Best AI tools for< Speech Researcher >

Moshi AI by Kyutai is an advanced native speech AI model that enables natural, expressive conversations. It can be installed locally and run offline, making it suitable for integration into smart home appliances and other local applications. The model, named Helium, has 7 billion parameters and is trained on text and audio codecs. Moshi AI supports native speech input and output, allowing for smooth communication with the AI. The application is community-supported, with plans for continuous improvement and adaptation.

TranscribeAudio is an AI-powered transcription tool that enables users to convert audio files into text quickly and accurately. It offers features like speaker identification, insights generation, and secure file handling. The tool is user-friendly, with a simple editor for reviewing and refining transcripts. TranscribeAudio provides a subscription-based service with a generous free tier and simple pricing. It is constantly updated with new features to enhance user experience.

Deepgram is a speech recognition and transcription service that uses artificial intelligence to convert audio into text. It is designed to be accurate, fast, and easy to use. Deepgram offers a variety of features, including: - Automatic speech recognition - Speaker diarization - Language identification - Custom acoustic models - Real-time transcription - Batch transcription - Webhooks - Integrations with popular platforms such as Zoom, Google Meet, and Microsoft Teams

ChatTTS is an open-source text-to-speech model designed for dialogue scenarios, supporting both English and Chinese speech generation. Trained on approximately 100,000 hours of Chinese and English data, it delivers speech quality comparable to human dialogue. The tool is particularly suitable for tasks involving large language model assistants and creating dialogue-based audio and video introductions. It provides developers with a powerful and easy-to-use tool based on open-source natural language processing and speech synthesis technologies.

Speak4Me is a text-to-speech application that converts any text file, including PDFs and websites, into audible content. It enables users to listen to their documents or school materials anytime, anywhere. With features like scanning physical or digital text, reading web pages aloud, and a new ChatWithMe function, Speak4Me aims to enhance reading experiences and improve focus for individuals with reading issues. The application is trusted by over 15,000 people on the App Store and offers a free version for schools, making education more accessible for everyone.

Voicetapp is a powerful cloud-based artificial intelligence software that helps you automatically convert audio to text with up to 100% accuracy. It supports over 170 languages and dialects, allowing you to quickly and accurately transcribe speech from audio and video files. Voicetapp also offers features such as speaker identification, live transcription, and multiple input formats, making it a versatile tool for various use cases.

OneAudio is an AI-powered tool that allows users to summarize, transcribe, and convert audio files into notes. With features like recording, summarization, and language selection, OneAudio helps users organize and transform their ideas efficiently. The tool leverages OpenAI GPT-4 and GPT-4o models to provide accurate transcriptions and summaries. Users can choose from different pricing plans based on their needs, from a free tier to a premium plan with unlimited features. OneAudio is designed to streamline the process of converting audio content into written notes, making it ideal for students, professionals, and anyone looking to enhance their productivity.

SpeechFlow is a powerful speech-to-text API that transcribes audio and video files into text with high accuracy. It supports 14 languages and offers features such as punctuation, easy deployment, scalability, and fast processing. SpeechFlow is ideal for businesses and individuals who need accurate and timely transcription services.

Read It is an AI-powered tool that allows users to convert newsletters and articles into podcasts effortlessly. By utilizing cutting-edge AI text-to-speech technology, users can listen to their favorite written content on the go. The tool provides users with a personal podcast feed URL upon sign-up, enabling them to add articles through email forwarding or using a bookmarklet. With a user-friendly interface and pay-as-you-go model, Read It offers a seamless experience for turning text-based content into audio podcasts.

Alice is a fast, accurate AI transcription and recorder application that prioritizes privacy and cost-effectiveness. It allows users to securely record audio and video, transcribe in multiple languages and accents with high accuracy, and offers real-time text streaming. Alice integrates with various tools, supports webhooks, and is trusted by journalists for its reliability and security features. The application is designed to be user-friendly, efficient, and suitable for a wide range of tasks, making it a valuable tool for journalists, freelancers, and anyone in need of transcription services.

WavoAI is an AI-powered transcription and summarization tool that helps users transcribe audio recordings quickly and accurately. It offers features such as speaker identification, annotations, and interactive AI insights, making it a valuable tool for a wide range of professionals, including academics, filmmakers, podcasters, and journalists.

Auri is an AI assistant that offers a range of writing, communication, and productivity tools. It includes an AI-powered keyboard with features like grammar checking, translation, and paraphrasing, as well as an AI chat, voice recorder and transcriber, and smart notes. Auri is available on iPhone, iPad, Mac, and Apple Watch.

Rev is a leading transcription service provider offering human and AI transcription solutions with high accuracy rates. The platform enables users to transcribe audio and video content efficiently, generate captions and subtitles in multiple languages, and access speech-to-text solutions for various industries such as news organizations, market research, video distribution, and legal services. Rev's AI-powered tools enhance content accessibility, global reach, and audience engagement, making it a versatile and reliable platform for transcription needs.

Sonix is a powerful and easy-to-use online audio and video transcription service. It uses advanced artificial intelligence (AI) to convert speech to text quickly and accurately. Sonix supports over 38 languages and offers a variety of features, including automatic transcription, translation, subtitling, and summarization. It is a valuable tool for journalists, researchers, students, businesses, and anyone who needs to transcribe audio or video content.

VideoToWords.ai is an AI-powered transcription tool that converts audio and video files into accurate written text. It utilizes advanced machine learning algorithms to transcribe files quickly and efficiently, catering to a wide range of users such as journalists, students, researchers, podcast hosts, filmmakers, content creators, marketers, and professionals from various industries. The platform supports multiple languages, offers convenient text editing and export options, and ensures data security and privacy for users.

MacWhisper is a native macOS application that utilizes OpenAI's Whisper technology for transcribing audio files into text. It offers a user-friendly interface for recording, transcribing, and editing audio, making it suitable for various use cases such as transcribing meetings, lectures, interviews, and podcasts. The application is designed to protect user privacy by performing all transcriptions locally on the device, ensuring that no data leaves the user's machine.

TTS Generator AI is a free online text-to-speech tool that leverages cutting-edge AI technology to convert written text into high-quality, natural-sounding audio. This tool is invaluable for a variety of users, including students who need auditory learning materials, researchers who want to listen to long documents, and professionals seeking to make their written content more accessible. One of the standout features of TTS Tool is its ability to support a range of text formats, from simple text files to complex PDFs, making it incredibly versatile.

TurboScribe.ai is an AI transcription tool that converts audio and video files into text with high accuracy and efficiency. It utilizes advanced AI algorithms to transcribe content quickly, making it ideal for professionals, students, and anyone needing transcription services. The tool ensures security by verifying user identity and connection before processing the transcription. TurboScribe.ai is powered by Cloudflare for enhanced performance and security.

Ragobble is an audio to LLM data tool that allows you to easily convert audio files into text data that can be used to train large language models (LLMs). With Ragobble, you can quickly and easily create high-quality training data for your LLM projects.

Meta AI is a research lab dedicated to advancing the field of artificial intelligence. Our mission is to build foundational AI technologies that will solve some of the world's biggest challenges, such as climate change, disease, and poverty.

Speech Trident is a repository focusing on speech/audio large language models, covering representation learning, neural codec, and language models. It explores speech representation models, speech neural codec models, and speech large language models. The repository includes contributions from various researchers and provides a comprehensive list of speech/audio language models, representation models, and codec models.

Lhotse is a Python library designed to make speech and audio data preparation flexible and accessible. It aims to attract a wider community to speech processing tasks by providing a Python-centric design and an expressive command-line interface. Lhotse offers standard data preparation recipes, PyTorch Dataset classes for speech tasks, and efficient data preparation for model training with audio cuts. It supports data augmentation, feature extraction, and feature-space cut mixing. The tool extends Kaldi's data preparation recipes with seamless PyTorch integration, human-readable text manifests, and convenient Python classes.

ChatTTS is a generative speech model optimized for dialogue scenarios, providing natural and expressive speech synthesis with fine-grained control over prosodic features. It supports multiple speakers and surpasses most open-source TTS models in terms of prosody. The model is trained with 100,000+ hours of Chinese and English audio data, and the open-source version on HuggingFace is a 40,000-hour pre-trained model without SFT. The roadmap includes open-sourcing additional features like VQ encoder, multi-emotion control, and streaming audio generation. The tool is intended for academic and research use only, with precautions taken to limit potential misuse.

HebTTS is a language modeling approach to diacritic-free Hebrew text-to-speech (TTS) system. It addresses the challenge of accurately mapping text to speech in Hebrew by proposing a language model that operates on discrete speech representations and is conditioned on a word-piece tokenizer. The system is optimized using weakly supervised recordings and outperforms diacritic-based Hebrew TTS systems in terms of content preservation and naturalness of generated speech.

WeST is a Speech Recognition/Transcript tool developed in 300 lines of code, inspired by SLAM-ASR and LLaMA 3.1. The model includes a Language Model (LLM), a Speech Encoder, and a trainable Projector. It requires training data in jsonl format with 'wav' and 'txt' entries. WeST can be used for training and decoding speech recognition models.

MockingBird is a toolbox designed for Mandarin speech synthesis using PyTorch. It supports multiple datasets such as aidatatang_200zh, magicdata, aishell3, and data_aishell. The toolbox can run on Windows, Linux, and M1 MacOS, providing easy and effective speech synthesis with pretrained encoder/vocoder models. It is webserver ready for remote calling. Users can train their own models or use existing ones for the encoder, synthesizer, and vocoder. The toolbox offers a demo video and detailed setup instructions for installation and model training.

LLaSA_training is a repository focused on training models for speech synthesis using a large amount of open-source speech data. The repository provides instructions for finetuning models and offers pre-trained models for multilingual speech synthesis. It includes tools for training, data downloading, and data processing using specialized tokenizers for text and speech sequences. The repository also supports direct usage on Hugging Face platform with specific codecs and collections.

Awesome-ChatTTS is an official recommended guide for ChatTTS beginners, compiling common questions and related resources. It provides a comprehensive overview of the project, including official introduction, quick experience options, popular branches, parameter explanations, voice seed details, installation guides, FAQs, and error troubleshooting. The repository also includes video tutorials, discussion community links, and project trends analysis. Users can explore various branches for different functionalities and enhancements related to ChatTTS.

KanTV is an open-source project that focuses on studying and practicing state-of-the-art AI technology in real applications and scenarios, such as online TV playback, transcription, translation, and video/audio recording. It is derived from the original ijkplayer project and includes many enhancements and new features, including: * Watching online TV and local media using a customized FFmpeg 6.1. * Recording online TV to automatically generate videos. * Studying ASR (Automatic Speech Recognition) using whisper.cpp. * Studying LLM (Large Language Model) using llama.cpp. * Studying SD (Text to Image by Stable Diffusion) using stablediffusion.cpp. * Generating real-time English subtitles for English online TV using whisper.cpp. * Running/experiencing LLM on Xiaomi 14 using llama.cpp. * Setting up a customized playlist and using the software to watch the content for R&D activity. * Refactoring the UI to be closer to a real commercial Android application (currently only supports English). Some goals of this project are: * To provide a well-maintained "workbench" for ASR researchers interested in practicing state-of-the-art AI technology in real scenarios on mobile devices (currently focusing on Android). * To provide a well-maintained "workbench" for LLM researchers interested in practicing state-of-the-art AI technology in real scenarios on mobile devices (currently focusing on Android). * To create an Android "turn-key project" for AI experts/researchers (who may not be familiar with regular Android software development) to focus on device-side AI R&D activity, where part of the AI R&D activity (algorithm improvement, model training, model generation, algorithm validation, model validation, performance benchmark, etc.) can be done very easily using Android Studio IDE and a powerful Android phone.

SenseVoice is a speech foundation model focusing on high-accuracy multilingual speech recognition, speech emotion recognition, and audio event detection. Trained with over 400,000 hours of data, it supports more than 50 languages and excels in emotion recognition and sound event detection. The model offers efficient inference with low latency and convenient finetuning scripts. It can be deployed for service with support for multiple client-side languages. SenseVoice-Small model is open-sourced and provides capabilities for Mandarin, Cantonese, English, Japanese, and Korean. The tool also includes features for natural speech generation and fundamental speech recognition tasks.

The 'nlp-llms-resources' repository is a comprehensive resource list for Natural Language Processing (NLP) and Large Language Models (LLMs). It covers a wide range of topics including traditional NLP datasets, data acquisition, libraries for NLP, neural networks, sentiment analysis, optical character recognition, information extraction, semantics, topic modeling, multilingual NLP, domain-specific LLMs, vector databases, ethics, costing, books, courses, surveys, aggregators, newsletters, papers, conferences, and societies. The repository provides valuable information and resources for individuals interested in NLP and LLMs.

CosyVoice is a tool designed for speech synthesis, offering pretrained models for zero-shot, sft, instruct inference. It provides a web demo for easy usage and supports advanced users with train and inference scripts. The tool can be deployed using grpc for service deployment. Users can download pretrained models and resources for immediate use or train their own models from scratch. CosyVoice is suitable for researchers, developers, linguists, AI engineers, and speech technology enthusiasts.

AudioLLMs is a curated collection of research papers focusing on developing, implementing, and evaluating language models for audio data. The repository aims to provide researchers and practitioners with a comprehensive resource to explore the latest advancements in AudioLLMs. It includes models for speech interaction, speech recognition, speech translation, audio generation, and more. Additionally, it covers methodologies like multitask audioLLMs and segment-level Q-Former, as well as evaluation benchmarks like AudioBench and AIR-Bench. Adversarial attacks such as VoiceJailbreak are also discussed.

Xorbits Inference (Xinference) is a powerful and versatile library designed to serve language, speech recognition, and multimodal models. With Xorbits Inference, you can effortlessly deploy and serve your or state-of-the-art built-in models using just a single command. Whether you are a researcher, developer, or data scientist, Xorbits Inference empowers you to unleash the full potential of cutting-edge AI models.

NeMo Framework is a generative AI framework built for researchers and pytorch developers working on large language models (LLMs), multimodal models (MM), automatic speech recognition (ASR), and text-to-speech synthesis (TTS). The primary objective of NeMo is to provide a scalable framework for researchers and developers from industry and academia to more easily implement and design new generative AI models by being able to leverage existing code and pretrained models.

The 'llms-tools' repository is a comprehensive collection of AI tools, open-source projects, and research related to Large Language Models (LLMs) and Chatbots. It covers a wide range of topics such as AI in various domains, open-source models, chats & assistants, visual language models, evaluation tools, libraries, devices, income models, text-to-image, computer vision, audio & speech, code & math, games, robotics, typography, bio & med, military, climate, finance, and presentation. The repository provides valuable resources for researchers, developers, and enthusiasts interested in exploring the capabilities of LLMs and related technologies.

The 'Conference-Acceptance-Rate' repository provides acceptance rates for top-tier AI-related conferences in the fields of Natural Language Processing, Computational Linguistics, Computer Vision, Pattern Recognition, Machine Learning, Learning Theory, Artificial Intelligence, Data Mining, Information Retrieval, Speech Processing, and Signal Processing. The data includes acceptance rates for long papers and short papers over several years for each conference, allowing researchers to track trends and make informed decisions about where to submit their work.

BrowserAI is a production-ready tool that allows users to run AI models directly in the browser, offering simplicity, speed, privacy, and open-source capabilities. It provides WebGPU acceleration for fast inference, zero server costs, offline capability, and developer-friendly features. Perfect for web developers, companies seeking privacy-conscious AI solutions, researchers experimenting with browser-based AI, and hobbyists exploring AI without infrastructure overhead. The tool supports various AI tasks like text generation, speech recognition, and text-to-speech, with pre-configured popular models ready to use. It offers a simple SDK with multiple engine support and seamless switching between MLC and Transformers engines.

SLAM-LLM is a deep learning toolkit designed for researchers and developers to train custom multimodal large language models (MLLM) focusing on speech, language, audio, and music processing. It provides detailed recipes for training and high-performance checkpoints for inference. The toolkit supports tasks such as automatic speech recognition (ASR), text-to-speech (TTS), visual speech recognition (VSR), automated audio captioning (AAC), spatial audio understanding, and music caption (MC). SLAM-LLM features easy extension to new models and tasks, mixed precision training for faster training with less GPU memory, multi-GPU training with data and model parallelism, and flexible configuration based on Hydra and dataclass.

Awesome Khmer Language is a comprehensive collection of resources for the Khmer language, including tools, datasets, research papers, projects/models, blogs/slides, and miscellaneous items. It covers a wide range of topics related to Khmer language processing, such as character normalization, word segmentation, part-of-speech tagging, optical character recognition, text-to-speech, and more. The repository aims to support the development of natural language processing applications for the Khmer language by providing a diverse set of resources and tools for researchers and developers.

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