Conversational Speech Synthesis (CSS) aims to produce speech appropriate for oral communication. However, the complexity of context dependency modeling poses significant challenges in the field of CSS, especially the mutual psychological influence between interlocutors. Previous studies have verified that prior commonsense knowledge helps machines understand subtle psychological information (e.g., feelings and intentions) in spontaneous oral dialogues. Therefore, to enhance context understanding and improve the naturalness of synthesized speech, we propose a novel conversational speech synthesis system (CMCU-CSS) that incorporates the Commonsense-based Multi-modal Context Understanding (CMCU) module to model the dynamic emotional interaction among interlocutors. Specifically, we first utilize three implicit states (intent state, internal state and external state) in CMCU to model the context dependency between inter/intra speakers with the help of commonsense knowledge. Furthermore, we infer emotion vectors from the fusion of these implicit states and multi-modal features to enhance the emotion discriminability of synthesized speech. This is the first attempt to combine commonsense knowledge with conversational speech synthesis, and its effect in terms of emotion discriminability of synthetic speech is evaluated by emotion recognition in conversation task. The results of subjective and objective evaluations demonstrate that the CMCU-CSS model achieves more natural speech with context-appropriate emotion and is equipped with the best emotion discriminability, surpassing that of other conversational speech synthesis models.
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The human brain can easily focus on one speaker and suppress others in scenarios such as a cocktail party. Recently, researchers found that auditory attention can be decoded from the electroencephalogram (EEG) data. However, most existing deep learning methods are difficult to use prior knowledge of different views (that is attended speech and EEG are task-related views) and extract an unsatisfactory representation. Inspired by Broadbent's filter model, we decode auditory attention in a multi-view paradigm and extract the most relevant and important information utilizing the missing view. Specifically, we propose an auditory attention decoding (AAD) method based on multi-view VAE with task-related multi-view contrastive (TMC) learning. Employing TMC learning in multi-view VAE can utilize the missing view to accumulate prior knowledge of different views into the fusion of representation, and extract the approximate task-related representation. We examine our method on two popular AAD datasets, and demonstrate the superiority of our method by comparing it to the state-of-the-art method.
Conversational Text-to-speech Synthesis (TTS) aims to generate speech with proper style in the user-agent conversation scenario. Although previous works have explored modeling the context in the dialogue history to provide style information for the agent, there are still deficiencies in modeling the role-aware multi-modal context. Moreover, previous works ignore the emotional dependencies between the user and the agent, which includes: 1) agent understands emotional states of users, and 2) agent expresses proper emotion in the generated speech. In this work, we propose an Emotionally Situated Text-to-speech Synthesis (EmoSit-TTS) framework to understand users' semantics and subtle emotional states, and generate speech with proper speaking style and emotional expression in the user-agent conversation. Experiments on the DailyTalk dataset show the superiority of our proposed framework for the user-agent conversational TTS, especially in terms of emotion-aware expressiveness, which outperforms other state-of-the-art methods by 0.69 on MOS. Demos of our proposed framework are available at https://anonydemo.github.io.
Emo-DNA: Emotion Decoupling and Alignment Learning for Cross-Corpus Speech Emotion Recognition
PDF ↗Cross-corpus speech emotion recognition (SER) seeks to generalize the ability of inferring speech emotion from a well-labeled corpus to an unlabeled one, which is a rather challenging task due to the significant discrepancy between two corpora. Existing methods, typically based on unsupervised domain adaptation (UDA), struggle to learn corpus-invariant features by global distribution alignment, but unfortunately, the resulting features are mixed with corpus-specific features or not class-discriminative. To tackle these challenges, we propose a novel Emotion Decoupling aNd Alignment learning framework (EMO-DNA) for cross-corpus SER, a novel UDA method to learn emotion-relevant corpus-invariant features. The novelties of EMO-DNA are two-fold: contrastive emotion decoupling and dual-level emotion alignment. On one hand, our contrastive emotion decoupling achieves decoupling learning via a contrastive decoupling loss to strengthen the separability of emotion-relevant features from corpus-specific ones. On the other hand, our dual-level emotion alignment introduces an adaptive threshold pseudo-labeling to select confident target samples for class-level alignment, and performs corpus-level alignment to jointly guide model for learning class-discriminative corpus-invariant features across corpora. Extensive experimental results demonstrate the superior performance of EMO-DNA over the state-of-the-art methods in several cross-corpus scenarios. Source code is available at https://github.com/Jiaxin-Ye/Emo-DNA.
In this paper, we introduce a novel approach to address the task of synthesizing speech from silent videos of any in-the-wild speaker solely based on lip movements. The traditional approach of directly generating speech from lip videos faces the challenge of not being able to learn a robust language model from speech alone, resulting in unsatisfactory outcomes. To overcome this issue, we propose incorporating noisy text supervision using a state-of-the-art lip-to-text network that instills language information into our model. The noisy text is generated using a pre-trained lip-to-text model, enabling our approach to work without text annotations during inference. We design a visual text-to-speech network that utilizes the visual stream to generate accurate speech, which is in-sync with the silent input video. We perform extensive experiments and ablation studies, demonstrating our approach's superiority over the current state-of-the-art methods on various benchmark datasets. Further, we demonstrate an essential practical application of our method in assistive technology by generating speech for an ALS patient who has lost the voice but can make mouth movements. Our demo video, code, and additional details can be found at http://cvit.iiit.ac.in/research/projects/cvit-projects/ms-l2s-itw.
Speech-driven 3D face animation technique, extending its applications to various multimedia fields.Previous research has generated promising realistic lip movements and facial expressions from audio signals. However, traditional regression models solely driven by data face several essential problems, such as difficulties in accessing precise labels and domain gaps between different modalities, leading to unsatisfactory results lacking precision and coherence.To enhance the visual accuracy of generated lip movement while reducing the dependence on labeled data, we propose a novel framework SelfTalk, by involving self-supervision in a cross-modals network system to learn 3D talking faces. The framework constructs a network system consisting of three modules: facial animator, speech recognizer, and lip-reading interpreter. The core of SelfTalk is a commutative training diagram that facilitates compatible features exchange among audio, text, and lip shape, enabling our models to learn the intricate connection between these factors. The proposed framework leverages the knowledge learned from the lip-reading interpreter to generate more plausible lip shapes. Extensive experiments and user studies demonstrate that our proposed approach achieves state-of-the-art performance both qualitatively and quantitatively. We recommend watching the supplementary video.
While recent research has made significant progress in speech-driven talking face generation, the quality of the generated video still lags behind that of real recordings. One reason for this is the use of handcrafted intermediate representations like facial landmarks and 3DMM coefficients, which are designed based on human knowledge and are insufficient to precisely describe facial movements. Additionally, these methods require an external pretrained model for extracting these representations, whose performance sets an upper bound on talking face generation. To address these limitations, we propose a novel method called DAE-Talker that leverages data-driven latent representations obtained from a diffusion autoencoder (DAE). DAE contains an image encoder that encodes an image into a latent vector and a DDIM-based image decoder that reconstructs the image from it. We train our DAE on talking face video frames and then extract their latent representations as the training target for a Conformer-based speech2latent model. During inference, DAE-Talker first predicts the latents from speech and then generates the video frames with the image decoder in DAE from the predicted latents. This allows DAE-Talker to synthesize full video frames and produce natural head movements that align with the content of speech, rather than relying on a predetermined head pose from a template video. We also introduce pose modelling in speech2latent for pose controllability. Additionally, we propose a novel method for generating continuous video frames with the DDIM-based image decoder trained on individual frames, eliminating the need for modelling the joint distribution of consecutive frames directly. Our experiments show that DAE-Talker outperforms existing popular methods in lip-sync, video fidelity, and pose naturalness. We also conduct ablation studies to analyze the effectiveness of the proposed techniques and demonstrate the pose controllability of DAE-Talker.
Audio-visual speaker diarization refers to the task of identifying "who spoke when" by using both audio and video data. Although previous fusion-based approaches have shown exceptional performance over audio-only methods, they have mainly focused on high-quality data and have not accounted for the impacts of acoustic noise or missing faces. To address these limitations, we propose a novel uncertainty-aware end-to-end audio-visual speaker diarization (UAV-SD) approach in this paper. Our approach leverages both framewise inter- and intra-modal confidence to achieve more effective and robust speaker diarization. By taking into account the uncertainty of the data, UAV-SD can achieve better diarization performance even in noisy or low-quality recordings. Additionally, our approach is compatible with multi-channel audio signals without the need to retrain the model, making it a more versatile solution. To evaluate the effectiveness of our approach, we conduct extensive experiments on the Multi-modal Information Based Speech Processing (MISP) 2022 Challenge datasets which consist of far-field audio and video data. The results show that UAV-SD is able to yield significant performance gains compared to baseline methods for both single and multi-channel data, demonstrating its effectiveness in real-world scenarios.
Cultural Self-Adaptive Multimodal Gesture Generation Based on Multiple Culture Gesture Dataset
PDF ↗Co-speech gesture generation is essential for multimodal chatbots and agents. Previous research extensively studies the relationship between text, audio, and gesture. Meanwhile, to enhance cross-culture communication, culture-specific gestures are crucial for chatbots to learn cultural differences and incorporate cultural cues. However, culture-specific gesture generation faces two challenges: lack of large-scale, high-quality gesture datasets that include diverse cultural groups, and lack of generalization across different cultures. Therefore, in this paper, we first introduce a Multiple Culture Gesture Dataset (MCGD), the largest freely available gesture dataset to date. It consists of ten different cultures, over 200 speakers, and 10,000 segmented sequences. We further propose a Cultural Self-adaptive Gesture Generation Network (CSGN) that takes multimodal relationships into consideration while generating gestures using a cascade architecture and learnable dynamic weight. The CSGN adaptively generates gestures with different cultural characteristics without the need to retrain a new network. It extracts cultural features from the multimodal inputs or a cultural style embedding space with a designated culture. We broadly evaluate our method across four large-scale benchmark datasets. Empirical results show that our method achieves multiple cultural gesture generation and improves comprehensiveness of multimodal inputs. Our method improves the state-of-the-art average FGD from 53.7 to 48.0 and culture deception rate (CDR) from 33.63% to 39.87%.
SpeechTripleNet: End-to-End Disentangled Speech Representation Learning for Content, Timbre and Prosody
PDF ↗Disentangled speech representation learning aims to separate different factors of variation from speech into disjoint representations. This paper focuses on disentangling speech into representations for three factors: spoken content, speaker timbre, and speech prosody. Many previous methods for speech disentanglement have focused on separating spoken content and speaker timbre. However, the lack of explicit modeling of prosodic information leads to degraded speech generation performance and uncontrollable prosody leakage into content and/or speaker representations. While some recent methods have utilized explicit speaker labels or pre-trained models to facilitate triple-factor disentanglement, there are no end-to-end methods to simultaneously disentangle three factors using only unsupervised or self-supervised learning objectives. This paper introduces SpeechTripleNet, an end-to-end method to disentangle speech into representations for content, timbre, and prosody. Based on VAE, SpeechTripleNet restricts the structures of the latent variables and the amount of information captured in them to induce disentanglement. It is a pure unsupervised/self-supervised learning method that only requires speech data and no additional labels. Our qualitative and quantitative results demonstrate that SpeechTripleNet is effective in achieving triple-factor speech disentanglement, as well as controllable speech editing concerning different factors.
With the proliferation of video platforms on the internet, recording musical performances by mobile devices has become commonplace. However, these recordings often suffer from degradation such as noise and reverberation, which negatively impact the listening experience. Consequently, the necessity for music audio enhancement (referred to as music enhancement from this point onward), involving the transformation of degraded audio recordings into pristine high-quality music, has surged to augment the auditory experience. To address this issue, we propose a music enhancement system based on the Conformer architecture that has demonstrated outstanding performance in speech enhancement tasks. Our approach explores the attention mechanisms of the Conformer and examines their performance to discover the best approach for the music enhancement task. Our experimental results show that our proposed model achieves state-of-the-art performance on single-stem music enhancement. Furthermore, our system can perform general music enhancement with multi-track mixtures, which has not been examined in previous work. Audio samples enhanced with our system are available at: https://tinyurl.com/smpls9999
Denoising diffusion probabilistic models (DDPMs) have shown promising performance for speech synthesis. However, a large number of iterative steps are required to achieve high sample quality, which restricts the inference speed. Maintaining sample quality while increasing sampling speed has become a challenging task. In this paper, we propose a Consistency Model-based Speech synthesis method, CoMoSpeech, which achieve speech synthesis through a single diffusion sampling step while achieving high audio quality. The consistency constraint is applied to distill a consistency model from a well-designed diffusion-based teacher model, which ultimately yields superior performances in the distilled CoMoSpeech. Our experiments show that by generating audio recordings by a single sampling step, the CoMoSpeech achieves an inference speed more than 150 times faster than real-time on a single NVIDIA A100 GPU, which is comparable to FastSpeech2, making diffusion-sampling based speech synthesis truly practical. Meanwhile, objective and subjective evaluations on text-to-speech and singing voice synthesis show that the proposed teacher models yield the best audio quality, and the one-step sampling based CoMoSpeech achieves the best inference speed with better or comparable audio quality to other conventional multi-step diffusion model baselines. Audio samples and codes are available at https://comospeech.github. https://comospeech.github.io/.
The automatic co-speech gesture generation draws much attention in computer animation. Previous works designed network structures on individual datasets, which resulted in a lack of data volume and generalizability across different motion capture standards. In addition, it is a challenging task due to the weak correlation between speech and gestures. To address these problems, we present UnifiedGesture, a novel diffusion model-based speech-driven gesture synthesis approach, trained on multiple gesture datasets with different skeletons. Specifically, we first present a retargeting network to learn latent homeomorphic graphs for different motion capture standards, unifying the representations of various gestures while extending the dataset. We then capture the correlation between speech and gestures based on a diffusion model architecture using cross-local attention and self-attention to generate better speech-matched and realistic gestures. To further align speech and gesture and increase diversity, we incorporate reinforcement learning on the discrete gesture units with a learned reward function. Extensive experiments show that UnifiedGesture outperforms recent approaches on speech-driven gesture generation in terms of CCA, FGD, and human-likeness.
Keyword spotting (KWS) presents a formidable challenge, particularly in high-noise environments. Traditional denoising algorithms that rely solely on speech have difficulty recovering speech that has been severely corrupted by noise. In this investigation, we develop an adaptive text-informed denoising model to bolster reliable keyword identification in the presence of considerable noise degradation. The whole proposed TE-KWS incorporates a tripartite branch structure, where the speech branch (SB) takes noisy speech as input which provides the raw speech information, the alignment branch (AB) accommodates aligned text input which facilitates accurate restoration of the corresponding speech when text with alignment is preserved, and the text branch (TB) handles unaligned text which prompts the model to autonomously learn the alignment between speech and text. To make the proposed denoising model more beneficial for KWS, following the training of the whole model,the alignment branch (AB) is frozen, and the model is fine-tuned by leveraging its speech restoration and forced alignment capabilities. Subsequently, the input for the text branch (TB) is supplanted with designated keywords, and a heavier denoising penalty is applied on the keywords period, thereby explicitly intensifying the speech restoration ability of the model for keywords. Finally, the Combined Adversarial Domain Adaptation (CADA) is implemented to enhance the robustness of KWS with regard to data pre-and post-speech enhancement (SE). Experimental results indicate that our approach not only markedly ameliorates highly corrupted speech, achieving SOTA performance for marginally corrupted speech, but also bolsters the efficacy and generalizability of prevailing mainstream KWS models.
Air Traffic Control (ATC) is a complicated, time-evolving, and real-time procedure to direct flight operations in a safer and ordered manner. Although enormous data storages are available during air traffic operations for over 40 years, data-driven intelligent application in aviation is still an emerging task due to the safety-critical issue. With the prevalence of the Next Generation ATC system, artificial intelligence (AI) -empowered research topics are attracting increasing attention from both industrial and academic domains and a high-quality dataset naturally becomes the prerequisite for such practices. However, almost all ATC-related datasets are only unimodal for certain tasks, which fails to comprehensively illustrate the traffic situation to further support real-world studies. To address this gap, a multimodal air traffic situation (M2ATS) dataset is constructed to advance AI-related research in the ATC domain, including airspace information, flight plan, trajectory, and speech. M2ATS covers 10362 flights ATC situation data, involving 110000+ utterances (104 hours) with diversity golden text annotations, 16 intents, and 51 slots. Considering the real-world ATC requirements, a total of 10 multimedia-related tasks (24 baselines) are designed to validate the proposed dataset, covering automatic speech recognition, natural language processing, and spatial-temporal data processing. New ATC-related metrics corresponding to ATC applications are proposed in addition to the common metrics to evaluate task performance. Extensive experiment results demonstrate that the selective baselines can achieve designed tasks on this new dataset, and further investigations are also required to address task and data specificities. It is believed that the proposed new dataset is a new practice to advance AI applications to an industrial scene, which not only promotes ATC-related applications but also provides diverse research topics in the common multimedia community.
Voice conversion as the style transfer task applied to speech, refers to converting one person's speech into a new speech that sounds like another person's. Up to now, there has been a lot of research devoted to better implementation of VC tasks. However, a good voice conversion model should not only match the timbre information of the target speaker, but also expressive information such as prosody, pace, pause, etc. In this context, prosody modeling is crucial for achieving expressive voice conversion that sounds natural and convincing. Unfortunately, prosody modeling is important but challenging, especially without text transcriptions. In this paper, we firstly propose a novel voice conversion framework named 'PMVC', which effectively separates and models the content, timbre, and prosodic information from the speech without text transcriptions. Specially, we introduce a new speech augmentation algorithm for robust prosody extraction. And building upon this, mask and predict mechanism is applied in the disentanglement of prosody and content information. The experimental results on the AIShell-3 corpus supports our improvement of naturalness and similarity of converted speech.
SpeechGPT: Empowering Large Language Models with Intrinsic Cross-Modal Conversational Abilities
PDF ↗Multi-modal large language models are regarded as a crucial step towards Artificial General Intelligence (AGI) and have garnered significant interest with the emergence of ChatGPT. However, current speech-language models typically adopt the cascade paradigm, preventing inter-modal knowledge transfer. In this paper, we propose SpeechGPT, a large language model with intrinsic cross-modal conversational abilities, capable of perceiving and generating multi-modal content. With discrete speech representations, we construct SpeechInstruct, the first large-scale cross-modal speech instruction dataset. Additionally, we employ a three-stage training strategy that includes modality-adaptation pre-training, cross-modal instruction fine-tuning, and chain-of-modality instruction fine-tuning. The experimental results demonstrate that SpeechGPT has an impressive capacity to follow cross-modal human instructions and highlight the potential of handling multiple modalities with one model. Code and models are available in https://github.com/0nutation/SpeechGPT. Demos are shown in https://0nutation.github.io/SpeechGPT.github.io/.
Does Listener Gaze in Face-to-Face Interaction Follow the Entropy Rate Constancy Principle: An Empirical Study
PDF ↗It is generally assumed that language (written and spoken) follows the entropy rate constancy (ERC) principle, which states that the information density of a text is constant over time. Recently, this has also been found for nonverbal gestures used in monologue, but it is still unclear whether the ERC principle also applies to listeners’ nonverbal signals. We focus on listeners’ gaze behaviour extracted from video-recorded conversations and trained a transformer-based neural sequence model to process the gaze data of the dialogues and compute its information density. We also compute the information density of the corresponding speech using a pre-trained language model. Our results show (1) that listeners’ gaze behaviour in dialogues roughly follows the ERC principle, as well as (2) a congruence between information density of speech and listeners’ gaze behaviour.
This paper investigates the challenges in building Swiss German speech translation systems, specifically focusing on the impact of dialect diversity and differences between Swiss German and Standard German. Swiss German is a spoken language with no formal writing system, it comprises many diverse dialects and is a low-resource language with only around 5 million speakers. The study is guided by two key research questions: how does the inclusion and exclusion of dialects during the training of speech translation models for Swiss German impact the performance on specific dialects, and how do the differences between Swiss German and Standard German impact the performance of the systems? We show that dialect diversity and linguistic differences pose significant challenges to Swiss German speech translation, which is in line with linguistic hypotheses derived from empirical investigations.
When training data are collected from human annotators, the design of the annotation instrument, the instructions given to annotators, the characteristics of the annotators, and their interactions can impact training data. This study demonstrates that design choices made when creating an annotation instrument also impact the models trained on the resulting annotations. We introduce the term annotation sensitivity to refer to the impact of annotation data collection methods on the annotations themselves and on downstream model performance and predictions. We collect annotations of hate speech and offensive language in five experimental conditions of an annotation instrument, randomly assigning annotators to conditions. We then fine-tune BERT models on each of the five resulting datasets and evaluate model performance on a holdout portion of each condition. We find considerable differences between the conditions for 1) the share of hate speech/offensive language annotations, 2) model performance, 3) model predictions, and 4) model learning curves. Our results emphasize the crucial role played by the annotation instrument which has received little attention in the machine learning literature. We call for additional research into how and why the instrument impacts the annotations to inform the development of best practices in instrument design.