Phoneme and sub-phoneme T-normalization for text-dependent speaker recognition

Proceedings of Odyssey 2008: The Speaker and Language Recognition Workshop, Stellenbosch, South AfricaTest normalization (T-Norm) is a score normalization technique that is regularly and successfully applied in the context of text-independent speaker recognition. It is less frequently applied, however, to text-dependent or text prompted speaker recognition, mainly because its improvement in this context is more modest. In this paper we present a novel way to improve the performance of T-Norm for text-dependent systems. It consists in applying score TNormalization at the phoneme or sub-phoneme level instead of at the sentence level. Experiments on the YOHO corpus show that, while using standard sentence-level T-Norm does not improve equal error rate (EER), phoneme and sub-phoneme level T-Norm produce a relative EER reduction of 18.9% and 20.1% respectively on a state-of-the-art HMM based text dependent speaker recognition system. Results are even better for working points with low false acceptance rates.This work was funded by the Spanish Ministry of Science and Technology under project TEC2006-13170-C02-01

Exploring convolutional, recurrent, and hybrid deep neural networks for speech and music detection in a large audio dataset

Audio signals represent a wide diversity of acoustic events, from background environmental noise to spoken communication. Machine learning models such as neural networks have already been proposed for audio signal modeling, where recurrent structures can take advantage of temporal dependencies. This work aims to study the implementation of several neural network-based systems for speech and music event detection over a collection of 77,937 10-second audio segments (216 h), selected from the Google AudioSet dataset. These segments belong to YouTube videos and have been represented as mel-spectrograms. We propose and compare two approaches. The first one is the training of two different neural networks, one for speech detection and another for music detection. The second approach consists on training a single neural network to tackle both tasks at the same time. The studied architectures include fully connected, convolutional and LSTM (long short-term memory) recurrent networks. Comparative results are provided in terms of classification performance and model complexity. We would like to highlight the performance of convolutional architectures, specially in combination with an LSTM stage. The hybrid convolutional-LSTM models achieve the best overall results (85% accuracy) in the three proposed tasks. Furthermore, a distractor analysis of the results has been carried out in order to identify which events in the ontology are the most harmful for the performance of the models, showing some difficult scenarios for the detection of music and speechThis work has been supported by project “DSSL: Redes Profundas y Modelos de Subespacios para Deteccion y Seguimiento de Locutor, Idioma y Enfermedades Degenerativas a partir de la Voz” (TEC2015-68172-C2-1-P), funded by the Ministry of Economy and Competitivity of Spain and FEDE

A quantitative study of disfluencies in formal, informal and media spontaneous speech in Spanish

Proceedings of IberSpeech 2012 (Madrid, Spain)A descriptive study of the prevalence of different types of disfluencies (fragmented words, restarts and vocalic supports) in spontaneous Spanish is presented based on a hand-annotated corpus. A quantitative account of differences among three types of registers (formal, informal and media) and several subtypes of text for each register is provided to analyze the importance of each disfluency class for a given register

An analysis of sound event detection under acoustic degradation using multi-resolution systems

The Sound Event Detection task aims to determine the temporal locations of acoustic events in audio clips. In recent years, the relevance of this field is rising due to the introduction of datasets such as Google AudioSet or DESED (Domestic Environment Sound Event Detection) and competitive evaluations like the DCASE Challenge (Detection and Classification of Acoustic Scenes and Events). In this paper, we analyze the performance of Sound Event Detection systems under diverse artificial acoustic conditions such as high-or low-pass filtering and clipping or dynamic range compression, as well as under an scenario of high overlap between events. For this purpose, the audio was obtained from the Evaluation subset of the DESED dataset, whereas the systems were trained in the context of the DCASE Challenge 2020 Task 4. Our systems are based upon the challenge baseline, which consists of a Convolutional-Recurrent Neural Network trained using the Mean Teacher method, and they employ a multiresolution approach which is able to improve the Sound Event Detection performance through the use of several resolutions during the extraction of Mel-spectrogram features. We provide insights on the benefits of this multiresolution approach in different acoustic settings, and compare the performance of the single-resolution systems in the aforementioned scenarios when using different resolutions. Furthermore, we complement the analysis of the performance in the high-overlap scenario by assessing the degree of overlap of each event category in sound event detection datasetsThis research and the APC were supported by project DSForSec (grant number RTI2018- 098091-B-I00) funded by the Ministry of Science, Innovation and Universities of Spain and the European Regional Development Fund (ERDF

Audio encoding for heart and breath sounds acquired with digital stethoscope

Proceedings of IberSpeech 2012 (Madrid, Spain)This paper presents a real-time audio encoding system for cardiorespiratory sounds acquired with a low cost Bluetooth digital stethoscope developed by EHAS (Enlace Hispano Americano de Salud) Foundation and Fundatel Foundation. The system is currently working with a G.722 speech coder for compression and subsequent transmission of cardiorespiratory signals. However, these signals have different frequency characteristics from speech. Therefore, we sought a better adapted alternative to encode these signals optimally, with an encoder not subject to the payment of any license, with low computational cost, good quality and low bandwidth. We have evaluated the proposed solution both using objective measures such as root mean squared error and using subjective opinions from four expert physicians

A Multi-Resolution CRNN-Based Approach for Semi-Supervised Sound Event Detection in DCASE 2020 Challenge

Sound Event Detection is a task with a rising relevance over the recent years in the field of audio signal processing, due to the creation of specific datasets such as Google AudioSet or DESED (Domestic Environment Sound Event Detection) and the introduction of competitive evaluations like the DCASE Challenge (Detection and Classification of Acoustic Scenes and Events). The different categories of acoustic events can present diverse temporal and spectral characteristics. However, most approaches use a fixed time-frequency resolution to represent the audio segments. This work proposes a multi-resolution analysis for feature extraction in Sound Event Detection, hypothesizing that different resolutions can be more adequate for the detection of different sound event categories, and that combining the information provided by multiple resolutions could improve the performance of Sound Event Detection systems. Experiments are carried out over the DESED dataset in the context of the DCASE 2020 Challenge, concluding that the combination of up to 5 resolutions allows a neural network-based system to obtain better results than single-resolution models in terms of event-based F1-score in every event category and in terms of PSDS (Polyphonic Sound Detection Score). Furthermore, we analyze the impact of score thresholding in the computation of F1-score results, finding that the standard value of 0.5 is suboptimal and proposing an alternative strategy based in the use of a specific threshold for each event category, which obtains further improvements in performanceThis work was supported in part by the Project Deep Speech for Forensics and Security (DSForSec) under Grant RTI2018-098091-B-I00, in part by the Ministry of Science, Innovation and Universities of Spain, and in part by the European Regional Development Fund (ERDF

Multivariate Cepstral Feature Compensation on Band-limited Data for Robust Speech Recognition

Proceedings of the 16th Nordic Conference of Computational Linguistics NODALIDA-2007. Editors: Joakim Nivre, Heiki-Jaan Kaalep, Kadri Muischnek and Mare Koit. University of Tartu, Tartu, 2007. ISBN 978-9985-4-0513-0 (online) ISBN 978-9985-4-0514-7 (CD-ROM) pp. 144-151

Inventario de frecuencias fonémicas y silábicas del castellano espontáneo y escrito

Actas de las IV Jornadas de Tecnología del Habla (JTH 2006)Este artículo presenta dos inventarios de frecuencias – fonémico y silábico – del castellano obtenidos a partir del corpus C-ORAL-ROM, que recoge español oral espontáneo en distintos contextos y registros. Estos inventarios se han desarrollado mediante un transcriptor fonológico y silábico cuyos resultados para el corpus CORAL-ROM han sido en su mayor parte revisados manualmente. Los inventarios incluyen la frecuencia absoluta de aparición de los diferentes fonemas y sílabas. Estos datos se han examinado junto a los extraídos de un corpus comparable de texto escrito y se ha hallado evidencia de que los inventarios de frecuencias obtenidos hasta ahora, basados fundamentalmente en textos, no describen adecuadamente el castellano oral espontáneo.Este trabajo ha sido finaciado parcialmente por el MEC CICYT (TIN2004-07588-C03-02) y la Comunidad Autónoma de Madrid y la Universidad Autónoma de Madrid (05/TIC/001)

Augmented set of features for confidence estimation in spoken term detection

Discriminative confidence estimation along with confidence normalisation have been shown to construct robust decision maker modules in spoken term detection (STD) systems. Discriminative confidence estimation, making use of termdependent features, has been shown to improve the widely used lattice-based confidence estimation in STD. In this work, we augment the set of these term-dependent features and show a significant improvement in the STD performance both in terms of ATWV and DET curves in experiments conducted on a Spanish geographical corpus. This work also proposes a multiple linear regression analysis to carry out the feature selection. Next, the most informative features derived from it are used within the discriminative confidence on the STD system

Using data-driven and phonetic units for speaker verification

Personal use of this material is permitted. Permission from IEEE must be obtained for all other uses, in any current or future media, including reprinting/republishing this material for advertising or promotional purposes, creating new collective works, for resale or redistribution to servers or lists, or reuse of any copyrighted component of this work in other works. A. E. Hannani, D. T. Toledano, D. Petrovska-Delacrétaz, A. Montero-Asenjo, J. Hennebert, "Using Data-driven and Phonetic Units for Speaker Verification" in Odyssey: The Speaker and Language Recognition Workshop, San Juan (Puerto Rico), 2006, pp.1 - 6Recognition of speaker identity based on modeling the streams produced by phonetic decoders (phonetic speaker recognition) has gained popularity during the past few years. Two of the major problems that arise when phone based systems are being developed are the possible mismatches between the development and evaluation data and the lack of transcribed databases. Data-driven segmentation techniques provide a potential solution to these problems because they do not use transcribed data and can easily be applied on development data minimizing the mismatches. In this paper we compare speaker recognition results using phonetic and data-driven decoders. To this end, we have compared the results obtained with a speaker recognition system based on data-driven acoustic units and phonetic speaker recognition systems trained on Spanish and English data. Results obtained on the NIST 2005 Speaker Recognition Evaluation data show that the data-driven approach outperforms the phonetic one and that further improvements can be achieved by combining both approache