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Ibrahim Almajai

dblp:75/8844 · DBLP profile ↗
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10ranked-venue papers
6as first author
0since 2021 · last 2016
—ORCID · none

Domains — the database's venue-derived domains; a paper can count in several

Graphics, computer vision, multimedia, augmented reality and games · 8 · 5 first-authorArtificial intelligence and machine learning · 6 · 3 first-author

Expertise — from the expertise taxonomy: the topics of the expert's papers under the CCF categories. A weight counts papers with recency: 1 for a paper about the topic, 0.3 when the topic is its context, halved every five years.

Artificial intelligence
1 paper
Speech recognition and synthesis · 100%

Topics — the 2 heaviest of 2, each with the papers that count most for it

TopicWeightPapersLastEvidence papers
Natural language and speech › Speech recognition and synthesis › speech enhancement
audio-visual speech enhancement
0.112011
Visually Derived Wiener Filters for Speech Enhancement · IEEE Trans. Speech Audio Process. 2011
Natural language and speech › Speech recognition and synthesis › speech analysis
voice activity detection
0.012011
Visually Derived Wiener Filters for Speech Enhancement · IEEE Trans. Speech Audio Process. 2011

Methods — techniques the papers use, named apart from their topics

wiener filter · 0.1maximum a posteriori estimation · 0.1hidden markov model · 0.1
YearPublicationVenuePosition
2016 Improved speaker independent lip reading using speaker adaptive training and deep neural networks
abstract
Recent improvements in tracking and feature extraction mean that speaker-dependent lip-reading of continuous speech using a medium size vocabulary (around 1000 words) is realistic. However, the recognition of previously unseen speakers has been found to be a very challenging task, because of the large variation in lip-shapes across speakers and the lack of large, tracked databases of visual features, which are very expensive to produce. By adapting a technique that is established in speech recognition but has not previously been used in lip-reading, we show that error-rates for speaker-independent lip-reading can be very significantly reduced. Furthermore, we show that error-rates can be even further reduced by the additional use of Deep Neural Networks (DNN). We also find that there is no need to map phonemes to visemes for context-dependent visual speech transcription.
Ibrahim Almajai, Stephen J. Cox, Richard W. Harvey, Yuxuan Lan
ICASSP1
2015 Voice Äpp: a mobile app for crowdsourcing Swiss German dialect data
Adrian Leemann, Marie-José Kolly, Jean-Philippe Goldman, Volker Dellwo, Ingrid Hove, Ibrahim Almajai, Sarah Grimm, Sylvain Robert, Daniel Wanitsch
INTERSPEECH6
2014 A Crowdsourcing Smartphone Application for Swiss German: Putting Language Documentation in the Hands of the Users
Jean-Philippe Goldman, Adrian Leemann, Marie-José Kolly, Ingrid Hove, Ibrahim Almajai, Volker Dellwo, Steven Moran
LREC5
2011 Visually Derived Wiener Filters for Speech Enhancement
abstract
The aim of this work is to examine whether visual speech information can be used to enhance audio speech that has been contaminated by noise. First, an analysis of audio and visual speech features is made, which identifies the pair with highest audio-visual correlation. The study also reveals that higher audio-visual correlation exists within individual phoneme sounds rather than globally across all speech. This correlation is exploited in the proposal of a visually derived Wiener filter that obtains clean speech and noise power spectrum statistics from visual speech features. Clean speech statistics are estimated from visual features using a maximum a posteriori framework that is integrated within the states of a network of hidden Markov models to provide phoneme localization. Noise statistics are obtained through a novel audio-visual voice activity detector which utilizes visual speech features to make robust speech/nonspeech classifications. The effectiveness of the visually derived Wiener filter is evaluated subjectively and objectively and is compared with three different audio-only enhancement methods over a range of signal-to-noise ratios.
Ibrahim Almajai, Ben P. Milner
IEEE Trans. Speech Audio Process.1
2010 Ball event recognition using hmm for automatic tennis annotation
abstract
A key prerequisite of automatic video indexing and summarisation is the description of events and actions. In the context of many sports, the motion of the ball and agents plays an essential role in describing events. However, the only existing solution for the tennis event recognition problem in the literature is the work in which relies on a set of heuristic rules such as proximity between ball and players or court lines to classify ball event candidates. We present hidden Markov models (HMMs) paradigm to automatically learn to identify events from ball trajectories and demonstrate that its ability to capture the dynamics of the ball movement lead to a much higher performance.
Ibrahim Almajai, Josef Kittler, Teófilo Emídio de Campos, William J. Christmas, Fei Yan 0001, David Windridge, Aftab Khan 0001
ICIP1
2009 Enhancing audio speech using visual speech features
abstract
This work presents a novel approach to speech enhancement by exploiting the bimodality of speech and the correlation that exists between audio and visual speech features. For speech enhancement, a visually-derived Wiener filter is developed. This obtains clean speech statistics from visual features by modelling their joint density and making a maximum a posteriori estimate of clean audio from visual speech features. Noise statistics for the Wiener filter utilise an audio-visual voice activity detector which classifies input audio as speech or nonspeech, enabling a noise model to be updated. Analysis shows estimation of speech and noise statistics to be effective with human listening tests measuring the effectiveness of the resulting Wiener filter. Index Terms: Audio-visual, speech enhancement, Wiener filter 1.
Ibrahim Almajai, Ben P. Milner
INTERSPEECH1
2009 Reconstructing clean speech from noisy MFCC vectors
abstract
The aim of this work is to reconstruct clean speech solely from a stream of noise-contaminated MFCC vectors, as may be encountered in distributed speech recognition systems. Speech reconstruction is performed using the ETSI Aurora back-end speech reconstruction standard which requires MFCC vectors, fundamental frequency and voicing information. In this work, fundamental frequency and voicing are obtained using maximum a posteriori prediction from input MFCC vectors, thereby allowing speech reconstruction solely from a stream of MFCC vectors. Two different methods to improve prediction accuracy in noisy conditions are then developed. Experimental results first establish that improved fundamental frequency and voicing prediction is obtained when noise compensation is applied. A series of human listening tests are then used to analyse the reconstructed speech quality, which determine the effectiveness of noise compensation in terms of mean opinion scores.
Ben P. Milner, Jonathan Darch, Ibrahim Almajai
INTERSPEECH3
2007 Visually-Derived Wiener Filters for Speech Enhancement
abstract
This work begins by examining the correlation between audio and visual speech features and reveals higher correlation to exist within individual phoneme sounds rather than globally across all speech. Utilising this correlation, a visually-derived Wiener filter is proposed in which clean power spectrum estimates are obtained from visual speech features. Two methods of extracting clean power spectrum estimates are made; first from a global estimate using a single Gaussian mixture model (GMM), and second from phoneme-specific estimates using a hidden Markov model (HMM)-GMM structure. Measurement of estimation accuracy reveals that the phoneme-specific (HMM-GMM) system leads to lower estimation errors than the global (GMM) system. Finally, the effectiveness of visually-derived Wiener filtering is examined.
Ibrahim Almajai, Ben P. Milner, Jonathan Darch, Saeed Vaseghi
ICASSP (4)1
2007 An Investigation into the Correlation and Prediction of Acoustic Speech Features from MFCC Vectors
abstract
This work develops a statistical framework to predict acoustic features (fundamental frequency, formant frequencies and voicing) from MFCC vectors. An analysis of correlation between acoustic features and MFCCs is made both globally across all speech and within phoneme classes, and also from speaker-independent and speaker-dependent speech. This leads to the development of both a global prediction method, using a Gaussian mixture model (GMM) to model the joint density of acoustic features and MFCCs, and a phoneme-specific prediction method using a combined hidden Markov model (HMM)-GMM. Prediction accuracy measurements show the phoneme-dependent HMM-GMM system to be more accurate which agrees with the correlation analysis. Results also show prediction to be more accurate from speaker-dependent speech which also agrees with the correlation analysis.
Jonathan Darch, Ben P. Milner, Ibrahim Almajai, Saeed Vaseghi
ICASSP (4)3
2006 Analysis of correlation between audio and visual speech features for clean audio feature prediction in noise
abstract
The aim of this work is to examine the correlation between audio and visual speech features. The motivation is to find visual features that can provide clean audio feature estimates which can be used for speech enhancement when the original audio signal is corrupted by noise. Two audio features (MFCCs and formants) and three visual features (active appearance model, 2-D DCT and cross-DCT) are considered with correlation measured using multiple linear regression. The correlation is then exploited through the development of a maximum a posteriori (MAP) prediction of audio features solely from the visual features. Experiments reveal that features representing broad spectral information have higher correlation to visual features than those representing finer spectral detail. The accuracy of prediction follows the results found in the correlation measurements. Index Terms: audio-visual speech, correlation, AAM, formants 1.
Ibrahim Almajai, Ben P. Milner, Jonathan Darch
INTERSPEECH1