Cocktail-Party Effect with Computational Auditory Scene Analysis — Preliminary Report —

Cocktail party problem is a classical scientific problem that has been studied for decades. Humans have remarkable skills in segregating target speech from a complex auditory mixture obtained in a cocktail party environment. Computational modeling for such a mechanism is however extremely challenging. This chapter presents an overview of several recent techniques for the source separation issues associated with this problem, including independent component analysis/blind source separation, computational auditory scene analysis, model-based approaches, non-negative matrix factorization and sparse coding. As an example, a multistage approach for source separation is included. The application areas of cocktail party processing are explored. Potential future research directions are also discussed.

The sound of a busy environment, such as a city street, gives rise to a perception of numerous distinct events in a human listener--the 'auditory scene analysis' of the acoustic information. Recent advances in the understanding of this process from experimental psychoacoustics have led to several efforts to build a computer model capable of the same function. This work is known as 'computational auditory scene analysis'. The dominant approach to this problem has been as a sequence of modules, the output of one forming the input to the next. Sound is converted to its spectrum, cues are picked out, and representations of the cues are grouped into an abstract description of the initial input. This 'data-driven' approach has some specific weaknesses in comparison to the auditory system: it will interpret a given sound in the same way regardless of its context, and it cannot 'infer' the presence of a sound for which direct evidence is hidden by other components. The 'prediction-driven' approach is presented as an alternative, in which analysis is a process of reconciliation between the observed acoustic features and the predictions of an internal model of the sound-producing entities in the environment. In this way, predicted sound events will form part of the scene interpretation as long as they are consistent with the input sound, regardless of whether direct evidence is found. A blackboard-based implementation of this approach is described which analyzes dense, ambient sound examples into a vocabulary of noise clouds, transient clicks, and a correlogram-based representation of wide-band periodic energy called the weft. The system is assessed through experiments that firstly investigate subjects' perception of distinct events in ambient sound examples, and secondly collect quality judgments for sound events resynthesized by the system. Although rated as far from perfect, there was good agreement between the events detected by the model and by the listeners. In addition, the experimental procedure does not depend on special aspects of the algorithm (other than the generation of resyntheses), and is applicable to the assessment and comparison of other models of human auditory organization. (Copies available exclusively from MIT Libraries, Rm. 14-0551, Cambridge, MA 02139-4307. Ph. 617-253-5668; Fax 617-253-1690.)

Monaural speech separation is the process of separating the target speech from a noisy speech mixture recorded using single microphone. It can be used in wide range of applications including mobile telephony, hearing aid design and robust automatic speech and speaker recognition (ASR). Recently, researchers use computational auditory scene analysis (CASA) technique to successfully separate the target speech from the monaural noisy speech mixture. In CASA based monaural speech separation techniques, Ideal binary mask (IBM) and Ideal ratio mask (IRM) has been proposed as a computational goal to improve the speech intelligibility and speech quality. This paper reviews and reports various research works carried out using CASA techniques with IBM and IRM to improve speech intelligibility and quality. The experimental results show that CASA systems using IBM improves the speech intelligibility and using IRM improves the speech quality.

Auditory Scene Analysis: Computational Models

This paper introduces a speech separation system as a front-end processing step for automatic speech recognition (ASR). It employs computational auditory scene analysis (CASA) to separate the target speech from the interference speech. Specifically, the mixed speech is preprocessed based on auditory peripheral model. Then a pitch tracking is conducted and the dominant pitch is used as a main cue to find the target speech. Next, the time frequency (TF) units are merged into many segments. These segments are then combined into streams via CASA initial grouping. A regrouping strategy is employed to refine these streams via amplitude modulate (AM) cues, which are finally organized by the speaker recognition techniques into corresponding speakers. Finally, the output streams are reconstructed to compensate the missing data in the abovementioned processing steps by a cluster based feature reconstruction. Experimental results of ASR show that at low TMR (<-6dB) the proposed method offers significantly higher recognition accuracy. Index term: speech separation, CASA, speaker recognition, pitch tracking, units grouping, reconstruction

The segregation of concurrent speakers and other sound sources is an important aspect in improving the performance of audio technology, such as noise reduction and automatic speech recognition, ASR, in difficult acoustic conditions. This technology is relevant for applications like hearing aids, mobile audio devices, robotics, hands-free audio communication and speech-based computer interfaces. Computational auditory-scene analysis (CASA) techniques simulate aspects of processing properties of the human perceptual system using statistical signal-processing techniques to improve inferences about the causes of audio input received by the system. This study argues that CASA is a promising approach to achieve source separation and outlines several theoretical arguments to support this hypothesis. With a focus on computational binaural scene analysis, principles of CASA techniques are reviewed. Furthermore, in an experimental approach, the applicability of a recent model of binaural interaction to improve ASR performance in multi-speaker conditions with spatially separated moving speakers is explored. The binaural model provides input to a statistical inference filter that employs a priori information on possible movements of the sources in order to track the positions of the speakers. The tracks are used to adapt a beamformer that selects a specific speaker. The output of the beamformer is subsequently used for an ASR task. Compared to the unprocessed, that is, mixed, data in a two-speaker condition, the word recognition rates obtained with the enhanced signals based on binaural information were increased from 30.8 to 88.4 %, demonstrating the potential of the proposed CASA-based approach.

An approach to segregate the target speech form the mixture utterance in low signal noise ratio (SNR) was proposed. Within the framework of computational auditory scene analysis (CASA), phase was the cue for segregation, and short time Fourier transforms (STFT) was used to extract the phase of the signal. Binary masking was used to group the target speech units based on the difference of phase between the mixture, clean speech and noise. The threshold of the binary masks was not linear. It adapted with the frequency change, and obtained from pretest. Experiments illustrated that the improvement of signal to noise ratio was more than 20dB in babble, m109, white and machinegun noise in -30dB to -20dB. The waveform of the result signal shown it remained most detail of the original signal, and had a well intelligibility. Phase is a robust cue in monaural speech segregation.Keywordsspeech segregationsignal phasecomputational auditory scene analysis (CASA)short time Fourier transforms (STFT)

In this paper, we propose a new time-frequency mask method for computational auditory scene analysis (CASA) based on convex optimization of the binary mask. In the proposed method, the pitch estimation and segment segregation in conventional CASA are completely replaced by the convex optimization of speech power. Considering the cross-correlation between the power spectra of noisy speech and noise in each of a Gammatone filterbank channel, the objective function of speech power used for convex optimization is built. The speech power is estimated by gradient descent method. Thus, the time-frequency units dominated by speech and noise are labeled by comparing the powers of noisy and estimated speech, and noise. The erroneous local masks are also removed by using the Teager energy of the estimated speech and time-frequency unit smoothing. The results from the average segmental signal-to-noise ratio improvement, HIT-False Alarm rate and subjective test show that the performance of the proposed method outperforms the reference methods.

Speech separation is a process of segregating the target speech from the noisy mixture. Human auditory system naturally has the capability to separate the speech from background noise; however the machine implementation of the same has been failed. Computational Auditory Scene Analysis (CASA) is a recent approach which models the human auditory system by computational means to separate the target speech from the noisy mixture. Most of the speech separation system based on CASA uses energy as one of the reliable feature to segregate the target speech from the noisy mixture. This research work proposes a new approach using adaptive energy selection threshold to segregate the target speech. The experimental result shows significant improvement in the signal to noise ratio as compared to the conventional energy threshold method.

In this paper, I make the following claims: (1) Subjective experience is tremendously useful in guiding productive research. (2) Studies of auditory scene analysis (ASA) in adults, newborn infants, and non-human animals (e.g., in goldfish or pigeons) establish the generality of ASA and suggest that it has an innate foundation. (3) ASA theory does not favor one musical style over another. (4) The principles used in the composition of polyphony (slightly modified) apply not only to one particular musical style or culture but to any form of layered music. (5) Neural explanations of ASA do not supersede explanations in terms of capacities; the two are complementary. (6) In computational auditory scene analysis (CASA) – ASA by computer systems – or any adequate theory of ASA, the most difficult challenge will be to discover how the contributions of a very large number of types of acoustical evidence and top-down schemas (acquired knowledge about the sound sources in our environments), can be coordinated without producing conflict that disables the system. (7) Finally I argue that the movement of a listener within the auditory scene provides him/her/it with rich information that should not be ignored by ASA theorists and researchers.

We are engaged in research on computational auditory scene analysis to attain sophisticated robot (computer) human interaction by recognizing auditory awareness. The objective of our research is the understanding of an arbitrary sound mixture including nonspeech sounds and music as well as voiced speech, obtained by robot's ears (or microphones embedded in the robot). The main issues are sound source localization, separation, and recognition at signal processing levels, and signal-to-symbol transformation at the interface level to symbol processing levels. The latter is critical in developmental communication and we are developing an automatic onomatopoeia recognition system. This paper overviews our activities in robot audition, in particular, active direction-pass filter (ADPF) that separates sounds originating from a specific direction by integrating sound source localization and visual processing. ADPF is implemented on three kinds of robots and demonstrates separating and recognizing three simultaneous speeches with a pair of microphones.

It is relatively easy for a human listener to attend to a particular speaker at a cocktail party in the presence of other speakers, music and environmental sounds. To perform this task, the human listener needs to separate the target speech from a mixture of multiple concurrent sources reflected by various surfaces. This process is referred to as auditory scene analysis. While humans excel at this task using only two ears, machine separation based on two-microphone recordings has proven to be extremely challenging. By incorporating the mechanisms underlying the perception of sound by human listeners, computational auditory scene analysis (CASA) offers a new approach to sound segregation. Binaural hearing – hearing with two ears – employs the difference in sound source locations to improve sound segregation. In this chapter, we describe the principles of binaural processing and review the state-of-the-art in binaural CASA, particularly for speech segregation.

For the problem of separating sound mixtures, periodicity is a powerful cue used by both human listeners and automatic systems. Short-term autocorrelation of subband envelopes, as in the correlogram, accounts for much perceptual data. We present a discrete representation of common-period sounds, derived from the correlogram, for use in computational auditory scene analysis: The weft describes a sound in terms of a time-varying periodicity and a smoothed spectral envelope of the energy exhibiting that period. Wefts improve on several aspects of previous approaches by providing, without additional grouping, a single, invertible element for each detected signal, and also a provisional solution to detecting and dissociating energy of different periodicities in a single frequency channel (unlike systems which allocate whole frequency channels to one source). We define the weft, describe the analysis procedure we have devised, and illustrate its capacity to separate periodic sounds from other signals.

A denoising method using computational auditory scene analysis (CASA) for a wearable stethoscope is proposed in this paper. The denoising method imitates human ears to analyze the ambient auditory scene so that it can separate the heart sound signal from unwanted ambient noise. Simulation results show that the method greatly improves the signal noise ratio of the heart sound signal after denoising.

Automatic sound event detection and computational auditory scene analysis gain importance through the increasing prevalence of technical systems operating autonomously or in the background, since such operation requires awareness of the system's environment. In realistic scenes, reliable sound event detection, despite the big improvements of the related automatic speech recognition, still poses a difficult problem: general sounds often are less definable than speech and exhibit less regularities and rules; commonly, many sounds occur simultaneously and in all kinds of acoustic environments. Binaural robotic systems are particularly interesting due to their resemblance of human means, but they are also more limited through the restriction to two microphones, specifically regarding spatial acoustic scene analysis. Spatial hearing figures prominently in humans, but for automatic sound event detection so far has gone mostly unregarded. One of the core objectives running through the entire thesis is the development of fundamental systematic methodology with respect to (a) the building of robust sound event detection models, and (b) the elaborate analysis regarding their application in many different situations --- both is underrepresented in available research publications. In the hereinafter presented studies, sound event detection models are built in different training schemes and evaluated in detail with respect to their performance in various acoustic scene conditions. Analyses are conducted on scenes with one to four co-occurring sound events, with sound-to-sound energy ratios of -20 dB to +20 dB, with different spatial source distributions, and in diverse acoustic environments from anechoic to church aula. It is shown (i) to which extent models that have been trained under specific acoustic conditions specialize to these, and (ii) that even with simple algorithms like logistic regression, through acoustically multifarious training almost optimal performances as achieved by the specialized models can be obtained consistently. The influence of temporal information integration is investigated, and it is shown that algorithms able to model context over longer durations benefit particularly in demanding scenes and get more precise in their detection. Moreover, a method for joining sound event detection and source localization is presented by which coherent auditory objects can be created. The proposed system associates the attributes "sound type" and "source location" successfully; for measuring success of such joint systems -- almost uncharted territory --, performance measures are suggested. It is shown that in an active binaural system spatial sound event detection performance can be increased considerably through suited head orientation. Finally, all developed models get tested in a simulated "online"-robotic system and their potential for forming integral components in computational auditory scene analysis is demonstrated.