22 research outputs found
Speaker detection in the wild: Lessons learned from JSALT 2019
Submitted to ICASSP 2020This paper presents the problems and solutions addressed at the JSALT workshop when using a single microphone for speaker detection in adverse scenarios. The main focus was to tackle a wide range of conditions that go from meetings to wild speech. We describe the research threads we explored and a set of modules that was successful for these scenarios. The ultimate goal was to explore speaker detection; but our first finding was that an effective diarization improves detection, and not having a diarization stage impoverishes the performance. All the different configurations of our research agree on this fact and follow a main backbone that includes diarization as a previous stage. With this backbone, we analyzed the following problems: voice activity detection, how to deal with noisy signals, domain mismatch, how to improve the clustering; and the overall impact of previous stages in the final speaker detection. In this paper, we show partial results for speaker diarizarion to have a better understanding of the problem and we present the final results for speaker detection
Speaker detection in the wild: Lessons learned from JSALT 2019
International audienceThis paper presents the problems and solutions addressed at the JSALT workshop when using a single microphone for speaker detection in adverse scenarios. The main focus was to tackle a wide range of conditions that go from meetings to wild speech. We describe the research threads we explored and a set of modules that was successful for these scenarios. The ultimate goal was to explore speaker detection; but our first finding was that an effective diarization improves detection, and not having a diarization stage impoverishes the performance. All the different configurations of our research agree on this fact and follow a main backbone that includes diarization as a previous stage. With this backbone, we analyzed the following problems: voice activity detection, how to deal with noisy signals, domain mismatch, how to improve the clustering; and the overall impact of previous stages in the final speaker detection. In this paper, we show partial results for speaker diarizarion to have a better understanding of the problem and we present the final results for speaker detection
Learning spectro-temporal representations of complex sounds with parameterized neural networks
Deep Learning models have become potential candidates for auditory
neuroscience research, thanks to their recent successes on a variety of
auditory tasks. Yet, these models often lack interpretability to fully
understand the exact computations that have been performed. Here, we proposed a
parametrized neural network layer, that computes specific spectro-temporal
modulations based on Gabor kernels (Learnable STRFs) and that is fully
interpretable. We evaluated predictive capabilities of this layer on Speech
Activity Detection, Speaker Verification, Urban Sound Classification and Zebra
Finch Call Type Classification. We found out that models based on Learnable
STRFs are on par for all tasks with different toplines, and obtain the best
performance for Speech Activity Detection. As this layer is fully
interpretable, we used quantitative measures to describe the distribution of
the learned spectro-temporal modulations. The filters adapted to each task and
focused mostly on low temporal and spectral modulations. The analyses show that
the filters learned on human speech have similar spectro-temporal parameters as
the ones measured directly in the human auditory cortex. Finally, we observed
that the tasks organized in a meaningful way: the human vocalizations tasks
closer to each other and bird vocalizations far away from human vocalizations
and urban sounds tasks
A comparison study on patient-psychologist voice diarization
International audienceConversations between a clinician and a patient, in natural conditions, are valuable sources of information for medical follow-up. The automatic analysis of these dialogues could help extract new language markers and speed up the clinicians' reports. Yet, it is not clear which model is the most efficient to detect and identify the speaker turns, especially for individuals with speech disorders. Here, we proposed a split of the data that allows conducting a comparative evaluation of different diarization methods. We designed and trained end-to-end neural network architectures to directly tackle this task from the raw signal and evaluate each approach under the same metric. We also studied the effect of fine-tuning models to find the best performance. Experimental results are reported on naturalistic clinical conversations between Psychologists and Interviewees, at different stages of Huntington's disease, displaying a large panel of speech disorders. We found out that our best end-to-end model achieved 19.5% IER on the test set, compared to 23.6% achieved by the finetuning of the X-vector architecture. Finally, we observed that we could extract clinical markers directly from the automatic systems, highlighting the clinical relevance of our methods
Audio self-supervised learning: a survey
Inspired by the humans' cognitive ability to generalise knowledge and skills,
Self-Supervised Learning (SSL) targets at discovering general representations
from large-scale data without requiring human annotations, which is an
expensive and time consuming task. Its success in the fields of computer vision
and natural language processing have prompted its recent adoption into the
field of audio and speech processing. Comprehensive reviews summarising the
knowledge in audio SSL are currently missing. To fill this gap, in the present
work, we provide an overview of the SSL methods used for audio and speech
processing applications. Herein, we also summarise the empirical works that
exploit the audio modality in multi-modal SSL frameworks, and the existing
suitable benchmarks to evaluate the power of SSL in the computer audition
domain. Finally, we discuss some open problems and point out the future
directions on the development of audio SSL
âDid the speaker change?â: Temporal tracking for overlapping speaker segmentation in multi-speaker scenarios
Diarization systems are an essential part of many speech processing applications, such as speaker indexing, improving automatic speech recognition (ASR) performance and making single speaker-based algorithms available for use in multi-speaker domains. This thesis will focus on the first task of the diarization process, that being the task of speaker segmentation which can be thought of as trying to answer the question âDid the speaker change?â in an audio recording.
This thesis starts by showing that time-varying pitch properties can be used advantageously within the segmentation step of a multi-talker diarization system. It is then highlighted that an individualâs pitch is smoothly varying and, therefore, can be predicted by means of a Kalman filter. Subsequently, it is shown that if the pitch is not predictable, then this is most likely due to a change in the speaker. Finally, a novel system is proposed that uses this approach of pitch prediction for speaker change detection.
This thesis then goes on to demonstrate how voiced harmonics can be useful in detecting when more than one speaker is talking, such as during overlapping speaker activity. A novel system is proposed to track multiple harmonics simultaneously, allowing for the determination of onsets and end-points of a speakerâs utterance in the presence of an additional active speaker.
This thesis then extends this work to explore the use of a new multimodal approach for overlapping speaker segmentation that tracks both the fundamental frequency (F0) and direction of arrival (DoA) of each speaker simultaneously. The proposed multiple hypothesis tracking system, which simultaneously tracks both features, shows an improvement in segmentation performance when compared to tracking these features separately.
Lastly, this thesis focuses on the DoA estimation part of the newly proposed multimodal approach. It does this by exploring a polynomial extension to the multiple signal classification (MUSIC) algorithm, spatio-spectral polynomial (SSP)-MUSIC, and evaluating its performance when using speech sound sources.Open Acces