当前的身份验证和可信系统依赖于经典和生物识别方法来识别或授权用户。这些方法包括音频语音识别，眼睛和手指签名。最近的工具利用深度学习和变压器来实现更好的结果。在本文中，我们使用Wav2Vec2.0和Hubert音频表示学习工具开发了阿拉伯语扬声器识别的深度学习构建模型。端到端Wav2Vec2.0范例通过随机掩蔽一组特征向量获取上下文化语音表示了解，然后应用变压器神经网络。我们使用了一个MLP分类器，可以区分不变的标记类。我们展示了几种实验结果，可以保护拟议模型的高精度。实验确保了某些扬声器的任意波信号分别可以分别在Wav2Vec2.0和Hubert的情况下以98％和97.1％的精度识别。

学龄前评估至关重要，因为它为教师和父母提供了有关儿童成长和成长的关键知识。冠状病毒大流行强调了在线评估学龄前儿童的必要性。这种在线测试需要各种技术，从Web应用程序开发到各种标准（例如语音识别）的各种人工智能模型。由于声学的波动和儿童和成人之间语音频率的差异，因此很难采用自动语音识别（ASR）系统，因为它们是在成年人的声音上预先训练的。此外，培训新模型需要大量数据。为了解决此问题，我们使用具有新的预训练目标的WAV2VEC 2.0模型为认知测试系统构建了ASR，称为随机频率音调（RFP），而我们的新数据集则在无意义的单词（MW）和New DataSet上进行了测试（MW）和快速自动命名（RAN）测试。由于这两个测试的特殊性，我们探索了许多模型，包括卷积神经网络（CNN）和WAV2VEC 2.0模型。我们的新方法在CommonVoice数据集的波斯部分上达到6.45的单词错误率（WER）。此外，我们的新方法在零和少数场景中产生积极的结果。

构建可用的无线电监控自动语音识别（ASR）系统是资源不足的语言的一项挑战性任务，但这在广播是公众沟通和讨论的主要媒介的社会中至关重要。联合国在乌干达的最初努力证明了如何理解被社交媒体排除在社交媒体中的农村人的看法在国家规划中很重要。但是，由于缺乏转录的语音数据集，这些努力正受到挑战。在本文中，Makerere人工智能研究实验室发布了155小时的Luganda Radio演讲语料库。据我们所知，这是撒哈拉以南非洲第一个公开可用的广播数据集。本文描述了语音语料库的开发，并使用开源语音识别工具包Coqui STT Toolkit提出了基线Luganda ASR绩效结果。

口吃是一种言语障碍，在此期间，语音流被非自愿停顿和声音重复打断。口吃识别是一个有趣的跨学科研究问题，涉及病理学，心理学，声学和信号处理，使检测很难且复杂。机器和深度学习的最新发展已经彻底彻底改变了语音领域，但是对口吃的识别受到了最小的关注。这项工作通过试图将研究人员从跨学科领域聚集在一起来填补空白。在本文中，我们回顾了全面的声学特征，基于统计和深度学习的口吃/不足分类方法。我们还提出了一些挑战和未来的指示。

甚至人类智能系统也无法提供100％的准确性来识别特定个人的演讲。Machine Intelligence试图通过各种语音提取和语音建模技术来模仿说话者识别问题。本文提出了一种独立于文本的扬声器识别系统，该系统采用了MEL频率曲线系数（MFCC）进行特征提取和K-Nearest邻居（KNN）进行分类。获得的最大交叉验证精度为60％。这将在随后的研究中得到改善。

口吃是一种多种言语障碍，会损害个人的沟通能力。口吃（PWS）的人经常使用语音疗法来应对自己的病情。改善具有这种非典型语音或跟踪语音疗法的人的语音识别系统将需要能够检测功能障碍的系统，同时能够检测到治疗中获得的语​​音技术。本文表明，用于在含有口吃的语音上结结巴巴的口吃的微调2VEC 2.0 [1]，结合多任务的学习，增强了通用Purepose Wav2VEC 2.0的有效性，以检测语音在语音中检测说话的功能;内部和跨语言。我们通过训练支持向量机分类器评估我们的FluencyBank的方法[2]和以德国治疗为中心的Kassel Fluency（KSOF）[3]数据集[3]数据集，该数据集使用六种不同结肠相关的事件类型中提取的功能：块：块： ，延长，声音重复，单词重复，插入和 - 特定于治疗 - 语音修改。使用来自微调模型的嵌入式嵌入会导致相对分类的性能增长到高达27％W.R.T. F1得分。

Self-supervised approaches for speech representation learning are challenged by three unique problems: (1) there are multiple sound units in each input utterance, (2) there is no lexicon of input sound units during the pre-training phase, and (3) sound units have variable lengths with no explicit segmentation. To deal with these three problems, we propose the Hidden-Unit BERT (HuBERT) approach for self-supervised speech representation learning, which utilizes an offline clustering step to provide aligned target labels for a BERT-like prediction loss. A key ingredient of our approach is applying the prediction loss over the masked regions only, which forces the model to learn a combined acoustic and language model over the continuous inputs. HuBERT relies primarily on the consistency of the unsupervised clustering step rather than the intrinsic quality of the assigned cluster labels. Starting with a simple k-means teacher of 100 clusters, and using two iterations of clustering, the HuBERT model either matches or improves upon the state-ofthe-art wav2vec 2.0 performance on the Librispeech (960h) and Libri-light (60,000h) benchmarks with 10min, 1h, 10h, 100h, and 960h fine-tuning subsets. Using a 1B parameter model, HuBERT shows up to 19% and 13% relative WER reduction on the more challenging dev-other and test-other evaluation subsets. 1

已经证明了深度学习技术在各种任务中有效，特别是在语音识别系统的发展中，即旨在以一系列写词中的音频句子转录音频句子的系统。尽管该地区进展，但语音识别仍然可以被认为是困难的，特别是对于缺乏可用数据的语言，例如巴西葡萄牙语（BP）。从这个意义上讲，这项工作介绍了仅使用打开可用的音频数据的公共自动语音识别（ASR）系统的开发，从Wav2Vec 2.0 XLSR-53模型的微调，在许多语言中，通过BP数据进行了多种。最终模型在7个不同的数据集中呈现12.4％的平均误差率（在应用语言模型时10.5％）。根据我们的知识，这是开放ASR系统中BP的最佳结果。

这项工作的目的是通过利用视频中的音频和视觉流的自然共同发生来研究语音重建（视频到音频）对语音重建（视频到音频）的影响。我们提出了Lipsound2，其包括编码器 - 解码器架构和位置感知注意机制，可直接将面部图像序列映射到熔化谱图，而无需任何人类注释。提出的Lipsound2模型首先在$ 2400H的$ 2400h多语言（例如英语和德语）视听数据（VoxceleB2）上进行预先培训。为了验证所提出的方法的概括性，我们将在与以前的方法相比，微调在域特定数据集（网格，TCD-Timit）上进行预先训练的模型，以实现对语音质量和可懂度的显着提高扬声器依赖和依赖的设置。除了英语外，我们还在CMLR数据集上进行中文语音重建，以验证对转移性的影响。最后，我们通过微调在预先训练的语音识别系统上产生生成的音频并在英语和中文基准数据集中实现最先进的性能来培训级联唇读（视频到文本）系统。

语音识别是一种技术，它将人类语音信号转换为文本或单词或以任何形式，可以通过计算机或其他机器容易地理解。有一些关于Bangla Digit识别系统的研究，其中大多数使用的小型数据集几乎没有变体，年龄，方言和其他变量。孟加拉国人民的录音，各种性别，年龄和方言，用于在本研究中创造一个大语音数据集。这里，已记录400个噪声和无噪音样本，用于创建数据集。 MEL频率谱系数（MFCC）已被用于从原始语音数据中提取有意义的功能。然后，为了检测Bangla数字，利用卷积神经网络（CNNS）。建议的技术在整个数据集中识别出“0-9”Bangla口语数字，精度为97.1％。还使用10倍的交叉透过来评估模型的效率，其精度为96.7％。

In this modern era of technology with e-commerce developing at a rapid pace, it is very important to understand customer requirements and details from a business conversation. It is very crucial for customer retention and satisfaction. Extracting key insights from these conversations is very important when it comes to developing their product or solving their issue. Understanding customer feedback, responses, and important details of the product are essential and it would be done using Named entity recognition (NER). For extracting the entities we would be converting the conversations to text using the optimal speech-to-text model. The model would be a two-stage network in which the conversation is converted to text. Then, suitable entities are extracted using robust techniques using a NER BERT transformer model. This will aid in the enrichment of customer experience when there is an issue which is faced by them. If a customer faces a problem he will call and register his complaint. The model will then extract the key features from this conversation which will be necessary to look into the problem. These features would include details like the order number, and the exact problem. All these would be extracted directly from the conversation and this would reduce the effort of going through the conversation again.

This paper describes a simple yet efficient repetition-based modular system for speeding up air-traffic controllers (ATCos) training. E.g., a human pilot is still required in EUROCONTROL's ESCAPE lite simulator (see https://www.eurocontrol.int/simulator/escape) during ATCo training. However, this need can be substituted by an automatic system that could act as a pilot. In this paper, we aim to develop and integrate a pseudo-pilot agent into the ATCo training pipeline by merging diverse artificial intelligence (AI) powered modules. The system understands the voice communications issued by the ATCo, and, in turn, it generates a spoken prompt that follows the pilot's phraseology to the initial communication. Our system mainly relies on open-source AI tools and air traffic control (ATC) databases, thus, proving its simplicity and ease of replicability. The overall pipeline is composed of the following: (1) a submodule that receives and pre-processes the input stream of raw audio, (2) an automatic speech recognition (ASR) system that transforms audio into a sequence of words; (3) a high-level ATC-related entity parser, which extracts relevant information from the communication, i.e., callsigns and commands, and finally, (4) a speech synthesizer submodule that generates responses based on the high-level ATC entities previously extracted. Overall, we show that this system could pave the way toward developing a real proof-of-concept pseudo-pilot system. Hence, speeding up the training of ATCos while drastically reducing its overall cost.

音频是人类交流最常用的方式之一，但与此同时，它很容易被欺骗人们滥用。随着AI的革命，几乎每个人都可以访问相关技术，从而使罪犯犯罪和伪造变得简单。在这项工作中，我们引入了一种深度学习方法，以开发一种分类器，该分类器将盲目地将输入音频分类为真实或模仿。提出的模型接受了从大型音频数据集提取的一组重要功能的培训，以获取分类器，该分类器已在不同音频的相同功能上进行了测试。为这项工作创建了两个数据集；所有英语数据集和混合数据集（阿拉伯语和英语）。这些数据集已通过GitHub提供，可在https://github.com/sass7/dataset上使用研究社区。为了进行比较，还通过人类检查对音频进行了分类，主题是母语人士。随之而来的结果很有趣，并且表现出强大的精度。

毒性言论，也被称为仇恨言论，被认为是今天批评在线社交媒体的重要问题之一。最近关于有毒语音检测的工作受到文本的模型，没有现有的毒性检测从口语中的出口检测。在本文中，我们提出了一种从口语中检测毒性的新口语处理任务。我们介绍了排毒，这是英语演讲的第一个公开的毒性注释数据集，来自各种公开可用的语音数据库，包括超过200万个话语。最后，我们还提供了对毒性注释的语音语料库的分析可以帮助促进E2E模型的发展，更好地捕获语音中的各种韵律线索，从而提高了口语的毒性分类。

本文介绍了与萨特布-Naija的基础努力，这是一种非原生（L2）尼日利亚语言语的新型语料库。我们描述了如何创建和策划的语料库以及令人口气分类和学习尼日利亚口音嵌入的初步实验。语料库的初始版本包括L2英语尼日利亚语言的900多个录音，例如Yoruba，Igbo，Edo，Efik-Ibibio和Igala。我们进一步演示了Wav2VEC的预先训练模型上的微调如何产生适合于相关语音任务的表示，例如重音分类。Sautidb-Naija已发表于Zenodo，以便在灵活的创造性的公共许可证下使用。

自我监督的语音表示，如Wav2Vec 2.0和Hubert正在自动语音识别（ASR）中进行革命性进展。但是，未经监督模型没有完全证明在ASR以外的任务中产生更好的性能。在这项工作中，我们探索了Wav2Vec 2.0和Hubert预先训练模型的部分微调和整个微调，适用于三个非ASR语音任务：语音情感识别，发言者验证和口语理解。我们还比较带有/没有ASR微调的预训练型号。通过简单的下游框架，最佳分数对IEMocap上的语音情感识别的加权精度达到79.58％，扬声器验证对voxcereB1的2.36％，意图分类的准确性为87.51％，Slotp的槽填充的75.32％f1，因此为这三个基准设置新的最先进，证明了微调Wave2VEC 2.0和Hubert模型可以更好地学习韵律，语音印刷和语义表示。

Through solving pretext tasks, self-supervised learning leverages unlabeled data to extract useful latent representations replacing traditional input features in the downstream task. In audio/speech signal processing, a wide range of features where engineered through decades of research efforts. As it turns out, learning to predict such features (a.k.a pseudo-labels) has proven to be a particularly relevant pretext task, leading to useful self-supervised representations which prove to be effective for downstream tasks. However, methods and common practices for combining such pretext tasks for better performance on the downstream task have not been explored and understood properly. In fact, the process relies almost exclusively on a computationally heavy experimental procedure, which becomes intractable with the increase of the number of pretext tasks. This paper introduces a method to select a group of pretext tasks among a set of candidates. The method we propose estimates calibrated weights for the partial losses corresponding to the considered pretext tasks during the self-supervised training process. The experiments conducted on automatic speech recognition, speaker and emotion recognition validate our approach, as the groups selected and weighted with our method perform better than classic baselines, thus facilitating the selection and combination of relevant pseudo-labels for self-supervised representation learning.

Automatic Speech Recognition (ASR) for air traffic control is generally trained by pooling Air Traffic Controller (ATCO) and pilot data into one set. This is motivated by the fact that pilot's voice communications are more scarce than ATCOs. Due to this data imbalance and other reasons (e.g., varying acoustic conditions), the speech from ATCOs is usually recognized more accurately than from pilots. Automatically identifying the speaker roles is a challenging task, especially in the case of the noisy voice recordings collected using Very High Frequency (VHF) receivers or due to the unavailability of the push-to-talk (PTT) signal, i.e., both audio channels are mixed. In this work, we propose to (1) automatically segment the ATCO and pilot data based on an intuitive approach exploiting ASR transcripts and (2) subsequently consider an automatic recognition of ATCOs' and pilots' voice as two separate tasks. Our work is performed on VHF audio data with high noise levels, i.e., signal-to-noise (SNR) ratios below 15 dB, as this data is recognized to be helpful for various speech-based machine-learning tasks. Specifically, for the speaker role identification task, the module is represented by a simple yet efficient knowledge-based system exploiting a grammar defined by the International Civil Aviation Organization (ICAO). The system accepts text as the input, either manually verified annotations or automatically generated transcripts. The developed approach provides an average accuracy in speaker role identification of about 83%. Finally, we show that training an acoustic model for ASR tasks separately (i.e., separate models for ATCOs and pilots) or using a multitask approach is well suited for the noisy data and outperforms the traditional ASR system where all data is pooled together.

最近的语音情绪识别分析与使用MFCCS频谱图特征和实现诸如卷积神经网络（CNNS）的神经网络方法的实施进行了相当大的进展。胶囊网络（CAPSNET）对CNN的替代品感谢其具有较大容量的分层表示。为了解决这些问题，本研究介绍了独立于文本和独立的讲话者独立的SER新颖体系结构，其中基于结构特征提出了双通道长短短期内存压缩帽（DC-LSTM Compsnet）算法Capsnet。我们所提出的新型分类器可以确保语音情感识别中模型和足够的压缩方法的能效，这不会通过彩铃的原始结构提供。此外，网格搜索方法用于获得最佳解决方案。结果目睹了培训和测试运行时间的性能和减少。用于评估我们的算法的语音数据集是：阿拉伯语Emirati-Egrented语料库，模拟和实际压力语料库下的英语演讲，情感语音和歌曲语料库的英语Ryerson Audio-Visual数据库，以及人群源性情绪多模式演员数据集。这项工作揭示了与其他已知方法相比的最佳特征提取方法是MFCCS Delta-Delta。使用四个数据集和MFCCS Delta-Delta，DC-LSTM CompsNet超越了所有最先进的系统，古典分类器，CNN和原始帽。我们的结果表明，基于Capsnet的拟议工作产生了89.3％的平均情绪识别准确性，其结果表明，拟议的工作产生了89.3％的89.3％。 CNN，支持向量机，多层Perceptron，K-最近邻居，径向基函数和幼稚贝叶斯。

While the Turkish language is listed among low-resource languages, literature on Turkish automatic speech recognition (ASR) is relatively old. In this report, we present our findings on Turkish ASR with speech representation learning using HUBERT. We investigate pre-training HUBERT for Turkish with large-scale data curated from online resources. We pre-train our model using 6,500 hours of speech data from YouTube. The results show that the models are not ready for commercial use since they are not robust against disturbances that typically occur in real-world settings such as variations in accents, slang, background noise and interference. We analyze typical errors and the limitations of the models for use in commercial settings.