538 research outputs found
Improving speaker turn embedding by crossmodal transfer learning from face embedding
Learning speaker turn embeddings has shown considerable improvement in
situations where conventional speaker modeling approaches fail. However, this
improvement is relatively limited when compared to the gain observed in face
embedding learning, which has been proven very successful for face verification
and clustering tasks. Assuming that face and voices from the same identities
share some latent properties (like age, gender, ethnicity), we propose three
transfer learning approaches to leverage the knowledge from the face domain
(learned from thousands of images and identities) for tasks in the speaker
domain. These approaches, namely target embedding transfer, relative distance
transfer, and clustering structure transfer, utilize the structure of the
source face embedding space at different granularities to regularize the target
speaker turn embedding space as optimizing terms. Our methods are evaluated on
two public broadcast corpora and yield promising advances over competitive
baselines in verification and audio clustering tasks, especially when dealing
with short speaker utterances. The analysis of the results also gives insight
into characteristics of the embedding spaces and shows their potential
applications
Processing and Linking Audio Events in Large Multimedia Archives: The EU inEvent Project
In the inEvent EU project [1], we aim at structuring, retrieving, and sharing large archives of networked, and dynamically changing, multimedia recordings, mainly consisting of meetings, videoconferences, and lectures. More specifically, we are developing an integrated system that performs audiovisual processing of multimedia recordings, and labels them in terms of interconnected âhyper-events â (a notion inspired from hyper-texts). Each hyper-event is composed of simpler facets, including audio-video recordings and metadata, which are then easier to search, retrieve and share. In the present paper, we mainly cover the audio processing aspects of the system, including speech recognition, speaker diarization and linking (across recordings), the use of these features for hyper-event indexing and recommendation, and the search portal. We present initial results for feature extraction from lecture recordings using the TED talks. Index Terms: Networked multimedia events; audio processing: speech recognition; speaker diarization and linking; multimedia indexing and searching; hyper-events. 1
Jitter and Shimmer measurements for speaker diarization
Jitter and shimmer voice quality features have been successfully
used to characterize speaker voice traits and detect voice pathologies.
Jitter and shimmer measure variations in the fundamental frequency
and amplitude of speaker's voice, respectively. Due to their nature, they can be used to assess differences between speakers. In this paper, we investigate the usefulness of these voice quality features in the task of speaker diarization. The combination of voice quality features with the conventional spectral features, Mel-Frequency Cepstral Coefficients (MFCC), is addressed in the framework of Augmented Multiparty Interaction (AMI) corpus, a multi-party and spontaneous speech set of recordings. Both sets of features are independently modeled using mixture of Gaussians and fused together at the score likelihood level. The experiments carried out on the AMI corpus show that incorporating jitter and shimmer measurements to the baseline spectral features decreases the diarization error rate in most of the recordings.Peer ReviewedPostprint (published version
The Blame Game: Performance Analysis of Speaker Diarization System Components
In this paper we discuss the performance analysis of a speaker diarization system similar to the system that was submitted by ICSI at the NIST RT06s evaluation benchmark. The analysis that is based on a series of oracle experiments, provides a good understanding of the performance of each system component on a test set of twelve conference meetings used in previous NIST benchmarks. Our analysis shows that the speech activity detection component contributes most to the total diarization error rate (23%). The lack of ability to model verlapping speech is also a large source of errors (22%) followed by the component that creates the initial system models (15%)
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