Speaker Tracking System Research Articles

Ray based virtual time reversal method for the localization of sound sources in reverberant fields

Localization of sound sources in reverberant fields is significant for the research areas such as noise sources recognition, meeting speaker tracking system, intelligent robot design. Microphone arrays are usually used, however, in most applications in rooms it is practical and economical to decrease the microphones. In this paper, using only one microphone a virtual time reversal algorithm based on the ray-tracing method has been developed according to the reciprocity theorem and the time reversal invariance of linear wave equation. The algorithm has been validated by the localization experiments in a real room and a virtual room. Then the performance of the algorithm under the conditions of various reverberation time, source-receiver distance and sound reflection times has been investigated according to a specified ratio.

Robust time delay estimation for speech signals using information theory: A comparison study

Time delay estimation (TDE) is a fundamental subsystem for a speaker localization and tracking system. Most of the traditional TDE methods are based on second-order statistics (SOS) under Gaussian assumption for the source. This article resolves the TDE problem using two information-theoretic measures, joint entropy and mutual information (MI), which can be considered to indirectly include higher order statistics (HOS). The TDE solutions using the two measures are presented for both Gaussian and Laplacian models. We show that, for stationary signals, the two measures are equivalent for TDE. However, for non-stationary signals (e.g., noisy speech signals), maximizing MI gives more consistent estimate than minimizing joint entropy. Moreover, an existing idea of using modified MI to embed information about reverberation is generalized to the multiple microphones case. From the experimental results for speech signals, this scheme with Gaussian model shows the most robust performance in various noisy and reverberant environments.

Beamforming With a Maximum Negentropy Criterion

In this paper, we address a beamforming application based on the capture of far-field speech data from a single speaker in a real meeting room. After the position of the speaker is estimated by a speaker tracking system, we construct a subband-domain beamformer in generalized sidelobe canceller (GSC) configuration. In contrast to conventional practice, we then optimize the active weight vectors of the GSC so as to obtain an output signal with maximum negentropy (MN). This implies the beamformer output should be as non-Gaussian as possible. For calculating negentropy, we consider the Gamma and the generalized Gaussian (GG) pdfs. After MN beamforming, Zelinski postfiltering is performed to further enhance the speech by removing residual noise. Our beamforming algorithm can suppress noise and reverberation without the signal cancellation problems encountered in the conventional beamforming algorithms. We demonstrate this fact through a set of acoustic simulations. Moreover, we show the effectiveness of our proposed technique through a series of far-field automatic speech recognition experiments on the Multi-Channel Wall Street Journal Audio Visual Corpus (MC-WSJ-AV), a corpus of data captured with real far-field sensors, in a realistic acoustic environment, and spoken by real speakers. On the MC-WSJ-AV evaluation data, the delay-and-sum beamformer with postfiltering achieved a word error rate (WER) of 16.5%. MN beamforming with the Gamma pdf achieved a 15.8% WER, which was further reduced to 13.2% with the GG pdf, whereas the simple delay-and-sum beamformer provided a WER of 17.8%. To the best of our knowledge, no lower error rates at present have been reported in the literature on this automatic speech recognition (ASR) task.

Development of a Speaker Diarization System for Speaker Tracking in Audio Broadcast News: a Case Study

A system for speaker tracking in broadcast-news audio data is presented and the impacts of the main components of the system to the overall speaker-tracking performance are evaluated. The process of speaker tracking in continuous audio streams involves several processing tasks and is therefore treated as a multistage process. The main building blocks of such system include the components for audio segmentation, speech detection, speaker clustering and speaker identification. The aim of the first three processes is to find homogeneous regions in continuous audio streams that belong to one speaker and to join each region of the same speaker together. The task of organizing the audio data in this way is known as speaker diarization and plays an important role in various speech-processing applications. In our case the impact of speaker diarization was assessed in a speaker-tracking system by performing a comparative study of how each of the component influenced the overall speaker-detection results. The evaluation experiments were performed on broadcast-news audio data with a speaker-tracking system, which was capable of detecting 41 target speakers. We implemented several different approaches in each component of the system and compared their performances by inspecting the final speaker-tracking results. The evaluation results indicate the importance of the audio-segmentation and speech-detection components, while no significant improvement of the overall results was achieved by additionally including a speaker-clustering component to the speaker-tracking system.

Real-Time Speaker Tracking Using Particle Filter Sensor Fusion

Sensor fusion for object tracking has become an active research direction during the past few years. But how to do it in a robust and principled way is still an open problem. In this paper, we propose a new fusion framework that combines both the bottom-up and top-down approaches to probabilistically fuse multiple sensing modalities. At the lower level, individual vision and audio trackers are designed to generate effective proposals for the fuser. At the higher level, the fuser performs reliable tracking by verifying hypotheses over multiple likelihood models from multiple cues. Unlike traditional fusion algorithms, the proposed framework is a closed-loop system where the fuser and trackers coordinate their tracking information. Furthermore, to handle nonstationary situations, the proposed framework evaluates the performance of the individual trackers and dynamically updates their object states. We present a real-time speaker tracking system based on the proposed framework by fusing object contour, color and sound source location. We report robust tracking results.

Multiple Speaker Tracking and Detection: Handset Normalization and Duration Scoring

Sönmez, Kemal, Heck, Larry, and Weintraub, Mitchel, Multiple Speaker Tracking and Detection: Handset Normalization and Duration Scoring, Digital Signal Processing10(2000), 133–142.We describe SRI's speaker tracking and detection system in the NIST 1998 Speaker Detection and Tracking Development Evaluation. The system is designed for tracking switchboard conversations and uses a two-speaker and silence hidden Markov model (HMM) with a minimum state duration constraint and Gaussian mixture model (GMM) state distributions adapted from a single gender- and handset-independent imposter model distribution. Speaker tracking is used to segment waveforms for speaker detection, which is carried out by averaging frame scores of the Viterbi path and normalizing for handset variation via a novel parameter interpolation extension of HNORM for use with waveform segments of arbitrary lengths. A short-duration penalty to augment the acoustic scores is also introduced via a nonlinear combination function. Results on the NIST 1998 Speaker Detection and Tracking Development Evaluation dataset are reported.