We describe our current work towards a framework that establishes a hierarchy of devices (sensors and actuators) within a wireless multimedia node and uses frequent sampling of cheaper devices to trigger the activation of more energy-hungry devices. Within this framework, we consider the suitability of servos for Wireless Multimedia Sensor Networks (WMSNs) by examining their functional characteristics and energy consumption [2].

This paper is concerned with the application of wireless sensor network (WSN) technology to long-duration and large-scale environmental monitoring. The holy grail is a system that can be deployed and operated by domain specialists not engineers, but this remains some distance into the future. We present our views as to why this field has progressed less quickly than many envisaged it would over a decade ago. We use real examples taken from our own work in this field to illustrate the technological difficulties and challenges that are entailed in meeting end-user requirements for information gathering systems. Reliability and productivity are key concerns and influence the design choices for system hardware and software. We conclude with a discussion of long-term challenges for WSN technology in environmental monitoring and outline our vision of the future.

This paper considers the use of servo-mechanisms as part of a tightly integrated homogeneous wireless multimedia sensor network (WMSN). We describe the design of our second generation WMSN node platform, which has increased image resolution, in-built audio sensors, PIR sensors, and servomechanisms. These devices have a wide disparity in their energy consumption and in the information quality they return. As a result, we propose a framework that establishes a hierarchy of devices (sensors and actuators) within the node and uses frequent sampling of cheaper devices to trigger the activation of more energy-hungry devices. Within this framework, we consider the suitability of servos for WMSNs by examining the functional characteristics and by measuring the energy consumption of 2 analog and 2 digital servos, in order to determine their impact on overall node energy cost. We also implement a simple version of our hierarchical sampling framework to evaluate the energy consumption of servos relative to other node components. The evaluation results show that: (1) the energy consumption of servos is small relative to audio/image signal processing energy cost in WMSN nodes; (2) digital servos do not necessarily consume as much energy as is currently believed; and (3) the energy cost per degree panning is lower for larger panning angles.

A wireless sensor network application for detection of bird calls is described. The detection algorithm is based on a likelihood ratio test between statistical models of the target and non-target audio frames. The algorithm has been implemented on CSIROpsilas wireless sensor network platform, using a DSP coprocessor board on each node for signal processing and detection. Results are presented for detection of the logrunner bird species.

Cancer staging provides a basis for planning clinical management, but also allows for meaningful analysis of cancer outcomes and evaluation of cancer care services. Despite this, stage data in cancer registries is often incomplete, inaccurate, or simply not collected. This article describes a prototype software system (Cancer Stage Interpretation System, CSIS) that automatically extracts cancer staging information from medical reports. The system uses text classification techniques to train support vector machines (SVMs) to extract elements of stage listed in cancer staging guidelines. When processing new reports, CSIS identifies sentences relevant to the staging decision, and subsequently assigns the most likely stage. The system was developed using a database of staging data and pathology reports for 710 lung cancer patients, then validated in an independent set of 179 patients against pathologic stage assigned by two independent pathologists. CSIS achieved overall accuracy of 74% for tumor (T) staging and 87% for node (N) staging, and errors were observed to mirror disagreements between human experts.

Close-talk headset microphones have been traditionally used for speech acquisition in a number of applications, as they naturally provide a higher signal-to-noise ratio - needed for recognition tasks than single distant microphones. However, in multi-party conversational settings like meetings, microphone arrays represent an important alternative to close-talking microphones, as they allow for localisation and tracking of speakers and signal-independent enhancement, while providing a non-intrusive, hands-free operation mode. In this article, we investigate the use of an audio-visual sensor array, composed of a small table-top microphone array and a set of cameras, for speaker tracking and speech enhancement in meetings. Our methodology first fuses audio and video for person tracking, and then integrates the output of the tracker with a beamformer for speech enhancement. We compare and discuss the features of the resulting speech signal with respect to that obtained from single close-talking and table-top microphones.

The automatic processing of speech collected in conference style meetings has attracted considerable interest with several large scale projects devoted to this area. In this paper we explore the use of various meeting corpora for the purpose of automatic speech recognition. In particular we investigate the similarity of these resources and how to efficiently use them in the construction of a meeting transcription system. The analysis shows distinctive features for each resource. However the benefit in pooling data and hence the similarity seems sufficient to speak of a generic conference meeting domain . In this context this paper also presents work on development for the AMI meeting transcription system, a joint effort by seven sites working on the AMI (augmented multi-party interaction) project.

The paper investigates the recognition of group actions in meetings by modeling the joint behaviour of participants. Many meeting actions, such as presentations, discussions and consensus, are characterised by similar or complementary behaviour across participants. Recognising these meaningful actions is an important step towards the goal of providing effective browsing and summarisation of processed meetings. A corpus of meetings was collected in a room equipped with a number of microphones and cameras. The corpus was labeled in terms of a predefined set of meeting actions characterised by global behaviour. In experiments, audio and visual features for each participant are extracted from the raw data and the interaction of participants is modeled using HMM-based approaches. Initial results on the corpus demonstrate the ability of the system to recognise the set of meeting actions.

This paper investigates the use of microphone arrays to acquire and recognise speech in meetings. Meetings pose several interesting problems for speech processing, as they consist of multiple competing speakers within a small space, typically around a table. Due to their ability to provide hands-free acquisition and directional discrimination, microphone arrays present a potential alternative to close-talking microphones in such an application. We first propose an appropriate microphone array geometry and improved processing technique for this scenario, paying particular attention to speaker separation. during possible overlap segments. Data collection of a small vocabulary speech recognition corpus (Numbers) was performed in a real meeting room for a single speaker, and several overlapping speech scenarios. In speech recognition experiments on the acquired database, the performance of the microphone array system is compared to that of a close-talking lapel microphone, and a single table-top microphone.

We present a probabilistic method for audio-visual (AV) speaker tracking, using an uncalibrated wide-angle camera and a micro- phone array. The algorithm fuses 2-D object shape and audio information via importance particle filters (I-PFs), allowing for the asymmetrical integration of AV information in a way that efficiently exploits the complementary features of each modality. Audio localization information is used to generate an importance sampling (IS) function, which guides the random search process of a particle filter towards regions of the configuration space likely to contain the true configuration (a speaker). The measurement process integrates contour-based and audio observations, which results in reliable head tracking in realistic scenarios. We show that imperfect single modalities can be combined into an algorithm that automatically initializes and tracks a speaker, switches between multiple speakers, tolerates visual clutter, and recovers from total AV object occlusion, in the context of a multimodal meeting room.

Speaker turn detection is an important task for many speech processing applications. However, accurate segmentation can be hard to achieve if there are multiple concurrent speakers (overlap), as is typically the case in multi-party conversations. In such cases, the location of the speaker, as measured using a microphone array, may provide greater discrimination than traditional spectral features. This was verified in previous work which obtained a global segmentation in terms of single speaker classes, as well as possible overlap combinations. However, such a global strategy suffers from an explosion of the number of overlap classes, as each possible combination of concurrent speakers must be modeled explicitly. In this paper, we propose two alternative schemes that produce an individual segmentation decision for each speaker, implicitly handling all overlapping speaker combinations. The proposed approaches also allow straightforward online implementations. Experiments are presented comparing the segmentation with that obtained using the previous system. 1.