Codec Detection from Speech

Tato práce se zabývá detekcí kodeků z komprimovaného řečového signálu. Cílem bylo zjistit, jaké charakteristiky rozlišují jednotlivé kodeky a následně vytvořit prostředí vhodné pro experimenty s různými typy a konfiguracemi klasifikátorů. Použity byly Support vector machines a především neuronové sítě, které byly vytvořeny pomocí nástroje Keras. Hlavním přínosem této práce je experimentální část, ve které je analyzován vliv různých parametrů neuronové sítě. Po nalezení nejvhodnější kombinace parametrů dosáhla síť přesnosti klasifikace přes 98% na testovací sadě obsahující data z 6 kodeků.This thesis deals with codec detection from compressed speech signal. The primary goal was to identify which features distinguish selected codecs, and then create an environment facilitating experiments with various types of classifiers and their configurations. Support vector machines and neural networks, modeled using the Keras library, were used. The main contribution of this work is the experimental part, in which the effects of the neural networks parameters are discussed. After tuning the parameters and finding their optimal values, the network achieved accuracy over 98% on a test set comprising data from six different codecs.

Tatouage du flux compressé MPEG-4 AVC

La présente thèse aborde le sujet de tatouage du flux MPEG-4 AVC sur ses deux volets théoriques et applicatifs en considérant deux domaines applicatifs à savoir la protection du droit d auteur et la vérification de l'intégrité du contenu. Du point de vue théorique, le principal enjeu est de développer un cadre de tatouage unitaire en mesure de servir les deux applications mentionnées ci-dessus. Du point de vue méthodologique, le défi consiste à instancier ce cadre théorique pour servir les applications visées. La première contribution principale consiste à définir un cadre théorique pour le tatouage multi symboles à base de modulation d index de quantification (m-QIM). La règle d insertion QIM a été généralisée du cas binaire au cas multi-symboles et la règle de détection optimale (minimisant la probabilité d erreur à la détection en condition du bruit blanc, additif et gaussien) a été établie. Il est ainsi démontré que la quantité d information insérée peut être augmentée par un facteur de log2m tout en gardant les mêmes contraintes de robustesse et de transparence. Une quantité d information de 150 bits par minutes, soit environ 20 fois plus grande que la limite imposée par la norme DCI est obtenue. La deuxième contribution consiste à spécifier une opération de prétraitement qui permet d éliminer les impactes du phénomène du drift (propagation de la distorsion) dans le flux compressé MPEG-4 AVC. D abord, le problème a été formalisé algébriquement en se basant sur les expressions analytiques des opérations d encodage. Ensuite, le problème a été résolu sous la contrainte de prévention du drift. Une amélioration de la transparence avec des gains de 2 dB en PSNR est obtenueThe present thesis addresses the MPEG-4 AVC stream watermarking and considers two theoretical and applicative challenges, namely ownership protection and content integrity verification.From the theoretical point of view, the thesis main challenge is to develop a unitary watermarking framework (insertion/detection) able to serve the two above mentioned applications in the compressed domain. From the methodological point of view, the challenge is to instantiate this theoretical framework for serving the targeted applications. The thesis first main contribution consists in building the theoretical framework for the multi symbol watermarking based on quantization index modulation (m-QIM). The insertion rule is analytically designed by extending the binary QIM rule. The detection rule is optimized so as to ensure minimal probability of error under additive white Gaussian noise distributed attacks. It is thus demonstrated that the data payload can be increased by a factor of log2m, for prescribed transparency and additive Gaussian noise power. A data payload of 150 bits per minute, i.e. about 20 times larger than the limit imposed by the DCI standard, is obtained. The thesis second main theoretical contribution consists in specifying a preprocessing MPEG-4 AVC shaping operation which can eliminate the intra-frame drift effect. The drift represents the distortion spread in the compressed stream related to the MPEG encoding paradigm. In this respect, the drift distortion propagation problem in MPEG-4 AVC is algebraically expressed and the corresponding equations system is solved under drift-free constraints. The drift-free shaping results in gain in transparency of 2 dB in PSNREVRY-INT (912282302) / SudocSudocFranceF

Multi-modal Video Content Understanding

Video is an important format of information. Humans use videos for a variety of purposes such as entertainment, education, communication, information sharing, and capturing memories. To this date, humankind accumulated a colossal amount of video material online which is freely available. Manual processing at this scale is simply impossible. To this end, many research efforts have been dedicated to the automatic processing of video content. At the same time, human perception of the world is multi-modal. A human uses multiple senses to understand the environment and objects, and their interactions. When watching a video, we perceive the content via both audio and visual modalities, and removing one of these modalities results in less immersive experience. Similarly, if information in both modalities does not correspond, it may create a sense of dissonance. Therefore, joint modelling of multiple modalities (such as audio, visual, and text) within one model is an active research area. In the last decade, the fields of automatic video understanding and multi-modal modelling have seen exceptional progress due to the ubiquitous success of deep learning models and, more recently, transformer-based architectures in particular. Our work draws on these advances and pushes the state-of-the-art of multi-modal video understanding forward. Applications of automatic multi-modal video processing are broad and exciting! For instance, the content-based textual description of a video (video captioning) may allow a visually- or auditory-impaired person to understand the content and, thus, engage in brighter social interactions. However, prior work in video content description relies on the visual input alone, missing vital information only available in the audio stream. To this end, we proposed two novel multi-modal transformer models that encode audio and visual interactions simultaneously. More specifically, first, we introduced a late-fusion multi-modal transformer that is highly modular and allows the processing of an arbitrary set of modalities. Second, an efficient bi-modal transformer was presented to encode audio-visual cues starting from the lower network layers allowing more rich audio-visual features and stronger performance as a result. Another application is the automatic visually-guided sound generation that might help professional sound (foley) designers who spend hours searching a database for relevant audio for a movie scene. Previous approaches for automatic conditional audio generation support only one class (e. g. “dog barking”), while real-life applications may require generation for hundreds of data classes and one would need to train one model for every data class which can be infeasible. To bridge this gap, we introduced a novel two-stage model that, first, efficiently encodes audio as a set of codebook vectors (i. e. trains to make “building blocks”) and, then, learns to sample these audio vectors given visual inputs to make a relevant audio track for this visual input. Moreover, we studied the automatic evaluation of the conditional audio generation model and proposed metrics that measure both quality and relevance of the generated samples. Finally, as video editing is becoming more common among non-professionals due to the increased popularity of such services as YouTube, automatic assistance during video editing grows in demand, e. g. off-sync detection between audio and visual tracks. Prior work in audio-visual synchronization was devoted to solving the task on lip-syncing datasets with “dense” signals, such as interviews and presentations. In such videos, synchronization cues occur “densely” across time, and it is enough to process just a few tens of a second to synchronize the tracks. In contrast, opendomain videos mostly have only “sparse” cues that occur just once in a seconds-long video clip (e. g. “chopping wood”). To address this, we: a) proposed a novel dataset with “sparse” sounds; b) designed a model which can efficiently encode seconds-long audio-visual tracks in a small set of “learnable selectors” that is, then, used for synchronization. In addition, we explored the temporal artefacts that common audio and video compression algorithms leave in data streams. To prevent a model from learning to rely on these artefacts, we introduced a list of recommendations on how to mitigate them. This thesis provides the details of the proposed methodologies as well as a comprehensive overview of advances in relevant fields of multi-modal video understanding. In addition, we provide a discussion of potential research directions that can bring significant contributions to the field

MediaSync: Handbook on Multimedia Synchronization

This book provides an approachable overview of the most recent advances in the fascinating field of media synchronization (mediasync), gathering contributions from the most representative and influential experts. Understanding the challenges of this field in the current multi-sensory, multi-device, and multi-protocol world is not an easy task. The book revisits the foundations of mediasync, including theoretical frameworks and models, highlights ongoing research efforts, like hybrid broadband broadcast (HBB) delivery and users' perception modeling (i.e., Quality of Experience or QoE), and paves the way for the future (e.g., towards the deployment of multi-sensory and ultra-realistic experiences). Although many advances around mediasync have been devised and deployed, this area of research is getting renewed attention to overcome remaining challenges in the next-generation (heterogeneous and ubiquitous) media ecosystem. Given the significant advances in this research area, its current relevance and the multiple disciplines it involves, the availability of a reference book on mediasync becomes necessary. This book fills the gap in this context. In particular, it addresses key aspects and reviews the most relevant contributions within the mediasync research space, from different perspectives. Mediasync: Handbook on Multimedia Synchronization is the perfect companion for scholars and practitioners that want to acquire strong knowledge about this research area, and also approach the challenges behind ensuring the best mediated experiences, by providing the adequate synchronization between the media elements that constitute these experiences

Proceedings of the 8th Workshop on Detection and Classification of Acoustic Scenes and Events (DCASE 2023)

This volume gathers the papers presented at the Detection and Classification of Acoustic Scenes and Events 2023 Workshop (DCASE2023), Tampere, Finland, during 21–22 September 2023

XXIII Congreso Argentino de Ciencias de la Computación - CACIC 2017 : Libro de actas

Trabajos presentados en el XXIII Congreso Argentino de Ciencias de la Computación (CACIC), celebrado en la ciudad de La Plata los días 9 al 13 de octubre de 2017, organizado por la Red de Universidades con Carreras en Informática (RedUNCI) y la Facultad de Informática de la Universidad Nacional de La Plata (UNLP).Red de Universidades con Carreras en Informática (RedUNCI