Discriminality of statistically independent Gaussian noise tokens and random tone-burst complexes

Hanna (1984) has shown that noise tokens with a duration of 400 ms are harder to discriminate than noise tokens of 100 ms. This is remarkable because a 400-ms stimulus potentially contains four times as much information for judging dissimilarity than the 100-ms stimulus. Apparently, the ability to use all information in a stimulus is impaired by some kind of limitation, e.g. a memory limitation (cf. Cowan 2000) or a limitation in the ability to allocate attentional resources (cf. Kidd and Watson 1992). In a first experiment, this study examined the influence of stimulus duration and bandwidth of Gaussian noise tokens on the ability to perform an auditory discrimination task. In a second experiment, the amount of potential information in a stimulus was decoupled from its duration in order to more carefully examine the properties of the memory or attention limitation that results in the discrimination impairment. Finally, a computational model that limits the amount of perceptual information is introduced as an attempt to model the findings of the first and second experiment

A probabilistic model for robust acoustic localization based on an auditory front-end

Although extensive research has been done in the field of localization, the degrading effect of reverberation and the presence of multiple sources on localization performance has remained a major issue. The classical approach to localize an acoustic source in the horizontal space is to search for the main peak in the cross-correlation function, which corresponds to the interaural time difference (ITD) between both ears. Apart from ITD, the interaural level difference (ILD) can contribute to localization, especially at higher frequencies where the wavelength becomes smaller than the diameter of the head, leading to ambiguous ITD information. Motivated by the robust localization performance of the human auditory system, its peripheral stage is used as a front-end for binaural cue extraction. The interdependency of ITD and ILD on azimuth is a complex pattern that depends also on the room acoustics and is therefore learned by azimuth-dependent Gaussian mixture models. Multiconditional training is performed to incorporate the spread of the binaural features caused by multiple sources and the effect of reverberation. The trained localization model outperforms state-of-the-art localization techniques in simulated adverse acoustic conditions. Furthermore, the model is capable of generalizing to changes in the simulated room absorption and to unknown source/receiver combinations

Model simulations of masked thresholds for tones in dichotic noise maskers (A)

The study of masked thresholds in dichotic noise maskers is important for understanding the processing in binaural hearing. To simulate these thresholds a psychoacoustically motivated perception model was used [T. Dau et al. (1995). ``A quantitative model of the ``effective'' signal processing in the auditory system: I. Model structure,'' submitted to J. Acoust. Soc. Am.]. This model, which has been successfully applied to several monaural psychoacoustical experiments, was extended by an additional binaural processing unit. It consists of a filterbank, half-wave rectifier, low-pass filter, and adaptation loops, which model the temporal processing. The binaural processing unit detects the interaural correlation and makes decisions based on the difference between the signals from both ears. Masked thresholds in the NoS and NSo configurations, obtained as a function of noise masker frequency and bandwidth, were simulated and compared to new experimental measurements. The dependence on interaural delay and interaural decorrelation of the noise masker was also modeled and compared to data in the literature. In general, model simulations agree well with the main features seen in the measurements. [Work supported by DFG (Ho 1627/1-1) and by NIDCD (Grant DC00100).

Parametric coding of stereo audio

Parametric-stereo coding is a technique to efficiently code a stereo audio signal as a monaural signal plus a small amount of parametric overhead to describe the stereo image. The stereo properties are analyzed, encoded, and reinstated in a decoder according to spatial psychoacoustical principles. The monaural signal can be encoded using any (conventional) audio coder. Experiments show that the parameterized description of spatial properties enables a highly efficient, high-quality stereo audio representation

Binaural detection with spectrally nonoverlapping signals and maskers: evidence for masking by aural distortion products

Thresholds were measured for diotic tonal signals in the presence of interaurally delayed bands of Gaussian noise. When the signal frequency was 525 Hz, the spectrum of the noise was either below (highest frequency, 450 Hz) or above (lowest frequency, 600 Hz) the frequency of the signal. When the signal frequency was 450 Hz, the spectrum of the noise was always above the signal frequency (lowest frequency, 600 Hz). Signals had a 250-ms duration and were temporally centered within the 300-ms long bursts of noise. The spectrum level of the noise was 60 dB. Thresholds obtained in all three conditions varied essentially sinusoidally with the interaural delay of the noise. For signals below the spectrum of the noise, the periodicities within the data were close to, but not identical with, the periodicities of the signals. This outcome is discussed in terms of masking produced by aural distortion products stemming from interactions within the bands of noise [cf. van der Heijden and Kohlrausch, J. Acoust. Soc. Am. 98, 3125–3134 (1995)]. For signals above the spectrum of the noise, the periodicities in the data suggested that masking was produced by components within the band of noise. Patterns within the data are also discussed in terms of limitations concerning the magnitude of external delays that can be matched by internal delays that are incorporated in modern models of binaural processing

Bit-rate saving in multichannel sound:using a band-limited channel to transmit the center signal

A method is proposed to achieve full-frequency-range three-channel (left, right, and center) sound reproduction in systems that have only two full-range sound channels and some band-limited commentary channels. The low-frequency part of the center signal, which matches the bandwidth of the commentary channels, is added to the (multilingual) speech signals in each of the commentary channels. The remaining high-frequency part is added in the left and right channels as in conventional mixdowns. Sound reproduction of this signal by a conventional two-channel receiver remains unaltered. The low-frequency part of the center signal is mixed to the left and right signals together with the speech once the user has selected a commentary channel. Three-channel reproduction is obtained by routing the selected commentary channel to a central loudspeaker. Listening tests revealed that sound reproduction according to the proposed scheme could not be distinguished from original three-channel reproduction. This scheme can be applied to proposed standards such as D2MAC and MPEG2.</p