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1.
The extent to which context influences speech categorization can inform theories of pre-lexical speech perception. Across three conditions, listeners categorized speech targets preceded by speech context syllables. These syllables were presented as the sole context or paired with nonspeech tone contexts previously shown to affect speech categorization. Listeners' context-dependent categorization across these conditions provides evidence that speech and nonspeech context stimuli jointly influence speech processing. Specifically, when the spectral characteristics of speech and nonspeech context stimuli are mismatched such that they are expected to produce opposing effects on speech categorization the influence of nonspeech contexts may undermine, or even reverse, the expected effect of adjacent speech context. Likewise, when spectrally matched, the cross-class contexts may collaborate to increase effects of context. Similar effects are observed even when natural speech syllables, matched in source to the speech categorization targets, serve as the speech contexts. Results are well-predicted by spectral characteristics of the context stimuli.  相似文献   

2.
In order to investigate sound image quality, white and bandlimited noises with various cross-correlation coefficients are reproduced from two loudspeakers placed in anechoic or echoic chambers. Subjects are asked to make similarity judgments and some subjective evaluations of pairs of the noises. The experimental data are analyzed by Kruskal's multidimensional scaling (MDS) program. The analysis of the experimental data shows the following: (1) sound image quality depends mostly on the width and the distance of the sound image, (2) the width of the sound image depends on the absolute value of the cross-correlation coefficient, (3) the distance of the sound image depends on the cross-correlation coefficient itself, (4) with respect to physical and psychological factors governing sound image quality, there is no fundamental difference between anechoic and echoic chambers.  相似文献   

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This paper examines whether correlations between speech perception and speech production exist, and, if so, whether they might provide a way of evaluating different acoustic metrics. The cues listeners use for many phonemic distinctions are not known, often because many different acoustic cues are highly correlated with one another, making it difficult to distinguish among them. Perception-production correlations may provide a new means of doing so. In the present paper, correlations were examined between acoustic measures taken on listeners' perceptual prototypes for a given speech category and on their average production of members of that category. Significant correlations were found for VOT among stop consonants, and for spectral peaks (but not centroids or skewness) for voiceless fricatives. These results suggest that correlations between speech perception and production may provide a methodology for evaluating different proposed acoustic metrics.  相似文献   

5.
In English, voiced and voiceless syllable-initial stop consonants differ in both fundamental frequency at the onset of voicing (onset F0) and voice onset time (VOT). Although both correlates, alone, can cue the voicing contrast, listeners weight VOT more heavily when both are available. Such differential weighting may arise from differences in the perceptual distance between voicing categories along the VOT versus onset F0 dimensions, or it may arise from a bias to pay more attention to VOT than to onset F0. The present experiment examines listeners' use of these two cues when classifying stimuli in which perceptual distance was artificially equated along the two dimensions. Listeners were also trained to categorize stimuli based on one cue at the expense of another. Equating perceptual distance eliminated the expected bias toward VOT before training, but successfully learning to base decisions more on VOT and less on onset F0 was easier than vice versa. Perceptual distance along both dimensions increased for both groups after training, but only VOT-trained listeners showed a decrease in Garner interference. Results lend qualified support to an attentional model of phonetic learning in which learning involves strategic redeployment of selective attention across integral acoustic cues.  相似文献   

6.
The speech intelligibility in classroom can be influenced by background-noise levels, speech sound pressure level (SSPL), reverberation time and signal-to-noise ratio (SNR). The relationship between SSPL and subjective Chinese Mandarin speech intelligibility and the effect of different SNRs on Chinese Mandarin speech intelligibility in the simulated classroom were investigated through room acoustical simulation, auralisation technique and subjective evaluation. Chinese speech intelligibility test signals recorded in anechoic chamber were convolved with the simulated binaural room impulse responses, and then reproduced through the headphone by different SSPLs and SNRs. The results show that Chinese Mandarin speech intelligibility scores increase with increasing of SSPLs and SNRs within a certain range in simulated classrooms. Chinese Mandarin speech intelligibility scores have no significant difference with SNRs of no less than 15 dBA under the same reverberation time condition.  相似文献   

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Speech intelligibility is known to be relatively unaffected by certain deformations of the acoustic spectrum. These include translations, stretching or contracting dilations, and shearing of the spectrum (represented along the logarithmic frequency axis). It is argued here that such robustness reflects a synergy between vocal production and auditory perception. Thus, on the one hand, it is shown that these spectral distortions are produced by common and unavoidable variations among different speakers pertaining to the length, cross-sectional profile, and losses of their vocal tracts. On the other hand, it is argued that these spectral changes leave the auditory cortical representation of the spectrum largely unchanged except for translations along one of its representational axes. These assertions are supported by analyses of production and perception models. On the production side, a simplified sinusoidal model of the vocal tract is developed which analytically relates a few "articulatory" parameters, such as the extent and location of the vocal tract constriction, to the spectral peaks of the acoustic spectra synthesized from it. The model is evaluated by comparing the identification of synthesized sustained vowels to labeled natural vowels extracted from the TIMIT corpus. On the perception side a "multiscale" model of sound processing is utilized to elucidate the effects of the deformations on the representation of the acoustic spectrum in the primary auditory cortex. Finally, the implications of these results for the perception of generally identifiable classes of sound sources beyond the specific case of speech and the vocal tract are discussed.  相似文献   

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Accented speech recognition is more challenging than standard speech recognition due to the effects of phonetic and acoustic confusions. Phonetic confusion in accented speech occurs when an expected phone is pronounced as a different one, which leads to erroneous recognition. Acoustic confusion occurs when the pronounced phone is found to lie acoustically between two baseform models and can be equally recognized as either one. We propose that it is necessary to analyze and model these confusions separately in order to improve accented speech recognition without degrading standard speech recognition. Since low phonetic confusion units in accented speech do not give rise to automatic speech recognition errors, we focus on analyzing and reducing phonetic and acoustic confusability under high phonetic confusion conditions. We propose using likelihood ratio test to measure phonetic confusion, and asymmetric acoustic distance to measure acoustic confusion. Only accent-specific phonetic units with low acoustic confusion are used in an augmented pronunciation dictionary, while phonetic units with high acoustic confusion are reconstructed using decision tree merging. Experimental results show that our approach is effective and superior to methods modeling phonetic confusion or acoustic confusion alone in accented speech, with a significant 5.7% absolute WER reduction, without degrading standard speech recognition.  相似文献   

11.
利用一次南海海山环境下的声传播实验数据,研究了负梯度水文环境下海底山对声传播的影响。针对实验数据中的传播损失异常,从射线声学角度给出了合理的解释,表明海底山的存在引起传播损失在距离上剧烈波动。在距离接收阵较近的7.6 km处,声源位于海山斜坡上,斜坡的反射使接收传播损失减小约8 dB,体现出斜坡增强特征。当声源位于海山后,海底山的遮蔽作用使23.8 km处的传播损失增加超过20 dB,不同位置处海山遮蔽效应的差异使传播损失随距离起伏。利用抛物模型对实验环境下的声传播进行了定量仿真,仿真传播损失同实验结果符合,验证了实验数据中海底山的反射和遮蔽作用。此外,对实验环境下海山的遮蔽损失进行分析,发现在不同声源位置处,海山遮蔽损失在特定频带上同频率对数具有线性关系。  相似文献   

12.
利用一次南海海山环境下的声传播实验数据,研究了负梯度水文环境下海底山对声传播的影响。针对实验数据中的传播损失异常,从射线声学角度给出了合理的解释,表明海底山的存在引起传播损失在距离上剧烈波动。在距离接收阵较近的7.6 km处,声源位于海山斜坡上,斜坡的反射使接收传播损失减小约8 dB,体现出斜坡增强特征。当声源位于海山后,海底山的遮蔽作用使23.8 km处的传播损失增加超过20 dB,不同位置处海山遮蔽效应的差异使传播损失随距离起伏。利用抛物模型对实验环境下的声传播进行了定量仿真,仿真传播损失同实验结果符合,验证了实验数据中海底山的反射和遮蔽作用。此外,对实验环境下海山的遮蔽损失进行分析,发现在不同声源位置处,海山遮蔽损失在特定频带上同频率对数具有线性关系。  相似文献   

13.
为了分析垂直载荷下颗粒物质的声速、声衰减系数、谐波非线性等特性,本工作采用飞行时间法测量了不同含水量下声速随压强的变化规律,并利用傅里叶变换法分析了干、湿玻璃珠样品的声衰减和非线性声学特性。结果表明:干、湿玻璃珠样品中的声速、声衰减系数以及谐波非线性均随压强呈幂律变化;湿颗粒样品中随着液体含量增多,声速逐渐增加,超声波透过湿颗粒样品时的能量耗散和非线性逐渐减小。分析原因表明,压强和孔隙流体改变了颗粒之间的接触分布,使得颗粒体系的声速、声衰减以及谐波非线性等特性都随之发生变化。  相似文献   

14.
In this work, variation of sound speed with pressures under different water contents is measured by using the time-of-flight method. Acoustic attenuation and nonlinear acoustic characteristics of dry and wet glass bead samples are analyzed by using fast Fourier transform.The results show that sound speed, acoustic attenuation coefficient and second harmonic nonlinearity in both dry and wet glass bead samples vary with the pressure in the form of power law. Sound speed gradually increases while t...  相似文献   

15.
循环平稳声场近场声全息理论与实验研究   总被引:2,自引:1,他引:2  
万泉  蒋伟康 《声学学报》2005,30(4):379-384
提出一种用于分析循环平稳声场的近场声全息技术。此类声场信号的调制现象非常严重,频谱上存在着明显的边频带,由于无法有效地分离调制和载波信息,以往近场声全息技术的全息图会在边频带处出现虚假的能量。本技术用二阶循环统计量理论代替传统的傅里叶分析,并以声压的谱相关密度函数取代其频谱及功率谱密度做为重建物理量。由于谱相关密度函数可以对循环平稳信号进行解调处理,使得该技术的全息图上不会因为边频带的存在出现虚假能量。仿真分析及实验研究表明,本技术可以更准确地提取循环平稳声场的调制和载波信息。  相似文献   

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From the equation for the steady state sound pressure distribution produced in a rectangular reverberation chamber by a point source, and by using the usual high frequency approximations, it is shown that, for a random source position, the cross-correlation function for two points not too far apart approaches that of Cook et al. in the reverberant field of the chamber. When the same approach is used on the equation for sound pressure decay when the point source excitation is cut off, the cross-correlation function obtained for the initial portion of the decay corresponds with that determined experimentally by Balachandran and Robinson.  相似文献   

18.
An ensemble Kalman filter(EnKF) approach is proposed to perform sequential tracking of water column sound speed profile(SSP) using a moving acoustic source. First,the SSPs are discretized in depth and range, and are expressed by the empirical orthogonal functions(EOFs). Second, the acoustic source state information and the first three orders of EOF coefficients are expressed as the state variable, and the acoustic field information received by the vertical line array are the measured values. Successively, the state variables and measured values are used to establish the state-measure model. Last, the EnKF is utilized to track the state variables. The simulation results show that the root mean square error of SSP and the absolute error of source are all small, and thus the acoustic source tracking-positioning has high accuracy. Moreover, increasing the number of sample collection, the signal-to-noise ratio and the number of receiving elements can improve the tracking-positioning results. The method is verified using the experimental data of the East China Sea.  相似文献   

19.
浅海声速剖面与移动声源的跟踪定位   总被引:2,自引:0,他引:2       下载免费PDF全文
在水平非均匀分布的浅海环境中,针对移动声源跟踪时,声速剖面的变化会对声场产生影响,提出了一种利用集合卡尔曼滤波算法的声速剖面跟踪反演和移动声源跟踪定位的方法。首先,将声速剖面进行距离和深度的参数化表示,从而将对声速剖面的跟踪转化为对声速剖面前3阶经验正交函数系数的跟踪;其次,通过将声源状态信息和声速剖面信息表示为状态变量,而将垂直线列阵接收到的声场信息作为测量值建立状态-测量模型,然后利用集合卡尔曼滤波方法对模型状态变量进行跟踪。仿真结果得出:声速剖面跟踪反演的均方根误差和移动声源跟踪定位的绝对误差都非常小,对声源的跟踪定位精度很高。并且通过增加集合样本数、增加接收信号信噪比以及增加接收阵元数目都可以提高跟踪定位结果精度。最后,利用东海实验数据对本方法进行了验证。  相似文献   

20.
A review of the absorption mechanism of sound in air is given in simple terms and is followed by a brief report on the level to which architectural acoustic models require to be dried in order to match the air absorption in their full-sized counterpart. The level must necessarily be a compromise depending on the ultrasonic range of frequencies used in the model and is further complicated by the lack of absorption data available at very low percentage values of relative humidity.  相似文献   

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