Multi-channel stereo reproducing apparatus

In environments with many concurrent sound sources, it is often possible to hear out individual sounds and identify their sources. To accomplish this perceptual organization of the sound sources, listeners analyze the mixture of sounds to estimate the constituent components that correspond to one acoustic sound source. Signal components that are likely to come from the same source are grouped into an auditory object, that is segregated from the acoustic complex. The grouping of signal components into sources is based on identifying shared features of the components, like common spectral cues (the same fundamental frequency or frequency modulation), temporal cues (the same temporal envelope), or spatial cues (the same interaural differences). When signal components’ onsets fall within such a time window that they are perceived as being synchronous, the components are combined to form a new auditory object, that is monitored across time on cues that sustain their grouping. Given that spatial cues are inherent to every sound and already present at the onset, and that they enhance detection of a sound by a spatial release from masking, it is unclear whether and how much temporal perception depends on the presence of spatial cues. The research presented here addressed the contribution of the binaural hearing system, which processes a sound’s spatial cues, to the processing of its temporal cues, and the contribution of the sounds’ temporal cues to the processing of their spatial cues. In order to assess the contribution of spatial cues to temporal perception, experiments were conducted in which the temporal position of a target sound’s onset was controlled by the subjects to find its desired position relative to a regular series of reference sound onsets. Manipulation of the target’s interaural differences, its sensation level, and the target and reference sounds’ durations allowed comparison of their effects on temporal perception of the target’s onset. The main finding was that temporal perception precision degraded below a critical level, depending on the absolute target and masker levels and the difference in spatial configuration between target and masker that facilitated perception at much lower signal-to-noise ratios. The implication of a sound’s signal-to-noise ratio getting below this critical level is that, when the temporal window for grouping signal components’ onsets is enlarged due to the degraded precision, there is an increased probability for signal components to be grouped incorrectly into an auditory object, or to be prevented from grouping into a new auditory object and remain part of the acoustic complex until other evidence is gathered for the existence of their source. This may be a contributing factor to the decrease of speech intelligibility in noisy environments, where precise temporal perception of transients is required for the processing of speech signals. In order to assess the contribution of temporal cues to binaural processing, thresholds were measured for discrimination between the spatial configurations of two sounds that had identical spectral range and onset and offset times, but differed in their temporal envelope structure and had equally-sized directional information to opposite sides of the medial plane. The measured minimum interaural differences in time or level at which the sounds’ spatial configurations could successfully be discriminated represented the ability to process binaural cues, and allowed comparison between the effects of the various manipulations of bandwidths, center frequencies, and temporal envelopes to the sounds. The main findings were that discrimination based on interaural level differences was good for all tested conditions (although absolute thresholds depended on the specific conditions), and that discrimination based on interaural time differences was only possible when the temporal envelopes of the sounds were sufficiently different. The binaural system must, therefore, have been capable of processing the interaural differences within short periods of time when one of the sounds was dominant in the monaural temporal envelopes of the combined signals at each ear. Analysis of the monaural temporal envelopes influences the ability to use the binaural information to separate individual signal components, and allow tracking across time of the signals components with identical spatial cues to form a perceptual unity, the auditory object. In conditions where discrimination was not possible, the temporal envelopes of the individual sounds were too similar and such periods did not occur systematically in the temporal envelopes of the combined sounds. This finding suggests that a monaural analysis of a composite signal’s temporal envelope is a prerequisite for the binaural processing of the constituent signal components’ interaural time differences. The implication is that, if the signals’ temporal envelopes are too similar, their signal components may not be correctly grouped into separate and individually lateralized auditory objects, but remain fused as a single object with an averaged spatial image.

We describe the design, implementation and performance of a stand-alone media server. This server is intended to be used in voice over IP systems for audio interaction with users, NAT traversal, call monitoring and media recording. Its advantage is that it is application independent, which means that it can be used by any service and for any signaling protocol. In this paper we describe the design of the server, of its control protocol and of its functionalities such as file playing, DTMF detection, RTP proxying and audio recording. For file playing, we propose an approach to schedule RTP packet transmission that overcomes the irregularity of event scheduling of the operating system and for conversation recording, we propose an original approach to synchronize media streams that allows to avoid voice overlap while recording a conversation. We have conducted many experiments on the media server. We evaluated the accuracy of the file playing timing under different load conditions, the effect of the RTP proxying on the media quality and the subjective quality of recorded conversations. The results of these experiments show the great impact of the load of the media server on the performance of its functionalities, the improvement that the packet scheduling algorithm brings to the precision of inter-departure timing, and the good subjective quality of the recorded conversation using our synchronizing approach.

Disclosed is a method of encoding a multichannel signal including at least a first signal component (y) and a second signal component (r). The method comprises the steps of determining a set of filter parameters (Fp) of a prediction filter (201) such that the prediction filter provides an estimate (P) of the second signal component when receiving the first signal component as an input; and representing the multichannel signal as the first signal component and the set of filter parameters. Further disclosed are a corresponding arrangement for encoding a multichannel signals and a corresponding method and arrangement for decoding such a signal.

Information (audio, video, or data) to be recorded magnetically is first encoded into a sequence of time-varying electrical signals which are used to drive a writing head, the magnetic field of which creates, on the moving tape or disk, a spatially varying pattern of magnetization. The reading process uses another head (or the same head) to reconvert this magnetization pattern into time-varying electrical signals that can be amplified and used to drive a loudspeaker, or a TV receiver, or feed data to a computer. Two methods of encoding the information are used; analog and digital. In analog recording, as the adjective suggests, the pattern of magnetization on the recording medium should be a precise analog of the incoming stream of information and the signal obtained on reading should be an exact reproduction of the original. Digital recording involves first converting the information into a sequence of ones and zeros which can then be recorded as a sequence of regions of opposite magnetization. Analog recording has been widely used to store audio and video information while digital recording was used for data recording. This picture is changing and digital recording is now being used increasingly in audio and video recording principally because errors can be corrected dynamically so that the original information can be reconstructed and copied repeatedly and with great fidelity. Whether the recording method used is analog or digital, similar though not identical properties are required of the recording materials and so the distinction between the two methods will not be pursued further.

Described as "the most comprehensive book on digital audio to date", it is widely acclaimed as an industry "bible". Covering the very latest developments in digital audio technology, it provides an thorough introduction to the theory as well as acting as an authoritative and comprehensive professional reference source. Everything you need is here from the fundamental principles to the latest applications, written in an award-winning style with clear explanations from first principles. New material covered includes internet audio, PC audio technology, DVD, MPEG audio compression, digital audio broadcasting and audio networks.Whether you are in the field of audio engineering, sound recording, music technology, broadcasting and communications media or audio design and installation, this book has it all. Written by a leading international audio specialist, who conducts professional seminars and workshops around the world, the book has been road tested for many years by professional seminar attendees and students to ensure their needs are taken into account, and all the right information is covered.This new edition now includes:Internet audioPC Audio technologyDVDMPEG Audio compressionDigital Audio BroadcastingAudio networksDigital audio professionals will find everything they need here, from the fundamental principles to the latest applications, written in an award-winning style with clear explanations from first principles.John Watkinson is an international consultant in audio, video and data recording. He is a Fellow of the AES, a member of the British Computer Society and a chartered information systems practitioner. He presents lectures, seminars, conference papers and training courses worldwide. He is the author of many other Focal Press books, including: the Kraszna-Krausz award winning MPEG-2; The Art of Digital Audio; An Introduction to Digital Video; The Art of Sound Reproduction; An Introduction to Digital Audio; TV Fundamentals and Audio for Television. He is also co-author, with Francis Rumsey, of The Digital Interface Handbook, and contributor to the Loudspeaker and Headphone Handbook, 3rd edition.

Time-Scale Modification (TSM) is a well-researched field and allows for time-domain manipulation of a signal without modifying the pitch or timbre. Many TSM methods have been presented, however quantitative results on the quality of these methods are rare, with most methods reporting informal listening tests. This is likely due to the timecommitment and cost of subjective testing. Additionally, an objective measure of quality has not yet been developed that is suitable for timescaled signals. This dissertation describes the design of e ective objective measures of quality for TSM. TSM methods are, generally, single channel algorithms that give poor results when applied to multi-channel signals, as the phase relationship between channels must be maintained. This dissertation proposes a method and additional variant for maintaining the phase relationship between channels and retaining the presence in the centre of the stereo signal. The method involves pre- and post-processing the signal, with the variant processing each frame for real-time suitability. Sum and di erence transformations of the stereo signal are used for TSM and result in a large improvement in stereo phase coherence, consequently maintaining the stereo field. The proposed method produces a highquality stereo output and greatly improves quality over the independent channel processing method. It also allows for simple implementation around all existing TSM frameworks. A modification to the Epoch-Synchronous Overlap-Add (ESOLA) TSM algorithm is proposed in this dissertation. The proposed method, Fuzzy Epoch-Synchronous Overlap-Add, improves on the previous ESOLA method through cross-correlation of time-smeared epochs before overlap-adding. This reduces distortion and artefacts while the speaker's fundamental frequency is stable, as well as reducing artefacts during pitch modulation. The proposed method is tested against well-known TSM algorithms. It is preferred over ESOLA and gives similar performance to other TSM algorithms for voice signals. It is also shown that this algorithm can work effectively with solo instrument signals containing strong fundamental frequencies. No effective objective measure of quality for TSM exists. This dissertation details the creation, subjective evaluation and analysis of a dataset, for use in the development of an objective measure of quality for TSM. Comprising two parts, the training subset contains 88 source files processed using six TSM methods at 10 time-scales, while the testing subset contains 20 source files processed using three additional methods at four time-scales. The source material contains speech, solo harmonic and percussive instruments, sound effects and a range of music genres. 42,529 ratings were collected from 633 sessions using laboratory and remote collection methods. Analysis of results shows no correlation between age and quality of rating; equivalence between expert and non-expert listeners; negligible di erences between participants with and without hearing issues; and negligible di erences between testing modalities. Comparison of published objective measures and subjective scores shows the objective measures to be poor indicators of subjective quality. Initial results for a retrained objective measure of quality are presented with results approaching average loss and correlation values of subjective sessions. An objective measure of quality for time-scaled audio is proposed that makes use of the previously developed dataset and improves on reported results. The measure uses hand-crafted features and a fully connected network to predict subjective mean opinion scores. Basic and Advanced Perceptual Evaluation of Audio Quality features are used in addition to nine features specific to TSM artefacts. Six methods of alignment are explored, with interpolation of the reference magnitude spectrum to the length of the test magnitude spectrum giving the best performance. The proposed measure achieves an average Root Mean Squared Error (RMSE) of 0.490 and a mean Pearson Correlation Coe cient (PCC) of 0.864, equivalent to 97th and 82nd percentiles of subjective sessions respectively. The proposed measure is used to evaluate TSM algorithms, finding that Elastique gives the highest objective quality for solo instrument and voice signals, while the Identity Phase-Locking Phase Vocoder gives the highest objective quality for music signals and the best overall quality. Two single-ended objective quality measures for time-scaled audio are also proposed. These measure do not require a reference signal, nor alignment. Data driven features are created by either a convolutional neural network (CNN) or a bidirectional gated recurrent unit (BGRU) network, and are fed to a fully-connected network to predict subjective mean opinion scores. The proposed CNN and BGRU measures achieve an average RMSE of 0.608 and 0.576, and a mean PCC of 0.771 and 0.794, respectively. The proposed measures are used to evaluate TSM algorithms, and comparisons are provided for 16 TSM implementations. A literature review is included with required background knowledge. It includes the fundamentals of sound perception, sound capture, digital signal processing, time-scale modification methods used within research, and subjective and objective measures of quality. Full implementation of all proposed methods and measures can be found at github.com/zygurt/TSM, while the labelled dataset is available at http://ieee-dataport.org/1987.

Separation of components of a 1H NMR composite signal by nutation experiments under low amplitude radio-frequency fields. Application to the water signal in clays

Listeners with normal hearing can communicate successfully in noisy and reverberant conditions that distort temporal and spectral cues in the speech signal. Recent work has sought to identify auditory grouping processes that contribute to this ability. One property that has received attention is the fundamental frequency of voicing (F0). During voiced speech the pulsing of the vocal folds gives rise to a pattern of amplitude modulation in the waveform and harmonicity in the spectrum. When two or more voices compete for the attention of the listener, momentary differences in F0 can contribute to voice segregation in at least three ways. First, periodicity or harmonicity in the composite signal provides a basis for grouping together signal components that stem from a target voice. Second, waveform interactions generate moment-to-moment fluctuations in the signal-to-noise ratio that enable listeners to ‘‘glimpse’’ the acoustic features of the target voice. Third, time-varying changes in F0 provide a basis for tracking properties of the voice over time. The contributions of each of these processes to the perceptual segregation of speech from competing sounds will be reviewed. [Work supported by NIH/NIDCD.]

Decomposition of multichannel multicomponent nonstationary signals by combining the eigenvectors of autocorrelation matrix using genetic algorithm

During the COVID-19 pandemic, smart home requirements have shifted toward entertainment at home. The purpose of this research project was therefore to develop a robotic audio system for home automation. High-end audio systems normally refer to multichannel home theaters. Although multichannel audio systems enable people to enjoy surround sound as they do at the cinema, stereo audio systems have been popularly used since the 1980s. The major shortcoming of a stereo audio system is its narrow listening area. If listeners are out of the area, the system has difficulty providing a stable sound field. This is because of the head-shadow effect blocking the high-frequency sound. The proposed system, by integrating computer vision and robotics, can track the head movement of a user and adjust the directions of loudspeakers, thereby helping the sound wave travel through the air. Unlike previous studies, in which only a diminutive scenario was built, in this work, the idea was applied to a commercial 2.1 audio system, and listening tests were conducted. The theory and the simulation coincide with the experimental results. The approximate rate of audio quality improvement is 31%. The experimental results are encouraging, especially for high-pitched music.

The transmission frequency of power grids, i.e., electric network frequency (ENF), has become a common criterion to authenticate audio recordings during the past decade, drawing much attention from both the academic researchers and law enforcement agencies world widely. The properties of ENF enable forensic applications such as audio evidence timestamp verification and tampering detection. In this paper, based on a general review of existing works, we discuss several important practical problems and facts that have drawn less research attention or have not been formally studied, including ENF detection problems, limitations of the ENF-based tampering detection systems, and the difficulties in the ENF analysis. During ENF detection, the challenges come from not only the noise and the interference, but also the fact that audio recordings without captured ENF can still have signal components in the frequency band of interest (false positive). In ENF-based tampering detection systems, the weakness of commonly used assumptions and the limitations of several existing solutions are discussed. In addition, we reveal that in the most intensively studied ENF-based audio evidence timestamp verification, many works aiming at improving ENF estimation could only produce marginal performance improvement, while the main problems due to noise and interference remain open. All these analysis and discussions are related by a proposed big picture of ENF-based audio authentication systems. After that, we also investigate the strategies to design more reliable audio authentication systems based on the ENF, which consists of a series of research and investigation works.

The structure and performance of locally optimum detectors for composite signals in additive and multiplicative noise are investigated. A generalised model representing the effects of multiplicative noise and additive noise is considered for signal detection problems. The signals considered here are composite signals which contain both deterministic and stochastic signal components. Locally optimum detectors consist of nonlinearities followed by an accumulator, whose output is then compared with a threshold. The detection nonlinearities for a probability density function are obtained to show the detection structures. Finite sample-size performance characteristics of the locally optimum detector are obtained and are compared with those of other detectors.

This paper proposes a new method to extract fetal ECG from multi-channel signals. The fetal ECG is extracted in phase space, which is constructed by a pregnant woman's multiple thoracic and abdominal ECG signals. Based on corresponding points in phase space, we adopt least square algorithm to estimate the maternal components of the abdominal signals and extract the fetal signals by subtracting the estimated maternal signals from the abdominal signals. Compared with traditional separation method based on information fusion of multi-channel signals, such as Independent Component Analysis (ICA), this method could be realized more easily and effectively.

Joint extraction-based multivariate chirp mode decomposition and its application to fault diagnosis of rotating machinery under variable speed conditions

The aim of the study was to inspect the acoustic properties and sleep characteristics of a preapneic snoring sound. The feasibility of forecasting upcoming respiratory events by snoring sound was also investigated. Participants with habitual snoring or a heavy breathing sound during sleep were recruited consecutively. Polysomnography was conducted, and snoring-related breathing sound was recorded simultaneously. Acoustic features and sleep features were extracted from 30-second samples, and a machine learning algorithm was used to establish 2 prediction models. A total of 74 eligible participants were included. Model 1, tested by 5-fold cross-validation, achieved an accuracy of 0.92 and an area under the curve of 0.94 for respiratory event prediction. Model 2, with acoustic features and sleep information tested by Leave-One-Out cross-validation, had an accuracy of 0.78 and an area under the curve of 0.80. Sleep position was found to be the most important among all sleep features contributing to the performance of the 2 models. Preapneic sound presented unique acoustic characteristics, and snoring-related breathing sound could be deployed as a real-time apneic event predictor. The models, combined with sleep information, serve as a promising tool for an early warning system to forecast apneic events. Wang B, Yi X, Gao J, etal. Real-time prediction of upcoming respiratory events via machine learning using snoring sound signal. J Clin Sleep Med. 2021;17(9):1777-1784.