Automated speech and audio analysis for semantic access to multimedia

The deployment and integration of audio processing tools can enhance the semantic annotation of multimedia content, and as a consequence, improve the effectiveness of conceptual access tools. This paper overviews the various ways in which automatic speech and audio analysis can contribute to increased granularity of automatically extracted metadata. A number of techniques will be presented, including the alignment of speech and text resources, large vocabulary speech recognition, key word spotting and speaker classification. The applicability of techniques will be discussed from a media crossing perspective. The added value of the techniques and their potential contribution to the content value chain will be illustrated by the description of two (complementary) demonstrators for browsing broadcast news archives

Contrastive Learning-Based Audio to Lyrics Alignment for Multiple Languages

Lyrics alignment gained considerable attention in recent years. State-of-the-art systems either re-use established speech recognition toolkits, or design end-to-end solutions involving a Connectionist Temporal Classification (CTC) loss. However, both approaches suffer from specific weaknesses: toolkits are known for their complexity, and CTC systems use a loss designed for transcription which can limit alignment accuracy. In this paper, we use instead a contrastive learning procedure that derives cross-modal embeddings linking the audio and text domains. This way, we obtain a novel system that is simple to train end-to-end, can make use of weakly annotated training data, jointly learns a powerful text model, and is tailored to alignment. The system is not only the first to yield an average absolute error below 0.2 seconds on the standard Jamendo dataset but it is also robust to other languages, even when trained on English data only. Finally, we release word-level alignments for the JamendoLyrics Multi-Lang dataset.Comment: 5 pages, accepted at the International Conference on Acoustics, Speech, and Signal Processing (ICASSP) 202

Pushing the envelope: Evaluating speech rhythm with different envelope extraction techniques

The amplitude of the speech signal varies over time, and the speech envelope is an attempt to characterise this variation in the form of an acoustic feature. Although tacitly assumed, the similarity between the speech envelope-derived time series and that of phonetic objects (e.g., vowels) remains empirically unestablished. The current paper, therefore, evaluates several speech envelope extraction techniques, such as the Hilbert transform, by comparing different acoustic landmarks (e.g., peaks in the speech envelope) with manual phonetic annotation in a naturalistic and diverse dataset. Joint speech tasks are also introduced to determine which acoustic landmarks are most closely coordinated when voices are aligned. Finally, the acoustic landmarks are evaluated as predictors for the temporal characterisation of speaking style using classification tasks. The landmark that performed most closely to annotated vowel onsets was peaks in the first derivative of a human audition-informed envelope, consistent with converging evidence from neural and behavioural data. However, differences also emerged based on language and speaking style. Overall, the results show that both the choice of speech envelope extraction technique and the form of speech under study affect how sensitive an engineered feature is at capturing aspects of speech rhythm, such as the timing of vowels

A cross-lingual adaptation approach for rapid development of speech recognizers for learning disabled users

Building a voice-operated system for learning disabled users is a difficult task that requires a considerable amount of time and effort. Due to the wide spectrum of disabilities and their different related phonopathies, most approaches available are targeted to a specific pathology. This may improve their accuracy for some users, but makes them unsuitable for others. In this paper, we present a cross-lingual approach to adapt a general-purpose modular speech recognizer for learning disabled people. The main advantage of this approach is that it allows rapid and cost-effective development by taking the already built speech recognition engine and its modules, and utilizing existing resources for standard speech in different languages for the recognition of the users’ atypical voices. Although the recognizers built with the proposed technique obtain lower accuracy rates than those trained for specific pathologies, they can be used by a wide population and developed more rapidly, which makes it possible to design various types of speech-based applications accessible to learning disabled users.This research was supported by the project ‘Favoreciendo la vida autónoma de discapacitados intelectuales con problemas de comunicación oral mediante interfaces personalizados de reconocimiento automático del habla’, financed by the Centre of Initiatives for Development Cooperation (Centro de Iniciativas de Cooperación al Desarrollo, CICODE), University of Granada, Spain. This research was supported by the Student Grant Scheme 2014 (SGS) at the Technical University of Liberec

A computational model for studying L1’s effect on L2 speech learning

abstract: Much evidence has shown that first language (L1) plays an important role in the formation of L2 phonological system during second language (L2) learning process. This combines with the fact that different L1s have distinct phonological patterns to indicate the diverse L2 speech learning outcomes for speakers from different L1 backgrounds. This dissertation hypothesizes that phonological distances between accented speech and speakers' L1 speech are also correlated with perceived accentedness, and the correlations are negative for some phonological properties. Moreover, contrastive phonological distinctions between L1s and L2 will manifest themselves in the accented speech produced by speaker from these L1s. To test the hypotheses, this study comes up with a computational model to analyze the accented speech properties in both segmental (short-term speech measurements on short-segment or phoneme level) and suprasegmental (long-term speech measurements on word, long-segment, or sentence level) feature space. The benefit of using a computational model is that it enables quantitative analysis of L1's effect on accent in terms of different phonological properties. The core parts of this computational model are feature extraction schemes to extract pronunciation and prosody representation of accented speech based on existing techniques in speech processing field. Correlation analysis on both segmental and suprasegmental feature space is conducted to look into the relationship between acoustic measurements related to L1s and perceived accentedness across several L1s. Multiple regression analysis is employed to investigate how the L1's effect impacts the perception of foreign accent, and how accented speech produced by speakers from different L1s behaves distinctly on segmental and suprasegmental feature spaces. Results unveil the potential application of the methodology in this study to provide quantitative analysis of accented speech, and extend current studies in L2 speech learning theory to large scale. Practically, this study further shows that the computational model proposed in this study can benefit automatic accentedness evaluation system by adding features related to speakers' L1s.Dissertation/ThesisDoctoral Dissertation Speech and Hearing Science 201

Evaluating pause particles and their functions in natural and synthesized speech in laboratory and lecture settings

Pause-internal phonetic particles (PINTs) comprise a variety of phenomena including: phonetic-acoustic silence, inhalation and exhalation breath noises, filler particles “uh” and “um” in English, tongue clicks, and many others. These particles are omni-present in spontaneous speech, however, they are under-researched in both natural speech and synthetic speech. The present work explores the influence of PINTs in small-context recall experiments, develops a bespoke speech synthesis system that incorporates the PINTs pattern of a single speaker, and evaluates the influence of PINTs on recall for larger material lengths, namely university lectures. The benefit of PINTs on recall has been documented in natural speech in small-context laboratory settings, however, this area of research has been under-explored for synthetic speech. We devised two experiments to evaluate if PINTs have the same recall benefit for synthetic material that is found with natural material. In the first experiment, we evaluated the recollection of consecutive missing digits for a randomized 7-digit number. Results indicated that an inserted silence improved recall accuracy for digits immediately following. In the second experiment, we evaluated sentence recollection. Results indicated that sentences preceded by an inhalation breath noise were better recalled than those with no inhalation. Together, these results reveal that in single-sentence laboratory settings PINTs can improve recall for synthesized speech. The speech synthesis systems used in the small-context recall experiments did not provide much freedom in terms of controlling PINT type or location. Therefore, we endeavoured to develop bespoke speech synthesis systems. Two neural text-to-speech (TTS) systems were created: one that used PINTs annotation labels in the training data, and another that did not include any PINTs labeling in the training material. The first system allowed fine-tuned control for inserting PINTs material into the rendered material. The second system produced PINTs probabilistally. To the best of our knowledge, these are the first TTS systems to render tongue clicks. Equipped with greater control of synthesized PINTs, we returned to evaluating the recall benefit of PINTs. This time we evaluated the influence of PINTs on the recollection of key information in lectures, an ecologically valid task that focused on larger material lengths. Results indicated that key information that followed PINTs material was less likely to be recalled. We were unable to replicate the benefits of PINTs found in the small-context laboratory settings. This body of work showcases that PINTs improve recall for TTS in small-context environments just like previous work had indicated for natural speech. Additionally, we’ve provided a technological contribution via a neural TTS system that exerts finer control over PINT type and placement. Lastly, we’ve shown the importance of using material rendered by speech synthesis systems in perceptual studies.This research was funded by the Deutsche Forschungsgemeinschaft (DFG, German Research Foundation) within the project “Pause-internal phonetic particles in speech communication” (project number: 418659027; project IDs: MO 597/10-1 and TR 468/3-1). Associate member of SFB1102 “Information Density and Linguistic Encoding” (project number: 232722074)

Word-level embeddings for cross-task transfer learning in speech processing

Recent breakthroughs in deep learning often rely on representation learning and knowledge transfer. In recent years, unsupervised and self-supervised techniques for learning speech representation were developed to foster automatic speech recognition. Up to date, most of these approaches are task-specific and designed for within-task transfer learning between different datasets or setups of a particular task. In turn, learning task-independent representation of speech and cross-task applications of transfer learning remain less common. Here, we introduce an encoder capturing word-level representations of speech for cross-task transfer learning. We demonstrate the application of the pre-trained encoder in four distinct speech and audio processing tasks: (i) speech enhancement, (ii) language identification, (iii) speech, noise, and music classification, and (iv) speaker identification. In each task, we compare the performance of our cross-task transfer learning approach to task-specific baselines. Our results show that the speech representation captured by the encoder through the pre-training is transferable across distinct speech processing tasks and datasets. Notably, even simple applications of our pre-trained encoder outperformed task-specific methods, or were comparable, depending on the task

The timing bottleneck: Why timing and overlap are mission-critical for conversational user interfaces, speech recognition and dialogue systems

Speech recognition systems are a key intermediary in voice-driven human-computer interaction. Although speech recognition works well for pristine monologic audio, real-life use cases in open-ended interactive settings still present many challenges. We argue that timing is mission-critical for dialogue systems, and evaluate 5 major commercial ASR systems for their conversational and multilingual support. We find that word error rates for natural conversational data in 6 languages remain abysmal, and that overlap remains a key challenge (study 1). This impacts especially the recognition of conversational words (study 2), and in turn has dire consequences for downstream intent recognition (study 3). Our findings help to evaluate the current state of conversational ASR, contribute towards multidimensional error analysis and evaluation, and identify phenomena that need most attention on the way to build robust interactive speech technologies

The analysis of breathing and rhythm in speech

Speech rhythm can be described as the temporal patterning by which speech events, such as vocalic onsets, occur. Despite efforts to quantify and model speech rhythm across languages, it remains a scientifically enigmatic aspect of prosody. For instance, one challenge lies in determining how to best quantify and analyse speech rhythm. Techniques range from manual phonetic annotation to the automatic extraction of acoustic features. It is currently unclear how closely these differing approaches correspond to one another. Moreover, the primary means of speech rhythm research has been the analysis of the acoustic signal only. Investigations of speech rhythm may instead benefit from a range of complementary measures, including physiological recordings, such as of respiratory effort. This thesis therefore combines acoustic recording with inductive plethysmography (breath belts) to capture temporal characteristics of speech and speech breathing rhythms. The first part examines the performance of existing phonetic and algorithmic techniques for acoustic prosodic analysis in a new corpus of rhythmically diverse English and Mandarin speech. The second part addresses the need for an automatic speech breathing annotation technique by developing a novel function that is robust to the noisy plethysmography typical of spontaneous, naturalistic speech production. These methods are then applied in the following section to the analysis of English speech and speech breathing in a second, larger corpus. Finally, behavioural experiments were conducted to investigate listeners' perception of speech breathing using a novel gap detection task. The thesis establishes the feasibility, as well as limits, of automatic methods in comparison to manual annotation. In the speech breathing corpus analysis, they help show that speakers maintain a normative, yet contextually adaptive breathing style during speech. The perception experiments in turn demonstrate that listeners are sensitive to the violation of these speech breathing norms, even if unconsciously so. The thesis concludes by underscoring breathing as a necessary, yet often overlooked, component in speech rhythm planning and production