "glottal noise" are used to characterize the vocal function. However, acoustic measures extracted from continuous speech samples may well be required for accurate prediction of abnormal voice quality that is relevant to the client's "real world" experience. In contrast with sustained vowel research, there is relatively sparse literature on the effectiveness of acoustic measures extracted from continuous speech samples. This is partially due to the challenge of segmenting the speech signal into voiced, unvoiced, and silence periods before features can be extracted for vocal function characterization. In this paper we propose a joint time-frequency approach for classifying pathological voices using continuous speech signals that obviates the need for such segmentation. The speech signals were decomposed using an adaptive time-frequency transform algorithm, and several features such as the octave max, octave mean, energy ratio, length ratio, and frequency ratio were extracted from the decomposition parameters and analyzed using statistical pattern classification techniques. Experiments with a database consisting of continuous speech samples from 51 normal and 161 pathological talkers yielded a classification accuracy of 93.4%.

Algorithms↗
PubMed2005

An ultra-low-power programmable analog bionic ear processor.

We report a programmable analog bionic ear (cochlear implant) processor in a 1.5-microm BiCMOS technology with a power consumption of 211 microW and 77-dB dynamic range of operation. The 9.58 mm x 9.23 mm processor chip runs on a 2.8 V supply and has a power consumption that is lower than state-of-the-art analog-to-digital (A/D)-then-DSP designs by a factor of 25. It is suitable for use in fully implanted cochlear-implant systems of the future which require decades of operation on a 100-mAh rechargeable battery with a finite number of charge-discharge cycles. It may also be used as an ultra-low-power spectrum-analysis front end in portable speech-recognition systems. The power consumption of the processor includes the 100 microW power consumption of a JFET-buffered electret microphone and an associated on-chip microphone front end. An automatic gain control circuit compresses the 77-dB input dynamic range into a narrower internal dynamic range (IDR) of 57 dB at which each of the 16 spectral channels of the processor operate. The output bits of the processor are scanned and reported off chip in a format suitable for continuous-interleaved-sampling stimulation of electrodes. Power-supply-immune biasing circuits ensure robust operation of the processor in the high-RF-noise environment typical of cochlear implant systems.

Algorithms↗