Inter-channel Time Difference Research Articles

Multi-Sound-Source Localization Using Machine Learning for Small Autonomous Unmanned Vehicles with a Self-Rotating Bi-Microphone Array

While vision-based localization techniques have been widely studied for small autonomous unmanned vehicles (SAUVs), sound-source localization capabilities have not been fully enabled for SAUVs. This paper presents two novel approaches for SAUVs to perform three-dimensional (3D) multi-sound-sources localization (MSSL) using only the inter-channel time difference (ICTD) signal generated by a self-rotating bi-microphone array. The proposed two approaches are based on two machine learning techniques viz., Density-Based Spatial Clustering of Applications with Noise (DBSCAN) and Random Sample Consensus (RANSAC) algorithms, respectively, whose performances were tested and compared in both simulations and experiments. The results show that both approaches are capable of correctly identifying the number of sound sources along with their 3D orientations in a reverberant environment.

Localization Uncertainty in Time-Amplitude Stereophonic Reproduction

This article studies the effects of inter-channel time and level differences in stereophonic reproduction on perceived localization uncertainty, which is defined as how difficult it is for a listener to tell where a sound source is located. Towards this end, a computational model of localization uncertainty is proposed first. The model calculates inter-aural time and level difference cues, and compares them to those associated to free-field point-like sources. The comparison is carried out using a particular distance functional that replicates the increased uncertainty observed experimentally with inconsistent inter-aural time and level difference cues. The model is validated by formal listening tests, achieving a Pearson correlation of 0.99. The model is then used to predict localization uncertainty for stereophonic setups and a listener in central and off-central positions. Results show that amplitude methods achieve a slightly lower localization uncertainty for a listener positioned exactly in the center of the sweet spot. As soon as the listener moves away from that position, the situation reverses, with time-amplitude methods achieving a lower localization uncertainty.

Dual Microphone Voice Activity Detection Based on Reliable Spatial Cues.

Two main spatial cues that can be exploited for dual microphone voice activity detection (VAD) are the interchannel time difference (ITD) and the interchannel level difference (ILD). While both ITD and ILD provide information on the location of audio sources, they may be impaired in different manners by background noises and reverberation and therefore can have complementary information. Conventional approaches utilize the statistics from all frequencies with fixed weight, although the information from some time–frequency bins may degrade the performance of VAD. In this letter, we propose a dual microphone VAD scheme based on the spatial cues in reliable frequency bins only, considering the sparsity of the speech signal in the time–frequency domain. The reliability of each time–frequency bin is determined by three conditions on signal energy, ILD, and ITD. ITD-based and ILD-based VADs and statistics are evaluated using the information from selected frequency bins and then combined to produce the final VAD results. Experimental results show that the proposed frequency selective approach enhances the performances of VAD in realistic environments.

The Reduction of Vertical Interchannel Crosstalk: The Analysis of Localisation Thresholds for Natural Sound Sources

In subjective listening tests, natural sound sources were presented to subjects as vertically-oriented phantom images from two layers of loudspeakers, ‘height’ and ‘main’. Subjects were required to reduce the amplitude of the height layer until the position of the resultant sound source matched that of the same source presented from the main layer only (the localisation threshold). Delays of 0, 1 and 10 ms were applied to the height layer with respect to the main, with vertical stereophonic and quadraphonic conditions being tested. The results of the study showed that the localisation thresholds obtained were not significantly affected by sound source or presentation method. Instead, the only variable whose effect was significant was interchannel time difference (ICTD). For ICTD of 0 ms, the median threshold was −9.5 dB, which was significantly lower than the −7 dB found for both 1 and 10 ms. The results of the study have implications both for the recording of sound sources for three-dimensional (3D) audio reproduction formats and also for the rendering of 3D images.

Vertical Stereophonic Localization in the Presence of Interchannel Crosstalk: the Analysis of Frequency-Dependent Localization Thresholds

Listening tests were conducted in order to investigate the frequency dependency of localization thresholds in relation to vertical interchannel crosstalk. Octave band and broadband pink noise stimuli were presented to subjects as phantom images from vertically arranged stereophonic loudspeakers located directly in front of the listening position. With respect to the listening position the lower loudspeaker was not elevated; the upper loudspeaker was elevated by 30◦. Subjects completed a method of adjustment task in which they were required to reduce the amplitude of the upper loudspeaker until the resultant phantom image matched the position of the same stimulus presented from the lower loudspeaker alone. The upper loudspeaker was delayedwith respect to the lower by 0, 0.5, 1, 5, and 10 ms. The experimental data demonstrated that the main effect of frequency on the localization threshold was significant, with the low frequency stimuli (125 and 250 Hz) requiring significantly less level reduction (less than 6 dB) than the mid-high (1, 2, and 8 kHz) frequency stimuli (9–10.5 dB reduction). The main effect of interchannel time difference (ICTD) on the localization thresholds for each octave band was found to be non-significant. For all stimuli interchannel level difference (ICLD) was always necessary, indicating that the precedence effect is not a feature of median plane localization.

Frequency-Dependent Amplitude Panning for the Stereophonic Image Enhancement of Audio Recorded Using Two Closely Spaced Microphones

In this paper, we propose a new frequency-dependent amplitude panning method for stereophonic image enhancement applied to a sound source recorded using two closely spaced omni-directional microphones. The ability to detect the direction of such a sound source is limited due to weak spatial information, such as the inter-channel time difference (ICTD) and inter-channel level difference (ICLD). Moreover, when sound sources are recorded in a convolutive or a real room environment, the detection of sources is affected by reverberation effects. Thus, the proposed method first tries to estimate the source direction depending on the frequency using azimuth-frequency analysis. Then, a frequency-dependent amplitude panning technique is proposed to enhance the stereophonic image by modifying the stereophonic law of sines. To demonstrate the effectiveness of the proposed method, we compare its performance with that of a conventional method based on the beamforming technique in terms of directivity pattern, perceived direction, and quality degradation under three different recording conditions (anechoic, convolutive, and real reverberant). The comparison shows that the proposed method gives us better stereophonic images in a stereo loudspeaker reproduction than the conventional method without any annoying effects.

The Effect of Interchannel Time Difference on Localization in Vertical Stereophony

Listening tests were conducted in order to analyse the localisation of band-limited stimuli in vertical stereophony. The test stimuli were seven octave bands of pink noise, with centre frequencies ranging from 125–8000Hz, as well as broadband pink noise. Stimuli were presented from vertically arranged loudspeakers either monophonically or as vertical phantom images, created with the upper loudspeaker delayed with respect to the lower by 0, 0.5, 1, 5 and 10ms (i.e. interchannel time difference). The experimental data obtained showed that localisation under the aforementioned conditions is generally governed by the so-called “pitch-height” effect, with the high frequency stimuli generally being localised significantly higher than the low frequency stimuli for all conditions. The effect of interchannel time difference was found to be significant on localisation judgments for both the 1000-4000Hz octave bands and the broadband pink noise; it is suggested that this was related to the effects of comb filtering. Additionally, no evidence could be found to support the existence of the precedence effect in vertical stereophony.

Time-Shifting Based Primary-Ambient Extraction for Spatial Audio Reproduction

One of the key issues in spatial audio analysis and reproduction is to decompose a signal into primary and ambient components based on their directional and diffuse spatial features, respectively. Existing approaches employed in primary-ambient extraction (PAE), such as principal component analysis (PCA), are mainly based on a basic stereo signal model. The performance of these PAE approaches has not been well studied for the input signals that do not satisfy all the assumptions of the stereo signal model. In practice, one such case commonly encountered is that the primary components of the stereo signal are partially correlated at zero lag, referred to as the primary-complex case. In this paper, we take PCA as a representative of existing PAE approaches and investigate the performance degradation of PAE with respect to the correlation of the primary components in the primary-complex case. A time-shifting technique is proposed in PAE to alleviate the performance degradation due to the low correlation of the primary components in such stereo signals. This technique involves time-shifting the input signal according to the estimated inter-channel time difference of the primary component prior to the signal decomposition using conventional PAE approaches. To avoid the switching artifacts caused by the varied time-shifting in successive time frames, overlapped output mapping is suggested. Based on the results from our experiments, PAE approaches with the proposed time-shifting technique are found to be superior to the conventional PAE approaches in terms of extraction accuracy and spatial accuracy.

Three-dimensional Sound Source Localization Using Inter-channel Time Difference Trajectory

Sound source localization is one of the basic and essential techniques for intelligent robots in terms of human-robot interaction and has been utilized in various engineering fields. This paper suggests a new localization method using an inter-channel time difference trajectory, which is a new localization cue for efficient 3-D localization. As one of the ways to realize the proposed cue, a two-channel rotating array is employed. Two microphones are attached on the left and right sides of the spherical head. One microphone is in a circular motion on the right side, while the other is fixed on the left side. According to the rotating motion of the array, the (source) direction-dependent characteristics of the trajectories are analysed using the Ray-Tracing formula extended for 3-D models. In simulation, the synthesized signals generated by the fixed and rotating microphone signal models were used as the output signals of the two microphones. The simulation showed that the localization performance is strong...

Estimation of multiple sound source directions using artificial robot ears

Estimating the direction of a sound source is an important technique used in various engineering fields, including intelligent robots and surveillance systems. In a household where a user’s voice and noises emitted from electric appliances originate from arbitrary directions in 3-D space, robots need to recognize the directions of multiple sound sources in order to effectively interact with the user.This paper proposes an ear-based estimation (localization) system using two artificial robot ears, each consisting of a spiral-shaped pinna and two microphones, for application in humanoid robots. Four microphones are asymmetrically placed on the left and right sides of the head. The proposed localization algorithm is based on a spatially mapped generalized cross-correlation function which is transformed from the time domain to the space domain by using a measured inter-channel time difference map. For validation of the proposed localization method, two experiments (single- and multiple-source cases) were conducted using male speech. In the case of a single source, with the exception of laterally biased sources, the localization was achieved with an error of less than 10°. In a multiple-source environment, one source was fixed at the front side and the other source changed its direction; from the experimental results, the error rates on the localization of the fixed and varying sources are 0% and 36.9% respectively within an error bound of 15°.

Analysis and Design of Multichannel Systems for Perceptual Sound Field Reconstruction

This paper presents a systematic framework for the analysis and design of circular multichannel surround sound systems. Objective analysis based on the concept of active intensity fields shows that for stable rendition of monochromatic plane waves it is beneficial to render each such wave by no more than two channels. Based on that finding, we propose a methodology for the design of circular microphone arrays, in the same configuration as the corresponding loudspeaker system, which aims to capture inter-channel time and intensity differences that ensure accurate rendition of the auditory perspective. The methodology is applicable to regular and irregular microphone/speaker layouts, and a wide range of microphone array radii, including the special case of coincident arrays which corresponds to intensity-based systems. Several design examples, involving first and higher-order microphones are presented. Results of formal listening tests suggest that the proposed design methodology achieves a performance comparable to prior art in the center of the loudspeaker array and a more graceful degradation away from the center.

Virtual Sound Rendering in a Stereophonic Loudspeaker Setup

This paper presents a mathematical analysis of the effects of interchannel amplitude and time differences in two channel (stereophonic) sound systems. The analysis is developed by computing the acoustic conditions at the listener's ears as a function of the stereophonic signal feature. We also present separate approximations of head-related transfer function (HRTF)-based panning according to predefined frequency bands. We attempt to create non-sophisticated models of the stereophonic listening mechanism with approximations in both the time and frequency domains. The models are based upon psychoacoustical theories that present the frequency-dependent relative importance of acoustical cues. Based on the model, we propose new panning methods that can enhance the localization accuracy of conventional panning methods, such as amplitude panning and HRTF-based panning. The localization performances of the new panning techniques are evaluated and compared by means of auditory model simulations and listening tests. Through simulations and listening test results, it is shown that the proposed panning method makes substantial improvements in the localization of virtual sources.

Estimation of Interchannel Time Difference in Frequency Subbands Based on Nonuniform Discrete Fourier Transform

Estimation of Interchannel Time Difference in Frequency Subbands Based on Nonuniform Discrete Fourier Transform

Estimation of Interchannel Time Difference in Frequency Subbands Based on Nonuniform Discrete Fourier Transform

Binaural cue coding (BCC) is an efficient technique for spatial audio rendering by using the side information such as interchannel level difference (ICLD), interchannel time difference (ICTD), and interchannel correlation (ICC). Of the side information, the ICTD plays an important role to the auditory spatial image. However, inaccurate estimation of the ICTD may lead to the audio quality degradation. In this paper, we develop a novel ICTD estimation algorithm based on the nonuniform discrete Fourier transform (NDFT) and integrate it with the BCC approach to improve the decoded auditory image. Furthermore, a new subjective assessment method is proposed for the evaluation of auditory image widths of decoded signals. The test results demonstrate that the NDFT-based scheme can achieve much wider and more externalized auditory image than the existing BCC scheme based on the discrete Fourier transform (DFT). It is found that the present technique, regardless of the image width, does not deteriorate the sound quality at the decoder compared to the traditional scheme without ICTD estimation.

Estimation of Interchannel Time Difference in Frequency Subbands Based on Nonuniform Discrete Fourier Transform

Estimation of Interchannel Time Difference in Frequency Subbands Based on Nonuniform Discrete Fourier Transform

Localization of Epileptogenic Foci Using a Simple Reference-Subtraction Montage to Document Small Interchannel Time Differences

We present a modified EEG montage that detects small interchannel time differences and assists in localizing the epileptogenic focus. Regions with apparently synchronous epileptic discharges are displayed simultaneously in referential and subtraction derivations. The subtraction derivation is a bipolar configuration of two regions of interest that are not necessarily adjacent. The referential derivation reveals the polarity, voltage, and morphology of the two discharges, and the subtraction derivation detects asynchrony; the combined reference-subtraction derivation thus indicates the region that is activated first.

Perceptive deafness for low tones

In an attempt to locate lesions in the auditory system, the author examined 46 cases of perceptive low tone deafness by various methods of hearing were bilaterally.involved and five unilaterally. Hearing was estimated by performing the tment test, temporaly threshold shift test using a Bekesy type audiometer, distorted speech perception test with use of 1200 cps low pass and 1700 cps high pass filters, and binaural hearing test using an interchannel time difference (ITD) apparatus. The vestibular function was also examined by observing spontaneous, positional and positioning netic pattern tests. Differentiation between the peripheral and the retrolabyrinthine lesions was anticipated from the results.It was concluded that the lesion in the patient of perceptive low tone deafness was peripheral, retrolabyrinthine or both, suggesting a systemic disease in the auditory system.

Some Measurements on the Effects of Interchannel Intensity and Time Differences in Two Channel Sound Systems

The results of some practical measurements on the effects of interchannel intensity and time differences in two channel (stereophonic) sound systems are presented. The effects of alterations in the listener position are also covered. The test signals used ranged from single component tones to wide and narrow band noise and included running speech. A theory based on the assumption that the brain is sensitive to interaural time difference and its variation with head movement is developed and is shown to be in reasonable agreement with the practical findings.