Dr Saeid Safavi


Research fellow in adaptive predictive fault detection for connected autonomous systems
PhD
+44 (0)1483 684718
09 BB 01
8:00-17:00

Biography

Areas of specialism

Machine learning; Speech processing; Music analysis; Para-linguistic processing of speech signals

University roles and responsibilities

  • Research fellow under H2020 project (Audio Commons)

    My qualifications

    PhD
    University of Birmingham
    Meng
    University of Birmingham

    Previous roles

    01 November 2015 - 01 November 2017
    Research fellow under a H2020 project with the title of "Objective Control of TAlker VErication (OCTAVE)".
    University of Hertfordshire
    2015 - 2015
    Postdoctoral researcher at the University of Birmingham working on a EU funded project.
    University of Birmingham

    Research

    Research interests

    Research projects

    My publications

    Publications

    Saeid Safavi, Wenwu Wang, Mark Plumbley, Ali Janalizadeh Choobbasti, George Fazekas (2018)Predicting the Perceived Level of Reverberation using Features from Nonlinear Auditory Model, In: Proceedings of the 23rd FRUCT conferencepp. 527-531 Institute of Electrical and Electronics Engineers (IEEE)

    Perceptual measurements have typically been recognized as the most reliable measurements in assessing perceived levels of reverberation. In this paper, a combination of blind RT60 estimation method and a binaural, nonlinear auditory model is employed to derive signal-based measures (features) that are then utilized in predicting the perceived level of reverberation. Such measures lack the excess of effort necessary for calculating perceptual measures; not to mention the variations in either stimuli or assessors that may cause such measures to be statistically insignificant. As a result, the automatic extraction of objective measurements that can be applied to predict the perceived level of reverberation become of vital significance. Consequently, this work is aimed at discovering measurements such as clarity, reverberance, and RT60 which can automatically be derived directly from audio data. These measurements along with labels from human listening tests are then forwarded to a machine learning system seeking to build a model to estimate the perceived level of reverberation, which is labeled by an expert, autonomously. The data has been labeled by an expert human listener for a unilateral set of files from arbitrary audio source types. By examining the results, it can be observed that the automatically extracted features can aid in estimating the perceptual rates.

    Ali Janalizadeh Choobbasti, Mohammad Erfan Gholamian, Amir Vaheb, Saeid Safavi (2018)JSpeech: a multi-lingual conversational speech corpus, In: Proceedings of the 2018 IEEE Spoken Language Technology Workshop (SLT 2018) Institute of Electrical and Electronics Engineers (IEEE)

    Speech processing, automatic speech and speaker recognition are the major area of interests in the field of computational linguistics. Research and development of computer and human interaction, forensic technologies and dialogue systems have been the motivating factor behind this interest. In this paper, JSpeech is introduced, a multilingual corpus. This corpus contains 1332 hours of conversational speech from 47 different languages. This corpus can be used in a variety of studies, created from 106 public chat group the effect of language variability on the performance of speaker recognition systems and automatic language detection. To this end, we include speaker verification results obtained for this corpus using a state of the art method based on 3D convolutional neural network.

    Saeid Safavi, Andy Pearce, Wenwu Wang, Mark Plumbley (2018)Predicting the perceived level of reverberation using machine learning, In: Proceedings of the 52nd Asilomar Conference on Signals, Systems and Computers (ACSSC 2018) Institute of Electrical and Electronics Engineers (IEEE)

    Perceptual measures are usually considered more reliable than instrumental measures for evaluating the perceived level of reverberation. However, such measures are time consuming and expensive, and, due to variations in stimuli or assessors, the resulting data is not always statistically significant. Therefore, an (objective) measure of the perceived level of reverberation becomes desirable. In this paper, we develop a new method to predict the level of reverberation from audio signals by relating the perceptual listening test results with those obtained from a machine learned model. More specifically, we compare the use of a multiple stimuli test for within and between class architectures to evaluate the perceived level of reverberation. An expert set of 16 human listeners rated the perceived level of reverberation for a same set of files from different audio source types. We then train a machine learning model using the training data gathered for the same set of files and a variety of reverberation related features extracted from the data such as reverberation time, and direct to reverberation ratio. The results suggest that the machine learned model offers an accurate prediction of the perceptual scores.

    Amir Vaheb, Ali Janalizadeh Choobbasti, S. H. E. Mortazavi Najafabadi, Saeid Safavi (2018)Investigating Language Variability on the Performance of Speaker Verification Systems, In: Speech and Computer11096pp. 718-727 Springer Nature Switzerland

    In recent years, speaker verification technologies have received an extensive amount of attention. Designing and developing machines that could communicate with humans are believed to be one of the primary motivations behind such developments. Speaker verification technologies are applied to numerous fields such as security, Biometrics, and forensics. In this paper, the authors study the effects of different languages on the performance of the automatic speaker verification (ASV) system. The MirasVoice speech corpus (MVSC), a bilingual English and Farsi speech corpus, is used in this study. This study collects results from both an I-vector based ASV system and a GMM-UBM based ASV system. The experimental results show that a mismatch between the enrolled data used for training and verification data can lead to a significant decrease in the overall system efficiency. This study shows that it is best to use an i-vector based framework with data from the English language used in the enrollment phase to improve the robustness of the ASV systems. The achieved results in this study indicate that this can narrow the degradation gap caused by the language mismatch.

    Qingju Liu, Wenwu Wang, Philip Jackson, Saeid Safavi (2018)A Performance Evaluation of Several Deep Neural Networks for Reverberant Speech Separation, In: 52nd Asilomar Conference Proceedingspp. 689-693 IEEE

    In this paper, we compare different deep neural networks (DNN) in extracting speech signals from competing speakers in room environments, including the conventional fullyconnected multilayer perception (MLP) network, convolutional neural network (CNN), recurrent neural network (RNN), and the recently proposed capsule network (CapsNet). Each DNN takes input of both spectral features and converted spatial features that are robust to position mismatch, and outputs the separation mask for target source estimation. In addition, a psychacoustically-motivated objective function is integrated in each DNN, which explores perceptual importance of each TF unit in the training process. Objective evaluations are performed on the separated sounds using the converged models, in terms of PESQ, SDR as well as STOI. Overall, all the implemented DNNs have greatly improved the quality and speech intelligibility of the embedded target source as compared to the original recordings. In particular, bidirectional RNN, either along the temporal direction or along the frequency bins, outperforms the other DNN structures with consistent improvement.

    Saeid Safavi, Turab Iqbal, Wenwu Wang, Philip Coleman, Mark D. Plumbley (2020)Open-Window: A Sound Event Dataset For Window State Detection and Recognition, In: Proc. 5th International Workshop on Detection and Classification of Acoustic Scenes and Events (DCASE 2020)

    Situated in the domain of urban sound scene classification by humans and machines, this research is the first step towards mapping urban noise pollution experienced indoors and finding ways to reduce its negative impact in peoples’ homes. We have recorded a sound dataset, called Open-Window, which contains recordings from three different locations and four different window states; two stationary states (open and close) and two transitional states (open to close and close to open). We have then built our machine recognition base lines for different scenarios (open set versus closed set) using a deep learning framework. The human listening test is also performed to be able to compare the human and machine performance for detecting the window state just using the acoustic cues. Our experimental results reveal that when using a simple machine baseline system, humans and machines are achieving similar average performance for closed set experiments.