Best AI tools for< Speech Engineer >

Deepgram is a powerful API platform that provides developers with tools for building speech-to-text, text-to-speech, and intelligence applications. With Deepgram, developers can easily add speech recognition, text-to-speech, and other AI-powered features to their applications.

Vocapia Research develops leading-edge, multilingual speech processing technologies exploiting AI methods such as machine learning. These technologies enable large vocabulary continuous speech recognition, automatic audio segmentation, language identification, speaker diarization and audio-text synchronization. Vocapia's VoxSigma™ speech-to-text software suite delivers state-of-the-art performance in many languages for a variety of audio data types, including broadcast data, parliamentary hearings and conversational data.

SoundHound is a leading innovator of conversational intelligence and voice AI technologies. Our independent voice AI platform is built for more natural conversation, enabling businesses to create customized and scalable voice AI solutions for their specific industries and use cases. With SoundHound, you can build voice assistants, enhance smart devices, improve customer experiences, and drive business value.

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.

Speech Studio is a cloud-based speech-to-text and text-to-speech platform that enables developers to add speech capabilities to their applications. With Speech Studio, developers can easily transcribe audio and video files, generate synthetic speech, and build custom speech models. Speech Studio is a powerful tool that can be used to improve the accessibility, efficiency, and user experience of any application.

Interesting Engineering is a website that covers the latest news and developments in technology, science, innovation, and engineering. The website features articles, videos, and podcasts on a wide range of topics, including artificial intelligence, robotics, space exploration, and renewable energy. Interesting Engineering also offers a variety of educational resources, such as courses, workshops, and webinars.

LMNT is an ultrafast and lifelike AI speech application that offers a developer API for creating conversational apps, agents, and games. It provides lifelike voices with studio-quality voice clones, engineered by an ex-Google team for reliability under pressure. Users can create engaging product marketing videos, build lightning-fast conversational experiences, and simplify content creation at scale. The application features a user-friendly interface, versatile voice cloning options, and downloadable content for easy integration into projects. With Python and Node SDKs, low latency streaming, and robust documentation, LMNT empowers developers to enhance their applications with high-quality voice synthesis.

Respeecher is a voice cloning software that allows users to create synthetic voices that are indistinguishable from the original speaker. The software is used by content creators in a variety of industries, including film, television, gaming, advertising, and audiobooks. Respeecher's technology is based on artificial intelligence and machine learning, and it can replicate the voice of any person with just a few minutes of audio recording. The software is easy to use and can be accessed through a web interface. Respeecher offers a variety of features, including the ability to change the pitch, speed, and volume of the synthetic voice, as well as the ability to add effects such as reverb and delay. The software also includes a library of pre-recorded voices that can be used for a variety of purposes.

AssemblyAI is an industry-leading Speech AI tool that offers powerful SpeechAI models for accurate transcription and understanding of speech. It provides breakthrough speech-to-text models, real-time captioning, and advanced speech understanding capabilities. AssemblyAI is designed to help developers build world-class products with unmatched accuracy and transformative audio intelligence.

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.

Voam is a productive AI platform that helps you to automate your tasks and improve your productivity. With Voam, you can create custom AI models to automate any task, from simple data entry to complex decision-making. Voam is easy to use and requires no coding experience. You can create an AI model in minutes and start automating your tasks right away.

Auphonic is an AI-powered audio post-production web tool designed to help users achieve professional-quality audio results effortlessly. It offers a range of features such as Intelligent Leveler, Noise & Reverb Reduction, Filtering & AutoEQ, Cut Filler Words and Silence, Multitrack Algorithms, Loudness Specifications, Speech2Text & Automatic Shownotes, Video Support, Metadata & Chapters, and more. Auphonic is widely used by podcasters, educators, content creators, and audiobook producers to enhance their audio content and streamline their workflows. With its intuitive interface and advanced algorithms, Auphonic simplifies the audio editing process and ensures consistent audio quality across different platforms.

Picovoice is an on-device Voice AI and local LLM platform designed for enterprises. It offers a range of voice AI and LLM solutions, including speech-to-text, noise suppression, speaker recognition, speech-to-index, wake word detection, and more. Picovoice empowers developers to build virtual assistants and AI-powered products with compliance, reliability, and scalability in mind. The platform allows enterprises to process data locally without relying on third-party remote servers, ensuring data privacy and security. With a focus on cutting-edge AI technology, Picovoice enables users to stay ahead of the curve and adapt quickly to changing customer needs.

AssemblyAI is a leading AI tool that provides industry-leading Speech AI models for accurate speech-to-text transcription and understanding. The platform offers powerful SpeechAI models, including the Universal-1, for transforming speech into meaning. With features like speech-to-text transcription, streaming speech-to-text, and speech understanding, AssemblyAI empowers users to extract valuable insights from audio data. The tool is trusted by developers for its accuracy, reliability, and comprehensive documentation, making it a go-to choice for building world-class voice data products.

AppTek is a global leader in artificial intelligence (AI) and machine learning (ML) technologies for automatic speech recognition (ASR), neural machine translation (NMT), natural language processing/understanding (NLP/U) and text-to-speech (TTS) technologies. The AppTek platform delivers industry-leading solutions for organizations across a breadth of global markets such as media and entertainment, call centers, government, enterprise business, and more. Built by scientists and research engineers who are recognized among the best in the world, AppTek’s solutions cover a wide array of languages/ dialects, channels, domains and demographics.

Respeecher is an AI tool that combines technology and magic to deliver authentic voices across various industries. It uses cutting-edge public models and proprietary technology to provide high-quality voice solutions. The team of dedicated sound professionals at Respeecher ensures ethical use of synthetic media, making it a trusted choice for voice cloning and voice conversion services.

Globose Technology Solutions Pvt Ltd (GTS) is an AI data collection company that provides various datasets such as image datasets, video datasets, text datasets, speech datasets, etc., to train machine learning models. They offer premium data collection services with a human touch, aiming to refine AI vision and propel AI forward. With over 25+ years of experience, they specialize in data management, annotation, and effective data collection techniques for AI/ML. The company focuses on unlocking high-quality data, understanding AI's transformative impact, and ensuring data accuracy as the backbone of reliable AI.

Kingshiper is a versatile multimedia tool offering a wide range of audio, photo, and video conversion and editing features. It provides tools for screen recording, video compression, screen mirroring, audio editing, vocal removal, and more. With support for over 1000+ formats, Kingshiper aims to simplify multimedia processing tasks for users. Additionally, it offers utilities for office tasks, system tools, data solutions, and image processing, catering to various user needs. The software is designed to enhance productivity and creativity by providing efficient and user-friendly tools for multimedia and office-related tasks.

Critiqs.ai is a platform offering reviews, tutorials, and a comprehensive list of over 5000 AI tools. These tools cover various categories such as image editing, audio generation, productivity enhancement, business solutions, text generation, coding assistance, and more. AI tools are software systems powered by artificial intelligence that automate tasks requiring human intelligence, from chatbots for customer service to predictive analytics for supply chain management. Critiqs.ai caters to tech enthusiasts, developers, and businesses seeking cutting-edge AI solutions to streamline operations, enhance skills, and explore the benefits of AI technology.

Marvin is a lightweight toolkit for building natural language interfaces that are reliable, scalable, and easy to trust. It provides a variety of AI functions for text, images, audio, and video, as well as interactive tools and utilities. Marvin is designed to be easy to use and integrate, and it can be used to build a wide range of applications, from simple chatbots to complex AI-powered systems.

OpenedAI Speech is a free, private text-to-speech server compatible with the OpenAI audio/speech API. It offers custom voice cloning and supports various models like tts-1 and tts-1-hd. Users can map their own piper voices and create custom cloned voices. The server provides multilingual support with XTTS voices and allows fixing incorrect sounds with regex. Recent changes include bug fixes, improved error handling, and updates for multilingual support. Installation can be done via Docker or manual setup, with usage instructions provided. Custom voices can be created using Piper or Coqui XTTS v2, with guidelines for preparing audio files. The tool is suitable for tasks like generating speech from text, creating custom voices, and multilingual text-to-speech applications.

Speech-AI-Forge is a project developed around TTS generation models, implementing an API Server and a WebUI based on Gradio. The project offers various ways to experience and deploy Speech-AI-Forge, including online experience on HuggingFace Spaces, one-click launch on Colab, container deployment with Docker, and local deployment. The WebUI features include TTS model functionality, speaker switch for changing voices, style control, long text support with automatic text segmentation, refiner for ChatTTS native text refinement, various tools for voice control and enhancement, support for multiple TTS models, SSML synthesis control, podcast creation tools, voice creation, voice testing, ASR tools, and post-processing tools. The API Server can be launched separately for higher API throughput. The project roadmap includes support for various TTS models, ASR models, voice clone models, and enhancer models. Model downloads can be manually initiated using provided scripts. The project aims to provide inference services and may include training-related functionalities in the future.

ChatTTS-Forge is a powerful text-to-speech generation tool that supports generating rich audio long texts using a SSML-like syntax and provides comprehensive API services, suitable for various scenarios. It offers features such as batch generation, support for generating super long texts, style prompt injection, full API services, user-friendly debugging GUI, OpenAI-style API, Google-style API, support for SSML-like syntax, speaker management, style management, independent refine API, text normalization optimized for ChatTTS, and automatic detection and processing of markdown format text. The tool can be experienced and deployed online through HuggingFace Spaces, launched with one click on Colab, deployed using containers, or locally deployed after cloning the project, preparing models, and installing necessary dependencies.

RealtimeSTT is an easy-to-use, low-latency speech-to-text library for realtime applications. It listens to the microphone and transcribes voice into text, making it ideal for voice assistants and applications requiring fast and precise speech-to-text conversion. The library utilizes Voice Activity Detection, Realtime Transcription, and Wake Word Activation features. It supports GPU-accelerated transcription using PyTorch with CUDA support. RealtimeSTT offers various customization options for different parameters to enhance user experience and performance. The library is designed to provide a seamless experience for developers integrating speech-to-text functionality into their applications.

FireRedTTS is a foundation text-to-speech framework designed for industry-level generative speech applications. It offers a rich-punctuation model with expanded punctuation coverage and enhanced audio production consistency. The tool provides pre-trained checkpoints, inference code, and an interactive demo space. Users can clone the repository, create a conda environment, download required model files, and utilize the tool for synthesizing speech in various languages. FireRedTTS aims to enhance stability and provide controllable human-like speech generation capabilities.

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.

VSP-LLM (Visual Speech Processing incorporated with LLMs) is a novel framework that maximizes context modeling ability by leveraging the power of LLMs. It performs multi-tasks of visual speech recognition and translation, where given instructions control the task type. The input video is mapped to the input latent space of a LLM using a self-supervised visual speech model. To address redundant information in input frames, a deduplication method is employed using visual speech units. VSP-LLM utilizes Low Rank Adaptors (LoRA) for computationally efficient training.

Generative Fusion Decoding (GFD) is a novel shallow fusion framework that integrates Large Language Models (LLMs) into multi-modal text recognition systems such as automatic speech recognition (ASR) and optical character recognition (OCR). GFD operates across mismatched token spaces of different models by mapping text token space to byte token space, enabling seamless fusion during the decoding process. It simplifies the complexity of aligning different model sample spaces, allows LLMs to correct errors in tandem with the recognition model, increases robustness in long-form speech recognition, and enables fusing recognition models deficient in Chinese text recognition with LLMs extensively trained on Chinese. GFD significantly improves performance in ASR and OCR tasks, offering a unified solution for leveraging existing pre-trained models through step-by-step fusion.

Deepgram JavaScript SDK. Power your apps with world-class speech and Language AI models.

Call GPT is a voice application that utilizes Deepgram for Speech to Text, elevenlabs for Text to Speech, and OpenAI for GPT prompt completion. It allows users to chat with ChatGPT on the phone, providing better transcription, understanding, and speaking capabilities than traditional IVR systems. The app returns responses with low latency, allows user interruptions, maintains chat history, and enables GPT to call external tools. It coordinates data flow between Deepgram, OpenAI, ElevenLabs, and Twilio Media Streams, enhancing voice interactions.

**RVC_CLI: Retrieval-based Voice Conversion Command Line Interface** This command-line interface (CLI) provides a comprehensive set of tools for voice conversion, enabling you to modify the pitch, timbre, and other characteristics of audio recordings. It leverages advanced machine learning models to achieve realistic and high-quality voice conversions. **Key Features:** * **Inference:** Convert the pitch and timbre of audio in real-time or process audio files in batch mode. * **TTS Inference:** Synthesize speech from text using a variety of voices and apply voice conversion techniques. * **Training:** Train custom voice conversion models to meet specific requirements. * **Model Management:** Extract, blend, and analyze models to fine-tune and optimize performance. * **Audio Analysis:** Inspect audio files to gain insights into their characteristics. * **API:** Integrate the CLI's functionality into your own applications or workflows. **Applications:** The RVC_CLI finds applications in various domains, including: * **Music Production:** Create unique vocal effects, harmonies, and backing vocals. * **Voiceovers:** Generate voiceovers with different accents, emotions, and styles. * **Audio Editing:** Enhance or modify audio recordings for podcasts, audiobooks, and other content. * **Research and Development:** Explore and advance the field of voice conversion technology. **For Jobs:** * Audio Engineer * Music Producer * Voiceover Artist * Audio Editor * Machine Learning Engineer **AI Keywords:** * Voice Conversion * Pitch Shifting * Timbre Modification * Machine Learning * Audio Processing **For Tasks:** * Convert Pitch * Change Timbre * Synthesize Speech * Train Model * Analyze Audio

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.

MARS5 is a novel English speech model (TTS) developed by CAMB.AI, featuring a two-stage AR-NAR pipeline with a unique NAR component. The model can generate speech for various scenarios like sports commentary and anime with just 5 seconds of audio and a text snippet. It allows steering prosody using punctuation and capitalization in the transcript. Speaker identity is specified using an audio reference file, enabling 'deep clone' for improved quality. The model can be used via torch.hub or HuggingFace, supporting both shallow and deep cloning for inference. Checkpoints are provided for AR and NAR models, with hardware requirements of 750M+450M params on GPU. Contributions to improve model stability, performance, and reference audio selection are welcome.

This repository is a curated list of awesome large AI models in audio signal processing, focusing on the application of large language models to audio tasks. It includes survey papers, popular large audio models, automatic speech recognition, neural speech synthesis, speech translation, other speech applications, large audio models in music, and audio datasets. The repository aims to provide a comprehensive overview of recent advancements and challenges in applying large language models to audio signal processing, showcasing the efficacy of transformer-based architectures in various audio tasks.

This repository implements a speech-to-speech cascaded pipeline with consecutive parts including Voice Activity Detection (VAD), Speech to Text (STT), Language Model (LM), and Text to Speech (TTS). It aims to provide a fully open and modular approach by leveraging models available on the Transformers library via the Hugging Face hub. The code is designed for easy modification, with each component implemented as a class. Users can run the pipeline either on a server/client approach or locally, with detailed setup and usage instructions provided in the readme.

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.

SLAM-LLM is a deep learning toolkit for training 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 various 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). Users can easily extend to new models and tasks, utilize mixed precision training for faster training with less GPU memory, and perform multi-GPU training with data and model parallelism. Configuration is flexible based on Hydra and dataclass, allowing different configuration methods.

IntelliNode is a javascript module that integrates cutting-edge AI models like ChatGPT, LLaMA, WaveNet, Gemini, and Stable diffusion into projects. It offers functions for generating text, speech, and images, as well as semantic search, multi-model evaluation, and chatbot capabilities. The module provides a wrapper layer for low-level model access, a controller layer for unified input handling, and a function layer for abstract functionality tailored to various use cases.

The Cheat-Sheet Collection for DevOps, Engineers, IT professionals, and more is a curated list of cheat sheets for various tools and technologies commonly used in the software development and IT industry. It includes cheat sheets for Nginx, Docker, Ansible, Python, Go (Golang), Git, Regular Expressions (Regex), PowerShell, VIM, Jenkins, CI/CD, Kubernetes, Linux, Redis, Slack, Puppet, Google Cloud Developer, AI, Neural Networks, Machine Learning, Deep Learning & Data Science, PostgreSQL, Ajax, AWS, Infrastructure as Code (IaC), System Design, and Cyber Security.

Kuebiko is a Twitch Chat Bot that reads twitch chat and generates text-to-speech responses using Google Cloud API and OpenAI's GPT-3 text completion model. It allows users to set up their own VTuber AI similar to 'Neuro-Sama'. The project is built with Python and requires setting up various API keys and configurations to enable the bot functionality. Users can customize the voice of their VTuber and route audio using VBAudio Cable. Kuebiko provides a unique way to interact with viewers through chat responses and captions in OBS.

Casual and friendly AI guide for easy learning

Casual, inquiry-driven expert in Vocode, fluent in English.

Guiding you through the depths of deep learning with accuracy and respect.

专业生成计算机科学演讲稿的助手

输出简单任务描述，帮助你改进和完善提示词

A helpful coach for speech writing, offering constructive advice and support

I generate perfect speeches.

Expert Speech-Language Pathologist offering tailored medical consultations.

Create speech transcript parodies. Copyright (C) 2023, Sourceduty - All Rights Reserved.

Writes interpreting speeches based on specific information.

Crafts speeches from PowerPoint slides and reference materials, adding depth and context.

Your partner in crafting perfect wedding speeches. Let me be your guide to writing impactful, memorable speeches for unforgettable moments.

Eiken Test Prep with Focus on Expression

Your best man speechwriting expert

Phonetics and reading coach with interactive voice capabilities, tailored for adult beginners.

Your very own speech therapy assistant. Completely private and confidential.

assists speech-language therapy with EBM

Analyzes leaders' actions and speech for insights.

Your Feline Language Specialist for translating human speech to cat sounds.

Yoda's iconic speech and wisdom, modernized.