1,897 research outputs found

    Deep Learning for Audio Signal Processing

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    Given the recent surge in developments of deep learning, this article provides a review of the state-of-the-art deep learning techniques for audio signal processing. Speech, music, and environmental sound processing are considered side-by-side, in order to point out similarities and differences between the domains, highlighting general methods, problems, key references, and potential for cross-fertilization between areas. The dominant feature representations (in particular, log-mel spectra and raw waveform) and deep learning models are reviewed, including convolutional neural networks, variants of the long short-term memory architecture, as well as more audio-specific neural network models. Subsequently, prominent deep learning application areas are covered, i.e. audio recognition (automatic speech recognition, music information retrieval, environmental sound detection, localization and tracking) and synthesis and transformation (source separation, audio enhancement, generative models for speech, sound, and music synthesis). Finally, key issues and future questions regarding deep learning applied to audio signal processing are identified.Comment: 15 pages, 2 pdf figure

    Artificial Bandwidth Extension of Speech Signals using Neural Networks

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    Although mobile wideband telephony has been standardized for over 15 years, many countries still do not have a nationwide network with good coverage. As a result, many cellphone calls are still downgraded to narrowband telephony. The resulting loss of quality can be reduced by artificial bandwidth extension. There has been great progress in bandwidth extension in recent years due to the use of neural networks. The topic of this thesis is the enhancement of artificial bandwidth extension using neural networks. A special focus is given to hands-free calls in a car, where the risk is high that the wideband connection is lost due to the fast movement. The bandwidth of narrowband transmission is not only reduced towards higher frequencies above 3.5 kHz but also towards lower frequencies below 300 Hz. There are already methods that estimate the low-frequency components quite well, which will therefore not be covered in this thesis. In most bandwidth extension algorithms, the narrowband signal is initially separated into a spectral envelope and an excitation signal. Both parts are then extended separately in order to finally combine both parts again. While the extension of the excitation can be implemented using simple methods without reducing the speech quality compared to wideband speech, the estimation of the spectral envelope for frequencies above 3.5 kHz is not yet solved satisfyingly. Current bandwidth extension algorithms are just able to reduce the quality loss due to narrowband transmission by a maximum of 50% in most evaluations. In this work, a modification for an existing method for excitation extension is proposed which achieves slight improvements while not generating additional computational complexity. In order to enhance the wideband envelope estimation with neural networks, two modifications of the training process are proposed. On the one hand, the loss function is extended with a discriminative part to address the different characteristics of phoneme classes. On the other hand, by using a GAN (generative adversarial network) for the training phase, a second network is added temporarily to evaluate the quality of the estimation. The neural networks that were trained are compared in subjective and objective evaluations. A final listening test addressed the scenario of a hands-free call in a car, which was simulated acoustically. The quality loss caused by the missing high frequency components could be reduced by 60% with the proposed approach.Obwohl die mobile Breitbandtelefonie bereits seit ĂŒber 15 Jahren standardisiert ist, gibt es oftmals noch kein flĂ€chendeckendes Netz mit einer guten Abdeckung. Das fĂŒhrt dazu, dass weiterhin viele MobilfunkgesprĂ€che auf Schmalbandtelefonie heruntergestuft werden. Der damit einhergehende QualitĂ€tsverlust kann mit kĂŒnstlicher Bandbreitenerweiterung reduziert werden. Das Thema dieser Arbeit sind Methoden zur weiteren Verbesserungen der QualitĂ€t des erweiterten Sprachsignals mithilfe neuronaler Netze. Ein besonderer Fokus liegt auf der Freisprech-Telefonie im Auto, da dabei das Risiko besonders hoch ist, dass durch die schnelle Fortbewegung die Breitbandverbindung verloren geht. Bei der SchmalbandĂŒbertragung fehlen neben den hochfrequenten Anteilen (etwa 3.5–7 kHz) auch tiefe Frequenzen unterhalb von etwa 300 Hz. Diese tieffrequenten Anteile können mit bereits vorhandenen Methoden gut geschĂ€tzt werden und sind somit nicht Teil dieser Arbeit. In vielen Algorithmen zur Bandbreitenerweiterung wird das Schmalbandsignal zu Beginn in eine spektrale EinhĂŒllende und ein Anregungssignal aufgeteilt. Beide Anteile werden dann separat erweitert und schließlich wieder zusammengefĂŒhrt. WĂ€hrend die Erweiterung der Anregung nahezu ohne QualitĂ€tsverlust durch einfache Methoden umgesetzt werden kann ist die SchĂ€tzung der spektralen EinhĂŒllenden fĂŒr Frequenzen ĂŒber 3.5 kHz noch nicht zufriedenstellend gelöst. Mit aktuellen Methoden können im besten Fall nur etwa 50% der durch SchmalbandĂŒbertragung reduzierten QualitĂ€t zurĂŒckgewonnen werden. FĂŒr die Anregungserweiterung wird in dieser Arbeit eine Variation vorgestellt, die leichte Verbesserungen erzielt ohne dabei einen Mehraufwand in der Berechnung zu erzeugen. FĂŒr die SchĂ€tzung der EinhĂŒllenden des Breitbandsignals mithilfe neuronaler Netze werden zwei Änderungen am Trainingsprozess vorgeschlagen. Einerseits wird die Kostenfunktion um einen diskriminativen Anteil erweitert, der das Netz besser zwischen verschiedenen Phonemen unterscheiden lĂ€sst. Andererseits wird als Architektur ein GAN (Generative adversarial network) verwendet, wofĂŒr in der Trainingsphase ein zweites Netz verwendet wird, das die QualitĂ€t der SchĂ€tzung bewertet. Die trainierten neuronale Netze wurden in subjektiven und objektiven Tests verglichen. Ein abschließender Hörtest diente zur Evaluierung des Freisprechens im Auto, welches akustisch simuliert wurde. Der QualitĂ€tsverlust durch Wegfallen der hohen Frequenzanteile konnte dabei mit dem vorgeschlagenen Ansatz um etwa 60% reduziert werden

    Practical Hidden Voice Attacks against Speech and Speaker Recognition Systems

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    Voice Processing Systems (VPSes), now widely deployed, have been made significantly more accurate through the application of recent advances in machine learning. However, adversarial machine learning has similarly advanced and has been used to demonstrate that VPSes are vulnerable to the injection of hidden commands - audio obscured by noise that is correctly recognized by a VPS but not by human beings. Such attacks, though, are often highly dependent on white-box knowledge of a specific machine learning model and limited to specific microphones and speakers, making their use across different acoustic hardware platforms (and thus their practicality) limited. In this paper, we break these dependencies and make hidden command attacks more practical through model-agnostic (blackbox) attacks, which exploit knowledge of the signal processing algorithms commonly used by VPSes to generate the data fed into machine learning systems. Specifically, we exploit the fact that multiple source audio samples have similar feature vectors when transformed by acoustic feature extraction algorithms (e.g., FFTs). We develop four classes of perturbations that create unintelligible audio and test them against 12 machine learning models, including 7 proprietary models (e.g., Google Speech API, Bing Speech API, IBM Speech API, Azure Speaker API, etc), and demonstrate successful attacks against all targets. Moreover, we successfully use our maliciously generated audio samples in multiple hardware configurations, demonstrating effectiveness across both models and real systems. In so doing, we demonstrate that domain-specific knowledge of audio signal processing represents a practical means of generating successful hidden voice command attacks

    JUNIPR: a Framework for Unsupervised Machine Learning in Particle Physics

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    In applications of machine learning to particle physics, a persistent challenge is how to go beyond discrimination to learn about the underlying physics. To this end, a powerful tool would be a framework for unsupervised learning, where the machine learns the intricate high-dimensional contours of the data upon which it is trained, without reference to pre-established labels. In order to approach such a complex task, an unsupervised network must be structured intelligently, based on a qualitative understanding of the data. In this paper, we scaffold the neural network's architecture around a leading-order model of the physics underlying the data. In addition to making unsupervised learning tractable, this design actually alleviates existing tensions between performance and interpretability. We call the framework JUNIPR: "Jets from UNsupervised Interpretable PRobabilistic models". In this approach, the set of particle momenta composing a jet are clustered into a binary tree that the neural network examines sequentially. Training is unsupervised and unrestricted: the network could decide that the data bears little correspondence to the chosen tree structure. However, when there is a correspondence, the network's output along the tree has a direct physical interpretation. JUNIPR models can perform discrimination tasks, through the statistically optimal likelihood-ratio test, and they permit visualizations of discrimination power at each branching in a jet's tree. Additionally, JUNIPR models provide a probability distribution from which events can be drawn, providing a data-driven Monte Carlo generator. As a third application, JUNIPR models can reweight events from one (e.g. simulated) data set to agree with distributions from another (e.g. experimental) data set.Comment: 37 pages, 24 figure

    Deepfake detection and low-resource language speech recognition using deep learning

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    While deep learning algorithms have made significant progress in automatic speech recognition and natural language processing, they require a significant amount of labelled training data to perform effectively. As such, these applications have not been extended to languages that have only limited amount of data available, such as extinct or endangered languages. Another problem caused by the rise of deep learning is that individuals with malicious intents have been able to leverage these algorithms to create fake contents that can pose serious harm to security and public safety. In this work, we explore the solutions to both of these problems. First, we investigate different data augmentation methods and acoustic architecture designs to improve automatic speech recognition performance on low-resource languages. Data augmentation for audio often involves changing the characteristic of the audio without modifying the ground truth. For example, different background noise can be added to an utterance while maintaining the content of the speech. We also explored how different acoustic model paradigms and complexity affect performance on low-resource languages. These methods are evaluated on Seneca, an endangered language spoken by a Native American tribe, and Iban, a low-resource language spoken in Malaysia and Brunei. Secondly, we explore methods to determine speaker identification and audio spoofing detection. A spoofing attack involves using either a text-to-speech voice conversion application to generate audio that mimic the identity of a target speaker. These methods are evaluated on the ASVSpoof 2019 Logical Access dataset containing audio generated using various methods of voice conversion and text-to-speech synthesis

    CHORUS Deliverable 2.1: State of the Art on Multimedia Search Engines

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    Based on the information provided by European projects and national initiatives related to multimedia search as well as domains experts that participated in the CHORUS Think-thanks and workshops, this document reports on the state of the art related to multimedia content search from, a technical, and socio-economic perspective. The technical perspective includes an up to date view on content based indexing and retrieval technologies, multimedia search in the context of mobile devices and peer-to-peer networks, and an overview of current evaluation and benchmark inititiatives to measure the performance of multimedia search engines. From a socio-economic perspective we inventorize the impact and legal consequences of these technical advances and point out future directions of research

    Survey of the State of the Art in Natural Language Generation: Core tasks, applications and evaluation

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    This paper surveys the current state of the art in Natural Language Generation (NLG), defined as the task of generating text or speech from non-linguistic input. A survey of NLG is timely in view of the changes that the field has undergone over the past decade or so, especially in relation to new (usually data-driven) methods, as well as new applications of NLG technology. This survey therefore aims to (a) give an up-to-date synthesis of research on the core tasks in NLG and the architectures adopted in which such tasks are organised; (b) highlight a number of relatively recent research topics that have arisen partly as a result of growing synergies between NLG and other areas of artificial intelligence; (c) draw attention to the challenges in NLG evaluation, relating them to similar challenges faced in other areas of Natural Language Processing, with an emphasis on different evaluation methods and the relationships between them.Comment: Published in Journal of AI Research (JAIR), volume 61, pp 75-170. 118 pages, 8 figures, 1 tabl

    Novel Pitch Detection Algorithm With Application to Speech Coding

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    This thesis introduces a novel method for accurate pitch detection and speech segmentation, named Multi-feature, Autocorrelation (ACR) and Wavelet Technique (MAWT). MAWT uses feature extraction, and ACR applied on Linear Predictive Coding (LPC) residuals, with a wavelet-based refinement step. MAWT opens the way for a unique approach to modeling: although speech is divided into segments, the success of voicing decisions is not crucial. Experiments demonstrate the superiority of MAWT in pitch period detection accuracy over existing methods, and illustrate its advantages for speech segmentation. These advantages are more pronounced for gain-varying and transitional speech, and under noisy conditions

    A Compact CNN-Based Speech Enhancement With Adaptive Filter Design Using Gabor Function And Region-Aware Convolution

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    Speech enhancement (SE) is used in many applications, such as hearing devices, to improve speech intelligibility and quality. Convolutional neural network-based (CNN-based) SE algorithms in literature often employ generic convolutional filters that are not optimized for SE applications. This paper presents a CNN-based SE algorithm with an adaptive filter design (named ‘CNN-AFD’) using Gabor function and region-aware convolution. The proposed algorithm incorporates fixed Gabor functions into convolutional filters to model human auditory processing for improved denoising performance. The feature maps obtained from the Gabor-incorporated convolutional layers serve as learnable guided masks (tuned at backpropagation) for generating adaptive custom region-aware filters. The custom filters extract features from speech regions (i.e., ‘region-aware’) while maintaining translation-invariance. To reduce the high cost of inference of the CNN, skip convolution and activation analysis-wise pruning are explored. Employing skip convolution allowed the training time per epoch to be reduced by close to 40%. Pruning of neurons with high numbers of zero activations complements skip convolution and significantly reduces model parameters by more than 30%. The proposed CNN-AFD outperformed all four CNN-based SE baseline algorithms (i.e., a CNN-based SE employing generic filters, a CNN-based SE without region-aware convolution, a CNN-based SE trained with complex spectrograms and a CNN-based SE processing in the time-domain) with an average of 0.95, 1.82 and 0.82 in short-time objective intelligibility (STOI), perceptual evaluation of speech quality (PESQ) and logarithmic spectral distance (LSD) scores, respectively, when tasked to denoise speech contaminated with NOISEX-92 noises at −5, 0 and 5 dB signal-to-noise ratios (SNRs)
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