Stereo Channels Research Articles

Predicting virtual acoustic source properties in a stereophonic sound-field using the coherence properties of the acoustic velocity vector field

It is widely understood that the mutual coherence properties, including the relative phase, of the channel signals in a stereophonic sound reproduction system significantly influence a listener’s perception of the resulting virtual acoustic sources. This talk presents a quantitative model that combines wavefield coherence theory, the acoustic velocity vector, and known perceptual cues to make quantitative predictions of the perceived location and spatial spread of horizontally placed stereophonic virtual sources as a function of the mutual coherence properties of the stereo channel signals. It will be seen that a small number of parameters, including the intrinsic coherence time of the source signal, the relative phase of the channels, and perceptual averaging times, account for most of the variation in the cues that determine virtual acoustic source localization. The output of the coherence model will be compared to the results of earlier experiments, including ones investigating the “phaseyness” problem and precedence effect. Additionally, the potential of this model to account for reverberation in the listening space will be demonstrated.

Design of a compact triple-channel panoramic stereo imaging system.

We propose a compact form of the triple-channel panoramic annular lens (PAL) with a stereo field and no central blind area based on polarization technology, which solves the problem that the traditional stereo panoramic system always has a large and complex mirror in front to reflect light. Based on the traditional dual-channel structure, we apply polarization technology to the first reflective surface to create a third channel for the stereovision. The field of view (FoV) of the front channel is 360° × (0° - 40°), the FoV of the side channel is 360° × (40° - 105°) and the stereo FoV is 360° × (20° - 50°). The Airy radii of the front channel, the side channel, and the stereo channel are 3.374 µm, 3.372 µm, and 3.360 µm, respectively. The modulation transfer function at 147 lp/mm is greater than 0.13 in the front and stereo channels and greater than 0.42 in the side channel. The F - θ distortion of all FoVs is less than 10%. This system shows a promising way to achieve stereovision without adding complex structures on the original basis.

Shadow method retrievals of the atmospheric optical depth above Gale crater on Mars using HRSC images

The ‘Shadow Method’ is a tool to estimate the Atmospheric Optical Depth (AOD) on Mars from the brightness of shadows. This method is derived from the equations of radiative transfer, but there are several important simplifications that together invoke errors of several tens of percent. Work by us and by others show that these errors are largely systematic and can be minimized by adding an empirical ‘Correction-Factor’ to the equations. To quantify this factor, we compared shadow method retrievals from orbiter images with in-situ measurements by the Curiosity rover. We analysed a set of seven images that was taken by DLR's High Resolution Stereo Camera (HRSC) on-board ESA's orbiter Mars Express. The images were taken in seven channels: NIR, red, green, blue, and in three panchromatic stereo channels S1, Nadir, and S2. All these images show Gale crater and the exploration site of the Curiosity rover therein. Comparing the rover measurement of the AOD with shadow method retrievals from 5 close-by regions yielded the following correction factors: 0.53 ± 0.03 for Nadir, 0.60 ± 0.04 for S1, 0.62 ± 0.03 for S2, 0.66 ± 0.03 for NIR, 0.64 ± 0.03 for Red, 0.55 ± 0.03 for Green, and 0.57 ± 0.03 for Blue.We analysed 71 regions with varying altitudes between −4.6 km to +3.8 km and found that, on an average, the AOD decreases with increasing altitude. The available channels yielded the following averages of atmospheric optical depth before applying the correction factor: Nadir: 0.47; S1: 0.46; S2: 0.52; Red: 0.49; Green: 0.53; Blue: 0.57 and NIR: 0.44 while after applying the correction factor we obtain Nadir: 0.89; S1: 0.77; S2: 0.84; Red: 0.77; Green: 0.96; Blue: 1.0 and NIR: 0.67. The Curiosity rover, at an altitude of −4.87 km, measured a ground-truth AOD of 0.88.The shadowed and sunlit comparison regions that were used for this work range in altitudes from −4.6 km to +3.8 km. The AOD decreases with altitude and the decrease yields the following scale heights: S1: 12.5 + 0.48/−0.35 km; nadir: 11.1 + 0.53/−0.33 km; S2: 14.3 + 0.48/−0.44 km; NIR: 11.1 + 0.56/−0.47 km; red: 14.3 + 0.52/−0.49 km; green: 15.6 + 0.62/−0.59 km; blue: 14.3 + 0.44/−0.42 km. For this area and around that time of day, the Mars Climate Database predicts a pressure scale height of 11.6–12.2 km, which agrees well with the scale heights that we derived for Nadir, S1 and NIR. One region (number 42) contained a cloud with an optical depth of up to 0.6—0.8; i.e., around one third of the total AOD of 1.9 ± 0.01 measured for this area. Region 29 showed the highest AOD of our sample.

Simulation of Images and Digital Terrain Models for the Mission BepiColombo

BepiColombo mission will provide Digital Terrain Models of the surface of Mercury by means of the stereo channel of the SIMBIO-SYS (Spectrometer and Imaging for MPO BepiColombo Integrated Observatory SYStem) imaging package onboard. The work here described presents a novel approach for the creation of higher resolution stereo products using the high-resolution channel of SIMBIO-SYS. Being the camera rigidly integrated with the spacecraft, this latter must be tilted to acquire stereo pairs necessary for the 3D reconstruction. A new method for image simulation and stereo reconstruction is presented in this work, where the input data are chosen as closely as possible to the real mission parameters. Different simulations are executed changing the illumination conditions and the stereo angles. The Digital Terrain Models obtained are evaluated and an analysis of the best acquisition conditions is performed, helping to improve the image acquisition strategy of BepiColombo mission. In addition, a strategy for the creation of a mosaic from different images acquired with the high-resolution channel of SIMBIO-SYS is explained, giving the possibility to obtain tridimensional products of extended targets.

Generating Artificial Reverberation via Genetic Algorithms for Real-Time Applications.

We introduce a Virtual Studio Technology (VST) 2 audio effect plugin that performs convolution reverb using synthetic Room Impulse Responses (RIRs) generated via a Genetic Algorithm (GA). The parameters of the plugin include some of those defined under the ISO 3382-1 standard (e.g., reverberation time, early decay time, and clarity), which are used to determine the fitness values of potential RIRs so that the user has some control over the shape of the resulting RIRs. In the GA, these RIRs are initially generated via a custom Gaussian noise method, and then evolve via truncation selection, random weighted average crossover, and mutation via Gaussian multiplication in order to produce RIRs that resemble real-world, recorded ones. Binaural Room Impulse Responses (BRIRs) can also be generated by assigning two different RIRs to the left and right stereo channels. With the proposed audio effect, new RIRs that represent virtual rooms, some of which may even be impossible to replicate in the physical world, can be generated and stored. Objective evaluation of the GA shows that contradictory combinations of parameter values will produce RIRs with low fitness. Additionally, through subjective evaluation, it was determined that RIRs generated by the GA were still perceptually distinguishable from similar real-world RIRs, but the perceptual differences were reduced when longer execution times were used for generating the RIRs or the unprocessed audio signals were comprised of only speech.

Speech-Based Activity Recognition for Trauma Resuscitation.

We present a speech-based approach to recognize team activities in the context of trauma resuscitation. We first analyzed the audio recordings of trauma resuscitations in terms of activity frequency, noise-level, and activity-related keyword frequency to determine the dataset characteristics. We next evaluated different audio-preprocessing parameters (spectral feature types and audio channels) to find the optimal configuration. We then introduced a novel neural network to recognize the trauma activities using a modified VGG network that extracts features from the audio input. The output of the modified VGG network is combined with the output of a network that takes keyword text as input, and the combination is used to generate activity labels. We compared our system with several baselines and performed a detailed analysis of the performance results for specific activities. Our results show that our proposed architecture that uses Mel-spectrum spectral coefficients features with a stereo channel and activity-specific frequent keywords achieve the highest accuracy and average F1-score.

Individualized Treatment of Tinnitus during Sleep Using Combined Tinnitus Signal and Music

Introduction: Tinnitus is a widely seen otological symptom that interferes with daily activities and causes discomfort. Tinnitus treatments can be classified into 4 main groups: pharmacological treatments, cognitive and behavioral therapy, psychological treatments, and combined treatment approaches made up of at least 2 of these 3 treatment methods. Objective: The aim of this study was to assess whether it would be possible to develop an individualized treatment method of tinnitus by application of a combined tinnitus signal and music during sleep. Methods: Forty-three ears of 30 patients who had subjective tinnitus were included. The patients were evaluated using Tinnitus Handicap Inventory, Visual Analogue Scale, and Beck Depression Inventory. The psychoacoustic parameters of tinnitus, such as tinnitus frequency and loudness, and minimal masking levels, were determined. The patients were asked to select musical melodies that they liked. The tinnitus frequency of each patient was taken as the central frequency according to ANSI 2004. All sound files were prepared as stereo channels, with 16-bit resolution and 44,100 Hz sampling rate. The root mean square power value of the music and the band noise’s average root mean square power value were equalized with the “Amplification” command, and 70% of the music and 30% of wide/narrow-band noise were mixed as a stereo channel by the “Mix Paste” command. The patients were instructed to listen to that individualized music/narrow-band noise (tinnitus signal) for 2 h during sleep for a duration of 6 months. Results: Tinnitus frequencies of the patients measured prior to treatment and at the second, fourth, and sixth months of follow-up were not significantly different. A statistically significant decrease was seen in tinnitus loudness, minimal masking levels, and residual inhibition during the follow-up. Tinnitus Handicap Inventory scores decreased significantly during follow-up, and the number of patients who complained of tinnitus decreased (p < 0.05). The Visual Analogue Scale scores significantly decreased during follow-up (p < 0.05). Beck Depression Inventory scores decreased significantly during follow-up (p < 0.05). Conclusion: Stimulation of the auditory and limbic systems during sleep by the tinnitus signal combined with individualized musical melodies seems an alternative, effective, and cheap method in the treatment of tinnitus.

The psychophysics of stereopsis can be explained without invoking independent ON and OFF channels.

Early vision proceeds through distinct ON and OFF channels, which encode luminance increments and decrements respectively. It has been argued that these channels also contribute separately to stereoscopic vision. This is based on the fact that observers perform better on a noisy disparity discrimination task when the stimulus is a random-dot pattern consisting of equal numbers of black and white dots (a “mixed-polarity stimulus,” argued to activate both ON and OFF stereo channels), than when it consists of all-white or all-black dots (“same-polarity,” argued to activate only one). However, it is not clear how this theory can be reconciled with our current understanding of disparity encoding. Recently, a binocular convolutional neural network was able to replicate the mixed-polarity advantage shown by human observers, even though it was based on linear filters and contained no mechanisms which would respond separately to black or white dots. Here, we show that a subtle feature of the way the stimuli were constructed in all these experiments can explain the results. The interocular correlation between left and right images is actually lower for the same-polarity stimuli than for mixed-polarity stimuli with the same amount of disparity noise applied to the dots. Because our current theories suggest stereopsis is based on a correlation-like computation in primary visual cortex, this postulate can explain why performance was better for the mixed-polarity stimuli. We conclude that there is currently no evidence supporting separate ON and OFF channels in stereopsis.

SIMBIO-SYS/STC stereo camera calibration: Geometrical distortion.

The STereo imaging Channel (STC) is the first push-frame stereo camera on board an European Space Agency (ESA) satellite, i.e., the ESA-Japan Aerospace eXploration Agency mission BepiColombo. It was launched in October 2018, and it will reach its target, Mercury, in 2025. The STC main aim is to provide the global three-dimensional reconstruction of the Mercury surface. STC, the stereo channel of spectrometer and imagers for Mercury Planetary Orbiter BepiColombo-Integrated Observatory System, is based on an original optical design that incorporates the advantages of a compact unique detector instrument and the convenience of a double direction acquisition system. In fact, STC operates in a push-frame imaging mode and its two optical sub-channels will converge the incoming light on a single focal plane assembly allowing to minimize mass and volume. The focal plane of the instrument is housing six different filters: two panchromatic filters in the range 600-800 nm and four broadband filters with central wavelengths in the range 420-920 nm. In this paper, the geometrical calibration of the instrument, including the optical setups used, will be described. The methods used to derive the focal lengths, the boresights, and the reference systems of the different filter models are presented, and the related distortion results are discussed. The STC off-axis configuration forced to develop a distortion map model based on the RFM (rational function model). In contrast to other existing models, which allow linear estimates, the RFM is not referred to specific lens geometry, but it is sufficiently general to model a variety of distortion types, as it will be demonstrated in this particular case.

Production method for simulcast in 22.2 multichannel sound broadcasting

The 8K with 22.2 multichannel (22.2 ch) sound broadcasting is planned to be launched since December 2018 in Japan. NHK is going to simulcast 2-channel stereo and 5.1 surround together with the 22.2 ch sound for backward compatibility with existing home reproduction equipment, e.g., AV receiver and soundbar. However, individual production for each audio format is impractical because it needs many engineers and a lot of production equipment compared with a production in single format. Hence, an efficient production method for simulcast which does not demand additional production resources is required. The first proposal is a tone compensation for downmixing from the 22.2 ch sound to the other audio formats. The energy spectrum of the downmixed signal is compensated to harmonize with that of the 22.2 ch sound. The second proposal is a loudness chase to have the loudness level of the downmixed signal follow that of the 22.2 ch sound. The loudness level of the downmixed signal usually differs from that of the 22.2 ch sound, so that the loudness chase successively adjusts the loudness level without unnatural level change and matches them by the end of the program.

Benefits and Limitations of Common Directional Microphones in Real-World Sounds

We present extensive experimental data to objectively evaluate the benefits and limitations of common directional microphones in real-world sound fields. The microphones include a conventional directional microphone(DM), a balanced DM, etc., plus the Omni microphone (mic) as a benchmark. The evaluation focuses on noise outputs, signal-to-noise ratios(S/Ns) and distortions; the real-world sounds include male voices, female voices, babble noises, white noises and talking interferences. Each type of noises is at 4 or 5 levels, from 30 to 70 dB SPL, at 10 dB step, and each talking interference is at 3 levels: 50, 60 and 70 dB SPL. The research methods include analytically deriving sensitivity-gains, statistically calculating the three mics’ outputs, experimentally viewing waveforms and spectra, and using large-sample wave files for a high confidence level. According to the experimental results, this paper concludes that 1) for a conversation in a quiet field, in soft or low noise field, the common DMs achieve comfortable S/Ns: 7 to 33 dB, similar to what the Omni mic does; 2) for a conversation in low, competing or strong talking interference fields, the common DMs achieve about 16 dB better S/N than the Omni mic does; 3) for a conversation in competing or strong surrounding noise field, the common DMs do not achieve beneficial S/N to understand speech; the common DMs’ noises are close to the Omni mic noise; 4) in various experiments, the balanced DM preserve speech fidelity well as the Omni mic does, while the conventional DM does poorly. This paper further introduces the Simulink experimental manipulations, such as digital FIR filters’ design, stereo channels’ wave files creation, etc., in the Appendix.

Thresholds for sine-wave corrugations defined by binocular disparity in random dot stereograms: Factor analysis of individual differences reveals two stereoscopic mechanisms tuned for spatial frequency

Threshold functions for sinusoidal depth corrugations typically reach their minimum (highest sensitivity) at spatial frequencies of 0.2–0.4 cycles/degree (cpd), with lower thresholds for horizontal than vertical corrugations at low spatial frequencies. To elucidate spatial frequency and orientation tuning of stereoscopic mechanisms, we measured the disparity sensitivity functions, and used factor analytic techniques to estimate the existence of independent underlying stereo channels. The data set (N = 30 individuals) was for horizontal and vertical corrugations of spatial frequencies ranging from 0.1 to 1.6 cpd. A principal component analysis of disparity sensitivities (log-arcsec) revealed that two significant factors accounted for 70% of the variability. Following Varimax rotation to approximate “simple structure”, one factor clearly loaded onto low spatial frequencies (≤0.4 cpd), and a second was tuned to higher spatial frequencies (≥0.8 cpd). Each factor had nearly identical tuning (loadings) for horizontal and vertical patterns. The finding of separate factors for low and high spatial frequencies is consistent with previous studies. The failure to find separate factors for horizontal and vertical corrugations is somewhat surprising because the neuronal mechanisms are believed to be different. Following an oblique rotation (Direct Oblimin), the two factors correlated significantly, suggesting some interdependence rather than full independence between the two factors.

Headphone screening to facilitate web-based auditory experiments.

Psychophysical experiments conducted remotely over the internet permit data collection from large numbers of participants but sacrifice control over sound presentation and therefore are not widely employed in hearing research. To help standardize online sound presentation, we introduce a brief psychophysical test for determining whether online experiment participants are wearing headphones. Listeners judge which of three pure tones is quietest, with one of the tones presented 180° out of phase across the stereo channels. This task is intended to be easy over headphones but difficult over loudspeakers due to phase-cancellation. We validated the test in the lab by testing listeners known to be wearing headphones or listening over loudspeakers. The screening test was effective and efficient, discriminating between the two modes of listening with a small number of trials. When run online, a bimodal distribution of scores was obtained, suggesting that some participants performed the task over loudspeakers despite instructions to use headphones. The ability to detect and screen out these participants mitigates concerns over sound quality for online experiments, a first step toward opening auditory perceptual research to the possibilities afforded by crowdsourcing.

Sosialisasi Panduan Elektronis Pembelajaran Tajwid Berbasis Macromedia Flash (Tempat : Pondok Pesantren Modern Islam Assalaam Surakarta)

The purpose of this research is to develop and test an electronic learning system asan alternative learning media to read Al-Qur’an. The creation of this learning systemhas through some steps. The steps are references searching, data collecting, systematic creating, and testing. This process involves data collecting from tajwid books and video recording. The capacity of this system is 179 MB including 1.655 files consisting of 1 .exe file. 769 .swf files, 734 .flv video files, 151 mp3 files, and 3 folders. The resulted video has an extention .flv which has a size as big as 250 x 180 pixel. Frame Rate 15 fps, and Bit Rate 260 kbps. The resulted audio extents mp3 that has a Bit Rate 128 kbps, stereo channel and resolusion 16 bit. The questioners that are devided to the 20 users show that this programm is better and completer than tajwid books sold.

Stereo channel music signal separation based on non-negative tensor factorization with cepstrum regularization

Music signals are usually generated by mixing many music source signals, such as various instrumental sounds and vocal sounds, and they are often represented as 2-channel signals (i.e., stereo channel signals). Underdetermined source separation for separating the music signals into individual music source signals is a potential technique to develop various applications, such as music transcription, singer discrimination, and vocal extraction. One of the most powerful underdetermined source separation methods is Nonnegative Matrix Factorization (NMF) that models a power spectrogram of an observation signal as a product of two nonnegative matrices; basis and activation matrices. To apply NMF to the stereo channel music signal separation, we have proposed Nonnegative Tensor Factorization (NTF) by further implementing a gain matrix to represent mixing (i.e., panning) information. However, the separation performance of this method is insufficient owing to less prior information to model acoustic characteristics of the individual music source signals. To address this issue, we propose a cepstrum regularization method for NTF to make power spectral envelopes of the separated signals close to those of individual music source signals. We conduct experimental evaluations to investigate the effectiveness of the regularization method and show remaining problems to be addressed.

Sound source separation and synthesis for audio enhancement based on spectral amplitudes of two-channel stereo signals

A sound source separation algorithm based on the spectral amplitudes of 2-channel signals has been developed for the up-mixing playback of 2-channel stereo. Short-term Fourier transforms (STFT) of the signals on the left and right channels are first calculated. The coefficients of the discrete Fourier transform (DFT) are used to calculate the ratio of the spectral amplitudes of the left and right channels, which is termed the channel level difference (CLD). The DFT coefficients are then divided into multiple groups on the basis of the CLD, with each group representing a separated sound source. The signal-to-distortion ratio (SDR)is used to evaluate the signal separation performance. It was found that a rough estimate of the CLD threshold yielding the best SDR could be obtained by cross-correlating the separated sounds. For playback on a headset, each separated signal is convoluted with head-related transfer functions (HRTF) that represent the direction of that particular sound source. Subjective listening tests showed that the sound synthesized by this method is more realistic than that synthesized with HRTFs that represent only left and right speakers.

Sound field reproduction system with active reverberation

Sound field reproduction system can be categorized roughly into two types. The first is physically assured system which reproduce the wavefront or another physical parameters by engineering means. Typical examples are the boundary surface control system, the wave field synthesis, the ambisonics, and so on. The second type is a psychological system that reproduces phantom images one sound at any desired position by artificial manipulation with rather simple technique such as volume based amplitude panning. Examples are the two channel stereo system and the surround system using 5.1 or more loudspeakers. The latter can be recognized as somewhat “artistic” method, because it requires norms based on artistic sense. The present study aims to fuse the engineering and artistic systems mentioned above. The boundary surface control system, which consists of 24-channel narrow directional microphones and 24 channel loudspeakers, was used as the platform of research. Based on the conventional control technique using inverse filtering, additional manipulation such as simple reproduction scheme for high frequency was introduced. Also, the concept of variable reflection acoustic wall system, which adds the artificial reverberation to the sounds that listener makes, was involved in the reproduction system for the direction of the presence.

Electronical correction of misalignments between optical grid and pixel panel on autostereoscopic displays

Large-area displays and an especial sharp image quality are built by gradual increasing pixel density. This trend is predestinated for the field of autostereoscopic 3D displays. Though it is hardly possible to adjust the optical grid against the pixel panel, due to micrometer scaled structures. This paper presents a method to locally correct adjustment failure by reallocating the image information in fine steps by adapting existing methods. It is possible to use the applicable methods at single-user, multiview as well as integral-imaging displays, and it enables an adjustment to the observer positions. With the correction method developed by us, local misalignments can be detected in a test image, which arranges the individual stereo channels on an HSV-color cycle by color coding. The test image will be detected and measured by a photo-spectrometer in a preselected position. We also demonstrate the generation of a resulting correction-map, which contains the position, offset or shift value as well as the shift direction of the affected subpixels. The effectiveness of this correction method was proven by measurements and also determined its working scope.

Using Paired Distances of Signal Peaks in Stereo Channels as Fingerprints for Copy Identification

This article proposes to use the relative distances between adjacent envelope peaks detected in stereo audio as fingerprints for copy identification. The matching algorithm used is the rough longest common subsequence (RLCS) algorithm. The experimental results show that the proposed approach has better identification accuracy than an MPEG-7 based scheme for distorted and noisy audio. When compared with other schemes, the proposed scheme uses fewer bits with comparable performance. The proposed fingerprints can also be used in conjunction with the MPEG-7 based scheme for lower computational burden.

Development of Real-Time Adaptive Noise Canceller and Echo Canceller

In this paper, the adaptive cancellation structure is first developed based on the LMS algorithm and FIR adaptive filtering. Then the novel practical noise and echo cancellation systems are built using the proposed adaptive technique and implemented using TX320TMS67C13 DSKs, which are Texas Instruments’ Digital Signal Processing (TI DSP) boards. Although adaptive filtering is an exciting topic in which many real-life applications can be explored [1]-[6], [9], building such a real-time system is often challenging due to the use of theoretical math, advanced DSP knowledge and practical industrial hands-on experience [1],[4]-[6],[9]. Therefore, this paper indicates that it is possible to apply traditional mathematics in adaptive filtering theory to real-time practical DSP systems. With the MATLAB software tool, we can simulate and verify various adaptive filtering designs first. Then, development and implementation of different noise or echo cancellation systems with adaptive filtering techniques can be successfully performed using the floating-point digital signal processor, TX320TMS67C13 DSK. Furthermore, it can be shown that TX320TMS67C13 DSKs with their stereo channels offer more effective and flexible tools for various noise cancellation applications.